From evorster at gmail.com Mon Sep 1 02:54:41 2008 From: evorster at gmail.com (evert vorster) Date: Mon, 1 Sep 2008 00:54:41 +0000 Subject: [gst-devel] configure script fails to complete. Message-ID: <27f41a930808311754u142a4ed0nf5f57e4ff5c09f79@mail.gmail.com> Hi there, gstreamer developers. Before I get 1000's of "please report bugs in the bug tracker", I do not believe this is a bug in gstreamer. I get a very similar error on my system on the configure script of swig. I humbly request a bit of help from experts, and no one stares at the configure scripts of gstreamer and gst-plugins-base more than gstreamer developers, and might have seen this problem before, and might spread some illumination on what I am doing wrong on my system. I come here as a last resort, as I have already been struggling with this for about a month! None of the other users of Sorcerer Linux have the same issue. On to the issue: On running ./configure in the gstreamer and gst-plugins-base source directories, it fails with these errors: gstreamer: ----------snip --------------------------------------- checking for GLIB... yes ./configure: line 32020: syntax error near unexpected token `(' ./configure: line 32020: ` for ac_var in `(set) 2>&1 | sed -n 's/^\([a-zA-Z_][a-zA-Z0-9_]*\)=.*/\1/p'`; do' gst-plugins-base: -----------snip--------------------------------- checking for GLIB... yes ./configure: line 38202: syntax error near unexpected token `(' ./configure: line 38202: ` for ac_var in `(set) 2>&1 | sed -n 's/^\([a-zA-Z_][a-zA-Z0-9_]*\)=.*/\1/p'`; do' The version of gcc used is: 4.3.1 Bash : 3.2.039 ncurses: 5.6 linux: 2.6.26 CFLAGS/CXXFLAGS for all software installed is -march=core2 -O1 -pipe, LDFLAGS = "-Wl,-O1" What is most annoying is that I managed to get a clean compile on 23 Aug, but not since. I will be more than happy to provide compile logs/settings. Thank you for any insight. -Evert Vorster- -- You are the gardener in your own life. Be careful what you sow... From nicolas.m.zhang at gmail.com Mon Sep 1 03:51:50 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Mon, 1 Sep 2008 09:51:50 +0800 Subject: [gst-devel] Media Player development Questions In-Reply-To: <6438d8660808281108j2fa69eddtac006065927ecba2@mail.gmail.com> References: <6438d8660808281108j2fa69eddtac006065927ecba2@mail.gmail.com> Message-ID: Hi, Raj: To your second question, my way is: I encapsulated the gstreamer stuffs into a class(derived from GObject) and the GTK+ main program will construct this object and also the interface and accept user input. The gstreamer object will emit signals whenever there is something need to be shown in the interface. Eric Zhang 2008/8/29 Raj Swaminathan > Hi, > > I am new to Glib programming and gstreamer. I am trying to build a basic > media player application. So far i have gstreamer pipelines up and rendering > various media formats. > > My questions are: > > 1) Im trying to reference the gstalsamixer plugin for volume control and > muting. I understand its an interface and im trying to find out whats the > best way to access the interface methods like > gst_alsa_alsa_mixer_get_volume etc ... > Could somebody please provide a code snippet that can show this ? > > 2) When designing the media player, whats the best way to accept user > input. Do i spool a separate thread for rendering media while the main > thread waits for user input or is there a better way within gstreamer to do > this? > > > Thanks for your time. > > regards, > raj > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nicolas.m.zhang at gmail.com Mon Sep 1 03:56:58 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Mon, 1 Sep 2008 09:56:58 +0800 Subject: [gst-devel] question about gstreamer internal gstmessage and linux message queue In-Reply-To: <31093284.115831219979033275.JavaMail.coremail@bj163app56.163.com> References: <31093284.115831219979033275.JavaMail.coremail@bj163app56.163.com> Message-ID: Hi, Volter: I have no idea what your controllersys application is so I think maybe you can write a gstreamer plugin which communicates with your controllersys application? Eric Zhang 2008/8/29 Volter Yen > Hi all, > I met a problem when I using gstreamer framework to develope my media > player. I used the gstreamer application as the backend of my player system, > and there is controllersys application in my system too, they can > communicate with each other using the message queue provided by linux > os, but the question is that *the gstreamer backend could not receive and > respond to the message sent from the controllersys when the pipeline was > doing a state change*,so if there is some problem occurs during the > pipeline changing, eq, the pipeline if waiting for a live source to push > datum to downstream element, the gstreamer backend will be hung up... > > Any ideas? thank you! > > Best Regards! > > Volter > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nicolas.m.zhang at gmail.com Mon Sep 1 03:59:53 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Mon, 1 Sep 2008 09:59:53 +0800 Subject: [gst-devel] GStreamer General State and BIN Issue In-Reply-To: <8c192ddd0808290339q4f345fddl9d280c63aa8354b3@mail.gmail.com> References: <8c192ddd0808290339q4f345fddl9d280c63aa8354b3@mail.gmail.com> Message-ID: Hi, Manish: I think you should pause the pipeline and re-play it after you inserted audiotestsrc into pipeline again. Eric 2008/8/29 Manish Rana > Hi All, > > > I am facing one issue in Gstreamer, I use following sequance. > > 1. Pipaline is created. > 2. Followink elements are there in Pipeline: > a. audiotestsrc > b. caps ! queue ! amrnbenc ! rtpamrpay ( One BIN - Called SRC BIN > ) > b. RTPBIN > c. udpsink > > Now if I unlink and remove audiotestsrc from my pipeline, that is ok. > But if i again add this element to my pipeline there is no data on the > audiotestsrc (checked using pad PROBE data.) > > Please let me know where i am doing it wrong. > As i think logically there is something messed up :( > > > Thanks a lot.. > Manish > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nicolas.m.zhang at gmail.com Mon Sep 1 04:16:43 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Mon, 1 Sep 2008 10:16:43 +0800 Subject: [gst-devel] Video processing In-Reply-To: References: <399889.18452.qm@web27203.mail.ukl.yahoo.com> Message-ID: Hi, Bruno: If you wanna just to drop the frames in your probe callback function, that's very simple -- just return FALSE in your probe function. Refer to the API manual of `gst_pad_add_data_probe' function for more details. About the `leak' property of queue, Aurelien Grimaud already gives a good explanation. Eric Zhang 2008/8/29 Bruno > Okay thanks a lot for these precious informations. > > With a pad probe I was able to draw on the xvimagesink buffer, so that part > is okay now. > > But I still can't find how to drop the frame sent by camera during the > image processing, > I found that the queue element has a property called "leaky" which, seeing > the description of the element : > > "Where the queue leaks, if at all. > > Default value: Not Leaky" > > It should be what I'm looking for. If I change its value to 1, the queue > element should drop the coming frames until the processing of the next > element is not finished, isn't it ? > > I tried to change the value and didn't see any difference, so maybe I'm > wrong. > > Thanks again. > > Bruno > > > > > > > 2008/8/28, Eric Zhang : > >> Hi, Bruno: >> >> I think that what you mentioned is not a gstreamer related topic. What >> you are worrying about is the performance of image processing and there is a >> lot of ways to improve it, such as re-consider the algorithm of image >> processing, try to parallel the processing and so on. >> >> The `queue' element in gstreamer is just a container of buffers. It >> just receive the buffer and push it out until `limitation' >> reaches(max-bytes, max-time, max-buffers). We usually use this element to >> cache the buffers and implement `buffering' features which is useful in a >> media player. >> >> Eric Zhang >> >> >> 2008/8/27 Bruno >> >>> Ok thanks a lot Eric, I didn't know the buffer can be modified on the fly >>> this way. >>> >>> But what I don't really understand is how exactly the pipeline works. If >>> I do that way, will the pipeline take one frame from the camera, call the >>> buffer probe related function, doing the image processing calculation, >>> modify the buffer, then send it to the screen ? Because that processing >>> takes a long time, and lot of frame are coming from the cam during the >>> calculation. I'd like them to be dropped, then when the calculation from the >>> first frame is finished, the image processing callback has to take the next >>> coming frame. (or the one just before, but not the second one). >>> >>> Actually it seems that all frame are showed, like it does the calculation >>> for the first, then keep the next coming in memory, and when the first is >>> processed, it comes to the second. (or at least it seems like this when I >>> start the prog). Maybe it's due to the "queue" element ? (I took this from >>> the maemo-camera example, and I don't get what the queue element is for) >>> >>> Thanks again for explanations ! >>> >>> Kind regards, >>> Bruno >>> >>> >>> 2008/8/27, Eric Zhang : >>> >>>> Hi, gstreamer-devel: >>>> >>>> Hello, Bruno, actually I have not many experiences on this topic and >>>> I was just going to give you some clews. OK, AFAIK, you can add a buffer >>>> probe before the xvimagesink and check out the GstBuffer which is going to >>>> flow into it and add your rectangle on this GstBuffer, that's it. You don't >>>> need to draw the buffer yourself because xvimagesink will do it for you(just >>>> give your drawing area's window id to it, which you have achieved before). >>>> >>>> You can refer to Chapter 18 of gstreamer application develop manual, >>>> there is an example which inverts the image by adding a buffer probe and >>>> modify the buffer on the fly. Maybe this can help you a little. >>>> >>>> Eric Zhang >>>> >>>> 2008/8/26 Bruno >>>> >>>>> Could you be a little more precise please ? >>>>> How do I modify the buffer ? How do I draw it on the screen after ? >>>>> should I still declare my screen widget from appdata struct as a drawing >>>>> area ? >>>>> >>>>> Thanks for help >>>>> >>>>> >>>>> 2008/8/26, Eric Zhang : >>>>>> >>>>>> Hi, gstreamer-devel: >>>>>> >>>>>> Oh, I think you can use a buffer/data probe to achieve this. >>>>>> Modify the buffer, add a rectangle on every frame. Or, modify the >>>>>> xvimagesink. :) >>>>>> >>>>>> Eric Zhang >>>>>> >>>>>> 2008/8/26 Bruno >>>>>> >>>>>>> Yep that is what I was doing before. Do you think I can draw >>>>>>> rectangles over the cam image when using xvimagesink ? >>>>>>> >>>>>>> >>>>>>> 2008/8/26, Eric Zhang : >>>>>>> >>>>>>>> Hi, gstreamer-devel: >>>>>>>> >>>>>>>> If you only want to `ximagesink' or `xvimagesink' draws images >>>>>>>> in >>>>>>>> your GtkDrawingArea, there is a very simple way to achieve this: >>>>>>>> >>>>>>>> Just connect the `expose-event' signal of GtkDrawingArea and >>>>>>>> pass >>>>>>>> the window ID to the sink element: >>>>>>>> >>>>>>>> // Drawing on our drawing area >>>>>>>> g_signal_connect(G_OBJECT(area), "expose-event", >>>>>>>> G_CALLBACK(expose_cb), >>>>>>>> NULL); >>>>>>>> >>>>>>>> /* Callback to be called when the drawing area is exposed */ >>>>>>>> >>>>>>>> static gboolean expose_cb(GtkWidget * widget, GdkEventExpose * >>>>>>>> event, >>>>>>>> gpointer data) >>>>>>>> { >>>>>>>> >>>>>>>> // `play-videosink' is your video sink element >>>>>>>> gst_x_overlay_set_xwindow_id(GST_X_OVERLAY(play->videosink), >>>>>>>> >>>>>>>> GDK_WINDOW_XWINDOW(widget->window)); >>>>>>>> return FALSE; >>>>>>>> } >>>>>>>> >>>>>>>> >>>>>>>> That's it. If you want to draw the image yourself while not >>>>>>>> using >>>>>>>> `xvimagesink' or `ximagesink', then I think this is a Gtk+ problem, >>>>>>>> not >>>>>>>> a gstreamer issue. >>>>>>>> >>>>>>>> >>>>>>>> Eric Zhang >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Bruno wrote: >>>>>>>> > Hello all, >>>>>>>> > >>>>>>>> > I still have some questions about gstreamer. >>>>>>>> > >>>>>>>> > Actually I'd like to change the way my program work, in order to >>>>>>>> > display a frame from the camera only once the image processing has >>>>>>>> > been done, and with a rectangle over the face of the person. >>>>>>>> > >>>>>>>> > So I changed my pipeline (removed the screen_sink element), and >>>>>>>> I'd >>>>>>>> > like to send a buffer from my image processing function to the gtk >>>>>>>> > drawing area where the camera image was displayed before. I tried >>>>>>>> to >>>>>>>> > do it with gtk drawing area (and with gtk image too with no >>>>>>>> sucess), >>>>>>>> > but I can't find the way to change the image contained in the >>>>>>>> drawing >>>>>>>> > area. >>>>>>>> > >>>>>>>> > Here is my code : >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > ///// IMAGE PROCESSING CALLBACK >>>>>>>> > >>>>>>>> > /* Callback to be called when data goes through the pad */ >>>>>>>> > static gboolean process_frame(GstElement *video_sink, >>>>>>>> > GstBuffer *buffer, GstPad *pad, AppData *appdata) >>>>>>>> > { >>>>>>>> > int x, y; >>>>>>>> > // getting the pointer to camera buffer >>>>>>>> > unsigned char *data_photo = (unsigned char *) >>>>>>>> > GST_BUFFER_DATA(buffer); >>>>>>>> > >>>>>>>> > >>>>>>>> > // REMOVED PART WHERE THE COORDINATES OF THE POSITION OF THE FACE >>>>>>>> IS >>>>>>>> > CALCULATED // >>>>>>>> > >>>>>>>> > >>>>>>>> > // THIS PART IS WHAT I TRIED, BUT I HAVE A SEGMENTATION FAULT WHEN >>>>>>>> > CREATING PIXBUF // >>>>>>>> > GdkPixbuf *newscreen; >>>>>>>> > //newscreen = gdk_pixbuf_new_from_data(data_photo, >>>>>>>> > //GDK_COLORSPACE_RGB, /* RGB-colorspace */ >>>>>>>> > //FALSE, /* No alpha-channel */ >>>>>>>> > //8, /* Bits per RGB-component */ >>>>>>>> > //IMAGE_WIDTH, IMAGE_HEIGHT, /* Dimensions */ >>>>>>>> > //3*IMAGE_WIDTH, /* Number of bytes between >>>>>>>> lines >>>>>>>> > (ie stride) */ >>>>>>>> > //NULL, NULL); /* Callbacks */ >>>>>>>> > >>>>>>>> > >>>>>>>> > gdk_draw_pixmap(GDK_DRAWABLE(appdata->screen), >>>>>>>> > appdata->screen->style->black_gc, GDK_DRAWABLE(newscreen), 0, 0, >>>>>>>> 0, 0, >>>>>>>> > -1, -1); >>>>>>>> > >>>>>>>> > return TRUE; >>>>>>>> > } >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > /////// PIPELINE >>>>>>>> > >>>>>>>> > >>>>>>>> > /* Initialize the the Gstreamer pipeline. Below is a diagram >>>>>>>> > * of the pipeline that will be created: >>>>>>>> > * >>>>>>>> > * |Camera| |CSP | |Screen| |Screen| |Image | >>>>>>>> > * |src |->|Filter|->|queue |->|sink |-> >>>>>>>> |processing|-> Display >>>>>>>> > */ >>>>>>>> > static gboolean initialize_pipeline(AppData *appdata, >>>>>>>> > int *argc, char ***argv) >>>>>>>> > { >>>>>>>> > GstElement *pipeline, *camera_src, *screen_sink; >>>>>>>> > GstElement *screen_queue; >>>>>>>> > GstElement *csp_filter; >>>>>>>> > GstCaps *caps; >>>>>>>> > GstBus *bus; >>>>>>>> > GstPad *sinkpad; >>>>>>>> > >>>>>>>> > /* Initialize Gstreamer */ >>>>>>>> > gst_init(argc, argv); >>>>>>>> > >>>>>>>> > /* Create pipeline and attach a callback to it's >>>>>>>> > * message bus */ >>>>>>>> > pipeline = gst_pipeline_new("test-camera"); >>>>>>>> > >>>>>>>> > bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline)); >>>>>>>> > gst_bus_add_watch(bus, (GstBusFunc)bus_callback, appdata); >>>>>>>> > gst_object_unref(GST_OBJECT(bus)); >>>>>>>> > >>>>>>>> > /* Save pipeline to the AppData structure */ >>>>>>>> > appdata->pipeline = pipeline; >>>>>>>> > >>>>>>>> > /* Create elements */ >>>>>>>> > /* Camera video stream comes from a Video4Linux driver */ >>>>>>>> > camera_src = gst_element_factory_make(VIDEO_SRC, >>>>>>>> "camera_src"); >>>>>>>> > /* Colorspace filter is needed to make sure that sinks >>>>>>>> understands >>>>>>>> > * the stream coming from the camera */ >>>>>>>> > csp_filter = gst_element_factory_make("ffmpegcolorspace", >>>>>>>> > "csp_filter"); >>>>>>>> > /* Queue creates new thread for the stream */ >>>>>>>> > screen_queue = gst_element_factory_make("queue", >>>>>>>> "screen_queue"); >>>>>>>> > /* Sink that shows the image on screen. Xephyr doesn't support >>>>>>>> XVideo >>>>>>>> > * extension, so it needs to use ximagesink, but the device >>>>>>>> uses >>>>>>>> > * xvimagesink */ >>>>>>>> > //screen_sink = gst_element_factory_make(VIDEO_SINK, >>>>>>>> "screen_sink"); >>>>>>>> > >>>>>>>> > sinkpad = gst_element_get_static_pad(screen_queue,"sink"); >>>>>>>> > gst_pad_add_buffer_probe(sinkpad,G_CALLBACK(process_frame), >>>>>>>> appdata); >>>>>>>> > >>>>>>>> > >>>>>>>> > /* Check that elements are correctly initialized */ >>>>>>>> > if(!(pipeline && camera_src /*&& screen_sink*/ && csp_filter >>>>>>>> && >>>>>>>> > screen_queue)) >>>>>>>> > { >>>>>>>> > g_critical("Couldn't create pipeline elements"); >>>>>>>> > return FALSE; >>>>>>>> > } >>>>>>>> > >>>>>>>> > >>>>>>>> > /* Add elements to the pipeline. This has to be done prior to >>>>>>>> > * linking them */ >>>>>>>> > gst_bin_add_many(GST_BIN(pipeline), camera_src, csp_filter, >>>>>>>> > screen_queue, /*screen_sink,*/ NULL); >>>>>>>> > >>>>>>>> > /* Specify what kind of video is wanted from the camera */ >>>>>>>> > caps = gst_caps_new_simple("video/x-raw-rgb", >>>>>>>> > "width", G_TYPE_INT, IMAGE_WIDTH, >>>>>>>> > "height", G_TYPE_INT, IMAGE_HEIGHT, >>>>>>>> > "framerate", GST_TYPE_FRACTION, FRAMERATE, 1, >>>>>>>> > NULL); >>>>>>>> > >>>>>>>> > >>>>>>>> > /* Link the camera source and colorspace filter using >>>>>>>> capabilities >>>>>>>> > * specified */ >>>>>>>> > if(!gst_element_link_filtered(camera_src, csp_filter, caps)) >>>>>>>> > { >>>>>>>> > return FALSE; >>>>>>>> > } >>>>>>>> > gst_caps_unref(caps); >>>>>>>> > >>>>>>>> > /* Connect Colorspace Filter -> Screen Queue -> Screen Sink >>>>>>>> > * This finalizes the initialization of the screen-part of the >>>>>>>> > pipeline */ >>>>>>>> > if(!gst_element_link_many(csp_filter, screen_queue, >>>>>>>> /*screen_sink, >>>>>>>> > */NULL)) >>>>>>>> > { >>>>>>>> > return FALSE; >>>>>>>> > } >>>>>>>> > >>>>>>>> > gst_element_set_state(pipeline, GST_STATE_PAUSED); >>>>>>>> > >>>>>>>> > return TRUE; >>>>>>>> > } >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > /////// MAIN FUNCTION >>>>>>>> > >>>>>>>> > >>>>>>>> > int main(int argc, char **argv) >>>>>>>> > { >>>>>>>> > // variables for face detection >>>>>>>> > // main structure for vjdetect >>>>>>>> > >>>>>>>> > pdata = (mainstruct*) calloc(1, sizeof(mainstruct)); >>>>>>>> > // Allocate memory for array of face detections returned by >>>>>>>> > facedetector (VjDetect). >>>>>>>> > pdata->pFaceDetections = (FLY_Rect >>>>>>>> > *)calloc(MAX_NUMBER_OF_FACE_DETECTIONS, sizeof(FLY_Rect)); >>>>>>>> > init(pdata); >>>>>>>> > >>>>>>>> > AppData appdata; >>>>>>>> > appdata.expression = 0; >>>>>>>> > GtkWidget *hbox, *vbox_button, *vbox, *button1, *button2; >>>>>>>> > >>>>>>>> > >>>>>>>> > /* Initialize and create the GUI */ >>>>>>>> > >>>>>>>> > example_gui_initialize( >>>>>>>> > &appdata.program, &appdata.window, >>>>>>>> > &argc, &argv, "Expression Detector"); >>>>>>>> > >>>>>>>> > vbox = gtk_vbox_new(FALSE, 0); >>>>>>>> > hbox = gtk_hbox_new(FALSE, 0); >>>>>>>> > vbox_button = gtk_vbox_new(FALSE, 0); >>>>>>>> > >>>>>>>> > gtk_box_pack_start(GTK_BOX(hbox), vbox, FALSE, FALSE, 0); >>>>>>>> > gtk_box_pack_start(GTK_BOX(hbox), vbox_button, FALSE, FALSE, >>>>>>>> 0); >>>>>>>> > >>>>>>>> > appdata.screen = gtk_drawing_area_new(); >>>>>>>> > gtk_widget_set_size_request(appdata.screen, 500, 380); >>>>>>>> > gtk_box_pack_start(GTK_BOX(vbox), appdata.screen, FALSE, >>>>>>>> FALSE, 0); >>>>>>>> > >>>>>>>> > button1 = gtk_toggle_button_new_with_label("Run/Stop"); >>>>>>>> > gtk_widget_set_size_request(button1, 170, 75); >>>>>>>> > gtk_box_pack_start(GTK_BOX(vbox_button), button1, FALSE, >>>>>>>> FALSE, 0); >>>>>>>> > >>>>>>>> > button2 = gtk_toggle_button_new_with_label("Expressions >>>>>>>> ON/OFF"); >>>>>>>> > gtk_widget_set_size_request(button2, 170, 75); >>>>>>>> > gtk_box_pack_start(GTK_BOX(vbox_button), button2, FALSE, >>>>>>>> FALSE, 0); >>>>>>>> > >>>>>>>> > >>>>>>>> > appdata.anger = >>>>>>>> gtk_image_new_from_file("./smileys/anger.jpg"); >>>>>>>> > gtk_widget_set_size_request(appdata.anger, 160, 180); >>>>>>>> > appdata.disgust = >>>>>>>> gtk_image_new_from_file("./smileys/disgust.jpg"); >>>>>>>> > gtk_widget_set_size_request(appdata.disgust, 160, 180); >>>>>>>> > appdata.fear = gtk_image_new_from_file("./smileys/fear.jpg"); >>>>>>>> > gtk_widget_set_size_request(appdata.fear, 160, 180); >>>>>>>> > appdata.happy = >>>>>>>> gtk_image_new_from_file("./smileys/happy.jpg"); >>>>>>>> > gtk_widget_set_size_request(appdata.happy, 160, 180); >>>>>>>> > appdata.neutral = >>>>>>>> gtk_image_new_from_file("./smileys/neutral.jpg"); >>>>>>>> > gtk_widget_set_size_request(appdata.neutral, 160, 180); >>>>>>>> > appdata.sad = gtk_image_new_from_file("./smileys/sad.jpg"); >>>>>>>> > gtk_widget_set_size_request(appdata.sad, 160, 180); >>>>>>>> > appdata.surprise = >>>>>>>> gtk_image_new_from_file("./smileys/surprise.jpg"); >>>>>>>> > gtk_widget_set_size_request(appdata.surprise, 160, 180); >>>>>>>> > appdata.unknown = >>>>>>>> gtk_image_new_from_file("./smileys/unknown.jpg"); >>>>>>>> > gtk_widget_set_size_request(appdata.unknown, 160, 180); >>>>>>>> > >>>>>>>> > appdata.smiley = >>>>>>>> gtk_image_new_from_file("./smileys/unknown.jpg"); >>>>>>>> > gtk_widget_set_size_request(appdata.smiley, 160, 180); >>>>>>>> > gtk_box_pack_start(GTK_BOX(vbox_button), appdata.smiley, >>>>>>>> FALSE, >>>>>>>> > FALSE, 0); >>>>>>>> > >>>>>>>> > g_signal_connect(G_OBJECT(button1), "clicked", >>>>>>>> > G_CALLBACK(button1_pressed), &appdata); >>>>>>>> > >>>>>>>> > g_signal_connect(G_OBJECT(button2), "clicked", >>>>>>>> > G_CALLBACK(button2_pressed), &appdata); >>>>>>>> > >>>>>>>> > >>>>>>>> > gtk_container_add(GTK_CONTAINER(appdata.window), hbox); >>>>>>>> > >>>>>>>> > /* Initialize the GTK pipeline */ >>>>>>>> > if(!initialize_pipeline(&appdata, &argc, &argv)) >>>>>>>> > { >>>>>>>> > hildon_banner_show_information( >>>>>>>> > GTK_WIDGET(appdata.window), >>>>>>>> > "gtk-dialog-error", >>>>>>>> > "Failed to initialize pipeline"); >>>>>>>> > } >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > g_signal_connect(G_OBJECT(appdata.window), "destroy", >>>>>>>> > G_CALLBACK(destroy_pipeline), &appdata); >>>>>>>> > >>>>>>>> > >>>>>>>> > /* Begin the main application */ >>>>>>>> > example_gui_run(appdata.program, appdata.window); >>>>>>>> > >>>>>>>> > /* Free the gstreamer resources. Elements added >>>>>>>> > * to the pipeline will be freed automatically */ >>>>>>>> > >>>>>>>> > return 0; >>>>>>>> > } >>>>>>>> > >>>>>>>> > >>>>>>>> > What I'd like to do is to modify the data_photo buffer to draw a >>>>>>>> > rectangle in it (in the process_frame function), and draw the >>>>>>>> content >>>>>>>> > in the appdata.screen GtkWidget. (by the way screen is declared as >>>>>>>> a >>>>>>>> > GtkWidget * in the appdata structure). >>>>>>>> > >>>>>>>> > Thanks in advance for your help ! >>>>>>>> > Bruno >>>>>>>> > >>>>>>>> >>>>>>>> > >>>>>>>> ------------------------------------------------------------------------ >>>>>>>> >>>>>>>> > >>>>>>>> > >>>>>>>> ------------------------------------------------------------------------- >>>>>>>> > This SF.Net email is sponsored by the Moblin Your Move Developer's >>>>>>>> challenge >>>>>>>> > Build the coolest Linux based applications with Moblin SDK & win >>>>>>>> great prizes >>>>>>>> > Grand prize is a trip for two to an Open Source event anywhere in >>>>>>>> the world >>>>>>>> > http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>>>>>> > >>>>>>>> ------------------------------------------------------------------------ >>>>>>>> > >>>>>>>> > _______________________________________________ >>>>>>>> > gstreamer-devel mailing list >>>>>>>> > gstreamer-devel at lists.sourceforge.net >>>>>>>> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>> > >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> ------------------------------------------------------------------------- >>>>>>>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>>>>>>> challenge >>>>>>>> Build the coolest Linux based applications with Moblin SDK & win >>>>>>>> great prizes >>>>>>>> Grand prize is a trip for two to an Open Source event anywhere in >>>>>>>> the world >>>>>>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>>>>>> _______________________________________________ >>>>>>>> gstreamer-devel mailing list >>>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------- >>>>>>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>>>>>> challenge >>>>>>> Build the coolest Linux based applications with Moblin SDK & win >>>>>>> great prizes >>>>>>> Grand prize is a trip for two to an Open Source event anywhere in the >>>>>>> world >>>>>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>>>>> _______________________________________________ >>>>>>> gstreamer-devel mailing list >>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------- >>>>>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>>>>> challenge >>>>>> Build the coolest Linux based applications with Moblin SDK & win great >>>>>> prizes >>>>>> Grand prize is a trip for two to an Open Source event anywhere in the >>>>>> world >>>>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>>>> _______________________________________________ >>>>>> gstreamer-devel mailing list >>>>>> gstreamer-devel at lists.sourceforge.net >>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>> >>>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------- >>>>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>>>> challenge >>>>> Build the coolest Linux based applications with Moblin SDK & win great >>>>> prizes >>>>> Grand prize is a trip for two to an Open Source event anywhere in the >>>>> world >>>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> gstreamer-devel at lists.sourceforge.net >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>> >>>> >>>> ------------------------------------------------------------------------- >>>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>>> challenge >>>> Build the coolest Linux based applications with Moblin SDK & win great >>>> prizes >>>> Grand prize is a trip for two to an Open Source event anywhere in the >>>> world >>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>> >>> ------------------------------------------------------------------------- >>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>> challenge >>> Build the coolest Linux based applications with Moblin SDK & win great >>> prizes >>> Grand prize is a trip for two to an Open Source event anywhere in the >>> world >>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> challenge >> Build the coolest Linux based applications with Moblin SDK & win great >> prizes >> Grand prize is a trip for two to an Open Source event anywhere in the >> world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ds at schleef.org Mon Sep 1 06:17:49 2008 From: ds at schleef.org (David Schleef) Date: Sun, 31 Aug 2008 21:17:49 -0700 Subject: [gst-devel] configure script fails to complete. In-Reply-To: <27f41a930808311754u142a4ed0nf5f57e4ff5c09f79@mail.gmail.com> References: <27f41a930808311754u142a4ed0nf5f57e4ff5c09f79@mail.gmail.com> Message-ID: <20080901041749.GA12888@bigkitten.com> On Mon, Sep 01, 2008 at 12:54:41AM +0000, evert vorster wrote: > On to the issue: > > On running ./configure in the gstreamer and gst-plugins-base source > directories, it fails with these errors: > > gstreamer: > ----------snip --------------------------------------- > checking for GLIB... yes > ./configure: line 32020: syntax error near unexpected token `(' > ./configure: line 32020: ` for ac_var in `(set) 2>&1 | sed -n > 's/^\([a-zA-Z_][a-zA-Z0-9_]*\)=.*/\1/p'`; do' That line comes from a very low-level autoconf macro in autoconf.m4. It is technically valid shell, although cryptic. That is, it is valid in *my* configure script. You could have a different version of autoconf that has broken the generated shell script. However, on the surface it looks more like your shell interpreter (i.e., /bin/sh, which may not be bash) is buggy. dave... From evorster at gmail.com Mon Sep 1 06:26:46 2008 From: evorster at gmail.com (evert vorster) Date: Mon, 1 Sep 2008 04:26:46 +0000 Subject: [gst-devel] configure script fails to complete. In-Reply-To: <20080901041749.GA12888@bigkitten.com> References: <27f41a930808311754u142a4ed0nf5f57e4ff5c09f79@mail.gmail.com> <20080901041749.GA12888@bigkitten.com> Message-ID: <27f41a930808312126w358b06adldfd15f412109ecc7@mail.gmail.com> On Mon, Sep 1, 2008 at 4:17 AM, David Schleef wrote: > On Mon, Sep 01, 2008 at 12:54:41AM +0000, evert vorster wrote: >> On to the issue: >> >> On running ./configure in the gstreamer and gst-plugins-base source >> directories, it fails with these errors: >> >> gstreamer: >> ----------snip --------------------------------------- >> checking for GLIB... yes >> ./configure: line 32020: syntax error near unexpected token `(' >> ./configure: line 32020: ` for ac_var in `(set) 2>&1 | sed -n >> 's/^\([a-zA-Z_][a-zA-Z0-9_]*\)=.*/\1/p'`; do' > > That line comes from a very low-level autoconf macro in autoconf.m4. > It is technically valid shell, although cryptic. That is, it is > valid in *my* configure script. You could have a different version > of autoconf that has broken the generated shell script. > > However, on the surface it looks more like your shell interpreter > (i.e., /bin/sh, which may not be bash) is buggy. Thank you for the reply and added insight, Dave. I will go and take a second look at autoconf and bash, and see where it all went funny. -Evert- From huan.zheng at intel.com Mon Sep 1 06:11:22 2008 From: huan.zheng at intel.com (Zheng, Huan) Date: Mon, 1 Sep 2008 12:11:22 +0800 Subject: [gst-devel] A question about normal_seek and index_seek Message-ID: Hi, dear developers I'm a little confused by normal_seek and index_seek when reading the code of mad plugin. What are the differences between them? Thanks for your response! Best Regards, Zheng, Huan(ZBT) OTC/SSD/SSG Intel Aisa-Pacific Research & Developement Ltd Tel: 021-6116 6435 Inet: 8821 6435 Cub: 3W035 -------------- next part -------------- An HTML attachment was scrubbed... URL: From manish.rana at gmail.com Mon Sep 1 07:00:25 2008 From: manish.rana at gmail.com (Manish Rana) Date: Mon, 1 Sep 2008 10:30:25 +0530 Subject: [gst-devel] GStreamer General State and BIN Issue In-Reply-To: References: <8c192ddd0808290339q4f345fddl9d280c63aa8354b3@mail.gmail.com> Message-ID: <8c192ddd0808312200l6920d460q67411ff787252759@mail.gmail.com> Dear Eric, Thanks a lot for your suggestion... I can't put the pipeline in the Paused State as there are other bins also in the pipeline and they need to be running.... Also what effect will it cause on the live source in my pipeline..as it has udpsink as well........... BR Manish On Mon, Sep 1, 2008 at 7:29 AM, Eric Zhang wrote: > Hi, Manish: > > I think you should pause the pipeline and re-play it after you inserted > audiotestsrc into pipeline again. > > Eric > > 2008/8/29 Manish Rana > >> Hi All, >> >> >> I am facing one issue in Gstreamer, I use following sequance. >> >> 1. Pipaline is created. >> 2. Followink elements are there in Pipeline: >> a. audiotestsrc >> b. caps ! queue ! amrnbenc ! rtpamrpay ( One BIN - Called SRC >> BIN ) >> b. RTPBIN >> c. udpsink >> >> Now if I unlink and remove audiotestsrc from my pipeline, that is ok. >> But if i again add this element to my pipeline there is no data on the >> audiotestsrc (checked using pad PROBE data.) >> >> Please let me know where i am doing it wrong. >> As i think logically there is something messed up :( >> >> >> Thanks a lot.. >> Manish >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> challenge >> Build the coolest Linux based applications with Moblin SDK & win great >> prizes >> Grand prize is a trip for two to an Open Source event anywhere in the >> world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ajaygautam1981 at gmail.com Mon Sep 1 07:13:19 2008 From: ajaygautam1981 at gmail.com (AJAY GAUTAM) Date: Mon, 1 Sep 2008 10:43:19 +0530 Subject: [gst-devel] Media Player development Questions In-Reply-To: <6438d8660808281108j2fa69eddtac006065927ecba2@mail.gmail.com> References: <6438d8660808281108j2fa69eddtac006065927ecba2@mail.gmail.com> Message-ID: <9e93fd980808312213haeae69fna46ac6c409fa8a60@mail.gmail.com> Hi Raj, For first question i think this link will help you: http://cmus.sourcearchive.com/documentation/2.0.4-1/mixer__alsa_8c-source.html On Thu, Aug 28, 2008 at 11:38 PM, Raj Swaminathan wrote: > Hi, > > I am new to Glib programming and gstreamer. I am trying to build a basic > media player application. So far i have gstreamer pipelines up and rendering > various media formats. > > My questions are: > > 1) Im trying to reference the gstalsamixer plugin for volume control and > muting. I understand its an interface and im trying to find out whats the > best way to access the interface methods like > gst_alsa_alsa_mixer_get_volume etc ... > Could somebody please provide a code snippet that can show this ? > > 2) When designing the media player, whats the best way to accept user > input. Do i spool a separate thread for rendering media while the main > thread waits for user input or is there a better way within gstreamer to do > this? > > > Thanks for your time. > > regards, > raj > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Thanx & Regards Ajay Gautam +91-9741083000 -------------- next part -------------- An HTML attachment was scrubbed... URL: From sachinpandhare at gmail.com Mon Sep 1 07:26:16 2008 From: sachinpandhare at gmail.com (Sachin Pandhare) Date: Mon, 1 Sep 2008 10:56:16 +0530 Subject: [gst-devel] amrnb parser In-Reply-To: <48B7E716.8060705@hora-obscura.de> References: <72cf309c0808282332r2351355eo12ba8c35d859688f@mail.gmail.com> <48B7E716.8060705@hora-obscura.de> Message-ID: <72cf309c0808312226s46c87db9h9a59e8365bafa127@mail.gmail.com> Thanks Stefan for giving reference to earlier post :) I think it will be a nice idea to have (external) library dependent code separated out from the parsing which, i think, is not (either free or non free) library dependent. Sachin On Fri, Aug 29, 2008 at 5:39 PM, Stefan Kost wrote: > hi, > > I sent an email earlier regading this (look for "restructuring amr plugins" > from > the 11th of Aug. 2008). It would solve you problem. Need to persuade the > others > still. > > Stefan > > Sachin Pandhare schrieb: > > Hi, > > i am using latest gstreamer plugins. But it fails to compile AMR plugin. > > i want to use only AMR NB parser from this plaugin. > > > > As per my understanding > > - 'amrnb' plugin has parser, decoder, encoder. > > - it has a dependency on external AMR codec library (libamr??) > > > > my query is : > > - is it possible to disable AMR encoder and decoder from this plugin? > > - if yes how? [configure --disable-amrnb will disable the whole plugin :( > ] > > - will it impact the other plugins/application? > > > > thanks, > > Sachin > > > > > > ------------------------------------------------------------------------ > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Mon Sep 1 10:05:43 2008 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Mon, 01 Sep 2008 09:05:43 +0100 Subject: [gst-devel] Media Player development Questions In-Reply-To: <6438d8660808281108j2fa69eddtac006065927ecba2@mail.gmail.com> References: <6438d8660808281108j2fa69eddtac006065927ecba2@mail.gmail.com> Message-ID: <1220256343.6287.6.camel@mini.centricular.net> On Thu, 2008-08-28 at 13:08 -0500, Raj Swaminathan wrote: Hi, > I am new to Glib programming and gstreamer. I am trying to build a > basic media player application. So far i have gstreamer pipelines up > and rendering various media formats. For a media player you'll probably want to use playbin (or even playbin2) from gst-plugins-base. There's no need to create your own pipelines. > 1) Im trying to reference the gstalsamixer plugin for volume control > and muting. I understand its an interface and im trying to find out > whats the best way to access the interface methods like > gst_alsa_alsa_mixer_get_volume etc ... > Could somebody please provide a code snippet that can show this ? For stream volume control you should use playbin's "volume" property. The GStreamer mixer interface (GstMixer) is to control hardware mixer levels, you usually don't want that in a playback application. It's also not very nice to use. If you don't use playbin, add a volume element to your pipeline (alternatively: audiosinks which support stream volumes will also have a "volume" property). > 2) When designing the media player, whats the best way to accept user > input. Do i spool a separate thread for rendering media while the main > thread waits for user input or is there a better way within gstreamer > to do this? GStreamer does all its playback in threads of its own anyway, so the main thread is yours. All you need to do is check the pipeline's/playbin's GstBus for messages (errors, tags, state changes) from time to time. Cheers -Tim From Martin.Lehmann at telekom.de Mon Sep 1 12:36:05 2008 From: Martin.Lehmann at telekom.de (Martin.Lehmann at telekom.de) Date: Mon, 1 Sep 2008 12:36:05 +0200 Subject: [gst-devel] connect to certain channels on alsa or jack In-Reply-To: <1220256343.6287.6.camel@mini.centricular.net> References: <6438d8660808281108j2fa69eddtac006065927ecba2@mail.gmail.com> <1220256343.6287.6.camel@mini.centricular.net> Message-ID: <547A7B079BC8EF4685873738492ADA430921F167@S4DE9JSAAMU.ost.t-com.de> Hey, i want to connect to certain channels in a 56channels setup (using either alsasink or jackaudiosink) in this case, when I have a 2channels-source. e.g. connect to the 42nd and 43rd channel. How does that work?! I read something about pad_connect, but didn't find some useful code-example. Does anybody use such a great setup and could help me please? Regards, Martin From sachinpandhare at gmail.com Mon Sep 1 14:41:15 2008 From: sachinpandhare at gmail.com (Sachin Pandhare) Date: Mon, 1 Sep 2008 18:11:15 +0530 Subject: [gst-devel] encode / decode chain didn't work Message-ID: <72cf309c0809010541m28186ah692486bab3337c80@mail.gmail.com> Hi, i was testing basic encode and decode chains. gst-launch videotestsrc ! ffmpegcolorspace ! theoraenc ! filesink location=/test_theoraenc.bin this created a file. then i tried gst-launch filesrc location=/test_theoraenc.bin ! theoradec ! ffmpegcolorspace ! xvimagesink but this gave me following error: -------------------------------------------------------------------------------------------------- ERROR: from element /pipeline0/theoradec0: Could not decode stream. Additional debug info: theoradec.c(1166): theora_handle_data_packet (): /pipeline0/theoradec0: no header sent yet -------------------------------------------------------------------------------------------------- was there any mistake in my commands? Thanks, Sachin -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Mon Sep 1 14:47:42 2008 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Mon, 1 Sep 2008 18:17:42 +0530 Subject: [gst-devel] encode / decode chain didn't work In-Reply-To: <72cf309c0809010541m28186ah692486bab3337c80@mail.gmail.com> References: <72cf309c0809010541m28186ah692486bab3337c80@mail.gmail.com> Message-ID: <785339900809010547g1ad0b6bbob2b69c61f7631bd7@mail.gmail.com> Hi , use following pipeline :- gst-launch videotestsrc num-buffers=100 ! ffmpegcolorspace ! theoraenc ! filesink location=/test_theoraenc.bin and let it play untill pipeline gets closed by itself , dont break it using CTL+ C . On Mon, Sep 1, 2008 at 6:11 PM, Sachin Pandhare wrote: > Hi, > i was testing basic encode and decode chains. > > gst-launch videotestsrc ! ffmpegcolorspace ! theoraenc ! filesink > location=/test_theoraenc.bin > > this created a file. then i tried > > gst-launch filesrc location=/test_theoraenc.bin ! theoradec ! > ffmpegcolorspace ! xvimagesink > > but this gave me following error: > > -------------------------------------------------------------------------------------------------- > ERROR: from element /pipeline0/theoradec0: Could not decode stream. > Additional debug info: > theoradec.c(1166): theora_handle_data_packet (): /pipeline0/theoradec0: > no header sent yet > > -------------------------------------------------------------------------------------------------- > > was there any mistake in my commands? > > Thanks, > Sachin > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Mon Sep 1 14:59:26 2008 From: wim.taymans at gmail.com (Wim Taymans) Date: Mon, 01 Sep 2008 14:59:26 +0200 Subject: [gst-devel] encode / decode chain didn't work In-Reply-To: <72cf309c0809010541m28186ah692486bab3337c80@mail.gmail.com> References: <72cf309c0809010541m28186ah692486bab3337c80@mail.gmail.com> Message-ID: <1220273966.6781.8.camel@metal> On Mon, 2008-09-01 at 18:11 +0530, Sachin Pandhare wrote: > Hi, > i was testing basic encode and decode chains. > > gst-launch videotestsrc ! ffmpegcolorspace ! theoraenc ! filesink > location=/test_theoraenc.bin > this created a file. then i tried > > gst-launch filesrc location=/test_theoraenc.bin ! theoradec ! > ffmpegcolorspace ! xvimagesink > > but this gave me following error: > -------------------------------------------------------------------------------------------------- > ERROR: from element /pipeline0/theoradec0: Could not decode stream. > Additional debug info: > theoradec.c(1166): theora_handle_data_packet > (): /pipeline0/theoradec0: > no header sent yet > -------------------------------------------------------------------------------------------------- > > was there any mistake in my commands? yes, theora is a format that should be encapsulated into a container such as ogg, like this: gst-launch videotestsrc ! ffmpegcolorspace ! theoraenc ! oggmux ! filesink location=/test_theoraenc.ogg and to play it back: gst-launch filesrc location=/test_theoraenc.ogg ! oggdemux ! theoradec ! ffmpegcolorspace ! xvimagesink The reason is that theora relies on the container format to feed it exactly one encoded frame, there are no fame markers in the raw theora bitstream otherwise. Wim > > > Thanks, > Sachin > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From lassi.vaatamoinen at tut.fi Mon Sep 1 10:24:31 2008 From: lassi.vaatamoinen at tut.fi (Lassi =?iso-8859-1?q?V=E4=E4t=E4m=F6inen?=) Date: Mon, 1 Sep 2008 11:24:31 +0300 Subject: [gst-devel] GstRtpbin and RTCP sender reports Message-ID: <200809011124.31770.lassi.vaatamoinen@tut.fi> Hi, I was wondering whether the Gstrtpbin is capable of sending RTCP SR packets? I managed to build a pipeline using the gstrtpbin and gst-launch, but only RR packets were sent from the sender. Another alternative would be to use Rtpbin, but I am not sure how to use this, so an example would be nice. Best regards, -Lassi -- Lassi V??t?m?inen M.Sc., Researcher Tampere University of Technology / Department of Communications Engineering Room TH209, tel. 03-3115 5124 From spamtrap at gmx.de Mon Sep 1 16:18:52 2008 From: spamtrap at gmx.de (Hendrik Schober) Date: Mon, 01 Sep 2008 16:18:52 +0200 Subject: [gst-devel] getting data format info Message-ID: <48BBF9CC.8040304@gmx.de> Hi, we are using (or is that abusing?) gstreamer in order to capture video data from an USB camera. For this we have setup a v4lsrc and a fake sink and grab the data in the handoff hook of the latter. So far, so good. However, despite several hours of tinkering we haven't found out how to determine what format (width, height, encoding) the data comes in. What do we need to do to get these values? TIA, Schobi From thiagossantos at gmail.com Mon Sep 1 16:43:09 2008 From: thiagossantos at gmail.com (thiagoss) Date: Mon, 1 Sep 2008 11:43:09 -0300 Subject: [gst-devel] getting data format info In-Reply-To: <48BBF9CC.8040304@gmx.de> References: <48BBF9CC.8040304@gmx.de> Message-ID: On Mon, Sep 1, 2008 at 11:18 AM, Hendrik Schober wrote: > Hi, > > we are using (or is that abusing?) gstreamer in order to capture > video data from an USB camera. For this we have setup a v4lsrc > and a fake sink and grab the data in the handoff hook of the > latter. So far, so good. However, despite several hours of > tinkering we haven't found out how to determine what format > (width, height, encoding) the data comes in. > What do we need to do to get these values? Check the 'caps' in the source pad of the v4lsrc element. > > > TIA, > > Schobi > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) -------------- next part -------------- An HTML attachment was scrubbed... URL: From xharada at gmail.com Mon Sep 1 16:43:26 2008 From: xharada at gmail.com (Danilo Freire) Date: Mon, 1 Sep 2008 11:43:26 -0300 Subject: [gst-devel] Google Talk = Open Source? In-Reply-To: References: Message-ID: <98b1b78e0809010743n1bf8b5d1r5b2b744486e1d5b4@mail.gmail.com> Hi, I dont know where is the code, but, probably, they use the Telepathy framework with Jabber protocol. http://telepathy.freedesktop.org/ []s On Wed, Aug 27, 2008 at 6:32 PM, Merrick Fonnesbeck < MFonnesbeck at sorenson.com> wrote: > Does anyone know if the Google Talk (Video) application on Nokia N8x0 > internet tablets are Open Source? Or is the source somewhere out there for > looking and learning? If someone knows please let me know and please > include the URL for the code. I am really wondering how they are doing > capturing from the camera and streaming the video out across the internet. > Thanks. > > MFonnesbeck > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Danilo Freire Laborat?rio de Sistemas Embarcados e Computa??o Pervasiva Centro de Engenharia El?trica e Inform?tica - CEEI Universidade Federal de Campina Grande - UFCG -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Mon Sep 1 16:48:17 2008 From: wim.taymans at gmail.com (Wim Taymans) Date: Mon, 01 Sep 2008 16:48:17 +0200 Subject: [gst-devel] GstRtpbin and RTCP sender reports In-Reply-To: <200809011124.31770.lassi.vaatamoinen@tut.fi> References: <200809011124.31770.lassi.vaatamoinen@tut.fi> Message-ID: <1220280497.6781.15.camel@metal> On Mon, 2008-09-01 at 11:24 +0300, Lassi V??t?m?inen wrote: > Hi, > > I was wondering whether the Gstrtpbin is capable of sending RTCP SR packets? I > managed to build a pipeline using the gstrtpbin and gst-launch, but only RR > packets were sent from the sender. > When gstrtpbin is sending data (using the send_rtp_sink_%d request pad) and when a send_rtcp_src_%d request pad is requested from gstrtpbin, it will send SR reports on the send_rtcp_src_%d pad. You need to request these pads with the same number filled in for the %d in the above padtemplate names. > Another alternative would be to use Rtpbin, but I am not sure how to use this, > so an example would be nice. don't use this element, it's deprecated and does not work correctly. Wim > > Best regards, > -Lassi > From benoit.fouet at purplelabs.com Mon Sep 1 16:47:26 2008 From: benoit.fouet at purplelabs.com (Benoit Fouet) Date: Mon, 01 Sep 2008 16:47:26 +0200 Subject: [gst-devel] getting data format info In-Reply-To: <48BBF9CC.8040304@gmx.de> References: <48BBF9CC.8040304@gmx.de> Message-ID: <48BC007E.9030401@purplelabs.com> Hendrik Schober wrote: > Hi, > > we are using (or is that abusing?) gstreamer in order to capture > video data from an USB camera. For this we have setup a v4lsrc > and a fake sink and grab the data in the handoff hook of the > latter. So far, so good. However, despite several hours of > tinkering we haven't found out how to determine what format > (width, height, encoding) the data comes in. > What do we need to do to get these values? > > if you're using gst-laucnh to do that, you can add -v to your command line -- Benoit Fouet Purple Labs S.A. www.purplelabs.com From spamtrap at gmx.de Mon Sep 1 16:56:23 2008 From: spamtrap at gmx.de (Hendrik Schober) Date: Mon, 01 Sep 2008 16:56:23 +0200 Subject: [gst-devel] getting data format info In-Reply-To: References: <48BBF9CC.8040304@gmx.de> Message-ID: thiagoss wrote: > > On Mon, Sep 1, 2008 at 11:18 AM, Hendrik Schober > wrote: > > Hi, > > we are using (or is that abusing?) gstreamer in order to capture > video data from an USB camera. For this we have setup a v4lsrc > and a fake sink and grab the data in the handoff hook of the > latter. So far, so good. However, despite several hours of > tinkering we haven't found out how to determine what format > (width, height, encoding) the data comes in. > What do we need to do to get these values? > > > Check the 'caps' in the source pad of the v4lsrc element. Thanks for your answer. In fact, we already did that -- and got back a rather long list of caps. But that doesn't tell us which one's currently used. (Also, they specify a range for width and height, while we need the ones that actually are negotiated.) Schobi From spamtrap at gmx.de Mon Sep 1 16:57:27 2008 From: spamtrap at gmx.de (Hendrik Schober) Date: Mon, 01 Sep 2008 16:57:27 +0200 Subject: [gst-devel] getting data format info In-Reply-To: <48BC007E.9030401@purplelabs.com> References: <48BBF9CC.8040304@gmx.de> <48BC007E.9030401@purplelabs.com> Message-ID: Benoit Fouet wrote: > Hendrik Schober wrote: >> Hi, >> >> we are using (or is that abusing?) gstreamer in order to capture >> video data from an USB camera. For this we have setup a v4lsrc >> and a fake sink and grab the data in the handoff hook of the >> latter. So far, so good. However, despite several hours of >> tinkering we haven't found out how to determine what format >> (width, height, encoding) the data comes in. >> What do we need to do to get these values? >> >> > > if you're using gst-laucnh to do that, you can add -v to your command line No, we're doing this in code (C++). Schobi From thiagossantos at gmail.com Mon Sep 1 17:18:36 2008 From: thiagossantos at gmail.com (thiagoss) Date: Mon, 1 Sep 2008 12:18:36 -0300 Subject: [gst-devel] getting data format info In-Reply-To: References: <48BBF9CC.8040304@gmx.de> Message-ID: On Mon, Sep 1, 2008 at 11:56 AM, Hendrik Schober wrote: > thiagoss wrote: > > > > On Mon, Sep 1, 2008 at 11:18 AM, Hendrik Schober > > wrote: > > > > Hi, > > > > we are using (or is that abusing?) gstreamer in order to capture > > video data from an USB camera. For this we have setup a v4lsrc > > and a fake sink and grab the data in the handoff hook of the > > latter. So far, so good. However, despite several hours of > > tinkering we haven't found out how to determine what format > > (width, height, encoding) the data comes in. > > What do we need to do to get these values? > > > > > > Check the 'caps' in the source pad of the v4lsrc element. > > Thanks for your answer. > In fact, we already did that -- and got back a rather long > list of caps. But that doesn't tell us which one's currently > used. (Also, they specify a range for width and height, while > we need the ones that actually are negotiated.) > You should check the caps in the buffers you get from the handoff buffers instead. > > Schobi > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) -------------- next part -------------- An HTML attachment was scrubbed... URL: From spamtrap at gmx.de Mon Sep 1 17:59:05 2008 From: spamtrap at gmx.de (Hendrik Schober) Date: Mon, 01 Sep 2008 17:59:05 +0200 Subject: [gst-devel] getting data format info In-Reply-To: References: <48BBF9CC.8040304@gmx.de> Message-ID: thiagoss wrote: > > > On Mon, Sep 1, 2008 at 11:56 AM, Hendrik Schober > wrote: > > thiagoss wrote: > > > > On Mon, Sep 1, 2008 at 11:18 AM, Hendrik Schober > > >> wrote: > > > > Hi, > > > > we are using (or is that abusing?) gstreamer in order to capture > > video data from an USB camera. For this we have setup a v4lsrc > > and a fake sink and grab the data in the handoff hook of the > > latter. So far, so good. However, despite several hours of > > tinkering we haven't found out how to determine what format > > (width, height, encoding) the data comes in. > > What do we need to do to get these values? > > > > > > Check the 'caps' in the source pad of the v4lsrc element. > > Thanks for your answer. > In fact, we already did that -- and got back a rather long > list of caps. But that doesn't tell us which one's currently > used. (Also, they specify a range for width and height, while > we need the ones that actually are negotiated.) > > > You should check the caps in the buffers you get from the handoff > buffers instead. Thanks, we hadn't even found those yet. However, this now leads to another question: Since these are rather suboptimal for us, we need to change them. We could use the ffmpegcolorspace conversion plugin, but IIUC we need to make this and the fakesink use the right pad. Where and how would we do this? When we setup the pipeline? By forcing some caps/pad onto the fakesink's source? Schobi From thiagossantos at gmail.com Mon Sep 1 18:34:24 2008 From: thiagossantos at gmail.com (thiagoss) Date: Mon, 1 Sep 2008 13:34:24 -0300 Subject: [gst-devel] getting data format info In-Reply-To: References: <48BBF9CC.8040304@gmx.de> Message-ID: On Mon, Sep 1, 2008 at 12:59 PM, Hendrik Schober wrote: > thiagoss wrote: > > > > > > On Mon, Sep 1, 2008 at 11:56 AM, Hendrik Schober > > wrote: > > > > thiagoss wrote: > > > > > > On Mon, Sep 1, 2008 at 11:18 AM, Hendrik Schober > > > > >> wrote: > > > > > > Hi, > > > > > > we are using (or is that abusing?) gstreamer in order to > capture > > > video data from an USB camera. For this we have setup a v4lsrc > > > and a fake sink and grab the data in the handoff hook of the > > > latter. So far, so good. However, despite several hours of > > > tinkering we haven't found out how to determine what format > > > (width, height, encoding) the data comes in. > > > What do we need to do to get these values? > > > > > > > > > Check the 'caps' in the source pad of the v4lsrc element. > > > > Thanks for your answer. > > In fact, we already did that -- and got back a rather long > > list of caps. But that doesn't tell us which one's currently > > used. (Also, they specify a range for width and height, while > > we need the ones that actually are negotiated.) > > > > > > You should check the caps in the buffers you get from the handoff > > buffers instead. > > Thanks, we hadn't even found those yet. > However, this now leads to another question: > Since these are rather suboptimal for us, we need to change > them. We could use the ffmpegcolorspace conversion plugin, > but IIUC we need to make this and the fakesink use the right > pad. > Where and how would we do this? When we setup the pipeline? > By forcing some caps/pad onto the fakesink's source? > You can do like this: v4lsrc ! ffmpegcolorspace ! capsfilter caps="yourdesiredcaps" ! fakesink If you need scaling put a videoscale between ffmpegcolorspace and capsfilter. > > Schobi > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Mon Sep 1 19:48:49 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Mon, 01 Sep 2008 20:48:49 +0300 Subject: [gst-devel] Google Talk = Open Source? In-Reply-To: References: Message-ID: <48BC2B01.4020602@hora-obscura.de> hi, Merrick Fonnesbeck schrieb: > Does anyone know if the Google Talk (Video) application on Nokia N8x0 > internet tablets are Open Source? Or is the source somewhere out there > for looking and learning? If someone knows please let me know and > please include the URL for the code. I am really wondering how they are > doing capturing from the camera and streaming the video out across the > internet. Thanks. it uses telepathy with is fully open source. The ui of the client isn't oepn source though. http://telepathy.freedesktop.org/wiki/ Stefan > > MFonnesbeck > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From gstelzz at yahoo.fr Mon Sep 1 21:44:42 2008 From: gstelzz at yahoo.fr (Aurelien Grimaud) Date: Mon, 01 Sep 2008 21:44:42 +0200 Subject: [gst-devel] GStreamer General State and BIN Issue In-Reply-To: <8c192ddd0808312200l6920d460q67411ff787252759@mail.gmail.com> References: <8c192ddd0808290339q4f345fddl9d280c63aa8354b3@mail.gmail.com> <8c192ddd0808312200l6920d460q67411ff787252759@mail.gmail.com> Message-ID: <48BC462A.70705@yahoo.fr> Hi, Try to set your new audiotestsrc in RUNNING state after linking. Another solution will be to not get rid of audiotestsrc. One way to change the pipeline behavior without adding or removing any element is to use input-selector, output-selector and tee. Pipeline works better if they are always linked from src to sink. I use "tee ! fakesink" to get an always running pipeline. I guess input-selector ! fakesink should do the trick too. I can then add or remove any modifier after the tee or selector. Aurelien Manish Rana a ?crit : > Dear Eric, > > Thanks a lot for your suggestion... > > I can't put the pipeline in the Paused State as there are other bins > also in the pipeline and they need to be running.... > Also what effect will it cause on the live source in my pipeline..as it > has udpsink as well........... > > BR > Manish > > On Mon, Sep 1, 2008 at 7:29 AM, Eric Zhang > wrote: > > Hi, Manish: > > I think you should pause the pipeline and re-play it after you > inserted audiotestsrc into pipeline again. > > Eric > > 2008/8/29 Manish Rana > > > Hi All, > > > I am facing one issue in Gstreamer, I use following sequance. > > 1. Pipaline is created. > 2. Followink elements are there in Pipeline: > a. audiotestsrc > b. caps ! queue ! amrnbenc ! rtpamrpay ( One BIN - > Called SRC BIN ) > b. RTPBIN > c. udpsink > > Now if I unlink and remove audiotestsrc from my pipeline, that > is ok. > But if i again add this element to my pipeline there is no data > on the audiotestsrc (checked using pad PROBE data.) > > Please let me know where i am doing it wrong. > As i think logically there is something messed up :( > > > Thanks a lot.. > Manish > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move > Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win > great prizes > Grand prize is a trip for two to an Open Source event anywhere > in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win > great prizes > Grand prize is a trip for two to an Open Source event anywhere in > the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From guillaume.dorchies at gmail.com Mon Sep 1 23:48:38 2008 From: guillaume.dorchies at gmail.com (Guillaume Dorchies) Date: Mon, 1 Sep 2008 23:48:38 +0200 Subject: [gst-devel] gstrtpbin and langage C or ruby Message-ID: <6a26c9a30809011448w1dd20904idd1979dcdc2f566e@mail.gmail.com> Hello I try to create a little rtp server with gstreamer in C or in ruby. Where can I find a translation of this line "gst-launch-0.10 -v gstrtpbin name=rtpbin \ filesrc location="/data/file.mp3" ! mad ! audioconvert ! rtpL16pay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink port=5000 " in C or in ruby. I wrote this code in ruby def create_pipeline_hard # Create a pipeline to hold the elements pipeline = Gst::Pipeline.new("my_pipeline") filesrc=Gst::ElementFactory.make("filesrc","filesrc") filesrc.location=FILE_ONE mad=Gst::ElementFactory.make("mad") audioconvert=Gst::ElementFactory.make("audioconvert") pay=Gst::ElementFactory.make("rtpL16pay") @rtpbin=Gst::ElementFactory.make('gstrtpbin','rtpbin') puts @rtpbin.public_methods.inspect udpsink=Gst::ElementFactory.make('udpsink') pipeline.add( filesrc, mad, audioconvert, pay, @rtpbin, udpsink) @rtpbin.signal_connect("pad-added") do | source, pad | puts "in pad-added #{pad.name}" if pad.name.include?('send_rtp_src') pad.link(udpsink.get_pad('sink')) end end filesrc >> mad >> audioconvert >> pay #Link the pads # send_rtp_sink=@rtpbin.get_request_pad('send_rtp_sink_%d') send_rtp_sink=@rtpbin.get_request_pad('send_rtp_sink_0') pay.get_pad('src').link(send_rtp_sink) return pipeline end the pipeline work but it don't send music Thanks a lot Guillaume -------------- next part -------------- An HTML attachment was scrubbed... URL: From guillaume.dorchies at gmail.com Mon Sep 1 23:38:27 2008 From: guillaume.dorchies at gmail.com (Guillaume Dorchies) Date: Mon, 1 Sep 2008 23:38:27 +0200 Subject: [gst-devel] gstrtpbin and langage C Message-ID: <6a26c9a30809011438pb18cc40qc4e77eb0efad0386@mail.gmail.com> Hello I try to create a little rtp server with gstreamer in C or in ruby. Where can I find a translation of this line "gst-launch-0.10 -v gstrtpbin name=rtpbin \ filesrc location="/data/file.mp3" ! mad ! audioconvert ! rtpL16pay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink port=5000 " in C or in ruby. I wrote this code in ruby def create_pipeline_hard # Create a pipeline to hold the elements pipeline = Gst::Pipeline.new("my_pipeline") filesrc=Gst::ElementFactory.make("filesrc","filesrc") filesrc.location=FILE_ONE mad=Gst::ElementFactory.make("mad") audioconvert=Gst::ElementFactory.make("audioconvert") pay=Gst::ElementFactory.make("rtpL16pay") @rtpbin=Gst::ElementFactory.make('gstrtpbin','rtpbin') puts @rtpbin.public_methods.inspect udpsink=Gst::ElementFactory.make('udpsink') pipeline.add( filesrc, mad, audioconvert, pay, @rtpbin, udpsink) @rtpbin.signal_connect("pad-added") do | source, pad | puts "in pad-added #{pad.name}" if pad.name.include?('send_rtp_src') pad.link(udpsink.get_pad('sink')) end end filesrc >> mad >> audioconvert >> pay #Link the pads # send_rtp_sink=@rtpbin.get_request_pad('send_rtp_sink_%d') send_rtp_sink=@rtpbin.get_request_pad('send_rtp_sink_0') pay.get_pad('src').link(send_rtp_sink) return pipeline end the pipeline work but it don't send music Thanks a lot Guillaume -------------- next part -------------- An HTML attachment was scrubbed... URL: From sachinpandhare at gmail.com Tue Sep 2 07:00:19 2008 From: sachinpandhare at gmail.com (Sachin Pandhare) Date: Tue, 2 Sep 2008 10:30:19 +0530 Subject: [gst-devel] encode / decode chain didn't work In-Reply-To: <1220273966.6781.8.camel@metal> References: <72cf309c0809010541m28186ah692486bab3337c80@mail.gmail.com> <1220273966.6781.8.camel@metal> Message-ID: <72cf309c0809012200v59d0d352o6f5a467649b57243@mail.gmail.com> Thanks Wim for this information, it works fine :) Sudarshan, I was able to encode the data in the file with the command that i used and terminate it with Ctrl+C. but as i was missing the container it didn't decode data correctly. On Mon, Sep 1, 2008 at 6:29 PM, Wim Taymans wrote: > On Mon, 2008-09-01 at 18:11 +0530, Sachin Pandhare wrote: > > Hi, > > i was testing basic encode and decode chains. > > > > gst-launch videotestsrc ! ffmpegcolorspace ! theoraenc ! filesink > > location=/test_theoraenc.bin > > > this created a file. then i tried > > > > gst-launch filesrc location=/test_theoraenc.bin ! theoradec ! > > ffmpegcolorspace ! xvimagesink > > > > but this gave me following error: > > > -------------------------------------------------------------------------------------------------- > > ERROR: from element /pipeline0/theoradec0: Could not decode stream. > > Additional debug info: > > theoradec.c(1166): theora_handle_data_packet > > (): /pipeline0/theoradec0: > > no header sent yet > > > -------------------------------------------------------------------------------------------------- > > > > was there any mistake in my commands? > > yes, theora is a format that should be encapsulated into a container > such as ogg, like this: > > gst-launch videotestsrc ! ffmpegcolorspace ! theoraenc ! oggmux ! > filesink location=/test_theoraenc.ogg > > and to play it back: > > gst-launch filesrc location=/test_theoraenc.ogg ! oggdemux ! theoradec ! > ffmpegcolorspace ! xvimagesink > > The reason is that theora relies on the container format to feed it > exactly one encoded frame, there are no fame markers in the raw theora > bitstream otherwise. > > Wim > > > > > > Thanks, > > Sachin > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Philipp.Leibfried at gmx.de Tue Sep 2 09:54:42 2008 From: Philipp.Leibfried at gmx.de (Philipp Leibfried) Date: Tue, 02 Sep 2008 09:54:42 +0200 Subject: [gst-devel] Trouble playing a-Law encoded file over RTP/UDP Message-ID: <20080902075442.81710@gmx.net> Hi again, I have now 'repaired' my sender pipeline to look like this (the file I use actually is a WAV container). gst-launch-0.10 filesrc location=/home/pl/Projects/gstSender/debug/src/2079.wav ! wavparse ! rtppcmapay max-ptime=20000000 ! udpsink host="localhost" port=4044 Wireshark tells me that there is no significant jitter on the RTP "stream" I'm sending (the jitter is 0.01 msec). However, my receiver pipeline gst-launch-0.10 udpsrc port=4044 ! rtppcmadepay ! audio/x-alaw, channels=1, rate=8000 ! alawdec ! alsasink still tells me there is a timestamp disontinuity. gstbaseaudiosink.c(1188): gst_base_audio_sink_render (): /pipeline0/alsasink0: Unexpected discontinuity in audio timestamps of more than half a second (0:00:02.049250000), resyncing WARNING: from element /pipeline0/alsasink0: Compensating for audio synchronisation problems Forgive my ignorance, but is there something I still need to do on the receiver side? According to Wireshark, my timestamps are good, or is this a misunderstanding on my part? Thanks -Philipp -- Der GMX SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! Ideal f?r Modem und ISDN: http://www.gmx.net/de/go/smartsurfer From sachinpandhare at gmail.com Tue Sep 2 10:38:03 2008 From: sachinpandhare at gmail.com (Sachin Pandhare) Date: Tue, 2 Sep 2008 14:08:03 +0530 Subject: [gst-devel] AMR support in AV mux/demux Message-ID: <72cf309c0809020138s7be1382am3e72f899e381b299@mail.gmail.com> Hi, could you please let me know if there is any AV mux/demux with AMR and AAC support? Thanks, Sachin -------------- next part -------------- An HTML attachment was scrubbed... URL: From lassi.vaatamoinen at tut.fi Tue Sep 2 09:42:31 2008 From: lassi.vaatamoinen at tut.fi (Lassi =?iso-8859-1?q?V=E4=E4t=E4m=F6inen?=) Date: Tue, 2 Sep 2008 10:42:31 +0300 Subject: [gst-devel] GstRtpbin and RTCP sender reports Message-ID: <200809021042.31719.lassi.vaatamoinen@tut.fi> 2008/9/1 Wim Taymans > On Mon, 2008-09-01 at 11:24 +0300, Lassi V??t?m?inen wrote: > > Hi, > > > > I was wondering whether the Gstrtpbin is capable of sending RTCP SR > > packets? I managed to build a pipeline using the gstrtpbin and > > gst-launch, but only RR packets were sent from the sender. > > When gstrtpbin is sending data (using the send_rtp_sink_%d request pad) > and when a send_rtcp_src_%d request pad is requested from gstrtpbin, it > will send SR reports on the send_rtcp_src_%d pad. You need to request > these pads with the same number filled in for the %d in the above > padtemplate names. I think I have these already, but I'm not sure if I am using them properly. Here is my pipeline: #!/bin/sh VPORT=8234 APORT=8236 HOST=${1:-130.230.52.193} gst-launch-0.10 gstrtpbin name=rtpbin \ gconfv4l2src ! \ video/x-raw-yuv,width=176,height=144,framerate=\(fraction\15/1 ! \ hantro4200enc stream-type=1 profile-and-level=1001 ! \ video/x-h263,framerate=\(fraction\)15/1 ! \ rtph263ppay mtu=1438 ! \ rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=$HOST port=$VPORT \ rtpbin.send_rtcp_src_0 ! \ udpsink host=$HOST port=8235 sync=false async=false \ udpsrc port=8239 ! rtpbin.recv_rtcp_sink_0 \ dsppcmsrc ! queue ! \ audio/x-raw-int,channels=1,rate=8000 ! mulawenc ! rtppcmupay mtu=1438 ! \ rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink host=$HOST port=$APORT \ rtpbin.send_rtcp_src_1 ! udpsink host=$HOST port=8237 \ sync=false async=false \ udpsrc port=8241 ! rtpbin.recv_rtcp_sink_1 I apologize for the poor formatting :) Oh, and this is used on N800, so that's why the hantro encoder :) -Lassi From spamtrap at gmx.de Tue Sep 2 10:34:58 2008 From: spamtrap at gmx.de (Hendrik Schober) Date: Tue, 02 Sep 2008 10:34:58 +0200 Subject: [gst-devel] getting data format info In-Reply-To: References: <48BBF9CC.8040304@gmx.de> Message-ID: thiagoss wrote: > On Mon, Sep 1, 2008 at 12:59 PM, Hendrik Schober > wrote: > > thiagoss wrote: > > > > > > On Mon, Sep 1, 2008 at 11:56 AM, Hendrik Schober > > >> wrote: > > > > thiagoss wrote: > > > > > > On Mon, Sep 1, 2008 at 11:18 AM, Hendrik Schober > > > > > > > > >>> wrote: > > > > > > Hi, > > > > > > we are using (or is that abusing?) gstreamer in order > to capture > > > video data from an USB camera. For this we have setup > a v4lsrc > > > and a fake sink and grab the data in the handoff hook > of the > > > latter. So far, so good. However, despite several hours of > > > tinkering we haven't found out how to determine what > format > > > (width, height, encoding) the data comes in. > > > What do we need to do to get these values? > > > > > > > > > Check the 'caps' in the source pad of the v4lsrc element. > > > > Thanks for your answer. > > In fact, we already did that -- and got back a rather long > > list of caps. But that doesn't tell us which one's currently > > used. (Also, they specify a range for width and height, while > > we need the ones that actually are negotiated.) > > > > > > You should check the caps in the buffers you get from the handoff > > buffers instead. > > Thanks, we hadn't even found those yet. > However, this now leads to another question: > Since these are rather suboptimal for us, we need to change > them. We could use the ffmpegcolorspace conversion plugin, > but IIUC we need to make this and the fakesink use the right > pad. > Where and how would we do this? When we setup the pipeline? > By forcing some caps/pad onto the fakesink's source? > > > You can do like this: > > v4lsrc ! ffmpegcolorspace ! capsfilter caps="yourdesiredcaps" ! fakesink > > If you need scaling put a videoscale between ffmpegcolorspace and > capsfilter. That did the trick! We got it working now. Thank you for your support. Schobi From lfarkas at lfarkas.org Tue Sep 2 11:32:58 2008 From: lfarkas at lfarkas.org (Farkas Levente) Date: Tue, 02 Sep 2008 11:32:58 +0200 Subject: [gst-devel] GstRtpbin and RTCP sender reports In-Reply-To: <1220280497.6781.15.camel@metal> References: <200809011124.31770.lassi.vaatamoinen@tut.fi> <1220280497.6781.15.camel@metal> Message-ID: <48BD084A.6020708@lfarkas.org> Wim Taymans wrote: >> Another alternative would be to use Rtpbin, but I am not sure how to use this, >> so an example would be nice. > > don't use this element, it's deprecated and does not work correctly. is something deprecated why not documented? according to the page (which is come from the main documentation page): http://gstreamer.freedesktop.org/documentation/rtp.html "The RTP stack described below on this page is in place and works, but a new and improved one is in development and you can find the latest details on that on the Farsight Wiki page." so for me it seems the farsight is the newer (what's more gstrtpbin is in gst-plugins-bad) although you said it's deprecated. it's very clear for this list that almost half of the mail is about rtp. it can be two reason: it's very popular or it's not clear to most people how to use it. imho it'd be better to everyone to make these thinks cleaner... -- Levente "Si vis pacem para bellum!" From lassi.vaatamoinen at tut.fi Tue Sep 2 12:11:39 2008 From: lassi.vaatamoinen at tut.fi (Lassi =?iso-8859-1?q?V=E4=E4t=E4m=F6inen?=) Date: Tue, 2 Sep 2008 13:11:39 +0300 Subject: [gst-devel] GstRtpbin and RTCP sender reports In-Reply-To: <200809021042.31719.lassi.vaatamoinen@tut.fi> References: <200809021042.31719.lassi.vaatamoinen@tut.fi> Message-ID: <200809021311.39599.lassi.vaatamoinen@tut.fi> On Tuesday 02 September 2008 10:42:31 Lassi V??t?m?inen wrote: > 2008/9/1 Wim Taymans > > > On Mon, 2008-09-01 at 11:24 +0300, Lassi V??t?m?inen wrote: > > > Hi, > > > > > > I was wondering whether the Gstrtpbin is capable of sending RTCP SR > > > packets? I managed to build a pipeline using the gstrtpbin and > > > gst-launch, but only RR packets were sent from the sender. > > > > When gstrtpbin is sending data (using the send_rtp_sink_%d request pad) > > and when a send_rtcp_src_%d request pad is requested from gstrtpbin, it > > will send SR reports on the send_rtcp_src_%d pad. You need to request > > these pads with the same number filled in for the %d in the above > > padtemplate names. > > I think I have these already, but I'm not sure if I am using them properly. > Here is my pipeline: > > #!/bin/sh > > > VPORT=8234 > APORT=8236 > HOST=${1:-130.230.52.193} > > gst-launch-0.10 gstrtpbin name=rtpbin \ > gconfv4l2src ! \ > video/x-raw-yuv,width=176,height=144,framerate=\(fraction\15/1 ! \ > hantro4200enc stream-type=1 profile-and-level=1001 ! \ > video/x-h263,framerate=\(fraction\)15/1 ! \ > rtph263ppay mtu=1438 ! \ > rtpbin.send_rtp_sink_0 \ > rtpbin.send_rtp_src_0 ! udpsink host=$HOST port=$VPORT \ > rtpbin.send_rtcp_src_0 ! \ > udpsink host=$HOST port=8235 sync=false async=false \ > udpsrc port=8239 ! rtpbin.recv_rtcp_sink_0 \ > dsppcmsrc ! queue ! \ > audio/x-raw-int,channels=1,rate=8000 ! mulawenc ! rtppcmupay mtu=1438 ! \ > rtpbin.send_rtp_sink_1 \ > rtpbin.send_rtp_src_1 ! udpsink host=$HOST port=$APORT \ > rtpbin.send_rtcp_src_1 ! udpsink host=$HOST port=8237 \ > sync=false async=false \ > udpsrc port=8241 ! rtpbin.recv_rtcp_sink_1 > > > I apologize for the poor formatting :) Oh, and this is used on N800, so > that's why the hantro encoder :) OK, I managed to debug my problem to a point, where it seems that the version of gstrtpbin is not completely bug-free on the N800 :) I tried the example pipeline from http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html with slight modifications (ulaw encoder instead of amr-nb) on my desktop-PC, and was able to capture SR packets. The PC version info shows me the following: Plugin Details: Name: gstrtpmanager Description: RTP session management plugin library Filename: /usr/lib/gstreamer-0.10/libgstrtpmanager.so Version: 0.10.6 License: LGPL Source module: gst-plugins-bad Binary package: GStreamer Bad Plugins (Ubuntu) Origin URL: https://launchpad.net/distros/ubuntu/+source/gst-plugins-bad0.10 On N800 the Version number is 1.0, instead of the format 0.10.X . So i assume the bug is evident? :) -Lassi -- Lassi V??t?m?inen M.Sc., Researcher Tampere University of Technology / Department of Communications Engineering Room TH209, tel. 03-3115 5124 From zaheermerali at gmail.com Tue Sep 2 13:01:24 2008 From: zaheermerali at gmail.com (Zaheer Merali) Date: Tue, 2 Sep 2008 12:01:24 +0100 Subject: [gst-devel] re-license and move of Fluendo mpeg ps and ts demuxers Message-ID: <15e616860809020401t348c31d9nbeda0164e03b37d0@mail.gmail.com> Fluendo is dual licensing the Fluendo mpeg demuxers as MPL/LGPL and is moving them to GStreamer CVS. This should address any licensing issues that people may have had with the demuxers. I am going to commit the current trunk into gst-plugins-bad not because the elements are bad, but of course it has to go through the qualification process if they are to move into ugly or good. We use the ts demuxer at Flumotion in production and it is very robust so do not really want to see any major refactoring so will be pushing for it to hit good or ugly soon. Zaheer From sachinpandhare at gmail.com Tue Sep 2 13:19:06 2008 From: sachinpandhare at gmail.com (Sachin Pandhare) Date: Tue, 2 Sep 2008 16:49:06 +0530 Subject: [gst-devel] aac parser Message-ID: <72cf309c0809020419w17e87d85yd86165633f682471@mail.gmail.com> Hi, if aac parser needs to be developed can we take some plugin code as a reference and which one will be a suitable candidate for this? thanks, Sachin -------------- next part -------------- An HTML attachment was scrubbed... URL: From Philipp.Leibfried at gmx.de Tue Sep 2 14:42:50 2008 From: Philipp.Leibfried at gmx.de (Philipp Leibfried) Date: Tue, 02 Sep 2008 14:42:50 +0200 Subject: [gst-devel] Trouble playing a-Law encoded file over RTP/UDP (CONT'D) Message-ID: <20080902124250.97270@gmx.net> Hi again, I have found out that the file gets played correctly if I start the "sender" first. Does that mean that as long as no UDP packets arrive at the selected port, the receiver is playing "silence" with some sort of arbitratry timestamping of its own, and then gets thrown off once the transmission starts? Regards -Philipp -- GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen! Jetzt dabei sein: http://www.shortview.de/wasistshortview.php?mc=sv_ext_mf at gmx From hadess at hadess.net Tue Sep 2 15:08:16 2008 From: hadess at hadess.net (Bastien Nocera) Date: Tue, 02 Sep 2008 14:08:16 +0100 Subject: [gst-devel] re-license and move of Fluendo mpeg ps and ts demuxers In-Reply-To: <15e616860809020401t348c31d9nbeda0164e03b37d0@mail.gmail.com> References: <15e616860809020401t348c31d9nbeda0164e03b37d0@mail.gmail.com> Message-ID: <1220360896.4757.4.camel@cookie.hadess.net> On Tue, 2008-09-02 at 12:01 +0100, Zaheer Merali wrote: > Fluendo is dual licensing the Fluendo mpeg demuxers as MPL/LGPL and is > moving them to GStreamer CVS. This should address any licensing > issues that people may have had with the demuxers. I am going to > commit the current trunk into gst-plugins-bad not because the elements > are bad, but of course it has to go through the qualification process > if they are to move into ugly or good. We use the ts demuxer at > Flumotion in production and it is very robust so do not really want to > see any major refactoring so will be pushing for it to hit good or > ugly soon. Success! Thanks. Can you let us know when you've moved the code, so I can move the few bugs I have in the trac pages? Cheers From ensonic at hora-obscura.de Tue Sep 2 15:24:14 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 02 Sep 2008 16:24:14 +0300 Subject: [gst-devel] aac parser In-Reply-To: <72cf309c0809020419w17e87d85yd86165633f682471@mail.gmail.com> References: <72cf309c0809020419w17e87d85yd86165633f682471@mail.gmail.com> Message-ID: <48BD3E7E.9060000@hora-obscura.de> hi, there will be one in gst-plugin-bad soon. Stefan Sachin Pandhare schrieb: > Hi, > if aac parser needs to be developed can we take some plugin code as a > reference and which one will be a suitable candidate for this? > thanks, > Sachin > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ensonic at hora-obscura.de Tue Sep 2 15:29:33 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 02 Sep 2008 16:29:33 +0300 Subject: [gst-devel] AMR support in AV mux/demux In-Reply-To: <72cf309c0809020138s7be1382am3e72f899e381b299@mail.gmail.com> References: <72cf309c0809020138s7be1382am3e72f899e381b299@mail.gmail.com> Message-ID: <48BD3FBD.2070000@hora-obscura.de> hi, I belive the soon to be merged qtmux has it. At least formaly mp4/3gp can have both and qtdemux can do them already. Stefan Sachin Pandhare schrieb: > Hi, > > could you please let me know if there is any AV mux/demux with AMR and > AAC support? > > Thanks, > Sachin > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From manish.rana at gmail.com Tue Sep 2 15:31:26 2008 From: manish.rana at gmail.com (Manish Rana) Date: Tue, 2 Sep 2008 19:01:26 +0530 Subject: [gst-devel] ffdec_mpeg4 Bitrate Issue.... Message-ID: <8c192ddd0809020631h1f0d3abboe183c9984c993aa2@mail.gmail.com> Hi All, I am trying to set bitrate=48000 on ffdec_mpeg4, and then transmitting the data using gstRTPbin and UDP. On the other end i tried calculating the data rate coming. I used WireShark for the same. After removing RTP/UDP/eth0 and all other header's the data rate is coming around 200kbps. FPS at the VideoSrc is 10. GOP size is 30. Can someone guide me what do i need to get the lower and constant bit rate from encoder. As this is eating all the bandwidth. and all my application is going for toss :"( Also i have seen there are some properties of ffenc_mpeg4 that are used for Rate Control. Please let me know. Also suggest to me to reduce overhead and ways to reduce the bitrate in RTP. I am in urgent need, Thanks a lot Manish -------------- next part -------------- An HTML attachment was scrubbed... URL: From gstelzz at yahoo.fr Tue Sep 2 15:35:42 2008 From: gstelzz at yahoo.fr (Aurelien Grimaud) Date: Tue, 02 Sep 2008 15:35:42 +0200 Subject: [gst-devel] AMR support in AV mux/demux In-Reply-To: <48BD3FBD.2070000@hora-obscura.de> References: <72cf309c0809020138s7be1382am3e72f899e381b299@mail.gmail.com> <48BD3FBD.2070000@hora-obscura.de> Message-ID: <48BD412E.6070909@yahoo.fr> What about ffmux_mov from gst-ffmpeg ? Aurelien Stefan Kost a ?crit : > hi, > > I belive the soon to be merged qtmux has it. At least formaly mp4/3gp > can have both and qtdemux can do them already. > > Stefan > > Sachin Pandhare schrieb: >> Hi, >> >> could you please let me know if there is any AV mux/demux with AMR and >> AAC support? >> >> Thanks, >> Sachin >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's challenge >> Build the coolest Linux based applications with Moblin SDK & win great prizes >> Grand prize is a trip for two to an Open Source event anywhere in the world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From zaheermerali at gmail.com Tue Sep 2 18:23:48 2008 From: zaheermerali at gmail.com (Zaheer Merali) Date: Tue, 2 Sep 2008 17:23:48 +0100 Subject: [gst-devel] re-license and move of Fluendo mpeg ps and ts demuxers In-Reply-To: <1220360896.4757.4.camel@cookie.hadess.net> References: <15e616860809020401t348c31d9nbeda0164e03b37d0@mail.gmail.com> <1220360896.4757.4.camel@cookie.hadess.net> Message-ID: <15e616860809020923x18f4e686ve86946f41fb2e37@mail.gmail.com> On Tue, Sep 2, 2008 at 2:08 PM, Bastien Nocera wrote: > On Tue, 2008-09-02 at 12:01 +0100, Zaheer Merali wrote: >> Fluendo is dual licensing the Fluendo mpeg demuxers as MPL/LGPL and is >> moving them to GStreamer CVS. This should address any licensing >> issues that people may have had with the demuxers. I am going to >> commit the current trunk into gst-plugins-bad not because the elements >> are bad, but of course it has to go through the qualification process >> if they are to move into ugly or good. We use the ts demuxer at >> Flumotion in production and it is very robust so do not really want to >> see any major refactoring so will be pushing for it to hit good or >> ugly soon. > > Success! Thanks. > > Can you let us know when you've moved the code, so I can move the few > bugs I have in the trac pages? > > Cheers > Code Moved. Zaheer From bilboed at gmail.com Tue Sep 2 18:34:51 2008 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 02 Sep 2008 18:34:51 +0200 Subject: [gst-devel] re-license and move of Fluendo mpeg ps and ts demuxers In-Reply-To: <15e616860809020923x18f4e686ve86946f41fb2e37@mail.gmail.com> References: <15e616860809020401t348c31d9nbeda0164e03b37d0@mail.gmail.com> <1220360896.4757.4.camel@cookie.hadess.net> <15e616860809020923x18f4e686ve86946f41fb2e37@mail.gmail.com> Message-ID: <1220373291.6593.15.camel@localhost> Excellent news ! Alas... one small issue worth mentionning... The names of the plugin and elementfactories are exactly the same are the ones from upstream... which creates some conflicts. We should rename them ASAP. I created a bug report regarding that, comments welcome http://bugzilla.gnome.org/show_bug.cgi?id=550468 Edward On Tue, 2008-09-02 at 17:23 +0100, Zaheer Merali wrote: > On Tue, Sep 2, 2008 at 2:08 PM, Bastien Nocera wrote: > > On Tue, 2008-09-02 at 12:01 +0100, Zaheer Merali wrote: > >> Fluendo is dual licensing the Fluendo mpeg demuxers as MPL/LGPL and is > >> moving them to GStreamer CVS. This should address any licensing > >> issues that people may have had with the demuxers. I am going to > >> commit the current trunk into gst-plugins-bad not because the elements > >> are bad, but of course it has to go through the qualification process > >> if they are to move into ugly or good. We use the ts demuxer at > >> Flumotion in production and it is very robust so do not really want to > >> see any major refactoring so will be pushing for it to hit good or > >> ugly soon. > > > > Success! Thanks. > > > > Can you let us know when you've moved the code, so I can move the few > > bugs I have in the trac pages? > > > > Cheers > > > > Code Moved. > > Zaheer > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Edward Hervey From hadess at hadess.net Tue Sep 2 18:37:06 2008 From: hadess at hadess.net (Bastien Nocera) Date: Tue, 02 Sep 2008 17:37:06 +0100 Subject: [gst-devel] re-license and move of Fluendo mpeg ps and ts demuxers In-Reply-To: <1220373291.6593.15.camel@localhost> References: <15e616860809020401t348c31d9nbeda0164e03b37d0@mail.gmail.com> <1220360896.4757.4.camel@cookie.hadess.net> <15e616860809020923x18f4e686ve86946f41fb2e37@mail.gmail.com> <1220373291.6593.15.camel@localhost> Message-ID: <1220373426.4757.15.camel@cookie.hadess.net> On Tue, 2008-09-02 at 18:34 +0200, Edward Hervey wrote: > Excellent news ! > > Alas... one small issue worth mentionning... The names of the plugin > and elementfactories are exactly the same are the ones from upstream... > which creates some conflicts. We should rename them ASAP. > > I created a bug report regarding that, comments welcome > > http://bugzilla.gnome.org/show_bug.cgi?id=550468 We could kill the old plugins in gst already, and rename the new plugins to that? From bilboed at gmail.com Tue Sep 2 18:40:17 2008 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 02 Sep 2008 18:40:17 +0200 Subject: [gst-devel] re-license and move of Fluendo mpeg ps and ts demuxers In-Reply-To: <1220373426.4757.15.camel@cookie.hadess.net> References: <15e616860809020401t348c31d9nbeda0164e03b37d0@mail.gmail.com> <1220360896.4757.4.camel@cookie.hadess.net> <15e616860809020923x18f4e686ve86946f41fb2e37@mail.gmail.com> <1220373291.6593.15.camel@localhost> <1220373426.4757.15.camel@cookie.hadess.net> Message-ID: <1220373617.6593.17.camel@localhost> Considering I'm getting crashes with the new mpegpsdemuxer I didn't have with the legacy ones... I very much doubt that. There's nothing wrong with having more than one demxuer for the time being. This is going to require a LOT more testing before deciding which one is better.... at least for mpeg-ps demuxing. Edward On Tue, 2008-09-02 at 17:37 +0100, Bastien Nocera wrote: > On Tue, 2008-09-02 at 18:34 +0200, Edward Hervey wrote: > > Excellent news ! > > > > Alas... one small issue worth mentionning... The names of the plugin > > and elementfactories are exactly the same are the ones from upstream... > > which creates some conflicts. We should rename them ASAP. > > > > I created a bug report regarding that, comments welcome > > > > http://bugzilla.gnome.org/show_bug.cgi?id=550468 > > We could kill the old plugins in gst already, and rename the new plugins > to that? > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From t.i.m at zen.co.uk Tue Sep 2 18:57:35 2008 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 02 Sep 2008 17:57:35 +0100 Subject: [gst-devel] re-license and move of Fluendo mpeg ps and ts demuxers In-Reply-To: <1220373426.4757.15.camel@cookie.hadess.net> References: <15e616860809020401t348c31d9nbeda0164e03b37d0@mail.gmail.com> <1220360896.4757.4.camel@cookie.hadess.net> <15e616860809020923x18f4e686ve86946f41fb2e37@mail.gmail.com> <1220373291.6593.15.camel@localhost> <1220373426.4757.15.camel@cookie.hadess.net> Message-ID: <1220374655.21105.8.camel@mini.centricular.net> On Tue, 2008-09-02 at 17:37 +0100, Bastien Nocera wrote: > We could kill the old plugins in gst already, and rename the new plugins > to that? Can't do that, since the behaviour of the two demuxers is different in some aspects (pad names, media format output etc.), which might break existing applications using the current -ugly demuxer (-ugly is supposed to have the same stability guarantees as -good if I'm not mistaken). We can deprecate the one in -ugly though and declare the new one the one to use if we find it's better or more hackable. Cheers -Tim From msmith at xiph.org Tue Sep 2 19:34:17 2008 From: msmith at xiph.org (Michael Smith) Date: Tue, 2 Sep 2008 10:34:17 -0700 Subject: [gst-devel] aac parser In-Reply-To: <72cf309c0809020419w17e87d85yd86165633f682471@mail.gmail.com> References: <72cf309c0809020419w17e87d85yd86165633f682471@mail.gmail.com> Message-ID: <3c1737210809021034h7e9608cah2ef234f3c69abecb@mail.gmail.com> Assuming you're willing license your code as LGPL, then the mp3parse element is probably a good starting point. Mike On Tue, Sep 2, 2008 at 4:19 AM, Sachin Pandhare wrote: > Hi, > if aac parser needs to be developed can we take some plugin code as a > reference and which one will be a suitable candidate for this? > thanks, > Sachin > From levi.pope at gmail.com Tue Sep 2 22:51:45 2008 From: levi.pope at gmail.com (Levi Pope) Date: Tue, 2 Sep 2008 15:51:45 -0500 Subject: [gst-devel] Duration and Seek in MPEG TS File Message-ID: <3afe75670809021351v2ba84d86k1a38074f1cf51373@mail.gmail.com> Has anyone been able to seek or get the duration of a TS file with Gstreamer. I have tried using flutsdemux and it does not seem to work. Thanks Levi -------------- next part -------------- An HTML attachment was scrubbed... URL: From tristan at sat.qc.ca Tue Sep 2 23:38:17 2008 From: tristan at sat.qc.ca (Tristan Matthews) Date: Tue, 02 Sep 2008 17:38:17 -0400 Subject: [gst-devel] add latency to audio in gstrtpbin Message-ID: <48BDB249.1040009@sat.qc.ca> Hi, If I have a pipeline using gstrtpbin (similar to the example in the documentation) to send audio and video, what is the best/most reliable way of adding latency to the audio? Would gst_event_new_latency work (and if so, how), or am I missing its intent: http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/gstreamer/html/gstreamer-GstEvent.html#gst-event-new-latency Basically my concern is that if video capture is too slow, can I manually adjust the audio latency to match. Best, Tristan -- Tristan Matthews Soci?t? des arts technologiques [SAT] email: tristan at sat.qc.ca web: http://www.music.mcgill.ca/~tmatthews From zaheermerali at gmail.com Tue Sep 2 23:44:38 2008 From: zaheermerali at gmail.com (Zaheer Merali) Date: Tue, 2 Sep 2008 22:44:38 +0100 Subject: [gst-devel] Duration and Seek in MPEG TS File In-Reply-To: <3afe75670809021351v2ba84d86k1a38074f1cf51373@mail.gmail.com> References: <3afe75670809021351v2ba84d86k1a38074f1cf51373@mail.gmail.com> Message-ID: <15e616860809021444s2305440cn14ad08d962708eb8@mail.gmail.com> On Tue, Sep 2, 2008 at 9:51 PM, Levi Pope wrote: > Has anyone been able to seek or get the duration of a TS file with > Gstreamer. > I have tried using flutsdemux and it does not seem to work. > > Thanks > Levi This is a missing feature. Zaheer From olivier.crete at collabora.co.uk Tue Sep 2 23:45:29 2008 From: olivier.crete at collabora.co.uk (Olivier =?ISO-8859-1?Q?Cr=EAte?=) Date: Tue, 02 Sep 2008 17:45:29 -0400 Subject: [gst-devel] add latency to audio in gstrtpbin In-Reply-To: <48BDB249.1040009@sat.qc.ca> References: <48BDB249.1040009@sat.qc.ca> Message-ID: <1220391929.4342.13.camel@TesterTop3.tester.ca> On Tue, 2008-09-02 at 17:38 -0400, Tristan Matthews wrote: > Hi, > > If I have a pipeline using gstrtpbin (similar to the example in the > documentation) to send audio and video, what is the best/most reliable > way of adding latency to the audio? Would gst_event_new_latency work > (and if so, how), or am I missing its intent: > http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/gstreamer/html/gstreamer-GstEvent.html#gst-event-new-latency > > Basically my concern is that if video capture is too slow, can I > manually adjust the audio latency to match. You want to play latency properties on the audio source.. The new latency event is inform pipeline elements of the overlay latency of the pipeline. -- Olivier Cr?te olivier.crete at collabora.co.uk Collabora Ltd -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: This is a digitally signed message part URL: From levi.pope at gmail.com Wed Sep 3 01:00:49 2008 From: levi.pope at gmail.com (Levi Pope) Date: Tue, 2 Sep 2008 18:00:49 -0500 Subject: [gst-devel] Duration and Seek in MPEG TS File In-Reply-To: <15e616860809021444s2305440cn14ad08d962708eb8@mail.gmail.com> References: <3afe75670809021351v2ba84d86k1a38074f1cf51373@mail.gmail.com> <15e616860809021444s2305440cn14ad08d962708eb8@mail.gmail.com> Message-ID: <3afe75670809021600g3b3de56avd075555a8a855be8@mail.gmail.com> I would like to implement this feature but I am not sure were to start. Is there any information on how Gstreamer performs seeks and duration queries? On Tue, Sep 2, 2008 at 4:44 PM, Zaheer Merali wrote: > On Tue, Sep 2, 2008 at 9:51 PM, Levi Pope wrote: > > Has anyone been able to seek or get the duration of a TS file with > > Gstreamer. > > I have tried using flutsdemux and it does not seem to work. > > > > Thanks > > Levi > > This is a missing feature. > > Zaheer > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From huan.zheng at intel.com Wed Sep 3 04:28:25 2008 From: huan.zheng at intel.com (Zheng, Huan) Date: Wed, 3 Sep 2008 10:28:25 +0800 Subject: [gst-devel] aac parser In-Reply-To: <48BD3E7E.9060000@hora-obscura.de> References: <72cf309c0809020419w17e87d85yd86165633f682471@mail.gmail.com> <48BD3E7E.9060000@hora-obscura.de> Message-ID: Stefan Could you please elaborate on this "Soon"? One month or several months? :) It would be very nice to have an aac parser in gstreamer. And one more question: Is this parser able to parse ADIF stream into single frames? Because what I heard now is that ADIF has only one header at the beginning, and the offset of each frame can not be located unless you have finished decoding. Thanks! Best Regards, Zheng, Huan(ZBT) OTC/SSD/SSG Intel Aisa-Pacific Research & Developement Ltd Tel: 021-6116 6435 Inet: 8821 6435 Cub: 3W035 -----Original Message----- From: gstreamer-devel-bounces at lists.sourceforge.net [mailto:gstreamer-devel-bounces at lists.sourceforge.net] On Behalf Of Stefan Kost Sent: 2008?9?2? 21:24 To: Discussion of the development of GStreamer Subject: Re: [gst-devel] aac parser hi, there will be one in gst-plugin-bad soon. Stefan Sachin Pandhare schrieb: > Hi, > if aac parser needs to be developed can we take some plugin code as a > reference and which one will be a suitable candidate for this? > thanks, > Sachin > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From Levi.Pope at L-3com.com Wed Sep 3 00:58:47 2008 From: Levi.Pope at L-3com.com (Levi.Pope at L-3com.com) Date: Tue, 2 Sep 2008 17:58:47 -0500 Subject: [gst-devel] Duration and Seek in MPEG TS File In-Reply-To: <15e616860809021444s2305440cn14ad08d962708eb8@mail.gmail.com> References: <3afe75670809021351v2ba84d86k1a38074f1cf51373@mail.gmail.com> <15e616860809021444s2305440cn14ad08d962708eb8@mail.gmail.com> Message-ID: I would like to implement this feature but I am not sure were to start. Is there any information on how Gstreamer performs seeks and duration queries? -----Original Message----- From: gstreamer-devel-bounces at lists.sourceforge.net [mailto:gstreamer-devel-bounces at lists.sourceforge.net] On Behalf Of Zaheer Merali Sent: Tuesday, September 02, 2008 4:45 PM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] Duration and Seek in MPEG TS File On Tue, Sep 2, 2008 at 9:51 PM, Levi Pope wrote: > Has anyone been able to seek or get the duration of a TS file with > Gstreamer. > I have tried using flutsdemux and it does not seem to work. > > Thanks > Levi This is a missing feature. Zaheer ------------------------------------------------------------------------ - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bilboed at gmail.com Wed Sep 3 11:43:01 2008 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 03 Sep 2008 11:43:01 +0200 Subject: [gst-devel] Duration and Seek in MPEG TS File In-Reply-To: <15e616860809021444s2305440cn14ad08d962708eb8@mail.gmail.com> References: <3afe75670809021351v2ba84d86k1a38074f1cf51373@mail.gmail.com> <15e616860809021444s2305440cn14ad08d962708eb8@mail.gmail.com> Message-ID: <1220434981.7548.0.camel@localhost> Hi all, I opened a bug on bugzilla regarding that. Comments and patches welcome. http://bugzilla.gnome.org/show_bug.cgi?id=550634 Edward On Tue, 2008-09-02 at 22:44 +0100, Zaheer Merali wrote: > On Tue, Sep 2, 2008 at 9:51 PM, Levi Pope wrote: > > Has anyone been able to seek or get the duration of a TS file with > > Gstreamer. > > I have tried using flutsdemux and it does not seem to work. > > > > Thanks > > Levi > > This is a missing feature. > > Zaheer > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From julien.isorce at gmail.com Wed Sep 3 14:53:29 2008 From: julien.isorce at gmail.com (Julien Isorce) Date: Wed, 3 Sep 2008 14:53:29 +0200 Subject: [gst-devel] dshowVIDEOsrc Message-ID: <180a127d0809030553s3ae42ed7i163eb01d64329037@mail.gmail.com> Hi, Can someone commit this patch ? http://bugzilla.gnome.org/show_bug.cgi?id=517203 I am waiting for a long time and I am using it every week so it seems to be stable. But the *main reason* of this request is to avoid to rewrite (again) the patch if someone else commit an other small patch. I know that some guys think that the dshowvideosrc is not correctly written (design) but we have nothing else for now. And the dshowvideosrc element without my patch is not usable (cannot change the videosize or the framerate). The patch resolves this problem. Julien Isorce -------------- next part -------------- An HTML attachment was scrubbed... URL: From chase.maupin at ti.com Wed Sep 3 16:36:08 2008 From: chase.maupin at ti.com (Maupin, Chase) Date: Wed, 3 Sep 2008 09:36:08 -0500 Subject: [gst-devel] Problem inheriting from GstVideoSink class Message-ID: <131E5DFBE7373E4C8D813795A6AA7F080127A04F75@dlee06.ent.ti.com> All, I am working on writing a video sink and I am trying to inherit from the GstVideoSink class so that I get functionality such as dropping frames that are past their display time. I have a reference video sink which uses this class and for experimentation purposes I placed a 1 second delay in the render function. I observed that the display times in the buffers given to the sink were now 1 second apart and the video was skipping as I would expect. I put the same delay in my video sink and I am still seeing time stamps that are only .033 seconds apart. Basically the frames that should have been thrown away during the delay are just getting queued up and arrive in order after the delay. I am doing the following to inherit from the GstVideoSink class: - In my .c code I am using #define _do_init(bla) \ GST_DEBUG_CATEGORY_INIT (gst_tidmaivideosink_debug, "TIDmaiVideoSink", 0, "TIDmaiVideoSink Element"); GST_BOILERPLATE_FULL (GstTIDmaiVideoSink, gst_tidmaivideosink, GstVideoSink, GST_TYPE_VIDEO_SINK, _do_init); - In my .h code I am doing the following - Define a member GstVideoSink videosink as the first member of my GstTIDmaiVideoSink structure - Define a member GstVideoSinkClass parent_class as the first member of my GstTIDmaiVideoSinkClass structure Lastly, in my makefile I am adding $(GST_BASE_LIBS), $(GST_PLUGINS_BASE_LIBS) and -lgstvideo-0.10 to my LIBADD variable. Even with all of this I don't seem to be getting the lateness property set correctly. I ran the compile of my plugin and stopped after the pre-processor to see what the GST_BOILERPLATE_FULL macro was expanding out to. I see the following output: static void gst_tidmaivideosink_base_init (gpointer g_class); static void gst_tidmaivideosink_class_init (GstTIDmaiVideoSinkClass *g_class); static void gst_tidmaivideosink_init (GstTIDmaiVideoSink *object, GstTIDmaiVideoSinkClass *g_class); static GstVideoSinkClass *parent_class = ((void *)0); static void gst_tidmaivideosink_class_init_trampoline (gpointer g_class, gpointer data) { parent_class = (GstVideoSinkClass *) g_type_class_peek_parent (g_class); gst_tidmaivideosink_class_init ((GstTIDmaiVideoSinkClass *)g_class); } GType gst_tidmaivideosink_get_type (void); GType gst_tidmaivideosink_get_type (void) { static GType object_type = 0; if ((__builtin_expect (__extension__ ({ int _g_boolean_var_; if (object_type == 0) _g_boolean_var_ = 1; else _g_boolean_var_ = 0; _g_boolean_var_; }), 0))) { static const GTypeInfo object_info = { sizeof (GstTIDmaiVideoSinkClass), gst_tidmaivideosink_base_init, ((void *)0), gst_tidmaivideosink_class_init_trampoline, ((void *)0), ((void *)0), sizeof (GstTIDmaiVideoSink), 0, (GInstanceInitFunc) gst_tidmaivideosink_init }; object_type = g_type_register_static ((gst_video_sink_get_type()), "GstTIDmaiVideoSink", &object_info, (GTypeFlags) 0); (void) __extension__ ({ if (gst_tidmaivideosink_debug == ((void *)0)) gst_tidmaivideosink_debug = _gst_debug_category_new ("TIDmaiVideoSink",0,"TIDmaiVideoSink Element"); });; } return object_type; }; Notice that object_type is being set to use gst_video_sink_get_type() for the parent class. Can anyone please tell me what I am missing about inheriting from the VideoSink class and why I don't seem to be getting the default max_lateness behavior that the VideoSink class defines. I will provide full source code if needed but I didn't want to make this e-mail too ugly to begin. Sincerely, Chase Maupin Software Applications Catalog DSP Products e-mail: chase.maupin at ti.com From bilboed at gmail.com Wed Sep 3 17:03:22 2008 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 03 Sep 2008 17:03:22 +0200 Subject: [gst-devel] RELEASE: GStreamer FFmpeg Plug-ins 0.10.5 'This little piggy went to market' Message-ID: <1220454202.7548.3.camel@localhost> This mail announces the release of the GStreamer FFmpeg Plug-ins 0.10.5 'This little piggy went to market'. GStreamer FFmpeg plug-in contains elements using the FFmpeg library code. It contains most popular decoders as well as very fast colorspace conversion elements. For more information, see http://gstreamer.freedesktop.org/modules/gst-ffmpeg.html To file bugs, go to http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-ffmpeg -------------- next part -------------- Release notes for GStreamer FFmpeg Plug-ins??0.10.5 "This little piggy went to market" The GStreamer team is proud to announce a new release in the 0.10.x stable series of the GStreamer FFmpeg Plug-ins. The 0.10.x series is a stable series targeted at end users. It is not API or ABI compatible with the stable 0.8.x series. It is, however, parallel installable with the 0.8.x series. This module contains plug-ins using libraries from the FFmpeg project. Other modules containing plug-ins are: gst-plugins-base contains a basic set of well-supported plug-ins gst-plugins-ugly contains a set of well-supported plug-ins, but might pose problems for distributors gst-plugins-bad contains a set of less supported plug-ins that haven't passed the rigorous quality testing we expect Features of this release * Updated to upstream ffmpeg revision r15004 (28th Aug 2008) * Parallel installability with 0.8.x series * Threadsafe design and API Bugs fixed in this release * 371939 : mov/mp4/m4a/3gp/3g2 muxers create wrong durations * 383420 : [ffmpeg] ISO-derivative muxers don't handle audio correctly * 518705 : Can't play streams from Rai.it * 533708 : broken mpeg-ts typefinding? * 534371 : autogen.sh not dist'ed * 534390 : Patch: use av_picture_copy instead of swscale to copy pic... * 534392 : PATCH: never use ffdec_faad * 534783 : Remove FLV demuxer * 540401 : Garbled sound instead of music * 549799 : all audio codecs claim to support up to 6 channels Download You can find source releases of gst-ffmpeg in the download directory: http://gstreamer.freedesktop.org/src/gst-ffmpeg/ GStreamer Homepage More details can be found on the project's website: http://gstreamer.freedesktop.org/ Support and Bugs We use GNOME's bugzilla for bug reports and feature requests: http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer Developers CVS is hosted on cvs.freedesktop.org. All code is in CVS and can be checked out from there. Interested developers of the core library, plug-ins, and applications should subscribe to the gstreamer-devel list. If there is sufficient interest we will create more lists as necessary. Applications Contributors to this release * Aurelien Grimaud * Edward Hervey * Jan Schmidt * Mark Nauwelaerts * Michael Smith * Sebastian Dr??ge * Thijs Vermeir * Tim-Philipp M??ller * Wim Taymans ?? From chase.maupin at ti.com Wed Sep 3 19:51:27 2008 From: chase.maupin at ti.com (Maupin, Chase) Date: Wed, 3 Sep 2008 12:51:27 -0500 Subject: [gst-devel] Problem inheriting from GstVideoSink class In-Reply-To: <131E5DFBE7373E4C8D813795A6AA7F080127A04F75@dlee06.ent.ti.com> References: <131E5DFBE7373E4C8D813795A6AA7F080127A04F75@dlee06.ent.ti.com> Message-ID: <131E5DFBE7373E4C8D813795A6AA7F080127A0500A@dlee06.ent.ti.com> I just wanted to let everyone know that I figured out my problem. I did not have the sync member in the base sync class set which is why I was not seeing frames dropped. Sincerely, Chase Maupin Software Applications Catalog DSP Products e-mail: chase.maupin at ti.com phone: (281) 274-3285 > -----Original Message----- > From: gstreamer-devel-bounces at lists.sourceforge.net > [mailto:gstreamer-devel-bounces at lists.sourceforge.net] On > Behalf Of Maupin, Chase > Sent: Wednesday, September 03, 2008 9:36 AM > To: gstreamer-devel at lists.sourceforge.net > Subject: [gst-devel] Problem inheriting from GstVideoSink class > > All, > > I am working on writing a video sink and I am trying to > inherit from the GstVideoSink class so that I get > functionality such as dropping frames that are past their > display time. I have a reference video sink which uses this > class and for experimentation purposes I placed a 1 second > delay in the render function. I observed that the display > times in the buffers given to the sink were now 1 second > apart and the video was skipping as I would expect. I put > the same delay in my video sink and I am still seeing time > stamps that are only .033 seconds apart. Basically the > frames that should have been thrown away during the delay are > just getting queued up and arrive in order after the delay. > > I am doing the following to inherit from the GstVideoSink class: > > - In my .c code I am using > #define _do_init(bla) \ > GST_DEBUG_CATEGORY_INIT > (gst_tidmaivideosink_debug, "TIDmaiVideoSink", 0, > "TIDmaiVideoSink Element"); > GST_BOILERPLATE_FULL (GstTIDmaiVideoSink, > gst_tidmaivideosink, GstVideoSink, GST_TYPE_VIDEO_SINK, _do_init); > > - In my .h code I am doing the following > - Define a member GstVideoSink videosink as the first > member of my GstTIDmaiVideoSink structure > - Define a member GstVideoSinkClass parent_class as the > first member of my GstTIDmaiVideoSinkClass structure > > Lastly, in my makefile I am adding $(GST_BASE_LIBS), > $(GST_PLUGINS_BASE_LIBS) and -lgstvideo-0.10 to my LIBADD variable. > > Even with all of this I don't seem to be getting the lateness > property set correctly. I ran the compile of my plugin and > stopped after the pre-processor to see what the > GST_BOILERPLATE_FULL macro was expanding out to. I see the > following output: > > static void gst_tidmaivideosink_base_init (gpointer g_class); > static void gst_tidmaivideosink_class_init > (GstTIDmaiVideoSinkClass *g_class); static void > gst_tidmaivideosink_init (GstTIDmaiVideoSink *object, > GstTIDmaiVideoSinkClass *g_class); static GstVideoSinkClass > *parent_class = ((void *)0); static void > gst_tidmaivideosink_class_init_trampoline (gpointer g_class, > gpointer data) { > parent_class = (GstVideoSinkClass *) > g_type_class_peek_parent (g_class); > gst_tidmaivideosink_class_init ((GstTIDmaiVideoSinkClass > *)g_class); } GType gst_tidmaivideosink_get_type (void); > GType gst_tidmaivideosink_get_type (void) { > static GType object_type = 0; > if ((__builtin_expect (__extension__ ({ int > _g_boolean_var_; if (object_type == 0) _g_boolean_var_ = 1; > else _g_boolean_var_ = 0; _g_boolean_var_; }), 0))) { > static const GTypeInfo object_info = { sizeof > (GstTIDmaiVideoSinkClass), gst_tidmaivideosink_base_init, > ((void *)0), gst_tidmaivideosink_class_init_trampoline, > ((void *)0), ((void *)0), sizeof (GstTIDmaiVideoSink), 0, > (GInstanceInitFunc) gst_tidmaivideosink_init }; > object_type = g_type_register_static > ((gst_video_sink_get_type()), "GstTIDmaiVideoSink", > &object_info, (GTypeFlags) 0); > (void) __extension__ ({ if (gst_tidmaivideosink_debug == > ((void *)0)) gst_tidmaivideosink_debug = > _gst_debug_category_new ("TIDmaiVideoSink",0,"TIDmaiVideoSink > Element"); });; } return object_type; }; > > Notice that object_type is being set to use > gst_video_sink_get_type() for the parent class. > > Can anyone please tell me what I am missing about inheriting > from the VideoSink class and why I don't seem to be getting > the default max_lateness behavior that the VideoSink class > defines. I will provide full source code if needed but I > didn't want to make this e-mail too ugly to begin. > > Sincerely, > Chase Maupin > Software Applications > Catalog DSP Products > e-mail: chase.maupin at ti.com > > > -------------------------------------------------------------- > ----------- > This SF.Net email is sponsored by the Moblin Your Move > Developer's challenge Build the coolest Linux based > applications with Moblin SDK & win great prizes Grand prize > is a trip for two to an Open Source event anywhere in the > world http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From ole.andre.ravnas at tandberg.com Thu Sep 4 02:52:32 2008 From: ole.andre.ravnas at tandberg.com (=?iso-8859-1?Q?Ole_Andre_Vadla_Ravn=E5s?=) Date: Thu, 4 Sep 2008 02:52:32 +0200 Subject: [gst-devel] Freshly baked Windows binaries of HEAD available Message-ID: Hi all, I just whipped together some binaries of GStreamer HEAD built against msvcrt.dll (the ReleaseWdkCrt configuration). The snapshots can be found here: http://people.collabora.co.uk/~oleavr/OABuild/snapshots/ 33e44c77f4b8f413db6ffc5e47f2a8ab OABuild-20080904-glib-dbg.exe 0f717766f27fab1b4905a28123a195cc OABuild-20080904-glib-dev.exe 708f0ca2d3b5934c2d1ad91ea3713e2e OABuild-20080904-glib.exe ffaff7b80693d76f1476c6ab809bbe75 OABuild-20080904-gstreamer-dbg.exe c02d961fd4f80900fa338a3049731b8b OABuild-20080904-gstreamer-dev.exe 8e29f516806bddc46a5435898e3cf204 OABuild-20080904-gstreamer.exe Please note that this was done in a hurry, so the naming scheme, which files go where, etc., will likely change in the future. Extract OABuild-20080904-glib.exe and OABuild-20080904-gstreamer.exe to get a pure runtime installation (with gst-inspect and gst-launch, likely to be split out in the future), dbg for debug symbols (.pdb files), and dev for development headers, .lib files and stuff like that. Also note that I just realized that the binaries from gst-plugins-good are off by one micro version as I forgot to sync GstPluginsGoodVersion.vsprops against configure.ac... This is on my TODO list of things to automate, and will obviously be fixed in future snapshots. Lastly, if someone feels like contributing automated .msi/.msm packaging for future snapshots then that would be awesome! :) Cheers, Ole Andr? From nicolas.m.zhang at gmail.com Thu Sep 4 03:30:02 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Thu, 4 Sep 2008 09:30:02 +0800 Subject: [gst-devel] add latency to audio in gstrtpbin In-Reply-To: <48BDB249.1040009@sat.qc.ca> References: <48BDB249.1040009@sat.qc.ca> Message-ID: Hi, Tristan: You should not adjust video/audio latency manually because RTP provides a mechanism to accomplish this, called `lip-synchronization'. Refer to RFC 3550 or book `RTP: Video and Audio for the Internet' for more details. These will help you a lot. Eric Zhang 2008/9/3 Tristan Matthews > Hi, > > If I have a pipeline using gstrtpbin (similar to the example in the > documentation) to send audio and video, what is the best/most reliable > way of adding latency to the audio? Would gst_event_new_latency work > (and if so, how), or am I missing its intent: > > http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/gstreamer/html/gstreamer-GstEvent.html#gst-event-new-latency > > Basically my concern is that if video capture is too slow, can I > manually adjust the audio latency to match. > > Best, > > Tristan > > -- > Tristan Matthews > Soci?t? des arts technologiques [SAT] > email: tristan at sat.qc.ca > web: http://www.music.mcgill.ca/~tmatthews > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From manish.rana at gmail.com Thu Sep 4 11:22:31 2008 From: manish.rana at gmail.com (Manish Rana) Date: Thu, 4 Sep 2008 14:52:31 +0530 Subject: [gst-devel] ffdec_mpeg4 Bitrate Issue.... In-Reply-To: <8c192ddd0809020631h1f0d3abboe183c9984c993aa2@mail.gmail.com> References: <8c192ddd0809020631h1f0d3abboe183c9984c993aa2@mail.gmail.com> Message-ID: <8c192ddd0809040222k771ac89cq2fb4d77b8e3c1e1@mail.gmail.com> On Tue, Sep 2, 2008 at 7:01 PM, Manish Rana wrote: > Hi All, > > I am trying to set bitrate=48000 on ffdec_mpeg4, and then transmitting the > data using gstRTPbin and UDP. > On the other end i tried calculating the data rate coming. I used WireShark > for the same. > > After removing RTP/UDP/eth0 and all other header's the data rate is coming > around 200kbps. > > FPS at the VideoSrc is 10. GOP size is 30. > > Can someone guide me what do i need to get the lower and constant bit rate > from encoder. > As this is eating all the bandwidth. and all my application is going for > toss :"( > > Also i have seen there are some properties of ffenc_mpeg4 that are used for > Rate Control. > > Please let me know. Also suggest to me to reduce overhead and ways to > reduce the bitrate in RTP. > > I am in urgent need, > > Thanks a lot > Manish > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liangzhihong1984 at 126.com Thu Sep 4 14:36:28 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Thu, 4 Sep 2008 20:36:28 +0800 (CST) Subject: [gst-devel] Can GStreamer work with Directshow? Message-ID: <3122839.910061220531788865.JavaMail.coremail@bj126app68.126.com> I mean, use DirectShow to send video data to a linux Machine, can I receive, decode and display it with GStreamer on this machine? -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Thu Sep 4 15:01:24 2008 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 04 Sep 2008 15:01:24 +0200 Subject: [gst-devel] Can GStreamer work with Directshow? In-Reply-To: <3122839.910061220531788865.JavaMail.coremail@bj126app68.126.com> References: <3122839.910061220531788865.JavaMail.coremail@bj126app68.126.com> Message-ID: <1220533284.2510.0.camel@localhost> Can you be more specific ? What format ? What codec ? What data transport (rtp, http, ...) ? Edward On Thu, 2008-09-04 at 20:36 +0800, liangzhihong1984 wrote: > I mean, use DirectShow to send video data to a linux Machine, can I > receive, decode and display it with GStreamer on this machine? > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From tristan at sat.qc.ca Thu Sep 4 17:34:42 2008 From: tristan at sat.qc.ca (Tristan Matthews) Date: Thu, 04 Sep 2008 11:34:42 -0400 Subject: [gst-devel] add latency to audio in gstrtpbin In-Reply-To: References: <48BDB249.1040009@sat.qc.ca> Message-ID: <48C00012.8040204@sat.qc.ca> Thanks for the feedback Eric. I had seen the RFC before but that book looks pretty useful as well. My concern isn't that the AV will get out of sync in gstreamer/gstrtpbin, but rather that it will be out of sync immediately at the capture stage, i.e. if i have video input from a camera with more latency than a separate audio source. An example scenario is given here: http://chris.pirillo.com/2007/07/11/audio-video-capture/ where you have video of someone clapping and you have to adjust the delay to make the sound and image of the clap line up. I would need to be able to manually adjust the latency to ensure the video and audio are sync'd. Should I just put the audio through a Ladspa-delay or is their a better solution in gstreamer? -T Eric Zhang wrote: > Hi, Tristan: > > You should not adjust video/audio latency manually because RTP > provides a mechanism to accomplish this, called `lip-synchronization'. > Refer to RFC 3550 or book `RTP: Video and Audio for the Internet' for > more details. These will help you a lot. > > Eric Zhang > > 2008/9/3 Tristan Matthews > > > Hi, > > If I have a pipeline using gstrtpbin (similar to the example in the > documentation) to send audio and video, what is the best/most reliable > way of adding latency to the audio? Would gst_event_new_latency work > (and if so, how), or am I missing its intent: > http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/gstreamer/html/gstreamer-GstEvent.html#gst-event-new-latency > > Basically my concern is that if video capture is too slow, can I > manually adjust the audio latency to match. > > Best, > > Tristan > > -- > Tristan Matthews > Soci?t? des arts technologiques [SAT] > email: tristan at sat.qc.ca > web: http://www.music.mcgill.ca/~tmatthews > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win > great prizes > Grand prize is a trip for two to an Open Source event anywhere in > the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Tristan Matthews Soci?t? des arts technologiques [SAT] email: tristan at sat.qc.ca web: http://www.music.mcgill.ca/~tmatthews From wim.taymans at gmail.com Thu Sep 4 18:54:36 2008 From: wim.taymans at gmail.com (Wim Taymans) Date: Thu, 04 Sep 2008 18:54:36 +0200 Subject: [gst-devel] add latency to audio in gstrtpbin In-Reply-To: <48C00012.8040204@sat.qc.ca> References: <48BDB249.1040009@sat.qc.ca> <48C00012.8040204@sat.qc.ca> Message-ID: <1220547276.6781.96.camel@metal> On Thu, 2008-09-04 at 11:34 -0400, Tristan Matthews wrote: > Thanks for the feedback Eric. I had seen the RFC before but that book > looks pretty useful as well. > My concern isn't that the AV will get out of sync in > gstreamer/gstrtpbin, but rather that it will be out of sync immediately > at the capture stage, i.e. if i have video input from a camera with more > latency than a separate audio source. An example scenario is given here: > http://chris.pirillo.com/2007/07/11/audio-video-capture/ > where you have video of someone clapping and you have to adjust the > delay to make the sound and image of the clap line up. > I would need to be able to manually adjust the latency to ensure the > video and audio are sync'd. Should I just put the audio through a > Ladspa-delay or is their a better solution in gstreamer? > This situation can not occur in gstreamer when you have a well written source element that generates correct timestamps and reports its latency correctly. You could write an element that applies an offset to all timestamps on the buffers. Wim > -T > > Eric Zhang wrote: > > Hi, Tristan: > > > > You should not adjust video/audio latency manually because RTP > > provides a mechanism to accomplish this, called `lip-synchronization'. > > Refer to RFC 3550 or book `RTP: Video and Audio for the Internet' for > > more details. These will help you a lot. > > > > Eric Zhang > > > > 2008/9/3 Tristan Matthews > > > > > Hi, > > > > If I have a pipeline using gstrtpbin (similar to the example in the > > documentation) to send audio and video, what is the best/most reliable > > way of adding latency to the audio? Would gst_event_new_latency work > > (and if so, how), or am I missing its intent: > > http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/gstreamer/html/gstreamer-GstEvent.html#gst-event-new-latency > > > > Basically my concern is that if video capture is too slow, can I > > manually adjust the audio latency to match. > > > > Best, > > > > Tristan > > > > -- > > Tristan Matthews > > Soci?t? des arts technologiques [SAT] > > email: tristan at sat.qc.ca > > web: http://www.music.mcgill.ca/~tmatthews > > > > > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > > challenge > > Build the coolest Linux based applications with Moblin SDK & win > > great prizes > > Grand prize is a trip for two to an Open Source event anywhere in > > the world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > ------------------------------------------------------------------------ > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > > Build the coolest Linux based applications with Moblin SDK & win great prizes > > Grand prize is a trip for two to an Open Source event anywhere in the world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > From thiagossantos at gmail.com Thu Sep 4 19:11:51 2008 From: thiagossantos at gmail.com (thiagoss) Date: Thu, 4 Sep 2008 14:11:51 -0300 Subject: [gst-devel] add latency to audio in gstrtpbin In-Reply-To: <1220547276.6781.96.camel@metal> References: <48BDB249.1040009@sat.qc.ca> <48C00012.8040204@sat.qc.ca> <1220547276.6781.96.camel@metal> Message-ID: On Thu, Sep 4, 2008 at 1:54 PM, Wim Taymans wrote: > On Thu, 2008-09-04 at 11:34 -0400, Tristan Matthews wrote: > > Thanks for the feedback Eric. I had seen the RFC before but that book > > looks pretty useful as well. > > My concern isn't that the AV will get out of sync in > > gstreamer/gstrtpbin, but rather that it will be out of sync immediately > > at the capture stage, i.e. if i have video input from a camera with more > > latency than a separate audio source. An example scenario is given here: > > http://chris.pirillo.com/2007/07/11/audio-video-capture/ > > where you have video of someone clapping and you have to adjust the > > delay to make the sound and image of the clap line up. > > I would need to be able to manually adjust the latency to ensure the > > video and audio are sync'd. Should I just put the audio through a > > Ladspa-delay or is their a better solution in gstreamer? > > > > This situation can not occur in gstreamer when you have a well written > source element that generates correct timestamps and reports its latency > correctly. You could write an element that applies an offset to all > timestamps on the buffers. > > Wim > AFAIK, This element already exists and is called GstShift from gentrans plugins > > > > -T > > > > Eric Zhang wrote: > > > Hi, Tristan: > > > > > > You should not adjust video/audio latency manually because RTP > > > provides a mechanism to accomplish this, called `lip-synchronization'. > > > Refer to RFC 3550 or book `RTP: Video and Audio for the Internet' for > > > more details. These will help you a lot. > > > > > > Eric Zhang > > > > > > 2008/9/3 Tristan Matthews >> > > > > > > Hi, > > > > > > If I have a pipeline using gstrtpbin (similar to the example in the > > > documentation) to send audio and video, what is the best/most > reliable > > > way of adding latency to the audio? Would gst_event_new_latency > work > > > (and if so, how), or am I missing its intent: > > > > http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/gstreamer/html/gstreamer-GstEvent.html#gst-event-new-latency > > > > > > Basically my concern is that if video capture is too slow, can I > > > manually adjust the audio latency to match. > > > > > > Best, > > > > > > Tristan > > > > > > -- > > > Tristan Matthews > > > Soci?t? des arts technologiques [SAT] > > > email: tristan at sat.qc.ca > > > web: http://www.music.mcgill.ca/~tmatthews > > > > > > > > > > > > > ------------------------------------------------------------------------- > > > This SF.Net email is sponsored by the Moblin Your Move Developer's > > > challenge > > > Build the coolest Linux based applications with Moblin SDK & win > > > great prizes > > > Grand prize is a trip for two to an Open Source event anywhere in > > > the world > > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > > ------------------------------------------------------------------------- > > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) -------------- next part -------------- An HTML attachment was scrubbed... URL: From nicolas.m.zhang at gmail.com Fri Sep 5 03:16:53 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Fri, 5 Sep 2008 09:16:53 +0800 Subject: [gst-devel] add latency to audio in gstrtpbin In-Reply-To: <1220547276.6781.96.camel@metal> References: <48BDB249.1040009@sat.qc.ca> <48C00012.8040204@sat.qc.ca> <1220547276.6781.96.camel@metal> Message-ID: Hi, Tristan: I agree with Wim. According to your requirement and problems, this is not the gstreamer or the network makes the audio/video out of sync. So you should apply different latency to your video and audio. Try to set `latency' property to your video and audio RTP stream or just write a plugin/probe to adjust the timestamp manually. Eric 2008/9/5 Wim Taymans > On Thu, 2008-09-04 at 11:34 -0400, Tristan Matthews wrote: > > Thanks for the feedback Eric. I had seen the RFC before but that book > > looks pretty useful as well. > > My concern isn't that the AV will get out of sync in > > gstreamer/gstrtpbin, but rather that it will be out of sync immediately > > at the capture stage, i.e. if i have video input from a camera with more > > latency than a separate audio source. An example scenario is given here: > > http://chris.pirillo.com/2007/07/11/audio-video-capture/ > > where you have video of someone clapping and you have to adjust the > > delay to make the sound and image of the clap line up. > > I would need to be able to manually adjust the latency to ensure the > > video and audio are sync'd. Should I just put the audio through a > > Ladspa-delay or is their a better solution in gstreamer? > > > > This situation can not occur in gstreamer when you have a well written > source element that generates correct timestamps and reports its latency > correctly. You could write an element that applies an offset to all > timestamps on the buffers. > > Wim > > > > -T > > > > Eric Zhang wrote: > > > Hi, Tristan: > > > > > > You should not adjust video/audio latency manually because RTP > > > provides a mechanism to accomplish this, called `lip-synchronization'. > > > Refer to RFC 3550 or book `RTP: Video and Audio for the Internet' for > > > more details. These will help you a lot. > > > > > > Eric Zhang > > > > > > 2008/9/3 Tristan Matthews >> > > > > > > Hi, > > > > > > If I have a pipeline using gstrtpbin (similar to the example in the > > > documentation) to send audio and video, what is the best/most > reliable > > > way of adding latency to the audio? Would gst_event_new_latency > work > > > (and if so, how), or am I missing its intent: > > > > http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/gstreamer/html/gstreamer-GstEvent.html#gst-event-new-latency > > > > > > Basically my concern is that if video capture is too slow, can I > > > manually adjust the audio latency to match. > > > > > > Best, > > > > > > Tristan > > > > > > -- > > > Tristan Matthews > > > Soci?t? des arts technologiques [SAT] > > > email: tristan at sat.qc.ca > > > web: http://www.music.mcgill.ca/~tmatthews > > > > > > > > > > > > > ------------------------------------------------------------------------- > > > This SF.Net email is sponsored by the Moblin Your Move Developer's > > > challenge > > > Build the coolest Linux based applications with Moblin SDK & win > > > great prizes > > > Grand prize is a trip for two to an Open Source event anywhere in > > > the world > > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > > ------------------------------------------------------------------------- > > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liangzhihong1984 at 126.com Fri Sep 5 03:58:14 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Fri, 5 Sep 2008 09:58:14 +0800 (CST) Subject: [gst-devel] Can GStreamer work with Directshow? Message-ID: <28595195.79471220579894887.JavaMail.coremail@bj126app68.126.com> For example, the Windows client grab video data from HDV device through 1394 firewire, codec is MPEG2-TS, then stream it. All above is done with DirectShow. The receiver client is a Linux machine and use GStreamer.My question is can GStreamer work in order? -------------- next part -------------- An HTML attachment was scrubbed... URL: From sachinpandhare at gmail.com Fri Sep 5 08:28:46 2008 From: sachinpandhare at gmail.com (Sachin Pandhare) Date: Fri, 5 Sep 2008 11:58:46 +0530 Subject: [gst-devel] libamrwb and gst-bad plugin 0.10.8 Message-ID: <72cf309c0809042328j7b202241hb93b4cc048610b74@mail.gmail.com> hi, i am working with gst-bad plugin 0.10.8. i downloaded amrwb-7.0.0.3.tar.bz2 from http://www.penguin.cz/~utx/amr. this needs codec sources from 3gp project. as i didn't have net connection i downloaded them from http://www.3gpp.org/FTP/Specs/archive/26_series/26.204/ . i have observed that gst-bad plugin 0.10.8 is not compatible with latest 26204-710.zip package at 3gpp. bad plugins searches for olders interface 3GPD_IF.... whereas latest package has D_IF.... has anybody already faced this problem? thanks. Sachin -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Fri Sep 5 11:25:50 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Fri, 05 Sep 2008 12:25:50 +0300 Subject: [gst-devel] aac parser In-Reply-To: References: <72cf309c0809020419w17e87d85yd86165633f682471@mail.gmail.com> <48BD3E7E.9060000@hora-obscura.de> Message-ID: <48C0FB1E.4070201@hora-obscura.de> Hi, Zheng, Huan schrieb: > Stefan > Could you please elaborate on this "Soon"? One month or several months? :) > Hopefully next week. > It would be very nice to have an aac parser in gstreamer. > And one more question: Is this parser able to parse ADIF stream into single frames? Because what I heard now is that ADIF has only one header at the beginning, and the offset of each frame can not be located unless you have finished decoding. > lets discuss this once its published. Stefan > Thanks! > > Best Regards, Zheng, Huan(ZBT) > OTC/SSD/SSG > Intel Aisa-Pacific Research & Developement Ltd > Tel: 021-6116 6435 > Inet: 8821 6435 > Cub: 3W035 > -----Original Message----- > From: gstreamer-devel-bounces at lists.sourceforge.net [mailto:gstreamer-devel-bounces at lists.sourceforge.net] On Behalf Of Stefan Kost > Sent: 2008?9?2? 21:24 > To: Discussion of the development of GStreamer > Subject: Re: [gst-devel] aac parser > > hi, > > there will be one in gst-plugin-bad soon. > > Stefan > > Sachin Pandhare schrieb: > >> Hi, >> if aac parser needs to be developed can we take some plugin code as a >> reference and which one will be a suitable candidate for this? >> thanks, >> Sachin >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's challenge >> Build the coolest Linux based applications with Moblin SDK & win great prizes >> Grand prize is a trip for two to an Open Source event anywhere in the world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From bilboed at gmail.com Fri Sep 5 12:01:38 2008 From: bilboed at gmail.com (Edward Hervey) Date: Fri, 05 Sep 2008 12:01:38 +0200 Subject: [gst-devel] Can GStreamer work with Directshow? In-Reply-To: <28595195.79471220579894887.JavaMail.coremail@bj126app68.126.com> References: <28595195.79471220579894887.JavaMail.coremail@bj126app68.126.com> Message-ID: <1220608898.20795.2.camel@localhost> Hi, Give us information... on what DirectShow IS SENDING OVER THE NETWORK ! What format it's sending, what codec it's sending, etc.... If you want us to tell you if GStreamer can decode some stream, we're only interested in what GStreamer will receive. We're not magicians who can guess what format you'll be streaming in. Edward On Fri, 2008-09-05 at 09:58 +0800, liangzhihong1984 wrote: > > > For example, the Windows client grab video data from HDV device > through 1394 firewire, codec is MPEG2-TS, then stream it. All above is > done with DirectShow. The receiver client is a Linux machine and use > GStreamer.My question is can GStreamer work in order? > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ensonic at hora-obscura.de Fri Sep 5 13:04:47 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Fri, 05 Sep 2008 14:04:47 +0300 Subject: [gst-devel] libamrwb and gst-bad plugin 0.10.8 In-Reply-To: <72cf309c0809042328j7b202241hb93b4cc048610b74@mail.gmail.com> References: <72cf309c0809042328j7b202241hb93b4cc048610b74@mail.gmail.com> Message-ID: <48C1124F.2050904@hora-obscura.de> hi, Sachin Pandhare schrieb: > hi, > i am working with gst-bad plugin 0.10.8. > i downloaded amrwb-7.0.0.3.tar.bz2 from http://www.penguin.cz/~utx/amr > . > this needs codec sources from 3gp project. as i didn't have net > connection i downloaded them from > http://www.3gpp.org/FTP/Specs/archive/26_series/26.204/. > > i have observed that gst-bad plugin 0.10.8 is not compatible with > latest 26204-710.zip package at 3gpp. bad plugins searches for olders > interface 3GPD_IF.... whereas latest package has D_IF.... I had installed it in the past. Please open a bug for the issue. Would be great if you can provide a patch as well. If its possible it would check for the version the user has installed and use the diffrent defines then. Stefan > > has anybody already faced this problem? > > thanks. > Sachin > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From alzhang03 at hotmail.com Fri Sep 5 14:50:53 2008 From: alzhang03 at hotmail.com (zhangal) Date: Fri, 5 Sep 2008 20:50:53 +0800 Subject: [gst-devel] How can I switch audio track when play back a multi-audio-track movie? Message-ID: Hi, I met some movies(like *.mkv and *.avi files) which include multi audio tracks and then my application can not switch audio when playback. I tried to just unlink the demuxer and the audio decoder and then called gst-element-get-pad() to get another audio pad which will be linked to the audio decoder that is able to decode the audio. But I can not hear anything. The code looks like following GstPad *src_pad, *sink_pad, *demux_src; FAIL_CHECK (NULL == (demux_src = gst_element_get_pad(demuxer,"audio_01")));//Get anotherdemuxer source padFAIL_CHECK (NULL == (sink_pad = gst_element_get_pad(a_decoder, "sink")));// Get audio decoder sink pad src_pad = gst_pad_get_peer(sink_pad);//get the audio pad connected to decoder sink padgst_element_set_state(player_status->bin, GST_STATE_PAUSED); gst_pad_unlink (src_pad, sink_pad);gst_pad_link(demux_src, sink_pad);gst_object_unref(src_pad); gst_object_unref(sink_pad); gst_object_unref(demux_src);gst_element_set_state(player_status->bin, GST_STATE_PLAYING);//End It seems that does not work well. Does anyone have the idea about this? Thanks a lot _________________________________________________________________ ?????MSN?????????? http://im.live.cn/click/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos.cirjan at gmail.com Sun Sep 7 01:21:15 2008 From: dragos.cirjan at gmail.com (Dragos Cirjan) Date: Sun, 7 Sep 2008 02:21:15 +0300 Subject: [gst-devel] need fast fast help Message-ID: <2f0090140809061621p13c0f1c5k6ebcf19320d3ca6b@mail.gmail.com> Guys, I really need some fast help. I need to enter on option of repeat-read into soup plugin, so the source should be read repeatedly. (I'm thinking of the Video over IP cameras who have only JPEG and not motion JPEG streams). In the same time, please take a look at this: http://dor.homelinux.com/gstcurlsrc.tgz. I get a very ugly glibc error telling me I free memory twice (double free or corruption). Started writing curlsrc because I couldn't modify soup plugin :(. Now I want to finish it whatever the result would be for soup. Thanks in advance. Chris P.S. There is a writen plugin with CURL, but it for gst-0.8. I need it for gst-0.10 :(. -- ----------------------------------------------------------------- Cristian - Dragos, Cirjan ----------------------------------------------------------------- Email: dragos.cirjan at yahoo.com Email: dragos.cirjan at itmediaconnect.ro, doru at bocancul-literar.ro Telefon: +40726355762 -------------- next part -------------- An HTML attachment was scrubbed... URL: From chhail at mail2.sysu.edu.cn Sun Sep 7 05:39:57 2008 From: chhail at mail2.sysu.edu.cn (Chen Hailiang) Date: Sun, 7 Sep 2008 11:39:57 +0800 Subject: [gst-devel] GStreamer beginer problem Message-ID: <20080907031708.M69442@mail2.sysu.edu.cn> Hi, I'm beginning to learn GStreamer this day.there is some problem. my linux is Kubuntu8.04.When the GStreamer installation completed,I checked it as the FAQ 5 said. It's OK.So I think my installation was correct. I tried to run the example in the application development manual(5.4). I used command "gcc -Wall $(pkg-config --cflags --libs gstreamer-0.10) xxx.c -o xxx" to compile it,and it return me that "undefined reference to `gst_element_factory_get_kalss'"compiled failed. my Gstreamer(version: 0.10) is in /usr/include/gstreamer-0.10/gst/ the Gstreamer lib is in /usr/lib/gstreamer-0.10/ I got this "undefined reference" error in using some other methods.like gst_element_unref() etc. is there something wrong with my installation? waiting for help.thinks! snYe -- Best regards From irfanshaikh at tataelxsi.co.in Sun Sep 7 10:06:42 2008 From: irfanshaikh at tataelxsi.co.in (Irfan Shaikh) Date: Sun, 7 Sep 2008 13:36:42 +0530 Subject: [gst-devel] Can .ASF be streamed using gstreamer pipeline? References: <2f0090140809061621p13c0f1c5k6ebcf19320d3ca6b@mail.gmail.com> Message-ID: <9D5E1752379A43408015F7FE98466115782748@CHNEXVS01.VSNLXCHANGE.COM> Hi guyz I m very new to Gstreamer. Can i know files with .ASF container format can be streamed using Gstreamer on a VLC player running on client PC. To my understanding I have tried a lot using the following pipeline using udp sink. gst-launch filesrc location = /root/Desktop/abc.asf ! udpsink host=10.60.3.78 port=4951 Please can any one help me... Also I m successfully able to stream audio file but i m unable to stream video files correctly using udpsink.... If possible plz mention some sample pipeline examples for playin video file, Thanks in advance......... Regards, Irfan. This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Sun Sep 7 11:24:56 2008 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sun, 07 Sep 2008 10:24:56 +0100 Subject: [gst-devel] GStreamer beginer problem In-Reply-To: <20080907031708.M69442@mail2.sysu.edu.cn> References: <20080907031708.M69442@mail2.sysu.edu.cn> Message-ID: <1220779496.13056.3.camel@mini.centricular.net> On Sun, 2008-09-07 at 11:39 +0800, Chen Hailiang wrote: Hi, > I tried to run the example in the application development manual(5.4). > I used command "gcc -Wall $(pkg-config --cflags --libs gstreamer-0.10) > xxx.c -o xxx" to compile it,and it return me that "undefined reference to > `gst_element_factory_get_kalss'"compiled failed. > my Gstreamer(version: 0.10) is in /usr/include/gstreamer-0.10/gst/ > the Gstreamer lib is in /usr/lib/gstreamer-0.10/ > I got this "undefined reference" error in using some other methods.like > gst_element_unref() etc. is there something wrong with my installation? > waiting for help.thinks! It sounds like you're reading a really old version of the application development manual which refers to historic GStreamer versions. The lastest version can be found here: http://gstreamer.freedesktop.org/documentation/ Cheers -Tim From timovwb at gmail.com Sun Sep 7 15:29:11 2008 From: timovwb at gmail.com (Timo) Date: Sun, 07 Sep 2008 15:29:11 +0200 Subject: [gst-devel] Visualization with Python Message-ID: <48c3d722.0309d00a.1388.ffffa25d@mx.google.com> Hello, I'm looking for a way to execute commands when music changes. Like visualizations in music players, but instead of changing weird lines on the screen, I want to do things. Now I found out about playbin, but I can't get this to work, and can't find that much documentation about the subject with Python. This is a bit that I could figure out, but can't get any further: pipeline = gst.element_factory_make('playbin', 'playbin') vis = gst.element_factory_make("goom") pipeline.set_property('vis-plugin', vis) And if this is possible, can this also be done without writing a plugin for a music player? But just use my own program and use the soundcard or something to hear the sounds? This way I don't have to write a plugin for a couple of players but just use my own GUI. Thanks in advance, Timo From bilboed at gmail.com Sun Sep 7 15:34:18 2008 From: bilboed at gmail.com (Edward Hervey) Date: Sun, 07 Sep 2008 15:34:18 +0200 Subject: [gst-devel] GIT test repositories Message-ID: <1220794458.2541.18.camel@putamadre> Hi all, Long time without any git updates, sincere apologies about that. There was some confusion/delays between Tim and myself regarding the conversion :( I've finally done a test conversion of all main modules from the state they were in 2 days ago. You can find them here under my personal repository section (~bilboed/): http://cgit.freedesktop.org/ These repositories include the full history contained in cvs (first commit is January 2000 by omega). I did a few fixups during the conversion, namely: * Using common as a git submodule git submodules are the natural way to share a module between several others. The problem was that we want to keep the coherence between checkouts of the parent module and common, so that if you check out a revision of core from a year ago, you will end up with the revision of common which was used at that time. All converted git modules have therefore been re-parsed to contain those updates and the initial .gitmodules file that contains the link to which submodule to check out and where (currently pointing to where I stored the common repository). > git checkout > git submodule update # you will end up with common being in the state it was when # was done. * Renaming .cvsignore to .gitignore Those special files have been renamed throughout the whole history of the modules. So you should end up with the same behaviour * Attributing authorship to patch authors GIT makes a difference between the person who committed a patch and the author of that patch. I parsed all the commit messages to extract (as much as I could) the author of the patches to set that properly. Example here : http://cgit.freedesktop.org/~bilboed/common/commit/?id=80627bb8053f6a10f28015e834778b098fc9c391 So at this point... we need testing ! Check out the modules, compile them, check out various revisions, report issues, etc... Final word of caution : hopefully these repositories will not require any fixups, but DO NOT ASSUME that these will be the final official repositories. If we have to do some more fixups, it will change the hash of the revisions and the work/branches you based on these repositories will be lost (not quite lost, but it'll be painful to rebase your work against the new repositories). To sum up: *** do not use these repositories for production use yet ! *** ... but please test them all the same :) Edward From chhail at mail2.sysu.edu.cn Mon Sep 8 02:36:44 2008 From: chhail at mail2.sysu.edu.cn (Chen Hailiang) Date: Mon, 8 Sep 2008 08:36:44 +0800 Subject: [gst-devel] GStreamer beginer problem In-Reply-To: <1220779496.13056.3.camel@mini.centricular.net> References: <20080907031708.M69442@mail2.sysu.edu.cn> <1220779496.13056.3.camel@mini.centricular.net> Message-ID: <20080908002049.M35587@mail2.sysu.edu.cn> thinks Tim, I got the lastest version now,and found some unuseable methods has made up, such like gst_element_unref() was not used again. But I still can't use 'gst_element_factory_get_klass()', (error:elementInfo.c:(.text+0x60): undefined reference to `gst_element_factory_get_kalss'). my source codes were copied from <5.4.1. Getting information about an element using a factory>,the newest version of the application development manaul(0.10.20.1).may any other reason to bring about that? think you. On Sun, 07 Sep 2008 10:24:56 +0100, Tim-Philipp M?ller wrote > On Sun, 2008-09-07 at 11:39 +0800, Chen Hailiang wrote: > > Hi, > > > I tried to run the example in the application development manual(5.4). > > I used command "gcc -Wall $(pkg-config --cflags --libs gstreamer-0.10) > > xxx.c -o xxx" to compile it,and it return me that "undefined reference to > > `gst_element_factory_get_kalss'"compiled failed. > > my Gstreamer(version: 0.10) is in /usr/include/gstreamer-0.10/gst/ > > the Gstreamer lib is in /usr/lib/gstreamer-0.10/ > > I got this "undefined reference" error in using some other methods.like > > gst_element_unref() etc. is there something wrong with my installation? > > waiting for help.thinks! > > It sounds like you're reading a really old version of the application > development manual which refers to historic GStreamer versions. The > lastest version can be found here: > > http://gstreamer.freedesktop.org/documentation/ > > Cheers > -Tim > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win > great prizes Grand prize is a trip for two to an Open Source event > anywhere in the world http://moblin-contest.org/redirect.php? banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Best regards From nicolas.m.zhang at gmail.com Mon Sep 8 03:11:51 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Mon, 8 Sep 2008 09:11:51 +0800 Subject: [gst-devel] How can I switch audio track when play back a multi-audio-track movie? In-Reply-To: References: Message-ID: Hi, zhangal: Maybe the element `selector' can do some help for you. Eric Zhang 2008/9/5 zhangal > Hi, > > I met some movies(like *.mkv and *.avi files) which include multi audio > tracks and then my application can not switch audio when playback. > > I tried to just unlink the demuxer and the audio decoder and then > called gst-element-get-pad() to get another audio pad which will be linked > to the audio decoder that is able to decode the audio. But I can not hear > anything. The code looks like following > > GstPad *src_pad, *sink_pad, *demux_src; > > FAIL_CHECK (NULL == (demux_src = > gst_element_get_pad(demuxer,"audio_01")));//Get anotherdemuxer source pad > FAIL_CHECK (NULL == (sink_pad = gst_element_get_pad(a_decoder, > "sink")));// Get audio decoder sink pad > > src_pad = gst_pad_get_peer(sink_pad);//get the audio pad connected to > decoder sink pad > gst_element_set_state(player_status->bin, GST_STATE_PAUSED); > > gst_pad_unli nk (src_pad, sink_pad); > > gst_pad_link(demux_src, sink_pad); > > gst_object_unref(src_pad); > gst_object_unref(sink_pad); > gst_object_unref(demux_src); > > gst_element_set_state(player_status->bin, GST_STATE_PLAYING); > > //End > > It seems that does not work well. Does anyone have the idea about this? > > Thanks a lot > > > > ------------------------------ > MSN????????????? ??????? > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nicolas.m.zhang at gmail.com Mon Sep 8 03:19:09 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Mon, 8 Sep 2008 09:19:09 +0800 Subject: [gst-devel] Visualization with Python In-Reply-To: <48c3d722.0309d00a.1388.ffffa25d@mx.google.com> References: <48c3d722.0309d00a.1388.ffffa25d@mx.google.com> Message-ID: Hi, Timo: If you don't mind trying playbin2, I know that it will emit signal `about-to-finish' while changing the playing stuffs to next. You can connect this signal and do works you want. Playbin hasn't this signal yet. Refer to /gst/playback/gstplaybin2.c for more details. Eric 2008/9/7 Timo > Hello, I'm looking for a way to execute commands when music changes. > Like visualizations in music players, but instead of changing weird > lines on the screen, I want to do things. > Now I found out about playbin, but I can't get this to work, and can't > find that much documentation about the subject with Python. > This is a bit that I could figure out, but can't get any further: > pipeline = gst.element_factory_make('playbin', 'playbin') > vis = gst.element_factory_make("goom") > pipeline.set_property('vis-plugin', vis) > > > And if this is possible, can this also be done without writing a plugin > for a music player? But just use my own program and use the soundcard or > something to hear the sounds? This way I don't have to write a plugin > for a couple of players but just use my own GUI. > > > Thanks in advance, > > Timo > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nicolas.m.zhang at gmail.com Mon Sep 8 03:22:26 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Mon, 8 Sep 2008 09:22:26 +0800 Subject: [gst-devel] GStreamer beginer problem In-Reply-To: <20080908002049.M35587@mail2.sysu.edu.cn> References: <20080907031708.M69442@mail2.sysu.edu.cn> <1220779496.13056.3.camel@mini.centricular.net> <20080908002049.M35587@mail2.sysu.edu.cn> Message-ID: Hi, Chen: `gst_element_factory_get_kalss'? Is this a spelling mistake? Eric 2008/9/8 Chen Hailiang > thinks Tim, I got the lastest version now,and found some unuseable methods > has made up, such like gst_element_unref() was not used again. But I still > can't use 'gst_element_factory_get_klass()', > (error:elementInfo.c:(.text+0x60): undefined reference to > `gst_element_factory_get_kalss'). > my source codes were copied from <5.4.1. Getting information about an > element > using a factory>,the newest version of the application development > manaul(0.10.20.1).may any other reason to bring about that? > think you. > > On Sun, 07 Sep 2008 10:24:56 +0100, Tim-Philipp M?ller wrote > > On Sun, 2008-09-07 at 11:39 +0800, Chen Hailiang wrote: > > > > Hi, > > > > > I tried to run the example in the application development > manual(5.4). > > > I used command "gcc -Wall $(pkg-config --cflags --libs > gstreamer-0.10) > > > xxx.c -o xxx" to compile it,and it return me that "undefined reference > to > > > `gst_element_factory_get_kalss'"compiled failed. > > > my Gstreamer(version: 0.10) is in /usr/include/gstreamer-0.10/gst/ > > > the Gstreamer lib is in /usr/lib/gstreamer-0.10/ > > > I got this "undefined reference" error in using some other > methods.like > > > gst_element_unref() etc. is there something wrong with my installation? > > > waiting for help.thinks! > > > > It sounds like you're reading a really old version of the application > > development manual which refers to historic GStreamer versions. The > > lastest version can be found here: > > > > http://gstreamer.freedesktop.org/documentation/ > > > > Cheers > > -Tim > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win > > great prizes Grand prize is a trip for two to an Open Source event > > anywhere in the world http://moblin-contest.org/redirect.php? > banner_id=100&url=/ > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > -- > Best regards > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nicolas.m.zhang at gmail.com Mon Sep 8 03:30:01 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Mon, 8 Sep 2008 09:30:01 +0800 Subject: [gst-devel] Decodebin doesn't work whereas playbin works, why? Message-ID: Hi, gstreamer: Recently, I wanna play movies from a rtsp server. I installed DarwinStreamServer on my laptop and use command line: gst-launch playbin uri="rtsp://localhost/test.mp4" It works well. But if I use decodebin instead of playbin, error occurs: gst-launch rtspsrc location="rtsp://localhost/test.mp4" protocols=1 name=d d. ! decodebin ! queue ! ffmpegcolorspace ! xvimagesink d. ! decodebin ! queue ! audioconvert ! alsasink The error message is: ========================= Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock ERROR: from element /pipeline0/d/udpsrc0: Internal data flow error. Additional debug info: gstbasesrc.c(2240): gst_base_src_loop (): /pipeline0/d/udpsrc0: streaming task paused, reason not-linked (-1) Execution ended after 117342031 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... FREEING pipeline ... ================================== I noticed sometimes decodebin works but mostly it fails. What tricks does playbin do to make it work? I skimed the source codes of playbin and found out that it also use decodebin inside. Thanks a lot. I really appreciate your suggestions. Eric Zhang -------------- next part -------------- An HTML attachment was scrubbed... URL: From huan.zheng at intel.com Mon Sep 8 04:18:11 2008 From: huan.zheng at intel.com (Zheng, Huan) Date: Mon, 8 Sep 2008 10:18:11 +0800 Subject: [gst-devel] aac parser In-Reply-To: <48C0FB1E.4070201@hora-obscura.de> References: <72cf309c0809020419w17e87d85yd86165633f682471@mail.gmail.com> <48BD3E7E.9060000@hora-obscura.de> <48C0FB1E.4070201@hora-obscura.de> Message-ID: Hi, Stefan Where will you post the announcement? I'm eager to see the code. :) Best Regards, Zheng, Huan(ZBT) OTC/SSD/SSG Intel Aisa-Pacific Research & Developement Ltd Tel: 021-6116 6435 Inet: 8821 6435 Cub: 3W035 -----Original Message----- From: gstreamer-devel-bounces at lists.sourceforge.net [mailto:gstreamer-devel-bounces at lists.sourceforge.net] On Behalf Of Stefan Kost Sent: 2008?9?5? 17:26 To: Discussion of the development of GStreamer Subject: Re: [gst-devel] aac parser Hi, Zheng, Huan schrieb: > Stefan > Could you please elaborate on this "Soon"? One month or several months? :) > Hopefully next week. > It would be very nice to have an aac parser in gstreamer. > And one more question: Is this parser able to parse ADIF stream into single frames? Because what I heard now is that ADIF has only one header at the beginning, and the offset of each frame can not be located unless you have finished decoding. > lets discuss this once its published. Stefan > Thanks! > > Best Regards, Zheng, Huan(ZBT) > OTC/SSD/SSG > Intel Aisa-Pacific Research & Developement Ltd > Tel: 021-6116 6435 > Inet: 8821 6435 > Cub: 3W035 > -----Original Message----- > From: gstreamer-devel-bounces at lists.sourceforge.net [mailto:gstreamer-devel-bounces at lists.sourceforge.net] On Behalf Of Stefan Kost > Sent: 2008?9?2? 21:24 > To: Discussion of the development of GStreamer > Subject: Re: [gst-devel] aac parser > > hi, > > there will be one in gst-plugin-bad soon. > > Stefan > > Sachin Pandhare schrieb: > >> Hi, >> if aac parser needs to be developed can we take some plugin code as a >> reference and which one will be a suitable candidate for this? >> thanks, >> Sachin >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's challenge >> Build the coolest Linux based applications with Moblin SDK & win great prizes >> Grand prize is a trip for two to an Open Source event anywhere in the world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From nicolas.m.zhang at gmail.com Mon Sep 8 07:23:08 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Mon, 8 Sep 2008 13:23:08 +0800 Subject: [gst-devel] Decodebin doesn't work whereas playbin works, why? Message-ID: Hi, gstreamer: Recently, I wanna play movies from a rtsp server. I installed DarwinStreamServer on my laptop and use command line: gst-launch playbin uri="rtsp://localhost/test.mp4" It works well. But if I use decodebin instead of playbin, error occurs: gst-launch rtspsrc location="rtsp://localhost/test.mp4" protocols=1 name=d d. ! decodebin ! queue ! ffmpegcolorspace ! xvimagesink d. ! decodebin ! queue ! audioconvert ! alsasink The error message is: ========================= Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock ERROR: from element /pipeline0/d/udpsrc0: Internal data flow error. Additional debug info: gstbasesrc.c(2240): gst_base_src_loop (): /pipeline0/d/udpsrc0: streaming task paused, reason not-linked (-1) Execution ended after 117342031 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... FREEING pipeline ... ================================== I noticed sometimes decodebin works but mostly it fails. What tricks does playbin do to make it work? I skimed the source codes of playbin and found out that it also use decodebin inside. Thanks a lot. I really appreciate your suggestions. Eric Zhang -------------- next part -------------- An HTML attachment was scrubbed... URL: From ketandesh at yahoo.com Mon Sep 8 08:38:12 2008 From: ketandesh at yahoo.com (ketan deshpande) Date: Sun, 7 Sep 2008 23:38:12 -0700 (PDT) Subject: [gst-devel] Gstreamer v4l2 to fb on DM6446 Message-ID: <652954.65319.qm@web32208.mail.mud.yahoo.com> Hello everybody, I am working on TI's DM6446. I was trying to run the pipeline from V4L2 to FB(released by TI). gst-launch-0.10 v4l2src ! fbvideosink device=/dev/fb/3 but I keep getting the set of following warnings which lateron fails saying cannot negotiate the caps. WARNING: from element /pipeline0/v4l2src0: Got unexpected frame size of 884736 instead of 829440. Additional debug info: gstv4l2src.c(1082): gst_v4l2src_get_mmap (): /pipeline0/v4l2src0 I tried to give the exact width and height along with the exact fourcc format and framerate but couldnt achieve any success. I am using the good plugins version 0.10.8. Can somebody suggest us as to what might be wrong in our case? regards, -Ketan Unlimited freedom, unlimited storage. Get it now, on http://help.yahoo.com/l/in/yahoo/mail/yahoomail/tools/tools-08.html/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From manauw at skynet.be Mon Sep 8 10:00:16 2008 From: manauw at skynet.be (Mark Nauwelaerts) Date: Mon, 08 Sep 2008 10:00:16 +0200 Subject: [gst-devel] [Fwd: FFmpeg on win32] Message-ID: <48C4DB90.8090704@skynet.be> Any suggestions on question below ? As far as I know, this would come down to some do-it-yourself building, maybe with some help from http://people.collabora.co.uk/~oleavr/OABuild/ Mark. -------- Original Message -------- Subject: FFmpeg on win32 Date: Mon, 8 Sep 2008 09:14:24 +0200 From: Lallement Lucas To: Hello, I noticed you contributed on the 0.10.5 release of Gstreamer's new FFmpeg. I really need this release; I am programming on Windows and the last FFmpeg made available is 0.10.2 and that version crashes with xvid video dued to treading problems. Could you help? Many thanks. Sincerely, Lucas Lallement From timovwb at gmail.com Mon Sep 8 12:00:40 2008 From: timovwb at gmail.com (Timo) Date: Mon, 08 Sep 2008 12:00:40 +0200 Subject: [gst-devel] Visualization with Python In-Reply-To: References: <48c3d722.0309d00a.1388.ffffa25d@mx.google.com> Message-ID: <48c4f7c2.0506d00a.447f.ffffc24f@mx.google.com> Hey Eric, Thanks for your answer, but I know near to nothing about C, so it's a bit hard for me. I think I didn't make myself that clear. I don't want to know when the music stops or changes song. But the actual changes in the music (pitch, frequency, etc.), just like a visualizer. But instead of changing colors according to the music, I want to execute commands according to the music. And all of this in Python. Thanks, Timo Eric Zhang schreef: > Hi, Timo: > > If you don't mind trying playbin2, I know that it will emit signal > `about-to-finish' while changing the playing stuffs to next. You can > connect this signal and do works you want. Playbin hasn't this signal > yet. Refer to /gst/playback/gstplaybin2.c for more > details. > > Eric > > 2008/9/7 Timo > > > Hello, I'm looking for a way to execute commands when music changes. > Like visualizations in music players, but instead of changing weird > lines on the screen, I want to do things. > Now I found out about playbin, but I can't get this to work, and can't > find that much documentation about the subject with Python. > This is a bit that I could figure out, but can't get any further: > pipeline = gst.element_factory_make('playbin', 'playbin') > vis = gst.element_factory_make("goom") > pipeline.set_property('vis-plugin', vis) > > > And if this is possible, can this also be done without writing a > plugin > for a music player? But just use my own program and use the > soundcard or > something to hear the sounds? This way I don't have to write a plugin > for a couple of players but just use my own GUI. > > > Thanks in advance, > > Timo > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win > great prizes > Grand prize is a trip for two to an Open Source event anywhere in > the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From xerxes2 at gmail.com Mon Sep 8 13:00:07 2008 From: xerxes2 at gmail.com (Jens Persson) Date: Mon, 08 Sep 2008 13:00:07 +0200 Subject: [gst-devel] Decodebin doesn't work whereas playbin works, why? In-Reply-To: References: Message-ID: <48C505B7.8020503@gmail.com> Hmm, try this instead: gst-launch rtspsrc location="rtsp://localhost/test.mp4" protocols=1 ! decodebin name=d ! queue ! ffmpegcolorspace ! xvimagesink d. ! queue ! audioconvert ! alsasink Greets Jens Eric Zhang wrote: > Hi, gstreamer: > > Recently, I wanna play movies from a rtsp server. I installed > DarwinStreamServer on my laptop and use command line: > > gst-launch playbin uri="rtsp://localhost/test.mp4" > > It works well. But if I use decodebin instead of playbin, error > occurs: > > gst-launch rtspsrc location="rtsp://localhost/test.mp4" > protocols=1 name=d d. ! decodebin ! queue ! ffmpegcolorspace ! > xvimagesink d. ! decodebin ! queue ! audioconvert ! alsasink > > The error message is: > > ========================= > Setting pipeline to PAUSED ... > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > ERROR: from element /pipeline0/d/udpsrc0: Internal data flow error. > Additional debug info: > gstbasesrc.c(2240): gst_base_src_loop (): /pipeline0/d/udpsrc0: > streaming task paused, reason not-linked (-1) > Execution ended after 117342031 ns. > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > Setting pipeline to NULL ... > FREEING pipeline ... > ================================== > > I noticed sometimes decodebin works but mostly it fails. What > tricks does playbin do to make it work? I skimed the source codes of > playbin and found out that it also use decodebin inside. > > Thanks a lot. I really appreciate your suggestions. > > Eric Zhang > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From fthiery at gmail.com Mon Sep 8 13:26:53 2008 From: fthiery at gmail.com (Florent) Date: Mon, 8 Sep 2008 13:26:53 +0200 Subject: [gst-devel] Visualization with Python In-Reply-To: <48c4f7c2.0506d00a.447f.ffffc24f@mx.google.com> References: <48c3d722.0309d00a.1388.ffffa25d@mx.google.com> <48c4f7c2.0506d00a.447f.ffffc24f@mx.google.com> Message-ID: <1efe3a6e0809080426l261a9b01tb5a3d02794aa9cfd@mail.gmail.com> Hi, I think I didn't make myself that clear. I don't want to know when the > music stops or changes song. But the actual changes in the music (pitch, > frequency, etc.), just like a visualizer. But instead of changing colors > according to the music, I want to execute commands according to the music. > And all of this in Python. Well, you could use analysis elements such as level or spectrum, and use the bus messages with thresholds to launch system events ... What kind of commands do you want to launch ? Florent -------------- next part -------------- An HTML attachment was scrubbed... URL: From sameer at nextbitcpu.com Sun Sep 7 08:19:14 2008 From: sameer at nextbitcpu.com (Sameer Naik) Date: Sun, 7 Sep 2008 11:49:14 +0530 Subject: [gst-devel] GStreamer beginer problem In-Reply-To: <20080907031708.M69442@mail2.sysu.edu.cn> References: <20080907031708.M69442@mail2.sysu.edu.cn> Message-ID: <200809071149.14336.sameer@nextbitcpu.com> Hi, If you have installed gstreamer using apt, then you will also require the devel packages for gstreamer. the devel packages are required since they have the gstreamer API headers. Regards ~Sameer On Sunday 07 September 2008 09:09:57 am Chen Hailiang wrote: > Hi, > I'm beginning to learn GStreamer this day.there is some problem. > my linux is Kubuntu8.04.When the GStreamer installation completed,I > checked it as the FAQ 5 said. It's OK.So I think my installation was > correct. > I tried to run the example in the application development manual(5.4). > I used command "gcc -Wall $(pkg-config --cflags --libs gstreamer-0.10) > xxx.c -o xxx" to compile it,and it return me that "undefined reference to > `gst_element_factory_get_kalss'"compiled failed. > my Gstreamer(version: 0.10) is in /usr/include/gstreamer-0.10/gst/ > the Gstreamer lib is in /usr/lib/gstreamer-0.10/ > I got this "undefined reference" error in using some other methods.like > gst_element_unref() etc. is there something wrong with my installation? > waiting for help.thinks! > > snYe > -- > Best regards From lassi.vaatamoinen at tut.fi Sun Sep 7 09:56:11 2008 From: lassi.vaatamoinen at tut.fi (Lassi =?utf-8?q?V=C3=A4=C3=A4t=C3=A4m=C3=B6inen?=) Date: Sun, 7 Sep 2008 10:56:11 +0300 Subject: [gst-devel] GStreamer beginer problem In-Reply-To: <20080907031708.M69442@mail2.sysu.edu.cn> References: <20080907031708.M69442@mail2.sysu.edu.cn> Message-ID: <200809071056.11180.lassi.vaatamoinen@tut.fi> On Sunday 07 September 2008 06:39:57 Chen Hailiang wrote: > Hi, > I'm beginning to learn GStreamer this day.there is some problem. > my linux is Kubuntu8.04.When the GStreamer installation completed,I > checked it as the FAQ 5 said. It's OK.So I think my installation was > correct. > I tried to run the example in the application development manual(5.4). > I used command "gcc -Wall $(pkg-config --cflags --libs gstreamer-0.10) > xxx.c -o xxx" to compile it, At least the compilation command is invalid, I think. You should put the pkg-config command in `` marks, as follows: "gcc -Wall `pkg-config --cflags --libs gstreamer-0.10` xxx.c -o xxx" This way the pkg-config is run first, and the output from that command is inserted into the gcc command line. -Lassi From manish.rana at gmail.com Mon Sep 8 15:48:03 2008 From: manish.rana at gmail.com (Manish Rana) Date: Mon, 8 Sep 2008 19:18:03 +0530 Subject: [gst-devel] RTP Session Release Message-ID: <8c192ddd0809080648jd010ec6n22a8b394fce3e2e2@mail.gmail.com> Hi All, Senario: Audio and Video Data Transfer using RTP and UDP. ( psVT ) I am removing Video Pipelines, keeping the Audio . ( To remove Video Pipelines i am sending EOS to the pipelines, then unlinking and removing it from the main pipe. Also i release the requested pad. ) Till now every thing works fine. Now if again i want the Video Pipelines to be added I am not able to get the RTP session :( Also if i do not release pad and try reusing the same session, it doesnt work...... Please tell me what i am doing wrong. I looked in to gstrtpbin.c code as well. There the function to release request pad is empty:( ( Is this a bug ????? ) I tried putting prints there it never came till that point.....looks i am doing something wrong :( In Short my senario is VT-> VoIP -> VT Please help Thanks a lot Manish -------------- next part -------------- An HTML attachment was scrubbed... URL: From levi.pope at gmail.com Mon Sep 8 20:00:21 2008 From: levi.pope at gmail.com (Levi Pope) Date: Mon, 8 Sep 2008 13:00:21 -0500 Subject: [gst-devel] Catching Bus messages on Windows Message-ID: <3afe75670809081100m43b6c4b5je1b9deafb5193637@mail.gmail.com> I have a GTK# app that PInvokes a c library that uses gstreamer for video playback. Every thing works fine on both Windows and Linux but I can not get any of my bus messages on Windows. Does anyone know why this is? Thanks Levi -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Mon Sep 8 20:14:00 2008 From: msmith at xiph.org (Michael Smith) Date: Mon, 8 Sep 2008 11:14:00 -0700 Subject: [gst-devel] Catching Bus messages on Windows In-Reply-To: <3afe75670809081100m43b6c4b5je1b9deafb5193637@mail.gmail.com> References: <3afe75670809081100m43b6c4b5je1b9deafb5193637@mail.gmail.com> Message-ID: <3c1737210809081114w269dcc5cw6226cde9f821f3d1@mail.gmail.com> On Mon, Sep 8, 2008 at 11:00 AM, Levi Pope wrote: > I have a GTK# app that PInvokes a c library that uses gstreamer for video > playback. > Every thing works fine on both Windows and Linux but I can not get any of my > bus messages > on Windows. Does anyone know why this is? You should probably provide a testcase (or at a minimum show the code you're talking about) if you want help with this sort of thing. Alternatively, run with GST_DEBUG and see what might be going wrong. Mike From levi.pope at gmail.com Mon Sep 8 21:28:02 2008 From: levi.pope at gmail.com (Levi Pope) Date: Mon, 8 Sep 2008 14:28:02 -0500 Subject: [gst-devel] Catching Bus messages on Windows In-Reply-To: <3c1737210809081114w269dcc5cw6226cde9f821f3d1@mail.gmail.com> References: <3afe75670809081100m43b6c4b5je1b9deafb5193637@mail.gmail.com> <3c1737210809081114w269dcc5cw6226cde9f821f3d1@mail.gmail.com> Message-ID: <3afe75670809081228j6e561f8er3399fceb1d656a26@mail.gmail.com> The GST_DEBUG shows me that the states are changing but I still do not see the messages. Also None of my queries seem to work. (Duration, Position) It seems like there is an issue with the gmainloop or something. Here is the code that creates the pipe. __declspec(dllexport) gboolean video_pipeline_construct_ts (const gchar *uri) { GstState state; GstEleemnt *pipe; GstElement *videosink; GstElement *filesrc; GstElement *demux; GstElement *Decode; GstElement *ColorSpace; GstElement *MultiQueue; GstBus *bus; GstPad *pad; pipe = NULL; pipe = gst_pipeline_new("Pipe"); g_return_val_if_fail (pipe != NULL, FALSE); videosink= gst_element_factory_make ("xvimagesink", "videosink"); if(!G_IS_OBJECT(videosink)) g_warning("****************videosink == NULL!"); filesrc = gst_element_factory_make("filesrc","source1"); if(!G_IS_OBJECT(filesrc)) g_warning("****************filesrc == NULL!"); demux = gst_element_factory_make("flutsdemux","demux"); if(!G_IS_OBJECT(demux)) g_warning("****************flutsdemux == NULL!"); Decode= gst_element_factory_make("decodebin","decoder"); if(!G_IS_OBJECT(Decode)) g_warning("****************Decode == NULL!"); ColorSpace = gst_element_factory_make("ffmpegcolorspace","colorspace"); if(!G_IS_OBJECT(ColorSpace)) g_warning("****************ColorSpace == NULL!"); MultiQueue = gst_element_factory_make("multiqueue","multiqueue"); if(!G_IS_OBJECT(MultiQueue)) g_warning("****************MultiQueue == NULL!"); g_signal_connect (demux, "pad-added", G_CALLBACK (new_demux_pad), player); g_signal_connect (MultiQueue, "pad-added", G_CALLBACK (new_multiqueue_pad), player); g_signal_connect (Decode, "new-decoded-pad", G_CALLBACK (cb_TS_DEC_New_Pad), player); gst_bin_add(GST_BIN(pipe),filesrc); gst_bin_add(GST_BIN(pipe),demux); gst_bin_add(GST_BIN(pipe),Multiqueue); gst_bin_add(GST_BIN(pipe),Decode); vbin = gst_bin_new("vbin"); g_assert(vbin); gst_bin_add_many(GST_BIN(vbin) ColorSpace, videosink, NULL); if(!gst_element_link_many(ColorSpace, videosink, NULL)) { debug("****************Failed to link vbin elements!"); } if(!gst_element_link_many(filesrc,demux, NULL)) { debug("****************Failed to link pipe elements!"); } pad = gst_element_get_pad (ColorSpace, "sink"); gst_element_add_pad (vbin, gst_ghost_pad_new ("sink", pad)); gst_object_unref (pad); gst_bin_add(GST_BIN(pipe),vbin); bus = gst_pipeline_get_bus (GST_PIPELINE (pipe)); if(bus) { debug ("got video BUS."); gst_bus_add_watch (bus, pipeline_bus_callback, NULL); gst_bus_set_sync_handler (bus, gst_bus_sync_signal_handler, NULL); g_signal_connect (bus, "sync-message::element", G_CALLBACK (video_bus_element_sync_message), NULL); gst_object_unref (bus); } else { debug ("Could not get video BUS."); } // Set the pipeline to the proper state gst_element_get_state (pipe, &state, NULL, 0); if (state >= GST_STATE_PAUSED) { gst_element_set_state (pipe, GST_STATE_READY); } debug ("Setting Filesrc Location."); if(filesrc != NULL) { debug ("setting location to = %s",uri); g_object_set (G_OBJECT (filesrc), "location", uri, NULL); } return TRUE; } The gst_bus_sync_signal_handler gets called to set up the xoverlay interface. But the pipeline_bus_callback() never gets called. Let me know if this is not the code you were looking for. Thanks On Mon, Sep 8, 2008 at 1:14 PM, Michael Smith wrote: > On Mon, Sep 8, 2008 at 11:00 AM, Levi Pope wrote: > > I have a GTK# app that PInvokes a c library that uses gstreamer for video > > playback. > > Every thing works fine on both Windows and Linux but I can not get any of > my > > bus messages > > on Windows. Does anyone know why this is? > > You should probably provide a testcase (or at a minimum show the code > you're talking about) if you want help with this sort of thing. > > Alternatively, run with GST_DEBUG and see what might be going wrong. > > Mike > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From manauw at skynet.be Mon Sep 8 22:42:21 2008 From: manauw at skynet.be (Mark Nauwelaerts) Date: Mon, 08 Sep 2008 22:42:21 +0200 Subject: [gst-devel] Announcing GEntrans 0.10.1 Message-ID: <48C58E2D.1090502@skynet.be> This mail announces a new release of GEntrans. ------------ Description: ------------ GEntrans targets GStreamer application developers and experienced end-users and has the following parts and purposes. Primarily; a python script, entrans, as an alternative to gst-launch, but with additional features geared towards transcoding purposes, such as: - specifying pipeline to execute either manually or have it constructed based on a sparse, partial description - non-linear editing of input - progress reporting, if applicable - (element) configuration management Secondly, a set of filter elements (for denoising, sharpening, etc) ported from other applications such as mplayer, avidemux, etc Finally, documentation for all of the above, including a fairly introductory GStreamer transcoding howto, all of which is also available online (e.g. http://gentrans.sourceforge.net/docs/head/manual/html/index.html) Refer to the project page http://gentrans.sourceforge.net for further information or download. ------------------- Changes in v0.10.1: ------------------- * make entrans.py also multiqueue and queue2 aware * allow re-ordering of input streams to output streams * processing pipeline can now also be provided per specific stream (rather than only per type of stream) * seek for selected sections no longer restricted to FORMAT_TIME; now also possible in terms of a custom format (e.g. DVD chapter) * monitor for e.g. no-more-pads time-out, and try to correct/simulate * new generic element bufferjoin: joins subsequent buffers with identical timestamps (e.g. codec sequence headers and subsequent frame) * recognize and support some more container format file extensions * upgrades and adjustments to deprecations and new bindings * misc. updates and fixes Regards, Mark. From chhail at mail2.sysu.edu.cn Tue Sep 9 02:43:15 2008 From: chhail at mail2.sysu.edu.cn (Chen Hailiang) Date: Tue, 9 Sep 2008 08:43:15 +0800 Subject: [gst-devel] GStreamer beginer problem In-Reply-To: References: <20080907031708.M69442@mail2.sysu.edu.cn> <1220779496.13056.3.camel@mini.centricular.net> <20080908002049.M35587@mail2.sysu.edu.cn> Message-ID: <20080909003751.M69008@mail2.sysu.edu.cn> On Mon, 8 Sep 2008 09:22:26 +0800, Eric Zhang wrote Hi, everyone: thaks for helping me.I fixed this 'undefined reference' error now.Eric is right, a spelling mistake. I am sorry to have troubled everyone. Chen > Hi, Chen: > > `gst_element_factory_get_kalss'? Is this a spelling mistake? > > Eric > > 2008/9/8 Chen Hailiang > > > thinks Tim, I got the lastest version now,and found some unuseable methods > > has made up, such like gst_element_unref() was not used again. But I still > > can't use 'gst_element_factory_get_klass()', > > (error:elementInfo.c:(.text+0x60): undefined reference to > > `gst_element_factory_get_kalss'). > > my source codes were copied from <5.4.1. Getting information about an > > element > > using a factory>,the newest version of the application development > > manaul(0.10.20.1).may any other reason to bring about that? > > think you. > > > > On Sun, 07 Sep 2008 10:24:56 +0100, Tim-Philipp M�ler wrote > > > On Sun, 2008-09-07 at 11:39 +0800, Chen Hailiang wrote: > > > > > > Hi, > > > > > > > I tried to run the example in the application development > > manual(5.4). > > > > I used command "gcc -Wall $(pkg-config --cflags --libs > > gstreamer-0.10) > > > > xxx.c -o xxx" to compile it,and it return me that "undefined reference > > to > > > > `gst_element_factory_get_kalss'"compiled failed. > > > > my Gstreamer(version: 0.10) is in /usr/include/gstreamer-0.10/gst/ > > > > the Gstreamer lib is in /usr/lib/gstreamer-0.10/ > > > > I got this "undefined reference" error in using some other > > methods.like > > > > gst_element_unref() etc. is there something wrong with my installation? > > > > waiting for help.thinks! > > > > > > It sounds like you're reading a really old version of the application > > > development manual which refers to historic GStreamer versions. The > > > lastest version can be found here: > > > > > > http://gstreamer.freedesktop.org/documentation/ > > > > > > Cheers > > > -Tim > > > > > > ------------------------------------------------------------------------- > > > This SF.Net email is sponsored by the Moblin Your Move Developer's > > challenge > > > Build the coolest Linux based applications with Moblin SDK & win > > > great prizes Grand prize is a trip for two to an Open Source event > > > anywhere in the world http://moblin-contest.org/redirect.php? > > banner_id=100&url=/ > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > -- > > Best regards > > > > > > > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > > challenge > > Build the coolest Linux based applications with Moblin SDK & win great > > prizes > > Grand prize is a trip for two to an Open Source event anywhere in the world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > -- Best regards From nicolas.m.zhang at gmail.com Tue Sep 9 03:14:31 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Tue, 9 Sep 2008 09:14:31 +0800 Subject: [gst-devel] Decodebin doesn't work whereas playbin works, why? In-Reply-To: <48C505B7.8020503@gmail.com> References: <48C505B7.8020503@gmail.com> Message-ID: Hi, Jens: Thanks for your reply but it doesn't work. Rtspsrc isn't like http-src element, the video & audio stream are separated in it while not been separated in any demux element. So that means there are 2 streams coming from rtspsrc and each of them should be connected to a decodebin. I tried the command you mentioned, the result is - nothing happened. The output is: ============================== eric at eric-desktop:~$ gst-launch rtspsrc location="rtsp://localhost/test.mp4" protocols=1 ! decodebin name=d ! queue ! ffmpegcolorspace ! xvimagesink d. ! queue ! audioconvert ! alsasink Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock ============================== Eric Zhang 2008/9/8 Jens Persson > Hmm, try this instead: > gst-launch rtspsrc location="rtsp://localhost/test.mp4" protocols=1 ! > decodebin name=d ! queue ! ffmpegcolorspace ! xvimagesink d. ! queue ! > audioconvert ! alsasink > > Greets Jens > > Eric Zhang wrote: > > Hi, gstreamer: > > > > Recently, I wanna play movies from a rtsp server. I installed > > DarwinStreamServer on my laptop and use command line: > > > > gst-launch playbin uri="rtsp://localhost/test.mp4" > > > > It works well. But if I use decodebin instead of playbin, error > > occurs: > > > > gst-launch rtspsrc location="rtsp://localhost/test.mp4" > > protocols=1 name=d d. ! decodebin ! queue ! ffmpegcolorspace ! > > xvimagesink d. ! decodebin ! queue ! audioconvert ! alsasink > > > > The error message is: > > > > ========================= > > Setting pipeline to PAUSED ... > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > New clock: GstSystemClock > > ERROR: from element /pipeline0/d/udpsrc0: Internal data flow error. > > Additional debug info: > > gstbasesrc.c(2240): gst_base_src_loop (): /pipeline0/d/udpsrc0: > > streaming task paused, reason not-linked (-1) > > Execution ended after 117342031 ns. > > Setting pipeline to PAUSED ... > > Setting pipeline to READY ... > > Setting pipeline to NULL ... > > FREEING pipeline ... > > ================================== > > > > I noticed sometimes decodebin works but mostly it fails. What > > tricks does playbin do to make it work? I skimed the source codes of > > playbin and found out that it also use decodebin inside. > > > > Thanks a lot. I really appreciate your suggestions. > > > > Eric Zhang > > ------------------------------------------------------------------------ > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Tue Sep 9 10:30:56 2008 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Tue, 9 Sep 2008 14:00:56 +0530 Subject: [gst-devel] ffmpeg - output_example Message-ID: <785339900809090130h4a8a7ffdt9587802f386159f9@mail.gmail.com> Hi all , In gst-ffmpeg-0.10.4\gst-libs\ext\ffmpeg location there is a file named "output_example.c" which creates a container format file on the basis of given extension name in the input file . It internally gererates audio samples and video frames and mux them into a desired container format . But i am not getting that how to make executable out of this so that i can test this . -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From thaytan at noraisin.net Tue Sep 9 10:38:03 2008 From: thaytan at noraisin.net (Jan Schmidt) Date: Tue, 09 Sep 2008 09:38:03 +0100 Subject: [gst-devel] Freezes tonight: Core/Base/Python Message-ID: <1220949484.1673.5.camel@fancy-ubuntu> Freezing Core/Base/Python to make 0.10.20.2, 0.10.20.2 and 0.10.12.2 respectively. For details, see the release schedule: http://gstreamer.freedesktop.org/wiki/ReleasePlanning2008 Cheers, Jan. -- Jan Schmidt From bilboed at gmail.com Tue Sep 9 10:42:01 2008 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 09 Sep 2008 10:42:01 +0200 Subject: [gst-devel] ffmpeg - output_example In-Reply-To: <785339900809090130h4a8a7ffdt9587802f386159f9@mail.gmail.com> References: <785339900809090130h4a8a7ffdt9587802f386159f9@mail.gmail.com> Message-ID: <1220949721.8650.0.camel@localhost> Hi, That's an ffmpeg specific question, you should ask on their mailing lists. Edward On Tue, 2008-09-09 at 14:00 +0530, sudarshan bisht wrote: > Hi all , > > In gst-ffmpeg-0.10.4\gst-libs\ext\ffmpeg location there > is a file named "output_example.c" which creates a container format > file on the basis of given extension name in the input file . It > internally gererates audio samples and video frames and mux them into > a desired container format . > > But i am not getting that how to make executable out of this so that i > can test this . > > > -- > Regards, > > Sudarshan Bisht > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bilboed at gmail.com Tue Sep 9 10:48:22 2008 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 09 Sep 2008 10:48:22 +0200 Subject: [gst-devel] GIT test repositories In-Reply-To: <1220794458.2541.18.camel@putamadre> References: <1220794458.2541.18.camel@putamadre> Message-ID: <1220950102.8650.2.camel@localhost> Hi again, At the general request, I've also created a commit message filter in order to have a saner view of the commits. The general rule is that I tried to condense as much information in one line (a summary if you wish). I haven't checked ALL commits to make sure there aren't any inconsistencies, but the original cvs commit message is present after the one-liner summary (for historical purposes). As mentionned previously... this breaks compatibility with the previously released repositories. As usual, comments/feedbacks/.. are welcome, Edward From dragos.cirjan at gmail.com Tue Sep 9 10:54:46 2008 From: dragos.cirjan at gmail.com (Dragos Cirjan) Date: Tue, 9 Sep 2008 11:54:46 +0300 Subject: [gst-devel] SoupHttpSrc repeat read option Message-ID: <2f0090140809090154l56a789dl8d2b2dbd3e942bd@mail.gmail.com> Hi there. I need a repeat read option for souphttpsrc. The problem that I have sounds like this: there are Video over IP cameras that have neigther MPJEG neigther MPEG sources but JPEG ones. The conclusion is that u have to reread the source over and over again. Can anyone tell me what to modifiy to be able to do this? Thanks. Chris -- ----------------------------------------------------------------- Cristian - Dragos, Cirjan ----------------------------------------------------------------- Email: dragos.cirjan at yahoo.com Email: dragos.cirjan at itmediaconnect.ro, doru at bocancul-literar.ro Telefon: +40726355762 -------------- next part -------------- An HTML attachment was scrubbed... URL: From timovwb at gmail.com Tue Sep 9 12:18:08 2008 From: timovwb at gmail.com (Timo) Date: Tue, 09 Sep 2008 12:18:08 +0200 Subject: [gst-devel] Visualization with Python In-Reply-To: <1efe3a6e0809080426l261a9b01tb5a3d02794aa9cfd@mail.gmail.com> References: <48c3d722.0309d00a.1388.ffffa25d@mx.google.com> <48c4f7c2.0506d00a.447f.ffffc24f@mx.google.com> <1efe3a6e0809080426l261a9b01tb5a3d02794aa9cfd@mail.gmail.com> Message-ID: <48c64d5c.0a04d00a.2d25.ffffac8b@mx.google.com> Is there some documentation on which bus messages exist? Cause I found some, but none of them useable for me. It would be just system commands: os.system(). Timo Florent schreef: > Hi, > > I think I didn't make myself that clear. I don't want to know when the > music stops or changes song. But the actual changes in the music > (pitch, > frequency, etc.), just like a visualizer. But instead of changing > colors > according to the music, I want to execute commands according to > the music. > And all of this in Python. > > > Well, you could use analysis elements such as level or spectrum, and > use the bus messages with thresholds to launch system events ... > > What kind of commands do you want to launch ? > > Florent > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From xerxes2 at gmail.com Tue Sep 9 13:33:57 2008 From: xerxes2 at gmail.com (Jens Persson) Date: Tue, 09 Sep 2008 13:33:57 +0200 Subject: [gst-devel] Visualization with Python In-Reply-To: <48c64d5c.0a04d00a.2d25.ffffac8b@mx.google.com> References: <48c3d722.0309d00a.1388.ffffa25d@mx.google.com> <48c4f7c2.0506d00a.447f.ffffc24f@mx.google.com> <1efe3a6e0809080426l261a9b01tb5a3d02794aa9cfd@mail.gmail.com> <48c64d5c.0a04d00a.2d25.ffffac8b@mx.google.com> Message-ID: <48C65F25.9040000@gmail.com> http://pygstdocs.berlios.de/pygst-reference/gst-constants.html#gst-message-constants They don't do what you're asking of though. :P There are a bunch of other pygst audio projects that might be of more use -> http://pygstdocs.berlios.de/#audio Greets Jens Timo wrote: > Is there some documentation on which bus messages exist? Cause I found > some, but none of them useable for me. > > It would be just system commands: os.system(). > > > Timo > From xerxes2 at gmail.com Tue Sep 9 13:43:45 2008 From: xerxes2 at gmail.com (Jens Persson) Date: Tue, 09 Sep 2008 13:43:45 +0200 Subject: [gst-devel] Decodebin doesn't work whereas playbin works, why? In-Reply-To: References: <48C505B7.8020503@gmail.com> Message-ID: <48C66171.2000101@gmail.com> Ok, but it might give you some more clues. Perhaps like this: gst-launch rtspsrc location="rtsp://localhost/test.mp4" protocols=1 name=d ! decodebin ! queue ! ffmpegcolorspace ! xvimagesink d. ! decodebin ! queue ! audioconvert ! alsasink I've never used rtsp before. :D Greets Jens Eric Zhang wrote: > Hi, Jens: > > Thanks for your reply but it doesn't work. Rtspsrc isn't like > http-src element, the video & audio stream are separated in it while > not been separated in any demux element. So that means there are 2 > streams coming from rtspsrc and each of them should be connected to a > decodebin. > > I tried the command you mentioned, the result is - nothing > happened. The output is: > > ============================== > eric at eric-desktop:~$ gst-launch rtspsrc > location="rtsp://localhost/test.mp4" protocols=1 ! decodebin name=d ! > queue ! ffmpegcolorspace ! xvimagesink d. ! queue ! audioconvert ! > alsasink > Setting pipeline to PAUSED ... > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > ============================== > > Eric Zhang From xerxes2 at gmail.com Tue Sep 9 13:57:46 2008 From: xerxes2 at gmail.com (Jens Persson) Date: Tue, 09 Sep 2008 13:57:46 +0200 Subject: [gst-devel] Decodebin doesn't work whereas playbin works, why? In-Reply-To: <48C66171.2000101@gmail.com> References: <48C505B7.8020503@gmail.com> <48C66171.2000101@gmail.com> Message-ID: <48C664BA.7020306@gmail.com> And perhaps move the queues one step back: gst-launch rtspsrc location="rtsp://localhost/test.mp4" protocols=1 name=d ! queue ! decodebin ! ffmpegcolorspace ! xvimagesink d. ! queue ! decodebin ! audioconvert ! alsasink Greets Jens Jens Persson wrote: > Ok, but it might give you some more clues. Perhaps like this: > > gst-launch rtspsrc location="rtsp://localhost/test.mp4" protocols=1 > name=d ! decodebin ! queue ! ffmpegcolorspace ! xvimagesink d. ! > decodebin ! queue ! audioconvert ! alsasink > > I've never used rtsp before. :D > > Greets Jens > > Eric Zhang wrote: >> Hi, Jens: >> >> Thanks for your reply but it doesn't work. Rtspsrc isn't like >> http-src element, the video & audio stream are separated in it while >> not been separated in any demux element. So that means there are 2 >> streams coming from rtspsrc and each of them should be connected to a >> decodebin. >> >> I tried the command you mentioned, the result is - nothing >> happened. The output is: >> >> ============================== >> eric at eric-desktop:~$ gst-launch rtspsrc >> location="rtsp://localhost/test.mp4" protocols=1 ! decodebin name=d ! >> queue ! ffmpegcolorspace ! xvimagesink d. ! queue ! audioconvert ! >> alsasink >> Setting pipeline to PAUSED ... >> Pipeline is live and does not need PREROLL ... >> Setting pipeline to PLAYING ... >> New clock: GstSystemClock >> ============================== >> >> Eric Zhang > > From gstelzz at yahoo.fr Tue Sep 9 15:26:44 2008 From: gstelzz at yahoo.fr (Aurelien Grimaud) Date: Tue, 9 Sep 2008 13:26:44 +0000 (GMT) Subject: [gst-devel] Decodebin doesn't work whereas playbin works, why? In-Reply-To: Message-ID: <938368.32674.qm@web27208.mail.ukl.yahoo.com> Do you use CVS Head ? Have you got video with gst-launch -v rtspsrc location="rtsp://localhost/test.mp4" debug=1 ! "application/x-rtp, media=video" ! decodebin ! xvimagesink Have you got audio with gst-launch -v rtspsrc location="rtsp://localhost/test.mp4" debug=1 ! "application/x-rtp, media=audio" ! decodebin ! audioconvert ! alsasink To figure out which are the differences between two pipelines (playbin/decodebin), one can dump dot files and draw graphs. export GST_DEBUG_DUMP_DOT_DIR=/tmp gst-launch ... dot files will be generated on every pipeline state change.(since gstreamer-0.10.18) Aurelien --- En date de?: Mar 9.9.08, Eric Zhang a ?crit?: De: Eric Zhang Objet: Re: [gst-devel] Decodebin doesn't work whereas playbin works, why? ?: "Discussion of the development of GStreamer" Date: Mardi 9 Septembre 2008, 3h14 Hi, Jens: ??? Thanks for your reply but it doesn't work. Rtspsrc isn't like http-src element, the video & audio stream are separated in it while not been separated in any demux element. So that means there are 2 streams coming from rtspsrc and each of them should be connected to a decodebin. ??? I tried the command you mentioned, the result is - nothing happened. The output is: ============================== eric at eric-desktop:~$ gst-launch rtspsrc location="rtsp://localhost/test.mp4" protocols=1 ! decodebin name=d ! queue ! ffmpegcolorspace ! xvimagesink? d. ! queue ! audioconvert ! alsasink Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock ============================== Eric Zhang 2008/9/8 Jens Persson Hmm, try this instead: gst-launch rtspsrc location="rtsp://localhost/test.mp4" protocols=1 ! decodebin name=d ! queue ! ffmpegcolorspace ! xvimagesink ?d. ! queue ! audioconvert ! alsasink Greets Jens Eric Zhang wrote: > Hi, gstreamer: > > ? ? Recently, I wanna play movies from a rtsp server. I installed > DarwinStreamServer on my laptop and use command line: > > ? ? gst-launch playbin uri="rtsp://localhost/test.mp4" > > ? ? It works well. But if I use decodebin instead of playbin, error > occurs: > > ? ? gst-launch rtspsrc location="rtsp://localhost/test.mp4" > protocols=1 name=d d. ! decodebin ! queue ! ffmpegcolorspace ! > xvimagesink d. ! decodebin ! queue ! audioconvert ! alsasink > > ? ? The error message is: > > ========================= > Setting pipeline to PAUSED ... > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > ERROR: from element /pipeline0/d/udpsrc0: Internal data flow error. > Additional debug info: > gstbasesrc.c(2240): gst_base_src_loop (): /pipeline0/d/udpsrc0: > streaming task paused, reason not-linked (-1) > Execution ended after 117342031 ns. > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > Setting pipeline to NULL ... > FREEING pipeline ... > ================================== > > ? ? I noticed sometimes decodebin works but mostly it fails. What > tricks does playbin do to make it work? I skimed the source codes of > playbin and found out that it also use decodebin inside. > > ? ? Thanks a lot. I really appreciate your suggestions. > > Eric Zhang > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/_______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From andres.colubri at gmail.com Tue Sep 9 16:36:56 2008 From: andres.colubri at gmail.com (Andres Colubri) Date: Tue, 9 Sep 2008 11:36:56 -0300 Subject: [gst-devel] [Fwd: FFmpeg on win32] In-Reply-To: <48C4DB90.8090704@skynet.be> References: <48C4DB90.8090704@skynet.be> Message-ID: <48017cb00809090736g60c8c380p7293de13ed6dabe9@mail.gmail.com> OABuild comes with a stripped-down version of FFMpeg in order to make it compatible with MSVC++, and so it misses lots of useful codecs. But I was able compile the code of FFMpeg pulled a few days ago from SVN using mingw and msys. However, the version of gcc that comes with mingw by default needs to be replaced by gcc 4.2.4 available here: http://www.tdragon.net/recentgcc That was pretty much the only trick I had to use, then the compilation process went smoothly with the following configure parameters: configure --enable-shared --disable-static --enable-memalign-hack --enable-w32threads Andres On Mon, Sep 8, 2008 at 5:00 AM, Mark Nauwelaerts wrote: > > Any suggestions on question below ? > As far as I know, this would come down to some do-it-yourself building, > maybe > with some help from http://people.collabora.co.uk/~oleavr/OABuild/ > > Mark. > > -------- Original Message -------- > Subject: FFmpeg on win32 > Date: Mon, 8 Sep 2008 09:14:24 +0200 > From: Lallement Lucas > To: > > > > Hello, > > I noticed you contributed on the 0.10.5 release of Gstreamer's new FFmpeg. > I really need this release; I am programming on Windows and the last > FFmpeg made available is > 0.10.2 and that version crashes with xvid video dued to treading problems. > Could you help? > > Many thanks. > > Sincerely, > > Lucas Lallement > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien.isorce at gmail.com Tue Sep 9 18:19:10 2008 From: julien.isorce at gmail.com (Julien Isorce) Date: Tue, 9 Sep 2008 18:19:10 +0200 Subject: [gst-devel] video/x-raw-yuv, format=(fourcc)HDYC Message-ID: <180a127d0809090919j679eaf42p6fbd9d5ee70bf810@mail.gmail.com> Hi, *Any plan to support HDYC format in gstreamer ?* (We have a Video Capture Device that delivers HDYC 1920*1080 only) Some docs here: http://www.fourcc.org/yuv.php#HDYC It seems that the difference bettween UYVY and HDYC is just the coefficients used in the yuv to rgb conversion. T470 coefficients in UYVY to rgb conversion (SD likes) and BT709 coefficients in HDYC to rgb conversion (HD likes) Sincerely J. I. -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien.isorce at gmail.com Tue Sep 9 18:21:49 2008 From: julien.isorce at gmail.com (Julien Isorce) Date: Tue, 9 Sep 2008 18:21:49 +0200 Subject: [gst-devel] video/x-raw-yuv, format=(fourcc)HDYC Message-ID: <180a127d0809090921u5bd8c803v958f718b7888add8@mail.gmail.com> Hi, (I send this mail again, because gmail was waitting etc... so I am not sure it was sent) *Any plan to support HDYC format in gstreamer ?* (We have a Video Capture Device that delivers HDYC 1920*1080 only) Some docs here: http://www.fourcc.org/yuv.php#HDYC It seems that the difference bettween UYVY and HDYC is just the coefficients used in the yuv to rgb conversion. T470 coefficients in UYVY to rgb conversion (SD likes) and BT709 coefficients in HDYC to rgb conversion (HD likes) Sincerely J. I. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Tue Sep 9 18:40:17 2008 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 09 Sep 2008 18:40:17 +0200 Subject: [gst-devel] video/x-raw-yuv, format=(fourcc)HDYC In-Reply-To: <180a127d0809090921u5bd8c803v958f718b7888add8@mail.gmail.com> References: <180a127d0809090921u5bd8c803v958f718b7888add8@mail.gmail.com> Message-ID: <1220978417.10012.8.camel@localhost> Hi, On Tue, 2008-09-09 at 18:21 +0200, Julien Isorce wrote: > Hi, > > (I send this mail again, because gmail was waitting etc... so I am not > sure it was sent) You're too impatient :) > > Any plan to support HDYC format in gstreamer ? If you have elements that support that format... you can already do it now. After all, it's just a matter of creating a caps (like "video/x-raw-yuv,format=(fourcc)HDYC"). The *real* problem is having elements that can convert (properly) from that colourspace/layout to other ones (like the ones used by video sinks or encoders). I don't know if anybody's planning that, but that wouldn't be the only issue with colourspaces in GStreamer (Anyone heard of chroma placement in subsampled video ?). It all comes down to having: * A proper definition of colourspaces in raw video caps * An element that *properly* does that conversion taking into account the correct conversion coefficients/clamping and chroma placement. Oh, and preferably with readable code (unlike ffmpegcolorspace). Patches welcome :) Edward > > (We have a Video Capture Device that delivers HDYC 1920*1080 only) > > Some docs here: http://www.fourcc.org/yuv.php#HDYC > > It seems that the difference bettween UYVY and HDYC is just the > coefficients used in the yuv to rgb conversion. > T470 coefficients in UYVY to rgb conversion (SD likes) and > BT709 coefficients in HDYC to rgb conversion (HD likes) > > Sincerely > > J. I. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From julien.isorce at gmail.com Wed Sep 10 00:11:40 2008 From: julien.isorce at gmail.com (Julien Isorce) Date: Wed, 10 Sep 2008 00:11:40 +0200 Subject: [gst-devel] video/x-raw-yuv, format=(fourcc)HDYC In-Reply-To: <1220978417.10012.8.camel@localhost> References: <180a127d0809090921u5bd8c803v958f718b7888add8@mail.gmail.com> <1220978417.10012.8.camel@localhost> Message-ID: <180a127d0809091511h6883177cl5e5116285d95c2ae@mail.gmail.com> 2008/9/9 Edward Hervey > > > Any plan to support HDYC format in gstreamer ? > > If you have elements that support that format... you can already do > it now. After all, it's just a matter of creating a caps (like > "video/x-raw-yuv,format=(fourcc)HDYC"). > > The *real* problem is having elements that can convert (properly) > from that colourspace/layout to other ones (like the ones used by video > sinks or encoders). > > I don't know if anybody's planning that, but that wouldn't be the > only issue with colourspaces in GStreamer (Anyone heard of chroma > placement in subsampled video ?). > > It all comes down to having: > * A proper definition of colourspaces in raw video caps > * An element that *properly* does that conversion taking into account > the correct conversion coefficients/clamping and chroma placement. Oh, > and preferably with readable code (unlike ffmpegcolorspace). Ok. I have some basic knowledges but I think I have not the background to make a such gst element (colorspace converter). It requires to know a lot of video formats in order to start good abstractions. And use appropriate libraries (liboil ?) For now I can start to add the HDYC to RGB convertion in the ffmegcolorspace element. (just to not forget this format ...) Is there any thing about HDYC in the ffmpeg libs ? > > > Patches welcome :) > > Edward Thx Julien > > > > > > > (We have a Video Capture Device that delivers HDYC 1920*1080 only) > > > > Some docs here: http://www.fourcc.org/yuv.php#HDYC > > > > It seems that the difference bettween UYVY and HDYC is just the > > coefficients used in the yuv to rgb conversion. > > T470 coefficients in UYVY to rgb conversion (SD likes) and > > BT709 coefficients in HDYC to rgb conversion (HD likes) > > > > Sincerely > > > > J. I. > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nicolas.m.zhang at gmail.com Wed Sep 10 03:33:34 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Wed, 10 Sep 2008 09:33:34 +0800 Subject: [gst-devel] Decodebin doesn't work whereas playbin works, why? In-Reply-To: <938368.32674.qm@web27208.mail.ukl.yahoo.com> References: <938368.32674.qm@web27208.mail.ukl.yahoo.com> Message-ID: Hi, Aurelien: Thanks guys. I tried the way Aurelien mentioned and found out that when I play separate video or audio streams -- that was OK. But if I combine the two gst-launch command, error occurs. I think maybe this is a bug with decodebin or gst-launch. According to the error message, seems that the reason is trying to play the stream while the pipeline was not linked completely. Next I am going to upgrade the gstreamer, including all plugin packages, then try again. Thanks for all helps. Eric Zhang 2008/9/9 Aurelien Grimaud > Do you use CVS Head ? > Have you got video with gst-launch -v rtspsrc > location="rtsp://localhost/test.mp4" debug=1 ! "application/x-rtp, > media=video" ! decodebin ! xvimagesink > Have you got audio with gst-launch -v rtspsrc > location="rtsp://localhost/test.mp4" debug=1 ! "application/x-rtp, > media=audio" ! decodebin ! audioconvert ! alsasink > > To figure out which are the differences between two pipelines > (playbin/decodebin), one can dump dot files and draw graphs. > > export GST_DEBUG_DUMP_DOT_DIR=/tmp > gst-launch ... > dot files will be generated on every pipeline state change.(since > gstreamer-0.10.18) > > Aurelien > > --- En date de : *Mar 9.9.08, Eric Zhang * a > ?crit : > > De: Eric Zhang > Objet: Re: [gst-devel] Decodebin doesn't work whereas playbin works, why? > ?: "Discussion of the development of GStreamer" < > gstreamer-devel at lists.sourceforge.net> > Date: Mardi 9 Septembre 2008, 3h14 > > > Hi, Jens: > > Thanks for your reply but it doesn't work. Rtspsrc isn't like http-src > element, the video & audio stream are separated in it while not been > separated in any demux element. So that means there are 2 streams coming > from rtspsrc and each of them should be connected to a decodebin. > > I tried the command you mentioned, the result is - nothing happened. > The output is: > > ============================== > eric at eric-desktop:~$ gst-launch rtspsrc > location="rtsp://localhost/test.mp4" protocols=1 ! decodebin name=d ! queue > ! ffmpegcolorspace ! xvimagesink d. ! queue ! audioconvert ! alsasink > Setting pipeline to PAUSED ... > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > ============================== > > Eric Zhang > > > 2008/9/8 Jens Persson > >> Hmm, try this instead: >> gst-launch rtspsrc location="rtsp://localhost/test.mp4" protocols=1 ! >> decodebin name=d ! queue ! ffmpegcolorspace ! xvimagesink d. ! queue ! >> audioconvert ! alsasink >> >> Greets Jens >> >> Eric Zhang wrote: >> > Hi, gstreamer: >> > >> > Recently, I wanna play movies from a rtsp server. I installed >> > DarwinStreamServer on my laptop and use command line: >> > >> > gst-launch playbin uri="rtsp://localhost/test.mp4" >> > >> > It works well. But if I use decodebin instead of playbin, error >> > occurs: >> > >> > gst-launch rtspsrc location="rtsp://localhost/test.mp4" >> > protocols=1 name=d d. ! decodebin ! queue ! ffmpegcolorspace ! >> > xvimagesink d. ! decodebin ! queue ! audioconvert ! alsasink >> > >> > The error message is: >> > >> > ========================= >> > Setting pipeline to PAUSED ... >> > Pipeline is live and does not need PREROLL ... >> > Setting pipeline to PLAYING ... >> > New clock: GstSystemClock >> > ERROR: from element /pipeline0/d/udpsrc0: Internal data flow error. >> > Additional debug info: >> > gstbasesrc.c(2240): gst_base_src_loop (): /pipeline0/d/udpsrc0: >> > streaming task paused, reason not-linked (-1) >> > Execution ended after 117342031 ns. >> > Setting pipeline to PAUSED ... >> > Setting pipeline to READY ... >> > Setting pipeline to NULL ... >> > FREEING pipeline ... >> > ================================== >> > >> > I noticed sometimes decodebin works but mostly it fails. What >> > tricks does playbin do to make it work? I skimed the source codes of >> > playbin and found out that it also use decodebin inside. >> > >> > Thanks a lot. I really appreciate your suggestions. >> > >> > Eric Zhang >> > ------------------------------------------------------------------------ >> > >> > >> ------------------------------------------------------------------------- >> > This SF.Net email is sponsored by the Moblin Your Move Developer's >> challenge >> > Build the coolest Linux based applications with Moblin SDK & win great >> prizes >> > Grand prize is a trip for two to an Open Source event anywhere in the >> world >> > http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > gstreamer-devel mailing list >> > gstreamer-devel at lists.sourceforge.net >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > >> >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> challenge >> Build the coolest Linux based applications with Moblin SDK & win great >> prizes >> Grand prize is a trip for two to an Open Source event anywhere in the >> world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nicolas.m.zhang at gmail.com Wed Sep 10 03:37:01 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Wed, 10 Sep 2008 09:37:01 +0800 Subject: [gst-devel] SoupHttpSrc repeat read option In-Reply-To: <2f0090140809090154l56a789dl8d2b2dbd3e942bd@mail.gmail.com> References: <2f0090140809090154l56a789dl8d2b2dbd3e942bd@mail.gmail.com> Message-ID: Hi, Dragos: This is interesting. I have no idea on it -- maybe you can create a timeout callback using `g_timeout_add' and play the pipeline again and again? Or you can hook the `EOS' message and trying to play the pipeline again while not terminate the pipeline. Ask Wim, he always has a lot of ideas. :) Eric 2008/9/9 Dragos Cirjan > Hi there. > > I need a repeat read option for souphttpsrc. > > The problem that I have sounds like this: there are Video over IP cameras > that have neigther MPJEG neigther MPEG sources but JPEG ones. The conclusion > is that u have to reread the source over and over again. > > Can anyone tell me what to modifiy to be able to do this? > > Thanks. > > Chris > > -- > ----------------------------------------------------------------- > Cristian - Dragos, Cirjan > ----------------------------------------------------------------- > Email: dragos.cirjan at yahoo.com > Email: dragos.cirjan at itmediaconnect.ro, doru at bocancul-literar.ro > Telefon: +40726355762 > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From learning_gst at hotmail.com Wed Sep 10 09:35:54 2008 From: learning_gst at hotmail.com (learning gst) Date: Wed, 10 Sep 2008 07:35:54 +0000 Subject: [gst-devel] Is GStreamer X independent ? Is Framebuffer plugin ready? Message-ID: I read from the GStreamer's FAQ: Yes, we have no X dependency in any of our core modules. There are GStreamer applications that run fine without any need for X. However, until our Linux Framebuffer or libsvga plugin is ready, you will not be able to play videos without X. In the future, there will probably be lots of different output plugins for video available. I hope to know if Linux framebuffer plugin is ready? I hope to use GStreamer in QT-embedded system that does not support x server. Is it possible? Thanks _________________________________________________________________ Connect to the next generation of MSN Messenger? http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline -------------- next part -------------- An HTML attachment was scrubbed... URL: From francis.meyvis at gmail.com Wed Sep 10 09:50:29 2008 From: francis.meyvis at gmail.com (franchan) Date: Wed, 10 Sep 2008 09:50:29 +0200 Subject: [gst-devel] 2 questions: threading and faad Message-ID: <8456544a0809100050wc1fe123l207f5158a00e61ca@mail.gmail.com> Hello, I'm using a playbin base pipeline. Initially I thought gstreamer was running in its own thread context. But after setting the state to GST_STATE_PLAYING from the main thread context, I get the impression that I only return from the call, when the typefinding procedure has finished. My source takes it content from the internet (kind of progressive download). And I inform this source plug-in each time more data is available from the main thread. The source plug-in presents itself as a pull source because I support seeking in the downloaded part. And the qtdemux requires its source to operate pull based. (don't know if this pull is relevant for my question here). When the source is asked for data but the plug-in sees there's not enough, it blocks in a sleep (or returns an error and certain type finding that is not relevant fails e.g. apetag). Because this all happens in the main thread content, I cannot inform the source plug-in that new data did arrive. And so type finding halts waiting for data (that is there actually). Is my conclusion right? If so, why does the typefinding phase not use the normal playbin buffering mechanism and informs when to pause/play the pipeline for cause of buffering. I likely have to make sure the PLAY_STATE is only set after enough data is downloaded for doing a successful type finding? Or do you know another work around? My second question is regarding faad. I tweak the configure script of the bad plug-in to accept the faad-library installed on my platform and it seems to work well. Yet I get a feeling faad is not welcomed in gstreamer. What is the reason for this? Some people prefer to make a new AAC decoder and a plug-in? Is it impossible to fix faad? Thanks, franchan From dragos.cirjan at gmail.com Wed Sep 10 10:54:03 2008 From: dragos.cirjan at gmail.com (Dragos Cirjan) Date: Wed, 10 Sep 2008 11:54:03 +0300 Subject: [gst-devel] SoupHttpSrc repeat read option Message-ID: <2f0090140809100154w482df841hcdb20cdc35767e8c@mail.gmail.com> Hi there. I have to admit you are kinda talking foring languages to me :|. I'm not familiar at all with gnome lib. Even gstreamer (0.10) documentation has gaps for me. I wrote to the author, but it seems that the email is not valid anymore. Can anyone help me with a patch or smth pls? Thanks a lot in advance. Chris > > Hi, Dragos: > > This is interesting. I have no idea on it -- maybe you can create a > timeout callback using `g_timeout_add' and play the pipeline again and > again? > > Or you can hook the `EOS' message and trying to play the pipeline again > while not terminate the pipeline. Ask Wim, he always has a lot of ideas. :) > > Eric > > 2008/9/9 Dragos Cirjan > > > Hi there. > > > > I need a repeat read option for souphttpsrc. > > > > The problem that I have sounds like this: there are Video over IP cameras > > that have neigther MPJEG neigther MPEG sources but JPEG ones. The conclusion > > is that u have to reread the source over and over again. > > > > Can anyone tell me what to modifiy to be able to do this? > > > > Thanks. > > > > Chris -- ----------------------------------------------------------------- Cristian - Dragos, Cirjan ----------------------------------------------------------------- Email: dragos.cirjan at yahoo.com Email: dragos.cirjan at itmediaconnect.ro, doru at bocancul-literar.ro Telefon: +40726355762 From dragos.cirjan at gmail.com Wed Sep 10 11:00:21 2008 From: dragos.cirjan at gmail.com (Dragos Cirjan) Date: Wed, 10 Sep 2008 12:00:21 +0300 Subject: [gst-devel] help me write a curl src plugin Message-ID: <2f0090140809100200x49a43a98udf04a75303836616@mail.gmail.com> Hy guys. I'm writing a curl src plugin for gstreamer 0.10 but my knowledge is pretty low on both gnome & gstreamer SDK. I have a few problems puting the info in the buffer. Can anyone pls help me with debugging what I wrote till now? You can find the code here: http://dor.homelinux.com/gstcurlsrc.tgz Thanks in advance. P.S. There is a curl plugin arleady written, but for gst 0.8. I need one for 0.10. -- ----------------------------------------------------------------- Cristian - Dragos, Cirjan ----------------------------------------------------------------- Email: dragos.cirjan at yahoo.com Email: dragos.cirjan at itmediaconnect.ro, doru at bocancul-literar.ro Telefon: +40726355762 From ensonic at hora-obscura.de Wed Sep 10 11:18:20 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Wed, 10 Sep 2008 12:18:20 +0300 Subject: [gst-devel] Is GStreamer X independent ? Is Framebuffer plugin ready? In-Reply-To: References: Message-ID: <48C790DC.7070505@hora-obscura.de> learning gst schrieb: > I read from the GStreamer's FAQ: > > Yes, we have no X dependency in any of our core modules. There are > GStreamer applications that > run fine without any need for X. However, until our Linux Framebuffer > or libsvga plugin is ready, you > will not be able to play videos without X. In the future, there will > probably be lots of different output > plugins for video available. > > I hope to know if Linux framebuffer plugin is ready? I hope to use > GStreamer in QT-embedded system that does not support x server. Is it > possible? There is fbdevsink and dfbvideosink for framebuffer and directfb. Stefan > > Thanks > > ------------------------------------------------------------------------ > Connect to the next generation of MSN Messenger Get it now! > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ensonic at hora-obscura.de Wed Sep 10 11:25:56 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Wed, 10 Sep 2008 12:25:56 +0300 Subject: [gst-devel] muxers, timestamps, sparse and continuous recordings Message-ID: <48C792A4.8040700@hora-obscura.de> hi, i was wondering how muxers should handle timestamps on incoming buffers. Assume an applications that shows video from a camera. When you click a button it records to file, allowing to pause and unpause in between. The recorded file should have a continuous stream. If I don't do any special casing this is not the case. 1) When I playback the recorded file, I have an initial pause before video start (if I pressed record after two seconds, the video will start after two seconds). 2) If I pause in between, also in the playback there is a pause. Right now I work around with a pad probe that looks at disconts to aggregate a time_stamp_offset and correct all incoming buffers by subtracting that. It works but probably is not the right way. I believe this involves the use of segments, but I am not sure how. Also both behaviors might be valid (having a sparse and having a continuous stream). So the application would somehow be involved to select the desired behavior. Any comments? Stefan From t.i.m at zen.co.uk Wed Sep 10 11:48:35 2008 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 10 Sep 2008 10:48:35 +0100 Subject: [gst-devel] help me write a curl src plugin In-Reply-To: <2f0090140809100200x49a43a98udf04a75303836616@mail.gmail.com> References: <2f0090140809100200x49a43a98udf04a75303836616@mail.gmail.com> Message-ID: <1221040115.20459.1.camel@mini.centricular.net> On Wed, 2008-09-10 at 12:00 +0300, Dragos Cirjan wrote: > I'm writing a curl src plugin for gstreamer 0.10 but my knowledge is > pretty low on both gnome & gstreamer SDK. > > I have a few problems puting the info in the buffer. Can anyone pls > help me with debugging what I wrote till now? Have you looked at any of the 291 already existing http sources yet? Cheers -Tim From learning_gst at hotmail.com Wed Sep 10 13:59:09 2008 From: learning_gst at hotmail.com (learning gst) Date: Wed, 10 Sep 2008 11:59:09 +0000 Subject: [gst-devel] Which libs will be used if I use gstreamer as a backend of phonon in QT embeded Message-ID: Hello, I will use GStreamer as a backend of Phonon in QT embedded. I have created libphonon_gstreamer.so and libphonon.so.4.1.0. I will run my application that will use Phonon (QT embedded) with a QT-embedded device. Will I need more libs besides libphonon_gstreamer.so and libphonon.so.4.1.0.? Your help will be greatly appreciated? _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Wed Sep 10 17:28:02 2008 From: msmith at xiph.org (Michael Smith) Date: Wed, 10 Sep 2008 08:28:02 -0700 Subject: [gst-devel] 2 questions: threading and faad In-Reply-To: <8456544a0809100050wc1fe123l207f5158a00e61ca@mail.gmail.com> References: <8456544a0809100050wc1fe123l207f5158a00e61ca@mail.gmail.com> Message-ID: <3c1737210809100828o388c3682m2d89bb5626a601c8@mail.gmail.com> On Wed, Sep 10, 2008 at 12:50 AM, franchan wrote: > Hello, > > I'm using a playbin base pipeline. > > Initially I thought gstreamer was running in its own thread context. > But after setting the state to GST_STATE_PLAYING from the main thread context, > I get the impression that I only return from the call, > when the typefinding procedure has finished. This is incorrect, typefinding is never done from the main thread. It's possible, of course, for buggy plugins to make setting the pipeline state block - some of the http sources have bugs like that. Since you're writing your own source, I'd guess the problem is there. > > My source takes it content from the internet (kind of progressive download). > And I inform this source plug-in each time more data is available from > the main thread. > > The source plug-in presents itself as a pull source because > I support seeking in the downloaded part. > And the qtdemux requires its source to operate pull based. > (don't know if this pull is relevant for my question here). That's probably not a good idea. qtdemux works fine in push mode, and you can support seeking in push mode if you want. > > When the source is asked for data but the plug-in sees there's not enough, > it blocks in a sleep (or returns an error and > certain type finding that is not relevant fails e.g. apetag). > > Because this all happens in the main thread content, > I cannot inform the source plug-in that new data did arrive. > And so type finding halts waiting for data (that is there actually). This doesn't happen in the main thread, so you've diagnosed the problem incorrectly. > > Is my conclusion right? > If so, why does the typefinding phase not use > the normal playbin buffering mechanism > and informs when to pause/play the pipeline for cause of buffering. > > I likely have to make sure the PLAY_STATE is only set > after enough data is downloaded for doing a successful type finding? > Or do you know another work around? > > My second question is regarding faad. > I tweak the configure script of the bad plug-in to accept the faad-library > installed on my platform and it seems to work well. > > Yet I get a feeling faad is not welcomed in gstreamer. > What is the reason for this? > Some people prefer to make a new AAC decoder and a plug-in? > Is it impossible to fix faad? > The FAAD library is mostly unmaintained, and is generally not very nice to use. If someone were to step up and maintain it properly, and the gstreamer plugin, and add appropriate tests and documentation, we would be happy to move it to gst-plugins-ugly - all plugins have the same criteria for being moved out of -bad. Mike From msmith at xiph.org Wed Sep 10 17:23:13 2008 From: msmith at xiph.org (Michael Smith) Date: Wed, 10 Sep 2008 08:23:13 -0700 Subject: [gst-devel] Which libs will be used if I use gstreamer as a backend of phonon in QT embeded In-Reply-To: References: Message-ID: <3c1737210809100823n3a88a6e4l7e9699250a93ace1@mail.gmail.com> 2008/9/10 learning gst : > Hello, > > I will use GStreamer as a backend of Phonon in QT embedded. > > I have created libphonon_gstreamer.so and libphonon.so.4.1.0. > > I will run my application that will use Phonon (QT embedded) with a > QT-embedded device. Will I need more libs besides libphonon_gstreamer.so and > libphonon.so.4.1.0.? You'll also need the gstreamer libraries. As for anything phonon-specific, I couldn't say - I imagine a phonon related list would be more likely to know. Mike From dragos.cirjan at gmail.com Wed Sep 10 18:02:33 2008 From: dragos.cirjan at gmail.com (Dragos Cirjan) Date: Wed, 10 Sep 2008 19:02:33 +0300 Subject: [gst-devel] help me write a curl src plugin Message-ID: <2f0090140809100902j1d8fff01n384a7ec13fda4790@mail.gmail.com> If you tell me where to find them, Tim, I so promise I shall look. On the Overview of the Plugin List secion on GST Documentation I can only find 2 (http://gstreamer.freedesktop.org/documentation/plugins.html). Now really. I'm a NOOB! I need to learn and GST has BIG BIG gaps in documentation. Thanks in advance. Chris Message: 6 Date: Wed, 10 Sep 2008 10:48:35 +0100 From: Tim-Philipp M?ller Subject: Re: [gst-devel] help me write a curl src plugin To: gstreamer-devel at lists.sourceforge.net Message-ID: <1221040115.20459.1.camel at mini.centricular.net> Content-Type: text/plain On Wed, 2008-09-10 at 12:00 +0300, Dragos Cirjan wrote: > I'm writing a curl src plugin for gstreamer 0.10 but my knowledge is > pretty low on both gnome & gstreamer SDK. > > I have a few problems puting the info in the buffer. Can anyone pls > help me with debugging what I wrote till now? Have you looked at any of the 291 already existing http sources yet? Cheers -Tim -- ----------------------------------------------------------------- Cristian - Dragos, Cirjan ----------------------------------------------------------------- Email: dragos.cirjan at yahoo.com Email: dragos.cirjan at itmediaconnect.ro, doru at bocancul-literar.ro Telefon: +40726355762 From bilboed at gmail.com Wed Sep 10 18:11:00 2008 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 10 Sep 2008 18:11:00 +0200 Subject: [gst-devel] help me write a curl src plugin In-Reply-To: <2f0090140809100902j1d8fff01n384a7ec13fda4790@mail.gmail.com> References: <2f0090140809100902j1d8fff01n384a7ec13fda4790@mail.gmail.com> Message-ID: <1221063060.12278.1.camel@putamadre> On Wed, 2008-09-10 at 19:02 +0300, Dragos Cirjan wrote: > If you tell me where to find them, Tim, I so promise I shall look. > > On the Overview of the Plugin List secion on GST Documentation I can > only find 2 (http://gstreamer.freedesktop.org/documentation/plugins.html). > Next to those plugins, click on the link (the gst-plugins-XXX one), and download the source from there. > Now really. I'm a NOOB! I need to learn and GST has BIG BIG gaps in > documentation. > > Thanks in advance. > > Chris > > Message: 6 > Date: Wed, 10 Sep 2008 10:48:35 +0100 > From: Tim-Philipp M?ller > Subject: Re: [gst-devel] help me write a curl src plugin > To: gstreamer-devel at lists.sourceforge.net > Message-ID: <1221040115.20459.1.camel at mini.centricular.net> > Content-Type: text/plain > > On Wed, 2008-09-10 at 12:00 +0300, Dragos Cirjan wrote: > > > I'm writing a curl src plugin for gstreamer 0.10 but my knowledge is > > pretty low on both gnome & gstreamer SDK. > > > > I have a few problems puting the info in the buffer. Can anyone pls > > help me with debugging what I wrote till now? > > Have you looked at any of the 291 already existing http sources yet? > > Cheers > -Tim > From dragos.cirjan at gmail.com Wed Sep 10 18:11:51 2008 From: dragos.cirjan at gmail.com (Dragos Cirjan) Date: Wed, 10 Sep 2008 19:11:51 +0300 Subject: [gst-devel] help me write a curl src plugin (complete answer) Message-ID: <2f0090140809100911y17505d4an7b19d9128528e79@mail.gmail.com> If you tell me where to find them, Tim, I so promise I shall look. On the Overview of the Plugin List secion on GST Documentation I can only find 2 (http://gstreamer.freedesktop.org/documentation/plugins.html). Now really. I'm a NOOB! I need to learn and GST has BIG BIG gaps in documentation. In fact even the thing I started there is based on the souphttpsrc sources, because I'm too dumm to understand the GST explainings from the documentation page. (And this only because I couldn't add another option for souphttpsrc to reread the source (make a reading loop) - I have absolutely no experience with gnome). Thanks in advance. Chris >Have you looked at any of the 291 already existing http sources yet? > >Cheers > -Tim >On Wed, 2008-09-10 at 12:00 +0300, Dragos Cirjan wrote: > >> I'm writing a curl src plugin for gstreamer 0.10 but my knowledge is >> pretty low on both gnome & gstreamer SDK. >> >> I have a few problems puting the info in the buffer. Can anyone pls >> help me with debugging what I wrote till now? -- ----------------------------------------------------------------- Cristian - Dragos, Cirjan ----------------------------------------------------------------- Email: dragos.cirjan at yahoo.com Email: dragos.cirjan at itmediaconnect.ro, doru at bocancul-literar.ro Telefon: +40726355762 From wim.taymans at gmail.com Wed Sep 10 17:39:09 2008 From: wim.taymans at gmail.com (Wim Taymans) Date: Wed, 10 Sep 2008 17:39:09 +0200 Subject: [gst-devel] 2 questions: threading and faad In-Reply-To: <3c1737210809100828o388c3682m2d89bb5626a601c8@mail.gmail.com> References: <8456544a0809100050wc1fe123l207f5158a00e61ca@mail.gmail.com> <3c1737210809100828o388c3682m2d89bb5626a601c8@mail.gmail.com> Message-ID: <1221061149.8268.16.camel@metal> On Wed, 2008-09-10 at 08:28 -0700, Michael Smith wrote: > On Wed, Sep 10, 2008 at 12:50 AM, franchan wrote: > > Hello, > > > > I'm using a playbin base pipeline. > > > > Initially I thought gstreamer was running in its own thread context. > > But after setting the state to GST_STATE_PLAYING from the main thread context, > > I get the impression that I only return from the call, > > when the typefinding procedure has finished. > > This is incorrect, typefinding is never done from the main thread. > It's possible, of course, for buggy plugins to make setting the > pipeline state block - some of the http sources have bugs like that. > Since you're writing your own source, I'd guess the problem is there. > The typefind element does typefinding in the state-change to PAUSED. This is arguably the wrong thing to do but it's currently done like that because we otherwise would need a scheduling mode switch from the streaming thread. I'm not quite sure how that would work... Wim > > > > > My source takes it content from the internet (kind of progressive download). > > And I inform this source plug-in each time more data is available from > > the main thread. > > > > The source plug-in presents itself as a pull source because > > I support seeking in the downloaded part. > > And the qtdemux requires its source to operate pull based. > > (don't know if this pull is relevant for my question here). > > That's probably not a good idea. qtdemux works fine in push mode, and > you can support seeking in push mode if you want. > > > > > When the source is asked for data but the plug-in sees there's not enough, > > it blocks in a sleep (or returns an error and > > certain type finding that is not relevant fails e.g. apetag). > > > > Because this all happens in the main thread content, > > I cannot inform the source plug-in that new data did arrive. > > And so type finding halts waiting for data (that is there actually). > > This doesn't happen in the main thread, so you've diagnosed the > problem incorrectly. > > > > > Is my conclusion right? > > If so, why does the typefinding phase not use > > the normal playbin buffering mechanism > > and informs when to pause/play the pipeline for cause of buffering. > > > > I likely have to make sure the PLAY_STATE is only set > > after enough data is downloaded for doing a successful type finding? > > Or do you know another work around? > > > > My second question is regarding faad. > > I tweak the configure script of the bad plug-in to accept the faad-library > > installed on my platform and it seems to work well. > > > > Yet I get a feeling faad is not welcomed in gstreamer. > > What is the reason for this? > > Some people prefer to make a new AAC decoder and a plug-in? > > Is it impossible to fix faad? > > > > The FAAD library is mostly unmaintained, and is generally not very > nice to use. If someone were to step up and maintain it properly, and > the gstreamer plugin, and add appropriate tests and documentation, we > would be happy to move it to gst-plugins-ugly - all plugins have the > same criteria for being moved out of -bad. > > Mike > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bilboed at gmail.com Wed Sep 10 18:27:44 2008 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 10 Sep 2008 18:27:44 +0200 Subject: [gst-devel] video/x-raw-yuv, format=(fourcc)HDYC In-Reply-To: <180a127d0809091511h6883177cl5e5116285d95c2ae@mail.gmail.com> References: <180a127d0809090921u5bd8c803v958f718b7888add8@mail.gmail.com> <1220978417.10012.8.camel@localhost> <180a127d0809091511h6883177cl5e5116285d95c2ae@mail.gmail.com> Message-ID: <1221064064.12278.4.camel@putamadre> On Wed, 2008-09-10 at 00:11 +0200, Julien Isorce wrote: > > > 2008/9/9 Edward Hervey > > > Any plan to support HDYC format in gstreamer ? > > > If you have elements that support that format... you can > already do > it now. After all, it's just a matter of creating a caps (like > "video/x-raw-yuv,format=(fourcc)HDYC"). > > The *real* problem is having elements that can convert > (properly) > from that colourspace/layout to other ones (like the ones used > by video > sinks or encoders). > > I don't know if anybody's planning that, but that wouldn't > be the > only issue with colourspaces in GStreamer (Anyone heard of > chroma > placement in subsampled video ?). > > It all comes down to having: > * A proper definition of colourspaces in raw video caps > * An element that *properly* does that conversion taking > into account > the correct conversion coefficients/clamping and chroma > placement. Oh, > and preferably with readable code (unlike ffmpegcolorspace). > > Ok. I have some basic knowledges but I think I have not the background > to make a such gst element (colorspace converter). > It requires to know a lot of video formats in order to start good > abstractions. And use appropriate libraries (liboil ?) > For now I can start to add the HDYC to RGB convertion in the > ffmegcolorspace element. (just to not forget this format ...) > > Is there any thing about HDYC in the ffmpeg libs ? There is one mention of HDYC in the source code (liabvcodec/raw.c) ... and that's to say it's UYVY422 (sigh). So they don't make a difference between YUY2, Y422, UYVY and HDYC. > > > > Patches welcome :) > > Edward > > Thx > > Julien > > > > > > > > > (We have a Video Capture Device that delivers HDYC 1920*1080 > only) > > > > Some docs here: http://www.fourcc.org/yuv.php#HDYC > > > > It seems that the difference bettween UYVY and HDYC is just > the > > coefficients used in the yuv to rgb conversion. > > T470 coefficients in UYVY to rgb conversion (SD likes) and > > BT709 coefficients in HDYC to rgb conversion (HD likes) > > > > Sincerely > > > > J. I. > > > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move > Developer's challenge > > Build the coolest Linux based applications with Moblin SDK & > win great prizes > > Grand prize is a trip for two to an Open Source event > anywhere in the world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move > Developer's challenge > Build the coolest Linux based applications with Moblin SDK & > win great prizes > Grand prize is a trip for two to an Open Source event anywhere > in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bilboed at gmail.com Wed Sep 10 18:29:56 2008 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 10 Sep 2008 18:29:56 +0200 Subject: [gst-devel] SoupHttpSrc repeat read option In-Reply-To: References: <2f0090140809090154l56a789dl8d2b2dbd3e942bd@mail.gmail.com> Message-ID: <1221064196.12278.7.camel@putamadre> Hi, Why don't you just use a fakesrc, connect to the 'handoff' signal and do the http GET yourself ? You can then give the data back to fakesrc that way. BTW, are you sure it's not feeding you the jpegs as multipart ? If you go on the address in a webpage... does the image refresh automatically ? Edward On Wed, 2008-09-10 at 09:37 +0800, Eric Zhang wrote: > Hi, Dragos: > > This is interesting. I have no idea on it -- maybe you can create > a timeout callback using `g_timeout_add' and play the pipeline again > and again? > > Or you can hook the `EOS' message and trying to play the pipeline > again while not terminate the pipeline. Ask Wim, he always has a lot > of ideas. :) > > Eric > > 2008/9/9 Dragos Cirjan > Hi there. > > I need a repeat read option for souphttpsrc. > > The problem that I have sounds like this: there are Video over > IP cameras that have neigther MPJEG neigther MPEG sources but > JPEG ones. The conclusion is that u have to reread the source > over and over again. > > Can anyone tell me what to modifiy to be able to do this? > > Thanks. > > Chris > > -- > ----------------------------------------------------------------- > Cristian - Dragos, Cirjan > ----------------------------------------------------------------- > Email: dragos.cirjan at yahoo.com > Email: dragos.cirjan at itmediaconnect.ro, > doru at bocancul-literar.ro > Telefon: +40726355762 > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move > Developer's challenge > Build the coolest Linux based applications with Moblin SDK & > win great prizes > Grand prize is a trip for two to an Open Source event anywhere > in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From msmith at xiph.org Wed Sep 10 18:39:41 2008 From: msmith at xiph.org (Michael Smith) Date: Wed, 10 Sep 2008 09:39:41 -0700 Subject: [gst-devel] 2 questions: threading and faad In-Reply-To: <1221061149.8268.16.camel@metal> References: <8456544a0809100050wc1fe123l207f5158a00e61ca@mail.gmail.com> <3c1737210809100828o388c3682m2d89bb5626a601c8@mail.gmail.com> <1221061149.8268.16.camel@metal> Message-ID: <3c1737210809100939n4f5e3ac6hee6dfc35111e5f43@mail.gmail.com> On Wed, Sep 10, 2008 at 8:39 AM, Wim Taymans wrote: > On Wed, 2008-09-10 at 08:28 -0700, Michael Smith wrote: >> On Wed, Sep 10, 2008 at 12:50 AM, franchan wrote: >> > Hello, >> > >> > I'm using a playbin base pipeline. >> > >> > Initially I thought gstreamer was running in its own thread context. >> > But after setting the state to GST_STATE_PLAYING from the main thread context, >> > I get the impression that I only return from the call, >> > when the typefinding procedure has finished. >> >> This is incorrect, typefinding is never done from the main thread. >> It's possible, of course, for buggy plugins to make setting the >> pipeline state block - some of the http sources have bugs like that. >> Since you're writing your own source, I'd guess the problem is there. >> > > The typefind element does typefinding in the state-change to PAUSED. > This is arguably the wrong thing to do but it's currently done like that > because we otherwise would need a scheduling mode switch from the > streaming thread. I'm not quite sure how that would work... > Ah, you're right. Sorry about that, franchan. Wim: we should think about a fix for that, though probably for 0.11? So: yes, apparently we do typefinding from the main thread, if we're in pull mode. That's unfortunate. However, it doesn't work this way in push mode. Since I recommended using push mode _anyway_, that's not too bad - you'll just need to change your plugin to work in push mode. There are a number of other network sources that operate in push mode and support streaming (e.g. the http sources), those might be useful examples for you. Mike From lutz at topfrose.de Wed Sep 10 19:10:30 2008 From: lutz at topfrose.de (Lutz =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 10 Sep 2008 19:10:30 +0200 Subject: [gst-devel] help me write a curl src plugin (complete answer) In-Reply-To: <2f0090140809100911y17505d4an7b19d9128528e79@mail.gmail.com> References: <2f0090140809100911y17505d4an7b19d9128528e79@mail.gmail.com> Message-ID: <1221066630.3934.4.camel@acer> On Wed, 2008-09-10 at 19:11 +0300, Dragos Cirjan wrote: > And this only because I couldn't add another option for souphttpsrc > to reread the source (make a reading loop) What exactly do you need/do you want to achieve? How? Regards Lutz From thaytan at noraisin.net Wed Sep 10 20:31:31 2008 From: thaytan at noraisin.net (Jan Schmidt) Date: Wed, 10 Sep 2008 19:31:31 +0100 Subject: [gst-devel] Freezes tonight: Core/Base/Python In-Reply-To: <1220949484.1673.5.camel@fancy-ubuntu> References: <1220949484.1673.5.camel@fancy-ubuntu> Message-ID: <1221071491.1673.29.camel@fancy-ubuntu> Pre-releases of GStreamer Core 0.10.20.2, Base 0.10.20.2 and Python bindings 0.10.12.2 are now available: http://gstreamer.freedesktop.org/data/src/gstreamer/pre/gstreamer-0.10.20.2.tar.bz2 http://gstreamer.freedesktop.org/data/src/gst-plugins-base/pre/gst-plugins-base-0.10.20.2.tar.bz2 and http://gstreamer.freedesktop.org/data/src/gst-python/pre/gst-python-0.10.12.2.tar.bz2 Please test them out, and file bugs in http://bugzilla.gnome.org/ New pre-releases Friday or Saturday as needed. Cheers, Jan. On Tue, 2008-09-09 at 09:38 +0100, Jan Schmidt wrote: > Freezing Core/Base/Python to make 0.10.20.2, 0.10.20.2 and 0.10.12.2 > respectively. > > For details, see the release schedule: > http://gstreamer.freedesktop.org/wiki/ReleasePlanning2008 > > Cheers, > Jan. -- Jan Schmidt From dragos.cirjan at gmail.com Wed Sep 10 20:32:52 2008 From: dragos.cirjan at gmail.com (Dragos Cirjan) Date: Wed, 10 Sep 2008 21:32:52 +0300 Subject: [gst-devel] SoupHttpSrc(CurlSrc) repeat read option Message-ID: <2f0090140809101132x6c818ee0m91adb838083f645e@mail.gmail.com> Since I kinda finished my plugin ( using curl ) also ( without help :)) ), I will merge the two threads I made. Btw. you can find my plugin here: http://dor.homelinux.com/gstcurlsrc.tgz(For now it's on ALPHA stage). I need to know how to repeat reading a jpeg image, because THERE ARE Video Cameras/Servers that DO NOT have mJPEG option, but only JPEG, and you really need to simulate mJPEG by reading the source repeatedly. The question remains: Can you please help me add such an option either to my plugin either to souphttp plugin? Edward, I need you to explain to me a little more detailed. If it is the way I understood, I can't do that. I'm suppose that in the end I should be able to run at least 32 threads (as in mpeg/jpeg/mjpeg links) in the same time, and it may become to complicated adding fakesrc's while I still belive I can add such an option. Tim, I have all the sources downloaded for a few weeks now, but still I couldn't find more than neon and soup as http sources. If there are others, their names do not sugest this. Thanks in advance. Chris P.S. I have an ideea, but I simply do not have the energy to work on it today anymore. If it comes out to be good, I'll probably let u know tomorow. Still I am 90% sure I still need your help. > On Wed, 2008-09-10 at 19:02 +0300, Dragos Cirjan wrote: > > If you tell me where to find them, Tim, I so promise I shall look. > > > > On the Overview of the Plugin List secion on GST Documentation I can > > only find 2 (http://gstreamer.freedesktop.org/documentation/plugins.html ). > > > > Next to those plugins, click on the link (the gst-plugins-XXX one), > and download the source from there. > > > Now really. I'm a NOOB! I need to learn and GST has BIG BIG gaps in > > documentation. > > > > Thanks in advance. > > > > Chris > > > > Message: 6 > > Date: Wed, 10 Sep 2008 10:48:35 +0100 > > From: Tim-Philipp M?ller > > Subject: Re: [gst-devel] help me write a curl src plugin > > To: gstreamer-devel at lists.sourceforge.net > > Message-ID: <1221040115.20459.1.camel at mini.centricular.net> > > Content-Type: text/plain > > > > On Wed, 2008-09-10 at 12:00 +0300, Dragos Cirjan wrote: > > > > > I'm writing a curl src plugin for gstreamer 0.10 but my knowledge is > > > pretty low on both gnome & gstreamer SDK. > > > > > > I have a few problems puting the info in the buffer. Can anyone pls > > > help me with debugging what I wrote till now? > > > > Have you looked at any of the 291 already existing http sources yet? > > > > Cheers > > -Tim > > > > =================================================== > > Hi, > > Why don't you just use a fakesrc, connect to the 'handoff' signal and > do the http GET yourself ? You can then give the data back to fakesrc > that way. > > BTW, are you sure it's not feeding you the jpegs as multipart ? If you > go on the address in a webpage... does the image refresh automatically ? > > Edward > > On Wed, 2008-09-10 at 09:37 +0800, Eric Zhang wrote: > > Hi, Dragos: > > > > This is interesting. I have no idea on it -- maybe you can create > > a timeout callback using `g_timeout_add' and play the pipeline again > > and again? > > > > Or you can hook the `EOS' message and trying to play the pipeline > > again while not terminate the pipeline. Ask Wim, he always has a lot > > of ideas. :) > > > > Eric > > > > 2008/9/9 Dragos Cirjan > > Hi there. > > > > I need a repeat read option for souphttpsrc. > > > > The problem that I have sounds like this: there are Video over > > IP cameras that have neigther MPJEG neigther MPEG sources but > > JPEG ones. The conclusion is that u have to reread the source > > over and over again. > > > > Can anyone tell me what to modifiy to be able to do this? > > > > Thanks. > > > > Chris -- ----------------------------------------------------------------- Cristian - Dragos, Cirjan ----------------------------------------------------------------- Email: dragos.cirjan at yahoo.com Email: dragos.cirjan at itmediaconnect.ro, doru at bocancul-literar.ro Telefon: +40726355762 -------------- next part -------------- An HTML attachment was scrubbed... URL: From wingo at pobox.com Wed Sep 10 21:47:30 2008 From: wingo at pobox.com (Andy Wingo) Date: Wed, 10 Sep 2008 21:47:30 +0200 Subject: [gst-devel] GIT test repositories In-Reply-To: <1220950102.8650.2.camel@localhost> (Edward Hervey's message of "Tue, 09 Sep 2008 10:48:22 +0200") References: <1220794458.2541.18.camel@putamadre> <1220950102.8650.2.camel@localhost> Message-ID: On Tue 09 Sep 2008 10:48, Edward Hervey writes: > At the general request, I've also created a commit message filter in > order to have a saner view of the commits. Edward this looks great. YOU ARE A HERO -- http://wingolog.org/ From airmind at gmail.com Wed Sep 10 22:34:15 2008 From: airmind at gmail.com (Alexandre) Date: Wed, 10 Sep 2008 17:34:15 -0300 Subject: [gst-devel] Link xvid with ffmpeg's mp4 muxer Message-ID: <48f4838d0809101334q2771ee18kfb6df1c349373f8b@mail.gmail.com> Hi, I'm trying to mux the GStreamer xvidenc element with ffmux_mp4, to create a Mpeg 4 file with XVid encoded video, but it's not being succesfull. The xvidenc caps are video/x-xvid, and ffmux supports either mpeg v4 video or video/x-divx. Is there a way to link them, or are they incompatible? -- Alexandre Rosenfeld EngComp 06 - USP S?o Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: From ylatuya at gmail.com Thu Sep 11 01:03:26 2008 From: ylatuya at gmail.com (Andoni Morales Alastruey) Date: Thu, 11 Sep 2008 01:03:26 +0200 Subject: [gst-devel] dv1394 Video Capture Message-ID: <1221087806.32279.21.camel@longo> Hello: I'm trying to create a pipeline to capture video streams from a dv1394 source with this behavior: Displaying the input stream Encode the video stream to a new file For this, I have created a decode bin, linked to a tee with 2 outputs: the first one linked to a display bin and the second one to an encode bin. The problem I'm facing is what to do when I decide to start the encoding, pause it or finally stop it. As all the elements are in the same pipeline I don't now how to tell the encode bin to go to the GST_STATE_PAUSED, or how to send it a eos event.In fact, what I don't really know is if two elements in the same pipeline can be in a different state. If so, the solution is easy. Another option I took in consideration is to remove the encode bin from the pipeline and set it to the GST_STATE_PAUSED state or inject an eos event to it sink pad when I decide to pause the encoding or stop it. Thanks! From lomesh.agarwal at intel.com Thu Sep 11 02:29:11 2008 From: lomesh.agarwal at intel.com (Agarwal, Lomesh) Date: Wed, 10 Sep 2008 17:29:11 -0700 Subject: [gst-devel] problem with flutsdemux Message-ID: I am trying to render content using gstreamer on Ubuntu machines. I have a program which mimics following pipeline - gst-launch -v gstrtpbin name=rtpbin udpsrc port=5000 caps="application/x-rtp,media=video,clock-rate=90000,encoding-name=mpegts" ! \ rtpbin.recv_rtp_sink_0 rtpbin. ! \ rtpmp2tdepay ! \ flutsdemux name=demuxer \ demuxer. ! queue max-size-buffers=0 max-size-time=0 ! flumcaacdec ! audioconvert ! volume volume=10 ! autoaudiosink \ demuxer. ! queue max-size-buffers=0 max-size-time=0 ! fluh264dec! autovideosink \ udpsrc port=5001 ! \ rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! \ udpsink port=5005 sync=false async=false -t I want to keep rendering the new streams sent by sender. So, on receiving "on-bye-ssrc" signal from gstrtpbin I unlink all the elements and keep the pipeline in playing state. When sender starts sending new stream I receive "on-new-ssrc" and "pad-added" signals from gstrtpbin. I link gstrtpbin, rtpmp2depay and flutsdemux on receiving "pad-added" signal. Then I wait for "pad-added" signal from flutsdemux for audio and video streams. But I never receive it. If I run the pipeline for the first time I receive those signals. In case of new ssrc I never receive it. I tried freeing the flutsdemux and creating it again and even then I don't receive the signal. Any help is appreciated. From nicolas.m.zhang at gmail.com Thu Sep 11 03:20:23 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Thu, 11 Sep 2008 09:20:23 +0800 Subject: [gst-devel] muxers, timestamps, sparse and continuous recordings In-Reply-To: <48C792A4.8040700@hora-obscura.de> References: <48C792A4.8040700@hora-obscura.de> Message-ID: Hi, Stefan: I think your pipeline is using GstSystemClock because you mentioned the source is a live element. If it is, the clock will keep increasing even if the pipeline is paused. This makes the timestamp noncontinuous. To generate a continuous timestamp, I think you can try to use the clock provided by your sink elements. Maybe this is not easy because the live element is different with other source elements. Eric 2008/9/10 Stefan Kost > hi, > > i was wondering how muxers should handle timestamps on incoming buffers. > Assume an applications that shows video from a camera. When you click a > button it records to file, allowing to pause and unpause in between. The > recorded file should have a continuous stream. If I don't do any special > casing this is not the case. > > 1) When I playback the recorded file, I have an initial pause before > video start (if I pressed record after two seconds, the video will start > after two seconds). > > 2) If I pause in between, also in the playback there is a pause. > > Right now I work around with a pad probe that looks at disconts to > aggregate a time_stamp_offset and correct all incoming buffers by > subtracting that. It works but probably is not the right way. I believe > this involves the use of segments, but I am not sure how. Also both > behaviors might be valid (having a sparse and having a continuous > stream). So the application would somehow be involved to select the > desired behavior. Any comments? > > > Stefan > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Thu Sep 11 07:23:59 2008 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Thu, 11 Sep 2008 10:53:59 +0530 Subject: [gst-devel] muxers, timestamps, sparse and continuous recordings In-Reply-To: References: <48C792A4.8040700@hora-obscura.de> Message-ID: <785339900809102223w104caf83gbdf06dacd3ed346c@mail.gmail.com> Hi , I had also tried this thing , I was getting same video frames for the particular duration . Its not a problem of muxers i guess . Because muxer creates its own timestamp and attach it to the buffer . Its problem of tea element where you connect your preview and record pipeline . Whenever you add a new pipeline to the running pipeline so what I think is all the elements of second pipelines should be in sync with the existing pipeline because source element is common here . So in order to do that tea element keep pushes same video frame for that particular duration that is same as the time till you press record. To fix this problem what I did is wrote a plugin which sends running pipeline's position as a timestamp for the first buffer , only for the very first time . And I insert this plugin right after tea element in record pipeline. This has fixed my problem . On Thu, Sep 11, 2008 at 6:50 AM, Eric Zhang wrote: > Hi, Stefan: > > I think your pipeline is using GstSystemClock because you mentioned the > source is a live element. If it is, the clock will keep increasing even if > the pipeline is paused. This makes the timestamp noncontinuous. To generate > a continuous timestamp, I think you can try to use the clock provided by > your sink elements. Maybe this is not easy because the live element is > different with other source elements. > > Eric > > 2008/9/10 Stefan Kost > > hi, >> >> i was wondering how muxers should handle timestamps on incoming buffers. >> Assume an applications that shows video from a camera. When you click a >> button it records to file, allowing to pause and unpause in between. The >> recorded file should have a continuous stream. If I don't do any special >> casing this is not the case. >> >> 1) When I playback the recorded file, I have an initial pause before >> video start (if I pressed record after two seconds, the video will start >> after two seconds). >> >> 2) If I pause in between, also in the playback there is a pause. >> >> Right now I work around with a pad probe that looks at disconts to >> aggregate a time_stamp_offset and correct all incoming buffers by >> subtracting that. It works but probably is not the right way. I believe >> this involves the use of segments, but I am not sure how. Also both >> behaviors might be valid (having a sparse and having a continuous >> stream). So the application would somehow be involved to select the >> desired behavior. Any comments? >> >> >> Stefan >> >> >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> challenge >> Build the coolest Linux based applications with Moblin SDK & win great >> prizes >> Grand prize is a trip for two to an Open Source event anywhere in the >> world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From francis.meyvis at gmail.com Thu Sep 11 09:28:57 2008 From: francis.meyvis at gmail.com (franchan) Date: Thu, 11 Sep 2008 09:28:57 +0200 Subject: [gst-devel] gstreamer-devel Digest, Vol 28, Issue 22 In-Reply-To: References: Message-ID: <8456544a0809110028p24ace434s510b7fe24b89906c@mail.gmail.com> Hello, First thanks for sharing your insights! >> The typefind element does typefinding in the state-change to PAUSED. >> This is arguably the wrong thing to do but it's currently done like that >> because we otherwise would need a scheduling mode switch from the >> streaming thread. I'm not quite sure how that would work... Well, I was inaccurate about that, the problem popped up in the paused state indeed. But by then the email was already send already. I'm not sure why it would be wrong doing it in the PAUSE state. Although I don't work long with gstreamer, it simple state model makes sense to me: READY state: don't make contact with server's yet but have allocated as much resources as necessary. PAUSE: ready for playing as soon as state changes to PLAYING. So by going from READY to the PAUSE state, it should start connecting to the network, get initial content to do typefinding. It would be nice if also in PAUSE transition, the playbin buffering mechanism would be in place already, so that the whole gmain_loop does not block in a "read()". > Ah, you're right. Sorry about that, franchan. Wim: we should think > about a fix for that, though probably for 0.11? > > So: yes, apparently we do typefinding from the main thread, if we're > in pull mode. That's unfortunate. However, it doesn't work this way in > push mode. Since I recommended using push mode _anyway_, that's not > too bad - you'll just need to change your plugin to work in push mode. Perhaps I should use another demuxer (now this was auto selected by playbin for the h264/AAC content) or use some API I did not find yet? At first I started with a progressive download plug-in for push operation. Then I found qtdemux refused to seek with this source plug-in. I made these conclusions back then when reading and tracing qtdemux.c (I write just what I remember from last weeks so I might miss) qtdemux_sink_activate_pull() sets its data member demux->pullbased=true. gst_qtdemux_handle_src_event() checks this data member in case of a GST_EVENT_SEEK event. And it concludes with GST_DEBUG_OBJECT (qtdemux, "cannot seek in streaming mode") if pullbased is false. qtdemux_sink_activate_pull() resets the pullbased flag when the qtdemux_sink_activate() concludes through the gst_pad_check_pull_range() that the source plug-in was push only because it missed a _check_range_get() function, typical for pull plug-ins ... so I thought ... So I decided to rewrite my plug-in at that time to not use a push model. I was surprised to see this demux code because gstreamer's documentation, seems to favor push operation, as you also tell here ... > There are a number of other network sources that operate in push mode > and support streaming (e.g. the http sources), those might be useful > examples for you. Well, I looked into the soup/neon based http source plug-ins, but these are complicated :-) Now the pull based plug-in is very simple (note that the HTTP downloading work is outside this plug-in and based on legacy code). For now the easiest for me is to have a delayed "PAUSE/PLAY". Once the legacy code tells me that it has a fair amount of buffered data, I put the pipeline into the PAUSE/PLAY state and typefinding phase can likely finish correctly. (I'm just afraid, how much data do I need exactly. But that's a completely different problem) For now about 300K seems to be enough to get the pipeline rolling ... (moov bag in front of the mp4 file of course) Thanks, franchan From ved.kpl at gmail.com Thu Sep 11 10:40:28 2008 From: ved.kpl at gmail.com (ved kpl) Date: Thu, 11 Sep 2008 14:10:28 +0530 Subject: [gst-devel] muxers, timestamps, sparse and continuous recordings In-Reply-To: <785339900809102223w104caf83gbdf06dacd3ed346c@mail.gmail.com> References: <48C792A4.8040700@hora-obscura.de> <785339900809102223w104caf83gbdf06dacd3ed346c@mail.gmail.com> Message-ID: <7496c23f0809110140q43c2c39eof52a00dba2cae7b2@mail.gmail.com> Hi, Stefan, I also got it working with pad probe. In my case, the pipeline has recordbin and previewbin right from the start. So the first data should flow right till the sinks, to make sure that the pipeline is prerolled and then trigerring the probe ON/Off for recording. Eric, I guess a new base time is set when the pipeline is set to paused. I dont think that should pose any problem (other than AV sync maybe). Please correct me on this. Ved On Thu, Sep 11, 2008 at 10:53 AM, sudarshan bisht wrote: > Hi , > I had also tried this thing , I was getting same video frames for > the particular duration . Its not a problem of muxers i guess . Because > muxer creates its own timestamp and attach it to the buffer . Its problem of > tea element where you connect your preview and record pipeline . Whenever > you add a new pipeline to the running pipeline so what I think is all the > elements of second pipelines should be in sync with the existing pipeline > because source element is common here . So in order to do that tea element > keep pushes same video frame for that particular duration that is same as > the time till you press record. To fix this problem what I did is wrote a > plugin which sends running pipeline's position as a timestamp for the first > buffer , only for the very first time . And I insert this plugin right after > tea element in record pipeline. > > This has fixed my problem . > > > On Thu, Sep 11, 2008 at 6:50 AM, Eric Zhang > wrote: >> >> Hi, Stefan: >> >> I think your pipeline is using GstSystemClock because you mentioned >> the source is a live element. If it is, the clock will keep increasing even >> if the pipeline is paused. This makes the timestamp noncontinuous. To >> generate a continuous timestamp, I think you can try to use the clock >> provided by your sink elements. Maybe this is not easy because the live >> element is different with other source elements. >> >> Eric >> >> 2008/9/10 Stefan Kost >>> >>> hi, >>> >>> i was wondering how muxers should handle timestamps on incoming buffers. >>> Assume an applications that shows video from a camera. When you click a >>> button it records to file, allowing to pause and unpause in between. The >>> recorded file should have a continuous stream. If I don't do any special >>> casing this is not the case. >>> >>> 1) When I playback the recorded file, I have an initial pause before >>> video start (if I pressed record after two seconds, the video will start >>> after two seconds). >>> >>> 2) If I pause in between, also in the playback there is a pause. >>> >>> Right now I work around with a pad probe that looks at disconts to >>> aggregate a time_stamp_offset and correct all incoming buffers by >>> subtracting that. It works but probably is not the right way. I believe >>> this involves the use of segments, but I am not sure how. Also both >>> behaviors might be valid (having a sparse and having a continuous >>> stream). So the application would somehow be involved to select the >>> desired behavior. Any comments? >>> >>> >>> Stefan >>> >>> >>> >>> ------------------------------------------------------------------------- >>> This SF.Net email is sponsored by the Moblin Your Move Developer's >>> challenge >>> Build the coolest Linux based applications with Moblin SDK & win great >>> prizes >>> Grand prize is a trip for two to an Open Source event anywhere in the >>> world >>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> challenge >> Build the coolest Linux based applications with Moblin SDK & win great >> prizes >> Grand prize is a trip for two to an Open Source event anywhere in the >> world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > -- > Regards, > > Sudarshan Bisht > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From kekko84 at gmail.com Thu Sep 11 10:46:17 2008 From: kekko84 at gmail.com (Francesco Argese) Date: Thu, 11 Sep 2008 10:46:17 +0200 Subject: [gst-devel] =?iso-8859-1?q?window_too_big_with_resolution_greater?= =?iso-8859-1?q?_than_352_=D7_288?= Message-ID: I have this problem: i'm trying to send towards udp a stream encoded with theora from Linux and to receive it on a windows machine (always with gstreamer) in a ARToolKit application (it do only a gst_parse_launch (pipeline_string, &error) to recover the video). The version of gstreamer used on Windows was cross-compiled by me (version 0.10.20 for core and base and 0.10.9 good, 0.10.8 bad, 0.10.9 ugly, ffmpeg lacks because i think it could require more time to cross-compile it). The two pipeline (reported below) works well if i use a low resolution (max 352 ? 288). If i try with a greater resolution, Gstreamer recognize a big window (1835099506 x 1327526501 ) but don't show me errors and don't open the window. The problem is that i have no debug informations on windows so i don't know how to find the possible error. I have built the same configuration with two Linux machine and it works well with all known resolution, higher than 320x240. I use the client with ARToolKit but the ARToolKit code doesn't give any error, too. The only signal seems to be the big dimension of the window. My two pipeline are the following: 1)gst-launch-0.10-0 videotestsrc ! video/x-raw-yuv,format=\(fourcc\)I420,width=640,height=480 ! theoraenc ! video/x-theora ! gdppay ! application/x-gdp ! udpsink host=127.0.0.1 port=5000 2)udpsrc port=5000 ! application/x-gdp ! gdpdepay ! video/x-theora ! theoradec ! video/x-raw-yuv,format=\(fourcc\)I420,width=640,height=480 ! ffmpegcolorspace ! capsfilter caps=video/x-raw-rgb,bpp=32,endianness=4321,red_mask=65280,green_mask=16711680,blue_mask=-16777216 ! identity name=artoolkit sync=false ! fakesink I'm trying it on localhost (both on Windows so) launching first 2), then 1) What could be the problem? Any suggestion? Thanks Francesco Argese From arnabsamanta at tataelxsi.co.in Thu Sep 11 12:05:43 2008 From: arnabsamanta at tataelxsi.co.in (arnabsamanta) Date: Thu, 11 Sep 2008 15:35:43 +0530 Subject: [gst-devel] queue element in gstreamer In-Reply-To: Message-ID: <00de01c913f5$f472a880$26033c0a@telxsi.com> Hi , can any body tel me what kind of input is taken by the QUEUE element in gstreamer ? is it frame by frame for a video file ? and i am going throught the code of the gstqueue.c .... how the queue->queue is getting the input ? in the init() queue->queue = g_queue_new (); is done ? it only creates a new queue . how the data flows in the queue ? regards, ~Arnab The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From julien.isorce at gmail.com Thu Sep 11 12:32:29 2008 From: julien.isorce at gmail.com (Julien Isorce) Date: Thu, 11 Sep 2008 12:32:29 +0200 Subject: [gst-devel] sub part of a movie Message-ID: <180a127d0809110332p88bee39uf3f5a3e1a7d8b6d4@mail.gmail.com> Hi, I would like to extract a sub part of a movie which is in an avi file. I can only extract a sub part from the beginnning with the following pipeline : (assume movie.avi is at 25 fps for 1 min) gst-launch filesrc location=movie.avi num_buffers=100 ! avidemux ! avimux ! filesink location=submovie.avi It extracts 4 sec from the beginning. Is there a way to extract 4 secs in the middle of the movie ? (I mean not from the beginnig, so *start* after N buffers, as I do to *end* after num_buffers) ? Thx Sincerely Julien -------------- next part -------------- An HTML attachment was scrubbed... URL: From fthiery at gmail.com Thu Sep 11 13:21:04 2008 From: fthiery at gmail.com (Florent) Date: Thu, 11 Sep 2008 13:21:04 +0200 Subject: [gst-devel] sub part of a movie In-Reply-To: <180a127d0809110332p88bee39uf3f5a3e1a7d8b6d4@mail.gmail.com> References: <180a127d0809110332p88bee39uf3f5a3e1a7d8b6d4@mail.gmail.com> Message-ID: <1efe3a6e0809110421ob7f5896w53e7e2c1b0d5b133@mail.gmail.com> Hi Julien; Did you take a look at gentrans ? C.f. "Example 2.5. Pass-through transcoding" in http://gentrans.sourceforge.net/docs/head/manual/html/entrans.html Florent -------------- next part -------------- An HTML attachment was scrubbed... URL: From fthiery at gmail.com Thu Sep 11 13:34:53 2008 From: fthiery at gmail.com (Florent) Date: Thu, 11 Sep 2008 13:34:53 +0200 Subject: [gst-devel] [gstreamer on windows] Error while network desktop streaming using screencapsrc if height is bigger than 200 Message-ID: <1efe3a6e0809110434q5f1c85fboa7b40df54b9231bd@mail.gmail.com> Hi; I tried experimenting using gstreamer on windows based on Elisa media center's binary distribution. I want to send the captured desktop (using dx9screencapsrc) towards a Linux machine. I noticed a quite strange error. Here's the pipeline: gst-launch-0.10.exe dx9screencapsrc width=1024 height=200 ! ffmpegcolorspace ! jpegenc ! udpsink host=$linux_ip port=1234 This works. However, if i specify a bigger height (ex: 768), i get: "gst_multiudpsink_render error: got send error: 0: No error ERROR: from element /pipeline0/udpsink0: GStreamer encoutered a general stream error basesrc gstbasesrc.c: gst_base_src_loop error: streaming task paused, reason error (-5)" How can i fix this ? I already tried: * playing with queues (in case there was some cpu/process overload) * using gdiscreencapsrc instead (no change) Thanks for any hint Florent -------------- next part -------------- An HTML attachment was scrubbed... URL: From techietone at gmail.com Thu Sep 11 14:02:25 2008 From: techietone at gmail.com (techie tone) Date: Thu, 11 Sep 2008 17:32:25 +0530 Subject: [gst-devel] how to determine number of channels from codec_data Message-ID: <6474b3950809110502o7aca1fa1je8a03c868dfef4e6@mail.gmail.com> Hi All, Can we derive the information about number of channels from codec_data? I want to extract the number of channels information from codec_data (for AAC-LC decoder) received from ffdemux_mov_mp4_m4a_3gp_3g2_mj2. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Thu Sep 11 14:07:23 2008 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 11 Sep 2008 14:07:23 +0200 Subject: [gst-devel] GIT test repositories In-Reply-To: References: <1220794458.2541.18.camel@putamadre> <1220950102.8650.2.camel@localhost> Message-ID: <1221134843.2551.3.camel@putamadre> On Wed, 2008-09-10 at 21:47 +0200, Andy Wingo wrote: > On Tue 09 Sep 2008 10:48, Edward Hervey writes: > > > At the general request, I've also created a commit message filter in > > order to have a saner view of the commits. > > Edward this looks great. YOU ARE A HERO > Cheers :) Alas, I had to take off my superhero stockings for a while, and realized that the conversion wasn't correct (the common submodule refs were wrong). I've been fixing the scripts used to do the conversion. I'll upload new converted modules and links to the conversion script later on today. Next step is to create a wiki page with information on the git workflow specific to GStreamer development (Changelog usage, patches, releases, etc...). If anyone's interested, help is most welcome. Edward -- It's a bird ? It's a plane ? No... it's git-filter-branch ! From thiagossantos at gmail.com Thu Sep 11 14:11:18 2008 From: thiagossantos at gmail.com (thiagoss) Date: Thu, 11 Sep 2008 09:11:18 -0300 Subject: [gst-devel] queue element in gstreamer In-Reply-To: <00de01c913f5$f472a880$26033c0a@telxsi.com> References: <00de01c913f5$f472a880$26033c0a@telxsi.com> Message-ID: The GstQueue takes *any kind* of stream and buffers it (FIFO), you should look at the chain method if you'd like to know how it receives the buffers. It also starts a new thread for pushing the data for the downstream element. On Thu, Sep 11, 2008 at 7:05 AM, arnabsamanta wrote: > Hi , > can any body tel me what kind of input is taken by the QUEUE element > in > gstreamer ? is it frame by frame for a video file ? > and i am going throught the code of the gstqueue.c .... > how the queue->queue is getting the input ? > in the init() > queue->queue = g_queue_new (); is done ? it only creates a > new queue . how > the data flows in the queue ? > regards, > ~Arnab > > > The information contained in this electronic message and any attachments to > this message are intended for the exclusive use of the addressee(s) and may > contain proprietary, confidential or privileged information. If you are not > the intended recipient, you should not disseminate, distribute or copy this > e-mail. Please notify the sender immediately and destroy all copies of this > message and any attachments contained in it. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) -------------- next part -------------- An HTML attachment was scrubbed... URL: From ved.kpl at gmail.com Thu Sep 11 14:22:03 2008 From: ved.kpl at gmail.com (ved kpl) Date: Thu, 11 Sep 2008 17:52:03 +0530 Subject: [gst-devel] queue element in gstreamer In-Reply-To: <00de01c913f5$f472a880$26033c0a@telxsi.com> References: <00de01c913f5$f472a880$26033c0a@telxsi.com> Message-ID: <7496c23f0809110522x7db17470g431151f773ca34e0@mail.gmail.com> Hi, GstQueue creates a thread boundary between the elements before it and the elements after it.The Gstqueue creates a Gqueue that is used to hold the input buffers. A separate thread reads these buffers from the Gqueue and gives out to the next element. So you have one thread(chain func being called as a result of gst_pad_push by previous element) filling the Gqueue and other(gst_queue_loop started on srcpad) emptying it. The GstQueue is an gstreamer element that encapsulates all this. The buffers could be anything .Frames, incomplete frames, etc. That depends on what the previous element is giving. If the GStQueue is right after the demuxer and the demuxer gives one video frame per Gstbuffer, then the Gstqueue will be getting a complete frame and storing it in Gqueue. Ved On Thu, Sep 11, 2008 at 3:35 PM, arnabsamanta wrote: > Hi , > can any body tel me what kind of input is taken by the QUEUE element in > gstreamer ? is it frame by frame for a video file ? > and i am going throught the code of the gstqueue.c .... > how the queue->queue is getting the input ? > in the init() > queue->queue = g_queue_new (); is done ? it only creates a new queue . how > the data flows in the queue ? > regards, > ~Arnab > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From wim.taymans at gmail.com Thu Sep 11 14:26:35 2008 From: wim.taymans at gmail.com (Wim Taymans) Date: Thu, 11 Sep 2008 14:26:35 +0200 Subject: [gst-devel] [gstreamer on windows] Error while network desktop streaming using screencapsrc if height is bigger than 200 In-Reply-To: <1efe3a6e0809110434q5f1c85fboa7b40df54b9231bd@mail.gmail.com> References: <1efe3a6e0809110434q5f1c85fboa7b40df54b9231bd@mail.gmail.com> Message-ID: <1221135995.6806.12.camel@metal> On Thu, 2008-09-11 at 13:34 +0200, Florent wrote: > Hi; > > I tried experimenting using gstreamer on windows based on Elisa media > center's binary distribution. > > I want to send the captured desktop (using dx9screencapsrc) towards a > Linux machine. I noticed a quite strange error. > > Here's the pipeline: > gst-launch-0.10.exe dx9screencapsrc width=1024 height=200 ! > ffmpegcolorspace ! jpegenc ! udpsink host=$linux_ip port=1234 > > This works. However, if i specify a bigger height (ex: 768), i get: > > "gst_multiudpsink_render error: got send error: 0: No error > ERROR: from element /pipeline0/udpsink0: GStreamer encoutered a > general stream error > basesrc gstbasesrc.c: gst_base_src_loop error: > streaming task paused, reason error (-5)" > Your compressed jpeg frame is likely larger than the maximum MTU or bigger than the maximum allowed size of a UDP packet. This will not work. You probably want to use some kind of RTP payloader if you send data over UDP. Wim > > How can i fix this ? I already tried: > * playing with queues (in case there was some cpu/process overload) > * using gdiscreencapsrc instead (no change) > > Thanks for any hint > > Florent > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ensonic at hora-obscura.de Thu Sep 11 15:31:43 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Thu, 11 Sep 2008 16:31:43 +0300 Subject: [gst-devel] muxers, timestamps, sparse and continuous recordings In-Reply-To: References: <48C792A4.8040700@hora-obscura.de> Message-ID: <48C91DBF.40901@hora-obscura.de> hi, Eric Zhang schrieb: > Hi, Stefan: > > I think your pipeline is using GstSystemClock because you > mentioned the source is a live element. If it is, the clock will keep > increasing even if the pipeline is paused. This makes the timestamp > noncontinuous. To generate a continuous timestamp, I think you can try > to use the clock provided by your sink elements. Maybe this is not > easy because the live element is different with other source elements. I meant pausing as on the application level. The videosrc ! tee name=t ! queue ! xvimagesink runs continously. It only t. ! queue ! encoder ! muxer ! filesink that get paused. Stefan > > Eric > > 2008/9/10 Stefan Kost > > > hi, > > i was wondering how muxers should handle timestamps on incoming > buffers. > Assume an applications that shows video from a camera. When you > click a > button it records to file, allowing to pause and unpause in > between. The > recorded file should have a continuous stream. If I don't do any > special > casing this is not the case. > > 1) When I playback the recorded file, I have an initial pause before > video start (if I pressed record after two seconds, the video will > start > after two seconds). > > 2) If I pause in between, also in the playback there is a pause. > > Right now I work around with a pad probe that looks at disconts to > aggregate a time_stamp_offset and correct all incoming buffers by > subtracting that. It works but probably is not the right way. I > believe > this involves the use of segments, but I am not sure how. Also both > behaviors might be valid (having a sparse and having a continuous > stream). So the application would somehow be involved to select the > desired behavior. Any comments? > > > Stefan > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win > great prizes > Grand prize is a trip for two to an Open Source event anywhere in > the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From francis.meyvis at gmail.com Thu Sep 11 16:08:28 2008 From: francis.meyvis at gmail.com (franchan) Date: Thu, 11 Sep 2008 16:08:28 +0200 Subject: [gst-devel] 2 questions: threading and faad In-Reply-To: <8456544a0809100050wc1fe123l207f5158a00e61ca@mail.gmail.com> References: <8456544a0809100050wc1fe123l207f5158a00e61ca@mail.gmail.com> Message-ID: <8456544a0809110708n24a6b656w22e64ad83c5f5e97@mail.gmail.com> Hello, Thanks for sharing your insights! >> The typefind element does typefinding in the state-change to PAUSED. >> This is arguably the wrong thing to do but it's currently done like that >> because we otherwise would need a scheduling mode switch from the >> streaming thread. I'm not quite sure how that would work... Well, I was inaccurate about that, the problem popped up in the paused state indeed. But by then the email was already send already. I'm not sure why it would be wrong doing it in the PAUSE state. Although I don't work long with gstreamer, it simple state model makes sense to me: READY state: don't make contact with server's yet but have allocated as much resources as necessary. PAUSE: ready for playing as soon as state changes to PLAYING. So by going from READY to the PAUSE state, it should start connecting to the network, get initial content to do typefinding. It would be nice if also in PAUSE transition, the playbin buffering mechanism would be in place already, so that the whole gmain_loop does not block in a "read()". > Ah, you're right. Sorry about that, franchan. Wim: we should think > about a fix for that, though probably for 0.11? > > So: yes, apparently we do typefinding from the main thread, if we're > in pull mode. That's unfortunate. However, it doesn't work this way in > push mode. Since I recommended using push mode _anyway_, that's not > too bad - you'll just need to change your plugin to work in push mode. Perhaps I should use another demuxer (now this was auto selected by playbin for the h264/AAC content) or use some API I did not find yet? At first I started with a progressive download plug-in for push operation. Then I found qtdemux refused to seek with this source plug-in. I made these conclusions back then when reading and tracing qtdemux.c (I write just what I remember from last weeks so I might miss) qtdemux_sink_activate_pull() sets its data member demux->pullbased=true. gst_qtdemux_handle_src_event() checks this data member in case of a GST_EVENT_SEEK event. And it concludes with GST_DEBUG_OBJECT (qtdemux, "cannot seek in streaming mode") if pullbased is false. qtdemux_sink_activate_pull() resets the pullbased flag when the qtdemux_sink_activate() concludes through the gst_pad_check_pull_range() that the source plug-in was push only because it missed a _check_range_get() function, typical for pull plug-ins ... so I thought ... So I decided to rewrite my plug-in at that time to not use a push model. I was surprised to see this demux code because gstreamer's documentation, seems to favor push operation, as you also tell here ... > There are a number of other network sources that operate in push mode > and support streaming (e.g. the http sources), those might be useful > examples for you. Well, I looked into the soup/neon based http source plug-ins, but these are complicated :-) Now the pull based plug-in is very simple (note that the HTTP downloading work is outside this plug-in and based on legacy code). For now the easiest for me is to have a delayed "PAUSE/PLAY". Once the legacy code tells me that it has a fair amount of buffered data, I put the pipeline into the PAUSE/PLAY state and typefinding phase can likely finish correctly. (I'm just afraid, how much data do I need exactly. But that's a completely different problem) For now about 300K seems to be enough to get the pipeline rolling ... (moov bag in front of the mp4 file of course) Thanks, franchan From bilboed at gmail.com Thu Sep 11 17:06:56 2008 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 11 Sep 2008 17:06:56 +0200 Subject: [gst-devel] GIT test repositories In-Reply-To: <1221134843.2551.3.camel@putamadre> References: <1220794458.2541.18.camel@putamadre> <1220950102.8650.2.camel@localhost> <1221134843.2551.3.camel@putamadre> Message-ID: <1221145616.2551.5.camel@putamadre> Hi again, So new *fixed* repositories are uploaded, at the same location. For those interested, the scripts I used are located here: http://git.collabora.co.uk/?p=user/edward/gst-git-migration;a=summary Edward On Thu, 2008-09-11 at 14:07 +0200, Edward Hervey wrote: > On Wed, 2008-09-10 at 21:47 +0200, Andy Wingo wrote: > > On Tue 09 Sep 2008 10:48, Edward Hervey writes: > > > > > At the general request, I've also created a commit message filter in > > > order to have a saner view of the commits. > > > > Edward this looks great. YOU ARE A HERO > > > > Cheers :) > > Alas, I had to take off my superhero stockings for a while, and > realized that the conversion wasn't correct (the common submodule refs > were wrong). > > I've been fixing the scripts used to do the conversion. I'll upload > new converted modules and links to the conversion script later on today. > > Next step is to create a wiki page with information on the git > workflow specific to GStreamer development (Changelog usage, patches, > releases, etc...). If anyone's interested, help is most welcome. > > Edward > > -- > It's a bird ? It's a plane ? > No... it's git-filter-branch ! > From knocte at gmail.com Thu Sep 11 17:26:40 2008 From: knocte at gmail.com (=?ISO-8859-1?Q?=22Andr=E9s_G=2E_Aragoneses=22?=) Date: Thu, 11 Sep 2008 17:26:40 +0200 Subject: [gst-devel] Where to report/fix this issue? Message-ID: Hello, I recently had a playback problem and I don't know where & how to report/fix it. Please refer to http://www.mail-archive.com/packman at links2linux.de/msg01686.html for more information (I don't want to crosspost to a lot of places). Thanks, any help will be appreciated. Andr?s -- From julien.isorce at gmail.com Thu Sep 11 18:07:43 2008 From: julien.isorce at gmail.com (Julien Isorce) Date: Thu, 11 Sep 2008 18:07:43 +0200 Subject: [gst-devel] sub part of a movie In-Reply-To: <1efe3a6e0809110421ob7f5896w53e7e2c1b0d5b133@mail.gmail.com> References: <180a127d0809110332p88bee39uf3f5a3e1a7d8b6d4@mail.gmail.com> <1efe3a6e0809110421ob7f5896w53e7e2c1b0d5b133@mail.gmail.com> Message-ID: <180a127d0809110907o25a44dc3v1546dee64a78fefa@mail.gmail.com> Hi Florent, Sounds good but when I tried this : (movie.avi is about 1 min) entrans.py -s seek-key -c 25-50 -- --raw filesrc location=movie.avi ! avidemux ! avimux ! filesink location=submovie.avi I got: 0:00:00.057245653 23852 0x8270cc8 ERROR python entrans.py:2354:excepthook: File "entrans.py", line 1825, in cb_started walk = self.walk_pipeline(self.nonlin.pipeline) File "entrans.py", line 1780, in walk_pipeline clone_element(element).get_property(pspec.name)): SystemError: error return without exception set <<<< Now reached PLAYING state >>>> 0:00:00.060644767 23852 0x8270cc8 ERROR python entrans.py:2354:excepthook: File "entrans.py", line 1902, in cb_playing walk = self.walk_pipeline(self.nonlin.pipeline) File "entrans.py", line 1780, in walk_pipeline clone_element(element).get_property(pspec.name)): SystemError: error return without exception set I am sure I am doing something wrong. I compiled gentrans myself so maybe I have not all the dependencies. Any help ? (why dam this : Cannot set property on dam0 before streaming has started ?) thx Julien 2008/9/11 Florent > Hi Julien; > > Did you take a look at gentrans ? > > C.f. "Example 2.5. Pass-through transcoding" in > http://gentrans.sourceforge.net/docs/head/manual/html/entrans.html > > Florent > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From manauw at skynet.be Thu Sep 11 18:56:48 2008 From: manauw at skynet.be (Mark Nauwelaerts) Date: Thu, 11 Sep 2008 18:56:48 +0200 Subject: [gst-devel] sub part of a movie In-Reply-To: <180a127d0809110907o25a44dc3v1546dee64a78fefa@mail.gmail.com> References: <180a127d0809110332p88bee39uf3f5a3e1a7d8b6d4@mail.gmail.com> <1efe3a6e0809110421ob7f5896w53e7e2c1b0d5b133@mail.gmail.com> <180a127d0809110907o25a44dc3v1546dee64a78fefa@mail.gmail.com> Message-ID: <48C94DD0.1010806@skynet.be> The exception is most likely an effect of bug http://bugzilla.gnome.org/show_bug.cgi?id=540221, the fix for which may not be present in your gst-python (or otherwise another bug very much like it :) ) It can be worked around (IIRC) by adding the option --ignore-prop '.*sink.*' (so it does not access the sink's property leading to the crash). I also have some doubts about the chances for success of the pipeline (e.g. no dam in it, entrans uses this element to direct the seek to ...) Using a more "conventional" pipeline as exampled in the link mentioned below may be advisable. Mark. Julien Isorce wrote: > Hi Florent, > > Sounds good but when I tried this : > > (movie.avi is about 1 min) > > entrans.py -s seek-key -c 25-50 -- --raw filesrc location=movie.avi ! > avidemux ! avimux ! filesink location=submovie.avi > > I got: > > 0:00:00.057245653 23852 0x8270cc8 ERROR python > entrans.py:2354:excepthook: > File "entrans.py", line 1825, in cb_started > walk = self.walk_pipeline(self.nonlin.pipeline) > File "entrans.py", line 1780, in walk_pipeline > clone_element(element).get_property(pspec.name )): > > SystemError: error return without exception set > > <<<< Now reached PLAYING state >>>> > 0:00:00.060644767 23852 0x8270cc8 ERROR python > entrans.py:2354:excepthook: > File "entrans.py", line 1902, in cb_playing > walk = self.walk_pipeline(self.nonlin.pipeline) > File "entrans.py", line 1780, in walk_pipeline > clone_element(element).get_property(pspec.name )): > > SystemError: error return without exception set > > > I am sure I am doing something wrong. > I compiled gentrans myself so maybe I have not all the dependencies. > > Any help ? > > (why dam this : Cannot set property on dam0 before streaming has started ?) > > thx > > Julien > > > 2008/9/11 Florent > > > Hi Julien; > > Did you take a look at gentrans ? > > C.f. "Example 2.5. Pass-through transcoding" in > http://gentrans.sourceforge.net/docs/head/manual/html/entrans.html > > Florent > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win > great prizes > Grand prize is a trip for two to an Open Source event anywhere in > the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From msmith at xiph.org Thu Sep 11 18:59:39 2008 From: msmith at xiph.org (Michael Smith) Date: Thu, 11 Sep 2008 09:59:39 -0700 Subject: [gst-devel] =?iso-8859-1?q?window_too_big_with_resolution_greater?= =?iso-8859-1?q?_than_352_=D7_288?= In-Reply-To: References: Message-ID: <3c1737210809110959n2bd87ae6n918bc4c3500c8518@mail.gmail.com> On Thu, Sep 11, 2008 at 1:46 AM, Francesco Argese wrote: > > My two pipeline are the following: > > 1)gst-launch-0.10-0 videotestsrc ! > video/x-raw-yuv,format=\(fourcc\)I420,width=640,height=480 ! theoraenc > ! video/x-theora ! gdppay ! application/x-gdp ! udpsink host=127.0.0.1 > port=5000 > > 2)udpsrc port=5000 ! application/x-gdp ! gdpdepay ! video/x-theora ! > theoradec ! video/x-raw-yuv,format=\(fourcc\)I420,width=640,height=480 > ! ffmpegcolorspace ! capsfilter > caps=video/x-raw-rgb,bpp=32,endianness=4321,red_mask=65280,green_mask=16711680,blue_mask=-16777216 > ! identity name=artoolkit sync=false ! fakesink GDP (which is what you're using, with gdppay and gdpdepay) is not designed for lossy transports. Probably the higher resolution means your encoded frames are larger, perhaps exceeding the MTU on the network, or just being dropped since you're using UDP. Use a TCP transport, or a network protocol designed to handle lossy transports. Mike From msmith at xiph.org Thu Sep 11 19:01:01 2008 From: msmith at xiph.org (Michael Smith) Date: Thu, 11 Sep 2008 10:01:01 -0700 Subject: [gst-devel] how to determine number of channels from codec_data In-Reply-To: <6474b3950809110502o7aca1fa1je8a03c868dfef4e6@mail.gmail.com> References: <6474b3950809110502o7aca1fa1je8a03c868dfef4e6@mail.gmail.com> Message-ID: <3c1737210809111001j7b6483feubbb6b1c1e3ec6504@mail.gmail.com> On Thu, Sep 11, 2008 at 5:02 AM, techie tone wrote: > Hi All, > > Can we derive the information about number of channels from codec_data? > > I want to extract the number of channels information from codec_data (for > AAC-LC decoder) received from ffdemux_mov_mp4_m4a_3gp_3g2_mj2. Do not use that demuxer. Use qtdemux instead, it works much much better. The codec_data for AAC does include information about the number of channels, but it's an AAC-specific thing, you'd need to parse it yourself to find out. The channel count is probably also in the caps anyway (from the demuxer, if you use a decent demuxer), so you don't need to do that. Mike From rajshyam at gmail.com Thu Sep 11 22:23:45 2008 From: rajshyam at gmail.com (Raj Swaminathan) Date: Thu, 11 Sep 2008 15:23:45 -0500 Subject: [gst-devel] Media Player development Questions In-Reply-To: <1220256343.6287.6.camel@mini.centricular.net> References: <6438d8660808281108j2fa69eddtac006065927ecba2@mail.gmail.com> <1220256343.6287.6.camel@mini.centricular.net> Message-ID: <6438d8660809111323p5d7599c4gfd4fbe87ce2394b6@mail.gmail.com> Tim, I followed your suggestions ... got playbin and used its volume property. I also created a separate thread for running the gst loop and checking bus messages and so far things are good. Thanks for your help. Thanks Ajay, Eric for your suggestions too. regards, raj Thanks for your responses. Sorry for getting back so late ... just in from vacation... *>For stream volume control you should use playbin's "volume" property. >The GStreamer mixer interface (GstMixer) is to control hardware mixer >levels, you usually don't want that in a playback application. It's also >not very nice to use. If you don't use playbin, add a volume element to >your pipeline (alternatively: audiosinks which support stream volumes >will also have a "volume" property). * Does this allow volume changing when the pipeline is playing .. if so how would you do that ? On Mon, Sep 1, 2008 at 3:05 AM, Tim-Philipp M?ller wrote: > On Thu, 2008-08-28 at 13:08 -0500, Raj Swaminathan wrote: > > Hi, > > > I am new to Glib programming and gstreamer. I am trying to build a > > basic media player application. So far i have gstreamer pipelines up > > and rendering various media formats. > > For a media player you'll probably want to use playbin (or even > playbin2) from gst-plugins-base. There's no need to create your own > pipelines. > > > > > 1) Im trying to reference the gstalsamixer plugin for volume control > > and muting. I understand its an interface and im trying to find out > > whats the best way to access the interface methods like > > gst_alsa_alsa_mixer_get_volume etc ... > > Could somebody please provide a code snippet that can show this ? > > For stream volume control you should use playbin's "volume" property. > The GStreamer mixer interface (GstMixer) is to control hardware mixer > levels, you usually don't want that in a playback application. It's also > not very nice to use. If you don't use playbin, add a volume element to > your pipeline (alternatively: audiosinks which support stream volumes > will also have a "volume" property). > > > > 2) When designing the media player, whats the best way to accept user > > input. Do i spool a separate thread for rendering media while the main > > thread waits for user input or is there a better way within gstreamer > > to do this? > > GStreamer does all its playback in threads of its own anyway, so the > main thread is yours. All you need to do is check the > pipeline's/playbin's GstBus for messages (errors, tags, state changes) > from time to time. > > Cheers > -Tim > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rajshyam at gmail.com Thu Sep 11 22:44:26 2008 From: rajshyam at gmail.com (Raj Swaminathan) Date: Thu, 11 Sep 2008 15:44:26 -0500 Subject: [gst-devel] Souphttpsrc and playbin Message-ID: <6438d8660809111344h1cafc389l75569f0bed9103f5@mail.gmail.com> Hi, I am trying to stream an Internet Radio Channel with playbin. I am going through a proxy and can use gst-launch in this way to successfully play the station: gst-launch souphttpsrc proxy=http://proxyURL location=http://myIRadio.com. ! mad ! alsasink I do not need to use gst-launch and am actually building my own software media player .... When i do not setup proxy but just run my mediaplayer with playbin, I just wait on an endless loop and get the following debug message on cancelling.. gstsouphttpsrc.c(683): gst_soup_http_src_finished_cb(): /play/source libsoup status code 1 Error: Cancelled So playbin is trying to use souphttpsrc but Im assuming the error is becoz the proxy is not set ..... Can someone plz suggest how to set the proxy property with souphttpsrc and still use playbin v1 ? Thanks, raj -------------- next part -------------- An HTML attachment was scrubbed... URL: From nicolas.m.zhang at gmail.com Fri Sep 12 03:24:19 2008 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Fri, 12 Sep 2008 09:24:19 +0800 Subject: [gst-devel] Souphttpsrc and playbin In-Reply-To: <6438d8660809111344h1cafc389l75569f0bed9103f5@mail.gmail.com> References: <6438d8660809111344h1cafc389l75569f0bed9103f5@mail.gmail.com> Message-ID: Hi, Raj: That's easy. First, connect `notify::source' signal of playbin: // connect signal g_signal_connect(G_OBJECT(playbin), "notify::source", G_CALLBACK(cb_playbin_notify_source), NULL); Then the callback function(I am working on rtsp recently so the example source below is a rtspsrc. In your program, this should be souphttpsrc): static void cb_playbin_notify_source(GObject *obj, GParamSpec *param, gpointer u_data) { // check whether this is rtsp source gchar *objname = GST_OBJECT_NAME(obj); g_message("objname is %s", objname); // check whether has a `protocols' property if (g_object_class_find_property(G_OBJECT_GET_CLASS(obj), "source")) { GObject *source_element; g_object_get(obj, "source", &source_element, NULL); if (g_object_class_find_property(G_OBJECT_GET_CLASS(source_element), "protocols")) { g_object_set(source_element, "protocols", 1, NULL); } g_object_unref(source_element); } } Eric Zhang 2008/9/12 Raj Swaminathan > Hi, > > I am trying to stream an Internet Radio Channel with playbin. I am going > through a proxy and can use gst-launch in this way to successfully play the > station: > > gst-launch souphttpsrc proxy=http://proxyURL location=http://myIRadio.com. > ! mad ! alsasink > > I do not need to use gst-launch and am actually building my own software > media player .... > > When i do not setup proxy but just run my mediaplayer with playbin, I just > wait on an endless loop and get the following debug message on cancelling.. > gstsouphttpsrc.c(683): gst_soup_http_src_finished_cb(): /play/source > libsoup status code 1 > Error: Cancelled > > So playbin is trying to use souphttpsrc but Im assuming the error is becoz > the proxy is not set ..... > Can someone plz suggest how to set the proxy property with souphttpsrc and > still use playbin v1 ? > > Thanks, > raj > > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From learning_gst at hotmail.com Fri Sep 12 08:15:07 2008 From: learning_gst at hotmail.com (learning gst) Date: Fri, 12 Sep 2008 06:15:07 +0000 Subject: [gst-devel] How can we compile GStreamer for ARM In-Reply-To: <3c1737210809112225i4e5f3c34q214a959851b7d58b@mail.gmail.com> References: <002401c9148d$416e6d30$340318ac@LGE.NET> <3c1737210809112225i4e5f3c34q214a959851b7d58b@mail.gmail.com> Message-ID: Hello, I hope to compile GStreammer for ARM. But I don't know how to compile? Please help me. Thanks _________________________________________________________________ Connect to the next generation of MSN Messenger? http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline -------------- next part -------------- An HTML attachment was scrubbed... URL: From vtorri at univ-evry.fr Fri Sep 12 08:31:27 2008 From: vtorri at univ-evry.fr (Vincent Torri) Date: Fri, 12 Sep 2008 08:31:27 +0200 (CEST) Subject: [gst-devel] How can we compile GStreamer for ARM In-Reply-To: References: <002401c9148d$416e6d30$340318ac@LGE.NET> <3c1737210809112225i4e5f3c34q214a959851b7d58b@mail.gmail.com> Message-ID: On Fri, 12 Sep 2008, learning gst wrote: > I hope to compile GStreammer for ARM. But I don't know how to compile? Please help me. use a cross-compilation toolchain. I'm sure google can help you (for example http://www.scratchbox.org/, but there are certainly other toolchain) So, I think that you have to: 1) set up the toolchain 2) compile the dependencies (iconv, glib, etc...) for the arm target 3) compile gstreamer for the arm target As an example, I plan to have gstreamer on Windows CE devices with cegcc. regards Vincent Torri From sachinpandhare at gmail.com Fri Sep 12 08:36:14 2008 From: sachinpandhare at gmail.com (Sachin Pandhare) Date: Fri, 12 Sep 2008 12:06:14 +0530 Subject: [gst-devel] audio / video sources in gstreamer Message-ID: <72cf309c0809112336g34c028ebj5512595a5cfbc23d@mail.gmail.com> Hi, could you please guide me on which are the audio/video sources in gstreamer which can be used with camera as a source for video capture (not just image capture)? i know about v4l2, dv1394src, alsasrc. is ximagesrc just for image capture or can it be used for video capture as well? what is the difference between dvbsrc present in dvb and dvbsrc Thanks, Sachin -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Fri Sep 12 08:58:40 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Fri, 12 Sep 2008 09:58:40 +0300 Subject: [gst-devel] Link xvid with ffmpeg's mp4 muxer In-Reply-To: <48f4838d0809101334q2771ee18kfb6df1c349373f8b@mail.gmail.com> References: <48f4838d0809101334q2771ee18kfb6df1c349373f8b@mail.gmail.com> Message-ID: <48CA1320.6090602@hora-obscura.de> Hi, Alexandre schrieb: > Hi, > > I'm trying to mux the GStreamer xvidenc element with ffmux_mp4, to > create a Mpeg 4 file with XVid encoded video, but it's not being > succesfull. The xvidenc caps are video/x-xvid, and ffmux supports either > mpeg v4 video or video/x-divx. Hopefully qtmux is merged to -plugins-bad soon. Lets make sure it support it then. Stefan > > Is there a way to link them, or are they incompatible? > > -- > Alexandre Rosenfeld > > EngComp 06 - USP S?o Carlos > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ensonic at hora-obscura.de Fri Sep 12 09:01:45 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Fri, 12 Sep 2008 10:01:45 +0300 Subject: [gst-devel] audio / video sources in gstreamer In-Reply-To: <72cf309c0809112336g34c028ebj5512595a5cfbc23d@mail.gmail.com> References: <72cf309c0809112336g34c028ebj5512595a5cfbc23d@mail.gmail.com> Message-ID: <48CA13D9.7090407@hora-obscura.de> Sachin Pandhare schrieb: > Hi, > could you please guide me on which are the audio/video sources in > gstreamer which can be used with camera as a source for video capture > (not just image capture)? > > i know about v4l2, dv1394src, alsasrc. Have a look at gst-inspector to browse elements by type. v4l2src, v4lsrc are both for video capture under linux. For audio there is alsasrc, pulsesrc, jacksrc, osssrc, oss4src .... > > is ximagesrc just for image capture or can it be used for video capture > as well? This can be used to capture your desktop. Stefan > > what is the difference between dvbsrc present in dvb and dvbsrc > > Thanks, > Sachin > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ensonic at hora-obscura.de Fri Sep 12 09:02:50 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Fri, 12 Sep 2008 10:02:50 +0300 Subject: [gst-devel] How can we compile GStreamer for ARM In-Reply-To: References: <002401c9148d$416e6d30$340318ac@LGE.NET> <3c1737210809112225i4e5f3c34q214a959851b7d58b@mail.gmail.com> Message-ID: <48CA141A.9010800@hora-obscura.de> Vincent Torri schrieb: > > On Fri, 12 Sep 2008, learning gst wrote: > >> I hope to compile GStreammer for ARM. But I don't know how to compile? Please help me. > > use a cross-compilation toolchain. I'm sure google can help you (for > example http://www.scratchbox.org/, but there are certainly other > toolchain) > > So, I think that you have to: > > 1) set up the toolchain > 2) compile the dependencies (iconv, glib, etc...) for the arm target > 3) compile gstreamer for the arm target > > As an example, I plan to have gstreamer on Windows CE devices with cegcc. Is glib buildable on WinCE therese days? Stefan > > regards > > Vincent Torri > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ensonic at hora-obscura.de Fri Sep 12 09:04:19 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Fri, 12 Sep 2008 10:04:19 +0300 Subject: [gst-devel] Gstreamer v4l2 to fb on DM6446 In-Reply-To: <652954.65319.qm@web32208.mail.mud.yahoo.com> References: <652954.65319.qm@web32208.mail.mud.yahoo.com> Message-ID: <48CA1473.2060306@hora-obscura.de> ketan deshpande schrieb: > Hello everybody, > > I am working on TI's DM6446. I was trying to run the pipeline from > V4L2 to FB(released by TI). > > gst-launch-0.10 v4l2src ! fbvideosink device=/dev/fb/3 > > but I keep getting the set of following warnings which lateron fails > saying cannot negotiate the caps. > > WARNING: from element /pipeline0/v4l2src0: Got unexpected frame size of > 884736 instead of 829440. > Additional debug info: > gstv4l2src.c(1082): gst_v4l2src_get_mmap (): /pipeline0/v4l2src0 > > I tried to give the exact width and height along with the exact > fourcc format and framerate but couldnt achieve any success. > I am using the good plugins version 0.10.8. Whats width, height and colorspace format? Stefan > > Can somebody suggest us as to what might be wrong in our case? > > regards, > -Ketan > > ------------------------------------------------------------------------ > Did you know? You can CHAT without downloading messenger. Click here > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From julien.isorce at gmail.com Fri Sep 12 11:10:38 2008 From: julien.isorce at gmail.com (Julien Isorce) Date: Fri, 12 Sep 2008 11:10:38 +0200 Subject: [gst-devel] sub part of a movie In-Reply-To: <48C94DD0.1010806@skynet.be> References: <180a127d0809110332p88bee39uf3f5a3e1a7d8b6d4@mail.gmail.com> <1efe3a6e0809110421ob7f5896w53e7e2c1b0d5b133@mail.gmail.com> <180a127d0809110907o25a44dc3v1546dee64a78fefa@mail.gmail.com> <48C94DD0.1010806@skynet.be> Message-ID: <180a127d0809120210w3237ef9dv7576306b77a89a64@mail.gmail.com> Hi Mark, Thanks to GEtrans I can now easily extract a sub part of a movie so that 's cool for my needs. ** This cammand works fine to cut a sub part of 15 sec of a movie. :P entrans.py --ignore-prop '.*sink*.' -s seek-key -c 25-40 --dam -- --raw filesrc location=movie.avi ! avidemux name=demux avimux name=mux ! filesink location=sub.avi demux.video_00 ! queue ! dam ! queue ! mux.video_0 demux.audio_00 ! queue ! dam ! queue ! mux.audio_00 ** The following seems to work but I cannot play the result sub2.avi: entrans.py --ignore-prop '.*sink.*' -s seek-key -c 25-40 --dam -- --raw filesrc location=movie.avi ! avidemux ! queue ! dam ! queue ! avimux ! filesink location=sub2.avi ** And the following causes a seg fault: entrans.py --ignore-prop '.*sink.*' -s seek-key -c 25-40 --dam -- --raw filesrc location=movie.avi ! avidemux ! dam ! avimux ! filesink location=sub3.avi I am sure I am not using correctly entrans in the 2 last pipelines :P (Also it was cool to see an element "detectinter" ...) Sincerely J.I. 2008/9/11 Mark Nauwelaerts > > The exception is most likely an effect of bug > http://bugzilla.gnome.org/show_bug.cgi?id=540221, the fix for which may > not be > present in your gst-python (or otherwise another bug very much like it :) ) > > It can be worked around (IIRC) by adding the option --ignore-prop > '.*sink.*' > (so it does not access the sink's property leading to the crash). > > I also have some doubts about the chances for success of the pipeline > (e.g. no dam in it, entrans uses this element to direct the seek to ...) > Using a more "conventional" pipeline as exampled in the link mentioned > below may > be advisable. > > Mark. > > Julien Isorce wrote: > > Hi Florent, > > > > Sounds good but when I tried this : > > > > (movie.avi is about 1 min) > > > > entrans.py -s seek-key -c 25-50 -- --raw filesrc location=movie.avi ! > > avidemux ! avimux ! filesink location=submovie.avi > > > > I got: > > > > 0:00:00.057245653 23852 0x8270cc8 ERROR python > > entrans.py:2354:excepthook: > > File "entrans.py", line 1825, in cb_started > > walk = self.walk_pipeline(self.nonlin.pipeline) > > File "entrans.py", line 1780, in walk_pipeline > > clone_element(element).get_property(pspec.name >)): > > > > SystemError: error return without exception set > > > > <<<< Now reached PLAYING state >>>> > > 0:00:00.060644767 23852 0x8270cc8 ERROR python > > entrans.py:2354:excepthook: > > File "entrans.py", line 1902, in cb_playing > > walk = self.walk_pipeline(self.nonlin.pipeline) > > File "entrans.py", line 1780, in walk_pipeline > > clone_element(element).get_property(pspec.name >)): > > > > SystemError: error return without exception set > > > > > > I am sure I am doing something wrong. > > I compiled gentrans myself so maybe I have not all the dependencies. > > > > Any help ? > > > > (why dam this : Cannot set property on dam0 before streaming has started > ?) > > > > thx > > > > Julien > > > > > > 2008/9/11 Florent > > > > > Hi Julien; > > > > Did you take a look at gentrans ? > > > > C.f. "Example 2.5. Pass-through transcoding" in > > http://gentrans.sourceforge.net/docs/head/manual/html/entrans.html > > > > Florent > > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > > challenge > > Build the coolest Linux based applications with Moblin SDK & win > > great prizes > > Grand prize is a trip for two to an Open Source event anywhere in > > the world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > ------------------------------------------------------------------------ > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From chhail at mail2.sysu.edu.cn Fri Sep 12 11:36:41 2008 From: chhail at mail2.sysu.edu.cn (Chen Hailiang) Date: Fri, 12 Sep 2008 17:36:41 +0800 Subject: [gst-devel] Beginner Problem:How to compile plugin? Message-ID: <20080912092506.M78596@mail2.sysu.edu.cn> Hi,all; I'm studying using 'make_element' tool to generate a filter plugin.but I don't know how to compile my source codes to the library. From google, it seems that "autogen.sh" or "libtool" may help. but my question is: 1. how to use autogen.sh? 2. I have installed the libtool using "apt-get install libtool"(my linux is Ubuntu),is it correct?if it's correct, how can I generate the Makefile?or someone can show me a template? thank's everyone and wish a good China mid-autumn festival to everybody all around the world. Chen -- Best regards From thiagossantos at gmail.com Fri Sep 12 11:50:44 2008 From: thiagossantos at gmail.com (thiagoss) Date: Fri, 12 Sep 2008 06:50:44 -0300 Subject: [gst-devel] Beginner Problem:How to compile plugin? In-Reply-To: <20080912092506.M78596@mail2.sysu.edu.cn> References: <20080912092506.M78596@mail2.sysu.edu.cn> Message-ID: On Fri, Sep 12, 2008 at 6:36 AM, Chen Hailiang wrote: > Hi,all; > I'm studying using 'make_element' tool to generate a filter plugin.but I > don't know how to compile my source codes to the library. From google, it > seems that "autogen.sh" or "libtool" may help. but my question is: > 1. how to use autogen.sh? Just run it as a script, it will complain if you don't have some lib installed, so you go and apt-get it. > > 2. I have installed the libtool using "apt-get install libtool"(my linux > is > Ubuntu),is it correct?if it's correct, how can I generate the Makefile?or > someone can show me a template? Autogen will create a makefile for you. > > thank's everyone and wish a good China mid-autumn festival to everybody > all around the world. > > > Chen > -- > Best regards > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Fri Sep 12 11:53:26 2008 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 12 Sep 2008 10:53:26 +0100 Subject: [gst-devel] Souphttpsrc and playbin In-Reply-To: <6438d8660809111344h1cafc389l75569f0bed9103f5@mail.gmail.com> References: <6438d8660809111344h1cafc389l75569f0bed9103f5@mail.gmail.com> Message-ID: <1221213206.18505.5.camel@mini.centricular.net> On Thu, 2008-09-11 at 15:44 -0500, Raj Swaminathan wrote: Hi, > So playbin is trying to use souphttpsrc but Im assuming the error is > becoz the proxy is not set ..... > Can someone plz suggest how to set the proxy property with souphttpsrc > and still use playbin v1 ? One possibility already mentioned by Eric is to connect to the notify::source signal and set up the source in the callback. This is the recommended way of configuring playbin sources (also for things like "device" properties and the like). In your case you could also just set the http_proxy environment variable (g_set_env) - if you're lucky souphttpsrc will take that into account and configure itself accordingly. Cheers -Tim From vtorri at univ-evry.fr Fri Sep 12 12:20:58 2008 From: vtorri at univ-evry.fr (Vincent Torri) Date: Fri, 12 Sep 2008 12:20:58 +0200 (CEST) Subject: [gst-devel] How can we compile GStreamer for ARM In-Reply-To: <48CA141A.9010800@hora-obscura.de> References: <002401c9148d$416e6d30$340318ac@LGE.NET> <3c1737210809112225i4e5f3c34q214a959851b7d58b@mail.gmail.com> <48CA141A.9010800@hora-obscura.de> Message-ID: On Fri, 12 Sep 2008, Stefan Kost wrote: > Vincent Torri schrieb: >> >> On Fri, 12 Sep 2008, learning gst wrote: >> >>> I hope to compile GStreammer for ARM. But I don't know how to compile? Please help me. >> >> use a cross-compilation toolchain. I'm sure google can help you (for >> example http://www.scratchbox.org/, but there are certainly other >> toolchain) >> >> So, I think that you have to: >> >> 1) set up the toolchain >> 2) compile the dependencies (iconv, glib, etc...) for the arm target >> 3) compile gstreamer for the arm target >> >> As an example, I plan to have gstreamer on Windows CE devices with cegcc. > > Is glib buildable on WinCE therese days? I've not tried yet. I'm stick on another set of libs before i go to gstreamer. Which means iconv then glib. I wonder which settings are best for embedded (configure options) Vincent From bilboed at gmail.com Fri Sep 12 12:36:57 2008 From: bilboed at gmail.com (Edward Hervey) Date: Fri, 12 Sep 2008 12:36:57 +0200 Subject: [gst-devel] How can we compile GStreamer for ARM In-Reply-To: References: <002401c9148d$416e6d30$340318ac@LGE.NET> <3c1737210809112225i4e5f3c34q214a959851b7d58b@mail.gmail.com> <48CA141A.9010800@hora-obscura.de> Message-ID: <1221215817.2566.3.camel@putamadre> On Fri, 2008-09-12 at 12:20 +0200, Vincent Torri wrote: > I wonder which settings are best for embedded (configure options) For testing... I wouldn't disable much else you'll find it hard to debug. Otherwise for production usage: --disable-gst-debug (you'll save some cpu/memory) --disable-loadsave (really not needed) --disable-parse (remove gst_parse_* features) --disable-option-parsing (not needed if you're going to be using just the library). That should slim it down enough. Oh, and CFLAGS="-Os" will optimize the resulting library size. And obviously only compile/install the plugins you need :) Edward > > Vincent > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From learning_gst at hotmail.com Fri Sep 12 13:24:31 2008 From: learning_gst at hotmail.com (learning gst) Date: Fri, 12 Sep 2008 11:24:31 +0000 Subject: [gst-devel] How can we compile GStreamer for ARM In-Reply-To: References: <002401c9148d$416e6d30$340318ac@LGE.NET> <3c1737210809112225i4e5f3c34q214a959851b7d58b@mail.gmail.com> Message-ID: Hello, Thanks for your reply. Now, the problem is that we do not use X server at all in our embedded system. In this case, is it possible for us to compile without any errors? Thanks! > Date: Fri, 12 Sep 2008 08:31:27 +0200 > From: vtorri at univ-evry.fr > To: gstreamer-devel at lists.sourceforge.net > CC: gstreamer-embedded at lists.sourceforge.net > Subject: Re: [gst-devel] How can we compile GStreamer for ARM > > > > On Fri, 12 Sep 2008, learning gst wrote: > > > I hope to compile GStreammer for ARM. But I don't know how to compile? Please help me. > > use a cross-compilation toolchain. I'm sure google can help you (for > example http://www.scratchbox.org/, but there are certainly other > toolchain) > > So, I think that you have to: > > 1) set up the toolchain > 2) compile the dependencies (iconv, glib, etc...) for the arm target > 3) compile gstreamer for the arm target > > As an example, I plan to have gstreamer on Windows CE devices with cegcc. > > regards > > Vincent Torri > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Fri Sep 12 13:34:20 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Fri, 12 Sep 2008 14:34:20 +0300 Subject: [gst-devel] [gst-embedded] How can we compile GStreamer for ARM In-Reply-To: <1221215817.2566.3.camel@putamadre> References: <002401c9148d$416e6d30$340318ac@LGE.NET> <3c1737210809112225i4e5f3c34q214a959851b7d58b@mail.gmail.com> <48CA141A.9010800@hora-obscura.de> <1221215817.2566.3.camel@putamadre> Message-ID: <48CA53BC.8070208@hora-obscura.de> Edward Hervey schrieb: > On Fri, 2008-09-12 at 12:20 +0200, Vincent Torri wrote: > >> I wonder which settings are best for embedded (configure options) > > For testing... I wouldn't disable much else you'll find it hard to > debug. > > Otherwise for production usage: > --disable-gst-debug (you'll save some cpu/memory) > --disable-loadsave (really not needed) > --disable-parse (remove gst_parse_* features) > --disable-option-parsing (not needed if you're going to be using just > the library). > only one to add: --disable-nls (if you don't mind english eror messages) Stefan > That should slim it down enough. Oh, and CFLAGS="-Os" will optimize > the resulting library size. > And obviously only compile/install the plugins you need :) > > > Edward > >> Vincent >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's challenge >> Build the coolest Linux based applications with Moblin SDK & win great prizes >> Grand prize is a trip for two to an Open Source event anywhere in the world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > Gstreamer-embedded mailing list > Gstreamer-embedded at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded From ensonic at hora-obscura.de Fri Sep 12 13:37:12 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Fri, 12 Sep 2008 14:37:12 +0300 Subject: [gst-devel] [gst-embedded] How can we compile GStreamer for ARM In-Reply-To: References: <002401c9148d$416e6d30$340318ac@LGE.NET> <3c1737210809112225i4e5f3c34q214a959851b7d58b@mail.gmail.com> Message-ID: <48CA5468.8050602@hora-obscura.de> learning gst schrieb: > Hello, > > Thanks for your reply. > > Now, the problem is that we do not use X server at all in our embedded > system. In this case, is it possible for us to compile without any errors? > It should be. Please let us know about errors you encounter (best via bugzilla). Most of us who build on linux have X and there might be error that we simply never get. Stefan > Thanks! > > > Date: Fri, 12 Sep 2008 08:31:27 +0200 > > From: vtorri at univ-evry.fr > > To: gstreamer-devel at lists.sourceforge.net > > CC: gstreamer-embedded at lists.sourceforge.net > > Subject: Re: [gst-devel] How can we compile GStreamer for ARM > > > > > > > > On Fri, 12 Sep 2008, learning gst wrote: > > > > > I hope to compile GStreammer for ARM. But I don't know how to > compile? Please help me. > > > > use a cross-compilation toolchain. I'm sure google can help you (for > > example http://www.scratchbox.org/, but there are certainly other > > toolchain) > > > > So, I think that you have to: > > > > 1) set up the toolchain > > 2) compile the dependencies (iconv, glib, etc...) for the arm tar get > > 3) compile gstreamer for the arm target > > > > As an example, I plan to have gstreamer on Windows CE devices with cegcc. > > > > regards > > > > Vincent Torri > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win > great prizes > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------ > Invite your mail contacts to join your friends list with Windows Live > Spaces. It's easy! Try it! > create&wx_url=/friends.aspx&mkt=en-us> > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > ------------------------------------------------------------------------ > > _______________________________________________ > Gstreamer-embedded mailing list > Gstreamer-embedded at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded From vtorri at univ-evry.fr Fri Sep 12 14:00:30 2008 From: vtorri at univ-evry.fr (Vincent Torri) Date: Fri, 12 Sep 2008 14:00:30 +0200 (CEST) Subject: [gst-devel] [gst-embedded] How can we compile GStreamer for ARM In-Reply-To: References: <002401c9148d$416e6d30$340318ac@LGE.NET> <3c1737210809112225i4e5f3c34q214a959851b7d58b@mail.gmail.com> Message-ID: On Fri, 12 Sep 2008, learning gst wrote: > > Hello, > > Thanks for your reply. > > Now, the problem is that we do not use X server at all in our embedded system. In this case, is it possible for us to compile without any errors? as I plan to port gst to Windows CE (and as gst compiles on Windows), you can compile it without having X11 at all. Some sinks requires X11 (x*imagesink, the opengl one, maybe others). Vincent Torri From felipe.contreras at gmail.com Fri Sep 12 14:07:23 2008 From: felipe.contreras at gmail.com (Felipe Contreras) Date: Fri, 12 Sep 2008 15:07:23 +0300 Subject: [gst-devel] [gst-embedded] How can we compile GStreamer for ARM In-Reply-To: References: <002401c9148d$416e6d30$340318ac@LGE.NET> <3c1737210809112225i4e5f3c34q214a959851b7d58b@mail.gmail.com> Message-ID: <94a0d4530809120507i68cfa60cj5e6357fa3172cf36@mail.gmail.com> On Fri, Sep 12, 2008 at 9:15 AM, learning gst wrote: > Hello, > > I hope to compile GStreammer for ARM. But I don't know how to compile? > Please help me. I've added a page in the wiki with my notes for compiling in ARM. http://gstreamer.freedesktop.org/wiki/HowToCompileForEmbedded I use --with-html-dir=/tmp/dump to avoid installing the documentation on the target, and --disable-static to avoid static libraries. Since I'm using this build for development purposes I keep the debugging stuff. Best regards. -- Felipe Contreras From remi.buisson at viotech.net Fri Sep 12 15:21:11 2008 From: remi.buisson at viotech.net (=?ISO-8859-1?Q?R=E9mi_BUISSON?=) Date: Fri, 12 Sep 2008 15:21:11 +0200 Subject: [gst-devel] A big problem !! HELP !!!!! Message-ID: <48CA6CC7.3080707@viotech.net> Hi everyone, I have to generate an XML code containing available plugins of gstreamer but I have a problem in listing gstreamer modules. I attach an archive with a bug and I don't know how to fix it ... maybe a gstreamer bug ? It looks like a pointer kind bug. With valgrind there is a lot of errors. Please compile it with: make depends && make execute : build/client Try to comment "xmlDocPtr doc;" in xml_parser.c" and re-test ... On my side I get 265 modules with the first step and 9 in the second ... Any idea ? R?mi -------------- next part -------------- A non-text attachment was scrubbed... Name: test.tgz Type: application/x-gtar Size: 3204 bytes Desc: not available URL: From ajitjohn at tataelxsi.co.in Fri Sep 12 16:06:32 2008 From: ajitjohn at tataelxsi.co.in (ajitjohn) Date: Fri, 12 Sep 2008 19:36:32 +0530 Subject: [gst-devel] how to stream .asf using gstreamer Message-ID: <003501c914e0$c3767540$68033c0a@telxsi.com> Hii all, I have tried streaming .asf file using gstreamer from 1 pc to a vlc player on other pc .The pipeline that i have used for this purpose is given below:- gst-launch filesrc location=/root/Desktop/audioVideo/mjpegi.asf ! udpsink host=10.60.3.104 port=4951 in the gstreamer side i am getting the following messages Setting pipeline to PAUSED ... Pipeline is PREROLLING ... Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock Got EOS from element "pipeline0". Execution ended after 414876240 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... FREEING pipeline ... and on the vlc player i am getting the following messages main error: recv() failed. Increase the mtu size (--mtu option) main error: recv() failed. Increase the mtu size (--mtu option) asf warning: unsupported packet header asf warning: p_peek[0]&0x80 != 0x80 asf warning: undeclared stream[Id 0x0] asf warning: p_peek[0]&0x80 != 0x80 asf warning: undeclared stream[Id 0x7f] asf warning: unsupported packet header asf warning: p_peek[0]&0x80 != 0x80 asf warning: undeclared stream[Id 0x67] asf warning: undeclared stream[Id 0x4c] asf warning: undeclared stream[Id 0x3c] asf warning: undeclared stream[Id 0x52] asf warning: undeclared stream[Id 0x71] asf warning: undeclared stream[Id 0x71] asf warning: undeclared stream[Id 0x30] asf warning: undeclared stream[Id 0x5d] asf warning: undeclared stream[Id 0x29] asf warning: undeclared stream[Id 0x38] asf warning: undeclared stream[Id 0x29] asf warning: undeclared stream[Id 0x6f] asf warning: undeclared stream[Id 0x9] asf warning: undeclared stream[Id 0x52] asf warning: undeclared stream[Id 0x30] asf warning: undeclared stream[Id 0x77] asf warning: undeclared stream[Id 0x3] asf warning: undeclared stream[Id 0x47] asf warning: undeclared stream[Id 0x4a] asf warning: unsupported packet header asf warning: unsupported packet header asf warning: p_peek[0]&0x80 != 0x80 asf warning: p_peek[0]&0x80 != 0x80 asf warning: p_peek[0]&0x80 != 0x80 ............................. I am not able to stream and receive the file.Can anyone tell me what changes i should make in the pipeline to stream the asf file through gstreamer and succesfully receive through vlc player. Regards, ~Ajit. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From andres.colubri at gmail.com Fri Sep 12 20:45:04 2008 From: andres.colubri at gmail.com (Andres Colubri) Date: Fri, 12 Sep 2008 11:45:04 -0700 Subject: [gst-devel] Freshly baked Windows binaries of HEAD available In-Reply-To: References: Message-ID: <48CAB8B0.8090801@gmail.com> Using the great work of Ole Andre as the starting point, I created a new set of simplified, up-to-date GStreamer installers for Windows (one for the binaries and libs, and another for development files). You can grab them from here: http://sourceforge.net/project/showfiles.php?group_id=225389&package_id=272648&release_id=625609 Please note that this is an initial, "preview" release so people could try it out and report their success (or failure). I have myself used it successfully on both Windows XP professional SP3 and Vista SP1 business edition. It includes ffmpeg libraries compiled from svn source grabbed a few days ago. For those interested in the steps I followed to create these installers, I have put up this post on my blog: http://codeanticode.wordpress.com/2008/09/12/the-gstreamer-adventure-part-i-creating-a-windows-installer/ Andres Ole Andre Vadla Ravn?s wrote: > Hi all, > > I just whipped together some binaries of GStreamer HEAD built against msvcrt.dll (the ReleaseWdkCrt configuration). > > The snapshots can be found here: > http://people.collabora.co.uk/~oleavr/OABuild/snapshots/ > > 33e44c77f4b8f413db6ffc5e47f2a8ab OABuild-20080904-glib-dbg.exe > 0f717766f27fab1b4905a28123a195cc OABuild-20080904-glib-dev.exe > 708f0ca2d3b5934c2d1ad91ea3713e2e OABuild-20080904-glib.exe > ffaff7b80693d76f1476c6ab809bbe75 OABuild-20080904-gstreamer-dbg.exe > c02d961fd4f80900fa338a3049731b8b OABuild-20080904-gstreamer-dev.exe > 8e29f516806bddc46a5435898e3cf204 OABuild-20080904-gstreamer.exe > > Please note that this was done in a hurry, so the naming scheme, which files go where, etc., will likely change in the future. Extract OABuild-20080904-glib.exe and OABuild-20080904-gstreamer.exe to get a pure runtime installation (with gst-inspect and gst-launch, likely to be split out in the future), dbg for debug symbols (.pdb files), and dev for development headers, .lib files and stuff like that. > > Also note that I just realized that the binaries from gst-plugins-good are off by one micro version as I forgot to sync GstPluginsGoodVersion.vsprops against configure.ac... This is on my TODO list of things to automate, and will obviously be fixed in future snapshots. > > Lastly, if someone feels like contributing automated .msi/.msm packaging for future snapshots then that would be awesome! :) > > Cheers, > Ole Andr? > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From ved.kpl at gmail.com Fri Sep 12 17:29:01 2008 From: ved.kpl at gmail.com (ved kpl) Date: Fri, 12 Sep 2008 20:59:01 +0530 Subject: [gst-devel] how to stream .asf using gstreamer In-Reply-To: <003501c914e0$c3767540$68033c0a@telxsi.com> References: <003501c914e0$c3767540$68033c0a@telxsi.com> Message-ID: <7496c23f0809120829h25a84440y5c9bbaaa19114d70@mail.gmail.com> Hi, Since the buffers are coming from the file, they are not timestamped and will quickly race to EOS . You probably need asf parse/reader in between that can parse and timestamp the asf data, before sending to the udpsink. Ved On Fri, Sep 12, 2008 at 7:36 PM, ajitjohn wrote: > Hii all, > > I have tried streaming .asf file using gstreamer from 1 pc to a vlc player > on other pc .The pipeline that i have used for this purpose is given below:- > > > gst-launch filesrc location=/root/Desktop/audioVideo/mjpegi.asf ! udpsink > host=10.60.3.104 port=4951 > > in the gstreamer side i am getting the following messages > > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > Pipeline is PREROLLED ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > Got EOS from element "pipeline0". > Execution ended after 414876240 ns. > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > Setting pipeline to NULL ... > FREEING pipeline ... > > and on the vlc player i am getting the following messages > > main error: recv() failed. Increase the mtu size (--mtu option) > main error: recv() failed. Increase the mtu size (--mtu option) > asf warning: unsupported packet header > asf warning: p_peek[0]&0x80 != 0x80 > asf warning: undeclared stream[Id 0x0] > asf warning: p_peek[0]&0x80 != 0x80 > asf warning: undeclared stream[Id 0x7f] > asf warning: unsupported packet header > asf warning: p_peek[0]&0x80 != 0x80 > asf warning: undeclared stream[Id 0x67] > asf warning: undeclared stream[Id 0x4c] > asf warning: undeclared stream[Id 0x3c] > asf warning: undeclared stream[Id 0x52] > asf warning: undeclared stream[Id 0x71] > asf warning: undeclared stream[Id 0x71] > asf warning: undeclared stream[Id 0x30] > asf warning: undeclared stream[Id 0x5d] > asf warning: undeclared stream[Id 0x29] > asf warning: undeclared stream[Id 0x38] > asf warning: undeclared stream[Id 0x29] > asf warning: undeclared stream[Id 0x6f] > asf warning: undeclared stream[Id 0x9] > asf warning: undeclared stream[Id 0x52] > asf warning: undeclared stream[Id 0x30] > asf warning: undeclared stream[Id 0x77] > asf warning: undeclared stream[Id 0x3] > asf warning: undeclared stream[Id 0x47] > asf warning: undeclared stream[Id 0x4a] > asf warning: unsupported packet header > asf warning: unsupported packet header > asf warning: p_peek[0]&0x80 != 0x80 > asf warning: p_peek[0]&0x80 != 0x80 > asf warning: p_peek[0]&0x80 != 0x80 ............................. > > I am not able to stream and receive the file.Can anyone tell me what > changes i should make in the pipeline to stream the asf file through > gstreamer and succesfully receive through vlc player. > > Regards, > ~Ajit. > > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ajitjohn at tataelxsi.co.in Fri Sep 12 17:52:28 2008 From: ajitjohn at tataelxsi.co.in (ajitjohn) Date: Fri, 12 Sep 2008 21:22:28 +0530 Subject: [gst-devel] how to stream .asf using gstreamer In-Reply-To: <7496c23f0809120829h25a84440y5c9bbaaa19114d70@mail.gmail.com> Message-ID: <003701c914ef$8fa738d0$68033c0a@telxsi.com> Hi , Thanks a lot for your reply. My actual requirement is to play the file on windows media player. I was testing on VLC player. If I use a asf parser the asf header will go.Then how can i play the streamed data on WMP. And also can u plzz inform me can we use rtpsend instead udpsink for play back on WMP Thanks in advance, Regards, Ajit.... -----Original Message----- From: ved kpl [mailto:ved.kpl at gmail.com] Sent: Friday, September 12, 2008 8:59 PM To: ajitjohn at tataelxsi.co.in; Discussion of the development of GStreamer Subject: Re: [gst-devel] how to stream .asf using gstreamer Hi, Since the buffers are coming from the file, they are not timestamped and will quickly race to EOS . You probably need asf parse/reader in between that can parse and timestamp the asf data, before sending to the udpsink. Ved On Fri, Sep 12, 2008 at 7:36 PM, ajitjohn wrote: > Hii all, > > I have tried streaming .asf file using gstreamer from 1 pc to a vlc player > on other pc .The pipeline that i have used for this purpose is given below:- > > > gst-launch filesrc location=/root/Desktop/audioVideo/mjpegi.asf ! udpsink > host=10.60.3.104 port=4951 > > in the gstreamer side i am getting the following messages > > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > Pipeline is PREROLLED ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > Got EOS from element "pipeline0". > Execution ended after 414876240 ns. > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > Setting pipeline to NULL ... > FREEING pipeline ... > > and on the vlc player i am getting the following messages > > main error: recv() failed. Increase the mtu size (--mtu option) > main error: recv() failed. Increase the mtu size (--mtu option) > asf warning: unsupported packet header > asf warning: p_peek[0]&0x80 != 0x80 > asf warning: undeclared stream[Id 0x0] > asf warning: p_peek[0]&0x80 != 0x80 > asf warning: undeclared stream[Id 0x7f] > asf warning: unsupported packet header > asf warning: p_peek[0]&0x80 != 0x80 > asf warning: undeclared stream[Id 0x67] > asf warning: undeclared stream[Id 0x4c] > asf warning: undeclared stream[Id 0x3c] > asf warning: undeclared stream[Id 0x52] > asf warning: undeclared stream[Id 0x71] > asf warning: undeclared stream[Id 0x71] > asf warning: undeclared stream[Id 0x30] > asf warning: undeclared stream[Id 0x5d] > asf warning: undeclared stream[Id 0x29] > asf warning: undeclared stream[Id 0x38] > asf warning: undeclared stream[Id 0x29] > asf warning: undeclared stream[Id 0x6f] > asf warning: undeclared stream[Id 0x9] > asf warning: undeclared stream[Id 0x52] > asf warning: undeclared stream[Id 0x30] > asf warning: undeclared stream[Id 0x77] > asf warning: undeclared stream[Id 0x3] > asf warning: undeclared stream[Id 0x47] > asf warning: undeclared stream[Id 0x4a] > asf warning: unsupported packet header > asf warning: unsupported packet header > asf warning: p_peek[0]&0x80 != 0x80 > asf warning: p_peek[0]&0x80 != 0x80 > asf warning: p_peek[0]&0x80 != 0x80 ............................. > > I am not able to stream and receive the file.Can anyone tell me what > changes i should make in the pipeline to stream the asf file through > gstreamer and succesfully receive through vlc player. > > Regards, > ~Ajit. > > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From ajitjohn at tataelxsi.co.in Fri Sep 12 17:57:48 2008 From: ajitjohn at tataelxsi.co.in (ajitjohn) Date: Fri, 12 Sep 2008 21:27:48 +0530 Subject: [gst-devel] how to stream .asf using gstreamer In-Reply-To: <7496c23f0809120829h25a84440y5c9bbaaa19114d70@mail.gmail.com> Message-ID: <003801c914f0$4e2ae3b0$68033c0a@telxsi.com> hii , Is there a plugin in gstreamer which wil serve the purpose? Regards, Ajit. -----Original Message----- From: ved kpl [mailto:ved.kpl at gmail.com] Sent: Friday, September 12, 2008 8:59 PM To: ajitjohn at tataelxsi.co.in; Discussion of the development of GStreamer Subject: Re: [gst-devel] how to stream .asf using gstreamer Hi, Since the buffers are coming from the file, they are not timestamped and will quickly race to EOS . You probably need asf parse/reader in between that can parse and timestamp the asf data, before sending to the udpsink. Ved On Fri, Sep 12, 2008 at 7:36 PM, ajitjohn wrote: > Hii all, > > I have tried streaming .asf file using gstreamer from 1 pc to a vlc player > on other pc .The pipeline that i have used for this purpose is given below:- > > > gst-launch filesrc location=/root/Desktop/audioVideo/mjpegi.asf ! udpsink > host=10.60.3.104 port=4951 > > in the gstreamer side i am getting the following messages > > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > Pipeline is PREROLLED ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > Got EOS from element "pipeline0". > Execution ended after 414876240 ns. > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > Setting pipeline to NULL ... > FREEING pipeline ... > > and on the vlc player i am getting the following messages > > main error: recv() failed. Increase the mtu size (--mtu option) > main error: recv() failed. Increase the mtu size (--mtu option) > asf warning: unsupported packet header > asf warning: p_peek[0]&0x80 != 0x80 > asf warning: undeclared stream[Id 0x0] > asf warning: p_peek[0]&0x80 != 0x80 > asf warning: undeclared stream[Id 0x7f] > asf warning: unsupported packet header > asf warning: p_peek[0]&0x80 != 0x80 > asf warning: undeclared stream[Id 0x67] > asf warning: undeclared stream[Id 0x4c] > asf warning: undeclared stream[Id 0x3c] > asf warning: undeclared stream[Id 0x52] > asf warning: undeclared stream[Id 0x71] > asf warning: undeclared stream[Id 0x71] > asf warning: undeclared stream[Id 0x30] > asf warning: undeclared stream[Id 0x5d] > asf warning: undeclared stream[Id 0x29] > asf warning: undeclared stream[Id 0x38] > asf warning: undeclared stream[Id 0x29] > asf warning: undeclared stream[Id 0x6f] > asf warning: undeclared stream[Id 0x9] > asf warning: undeclared stream[Id 0x52] > asf warning: undeclared stream[Id 0x30] > asf warning: undeclared stream[Id 0x77] > asf warning: undeclared stream[Id 0x3] > asf warning: undeclared stream[Id 0x47] > asf warning: undeclared stream[Id 0x4a] > asf warning: unsupported packet header > asf warning: unsupported packet header > asf warning: p_peek[0]&0x80 != 0x80 > asf warning: p_peek[0]&0x80 != 0x80 > asf warning: p_peek[0]&0x80 != 0x80 ............................. > > I am not able to stream and receive the file.Can anyone tell me what > changes i should make in the pipeline to stream the asf file through > gstreamer and succesfully receive through vlc player. > > Regards, > ~Ajit. > > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From ryankelln at gmail.com Fri Sep 12 20:43:00 2008 From: ryankelln at gmail.com (Ryan Kelln) Date: Fri, 12 Sep 2008 14:43:00 -0400 Subject: [gst-devel] Possible gstreamer errors: volume, dispose, loop detected - oh my! Message-ID: <48CAB834.5040302@gmail.com> Hi Gstreamer devs, (sorry to mods, I tried posting this without subscribing first, you can delete/disapprove the post waiting for approval) I'm using gstreamer through gstreamer-java inside of processing. I'm getting some odd errors and warnings and eventually a crash (perhaps a null pointer). I'm hoping you might have a bit of insight or guidance for how to avoid these problems or what I could do to fix them. Thanks. Environment: Ubuntu 8.04 gstreamer-team ppa (https://edge.launchpad.net/~gstreamer-team/+archive) Sun Java 1.06_07 Processing 148 gstreamer-java 0.8 The errors happen in a variety of orders, here is an example of each: ** (GSVideo:9501): CRITICAL **: volume_transform_ip: assertion `this->process != NULL' failed (GSVideo:9501): GStreamer-CRITICAL **: Trying to dispose element test, but it is not in the NULL state. You need to explicitly set elements to the NULL state before dropping the final reference, to allow them to clean up. (GSVideo:9501): GStreamer-WARNING **: loop detected in the graph of bin GStreamer Audio Data Extractor: mysoundfile.wav!! Then eventually Java will crash with a SIGSEGV fault in a native thread which I'm assuming is gstreamer. Looks like it is accessing a null pointer. Description of what I'm doing: Custom software that is playing back multiple audio files at once. It loads approx. 11 sound files (all wavs or aif errors and crash remain the same) and starts and stops the files quite often. It also checks the current volume and changes the volume each update (i.e. constantly adjusting the volume). In addition I'm using the audiopanorama plug-in to do constant panning. Occasionally I will get the "loop detected" warning on one or two of the sound files, but which sound file it is seems to change and it doesn't happen every run. The error appears to occur when the sound is first played, not when the object is created. The program continues to function after receiving this warning although the sound file doesn't play. I get the "volume_transform_ip" error quite often (every 5 minutes?) but the program doesn't immediately crash. I get the "dispose element test" error only on a multi-core processor (AMD Phenom) but not on an equivalent machine with just a single core. Again, this doesn't cause an immediate crash. As far as I know I'm not make any elements named "test", and I'm not disposing of anything until the program ends but I am starting and stopping the audio streams. Things I've tried: I've grepped the source for these and found the loop detected error in gstbin.c (gst_bin_sort_iterator_next() )and the "dispose element not NULL" in gstelement.c (gst_element_dispose() ) but I can't locate the "volume_transform_ip" error. I've tired running the program with sounds disabled and it runs without errors or crashes (for hours). If I only update the volumes of the sounds every other update (or every 5, etc) it seems to help, although its hard to tell if the errors occur that much slower or even more infrequently. If I don't adjust the volume at all it seems to run fine (survived 10+ mins without errors, still running tests). Some possible problem areas: When is it safe to set and check the volume (especially in regards to starting and stopping sounds)? Is it safe to set and check the volume in the same update... multiple times? I'm not sure if the errors are in gstreamer or just my use of it? I've got to deliver this project really soon so workarounds might be the best option if the error is gstreamers. If you're sure that gstreamer is at fault I'd love some tips on how to provide more info (I'm not sure how I'd debug gstreamer through all these other layers) and write up a bug report. Thanks very much, Ryan From jjcogliati-gstreamer at yahoo.com Fri Sep 12 22:09:35 2008 From: jjcogliati-gstreamer at yahoo.com (jjcogliati-gstreamer at yahoo.com) Date: Fri, 12 Sep 2008 13:09:35 -0700 (PDT) Subject: [gst-devel] Can a MPEG-1 with Audio Layers 1&2 plugin be in plugins-good (patentwise)? In-Reply-To: <48B674B7.4050205@joost.com> Message-ID: <735516.27489.qm@web62407.mail.re1.yahoo.com> --- On Thu, 8/28/08, Gabriel Bouvigne wrote: > From: Gabriel Bouvigne > Subject: Re: [gst-devel] Can a MPEG-1 with Audio Layers 1&2 plugin be in plugins-good (patentwise)? > To: jjcogliati-gstreamer at yahoo.com, "Discussion of the development of GStreamer" > Date: Thursday, August 28, 2008, 3:49 AM > jjcogliati-gstreamer at yahoo.com a ?crit : > > Is there anywhere that has a list of these patents? > > You posted such a list yourself: > http://scratchpad.wikia.com/wiki/MPEG_patent_status Yep, I put up a list of MPEG-2 patents and MPEG-1 Audio Layer 3 patents. But I didn't put up a list of MPEG-1 video patents or MPEG-1 Audio Layers 1 and 2 patents because I have not found anyone saying that they have them. So basically, I haven't found a list of MPEG-1 patents (other than MP3 patents), and no one on this list has managed to name any. > > I looked through the US MPEG-2 patents that Philips > has > > ( http://en.wikipedia.org/wiki/Talk:MPEG-2#Patents > based on http://www.mpegla.com/m2/m2-patentlist.cfm ) and > none sounded like audio patents. > > Probably because MPEG-LA is not a licensing authority for > MPEG1/2 Layer > I/II/III patents. > > -- > Gabriel From info at klauszeitler.de Fri Sep 12 22:38:25 2008 From: info at klauszeitler.de (Klaus Zeitler) Date: Fri, 12 Sep 2008 22:38:25 +0200 Subject: [gst-devel] GStreamer core error in banshee: StateChange Message-ID: <87hc8lgkvy.fsf@lysmata.klauszeitler.de> Hello, all of a sudden I can't play music anymore. Instead banshee spits out hundreds of the following 2 error messages: GStreamer core error: StateChange GStreamer resource error: NotFound Not very informative I'd say. Maybe somebody here has got an idea what happened or can point me towards the right place to ask (if this isn't the right one :-). My problem started after I had installed sound-juicer (somebody suggested to use sound-juicer to rip the CDs, since I just can't get banshee to fetch metadata). I installed sound-juicer and ripped 2 CDs (without any problems) and also played a CD. But next time I called banshee I got those odd GStreamer errors and now I also get them when I try to play music with sound-juicer. But then again that might be pure coincidence. Nevertheless I de-installed sound-juicer, but the errors didn't disappear for banshee. I'm at a total loss. Maybe I should try to reinstall gstreamer, but there are so many packages, that I have no idea what I should select. e.g. I see many gstreamer-0_10... packages as well as gstreamer010... ones with yast2. Can/should I install new gstreamer packages and if so which ones? OS: openSuse 10.3 BTW I also submitted a bug report for gstreamer. I appreciate any help. Thanks Klaus -- ------------------------------------------------------- | Klaus Zeitler | | Raiffeisenstr. 31 a 90427 N?rnberg Germany | | Telefon: 49 911 316261 Fax: 316281 | | Email: info at klauszeitler.de | ------------------------------------------------------- --- Illegal aliens have always been a problem in the United States. Ask any Indian. -- Robert Orben From airmind at gmail.com Sat Sep 13 00:22:11 2008 From: airmind at gmail.com (Alexandre) Date: Fri, 12 Sep 2008 19:22:11 -0300 Subject: [gst-devel] Link xvid with ffmpeg's mp4 muxer In-Reply-To: <48CA1320.6090602@hora-obscura.de> References: <48f4838d0809101334q2771ee18kfb6df1c349373f8b@mail.gmail.com> <48CA1320.6090602@hora-obscura.de> Message-ID: <48f4838d0809121522o5c755499gbd0074c1195f15d3@mail.gmail.com> Hi, I've been following the Qtmux development very closely (part of Google Summer Code, which I also participated). It seems to support H.264 now, which is very good. I'll take a look if it supports xvid. However, it will take some time to reach the trunk, then a release, and only then into packages and distros. I wanted something working now, with the ffmux_mp4. Could I force a cap in the ffmux? Will that have a problem with the data going through? I mean, the mux must support xvid, it's just Mpeg4 video. Thanks, Alexandre On Fri, Sep 12, 2008 at 3:58 AM, Stefan Kost wrote: > Hi, > > Alexandre schrieb: > > Hi, > > > > I'm trying to mux the GStreamer xvidenc element with ffmux_mp4, to > > create a Mpeg 4 file with XVid encoded video, but it's not being > > succesfull. The xvidenc caps are video/x-xvid, and ffmux supports either > > mpeg v4 video or video/x-divx. > Hopefully qtmux is merged to -plugins-bad soon. Lets make sure it support > it then. > > Stefan > > > > > Is there a way to link them, or are they incompatible? > > > > -- > > Alexandre Rosenfeld > > > > EngComp 06 - USP S?o Carlos > > > > > > ------------------------------------------------------------------------ > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Alexandre Rosenfeld EngComp 06 - USP S?o Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: From thiagossantos at gmail.com Sat Sep 13 00:42:04 2008 From: thiagossantos at gmail.com (thiagoss) Date: Fri, 12 Sep 2008 20:42:04 -0200 Subject: [gst-devel] Link xvid with ffmpeg's mp4 muxer In-Reply-To: <48f4838d0809121522o5c755499gbd0074c1195f15d3@mail.gmail.com> References: <48f4838d0809101334q2771ee18kfb6df1c349373f8b@mail.gmail.com> <48CA1320.6090602@hora-obscura.de> <48f4838d0809121522o5c755499gbd0074c1195f15d3@mail.gmail.com> Message-ID: Do you have any valid mp4 files with xvid? It would help me adding support for it in qtmux. On Fri, Sep 12, 2008 at 8:22 PM, Alexandre wrote: > Hi, > > I've been following the Qtmux development very closely (part of Google > Summer Code, which I also participated). It seems to support H.264 now, > which is very good. I'll take a look if it supports xvid. > However, it will take some time to reach the trunk, then a release, and > only then into packages and distros. I wanted something working now, with > the ffmux_mp4. > > Could I force a cap in the ffmux? Will that have a problem with the data > going through? I mean, the mux must support xvid, it's just Mpeg4 video. > > Thanks, > Alexandre > > > On Fri, Sep 12, 2008 at 3:58 AM, Stefan Kost wrote: > >> Hi, >> >> Alexandre schrieb: >> > Hi, >> > >> > I'm trying to mux the GStreamer xvidenc element with ffmux_mp4, to >> > create a Mpeg 4 file with XVid encoded video, but it's not being >> > succesfull. The xvidenc caps are video/x-xvid, and ffmux supports either >> > mpeg v4 video or video/x-divx. >> Hopefully qtmux is merged to -plugins-bad soon. Lets make sure it support >> it then. >> >> Stefan >> >> > >> > Is there a way to link them, or are they incompatible? >> > >> > -- >> > Alexandre Rosenfeld >> > >> > EngComp 06 - USP S?o Carlos >> > >> > >> > ------------------------------------------------------------------------ >> > >> > >> ------------------------------------------------------------------------- >> > This SF.Net email is sponsored by the Moblin Your Move Developer's >> challenge >> > Build the coolest Linux based applications with Moblin SDK & win great >> prizes >> > Grand prize is a trip for two to an Open Source event anywhere in the >> world >> > http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> > >> > >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > gstreamer-devel mailing list >> > gstreamer-devel at lists.sourceforge.net >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> challenge >> Build the coolest Linux based applications with Moblin SDK & win great >> prizes >> Grand prize is a trip for two to an Open Source event anywhere in the >> world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > -- > Alexandre Rosenfeld > > EngComp 06 - USP S?o Carlos > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) -------------- next part -------------- An HTML attachment was scrubbed... URL: From chalserogers at gmail.com Sat Sep 13 09:32:23 2008 From: chalserogers at gmail.com (Christopher James Halse Rogers) Date: Sat, 13 Sep 2008 17:32:23 +1000 Subject: [gst-devel] Link xvid with ffmpeg's mp4 muxer In-Reply-To: <48f4838d0809121522o5c755499gbd0074c1195f15d3@mail.gmail.com> References: <48f4838d0809101334q2771ee18kfb6df1c349373f8b@mail.gmail.com> <48CA1320.6090602@hora-obscura.de> <48f4838d0809121522o5c755499gbd0074c1195f15d3@mail.gmail.com> Message-ID: <1221291143.7676.6.camel@CowboyLaputopu.cooperteam.net.cooperteam.net> On Fri, 2008-09-12 at 19:22 -0300, Alexandre wrote: > Hi, > > I've been following the Qtmux development very closely (part of Google > Summer Code, which I also participated). It seems to support H.264 > now, which is very good. I'll take a look if it supports xvid. > However, it will take some time to reach the trunk, then a release, > and only then into packages and distros. I wanted something working > now, with the ffmux_mp4. > > Could I force a cap in the ffmux? Will that have a problem with the > data going through? I mean, the mux must support xvid, it's just Mpeg4 > video. Is this a bug in xvidenc? Is there any situation in which you need to know that xvid (or divx, barring a divx v3 stream) was the encoder? Why do the video/x-divx {version 4,5} and video/x-xvid caps exist at all - why shouldn't they be video/mpeg version=4? -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: This is a digitally signed message part URL: From ved.kpl at gmail.com Sat Sep 13 16:24:38 2008 From: ved.kpl at gmail.com (ved kpl) Date: Sat, 13 Sep 2008 19:54:38 +0530 Subject: [gst-devel] how to stream .asf using gstreamer In-Reply-To: <003801c914f0$4e2ae3b0$68033c0a@telxsi.com> References: <7496c23f0809120829h25a84440y5c9bbaaa19114d70@mail.gmail.com> <003801c914f0$4e2ae3b0$68033c0a@telxsi.com> Message-ID: <7496c23f0809130724p792d2b0aka83988191d600429@mail.gmail.com> Hi, I am not sure if theres a plugin that can stream the asf container format. For rtp u will probably need elementary streams, although mpeg2 TS can be delivered overRTP. On Fri, Sep 12, 2008 at 9:27 PM, ajitjohn wrote: > hii , > > Is there a plugin in gstreamer which wil serve the purpose? > > Regards, > Ajit. > > -----Original Message----- > From: ved kpl [mailto:ved.kpl at gmail.com] > Sent: Friday, September 12, 2008 8:59 PM > To: ajitjohn at tataelxsi.co.in; Discussion of the development of GStreamer > Subject: Re: [gst-devel] how to stream .asf using gstreamer > > > Hi, > > Since the buffers are coming from the file, they are not timestamped > and will quickly race to EOS . You probably need asf parse/reader in > between that can parse and timestamp the asf data, before sending to > the udpsink. > > Ved > > On Fri, Sep 12, 2008 at 7:36 PM, ajitjohn wrote: >> Hii all, >> >> I have tried streaming .asf file using gstreamer from 1 pc to a > vlc player >> on other pc .The pipeline that i have used for this purpose is given > below:- >> >> >> gst-launch filesrc location=/root/Desktop/audioVideo/mjpegi.asf ! > udpsink >> host=10.60.3.104 port=4951 >> >> in the gstreamer side i am getting the following messages >> >> Setting pipeline to PAUSED ... >> Pipeline is PREROLLING ... >> Pipeline is PREROLLED ... >> Setting pipeline to PLAYING ... >> New clock: GstSystemClock >> Got EOS from element "pipeline0". >> Execution ended after 414876240 ns. >> Setting pipeline to PAUSED ... >> Setting pipeline to READY ... >> Setting pipeline to NULL ... >> FREEING pipeline ... >> >> and on the vlc player i am getting the following messages >> >> main error: recv() failed. Increase the mtu size (--mtu option) >> main error: recv() failed. Increase the mtu size (--mtu option) >> asf warning: unsupported packet header >> asf warning: p_peek[0]&0x80 != 0x80 >> asf warning: undeclared stream[Id 0x0] >> asf warning: p_peek[0]&0x80 != 0x80 >> asf warning: undeclared stream[Id 0x7f] >> asf warning: unsupported packet header >> asf warning: p_peek[0]&0x80 != 0x80 >> asf warning: undeclared stream[Id 0x67] >> asf warning: undeclared stream[Id 0x4c] >> asf warning: undeclared stream[Id 0x3c] >> asf warning: undeclared stream[Id 0x52] >> asf warning: undeclared stream[Id 0x71] >> asf warning: undeclared stream[Id 0x71] >> asf warning: undeclared stream[Id 0x30] >> asf warning: undeclared stream[Id 0x5d] >> asf warning: undeclared stream[Id 0x29] >> asf warning: undeclared stream[Id 0x38] >> asf warning: undeclared stream[Id 0x29] >> asf warning: undeclared stream[Id 0x6f] >> asf warning: undeclared stream[Id 0x9] >> asf warning: undeclared stream[Id 0x52] >> asf warning: undeclared stream[Id 0x30] >> asf warning: undeclared stream[Id 0x77] >> asf warning: undeclared stream[Id 0x3] >> asf warning: undeclared stream[Id 0x47] >> asf warning: undeclared stream[Id 0x4a] >> asf warning: unsupported packet header >> asf warning: unsupported packet header >> asf warning: p_peek[0]&0x80 != 0x80 >> asf warning: p_peek[0]&0x80 != 0x80 >> asf warning: p_peek[0]&0x80 != 0x80 ............................. >> >> I am not able to stream and receive the file.Can anyone tell me > what >> changes i should make in the pipeline to stream the asf file through >> gstreamer and succesfully receive through vlc player. >> >> Regards, >> ~Ajit. >> >> >> >> The information contained in this electronic message and any attachments > to this message are intended for the exclusive use of the addressee(s) and > may contain proprietary, confidential or privileged information. If you are > not the intended recipient, you should not disseminate, distribute or copy > this e-mail. Please notify the sender immediately and destroy all copies of > this message and any attachments contained in it. >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge >> Build the coolest Linux based applications with Moblin SDK & win great > prizes >> Grand prize is a trip for two to an Open Source event anywhere in the > world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. > From ylatuya at gmail.com Sat Sep 13 17:23:54 2008 From: ylatuya at gmail.com (Andoni Morales) Date: Sat, 13 Sep 2008 17:23:54 +0200 Subject: [gst-devel] Freshly baked Windows binaries of HEAD available In-Reply-To: <48CAB8B0.8090801@gmail.com> References: <48CAB8B0.8090801@gmail.com> Message-ID: <772db3280809130823y74bff3b9u4f45db8d0803c479@mail.gmail.com> Hi One week ago I tested the OABuild binaries and I found some backward incompabilities with GLib. I'm actually working on a GTK app using gstreamer on windows, and the glib version provided by the gtk-sharp installer isn't compatible with the one used to compile gstreamer with the OABuild tools (I don't really know why, but there is an error with gsignal). So what I did, is to compile GStreamer using the OABuild tools against the older glib version and all went well. I tried to replace my installed glib version with the one used to link the OABuild binaries, but the GTK part failed. And with the glib version provided by GTK, the gstreamer part failed (always with a gsignal error when sending a signal). What I mean is: If you want to provide some GStreamer binaries, try to link it against the glib version actually provided by the windows gtk installers as many of us will use GStreamer in combination with GTK. If you don't do this, your gstreamer binaries will work well on a standalone app, but they won't work at all with any previous installation. I dont know if I was clear enough because my english is very poor. Want I wanted to let clear is that there could be some conflicts with GLib that you should take in consideration. I think the GLib version used to link the GStreamer binaries si too cooler for windows and won't be compatible with any other previous installation as the one provided by the windows gtk installer. Regards, Andoni Morales 2008/9/12 Andres Colubri > Using the great work of Ole Andre as the starting point, I created a new > set of simplified, up-to-date GStreamer installers for Windows (one for > the binaries and libs, and another for development files). You can grab > them from here: > > > http://sourceforge.net/project/showfiles.php?group_id=225389&package_id=272648&release_id=625609 > > Please note that this is an initial, "preview" release so people could > try it out and report their success (or failure). I have myself used it > successfully on both Windows XP professional SP3 and Vista SP1 business > edition. > > It includes ffmpeg libraries compiled from svn source grabbed a few days > ago. > > For those interested in the steps I followed to create these installers, > I have put up this post on my blog: > > http://codeanticode.wordpress.com/2008/09/12/the-gstreamer-adventure-part-i-creating-a-windows-installer/ > > Andres > > Ole Andre Vadla Ravn?s wrote: > > Hi all, > > > > I just whipped together some binaries of GStreamer HEAD built against > msvcrt.dll (the ReleaseWdkCrt configuration). > > > > The snapshots can be found here: > > http://people.collabora.co.uk/~oleavr/OABuild/snapshots/ > > > > 33e44c77f4b8f413db6ffc5e47f2a8ab OABuild-20080904-glib-dbg.exe > > 0f717766f27fab1b4905a28123a195cc OABuild-20080904-glib-dev.exe > > 708f0ca2d3b5934c2d1ad91ea3713e2e OABuild-20080904-glib.exe > > ffaff7b80693d76f1476c6ab809bbe75 OABuild-20080904-gstreamer-dbg.exe > > c02d961fd4f80900fa338a3049731b8b OABuild-20080904-gstreamer-dev.exe > > 8e29f516806bddc46a5435898e3cf204 OABuild-20080904-gstreamer.exe > > > > Please note that this was done in a hurry, so the naming scheme, which > files go where, etc., will likely change in the future. Extract > OABuild-20080904-glib.exe and OABuild-20080904-gstreamer.exe to get a pure > runtime installation (with gst-inspect and gst-launch, likely to be split > out in the future), dbg for debug symbols (.pdb files), and dev for > development headers, .lib files and stuff like that. > > > > Also note that I just realized that the binaries from gst-plugins-good > are off by one micro version as I forgot to sync > GstPluginsGoodVersion.vsprops against configure.ac... This is on my TODO > list of things to automate, and will obviously be fixed in future snapshots. > > > > Lastly, if someone feels like contributing automated .msi/.msm packaging > for future snapshots then that would be awesome! :) > > > > Cheers, > > Ole Andr? > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From andres.colubri at gmail.com Sat Sep 13 22:24:56 2008 From: andres.colubri at gmail.com (Andres Colubri) Date: Sat, 13 Sep 2008 13:24:56 -0700 Subject: [gst-devel] Freshly baked Windows binaries of HEAD available In-Reply-To: <772db3280809130823y74bff3b9u4f45db8d0803c479@mail.gmail.com> References: <48CAB8B0.8090801@gmail.com> <772db3280809130823y74bff3b9u4f45db8d0803c479@mail.gmail.com> Message-ID: <48CC2198.6070000@gmail.com> > What I mean is: If you want to provide some GStreamer binaries, try to > link it against the glib version actually provided by the windows gtk > installers as many of us will use GStreamer in combination with GTK. > If you don't do this, your gstreamer binaries will work well on a > standalone app, but they won't work at all with any previous installation. > I dont know if I was clear enough because my english is very poor. > Want I wanted to let clear is that there could be some conflicts with > GLib that you should take in consideration. I think the GLib version > used to link the GStreamer binaries si too cooler for windows and > won't be compatible with any other previous installation as the one > provided by the windows gtk installer. Hi Andoni, your comment is very clear. From what I understand, all that GStreamer uses from GTK on windows are the glib dlls (specifically libgio-2.0, libglib-2.0, libgmodule-2.0 and libgobject-2.0), so those are the only ones GStreamerOABuild needs to be linked against in case we want to create a GTK-compatible release of OABuild. From andres.colubri at gmail.com Sat Sep 13 22:42:07 2008 From: andres.colubri at gmail.com (Andres Colubri) Date: Sat, 13 Sep 2008 13:42:07 -0700 Subject: [gst-devel] Freshly baked Windows binaries of HEAD available In-Reply-To: <48CC2198.6070000@gmail.com> References: <48CAB8B0.8090801@gmail.com> <772db3280809130823y74bff3b9u4f45db8d0803c479@mail.gmail.com> <48CC2198.6070000@gmail.com> Message-ID: <48CC259F.1060507@gmail.com> >> What I mean is: If you want to provide some GStreamer binaries, try >> to link it against the glib version actually provided by the windows >> gtk installers as many of us will use GStreamer in combination with >> GTK. If you don't do this, your gstreamer binaries will work well on >> a standalone app, but they won't work at all with any previous >> installation. >> I dont know if I was clear enough because my english is very poor. >> Want I wanted to let clear is that there could be some conflicts with >> GLib that you should take in consideration. I think the GLib version >> used to link the GStreamer binaries si too cooler for windows and >> won't be compatible with any other previous installation as the one >> provided by the windows gtk installer. > Hi Andoni, your comment is very clear. From what I understand, all > that GStreamer uses from GTK on windows are the glib dlls > (specifically libgio-2.0, libglib-2.0, libgmodule-2.0 and > libgobject-2.0), so those are the only ones GStreamerOABuild needs to > be linked against in case we want to create a GTK-compatible release > of OABuild. I forgot to mention libgthread-2.0, it is also required by GStreamerOABuild. From ylatuya at gmail.com Sat Sep 13 19:19:05 2008 From: ylatuya at gmail.com (Andoni Morales) Date: Sat, 13 Sep 2008 19:19:05 +0200 Subject: [gst-devel] Freshly baked Windows binaries of HEAD available In-Reply-To: <48CC259F.1060507@gmail.com> References: <48CAB8B0.8090801@gmail.com> <772db3280809130823y74bff3b9u4f45db8d0803c479@mail.gmail.com> <48CC2198.6070000@gmail.com> <48CC259F.1060507@gmail.com> Message-ID: <772db3280809131019p568c3817nad50cdadf25b28ee@mail.gmail.com> Yes, thats all. As the glib version installed on a Windows SO is usually the one provided by the Gtk Windows installer, you should try to use this one to avoid incompatibilities. I've been also working on compiling ffmpeg with really full codec support and I've written a post on the Andres Colubri's blog with some guidelines to compile ffmpeg with more codecs support like faad, faac , mp3lame, x264, vorbis, dts, a52, xvid... check it out. I think we are very close to get GStreamer working really well on Windows!!! Regards, Andoni Morales 2008/9/13 Andres Colubri > > >> What I mean is: If you want to provide some GStreamer binaries, try > >> to link it against the glib version actually provided by the windows > >> gtk installers as many of us will use GStreamer in combination with > >> GTK. If you don't do this, your gstreamer binaries will work well on > >> a standalone app, but they won't work at all with any previous > >> installation. > >> I dont know if I was clear enough because my english is very poor. > >> Want I wanted to let clear is that there could be some conflicts with > >> GLib that you should take in consideration. I think the GLib version > >> used to link the GStreamer binaries si too cooler for windows and > >> won't be compatible with any other previous installation as the one > >> provided by the windows gtk installer. > > Hi Andoni, your comment is very clear. From what I understand, all > > that GStreamer uses from GTK on windows are the glib dlls > > (specifically libgio-2.0, libglib-2.0, libgmodule-2.0 and > > libgobject-2.0), so those are the only ones GStreamerOABuild needs to > > be linked against in case we want to create a GTK-compatible release > > of OABuild. > I forgot to mention libgthread-2.0, it is also required by > GStreamerOABuild. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ylatuya at gmail.com Sat Sep 13 19:49:10 2008 From: ylatuya at gmail.com (Andoni Morales) Date: Sat, 13 Sep 2008 19:49:10 +0200 Subject: [gst-devel] muxers, timestamps, sparse and continuous recordings In-Reply-To: <48C91DBF.40901@hora-obscura.de> References: <48C792A4.8040700@hora-obscura.de> <48C91DBF.40901@hora-obscura.de> Message-ID: <772db3280809131049p4c25d93dy1742e5a31030f88a@mail.gmail.com> hi, I've been working on a dv capturer and I implemented the play/pause recording doing the following: When I decide to start the recording, I add the encodebin to the main pipeline and then I link it to the tee. If you want to pause the encoding while previewing, unlink the encodebin, remove it from the main pipeline and I set it in the paused sate. If you want to restart the recording, readd it to the main pipeline and then relink it to tee. Finally, to stop the recording, send and eos event to the encodebin sink pad. The recorded file have a continous stream. Regards, Andoni Morales 2008/9/11 Stefan Kost > hi, > > Eric Zhang schrieb: > > Hi, Stefan: > > > > I think your pipeline is using GstSystemClock because you > > mentioned the source is a live element. If it is, the clock will keep > > increasing even if the pipeline is paused. This makes the timestamp > > noncontinuous. To generate a continuous timestamp, I think you can try > > to use the clock provided by your sink elements. Maybe this is not > > easy because the live element is different with other source elements. > > I meant pausing as on the application level. The videosrc ! tee name=t ! > queue ! xvimagesink runs continously. It only t. ! queue ! encoder ! > muxer ! filesink that get paused. > > Stefan > > > > > Eric > > > > 2008/9/10 Stefan Kost > > > > > > hi, > > > > i was wondering how muxers should handle timestamps on incoming > > buffers. > > Assume an applications that shows video from a camera. When you > > click a > > button it records to file, allowing to pause and unpause in > > between. The > > recorded file should have a continuous stream. If I don't do any > > special > > casing this is not the case. > > > > 1) When I playback the recorded file, I have an initial pause before > > video start (if I pressed record after two seconds, the video will > > start > > after two seconds). > > > > 2) If I pause in between, also in the playback there is a pause. > > > > Right now I work around with a pad probe that looks at disconts to > > aggregate a time_stamp_offset and correct all incoming buffers by > > subtracting that. It works but probably is not the right way. I > > believe > > this involves the use of segments, but I am not sure how. Also both > > behaviors might be valid (having a sparse and having a continuous > > stream). So the application would somehow be involved to select the > > desired behavior. Any comments? > > > > > > Stefan > > > > > > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > > challenge > > Build the coolest Linux based applications with Moblin SDK & win > > great prizes > > Grand prize is a trip for two to an Open Source event anywhere in > > the world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > ------------------------------------------------------------------------ > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Sat Sep 13 20:51:11 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 13 Sep 2008 21:51:11 +0300 Subject: [gst-devel] GStreamer core error in banshee: StateChange In-Reply-To: <87hc8lgkvy.fsf@lysmata.klauszeitler.de> References: <87hc8lgkvy.fsf@lysmata.klauszeitler.de> Message-ID: <48CC0B9F.6010106@hora-obscura.de> hi, Klaus Zeitler schrieb: > Hello, > all of a sudden I can't play music anymore. Instead banshee spits out hundreds > of the following 2 error messages: > > GStreamer core error: StateChange > GStreamer resource error: NotFound > > Not very informative I'd say. > Maybe somebody here has got an idea what happened or can point me towards the > right place to ask (if this isn't the right one :-). > Try running banshee from commandline like: GST_DEBUG="*:3" banshee Also please check gstreamer-properties if for some reason your preferred audiosink is invalid. You could also check other gstreamer based apps like totem. Stefan > > My problem started after I had installed sound-juicer (somebody suggested to > use sound-juicer to rip the CDs, since I just can't get banshee to fetch > metadata). I installed sound-juicer and ripped 2 CDs (without any problems) and > also played a CD. But next time I called banshee I got those odd GStreamer > errors and now I also get them when I try to play music with sound-juicer. > But then again that might be pure coincidence. > > Nevertheless I de-installed sound-juicer, but the errors didn't disappear for > banshee. > > I'm at a total loss. Maybe I should try to reinstall gstreamer, but there are > so many packages, that I have no idea what I should select. e.g. I see many > gstreamer-0_10... packages as well as gstreamer010... ones with yast2. > Can/should I install new gstreamer packages and if so which ones? > > OS: openSuse 10.3 > > BTW I also submitted a bug report for gstreamer. > > I appreciate any help. > > Thanks > > Klaus > From ensonic at hora-obscura.de Sat Sep 13 21:49:04 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 13 Sep 2008 22:49:04 +0300 Subject: [gst-devel] Possible gstreamer errors: volume, dispose, loop detected - oh my! In-Reply-To: <48CAB834.5040302@gmail.com> References: <48CAB834.5040302@gmail.com> Message-ID: <48CC1930.4050208@hora-obscura.de> hi, Ryan Kelln schrieb: > Hi Gstreamer devs, > (sorry to mods, I tried posting this without subscribing first, you can > delete/disapprove the post waiting for approval) > > I'm using gstreamer through gstreamer-java inside of processing. I'm > getting some odd errors and warnings and eventually a crash (perhaps a > null pointer). I'm hoping you might have a bit of insight or guidance > for how to avoid these problems or what I could do to fix them. Thanks. > > Environment: > Ubuntu 8.04 > gstreamer-team ppa (https://edge.launchpad.net/~gstreamer-team/+archive) > Sun Java 1.06_07 > Processing 148 > gstreamer-java 0.8 > > The errors happen in a variety of orders, here is an example of each: > > ** (GSVideo:9501): CRITICAL **: volume_transform_ip: assertion > `this->process != NULL' failed This means that the volume element is not configured. I use volume a lot and have never seen it. Could it be that the java bindungs handling something different here. > > (GSVideo:9501): GStreamer-CRITICAL **: > Trying to dispose element test, but it is not in the NULL state. > You need to explicitly set elements to the NULL state before > dropping the final reference, to allow them to clean up. > Do you set your pipeline to NULL state before releasing it. Dunnon how much this is abstracted by java-bindings again. > > (GSVideo:9501): GStreamer-WARNING **: loop detected in the graph of bin > GStreamer Audio Data Extractor: mysoundfile.wav!! No idea where this comes from. > > Then eventually Java will crash with a SIGSEGV fault in a native thread > which I'm assuming is gstreamer. Looks like it is accessing a null pointer. > > > Description of what I'm doing: > Custom software that is playing back multiple audio files at once. It > loads approx. 11 sound files (all wavs or aif errors and crash remain > the same) and starts and stops the files quite often. It also checks the > current volume and changes the volume each update (i.e. constantly > adjusting the volume). In addition I'm using the audiopanorama plug-in > to do constant panning. > > Occasionally I will get the "loop detected" warning on one or two of the > sound files, but which sound file it is seems to change and it doesn't > happen every run. The error appears to occur when the sound is first > played, not when the object is created. The program continues to > function after receiving this warning although the sound file doesn't play. > > I get the "volume_transform_ip" error quite often (every 5 minutes?) but > the program doesn't immediately crash. > > I get the "dispose element test" error only on a multi-core processor > (AMD Phenom) but not on an equivalent machine with just a single core. > Again, this doesn't cause an immediate crash. As far as I know I'm not > make any elements named "test", and I'm not disposing of anything until > the program ends but I am starting and stopping the audio streams. > > Things I've tried: > I've grepped the source for these and found the loop detected error in > gstbin.c (gst_bin_sort_iterator_next() )and the "dispose element not > NULL" in gstelement.c (gst_element_dispose() ) but I can't locate the > "volume_transform_ip" error. > > I've tired running the program with sounds disabled and it runs without > errors or crashes (for hours). If I only update the volumes of the > sounds every other update (or every 5, etc) it seems to help, although > its hard to tell if the errors occur that much slower or even more > infrequently. If I don't adjust the volume at all it seems to run fine > (survived 10+ mins without errors, still running tests). > > Some possible problem areas: > When is it safe to set and check the volume (especially in regards to > starting and stopping sounds)? Is it safe to set and check the volume in > the same update... multiple times? > I don't think the volume element causes these issues. Its safe to set this property at any time. Can you explain how you check for the volume? Are you checking the actual data or are you using the level element. > > I'm not sure if the errors are in gstreamer or just my use of it? I've > got to deliver this project really soon so workarounds might be the best > option if the error is gstreamers. If you're sure that gstreamer is at > fault I'd love some tips on how to provide more info (I'm not sure how > I'd debug gstreamer through all these other layers) and write up a bug > report. Try to get hold of the java binding maintainers. I suspect the issues there. Stefan > > Thanks very much, > Ryan > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From info at klauszeitler.de Sat Sep 13 22:50:46 2008 From: info at klauszeitler.de (Klaus Zeitler) Date: Sat, 13 Sep 2008 22:50:46 +0200 Subject: [gst-devel] GStreamer core error in banshee: StateChange In-Reply-To: <48CC0B9F.6010106@hora-obscura.de> (Stefan Kost's message of "Sat\, 13 Sep 2008 21\:51\:11 +0300") References: <87hc8lgkvy.fsf@lysmata.klauszeitler.de> <48CC0B9F.6010106@hora-obscura.de> Message-ID: <87vdwzhis9.fsf@lysmata.klauszeitler.de> >>>>> "Stefan" == Stefan Kost writes: Stefan> Stefan> Try running banshee from commandline like: Stefan> GST_DEBUG="*:3" banshee Stefan> Stefan> Also please check gstreamer-properties if for some reason your Stefan> preferred audiosink is invalid. gstreamer-properties shows: Audio output: Autodetect Audio input: ALSA BTW when I start gstreamer-properties I get gstreamer-properties-Message: Skipping unavailable plugin 'artsdsink' gstreamer-properties-Message: Skipping unavailable plugin 'pulsesink' gstreamer-properties-Message: Skipping unavailable plugin 'sdlvideosink' gstreamer-properties-Message: Skipping unavailable plugin 'v4lmjpegsrc' gstreamer-properties-Message: Skipping unavailable plugin 'qcamsrc' gstreamer-properties-Message: Skipping unavailable plugin 'v4l2src' gstreamer-properties-Message: Skipping unavailable plugin 'esdmon' gstreamer-properties-Message: Skipping unavailable plugin 'pulsesrc' Is that normal? Stefan> You could also check other gstreamer based apps like totem. That was a good hint. I tried totem, tried to play a FLAC file. No sound, but no error messages either. Then I started envy24control to check the sound output and finally I got an error message that I understood: No ICE1712 cards found Thus I de-installed and reinstalled my sound card and voila everything works just fine now. I have no idea what messed up the sound configuration. Nevertheless banshee and sound-juicer could do better than spitting out those 2 error messages I mentioned in my first mail thousands of times in such a case. Thanks a lot for your help Klaus -- ------------------------------------------------------- | Klaus Zeitler | | Raiffeisenstr. 31 a 90427 N?rnberg Germany | | Telefon: 49 911 316261 Fax: 316281 | | Email: info at klauszeitler.de | ------------------------------------------------------- --- Christian Fundamentalism: The doctrine that there is an absolutely powerful, infinitely knowledgeable, universe spanning entity that is deeply and personally concerned about my sex life. From kekko84 at gmail.com Sun Sep 14 09:47:42 2008 From: kekko84 at gmail.com (Francesco Argese) Date: Sun, 14 Sep 2008 09:47:42 +0200 Subject: [gst-devel] =?iso-8859-1?q?window_too_big_with_resolution_greater?= =?iso-8859-1?q?_than_352_=D7_288?= In-Reply-To: <3c1737210809110959n2bd87ae6n918bc4c3500c8518@mail.gmail.com> References: <3c1737210809110959n2bd87ae6n918bc4c3500c8518@mail.gmail.com> Message-ID: > Probably the higher resolution means your encoded frames are larger, > perhaps exceeding the MTU on the network, or just being dropped since > you're using UDP. Yes, probably this is the problem. I can confirm it because now I have tried with a version of gstreamer including debug (the same posted in this mailing and compiled with OABuild) and the application give me the following error: ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Could not read from resource. Additional debug info: ..\..\gst\udp\gstudpsrc.c(563): gst_udpsrc_create (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: receive error -1 (WSA error: 10040) Capturing udp packets toward the network analyzer Wireshark, I have understand that the packet exceeding a size near to 1500 are discarded by Windows (then return that error reported on this site (http://www.ultrabac.com/kb/ubq000192.htm) as message too long). With low resolution the packet are always under this limit so it works well. The error WSA error: 10040 is the following: A message sent on a datagram socket was larger than the internal message buffer or some other network limit, or the buffer used to receive a datagram into was smaller than the datagram itself. > Use a TCP transport, or a network protocol designed to handle lossy transports. I haven't tcp plugins in my gstreamer distribution on windows (i have checked it with a gst-inspect-0.10 | grep tcp). I have tried to use a different payloader also with codec h263 or mpeg4 but i have the following error at the receiver (it seems that the udpsrc problem is resolved in this manner): $ gst-launch-0.10 -v udpsrc port=5000 ! application/x-rtp,media=video,payload=9 6,encoding-name=MP4V-ES ! rtpmp4vdepay ! video/mpeg,mpegversion=4,systemstream= false ! ffdec_mpeg4 ! ffmpegcolorspace ! directdrawsink New clock: GstSystemClock Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... ** ** ERROR:(..\..\libs\gst\base\gstbasetransform.c:1141):gst_base_transform_prepare_output_buffer: assertion failed: (*out_buf != NULL) This application has requested the Runtime to terminate it in an unusual way. Please contact the application's support team for more information. The pipeline to send this flow is the following: gst-launch-0.10 videotestsrc ! video/x-raw-rgb,width=640,height=480 ! ffmpegc olorspace ! video/x-raw-yuv ! ffenc_mpeg4 ! video/mpeg,width=640,height=480,mpe gversion=4,systemstream=false ! rtpmp4vpay ! application/x-rtp,media=video,payl oad=96,encoding-name=MP4V-ES ! udpsink host=10.10.0.1 port=5000 New clock: GstSystemClock It seems that the buffer is not initialized. What could be the problem? Probably the rtp payloader? I have tried also with other encoders (h263, h263p) and their relative payloaders and depayloaders with the same result. These does not work also with lower resolution (the problem is different). Another question: is there a manner to specify the size of the sending udp packet? It probably could resolve my problems with gdppay that work well with small resolution (on Linux also with greater resolution). Thanks Francesco From ensonic at hora-obscura.de Sun Sep 14 13:31:55 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sun, 14 Sep 2008 14:31:55 +0300 Subject: [gst-devel] GStreamer core error in banshee: StateChange In-Reply-To: <87vdwzhis9.fsf@lysmata.klauszeitler.de> References: <87hc8lgkvy.fsf@lysmata.klauszeitler.de> <48CC0B9F.6010106@hora-obscura.de> <87vdwzhis9.fsf@lysmata.klauszeitler.de> Message-ID: <48CCF62B.60908@hora-obscura.de> Klaus Zeitler schrieb: >>>>>> "Stefan" == Stefan Kost writes: > Stefan> > Stefan> Try running banshee from commandline like: > Stefan> GST_DEBUG="*:3" banshee > Stefan> > Stefan> Also please check gstreamer-properties if for some reason your > Stefan> preferred audiosink is invalid. > > gstreamer-properties shows: > Audio output: Autodetect > Audio input: ALSA > > BTW when I start gstreamer-properties I get > gstreamer-properties-Message: Skipping unavailable plugin 'artsdsink' > gstreamer-properties-Message: Skipping unavailable plugin 'pulsesink' > gstreamer-properties-Message: Skipping unavailable plugin 'sdlvideosink' > gstreamer-properties-Message: Skipping unavailable plugin 'v4lmjpegsrc' > gstreamer-properties-Message: Skipping unavailable plugin 'qcamsrc' > gstreamer-properties-Message: Skipping unavailable plugin 'v4l2src' > gstreamer-properties-Message: Skipping unavailable plugin 'esdmon' > gstreamer-properties-Message: Skipping unavailable plugin 'pulsesrc' > > Is that normal? > > Stefan> You could also check other gstreamer based apps like totem. > > That was a good hint. I tried totem, tried to play a FLAC file. > No sound, but no error messages either. Then I started envy24control > to check the sound output and finally I got an error message that > I understood: No ICE1712 cards found > > Thus I de-installed and reinstalled my sound card and voila everything > works just fine now. I have no idea what messed up the sound configuration. > > Nevertheless banshee and sound-juicer could do better than spitting out > those 2 error messages I mentioned in my first mail thousands of times in > such a case. > > Thanks a lot for your help > > Klaus > Godd that you got it fixed. Feel free to add bug reports for banshee and soundjuicer. Error reporting is probably always a weak point in software. Stefan From ylatuya at gmail.com Sun Sep 14 13:48:02 2008 From: ylatuya at gmail.com (Andoni Morales) Date: Sun, 14 Sep 2008 13:48:02 +0200 Subject: [gst-devel] Freshly baked Windows binaries of HEAD available In-Reply-To: <772db3280809131019p568c3817nad50cdadf25b28ee@mail.gmail.com> References: <48CAB8B0.8090801@gmail.com> <772db3280809130823y74bff3b9u4f45db8d0803c479@mail.gmail.com> <48CC2198.6070000@gmail.com> <48CC259F.1060507@gmail.com> <772db3280809131019p568c3817nad50cdadf25b28ee@mail.gmail.com> Message-ID: <772db3280809140448rdcffe51l42e39fbda39e684@mail.gmail.com> Hi, The last think you should consider on building GStreamer for w?Windows is to add the dshowvideosrc patch from Julien Isorce ( http://bugzilla.gnome.org/show_bug.cgi?id=517203), as without it is not possible to use other video size than the default. 2008/9/13 Andoni Morales > Yes, thats all. > As the glib version installed on a Windows SO is usually the one provided > by the Gtk Windows installer, you should try to use this one to avoid > incompatibilities. > I've been also working on compiling ffmpeg with really full codec support > and I've written a post on the Andres Colubri's blog with some guidelines > to compile ffmpeg with more codecs support like faad, faac , mp3lame, x264, > vorbis, dts, a52, xvid... check it out. > I think we are very close to get GStreamer working really well on > Windows!!! > > Regards, > Andoni Morales > > 2008/9/13 Andres Colubri > > >> >> What I mean is: If you want to provide some GStreamer binaries, try >> >> to link it against the glib version actually provided by the windows >> >> gtk installers as many of us will use GStreamer in combination with >> >> GTK. If you don't do this, your gstreamer binaries will work well on >> >> a standalone app, but they won't work at all with any previous >> >> installation. >> >> I dont know if I was clear enough because my english is very poor. >> >> Want I wanted to let clear is that there could be some conflicts with >> >> GLib that you should take in consideration. I think the GLib version >> >> used to link the GStreamer binaries si too cooler for windows and >> >> won't be compatible with any other previous installation as the one >> >> provided by the windows gtk installer. >> > Hi Andoni, your comment is very clear. From what I understand, all >> > that GStreamer uses from GTK on windows are the glib dlls >> > (specifically libgio-2.0, libglib-2.0, libgmodule-2.0 and >> > libgobject-2.0), so those are the only ones GStreamerOABuild needs to >> > be linked against in case we want to create a GTK-compatible release >> > of OABuild. >> I forgot to mention libgthread-2.0, it is also required by >> GStreamerOABuild. >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> challenge >> Build the coolest Linux based applications with Moblin SDK & win great >> prizes >> Grand prize is a trip for two to an Open Source event anywhere in the >> world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gstelzz at yahoo.fr Sun Sep 14 19:29:12 2008 From: gstelzz at yahoo.fr (Aurelien Grimaud) Date: Sun, 14 Sep 2008 19:29:12 +0200 Subject: [gst-devel] =?iso-8859-1?q?window_too_big_with_resolution_greater?= =?iso-8859-1?q?_than_352_=D7_288?= In-Reply-To: References: <3c1737210809110959n2bd87ae6n918bc4c3500c8518@mail.gmail.com> Message-ID: <48CD49E8.4000402@yahoo.fr> Hi, Francesco Argese a ?crit : >> Probably the higher resolution means your encoded frames are larger, >> perhaps exceeding the MTU on the network, or just being dropped since >> you're using UDP. >> > > Yes, probably this is the problem. I can confirm it because now I have > tried with a version of gstreamer including debug (the same posted in > this mailing and compiled with OABuild) and the application give me > the following error: > > ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Could > not read from resource. > Additional debug info: > ..\..\gst\udp\gstudpsrc.c(563): gst_udpsrc_create (): > /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: > receive error -1 (WSA error: 10040) > > Capturing udp packets toward the network analyzer Wireshark, I have > understand that the packet exceeding a size near to 1500 are discarded > by Windows (then return that error reported on this site > (http://www.ultrabac.com/kb/ubq000192.htm) as message too long). With > low resolution the packet are always under this limit so it works > well. > > The error WSA error: 10040 is the following: A message sent on a > datagram socket was larger than the > internal message buffer or some other network limit, or the > buffer used to receive a datagram into was smaller than > the datagram itself. > > >> Use a TCP transport, or a network protocol designed to handle lossy transports. >> > > I haven't tcp plugins in my gstreamer distribution on windows (i have > checked it with a gst-inspect-0.10 | grep tcp). I have tried to use a > different payloader also with codec h263 or mpeg4 but i have the > following error at the receiver (it seems that the udpsrc problem is > resolved in this manner): > > $ gst-launch-0.10 -v udpsrc port=5000 ! application/x-rtp,media=video,payload=9 > 6,encoding-name=MP4V-ES ! rtpmp4vdepay ! video/mpeg,mpegversion=4,systemstream= > false ! ffdec_mpeg4 ! ffmpegcolorspace ! directdrawsink > New clock: GstSystemClock > Setting pipeline to PAUSED ... > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > ** > ** ERROR:(..\..\libs\gst\base\gstbasetransform.c:1141):gst_base_transform_prepare_output_buffer: > assertion failed: (*out_buf != NULL) > This application has requested the Runtime to terminate it in an unusual way. > Please contact the application's support team for more information. > > Check out http://bugzilla.gnome.org/show_bug.cgi?id=545853 http://bugzilla.gnome.org/show_bug.cgi?id=551509 the assertion failed problem is due to capsfilter. Specify full caps after udpsrc to get it working. clock-rate is missing. > The pipeline to send this flow is the following: > > gst-launch-0.10 videotestsrc ! video/x-raw-rgb,width=640,height=480 ! ffmpegc > olorspace ! video/x-raw-yuv ! ffenc_mpeg4 ! video/mpeg,width=640,height=480,mpe > gversion=4,systemstream=false ! rtpmp4vpay ! application/x-rtp,media=video,payl > oad=96,encoding-name=MP4V-ES ! udpsink host=10.10.0.1 port=5000 > New clock: GstSystemClock > > It seems that the buffer is not initialized. What could be the > problem? Probably the rtp payloader? I have tried also with other > encoders (h263, h263p) and their relative payloaders and depayloaders > with the same result. These does not work also with lower resolution > (the problem is different). > > Is first colorspace really necessary ? videotestsrc may produce x-raw-yuv. Try videotestsrc ! video/x-raw-yuv,width=640,height=480 ! ffenc_mpeg4 ... > Another question: is there a manner to specify the size of the sending > udp packet? It probably could resolve my problems with gdppay that > work well with small resolution (on Linux also with greater > resolution). > > Thanks > Francesco > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From ryankelln at gmail.com Mon Sep 15 06:29:41 2008 From: ryankelln at gmail.com (Ryan Kelln) Date: Mon, 15 Sep 2008 00:29:41 -0400 Subject: [gst-devel] Possible gstreamer errors: volume, dispose, loop detected - oh my! In-Reply-To: <48CC1930.4050208@hora-obscura.de> References: <48CAB834.5040302@gmail.com> <48CC1930.4050208@hora-obscura.de> Message-ID: <48CDE4B5.9070906@gmail.com> Thanks for the info Stefan, most helpful. Some comments below. Stefan Kost wrote: > Ryan Kelln schrieb: >> ** (GSVideo:9501): CRITICAL **: volume_transform_ip: assertion >> `this->process != NULL' failed > This means that the volume element is not configured. I use volume a > lot and have never seen it. Could it be that the java bindungs > handling something different here. > Yup, I'll take a look at the java that is between me and gstreamer, however on my end I don't every delete any sounds once they are around. They are started and stopped, seeked, panned and volume changed, but I don't see why the volume element would ever become unconfigured from any of those actions. If I slow down the rate at which the volume is checked and never start/stop the sounds then this is the only error I get and it drastically slows the crash (I've not seen it in 8+ hours of continuous running). Hmm, here is the gstreamer-java code between me and gstreamer. It seems simple enough, unless there is a bug in the jna library stuff that this relies on: public void setVolume(double volume) { gobj.g_object_set(this, "volume", volume); } public double getVolume() { double[] volume = { 0d }; gobj.g_object_get(this, "volume", volume); return volume[0]; } public interface GObjectAPI extends Library { static GObjectAPI gobj = GNative.loadLibrary("gobject-2.0", GObjectAPI.class, new HashMap() {{ put(Library.OPTION_TYPE_MAPPER, new GTypeMapper()); }}); ... void g_object_set(GObject obj, String propertyName, Object... data); void g_object_get(GObject obj, String propertyName, Object... data); See anything wrong with that? >> (GSVideo:9501): GStreamer-CRITICAL **: >> Trying to dispose element test, but it is not in the NULL state. >> You need to explicitly set elements to the NULL state before >> dropping the final reference, to allow them to clean up. > Do you set your pipeline to NULL state before releasing it. Dunnon how > much this is abstracted by java-bindings again. The problem I'm having tracking this down is one in is making an element called "test" specifically in any of the code. A grep or search of the code turns up nothing interesting for "test". And as I stated I'm not disposing any elements (until the software exits), unless the sounds or some element of the sounds is being disposed when stopped.... hmm, nope, that isn't the case. A question about your statement. Are you saying that before you call dispose() you need to stop the pipeline (set the state to NULL)? This does seem to be what is happening. >> (GSVideo:9501): GStreamer-WARNING **: loop detected in the graph of bin >> GStreamer Audio Data Extractor: mysoundfile.wav!! > > No idea where this comes from. Damn, I hoped this would be something that other people had run into: here it is in context (sadly the code makes no sense to me): gstbin.c: line 1747: /* get next element in iterator. the returned element has the * refcount increased */ static GstIteratorResult gst_bin_sort_iterator_next (GstBinSortIterator * bit, gpointer * result) { GstBin *bin = bit->bin; /* empty queue, we have to find a next best element */ if (g_queue_is_empty (bit->queue)) { GstElement *best; bit->best = NULL; bit->best_deg = G_MAXINT; g_list_foreach (bin->children, (GFunc) find_element, bit); if ((best = bit->best)) { if (bit->best_deg != 0) { /* we don't fail on this one yet */ GST_WARNING_OBJECT (bin, "loop dected in graph"); g_warning ("loop detected in the graph of bin %s!!", GST_ELEMENT_NAME (bin)); } /* best unhandled element, schedule as next element */ GST_DEBUG_OBJECT (bin, "queue empty, next best: %s", GST_ELEMENT_NAME (best)); gst_object_ref (best); HASH_SET_DEGREE (bit, best, -1); *result = best; } else { GST_DEBUG_OBJECT (bin, "queue empty, elements exhausted"); /* no more unhandled elements, we are done */ return GST_ITERATOR_DONE; } } else { /* everything added to the queue got reffed */ *result = g_queue_pop_head (bit->queue); } .... > I don't think the volume element causes these issues. Its safe to set > this property at any time. Can you explain how you check for the > volume? Are you checking the actual data or are you using the level > element. See above for how I'm checking the volume. Thanks again for the help, the more I test this bug and learn about gstreamer the more bizarre this seems to be. I'll test the entire thing with entirely different sounds tomorrow. Ryan From wim.taymans at gmail.com Mon Sep 15 07:00:26 2008 From: wim.taymans at gmail.com (Wim Taymans) Date: Sun, 14 Sep 2008 22:00:26 -0700 Subject: [gst-devel] Possible gstreamer errors: volume, dispose, loop detected - oh my! In-Reply-To: <48CDE4B5.9070906@gmail.com> References: <48CAB834.5040302@gmail.com> <48CC1930.4050208@hora-obscura.de> <48CDE4B5.9070906@gmail.com> Message-ID: <1221454826.6848.15.camel@metal> On Mon, 2008-09-15 at 00:29 -0400, Ryan Kelln wrote: > > >> (GSVideo:9501): GStreamer-WARNING **: loop detected in the graph of bin > >> GStreamer Audio Data Extractor: mysoundfile.wav!! > > > > No idea where this comes from. > Damn, I hoped this would be something that other people had run into: > here it is in context (sadly the code makes no sense to me): > gstbin.c: line 1747: it's very likely because of this bug: http://bugzilla.gnome.org/show_bug.cgi?id=510354 Wim From irfanshaikh at tataelxsi.co.in Mon Sep 15 08:30:26 2008 From: irfanshaikh at tataelxsi.co.in (irfanshaikh) Date: Mon, 15 Sep 2008 12:00:26 +0530 Subject: [gst-devel] Possible gstreamer errors: volume, dispose, loop detected - oh my! In-Reply-To: <1221454826.6848.15.camel@metal> Message-ID: <004c01c916fc$8adeb2b0$37033c0a@telxsi.com> Hi All, I want to stream a .asf file on Windows media player using following pipeline through Gstreamer. 1) gst-launch filesrc location=/root/Desktop/mjpegi.asf ! rtspwms ! tcpserversink port=554 host=10.60.3.55 2) gst-launch filesrc location=/root/Desktop/mjpegi.asf ! rtspwms ! udpsink port=5005 host=10.60.3.55 I am getting following warnings: WARNING: erroneous pipeline: could not link filesrc0 to rtspwms0 On windows media I have used Open URL : rtsp://: Please can me help me out how to use this element. Thankyou in advance, Irfan. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From irfanshaikh at tataelxsi.co.in Mon Sep 15 09:45:28 2008 From: irfanshaikh at tataelxsi.co.in (irfanshaikh) Date: Mon, 15 Sep 2008 13:15:28 +0530 Subject: [gst-devel] Possible gstreamer errors: volume, dispose, loop detected - oh my! In-Reply-To: <1221454826.6848.15.camel@metal> Message-ID: <006201c91707$06975240$37033c0a@telxsi.com> Hi Wim, Thanks for the understanding. My requirement is as follows: I have some encoded audio and video files.I need to put them into asf container and stream it using gstreamer and receive the same on client pc using wmp player.Can you suggest how can i acheive this? Thanks in advance, Regards, Irfan. -----Original Message----- From: gstreamer-devel-bounces at lists.sourceforge.net [mailto:gstreamer-devel-bounces at lists.sourceforge.net]On Behalf Of Wim Taymans Sent: Monday, September 15, 2008 10:30 AM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] Possible gstreamer errors: volume, dispose,loop detected - oh my! On Mon, 2008-09-15 at 00:29 -0400, Ryan Kelln wrote: > > >> (GSVideo:9501): GStreamer-WARNING **: loop detected in the graph of bin > >> GStreamer Audio Data Extractor: mysoundfile.wav!! > > > > No idea where this comes from. > Damn, I hoped this would be something that other people had run into: > here it is in context (sadly the code makes no sense to me): > gstbin.c: line 1747: it's very likely because of this bug: http://bugzilla.gnome.org/show_bug.cgi?id=510354 Wim ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From arnabsamanta at tataelxsi.co.in Mon Sep 15 10:44:51 2008 From: arnabsamanta at tataelxsi.co.in (arnabsamanta) Date: Mon, 15 Sep 2008 14:14:51 +0530 Subject: [gst-devel] pad creation and linking for any element In-Reply-To: Message-ID: <016701c9170f$51d4d220$26033c0a@telxsi.com> Hi, can any body tel me what exactly the macro GST_PAD_LINKFUNC does ? i searched in the http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/gstreamer/html/Gs tPad.html but i could not find it. is it required to use this macro and link the pads when ever a new elemant is to be created ? The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From kekko84 at gmail.com Mon Sep 15 10:44:42 2008 From: kekko84 at gmail.com (Francesco Argese) Date: Mon, 15 Sep 2008 10:44:42 +0200 Subject: [gst-devel] =?iso-8859-1?q?window_too_big_with_resolution_greater?= =?iso-8859-1?q?_than_352_=D7_288?= In-Reply-To: <48CD49E8.4000402@yahoo.fr> References: <3c1737210809110959n2bd87ae6n918bc4c3500c8518@mail.gmail.com> <48CD49E8.4000402@yahoo.fr> Message-ID: Ok. I have resolved my problem with these two pipelines: Receiver: gst-launch-0.10 -v udpsrc port=5000 ! capsfilter caps="application/x-rtp,media=(string)video,clock-rate=90000,encoding-name=(string)MP4V-ES,profile-level-id=(string)1,payload=(int)96" ! rtpmp4vdepay ! capsfilter caps="video/mpeg,width=(int)640,height=(int)480,framerate=(fraction)30/1,mpegversion=4,systemstream=(boolean)false" !!ffdec_mpeg4! ffmpegcolorspace ! directdrawsink Sender: gst-launch-0.10 -v videotestsrc is-live=true ! video/x-raw-rgb,width=640,height=480 ! ffmpegcolorspace ! ffenc_mpeg4 ! rtppmp4vpay ! udpsink host=127.0.0.1 port=5000 Thanks all Francesco From irfanshaikh at tataelxsi.co.in Mon Sep 15 12:53:41 2008 From: irfanshaikh at tataelxsi.co.in (irfanshaikh) Date: Mon, 15 Sep 2008 16:23:41 +0530 Subject: [gst-devel] Query : ASF streaming on client WMP player. In-Reply-To: <87hc8lgkvy.fsf@lysmata.klauszeitler.de> Message-ID: <008a01c91721$515fc400$37033c0a@telxsi.com> Hi All, I tried streaming a file in ASF file format using gstreamer to a client pc having wmp player. I hav tried various pipelines to stream asf file format using gst-launch. Finally after trying a lot, i am unable to find elements in GStreamer to implement a pipeline to stream a asf file on WMP player on client PC. Can you guys share information regarding my query. Do i need to write a plugin to achieve this. I am pretty unclear about it.I would be very thankful if you look into my problem.And suggest me different possible alternatives. Regards, Irfan. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From rbultje at ronald.bitfreak.net Mon Sep 15 13:39:24 2008 From: rbultje at ronald.bitfreak.net (Ronald S. Bultje) Date: Mon, 15 Sep 2008 07:39:24 -0400 Subject: [gst-devel] Fwd: regarding ffmpeg_mux In-Reply-To: <001601c91723$db210bc0$68033c0a@telxsi.com> References: <001601c91723$db210bc0$68033c0a@telxsi.com> Message-ID: <34539a480809150439s59ca1706y928a5f90efaf55f5@mail.gmail.com> Can someone help this guy? ---------- Forwarded message ---------- From: ajitjohn Date: Mon, Sep 15, 2008 at 7:11 AM Subject: regarding ffmpeg_mux To: Ronald Bultje Hii, I am doing a project in which i am supposed to stream .asf file using gstreamer to a client pc which has a wmp player.So to acheive this aim can i use the element ffmux_asf written by you and directly stream using udpsink.Is my understanding correct regarding this.Can you suggest all possible alternatives to stream asf file using gstreamer and receive throgh wmp player at the client side. regards, Ajit. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From learning_gst at hotmail.com Mon Sep 15 14:52:48 2008 From: learning_gst at hotmail.com (learning gst) Date: Mon, 15 Sep 2008 12:52:48 +0000 Subject: [gst-devel] libs/gst/check/gstcheck.h:32:19: check.h: No such file or directory Message-ID: Hello, I tried to build Gstreamer with scratchbox. But I got the error: libs/gst/check/gstcheck.h:32:19: check.h: No such file or directory I hope to know how to fix it. _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us -------------- next part -------------- An HTML attachment was scrubbed... URL: From felipe.contreras at nokia.com Mon Sep 15 15:15:48 2008 From: felipe.contreras at nokia.com (Felipe Contreras) Date: Mon, 15 Sep 2008 16:15:48 +0300 Subject: [gst-devel] [gst-embedded] libs/gst/check/gstcheck.h:32:19: check.h: No such file or directory In-Reply-To: References: Message-ID: <1221484548.3408.1.camel@localhost.localdomain> On Mon, 2008-09-15 at 12:52 +0000, ext learning gst wrote: > Hello, > > I tried to build Gstreamer with scratchbox. But I got the error: > libs/gst/check/gstcheck.h:32:19: check.h: No such file or directory > > I hope to know how to fix it. Try installing libcheck. Although It really doesn't make sense to check for libcheck when cross-compiling, somebody should fix that :) -- Felipe Contreras From ensonic at hora-obscura.de Mon Sep 15 16:23:28 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Mon, 15 Sep 2008 17:23:28 +0300 Subject: [gst-devel] aac parser In-Reply-To: References: <72cf309c0809020419w17e87d85yd86165633f682471@mail.gmail.com> <48BD3E7E.9060000@hora-obscura.de> <48C0FB1E.4070201@hora-obscura.de> Message-ID: <48CE6FE0.4050900@hora-obscura.de> hi, there you go http://bugzilla.gnome.org/show_bug.cgi?id=518857 Stefan Zheng, Huan schrieb: > Hi, Stefan > Where will you post the announcement? > I'm eager to see the code. :) > > Best Regards, Zheng, Huan(ZBT) > OTC/SSD/SSG > Intel Aisa-Pacific Research & Developement Ltd > Tel: 021-6116 6435 > Inet: 8821 6435 > Cub: 3W035 > -----Original Message----- > From: gstreamer-devel-bounces at lists.sourceforge.net [mailto:gstreamer-devel-bounces at lists.sourceforge.net] On Behalf Of Stefan Kost > Sent: 2008?9?5? 17:26 > To: Discussion of the development of GStreamer > Subject: Re: [gst-devel] aac parser > > Hi, > > Zheng, Huan schrieb: > >> Stefan >> Could you please elaborate on this "Soon"? One month or several months? :) >> >> > Hopefully next week. > > >> It would be very nice to have an aac parser in gstreamer. >> And one more question: Is this parser able to parse ADIF stream into single frames? Because what I heard now is that ADIF has only one header at the beginning, and the offset of each frame can not be located unless you have finished decoding. >> >> > lets discuss this once its published. > Stefan > > >> Thanks! >> >> Best Regards, Zheng, Huan(ZBT) >> OTC/SSD/SSG >> Intel Aisa-Pacific Research & Developement Ltd >> Tel: 021-6116 6435 >> Inet: 8821 6435 >> Cub: 3W035 >> -----Original Message----- >> From: gstreamer-devel-bounces at lists.sourceforge.net [mailto:gstreamer-devel-bounces at lists.sourceforge.net] On Behalf Of Stefan Kost >> Sent: 2008?9?2? 21:24 >> To: Discussion of the development of GStreamer >> Subject: Re: [gst-devel] aac parser >> >> hi, >> >> there will be one in gst-plugin-bad soon. >> >> Stefan >> >> Sachin Pandhare schrieb: >> >> >>> Hi, >>> if aac parser needs to be developed can we take some plugin code as a >>> reference and which one will be a suitable candidate for this? >>> thanks, >>> Sachin >>> ------------------------------------------------------------------------ >>> >>> ------------------------------------------------------------------------- >>> This SF.Net email is sponsored by the Moblin Your Move Developer's challenge >>> Build the coolest Linux based applications with Moblin SDK & win great prizes >>> Grand prize is a trip for two to an Open Source event anywhere in the world >>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >>> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's challenge >> Build the coolest Linux based applications with Moblin SDK & win great prizes >> Grand prize is a trip for two to an Open Source event anywhere in the world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's challenge >> Build the coolest Linux based applications with Moblin SDK & win great prizes >> Grand prize is a trip for two to an Open Source event anywhere in the world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ensonic at hora-obscura.de Mon Sep 15 16:43:47 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Mon, 15 Sep 2008 17:43:47 +0300 Subject: [gst-devel] [gst-embedded] libs/gst/check/gstcheck.h:32:19: check.h: No such file or directory In-Reply-To: <1221484548.3408.1.camel@localhost.localdomain> References: <1221484548.3408.1.camel@localhost.localdomain> Message-ID: <48CE74A3.6020209@hora-obscura.de> hi all, there is a report with a patch at: http://bugzilla.gnome.org/show_bug.cgi?id=551952 It works for me, please try it. Stefan Felipe Contreras schrieb: > On Mon, 2008-09-15 at 12:52 +0000, ext learning gst wrote: > >> Hello, >> >> I tried to build Gstreamer with scratchbox. But I got the error: >> libs/gst/check/gstcheck.h:32:19: check.h: No such file or directory >> >> I hope to know how to fix it. >> > > Try installing libcheck. > > Although It really doesn't make sense to check for libcheck when > cross-compiling, somebody should fix that :) > > From wim.taymans at gmail.com Mon Sep 15 16:55:54 2008 From: wim.taymans at gmail.com (Wim Taymans) Date: Mon, 15 Sep 2008 07:55:54 -0700 Subject: [gst-devel] pad creation and linking for any element In-Reply-To: <016701c9170f$51d4d220$26033c0a@telxsi.com> References: <016701c9170f$51d4d220$26033c0a@telxsi.com> Message-ID: <1221490554.6848.43.camel@metal> On Mon, 2008-09-15 at 14:14 +0530, arnabsamanta wrote: > Hi, > can any body tel me what exactly the macro GST_PAD_LINKFUNC does ? i > searched in the > http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/gstreamer/html/Gs > tPad.html but i could not find it. It is a macro to access the the linkfunc member in the GstPad object. It's not documented because you would not normally want to use it with the macro. > > is it required to use this macro and link the pads when ever a new elemant > is to be created ? No, this macro is only to be used by the core. If you want to link pads use a function like gst_pad_link() or gst_element_link(). Wim > > > > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From wim.taymans at gmail.com Mon Sep 15 17:42:46 2008 From: wim.taymans at gmail.com (Wim Taymans) Date: Mon, 15 Sep 2008 08:42:46 -0700 Subject: [gst-devel] rtspwms element query In-Reply-To: <001f01c91725$2a53e5e0$68033c0a@telxsi.com> References: <001f01c91725$2a53e5e0$68033c0a@telxsi.com> Message-ID: <1221493366.6848.75.camel@metal> On Mon, 2008-09-15 at 16:51 +0530, ajitjohn wrote: > Hii, > > I am doing a project in which i am supposed to stream .asf file using > gstreamer to a client pc which has a wmp player.So to acheive this aim can i > use the element ffmux_asf written by you and directly stream using > udpsink.Is my understanding correct regarding this.Can you suggest all > possible alternatives to stream asf file using gstreamer and receive throgh > wmp player at the client side. > Hello, Dude, stop posting this same question around to a zillion of people multiple times. Nobody is able to answer your question because it is underspecified. Also there is no easy answer to your question, you are asking people how to write a multimedia streaming server. Anyway, I'll start with the possibilities and give you an idea on how to implement them. First use case is where we assume the asf file is already available on the server. This can be, for example, achieved by transcoding (google gentrans for a gstreamer based transcoder) and uploading an asf file to the streaming server. - stream asf over http: Install a webserver, upload the file, stream the file with http. No GStreamer involved here although you could write a custom web module that hands a file descriptor to a gstreamer pipeline that uses fdsink to stream to the client. I believe flumotion has code for this in the proprietary extensions. - stream asf over mms. This is basically writing a custom mms server, documents on how mms works can be found on the internets although my experience is that most documents are not very accurate. There is no code in GStreamer that you can reuse for this task. I'm also not aware of an open source implementation of such a server. I would not recommend walking this path, microsoft is discontinuing mms in favour of rtsp. - stream asf over rtsp. You'll need to write an RTSP server that configures the gstreamer transport pipeline. You'll also need to write an asf payloader both for udp and tcp transports. A document on how an rtsp server would work in gstreamer can be found here: http://webcvs.freedesktop.org/gstreamer/gst-plugins-good/gst/rtsp/README?revision=1.5&view=markup There are a bunch of helper libraries in the gstreamer plugins base module that can assist you in creating and parsing RTSP messages, SDP messages and RTP packets. Second use case would be when you need to transcode or encode into an asf stream on the fly. We're talking about live capture or live transcoding. For this you will need wma/wmv encoders and a good asf muxer. ffmux_asf is not a good muxer, I would not use it. I understand Fluendo can provide these encoders and muxers to you. Again I would recommend using either a http server or an rtsp server to prepare the gstreamer transcoding/encoding pipelines. Hope this helps, Wim > regards, > Ajit. > > -----Original Message----- > From: Wim Taymans [mailto:wim.taymans at gmail.com] > Sent: Monday, September 15, 2008 12:57 PM > To: irfanshaikh at tataelxsi.co.in > Cc: Ajit S John > Subject: Re: rtspwms element query > > > On Mon, 2008-09-15 at 11:56 +0530, irfanshaikh wrote: > > Hi wim, > > > > I want to stream a .asf file on Windows media player using following > > pipeline through Gstreamer. > > > > 1) gst-launch filesrc location=/root/Desktop/mjpegi.asf ! rtspwms ! > > tcpserversink port=554 host=10.60.3.55 > > > > 2) gst-launch filesrc location=/root/Desktop/mjpegi.asf ! rtspwms ! > udpsink > > port=5005 host=10.60.3.55 > > > > > > I am getting following warnings: > > WARNING: erroneous pipeline: could not link filesrc0 to rtspwms0 > > rtpwms is an rtsp extension object, it's used to implement the windows > specific rtsp extensions. It is not meant to be used as an element in a > pipeline. > > We don't currently have elements in GStreamer to implement a windows > streaming server. We have however some support to receive asf streams > from an mms or rtsp windows server. > > Wim > > > > > On windows media I have used > > Open URL : rtsp://: > > > > Please can me help me out how to use this element. > > > > Thankyou in advance, > > > > Irfan. > > > > > > > > The information contained in this electronic message and any attachments > to this message are intended for the exclusive use of the addressee(s) and > may contain proprietary, confidential or privileged information. If you are > not the intended recipient, you should not disseminate, distribute or copy > this e-mail. Please notify the sender immediately and destroy all copies of > this message and any attachments contained in it. > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From julien.pauty at gmail.com Mon Sep 15 21:15:44 2008 From: julien.pauty at gmail.com (Julien Pauty) Date: Mon, 15 Sep 2008 21:15:44 +0200 Subject: [gst-devel] Writing a transport stream to the disk and reading it at the same time Message-ID: Hello, I have an application with two pipelines. The first pipeline writes to the disk the transport streams provided by a dvbbasebin. My second pipeline reads the transport streams from the disk. My problem is that playing stops after a few seconds. I'm waiting a few seconds before starting playing to be sure that enough data has been saved, but this is not sufficient. My pipeling looks like this: filesrc ! flutsdemux ! queue ! audiodec ! audiosink ! queue! videodec ! videosink . If I use the playbin instead of my custom pipeline, it works. However, I still want to use my pipeline, because it can seek through the transport stream, whereas I'm not able to seek with the playbin. With a big file I don't have this problem, suggesting some buffering problem. My debbuging session tends to show that, if the file is small (<10meg) the filesrc reads the whole file and send an EOS event. I thought that I may need some prerolling and I tried to insert a queue between the filesrc and the demuxer, but this does not help. I think it's possible to do it since the playbin can handle this. Unfortunally sources of the playbin are rather complex. Any idea is welcomed, Cheers, Julien -------------- next part -------------- An HTML attachment was scrubbed... URL: From andres.colubri at gmail.com Mon Sep 15 23:37:53 2008 From: andres.colubri at gmail.com (Andres Colubri) Date: Mon, 15 Sep 2008 14:37:53 -0700 Subject: [gst-devel] Freshly baked Windows binaries of HEAD available In-Reply-To: <772db3280809140448rdcffe51l42e39fbda39e684@mail.gmail.com> References: <48CAB8B0.8090801@gmail.com> <772db3280809130823y74bff3b9u4f45db8d0803c479@mail.gmail.com> <48CC2198.6070000@gmail.com> <48CC259F.1060507@gmail.com> <772db3280809131019p568c3817nad50cdadf25b28ee@mail.gmail.com> <772db3280809140448rdcffe51l42e39fbda39e684@mail.gmail.com> Message-ID: <48CED5B1.1050204@gmail.com> I applied those patches on the dshowwrapper plugin in OPBuild in order to generate the installer I posted the other day. But there seems to be something wrong. If you try dshowvideosrc from gst-launch (on windows vista) you get this error: > gst-launch-0.10.exe dshowvideosrc ! ffmpegcolorspace ! directdrawsink Setting pipeline to PAUSED ... ERROR: Pipeline doesn't want to pause. Setting pipeline to NULL ... FREEING pipeline ... Andoni Morales wrote: > Hi, > > The last think you should consider on building GStreamer for > w?Windows is to add the dshowvideosrc patch from Julien Isorce > (http://bugzilla.gnome.org/show_bug.cgi?id=517203), as without it is > not possible to use other video size than the default. > > > 2008/9/13 Andoni Morales > > > Yes, thats all. > As the glib version installed on a Windows SO is usually the one > provided by the Gtk Windows installer, you should try to use this > one to avoid incompatibilities. > I've been also working on compiling ffmpeg with really full > codec support and I've written a post on the Andres Colubri's > blog with some guidelines to compile ffmpeg with more codecs > support like faad, faac , mp3lame, x264, vorbis, dts, > a52, xvid... check it out. > I think we are very close to get GStreamer working really well on > Windows!!! > > Regards, > Andoni Morales > > 2008/9/13 Andres Colubri > > > > >> What I mean is: If you want to provide some GStreamer > binaries, try > >> to link it against the glib version actually provided by > the windows > >> gtk installers as many of us will use GStreamer in > combination with > >> GTK. If you don't do this, your gstreamer binaries will > work well on > >> a standalone app, but they won't work at all with any previous > >> installation. > >> I dont know if I was clear enough because my english is > very poor. > >> Want I wanted to let clear is that there could be some > conflicts with > >> GLib that you should take in consideration. I think the > GLib version > >> used to link the GStreamer binaries si too cooler for > windows and > >> won't be compatible with any other previous installation as > the one > >> provided by the windows gtk installer. > > Hi Andoni, your comment is very clear. From what I > understand, all > > that GStreamer uses from GTK on windows are the glib dlls > > (specifically libgio-2.0, libglib-2.0, libgmodule-2.0 and > > libgobject-2.0), so those are the only ones GStreamerOABuild > needs to > > be linked against in case we want to create a GTK-compatible > release > > of OABuild. > I forgot to mention libgthread-2.0, it is also required by > GStreamerOABuild. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move > Developer's challenge > Build the coolest Linux based applications with Moblin SDK & > win great prizes > Grand prize is a trip for two to an Open Source event anywhere > in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From irfanshaikh at tataelxsi.co.in Tue Sep 16 09:43:41 2008 From: irfanshaikh at tataelxsi.co.in (irfanshaikh) Date: Tue, 16 Sep 2008 13:13:41 +0530 Subject: [gst-devel] Writing a transport stream to the disk and reading itat the same time In-Reply-To: Message-ID: <001c01c917cf$f144d920$37033c0a@telxsi.com> Hi, I suggest you to first play the video file using mplayer. So that you come to know about the audio video codec sequence as to know whetehr audo or video comes first.Based on that you can use the respective decoders as per sequence in mplayer.If audio comes first use audio decoder, else video..... Try changing the demux or the decoder elements by trial and error.I also had faced the same problem, some time ago, it worked out when i changed the respective decoder. Hope it works for you too . Regards, Irfan -----Original Message----- From: gstreamer-devel-bounces at lists.sourceforge.net [mailto:gstreamer-devel-bounces at lists.sourceforge.net]On Behalf Of Julien Pauty Sent: Tuesday, September 16, 2008 12:46 AM To: gstreamer-devel at lists.sourceforge.net Subject: [gst-devel] Writing a transport stream to the disk and reading itat the same time Hello, I have an application with two pipelines. The first pipeline writes to the disk the transport streams provided by a dvbbasebin. My second pipeline reads the transport streams from the disk. My problem is that playing stops after a few seconds. I'm waiting a few seconds before starting playing to be sure that enough data has been saved, but this is not sufficient. My pipeling looks like this: filesrc ! flutsdemux ! queue ! audiodec ! audiosink ! queue! videodec ! videosink . If I use the playbin instead of my custom pipeline, it works. However, I still want to use my pipeline, because it can seek through the transport stream, whereas I'm not able to seek with the playbin. With a big file I don't have this problem, suggesting some buffering problem. My debbuging session tends to show that, if the file is small (<10meg) the filesrc reads the whole file and send an EOS event. I thought that I may need some prerolling and I tried to insert a queue between the filesrc and the demuxer, but this does not help. I think it's possible to do it since the playbin can handle this. Unfortunally sources of the playbin are rather complex. Any idea is welcomed, Cheers, Julien The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien.pauty at gmail.com Tue Sep 16 09:56:28 2008 From: julien.pauty at gmail.com (Julien Pauty) Date: Tue, 16 Sep 2008 09:56:28 +0200 Subject: [gst-devel] Writing a transport stream to the disk and reading itat the same time In-Reply-To: <001c01c917cf$f144d920$37033c0a@telxsi.com> References: <001c01c917cf$f144d920$37033c0a@telxsi.com> Message-ID: Hello, Thank you for your answer. 2008/9/16 irfanshaikh > Hi, > > I suggest you to first play the video file using mplayer. So that you come > to know about the audio video codec sequence as to know whetehr audo or > video comes first.Based on that you can use the respective decoders as per > sequence in mplayer.If audio comes first use audio decoder, else > video..... > I tried to read a saved transport stream with mplayer but it fails. Vlc can read the video part but there's no sound. So here, gstreamer is a winner. I'm using the capabilities of the pads created by the demuxer to determine the kind of stream that comes out of it, so I'm pretty sure I'm using the correct decoders. Julien -------------- next part -------------- An HTML attachment was scrubbed... URL: From ylatuya at gmail.com Tue Sep 16 10:57:12 2008 From: ylatuya at gmail.com (Andoni Morales) Date: Tue, 16 Sep 2008 10:57:12 +0200 Subject: [gst-devel] Freshly baked Windows binaries of HEAD available In-Reply-To: <48CED5B1.1050204@gmail.com> References: <48CAB8B0.8090801@gmail.com> <772db3280809130823y74bff3b9u4f45db8d0803c479@mail.gmail.com> <48CC2198.6070000@gmail.com> <48CC259F.1060507@gmail.com> <772db3280809131019p568c3817nad50cdadf25b28ee@mail.gmail.com> <772db3280809140448rdcffe51l42e39fbda39e684@mail.gmail.com> <48CED5B1.1050204@gmail.com> Message-ID: <772db3280809160157k16bf1ad0yc385cd135af810d3@mail.gmail.com> hi, > I applied those patches on the dshowwrapper plugin in OPBuild in order > to generate the installer I posted the other day. That's strange... He recently merged the 2 patches in 1. Did you used this one: http://bugzilla.gnome.org/attachment.cgi?id=114008&action=view I'm for 2 weeks out of home and I can't give any feedback and I've never tested this patch on Windows Vista, but for XP it works fine. You should better talk directly with Julien Isource (julien.isource at gmail.com) who did the patch, as he recently asked the GStreamer Team to commit the patch to the trunk. If so, don't include it for the moment, as for the default sizes the original works. 2008/9/15 Andres Colubri > I applied those patches on the dshowwrapper plugin in OPBuild in order > to generate the installer I posted the other day. > > But there seems to be something wrong. If you try dshowvideosrc from > gst-launch (on windows vista) you get this error: > > > gst-launch-0.10.exe dshowvideosrc ! ffmpegcolorspace ! directdrawsink > Setting pipeline to PAUSED ... > ERROR: Pipeline doesn't want to pause. > Setting pipeline to NULL ... > FREEING pipeline ... > > Andoni Morales wrote: > > Hi, > > > > The last think you should consider on building GStreamer for > > w?Windows is to add the dshowvideosrc patch from Julien Isorce > > (http://bugzilla.gnome.org/show_bug.cgi?id=517203), as without it is > > not possible to use other video size than the default. > > > > > > 2008/9/13 Andoni Morales > > > > > Yes, thats all. > > As the glib version installed on a Windows SO is usually the one > > provided by the Gtk Windows installer, you should try to use this > > one to avoid incompatibilities. > > I've been also working on compiling ffmpeg with really full > > codec support and I've written a post on the Andres Colubri's > > blog with some guidelines to compile ffmpeg with more codecs > > support like faad, faac , mp3lame, x264, vorbis, dts, > > a52, xvid... check it out. > > I think we are very close to get GStreamer working really well on > > Windows!!! > > > > Regards, > > Andoni Morales > > > > 2008/9/13 Andres Colubri > > > > > > > > >> What I mean is: If you want to provide some GStreamer > > binaries, try > > >> to link it against the glib version actually provided by > > the windows > > >> gtk installers as many of us will use GStreamer in > > combination with > > >> GTK. If you don't do this, your gstreamer binaries will > > work well on > > >> a standalone app, but they won't work at all with any previous > > >> installation. > > >> I dont know if I was clear enough because my english is > > very poor. > > >> Want I wanted to let clear is that there could be some > > conflicts with > > >> GLib that you should take in consideration. I think the > > GLib version > > >> used to link the GStreamer binaries si too cooler for > > windows and > > >> won't be compatible with any other previous installation as > > the one > > >> provided by the windows gtk installer. > > > Hi Andoni, your comment is very clear. From what I > > understand, all > > > that GStreamer uses from GTK on windows are the glib dlls > > > (specifically libgio-2.0, libglib-2.0, libgmodule-2.0 and > > > libgobject-2.0), so those are the only ones GStreamerOABuild > > needs to > > > be linked against in case we want to create a GTK-compatible > > release > > > of OABuild. > > I forgot to mention libgthread-2.0, it is also required by > > GStreamerOABuild. > > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move > > Developer's challenge > > Build the coolest Linux based applications with Moblin SDK & > > win great prizes > > Grand prize is a trip for two to an Open Source event anywhere > > in the world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > ------------------------------------------------------------------------ > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From learning_gst at hotmail.com Tue Sep 16 12:02:14 2008 From: learning_gst at hotmail.com (learning gst) Date: Tue, 16 Sep 2008 10:02:14 +0000 Subject: [gst-devel] checking for LIBOIL... configure: error: liboil-0.3.14 or later is required. In-Reply-To: References: Message-ID: Hello, I compiled gst-plugins-base with the command ./configure --prefix="$prefix" --disable-nls --disable-static --with-html-dir=/tmp/dump But I always get the error information: checking for LIBOIL... configure: error: liboil-0.3.14 or later is required. Actually I installed liboil in the :/usr/local/lib. How should I fix it? Thanks! _________________________________________________________________ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Tue Sep 16 12:19:06 2008 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 16 Sep 2008 11:19:06 +0100 Subject: [gst-devel] checking for LIBOIL... configure: error: liboil-0.3.14 or later is required. In-Reply-To: References: Message-ID: <1221560346.32479.38.camel@mini.centricular.net> On Tue, 2008-09-16 at 10:02 +0000, learning gst wrote: Hi, (there's no need to send mails to both gstreamer-devel and gstreamer-embedded, pick whichever you think is the most suitable list and then send mails only to that list, thanks). > I compiled > > gst-plugins-base > with the command ./configure --prefix="$prefix" --disable-nls --disable-static --with-html-dir=/tmp/dump > But I always get the error information: > > checking for LIBOIL... configure: error: liboil-0.3.14 or later is > required. > > Actually I installed liboil in the :/usr/local/lib. > > How should I fix it? export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig:$PKG_CONFIG_PATH export LD_LIBRARY_PATH=/usr/local/lib:$LD_LIBRARY_PATH Then re-run ./configure Cheers -Tim From julien.isorce at gmail.com Tue Sep 16 12:27:45 2008 From: julien.isorce at gmail.com (Julien Isorce) Date: Tue, 16 Sep 2008 12:27:45 +0200 Subject: [gst-devel] Freshly baked Windows binaries of HEAD available In-Reply-To: <772db3280809160157k16bf1ad0yc385cd135af810d3@mail.gmail.com> References: <48CAB8B0.8090801@gmail.com> <772db3280809130823y74bff3b9u4f45db8d0803c479@mail.gmail.com> <48CC2198.6070000@gmail.com> <48CC259F.1060507@gmail.com> <772db3280809131019p568c3817nad50cdadf25b28ee@mail.gmail.com> <772db3280809140448rdcffe51l42e39fbda39e684@mail.gmail.com> <48CED5B1.1050204@gmail.com> <772db3280809160157k16bf1ad0yc385cd135af810d3@mail.gmail.com> Message-ID: <180a127d0809160327h18e7c8c1v5a02c789c8a848c@mail.gmail.com> Hi, I never tested the dshwovideosrc element with or without my patch on Vista. *Can you test the dshowvideosrc element without the patch, on Vista ?* If the pipeline still does not want to pause: - how many video devices do you have on your computer ? you can list them through graphedt.exe in Video Capture Devices section. - If you have several video capture devices, please test each one with dshowvideosrc device-name='..' (you can get the correct name through graphedt.exe) - run the command with -v option, and/or with --gst-debug=dshowvideosrc:5 option. Then send us the log. - if your video device capture supports several ouput formats, force to use each one: dshwovideosrc ! "video/x-raw-yuv, format=(fourcc)I420" ! .. and dshwovideosrc ! "video/x-raw-rgb" ! .. and send us the log. (I have upgraded the patch (but not submited yet) because some special video devices need more "pin capabilities discovering" but I am waitting someone commit the one submited first) (I think this element is not yet mature (not enough devices have been tested and windows platforms) to start code refactoring.) J. > I applied those patches on the dshowwrapper plugin in OPBuild in order >> to generate the installer I posted the other day. >> >> But there seems to be something wrong. If you try dshowvideosrc from >> gst-launch (on windows vista) you get this error: >> >> > gst-launch-0.10.exe dshowvideosrc ! ffmpegcolorspace ! directdrawsink >> Setting pipeline to PAUSED ... >> ERROR: Pipeline doesn't want to pause. >> Setting pipeline to NULL ... >> FREEING pipeline ... >> >> Andoni Morales wrote: >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From irfanshaikh at tataelxsi.co.in Tue Sep 16 12:57:48 2008 From: irfanshaikh at tataelxsi.co.in (irfanshaikh) Date: Tue, 16 Sep 2008 16:27:48 +0530 Subject: [gst-devel] checking for LIBOIL... configure: error: liboil-0.3.14 or later is required. In-Reply-To: Message-ID: <002801c917eb$0f6d8940$37033c0a@telxsi.com> Hii, Export the library path and configure it once again. export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig:/usr/lib/pkgconfig export LD_LIBRARY_PATH=/usr/local/lib Regards, Irfan -----Original Message----- From: gstreamer-devel-bounces at lists.sourceforge.net [mailto:gstreamer-devel-bounces at lists.sourceforge.net]On Behalf Of learning gst Sent: Tuesday, September 16, 2008 3:32 PM To: gstreamer-devel at lists.sourceforge.net; gstreamer-embedded at lists.sourceforge.net Subject: [gst-devel] checking for LIBOIL... configure: error: liboil-0.3.14 or later is required. Hello, I compiled gst-plugins-base with the command ./configure --prefix="$prefix" --disable-nls --disable-static --with-html-di r=/tmp/dumpBut I always get the error information: checking for LIBOIL... configure: error: liboil-0.3.14 or later is required. Actually I installed liboil in the :/usr/local/lib. How should I fix it? Thanks! [irfanshaikh] ---------------------------------------------------------------------------- -- Get news, entertainment and everything you care about at Live.com. Check it out! The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien.isorce at gmail.com Tue Sep 16 17:29:15 2008 From: julien.isorce at gmail.com (Julien Isorce) Date: Tue, 16 Sep 2008 17:29:15 +0200 Subject: [gst-devel] mpegts encrypted Message-ID: <180a127d0809160829k2ab883e1w85e3ec5b3cdc492b@mail.gmail.com> Hi, Does anybody know if it's possible to generate a crypted mpegts flux with gstreamer ? thx (According to the mpeg ts norm the PID can indicate an encrypted stream) Sincerely Julien -------------- next part -------------- An HTML attachment was scrubbed... URL: From rajshyam at gmail.com Tue Sep 16 20:20:36 2008 From: rajshyam at gmail.com (Raj Swaminathan) Date: Tue, 16 Sep 2008 13:20:36 -0500 Subject: [gst-devel] Souphttpsrc and playbin In-Reply-To: <1221213206.18505.5.camel@mini.centricular.net> References: <6438d8660809111344h1cafc389l75569f0bed9103f5@mail.gmail.com> <1221213206.18505.5.camel@mini.centricular.net> Message-ID: <6438d8660809161120n218c8db7sa624866c801c440d@mail.gmail.com> Eric, Tim, Both methods beautifully.... Thanks a lot for your help. regards, raj On Fri, Sep 12, 2008 at 4:53 AM, Tim-Philipp M?ller wrote: > On Thu, 2008-09-11 at 15:44 -0500, Raj Swaminathan wrote: > > Hi, > > > So playbin is trying to use souphttpsrc but Im assuming the error is > > becoz the proxy is not set ..... > > Can someone plz suggest how to set the proxy property with souphttpsrc > > and still use playbin v1 ? > > One possibility already mentioned by Eric is to connect to the > notify::source signal and set up the source in the callback. This is the > recommended way of configuring playbin sources (also for things like > "device" properties and the like). > > In your case you could also just set the http_proxy environment variable > (g_set_env) - if you're lucky souphttpsrc will take that into account > and configure itself accordingly. > > Cheers > -Tim > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From lomesh.agarwal at intel.com Tue Sep 16 20:26:30 2008 From: lomesh.agarwal at intel.com (Agarwal, Lomesh) Date: Tue, 16 Sep 2008 11:26:30 -0700 Subject: [gst-devel] H.264 content streaming Message-ID: I am streaming H.264 content (in MPEG TS) between two Ubuntu machines using following pipelines - Server - gst-launch -v gstrtpbin name=rtpbin filesrc location=h264.ts ! \ queue2 max-size-buffers=65535 ! \ h264parse ! \ rtpmp2tpay ! \ queue2 max-size-buffers=65535 ! \ identity sync=true silent=true sleep-time=300 ! \ rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=192.168.1.2 port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink host=192.168.1.2 port=5001 sync=false async=false \ udpsrc port=5002 ! rtpbin.recv_rtcp_sink_0 client - gst-launch -v gstrtpbin name=rtpbin udpsrc buffer-size=200000 port=5000 caps="application/x-rtp,media=video,clock-rate=90000,encoding-name=mpegts" ! \ rtpbin.recv_rtp_sink_0 rtpbin. ! \ rtpmp2tdepay ! \ flutsdemux name=demuxer \ demuxer. ! queue2 max-size-buffers=0 max-size-time=0 ! fluh264dec ! autovideosink \ udpsrc port=5001 ! \ rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! \ udpsink host=192.168.1.1 port=5002 sync=false async=false -t If I don't use the identity element on server and don't increase the buffer-size on client (text in red) then on client side video is jittery and after some time client pipeline dies. Can someone clarify why does it work with the text in red? Thanks, Lomesh -------------- next part -------------- An HTML attachment was scrubbed... URL: From andres.colubri at gmail.com Tue Sep 16 21:59:31 2008 From: andres.colubri at gmail.com (Andres Colubri) Date: Tue, 16 Sep 2008 12:59:31 -0700 Subject: [gst-devel] Freshly baked Windows binaries of HEAD available In-Reply-To: <180a127d0809160327h18e7c8c1v5a02c789c8a848c@mail.gmail.com> References: <48CAB8B0.8090801@gmail.com> <772db3280809130823y74bff3b9u4f45db8d0803c479@mail.gmail.com> <48CC2198.6070000@gmail.com> <48CC259F.1060507@gmail.com> <772db3280809131019p568c3817nad50cdadf25b28ee@mail.gmail.com> <772db3280809140448rdcffe51l42e39fbda39e684@mail.gmail.com> <48CED5B1.1050204@gmail.com> <772db3280809160157k16bf1ad0yc385cd135af810d3@mail.gmail.com> <180a127d0809160327h18e7c8c1v5a02c789c8a848c@mail.gmail.com> Message-ID: <48D01023.7030509@gmail.com> Hello, When I run the unpatched version of dshowvideosrc I get a different error: >gst-launch-0.10.exe dshowvideosrc ! ffmpegcolorspace ! directdrawsink WARNING: erroneous pipeline: could not link dshowvideosrc0 to ffmpegcsp0 However, I remember using on Vista a version of the updated dll that Julien posted on the list a while ago, and it was working fine. It wasn't with the OABuild version of GStreamer, though, but with the latest "official" version available here: http://gstreamer.freedesktop.org/pkg/windows/releases/ GStreamer OABuild was compiled on MSVC 9 (visual studio 2008) using very recent code from the svn head. Perhaps some incompatibly between dshowvideosrc and a recent change in GStreamer...? Or with MSVC 9? Andres Julien Isorce wrote: > Hi, > > I never tested the dshwovideosrc element with or without my patch on > Vista. > _Can you test the dshowvideosrc element without the patch, on Vista ?_ > > If the pipeline still does not want to pause: > - how many video devices do you have on your computer ? you can list > them through graphedt.exe in Video Capture Devices section. > - If you have several video capture devices, please test each one with > dshowvideosrc device-name='..' > (you can get the correct name through graphedt.exe) > - run the command with -v option, and/or with > --gst-debug=dshowvideosrc:5 option. Then send us the log. > - if your video device capture supports several ouput formats, force > to use each one: > dshwovideosrc ! "video/x-raw-yuv, format=(fourcc)I420" ! .. and > dshwovideosrc ! "video/x-raw-rgb" ! .. > and send us the log. > > (I have upgraded the patch (but not submited yet) because some special > video devices need more "pin capabilities discovering" but I am > waitting someone commit the one submited first) > > (I think this element is not yet mature (not enough devices have been > tested and windows platforms) to start code refactoring.) > > J. > > > I applied those patches on the dshowwrapper plugin in OPBuild > in order > to generate the installer I posted the other day. > > But there seems to be something wrong. If you try > dshowvideosrc from > gst-launch (on windows vista) you get this error: > > > gst-launch-0.10.exe dshowvideosrc ! ffmpegcolorspace ! > directdrawsink > Setting pipeline to PAUSED ... > ERROR: Pipeline doesn't want to pause. > Setting pipeline to NULL ... > FREEING pipeline ... > > Andoni Morales wrote: > > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From andres.colubri at gmail.com Wed Sep 17 01:20:28 2008 From: andres.colubri at gmail.com (Andres Colubri) Date: Tue, 16 Sep 2008 16:20:28 -0700 Subject: [gst-devel] Freshly baked Windows binaries of HEAD available In-Reply-To: <48D01023.7030509@gmail.com> References: <48CAB8B0.8090801@gmail.com> <772db3280809130823y74bff3b9u4f45db8d0803c479@mail.gmail.com> <48CC2198.6070000@gmail.com> <48CC259F.1060507@gmail.com> <772db3280809131019p568c3817nad50cdadf25b28ee@mail.gmail.com> <772db3280809140448rdcffe51l42e39fbda39e684@mail.gmail.com> <48CED5B1.1050204@gmail.com> <772db3280809160157k16bf1ad0yc385cd135af810d3@mail.gmail.com> <180a127d0809160327h18e7c8c1v5a02c789c8a848c@mail.gmail.com> <48D01023.7030509@gmail.com> Message-ID: <48D03F3C.7080208@gmail.com> BTW, I have been able to do video capture using the ksvideosrc element included in the winks plugin of OABuild. Seems to work ok on both windows vista and xp. Andres Colubri wrote: > Hello, > > When I run the unpatched version of dshowvideosrc I get a different > error: > > >gst-launch-0.10.exe dshowvideosrc ! ffmpegcolorspace ! directdrawsink > WARNING: erroneous pipeline: could not link dshowvideosrc0 to ffmpegcsp0 > > However, I remember using on Vista a version of the updated dll that > Julien posted on the list a while ago, and it was working fine. > It wasn't with the OABuild version of GStreamer, though, but with the > latest "official" version available here: > http://gstreamer.freedesktop.org/pkg/windows/releases/ > > GStreamer OABuild was compiled on MSVC 9 (visual studio 2008) using > very recent code from the svn head. Perhaps some incompatibly between > dshowvideosrc and a recent change in GStreamer...? Or with MSVC 9? > > Andres > > Julien Isorce wrote: >> Hi, >> >> I never tested the dshwovideosrc element with or without my patch on >> Vista. >> _Can you test the dshowvideosrc element without the patch, on Vista ?_ >> >> If the pipeline still does not want to pause: >> - how many video devices do you have on your computer ? you can list >> them through graphedt.exe in Video Capture Devices section. >> - If you have several video capture devices, please test each one >> with dshowvideosrc device-name='..' >> (you can get the correct name through graphedt.exe) >> - run the command with -v option, and/or with >> --gst-debug=dshowvideosrc:5 option. Then send us the log. >> - if your video device capture supports several ouput formats, force >> to use each one: >> dshwovideosrc ! "video/x-raw-yuv, format=(fourcc)I420" ! .. and >> dshwovideosrc ! "video/x-raw-rgb" ! .. and send us the log. >> >> (I have upgraded the patch (but not submited yet) because some >> special video devices need more "pin capabilities discovering" but I >> am waitting someone commit the one submited first) >> >> (I think this element is not yet mature (not enough devices have been >> tested and windows platforms) to start code refactoring.) >> >> J. >> >> >> I applied those patches on the dshowwrapper plugin in OPBuild >> in order >> to generate the installer I posted the other day. >> >> But there seems to be something wrong. If you try >> dshowvideosrc from >> gst-launch (on windows vista) you get this error: >> >> > gst-launch-0.10.exe dshowvideosrc ! ffmpegcolorspace ! >> directdrawsink >> Setting pipeline to PAUSED ... >> ERROR: Pipeline doesn't want to pause. >> Setting pipeline to NULL ... >> FREEING pipeline ... >> >> Andoni Morales wrote: >> >> >> >> >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------- >> >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> challenge >> Build the coolest Linux based applications with Moblin SDK & win >> great prizes >> Grand prize is a trip for two to an Open Source event anywhere in the >> world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > From remi.buisson at viotech.net Wed Sep 17 08:54:22 2008 From: remi.buisson at viotech.net (=?ISO-8859-1?Q?R=E9mi_BUISSON?=) Date: Wed, 17 Sep 2008 08:54:22 +0200 Subject: [gst-devel] A big problem !! HELP !!!!! In-Reply-To: <48CA6CC7.3080707@viotech.net> References: <48CA6CC7.3080707@viotech.net> Message-ID: <48D0A99E.1020606@viotech.net> up ! R?mi BUISSON wrote: > Hi everyone, > > I have to generate an XML code containing available plugins of > gstreamer but I have a problem in listing gstreamer modules. > > I attach an archive with a bug and I don't know how to fix it ... > maybe a gstreamer bug ? > It looks like a pointer kind bug. > > With valgrind there is a lot of errors. > > Please compile it with: make depends && make > > execute : build/client > > Try to comment "xmlDocPtr doc;" in xml_parser.c" and re-test ... > > On my side I get 265 modules with the first step and 9 in the second ... > > Any idea ? > > R?mi From arnabsamanta at tataelxsi.co.in Wed Sep 17 10:55:54 2008 From: arnabsamanta at tataelxsi.co.in (arnabsamanta) Date: Wed, 17 Sep 2008 14:25:54 +0530 Subject: [gst-devel] filesrc , v4lsrc element In-Reply-To: Message-ID: <008901c918a3$322ebef0$26033c0a@telxsi.com> Hi , can any body tel how the output of the filesrc and the v4lsrc element is organised ? is it frame by frame or byte by byte ? or any other as such ? ~Arnab The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From t.i.m at zen.co.uk Wed Sep 17 13:16:03 2008 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 17 Sep 2008 12:16:03 +0100 Subject: [gst-devel] A big problem !! HELP !!!!! In-Reply-To: <48D0A99E.1020606@viotech.net> References: <48CA6CC7.3080707@viotech.net> <48D0A99E.1020606@viotech.net> Message-ID: <1221650164.26033.6.camel@mini.centricular.net> On Wed, 2008-09-17 at 08:54 +0200, R?mi BUISSON wrote: > up ! You may find http://www.catb.org/~esr/faqs/smart-questions.html a useful read. > R?mi BUISSON wrote: > > > > I have to generate an XML code containing available plugins of > > gstreamer You mean something like gst-xmlinspect-0.10 ? > > (...) but I have a problem in listing gstreamer modules. > > > > I attach an archive with a bug and I don't know how to fix it ... > > maybe a gstreamer bug ? > > It looks like a pointer kind bug. > > > > With valgrind there is a lot of errors. > > > > Please compile it with: make depends && make > > > > execute : build/client > > > > Try to comment "xmlDocPtr doc;" in xml_parser.c" and re-test ... > > > > On my side I get 265 modules with the first step and 9 in the second ... > > > > Any idea ? Chances that anyone will actually look at your code or even try it are much much higher if your code is one single uncompressed self-contained source file that doesn't need modifying before being run. Cheers -Tim From remi.buisson at viotech.net Wed Sep 17 13:45:33 2008 From: remi.buisson at viotech.net (=?UTF-8?B?UsOpbWkgQlVJU1NPTg==?=) Date: Wed, 17 Sep 2008 13:45:33 +0200 Subject: [gst-devel] A big problem !! HELP !!!!! In-Reply-To: <1221650164.26033.6.camel@mini.centricular.net> References: <48CA6CC7.3080707@viotech.net> <48D0A99E.1020606@viotech.net> <1221650164.26033.6.camel@mini.centricular.net> Message-ID: <48D0EDDD.7000108@viotech.net> Tim-Philipp M?ller wrote: > On Wed, 2008-09-17 at 08:54 +0200, R?mi BUISSON wrote: > > >> up ! >> > > You may find http://www.catb.org/~esr/faqs/smart-questions.html a useful read. > Indeed : stupid subject ;-) Sorry I'm not a very regular customer of mailing lists ... > > >> R?mi BUISSON wrote: >> >>> I have to generate an XML code containing available plugins of >>> gstreamer >>> > > You mean something like gst-xmlinspect-0.10 ? > Not so complete. I would like something like that : metadatademux multipartdemux mpegtsparse vorbisparse ... > >>> (...) but I have a problem in listing gstreamer modules. >>> >>> I attach an archive with a bug and I don't know how to fix it ... >>> maybe a gstreamer bug ? >>> It looks like a pointer kind bug. >>> >>> With valgrind there is a lot of errors. >>> >>> Please compile it with: make depends && make >>> >>> execute : build/client >>> >>> Try to comment "xmlDocPtr doc;" in xml_parser.c" and re-test ... >>> >>> On my side I get 265 modules with the first step and 9 in the second ... >>> >>> Any idea ? >>> > > Chances that anyone will actually look at your code or even try it are > much much higher if your code is one single uncompressed self-contained > source file that doesn't need modifying before being run. > More practical for you indeed ... :-S Actually I didn't want to modify the code I extracted from playbin source but I finally found a more convenient filter for my purpose : gboolean element_filter(GstPluginFeature *feature, FilterData *data) { const gchar *klass = NULL; /* we only care about element factories */ if(!GST_IS_ELEMENT_FACTORY(feature)) return FALSE; klass = gst_element_factory_get_klass(GST_ELEMENT_FACTORY(feature)); if(g_strrstr(klass, "Demux") == NULL && g_strrstr(klass, "Decoder") == NULL && g_strrstr(klass, "Depayloader") == NULL && g_strrstr(klass, "Parse") == NULL) return FALSE; return TRUE; } ... /* get the list of the modules */ modulesList = gst_default_registry_feature_filter((GstPluginFeatureFilter) element_filter, FALSE, &data); printDebug("%d modules available\n", (int) g_list_length(modulesList)); for(walk = modulesList; walk != NULL; walk = g_list_next(walk)) { mod = xmlNewNode(NULL, BAD_CAST NODE_MODULE); text = xmlNewDocText(doc, BAD_CAST (char *)gst_plugin_feature_get_name(walk->data)); tmp = xmlAddChildList(mod, text); tmp = xmlAddChildList(modLst, mod); } gst_plugin_feature_list_free(modulesList); ... So, problem resolved. Thanks for your reply and your precious link :-) > Cheers > -Tim > > Cheers, R?mi > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From thaytan at noraisin.net Wed Sep 17 15:31:26 2008 From: thaytan at noraisin.net (Jan Schmidt) Date: Wed, 17 Sep 2008 14:31:26 +0100 Subject: [gst-devel] New pre-releases of Core/Base uploaded - 0.10.20.3. In-Reply-To: <1221071491.1673.29.camel@fancy-ubuntu> References: <1220949484.1673.5.camel@fancy-ubuntu> <1221071491.1673.29.camel@fancy-ubuntu> Message-ID: <1221658286.6090.20.camel@fancy-ubuntu> Hi all, I've just uploaded new pre-release tarballs of Core & Base, which are now at 0.10.20.3. gst-python is still at 0.10.12.2 The tarballs are at: http://gstreamer.freedesktop.org/data/src/gstreamer/pre/gstreamer-0.10.20.3.tar.bz2 http://gstreamer.freedesktop.org/data/src/gst-plugins-base/pre/gst-plugins-base-0.10.20.3.tar.bz2 and http://gstreamer.freedesktop.org/data/src/gst-python/pre/gst-python-0.10.12.2.tar.bz2 The release is scheduled for next Monday, 22nd September, however there are 2 unresolved blocker bugs in Core relating to BaseTransform change that may delay it. Please test, and file bugs in http://bugzilla.gnome.org Cheers, Jan. On Wed, 2008-09-10 at 19:31 +0100, Jan Schmidt wrote: > Pre-releases of GStreamer Core 0.10.20.2, Base 0.10.20.2 and Python > bindings 0.10.12.2 are now available: > > http://gstreamer.freedesktop.org/data/src/gstreamer/pre/gstreamer-0.10.20.2.tar.bz2 > > http://gstreamer.freedesktop.org/data/src/gst-plugins-base/pre/gst-plugins-base-0.10.20.2.tar.bz2 > and > http://gstreamer.freedesktop.org/data/src/gst-python/pre/gst-python-0.10.12.2.tar.bz2 > > Please test them out, and file bugs in http://bugzilla.gnome.org/ > > New pre-releases Friday or Saturday as needed. > > Cheers, > Jan. > > On Tue, 2008-09-09 at 09:38 +0100, Jan Schmidt wrote: > > Freezing Core/Base/Python to make 0.10.20.2, 0.10.20.2 and 0.10.12.2 > > respectively. > > > > For details, see the release schedule: > > http://gstreamer.freedesktop.org/wiki/ReleasePlanning2008 > > > > Cheers, > > Jan. -- Jan Schmidt From ved.kpl at gmail.com Wed Sep 17 17:13:31 2008 From: ved.kpl at gmail.com (ved kpl) Date: Wed, 17 Sep 2008 20:43:31 +0530 Subject: [gst-devel] filesrc , v4lsrc element In-Reply-To: <008901c918a3$322ebef0$26033c0a@telxsi.com> References: <008901c918a3$322ebef0$26033c0a@telxsi.com> Message-ID: <7496c23f0809170813n556b3f93jbaa2c86eb2587356@mail.gmail.com> Hi, Filesrc gives 4Kbytes (blocksize property) of data. It is unaware of the data format it is reading so "frame by frame" concept is totally invalid here. V4l2src gives a complete raw video frame per GstBuffer.(1 frame per output). Ved On Wed, Sep 17, 2008 at 2:25 PM, arnabsamanta wrote: > Hi , > can any body tel how the output of the filesrc and the v4lsrc element is > organised ? > is it frame by frame or byte by byte ? or any other as such ? > > ~Arnab > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From Martin.Lehmann at telekom.de Wed Sep 17 18:16:11 2008 From: Martin.Lehmann at telekom.de (Martin.Lehmann at telekom.de) Date: Wed, 17 Sep 2008 18:16:11 +0200 Subject: [gst-devel] Problems with recording single impulses in wav In-Reply-To: <1221658286.6090.20.camel@fancy-ubuntu> References: <1220949484.1673.5.camel@fancy-ubuntu><1221071491.1673.29.camel@fancy-ubuntu> <1221658286.6090.20.camel@fancy-ubuntu> Message-ID: <547A7B079BC8EF4685873738492ADA430921F169@S4DE9JSAAMU.ost.t-com.de> Hello, i want to measure my Systems-latency, so created an impulse.wav-file with one single impulse (1.0) at the first sample. Then I wrote this commandline: gst-launch ! filesrc location=impulse.wav ! decodebin ! audioconvert ! alsasink alsasrc ! audioconvert ! wavenc ! filesink location=test.wav but when I look in my test.wav to count the samples, the signal took, only every 10th time I see the impulse. Is this a problem with the wav-format, that it is not really made for recording impulses? May there be another format, what is better for? Or is it a problem with buffers and blocks? In the alsasrc there is a problem changing the values for buffer-time and latency. Hope anybody has an idea. From amoebae at gmail.com Wed Sep 17 22:20:26 2008 From: amoebae at gmail.com (David Banks) Date: Wed, 17 Sep 2008 21:20:26 +0100 Subject: [gst-devel] Seek works strangely with filesink Message-ID: <5e058eb50809171320x42c6bf19v732c0cbacc846da@mail.gmail.com> Hi, I'm just getting started with GStreamer. I want to write a program to load an Ogg file, play it, and seek to a random point within the file every two seconds. In addition, I want to be able to save the output of the program to a file. I initially wrote the program using the pipeline: filesrc | oggdemux | vorbisdec | audioconvert | autoaudiosink Basically according to the example given in the application development manual. I added the function 'cb_timeout0' to detect the length and do the seek. That version worked perfectly and produced exactly what I expected. I then tried to make it write its output to a file. I initially tried to simply replace 'autoaudiosink' with 'filesink', but couldn't figure out the format of the raw samples in the output file. Since I wanted WAV output anyway, I tried using wavenc. After a little trial and error, I found the pipeline below: filesrc | oggdemux | vorbisdec | audioconvert | wavenc | filesink I converted it to the program below. This program displays an odd behaviour, however. Its output is always the size of the fully decoded Ogg file, no matter where you interrupt it. That is, even if you run it for only a few seconds (when it . In addition, the seeks in the output are erratic. For example, if you run the program for three minutes, the output file might contain only three or four seeks. They seem to occur at random points in the output. Why is the filesink output so different from the autoaudiosink output? Is it possible the seeks are happening in the *output* rather than the input file? I tried replacing the first parameter to gst_element_seek() with a pointer to the demuxer rather than the entire pipeline, but still the same behaviour. // gcc -o demo -Wall -g $(pkg-config --libs --cflags $(libs)) demo.c #include #include #include static gboolean cb_bus(GstBus *bus, GstMessage *msg, gpointer data); static void cb_pad_added(GstElement *element, GstPad *pad, gpointer data); gboolean cb_timeout0(gpointer data); gint32 pos = 0; int main(int argc, char **argv) { GMainLoop *loop; GstElement *pipeline, *source, *demuxer, *decoder, *conv1, *conv2, *sink; GstBus *bus; gst_init(&argc, &argv); loop = g_main_loop_new(NULL, FALSE); if (argc != 2) { g_printerr("Usage: %s \n", argv[0]); return 1; } pipeline = gst_pipeline_new("audio-player"); source = gst_element_factory_make("filesrc", "file-source"); demuxer = gst_element_factory_make("oggdemux", "ogg-demuxer"); decoder = gst_element_factory_make("vorbisdec", "vorbis-decoder"); conv1 = gst_element_factory_make("audioconvert", NULL); conv2 = gst_element_factory_make("wavenc", NULL); sink = gst_element_factory_make("filesink", "audio-output"); if (!pipeline || !source || !demuxer || !decoder || !conv1 || !conv2 || !sink) { g_printerr("pipeline failed to create properly, exiting\n"); return 1; } g_object_set(G_OBJECT(source), "location", argv[1], NULL); g_object_set(G_OBJECT(sink), "location", "/home/amoe/foo.wav", NULL); bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline)); gst_bus_add_watch(bus, cb_bus, loop); gst_object_unref(bus); gst_bin_add_many( GST_BIN(pipeline), source, demuxer, decoder, conv1, conv2, sink, NULL ); gst_element_link(source, demuxer); gst_element_link_many(decoder, conv1, conv2, sink, NULL); g_signal_connect( demuxer, "pad-added", G_CALLBACK(cb_pad_added), decoder ); g_print("Now playing: %s\n", argv[1]); gst_element_set_state(pipeline, GST_STATE_PLAYING); // seek g_timeout_add(2 * 1000, cb_timeout0, pipeline); g_print("running...\n"); g_main_loop_run(loop); g_print("returned, stopping playback\n"); gst_element_set_state(pipeline, GST_STATE_NULL); g_print("deleting pipeline\n"); gst_object_unref(GST_OBJECT(pipeline)); return 0; } gboolean cb_timeout0(gpointer data) { GstElement *pipeline = (GstElement *) data; GstFormat fmt = GST_FORMAT_TIME; gboolean test; gint64 len; guint64 len_seconds; test = gst_element_seek( pipeline, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET, pos * GST_SECOND, GST_SEEK_TYPE_NONE, -1 ); printf("seek: %d\n", test); test = gst_element_query_duration( pipeline, &fmt, &len ); len_seconds = len / GST_SECOND; printf("total time: %lld (%llds)\n", len, len_seconds); pos = g_random_int_range(0, len_seconds); printf("random: %d\n", pos); puts("timeout called"); return TRUE; } static gboolean cb_bus(GstBus *bus, GstMessage *msg, gpointer data) { return TRUE; } static void cb_pad_added(GstElement *element, GstPad *pad, gpointer data) { GstPad *sinkpad; GstElement *decoder = (GstElement *) data; puts("pad added callback"); g_print("dynamic pad created, linking demuxer/decoder\n"); sinkpad = gst_element_get_static_pad(decoder, "sink"); gst_pad_link(pad, sinkpad); gst_object_unref(sinkpad); } Thanks, -- David Banks From gstelzz at yahoo.fr Wed Sep 17 23:17:53 2008 From: gstelzz at yahoo.fr (Aurelien Grimaud) Date: Wed, 17 Sep 2008 23:17:53 +0200 Subject: [gst-devel] Seek works strangely with filesink In-Reply-To: <5e058eb50809171320x42c6bf19v732c0cbacc846da@mail.gmail.com> References: <5e058eb50809171320x42c6bf19v732c0cbacc846da@mail.gmail.com> Message-ID: <48D17401.7020700@yahoo.fr> Hi, audiosink is sync on the clock, not filesink. So your pipeline with filesink runs as fast as possible, while the audiosink one runs at real time. Your filesink pipelines misses a sync element. Try add identity with sync=TRUE in it to force clock sync. gst-launch -v filesrc location=/home/musique/Heidsieck-Bernard.mp3 \ ! decodebin ! identity sync=TRUE ! wavenc ! filesink location=/tmp/xx.wav Note: I was first expecting sync option on filesink to do the trick, but it fails : gst-launch -v filesrc location=/home/musique/Heidsieck-Bernard.mp3 \ ! decodebin ! wavenc ! identity ! filesink location=/tmp/xx.wav sync=TRUE So I tried gst-launch -v filesrc location=/home/musique/Heidsieck-Bernard.mp3 \ ! decodebin ! wavenc ! identity sync=TRUE ! filesink location=/tmp/xx.wav but it runs too fast too. Aurelien David Banks a ?crit : > Hi, I'm just getting started with GStreamer. I want to write a program to load > an Ogg file, play it, and seek to a random point within the file every two > seconds. In addition, I want to be able to save the output of the program to a > file. > > I initially wrote the program using the pipeline: > > filesrc | oggdemux | vorbisdec | audioconvert | autoaudiosink > > Basically according to the example given in the application development manual. > I added the function 'cb_timeout0' to detect the length and do the seek. That > version worked perfectly and produced exactly what I expected. > > I then tried to make it write its output to a file. I initially tried to simply > replace 'autoaudiosink' with 'filesink', but couldn't figure out the format of > the raw samples in the output file. Since I wanted WAV output anyway, I tried > using wavenc. After a little trial and error, I found the pipeline below: > > filesrc | oggdemux | vorbisdec | audioconvert | wavenc | filesink > > I converted it to the program below. This program displays an odd behaviour, > however. Its output is always the size of the fully decoded Ogg file, no matter > where you interrupt it. That is, even if you run it for only a few seconds > (when it . In addition, the seeks in the output are erratic. For > example, if you run the program for three minutes, the output file might contain > only three or four seeks. They seem to occur at random points in the output. > > Why is the filesink output so different from the autoaudiosink output? Is it > possible the seeks are happening in the *output* rather than the input file? I > tried replacing the first parameter to gst_element_seek() with a pointer to the > demuxer rather than the entire pipeline, but still the same behaviour. > > // gcc -o demo -Wall -g $(pkg-config --libs --cflags $(libs)) demo.c > > #include > > #include > #include > > static gboolean cb_bus(GstBus *bus, GstMessage *msg, gpointer data); > static void cb_pad_added(GstElement *element, GstPad *pad, gpointer data); > gboolean cb_timeout0(gpointer data); > > gint32 pos = 0; > > int main(int argc, char **argv) { > GMainLoop *loop; > GstElement *pipeline, *source, *demuxer, *decoder, *conv1, *conv2, *sink; > GstBus *bus; > > gst_init(&argc, &argv); > loop = g_main_loop_new(NULL, FALSE); > > if (argc != 2) { > g_printerr("Usage: %s \n", argv[0]); > return 1; > } > > pipeline = gst_pipeline_new("audio-player"); > source = gst_element_factory_make("filesrc", "file-source"); > demuxer = gst_element_factory_make("oggdemux", "ogg-demuxer"); > decoder = gst_element_factory_make("vorbisdec", "vorbis-decoder"); > conv1 = gst_element_factory_make("audioconvert", NULL); > conv2 = gst_element_factory_make("wavenc", NULL); > sink = gst_element_factory_make("filesink", "audio-output"); > > if (!pipeline || !source || !demuxer || !decoder || !conv1 || > !conv2 || !sink) { > g_printerr("pipeline failed to create properly, exiting\n"); > return 1; > } > > g_object_set(G_OBJECT(source), "location", argv[1], NULL); > g_object_set(G_OBJECT(sink), "location", "/home/amoe/foo.wav", NULL); > > bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline)); > gst_bus_add_watch(bus, cb_bus, loop); > gst_object_unref(bus); > > gst_bin_add_many( > GST_BIN(pipeline), > source, demuxer, decoder, conv1, conv2, sink, NULL > ); > > gst_element_link(source, demuxer); > gst_element_link_many(decoder, conv1, conv2, sink, NULL); > > > g_signal_connect( > demuxer, "pad-added", G_CALLBACK(cb_pad_added), decoder > ); > > g_print("Now playing: %s\n", argv[1]); > gst_element_set_state(pipeline, GST_STATE_PLAYING); > > // seek > g_timeout_add(2 * 1000, cb_timeout0, pipeline); > > g_print("running...\n"); > g_main_loop_run(loop); > > g_print("returned, stopping playback\n"); > gst_element_set_state(pipeline, GST_STATE_NULL); > > g_print("deleting pipeline\n"); > gst_object_unref(GST_OBJECT(pipeline)); > > return 0; > } > > gboolean cb_timeout0(gpointer data) { > GstElement *pipeline = (GstElement *) data; > GstFormat fmt = GST_FORMAT_TIME; > gboolean test; > gint64 len; > guint64 len_seconds; > > test = gst_element_seek( > pipeline, > 1.0, > GST_FORMAT_TIME, > GST_SEEK_FLAG_FLUSH, > GST_SEEK_TYPE_SET, > pos * GST_SECOND, > GST_SEEK_TYPE_NONE, > -1 > ); > printf("seek: %d\n", test); > > test = gst_element_query_duration( > pipeline, &fmt, &len > ); > > len_seconds = len / GST_SECOND; > printf("total time: %lld (%llds)\n", len, len_seconds); > pos = g_random_int_range(0, len_seconds); > printf("random: %d\n", pos); > > puts("timeout called"); > > return TRUE; > } > > static gboolean cb_bus(GstBus *bus, GstMessage *msg, gpointer data) { > > return TRUE; > } > > static void cb_pad_added(GstElement *element, GstPad *pad, gpointer data) { > > GstPad *sinkpad; > GstElement *decoder = (GstElement *) data; > > puts("pad added callback"); > g_print("dynamic pad created, linking demuxer/decoder\n"); > > sinkpad = gst_element_get_static_pad(decoder, "sink"); > gst_pad_link(pad, sinkpad); > > gst_object_unref(sinkpad); > > } > > Thanks, > From amoebae at gmail.com Wed Sep 17 23:58:31 2008 From: amoebae at gmail.com (David Banks) Date: Wed, 17 Sep 2008 22:58:31 +0100 Subject: [gst-devel] Seek works strangely with filesink In-Reply-To: <48D17401.7020700@yahoo.fr> References: <5e058eb50809171320x42c6bf19v732c0cbacc846da@mail.gmail.com> <48D17401.7020700@yahoo.fr> Message-ID: <5e058eb50809171458h77fad6bdh5d68093ac3d9d818@mail.gmail.com> 2008/9/17 Aurelien Grimaud : > Your filesink pipelines misses a sync element. > Try add identity with sync=TRUE in it to force clock sync. Thanks! This fixed it. -- David Banks From remi.buisson at viotech.net Thu Sep 18 08:49:31 2008 From: remi.buisson at viotech.net (=?ISO-8859-1?Q?R=E9mi_BUISSON?=) Date: Thu, 18 Sep 2008 08:49:31 +0200 Subject: [gst-devel] using SDP files with playbin Message-ID: <48D1F9FB.5080501@viotech.net> Hi everyone, I have some troubles using SDP files with playbin. Both client (192.168.1.30) and server (192.168.1.2) are under Ubuntu 8.04 LTS and gstreamer-0.10. On the server side : $ gst-launch -v gstrtpbin name=rtpbin \ filesrc location=filesrc location=../../../partage/Videos/superman_originale.avi ! decodebin name=dec \ dec. ! queue ! x264enc byte-stream=false bitrate=300 ! rtph264pay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.1.30 ts-offset=0 name=vrtpsink \ rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=192.168.1.30 sync=false async=false name=vrtcpsink \ udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \ dec. ! queue ! audioresample ! audioconvert ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink port=5002 host=192.168.1.30 ts-offset=0 name=artpsink \ rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=192.168.1.30 sync=false async=false name=artcpsink \ udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1 On the client side: $ gst-launch -vvv playbin uri=file:///media/disk/client.sdp Setting pipeline to PAUSED ... /playbin0/decodebin0/typefind.src: caps = application/sdp Pipeline is live and does not need PREROLL ... /playbin0/decodebin0/sdpdemux0.sink: caps = application/sdp Setting pipeline to PLAYING ... /playbin0/decodebin0/sdpdemux0/rtpbin0: latency = 200 /playbin0/decodebin0/sdpdemux0/udpsrc0: timeout = 10000000 /playbin0/decodebin0/sdpdemux0/udpsrc2: timeout = 10000000 /playbin0/decodebin0/sdpdemux0/rtpbin0/rtpsession1: ntp-ns-base = 3430706685285951000 /playbin0/decodebin0/sdpdemux0/rtpbin0/rtpsession0: ntp-ns-base = 3430706685285951000 ERROR: pipeline doesn't want to play. ERROR: from element /playbin0/decodebin0/sdpdemux0/udpsrc0: Could not get/set settings from/on resource. Additional debug info: gstudpsrc.c(843): gst_udpsrc_start (): /playbin0/decodebin0/sdpdemux0/udpsrc0: bind failed -1: Cannot assign requested address (99) Setting pipeline to NULL ... /playbin0/decodebin0/sdpdemux0.sink: caps = NULL /playbin0/decodebin0/typefind.src: caps = NULL FREEING pipeline ... with client.sdp: v=0 o=- 1188340656180883 1 IN IP4 192.168.1.2 s=Session streamed by GStreamer i=server.sh t=0 0 a=tool:GStreamer a=type:broadcast m=video 5000 RTP/AVP 96 c=IN IP4 192.168.1.2 a=rtpmap:96 H264/90000 m=audio 5002 RTP/AVP 8 c=IN IP4 192.168.1.2 It works with localhost address for gstreamer and using vlc 0.8.6. This works with gstreamer: $ gst-launch -v gstrtpbin name=rtpbin latency=200 \ udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" port=5000 ! rtpbin.recv_rtp_sink_0 \ rtpbin. ! rtph264depay ! decodebin ! xvimagesink \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=192.168.1.2 sync=false async=false \ udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" port=5002 ! rtpbin.recv_rtp_sink_1 \ rtpbin. ! rtppcmadepay ! decodebin ! audioconvert ! audioresample ! alsasink \ udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=192.168.1.2 sync=false async=false Thanks in advance. From kumarkm at tataelxsi.co.in Thu Sep 18 09:55:19 2008 From: kumarkm at tataelxsi.co.in (Kumar) Date: Thu, 18 Sep 2008 13:25:19 +0530 Subject: [gst-devel] ffmux_flv issues Message-ID: <002301c91963$e5a9ffb0$60033c0a@telxsi.com> Hi all, 1) Will ffmux_flv supports H264 stream? 2)I am trying to mux H264 ES to FLV format using the pipeline: gst-launch filesrc location=/home/Gstreamer/GStreamer/CodecH264.dat ! ffmux_flv ! filesink location=/home/Gstreamer/GStreamer/mux2.flv I am getting the following error: Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /pipeline0/ffmux_flv0: Internal GStreamer error: negotiation problem. Please file a bug at http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer. Additional debug info: gstffmpegmux.c(377): gst_ffmpegmux_collected (): /pipeline0/ffmux_flv0: no caps set on stream 0 (audio) ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... FREEING pipeline ... Pls help me out with this issue. Thanks & Regards, Kumar KM The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From arnabsamanta at tataelxsi.co.in Thu Sep 18 10:09:52 2008 From: arnabsamanta at tataelxsi.co.in (arnabsamanta) Date: Thu, 18 Sep 2008 13:39:52 +0530 Subject: [gst-devel] C code to gstreamer plugin conversion In-Reply-To: Message-ID: <006001c91965$ee171050$26033c0a@telxsi.com> Hi i have an application which has a number of C++ files and i need to make the application as a gstreamer plug in. what are the basic steps to do so ? i know how to create a source and a sink pad. do i need to declare all functions of all the C++ codes within the init functions ? regards, ~Arnab The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From bob at tataelxsi.co.in Thu Sep 18 10:34:50 2008 From: bob at tataelxsi.co.in (Bob) Date: Thu, 18 Sep 2008 14:04:50 +0530 Subject: [gst-devel] H.264 plugin Message-ID: <0D54F7DB54A34ECC9163EBCC14F5016D@telxsicovai.com> Hi, I want to write a plugin for H.264. Is there an H264 plugin available in gstreamer? I mean, the code. Bob The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien.pauty at gmail.com Thu Sep 18 11:06:33 2008 From: julien.pauty at gmail.com (Julien Pauty) Date: Thu, 18 Sep 2008 11:06:33 +0200 Subject: [gst-devel] Getting eit table from dvbsrc In-Reply-To: References: Message-ID: Ok I'm replying to myself, but this may useful to someone else. So the eit tables (and other DVB related table) can be retrieved via the mpegtsparse element. Simply put this element after the dvbsrc and before the flutsdemux. Another approach is to use the dvbbasebin which contains an mpegtsparse. The mpegtsparse will send message on the but containing the tables. Table are GstStructures and can be easily "parsed" to get electronic program informations. Cheers, Julien 2008/8/7 Julien Pauty > Hello, > > I'm trying to fetch the program timetable from dvb. Eit tables is > transfered on the PID 18, so I'm adding this pid when I setup the > dvbsrc.Problem is that I don't know how to fetch the data. Is it > parsed by the demuxer ? Sent on the bus ? Do I have to parse it > myself? Well any information is welcome. > > Thanks, > > Julien > -------------- next part -------------- An HTML attachment was scrubbed... URL: From chhail at mail2.sysu.edu.cn Thu Sep 18 11:16:13 2008 From: chhail at mail2.sysu.edu.cn (Chen Hailiang) Date: Thu, 18 Sep 2008 17:16:13 +0800 Subject: [gst-devel] Building gstreamer+sdl program Message-ID: <20080918090832.M22120@mail2.sysu.edu.cn> Hi all: I'm using gstreamer and sdl to make a very sample vedio player.I put all the codes in one file; video_test.c. My question is: how can i compile it? I tried some commands like"gcc -Wall `sdl-config --libs gstreamer-0.10` video_test.c -o video_player",it didn't work.Can someone can help? Chen -- Best regards From julien.pauty at gmail.com Thu Sep 18 11:22:10 2008 From: julien.pauty at gmail.com (Julien Pauty) Date: Thu, 18 Sep 2008 11:22:10 +0200 Subject: [gst-devel] Duration and Seek in MPEG TS File In-Reply-To: <1220434981.7548.0.camel@localhost> References: <3afe75670809021351v2ba84d86k1a38074f1cf51373@mail.gmail.com> <15e616860809021444s2305440cn14ad08d962708eb8@mail.gmail.com> <1220434981.7548.0.camel@localhost> Message-ID: Hello, I just want to mention that I'm able to seek and get the duration of an mpeg TS. I'm using a simple pipeline containing: filesrc ! flutsdemux ! queue ! videodec ! videosink ! queue ! audiodec! audiosink . To do the seek I'm calling seek_simple on the filesrc, with a byte position. Maybe this is not the best approach but it works. To convert bytes duration to time i'm calling query_convert on the pipeline itself. To get the duration of the TS I first call querry_duraiton on the filesrc (bytes format) and then I request a convertion to time format to the pipeline. That's the only way I found to do it. There may be an easier way, but in the mean time this is useable. Best, Julien 2008/9/3 Edward Hervey > Hi all, > > I opened a bug on bugzilla regarding that. Comments and patches > welcome. > > http://bugzilla.gnome.org/show_bug.cgi?id=550634 > > Edward > > On Tue, 2008-09-02 at 22:44 +0100, Zaheer Merali wrote: > > On Tue, Sep 2, 2008 at 9:51 PM, Levi Pope wrote: > > > Has anyone been able to seek or get the duration of a TS file with > > > Gstreamer. > > > I have tried using flutsdemux and it does not seem to work. > > > > > > Thanks > > > Levi > > > > This is a missing feature. > > > > Zaheer > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From christophe.dehais at gmail.com Thu Sep 18 11:58:42 2008 From: christophe.dehais at gmail.com (Christophe Dehais) Date: Thu, 18 Sep 2008 11:58:42 +0200 Subject: [gst-devel] Dynamically adding a filter to a video pipeline Message-ID: <4a305d880809180258m7f444e13lf14f45d4018d0fb4@mail.gmail.com> Hi everyone! I have this simple pipeline: videosrc ! identity ! videosink (videosrc is a bin embedding videotestsrc or v4l2src and colorspace, videosink is a bin embedding colorspace and xvimagesink) I simply want to replace identity by a videofilter (e.g. edgetv), while the pipeline is playing. So I made a little test app (see attachments) and I guess I'm following what's described in the design docs here: http://webcvs.freedesktop.org/gstreamer/gstreamer/docs/design/part-block.txt?view=markup except that I'm blocking the src pad synchronously. It doesn't work because when unblocking back the videosrc source pad, a negotiation problem occurs. If I replace 'identity' by 'shagadelictv' (which has the same in and out caps as edgetv), it works fine. So my question is two fold: 1) do I miss something in the design doc ? 2) how can I force the videosrc bin to renegociate with the rest of the pipeline ? thanks for any hints, Christophe -------------- next part -------------- A non-text attachment was scrubbed... Name: dyn_pipeline.py Type: text/x-python Size: 3604 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: dyn_pipeline.xml Type: text/xml Size: 1470 bytes Desc: not available URL: From irfanshaikh at tataelxsi.co.in Thu Sep 18 12:08:04 2008 From: irfanshaikh at tataelxsi.co.in (irfanshaikh) Date: Thu, 18 Sep 2008 15:38:04 +0530 Subject: [gst-devel] Query: ffmux_asf In-Reply-To: Message-ID: <003601c91976$719427a0$37033c0a@telxsi.com> Hi all, I am unable to mux the raw elementry stream (Audio/Video) in an ASF container using ffmux_asf provided by ffmpeg. Can i know how do i mux the audio/video elementry streams using ffmux_asf. I hav tried using ffmux_asf element provided by ffmpeg.I have created following pipelines to investigate ffmux_asf. 1) gst-launch filesrc location=/root/Desktop/teststreams/mpeg4/mpeg4_mp3.ASF ! ffdemux_asf ! ffmux_asf ! filesink location=/root/Desktop/test2.asf ffmpeg (FFmpeg M$ MPEG-4 v2) mp3lib (mp3lib MPEG layer-2, layer-3) Result : test2.asf contains only audio. 2) gst-launch filesrc location=/root/Desktop/audioVideo/mjpegi.AVI ! avidemux ! ffmux_asf ! filesink location=/root/Desktop/test.asf ffmpeg (FFmpeg MJPEG decoder) mp3lib (mp3lib MPEG layer-2, layer-3) Result : test.asf contains only audio. 3) H264 elementry stream gst-launch filesrc location=/root/Desktop/audioVideo/CodecH264.dat ! ffmux_asf ! filesink location=/root/Desktop/ajit1.asf ffmpeg H.264 Result : No output 4)MJPEG elementry stream gst-launch filesrc location=/root/Desktop/audioVideo/CodecH264.dat ! ffmux_asf ! filesink location=/root/Desktop/ajit1.asf Result : No output 5)gst-launch filesrc location=/root/Desktop/teststreams/g.711/alaw_8k_64kbps_mono.wav ! ffmux_asf ! filesink location=/root/Desktop/test6.asf alaw (aLaw) Result : No output 6)gst-launch filesrc location=/root/Desktop/teststreams/h.264/H264_Ray_QVGA__364kbps_HEAAC.asf ! asfdemux ! ffmux_asf ! filesink location=/root/Desktop/h2641.asf ffmpeg (FFmpeg H.264) Result : No output 7)gst-launch filesrc location=/root/Desktop/teststreams/h.264/h264_mp4.MP4 ! ffdemux_mov_mp4_m4a_3gp_3g2_mj2 ! ffmux_asf ! filesink location=/root/Desktop/abc.asf ffmpeg (FFmpeg H.264) Result : No output Regards, Irfan The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. -------------- next part -------------- An HTML attachment was scrubbed... URL: From tero.saarni at gmail.com Thu Sep 18 12:13:03 2008 From: tero.saarni at gmail.com (Tero Saarni) Date: Thu, 18 Sep 2008 13:13:03 +0300 Subject: [gst-devel] DVB pipeline consumes a lot of CPU? Message-ID: Hi, I'm seeing very high CPU usage when running pipeline for DVB-T stream. Setting sync=false like instructed in the example in the docs makes it consume less CPU but with the cost of loosing lipsync. I wonder if this is a well known problem and if there is a fix? Here's the pipeline: gst-launch \ dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=714000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 pids=0:256:512:650 ! \ flutsdemux name=demuxer \ demuxer. ! queue max-size-buffers=0 max-size-time=0 ! mpeg2dec ! ffmpegcolorspace ! xvimagesink sync=false \ demuxer. ! queue max-size-buffers=0 max-size-time=0 ! mad ! alsasink sync=false I'm running gstreamer 0.10.18 on Ubuntu Hardy. -- Tero From ved.kpl at gmail.com Thu Sep 18 13:57:38 2008 From: ved.kpl at gmail.com (ved kpl) Date: Thu, 18 Sep 2008 17:27:38 +0530 Subject: [gst-devel] Query: ffmux_asf In-Reply-To: <003601c91976$719427a0$37033c0a@telxsi.com> References: <003601c91976$719427a0$37033c0a@telxsi.com> Message-ID: <7496c23f0809180457y56d1b33co1fe49c4f75a0efd1@mail.gmail.com> Hi, "ffdemux_asf ! ffmux_asf" is not correct, although your pipeline will work. You need to specify the link. Your output file has only audio data, because the demux gets the audio data first and continues linking the audio pads of muxer & demuxer. Try This. gst-launch filesrc location=/root/Desktop/teststreams/mpeg4/mpeg4_mp3.ASF ! ffdemux_asf name=demux demux.audio_00 ! queue ! mux.audio_0 demux.video_00 ! queue ! mux.video_0 ffmux_asf name=mux ! filesink location=/root/Desktop/test2.asf "gst-launch filesrc location=/root/Desktop/audioVideo/CodecH264.dat ! ffmux_asf" is wrong. You are feeding some unknown(no caps) data (that too not even an elementary stream) to the muxer. Ved On Thu, Sep 18, 2008 at 3:38 PM, irfanshaikh wrote: > > > Hi all, > > I am unable to mux the raw elementry stream (Audio/Video) in an ASF > container using ffmux_asf provided by ffmpeg. > > Can i know how do i mux the audio/video elementry streams using ffmux_asf. > > > > I hav tried using ffmux_asf element provided by ffmpeg. > > I have created following pipelines to investigate > > ffmux_asf. > > > > 1) > > gst-launch filesrc location=/root/Desktop/teststreams/mpeg4/mpeg4_mp3.ASF ! > ffdemux_asf ! ffmux_asf ! filesink location=/root/Desktop/test2.asf > > ffmpeg (FFmpeg M$ MPEG-4 v2) > > mp3lib (mp3lib MPEG layer-2, layer-3) > > Result : test2.asf contains only audio > > . > > > > 2) > > gst-launch filesrc location=/root/Desktop/audioVideo/mjpegi.AVI ! avidemux ! > ffmux_asf ! filesink location=/root/Desktop/test.asf > > ffmpeg (FFmpeg MJPEG decoder) > > mp3lib (mp3lib MPEG layer-2, layer-3) > > Result : test.asf contains only audio. > > > > 3) H264 elementry stream > > gst-launch filesrc location=/root/Desktop/audioVideo/CodecH264.dat ! > ffmux_asf ! filesink location=/root/Desktop/ajit1.asf > > ffmpeg H.264 > > Result : No output > > > > 4)MJPEG elementry stream > > gst-launch filesrc location=/root/Desktop/audioVideo/CodecH264.dat ! > ffmux_asf ! filesink location=/root/Desktop/ajit1.asf > > Result : No output > > > > 5) > > gst-launch filesrc > location=/root/Desktop/teststreams/g.711/alaw_8k_64kbps_mono.wav ! ffmux_asf > ! filesink location=/root/Desktop/test6.asf > > alaw (aLaw) > > Result : No output > > 6) > > gst-launch filesrc > location=/root/Desktop/teststreams/h.264/H264_Ray_QVGA__364kbps_HEAAC.asf ! > asfdemux ! ffmux_asf ! filesink location=/root/Desktop/h2641.asf > > ffmpeg (FFmpeg H.264) > > Result : No output > > 7)gst-launch filesrc location=/root/Desktop/teststreams/h.264/h264_mp4.MP4 ! > ffdemux_mov_mp4_m4a_3gp_3g2_mj2 ! ffmux_asf ! filesink > location=/root/Desktop/abc.asf > > ffmpeg (FFmpeg H.264) > > Result : No output > > > Regards, > Irfan > > The information contained in this electronic message and any attachments to > this message are intended for the exclusive use of the addressee(s) and may > contain proprietary, confidential or privileged information. If you are not > the intended recipient, you should not disseminate, distribute or copy this > e-mail. Please notify the sender immediately and destroy all copies of this > message and any attachments contained in it. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From Martin.Lehmann at telekom.de Thu Sep 18 14:55:18 2008 From: Martin.Lehmann at telekom.de (Martin.Lehmann at telekom.de) Date: Thu, 18 Sep 2008 14:55:18 +0200 Subject: [gst-devel] Problems with recording single impulses in wav In-Reply-To: <547A7B079BC8EF4685873738492ADA430921F169@S4DE9JSAAMU.ost.t-com.de> References: <1220949484.1673.5.camel@fancy-ubuntu><1221071491.1673.29.camel@fancy-ubuntu><1221658286.6090.20.camel@fancy-ubuntu> <547A7B079BC8EF4685873738492ADA430921F169@S4DE9JSAAMU.ost.t-com.de> Message-ID: <547A7B079BC8EF4685873738492ADA430921F16A@S4DE9JSAAMU.ost.t-com.de> Ok I've tried it once more, adding some noise and 4000samples of silence, But still the same Problem, first it works, but when I want validate my Results, the impulse is gone... I really have no idea, can someone please give me a hint? there was an error in the pipeline (one "!" to much, I've also replaced the decodebin with the wavparse). Right one is: gst-launch filesrc location=impulse.wav ! wavparse ! audioconvert ! alsasink alsasrc ! audioconvert ! wavenc ! filesink location=test.wav thanks for answers, Martin -----Original Message----- From: gstreamer-devel-bounces at lists.sourceforge.net [mailto:gstreamer-devel-bounces at lists.sourceforge.net] On Behalf Of Lehmann, Martin Sent: Mittwoch, 17. September 2008 18:16 To: gstreamer-devel at lists.sourceforge.net Subject: [gst-devel] Problems with recording single impulses in wav Hello, i want to measure my Systems-latency, so created an impulse.wav-file with one single impulse (1.0) at the first sample. Then I wrote this commandline: gst-launch ! filesrc location=impulse.wav ! decodebin ! audioconvert ! alsasink alsasrc ! audioconvert ! wavenc ! filesink location=test.wav but when I look in my test.wav to count the samples, the signal took, only every 10th time I see the impulse. Is this a problem with the wav-format, that it is not really made for recording impulses? May there be another format, what is better for? Or is it a problem with buffers and blocks? In the alsasrc there is a problem changing the values for buffer-time and latency. Hope anybody has an idea. ------------------------------------------------------------------------ - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ved.kpl at gmail.com Thu Sep 18 16:10:35 2008 From: ved.kpl at gmail.com (ved kpl) Date: Thu, 18 Sep 2008 19:40:35 +0530 Subject: [gst-devel] Query: ffmux_asf In-Reply-To: <004601c91994$0b194730$37033c0a@telxsi.com> References: <7496c23f0809180457y56d1b33co1fe49c4f75a0efd1@mail.gmail.com> <004601c91994$0b194730$37033c0a@telxsi.com> Message-ID: <7496c23f0809180710t3423e911t886e6095c73b8337@mail.gmail.com> Hi, Please "REPLY TO ALL". Can you upload "mpeg4_mp3.ASF"? Secondly, if you are reading the elementary stream from file directly (filesrc ! mux), I guess it wont work. The filesrc wont attach any caps and no timestamp to the buffer and of course no parsed data. Ved On Thu, Sep 18, 2008 at 7:09 PM, irfanshaikh wrote: > Hi ved, > > Thanks a lot for ur helpful reply.......It was like you giving me hand > when i m sinking. > But brother i tried to use the same pipeline as you suggested, the pipeline > has stucked while prerolling. > > [root at localhost ffmpeg]# gst-launch filesrc > location=/root/Desktop/teststreams/mpeg4/mpeg4_mp3.ASF ! ffdemux_asf > name=demux demux.audio_00 ! queue ! mux.audio_0 demux.video_00 ! queue ! > mux.video_0 ffmux_asf name=mux ! filesink location=/root/Desktop/test2.asf > > (gst-launch-0.10:4308): GStreamer-WARNING **: Failed to load plugin > '/usr/local/lib/gstreamer-0.10/libgstrtppayloads.so': > /usr/local/lib/gstreamer-0.10/libgstrtppayloads.so: undefined symbol: > gst_rtp_g729_pay_plugin_init > Filesrc Base Init > Filesrc Class Init > Filesrc Init > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > > And one more thing i hav tried muxing the elementry stream that also is not > working, there is no output file at all. > > 1) H264 elementry stream > > gst-launch filesrc location=/root/Desktop/audioVideo/CodecH264.dat ! > ffmux_asf ! filesink location=/root/Desktop/ajit1.asf > > ffmpeg H.264 > Result : No output > > > > 2)MJPEG elementry stream > > gst-launch filesrc location=/root/Desktop/audioVideo/CodecH264.dat ! > ffmux_asf ! filesink location=/root/Desktop/ajit1.asf > > Result : No output > > > I would be thankful to you a lot if i am able to find some way to mux the > audio/video elementry stream. Or else if not possible suggest me where i can > luk to modify the code so as i get the proper muxed data. > > Thanks in advance > > Regards, > Irfan. > > > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. > From ved.kpl at gmail.com Thu Sep 18 16:13:23 2008 From: ved.kpl at gmail.com (ved kpl) Date: Thu, 18 Sep 2008 19:43:23 +0530 Subject: [gst-devel] Query: ffmux_asf In-Reply-To: <7496c23f0809180710t3423e911t886e6095c73b8337@mail.gmail.com> References: <7496c23f0809180457y56d1b33co1fe49c4f75a0efd1@mail.gmail.com> <004601c91994$0b194730$37033c0a@telxsi.com> <7496c23f0809180710t3423e911t886e6095c73b8337@mail.gmail.com> Message-ID: <7496c23f0809180713k3f0aad42s1ef1964efe3cb641@mail.gmail.com> Try asfdemux too. On Thu, Sep 18, 2008 at 7:40 PM, ved kpl wrote: > Hi, > > Please "REPLY TO ALL". > > Can you upload "mpeg4_mp3.ASF"? > Secondly, if you are reading the elementary stream from file directly > (filesrc ! mux), > I guess it wont work. The filesrc wont attach any caps and no > timestamp to the buffer and of course no parsed data. > > Ved > > On Thu, Sep 18, 2008 at 7:09 PM, irfanshaikh > wrote: >> Hi ved, >> >> Thanks a lot for ur helpful reply.......It was like you giving me hand >> when i m sinking. >> But brother i tried to use the same pipeline as you suggested, the pipeline >> has stucked while prerolling. >> >> [root at localhost ffmpeg]# gst-launch filesrc >> location=/root/Desktop/teststreams/mpeg4/mpeg4_mp3.ASF ! ffdemux_asf >> name=demux demux.audio_00 ! queue ! mux.audio_0 demux.video_00 ! queue ! >> mux.video_0 ffmux_asf name=mux ! filesink location=/root/Desktop/test2.asf >> >> (gst-launch-0.10:4308): GStreamer-WARNING **: Failed to load plugin >> '/usr/local/lib/gstreamer-0.10/libgstrtppayloads.so': >> /usr/local/lib/gstreamer-0.10/libgstrtppayloads.so: undefined symbol: >> gst_rtp_g729_pay_plugin_init >> Filesrc Base Init >> Filesrc Class Init >> Filesrc Init >> Setting pipeline to PAUSED ... >> Pipeline is PREROLLING ... >> >> And one more thing i hav tried muxing the elementry stream that also is not >> working, there is no output file at all. >> >> 1) H264 elementry stream >> >> gst-launch filesrc location=/root/Desktop/audioVideo/CodecH264.dat ! >> ffmux_asf ! filesink location=/root/Desktop/ajit1.asf >> >> ffmpeg H.264 >> Result : No output >> >> >> >> 2)MJPEG elementry stream >> >> gst-launch filesrc location=/root/Desktop/audioVideo/CodecH264.dat ! >> ffmux_asf ! filesink location=/root/Desktop/ajit1.asf >> >> Result : No output >> >> >> I would be thankful to you a lot if i am able to find some way to mux the >> audio/video elementry stream. Or else if not possible suggest me where i can >> luk to modify the code so as i get the proper muxed data. >> >> Thanks in advance >> >> Regards, >> Irfan. >> >> >> >> >> The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. >> > From bilboed at gmail.com Thu Sep 18 17:24:27 2008 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 18 Sep 2008 17:24:27 +0200 Subject: [gst-devel] H.264 plugin In-Reply-To: <0D54F7DB54A34ECC9163EBCC14F5016D@telxsicovai.com> References: <0D54F7DB54A34ECC9163EBCC14F5016D@telxsicovai.com> Message-ID: <1221751467.13024.6.camel@putamadre> Hi, Did you search for one on the gstreamer webpage ? Or by doing "gst-inspect-0.10 | grep 264" ? Edward On Thu, 2008-09-18 at 14:04 +0530, Bob wrote: > Hi, > > I want to write a plugin for H.264. Is there an H264 plugin available > in gstreamer? I mean, the code. > > Bob > The information contained in this electronic message and any > attachments to this message are intended for the exclusive use of the > addressee(s) and may contain proprietary, confidential or privileged > information. If you are not the intended recipient, you should not > disseminate, distribute or copy this e-mail. Please notify the sender > immediately and destroy all copies of this message and any attachments > contained in it. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ryankelln at gmail.com Thu Sep 11 19:20:15 2008 From: ryankelln at gmail.com (Ryan Kelln) Date: Thu, 11 Sep 2008 13:20:15 -0400 Subject: [gst-devel] Possible gstreamer errors: volume, dispose, loop detected - oh my! Message-ID: <48C9534F.1060800@gmail.com> Hi Gstreamer devs, I'm using gstreamer through gstreamer-java inside of processing. I'm getting some odd errors and warnings and eventually a crash (perhaps a null pointer). I'm hoping you might have a bit of insight or guidance for how to avoid these problems or what I could do to fix them. Thanks. Environment: Ubuntu 8.04 gstreamer-team ppa (https://edge.launchpad.net/~gstreamer-team/+archive) Sun Java 1.06_07 Processing 148 gstreamer-java 0.8 The errors happen in a variety of orders, here is an example of each: ** (GSVideo:9501): CRITICAL **: volume_transform_ip: assertion `this->process != NULL' failed (GSVideo:9501): GStreamer-CRITICAL **: Trying to dispose element test, but it is not in the NULL state. You need to explicitly set elements to the NULL state before dropping the final reference, to allow them to clean up. (GSVideo:9501): GStreamer-WARNING **: loop detected in the graph of bin GStreamer Audio Data Extractor: mysoundfile.wav!! Then eventually Java will crash with a SIGSEGV fault in a native thread which I'm assuming is gstreamer. Looks like it is accessing a null pointer. Description of what I'm doing: Custom software that is playing back multiple audio files at once. It loads approx. 11 sound files (all wavs or aif - errors and crash remain the same) and starts and stops the files quite often. It also checks the current volume and changes the volume each update (i.e. constantly adjusting the volume). In addition I'm using the audiopanorama plug-in to do constant panning. Occasionally I will get the "loop detected" warning (see below) on one or two of the sound files, but which sound file it is seems to change and it doesn't happen every run. The program continues to function after receiving this warning although the sound file doesn't play. I get the "volume_transform_ip" error quite often (every 5 minutes?) but the program doesn't immediately crash. I get the "dispose element test" error only on a multi-core processor (AMD Phenom) but not on an equivalent machine with just a single core. Again, this doesn't cause an immediate crash. As far as I know I'm not make any elements named "test", and I'm not disposing of anything until the program ends but I am starting and stopping the audio streams. I've grepped the source for these and found the loop detected error in gstbin.c (gst_bin_sort_iterator_next() )and the "dispose element not NULL" in gstelement.c (gst_element_dispose() ) but I can't locate the "volume_transform_ip" error. I've got to deliver this project really soon so workarounds might be the best option. I'm not sure if the errors are in gstreamer or just my use of it? If you're sure that gstreamer is at fault I'd love some tips on how to provide more info (I'm not sure how I'd debug gstreamer through all these other layers) and write up a bug report. Thanks very much, Ryan From vuntz at gnome.org Tue Sep 16 11:57:09 2008 From: vuntz at gnome.org (Vincent Untz) Date: Tue, 16 Sep 2008 11:57:09 +0200 Subject: [gst-devel] GStreamer tarballs and ftp.gnome.org Message-ID: <20080916095709.GS30241@vuntz.net> Hi, See [1] for past discussion about this :-) The latest gst-plugins-good we have on ftp.gnome.org is 0.10.6 while the latest version really is 0.10.10. What can we do to fix this? Thanks, Vincent [1] http://mail.gnome.org/archives/release-team/2007-June/msg00018.html -- Les gens heureux ne sont pas press?s. From xxopxe at gmail.com Thu Sep 18 18:28:19 2008 From: xxopxe at gmail.com (Jorge) Date: Thu, 18 Sep 2008 13:28:19 -0300 Subject: [gst-devel] Seeking multifilesrc Message-ID: <1221755299.16314.22.camel@matroskin> Hi to all. I'm trying to write a stop-motion creation app, in the vein of http://developer.skolelinux.no/info/studentgrupper/2005-hig-stopmotion/index.php but using python, gtk, and well, gstreamer. It's my first using all three technologies, tough :p What i'm stump on is on how step frame by frame forward and backwards in a video loaded trough multifilesrc. I was trying to seek using gst.FORMAT_BUFFERS as to move one frame at a time, but it doesn't work. When i attempt a query_position on the pipeline, i get a "query failed". My pipeline is multifilesrc ! jpegdec ! cspvideo !videosink. (multifilesrc has a "image/jpeg,framerate=15/1" cap for correct playing) I've tried a queue right after multifilesrc, but no difference. Is there something i have to do for a stream be seekable? Also, the idea is being able to keep adding image frames without having to reprocess everything each time... Any pointer? Jorge From chhail at mail2.sysu.edu.cn Fri Sep 19 02:17:52 2008 From: chhail at mail2.sysu.edu.cn (Chen Hailiang) Date: Fri, 19 Sep 2008 08:17:52 +0800 Subject: [gst-devel] testing video with SDL Message-ID: <20080919001704.M94981@mail2.sysu.edu.cn> Hi all: I'm using gstreamer and SDL to make a very sample vedio player.I put all the codes in one file; video_test.c. My question is: how can i compile it? I tried some commands like"gcc -Wall `sdl-config --libs gstreamer-0.10` video_test.c -o video_player",it didn't work.Can someone can help? Chen -- Best regards From liangzhihong1984 at 126.com Fri Sep 19 05:08:47 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Fri, 19 Sep 2008 11:08:47 +0800 (CST) Subject: [gst-devel] Problems about Webcam streaming Message-ID: <11294801.140541221793727672.JavaMail.coremail@bj126app56.126.com> I would like to capture video from a webcam, encode and send it through udp.Then receive and display the video at another machine, how should I build the pipeline? Thank you in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: From irfanshaikh at tataelxsi.co.in Fri Sep 19 06:49:35 2008 From: irfanshaikh at tataelxsi.co.in (irfanshaikh) Date: Fri, 19 Sep 2008 10:19:35 +0530 Subject: [gst-devel] Problems about Webcam streaming In-Reply-To: <11294801.140541221793727672.JavaMail.coremail@bj126app56.126.com> Message-ID: <008b01c91a13$1dbf31e0$37033c0a@telxsi.com> Hi 1)you can use a v4lsrc as source use the appropiate encoders required. Mux the Video in appropiate container and stream it using udpsink 2)Orelse you can use a v4lsrc as source use appropiate encoders required, packetize using appropiate payload packetizer and stream the data using rtpsend or rtpbin. Tell me if it wrks for you. Regards, ~irfan -----Original Message----- From: gstreamer-devel-bounces at lists.sourceforge.net [mailto:gstreamer-devel-bounces at lists.sourceforge.net]On Behalf Of liangzhihong1984 Sent: Friday, September 19, 2008 8:39 AM To: gstreamer-devel at lists.sourceforge.net Subject: [gst-devel] Problems about Webcam streaming I would like to capture video from a webcam, encode and send it through udp.Then receive and display the video at another machine, how should I build the pipeline? Thank you in advance. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. -------------- next part -------------- An HTML attachment was scrubbed... URL: From irfanshaikh at tataelxsi.co.in Fri Sep 19 07:09:59 2008 From: irfanshaikh at tataelxsi.co.in (irfanshaikh) Date: Fri, 19 Sep 2008 10:39:59 +0530 Subject: [gst-devel] ffmux_asf Query Message-ID: <009a01c91a15$f74f3b10$37033c0a@telxsi.com> Hi Ved, 1) As you replied, reading the elementary stream from file directly (filesrc ! mux),wont work. The filesrc wont attach any caps and no timestamp to the buffer and of course no parsed data. For the following: gst-launch filesrc location=/root/Desktop/audioVideo/CodecH264.dat ! ffmux_asf ! filesink location=/root/Desktop/ajit1.asf Can i know the alternative ? Or else if it is the implementation issue, can you suggest me where all in the plugin-code{functions} do i have to make changes so as i can be able to mux the audio/video data from the encoded audio/video files. Above issue is more critical to me. I would be thankful to you a lot. 2)[root at localhost ffmpeg]# gst-launch filesrc location=/root/Desktop/teststreams/mpeg4/mpeg4_mp3.ASF ! asfdemux name=demux demux.audio_00 ! queue ! mux.audio_0 demux.video_00 ! queue ! mux.video_0 ffmux_asf name=mux ! filesink location=/root/Desktop/test2.asf I hav tried using asfdemux but yet i am not able to MUX the audiodata.I am getting the o/p file of zero bytes. THANKING YOU, Regards, Irfan. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From ved.kpl at gmail.com Fri Sep 19 07:42:51 2008 From: ved.kpl at gmail.com (ved kpl) Date: Fri, 19 Sep 2008 11:12:51 +0530 Subject: [gst-devel] ffmux_asf Query In-Reply-To: <009a01c91a15$f74f3b10$37033c0a@telxsi.com> References: <009a01c91a15$f74f3b10$37033c0a@telxsi.com> Message-ID: <7496c23f0809182242u1ab7ce25x886d0e5682a0db5b@mail.gmail.com> HI, Is the dat file elementary stream or some container format?. If its just an elementary stream then u need a parse ( h264parse) in between. (filesrc ! 'video/x-h264' ! h264parse ! ... ) . But then its a DAT file, so probably you need to strip off the unwanted bytes in it before giving it to h264parse. ffmux_asf , guess does not take h264 so add h264 decoder and supported encoder also. What mpeg4 data does mpeg4_mp3.asf have? Check the formats that ffmux_asf supports. Ved On Fri, Sep 19, 2008 at 10:39 AM, irfanshaikh wrote: > > Hi Ved, > > 1) As you replied, reading the elementary stream from file directly (filesrc > ! mux),wont work. The filesrc wont attach any caps and no timestamp to the > buffer and of course no parsed data. > > For the following: > gst-launch filesrc location=/root/Desktop/audioVideo/CodecH264.dat ! > ffmux_asf ! filesink location=/root/Desktop/ajit1.asf > > Can i know the alternative ? Or else if it is the implementation issue, can > you suggest me where all in the plugin-code{functions} do i have to make > changes so as i can be able to mux the audio/video data from the encoded > audio/video files. > > Above issue is more critical to me. I would be thankful to you a lot. > > > 2)[root at localhost ffmpeg]# gst-launch filesrc > location=/root/Desktop/teststreams/mpeg4/mpeg4_mp3.ASF ! asfdemux name=demux > demux.audio_00 ! queue ! mux.audio_0 demux.video_00 ! queue ! mux.video_0 > ffmux_asf name=mux ! filesink location=/root/Desktop/test2.asf > > I hav tried using asfdemux but yet i am not able to MUX the audiodata.I am > getting the o/p file of zero bytes. > > THANKING YOU, > > Regards, > Irfan. > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. > From rahulmittal321 at gmail.com Fri Sep 19 09:18:42 2008 From: rahulmittal321 at gmail.com (Rahul Mittal) Date: Fri, 19 Sep 2008 12:48:42 +0530 Subject: [gst-devel] SSRC collision with Multicast RTP in Rtpbin Message-ID: Hi Guys,I have come across something in rtpsession.c I am trying to fetch a multicast RTP stream using RTPBIN. The problem is that when i send any RR packet, since its a multicast stream, the RR packets get looped back to us. RTPSession checks for collision and since for the first packet there are no known conflicts uptill that time, the rtpsession assumes that there is a collision of ssrc and turns the received_bye flag to true. This causes rtp packets to be dropped untill new ssrc is created. Just wanted to know whether this is a known bug and if not then shouldnt we be checking the netaddress of the loop backed RR packet with the netaddress of the machine. thanx Rahul -------------- next part -------------- An HTML attachment was scrubbed... URL: From christophe.dehais at gmail.com Fri Sep 19 12:05:05 2008 From: christophe.dehais at gmail.com (Christophe Dehais) Date: Fri, 19 Sep 2008 12:05:05 +0200 Subject: [gst-devel] Dynamically adding a filter to a video pipeline In-Reply-To: <4a305d880809180258m7f444e13lf14f45d4018d0fb4@mail.gmail.com> References: <4a305d880809180258m7f444e13lf14f45d4018d0fb4@mail.gmail.com> Message-ID: <4a305d880809190305h40145f51r5802be882d8d7a4b@mail.gmail.com> Ok, I found a workaround (see attached). The idea is to replace the element by a bin made of colorspace ! filter ! colorspace which will take care of caps negotiation around the real filter. (Now that I think of it, there is something similar in Rhythmbox, except that colorspace is replaced by audioconvert). That's not really ideal since the pipeline is then encumbered with a lot of colorspace converters but most of them will probably operate in passthough mode so the overhead should be low. The conclusion for me at this point is: if you want to dynamically replace an element by another, it has to be done in a way that doesn't require renegotiation of the pads concerned by the broken links. cheers, Christophe On Thu, Sep 18, 2008 at 11:58 AM, Christophe Dehais wrote: > Hi everyone! > > I have this simple pipeline: videosrc ! identity ! videosink > > (videosrc is a bin embedding videotestsrc or v4l2src and colorspace, > videosink is a bin embedding colorspace and xvimagesink) > > I simply want to replace identity by a videofilter (e.g. edgetv), > while the pipeline is playing. > > So I made a little test app (see attachments) and I guess I'm > following what's described in the design docs here: > http://webcvs.freedesktop.org/gstreamer/gstreamer/docs/design/part-block.txt?view=markup > > except that I'm blocking the src pad synchronously. > > It doesn't work because when unblocking back the videosrc source pad, > a negotiation problem occurs. If I replace 'identity' by > 'shagadelictv' (which has the same in and out caps as edgetv), it > works fine. > > So my question is two fold: > 1) do I miss something in the design doc ? > 2) how can I force the videosrc bin to renegociate with the rest of > the pipeline ? > > thanks for any hints, > Christophe > -------------- next part -------------- A non-text attachment was scrubbed... Name: dyn_pipeline.py Type: application/octet-stream Size: 3953 bytes Desc: not available URL: From rbultje at ronald.bitfreak.net Sat Sep 20 14:40:41 2008 From: rbultje at ronald.bitfreak.net (Ronald S. Bultje) Date: Sat, 20 Sep 2008 08:40:41 -0400 Subject: [gst-devel] Fwd: In-Reply-To: <9D5E1752379A43408015F7FE9846611578275C@CHNEXVS01.VSNLXCHANGE.COM> References: <9D5E1752379A43408015F7FE9846611578275C@CHNEXVS01.VSNLXCHANGE.COM> Message-ID: <34539a480809200540m60839a39nfcac4bc1d7bacd55@mail.gmail.com> Can someone help him? ---------- Forwarded message ---------- From: Irfan Shaikh Date: Sat, Sep 20, 2008 at 6:08 AM Subject: To: rbultje at ronald.bitfreak.net Cc: rbultje at ronald.bitfreak.net Hi Ronald, In gst-plugins-0.8.12/gst/asfdemux, How can gstasfmux.c can be used as ASF muxer plugin for GStreamer. Is it feaseable to specify the bit rate and play stream on WMP player.can you help me in knowing how that can be done? It is speacified that, stream does NOT work on Windows Media Player,because we do not specify bitrate. Can i know what all changes i have to do, So as to play back the ASF muxed file in WMP player. My requirement is to mux the, MPEG4, MJPEG, H264 video and G711, G726 audio in ASF file format. Can i know if possible, where all in the code i need to modify, THanking you, Irfan. This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. From jpuydt at free.fr Sat Sep 20 21:56:58 2008 From: jpuydt at free.fr (Julien Puydt) Date: Sat, 20 Sep 2008 21:56:58 +0200 Subject: [gst-devel] Using GstAppSink Message-ID: <48D5558A.4010300@free.fr> Hi, I'm trying to use GstAppSink in my ekiga code -- however I don't know exactly how to make those pipelines work : gst-launch videotestsrc ! appsink max_buffers=2 drop=true caps=video/x-raw-yuv,format=\(fourcc\)I420,width=320,height=240,framerate=30 name=ekiga_sink gst-launch v4lsrc ! appsink max_buffers=2 drop=true caps=video/x-raw-yuv,format=\(fourcc\)I420,width=320,height=240,framerate=30 name=ekiga_sink I'm pretty sure I lack some magic in-between my source and my sink... Snark on #ekiga From jpuydt at free.fr Sun Sep 21 08:44:09 2008 From: jpuydt at free.fr (Julien Puydt) Date: Sun, 21 Sep 2008 08:44:09 +0200 Subject: [gst-devel] Using GstAppSink In-Reply-To: <48D5558A.4010300@free.fr> References: <48D5558A.4010300@free.fr> Message-ID: <48D5ED39.8010104@free.fr> Julien Puydt a ?crit : > Hi, > > I'm trying to use GstAppSink in my ekiga code -- however I don't know > exactly how to make those pipelines work : > > gst-launch videotestsrc ! appsink max_buffers=2 drop=true > caps=video/x-raw-yuv,format=\(fourcc\)I420,width=320,height=240,framerate=30 > name=ekiga_sink > > gst-launch v4lsrc ! appsink max_buffers=2 drop=true > caps=video/x-raw-yuv,format=\(fourcc\)I420,width=320,height=240,framerate=30 > name=ekiga_sink > > I'm pretty sure I lack some magic in-between my source and my sink... It seems the "framerate=30" is the problem... Snark on #gstreamer From irfanshaikh at tataelxsi.co.in Mon Sep 22 09:37:04 2008 From: irfanshaikh at tataelxsi.co.in (Irfan Shaikh) Date: Mon, 22 Sep 2008 13:07:04 +0530 Subject: [gst-devel] FW: ffmux_asf Query References: <000e01c91c84$a4a35c40$37033c0a@telxsi.com> Message-ID: <9D5E1752379A43408015F7FE9846611578276C@CHNEXVS01.VSNLXCHANGE.COM> Hi ved, As per ur suggestions i hav been trying to use ffmux_asf. I hav been looking into the source code of ffmpegs muxer code. I have been trying to mux the videotestsrc and audiotestsrc using ffmux_asf. but i am not able to get any output file. Will you please help to achieve o/p ASF file. I used, gst-launch audiotestsrc ! ffenc_g726 ! queue ! mux.audio_0 videotestsrc ! ffdec_mpeg4 ! queue ! mux.video_0 ffmux_asf name=mux ! filesink location=/root/Desktop/vidaud.asf This is also not working, Please help me regarding this pipeline if this works i fine i will be half way through, Since i can conclude if i use appropiate encoders for file formats needs to be supported ffmux_asf can work properly. Will finish going through the source code by Evening so as to be more clear about ffmux_asf plug-in. Regards, Irfan -----Original Message----- From: ved kpl [mailto:ved.kpl at gmail.com] Sent: Friday, September 19, 2008 11:13 AM To: irfanshaikh at tataelxsi.co.in Cc: Discussion of the development of GStreamer Subject: Re: ffmux_asf Query HI, Is the dat file elementary stream or some container format?. If its just an elementary stream then u need a parse ( h264parse) in between. (filesrc ! 'video/x-h264' ! h264parse ! ... ) . But then its a DAT file, so probably you need to strip off the unwanted bytes in it before giving it to h264parse. ffmux_asf , guess does not take h264 so add h264 decoder and supported encoder also. What mpeg4 data does mpeg4_mp3.asf have? Check the formats that ffmux_asf supports. Ved On Fri, Sep 19, 2008 at 10:39 AM, irfanshaikh wrote: > > Hi Ved, > > 1) As you replied, reading the elementary stream from file directly (filesrc > ! mux),wont work. The filesrc wont attach any caps and no timestamp to the > buffer and of course no parsed data. > > For the following: > gst-launch filesrc location=/root/Desktop/audioVideo/CodecH264.dat ! > ffmux_asf ! filesink location=/root/Desktop/ajit1.asf > > Can i know the alternative ? Or else if it is the implementation issue, can > you suggest me where all in the plugin-code{functions} do i have to make > changes so as i can be able to mux the audio/video data from the encoded > audio/video files. > > Above issue is more critical to me. I would be thankful to you a lot. > > > 2)[root at localhost ffmpeg]# gst-launch filesrc > location=/root/Desktop/teststreams/mpeg4/mpeg4_mp3.ASF ! asfdemux name=demux > demux.audio_00 ! queue ! mux.audio_0 demux.video_00 ! queue ! mux.video_0 > ffmux_asf name=mux ! filesink location=/root/Desktop/test2.asf > > I hav tried using asfdemux but yet i am not able to MUX the audiodata.I am > getting the o/p file of zero bytes. > > THANKING YOU, > > Regards, > Irfan. > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. > This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ajitjohn at tataelxsi.co.in Mon Sep 22 10:00:31 2008 From: ajitjohn at tataelxsi.co.in (ajitjohn) Date: Mon, 22 Sep 2008 13:30:31 +0530 Subject: [gst-devel] H.264 plugin Message-ID: <001b01c91c89$49662ce0$68033c0a@telxsi.com> Hi bob, There are h264 plugins available in gstreamer like h264encoder ,decoder and h264packetiser just try doing gst-inspect | grep h264 u will get the list of h264 plugins installed in gstreamer. I have installed gstreamer-0.10.20 and i have the following h264 plugins installed ffmpeg: ffdec_h264: FFMPEG H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 decoder h264parse: h264parse: H264Parse rtp: rtph264pay: RTP packet payloader rtp: rtph264depay: RTP packet depayloader The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From ajitjohn at tataelxsi.co.in Mon Sep 22 09:40:02 2008 From: ajitjohn at tataelxsi.co.in (ajitjohn) Date: Mon, 22 Sep 2008 13:10:02 +0530 Subject: [gst-devel] mjpeg packetiser plugin Message-ID: <001601c91c86$6ce37090$68033c0a@telxsi.com> Hii all, Does gstreamer have mjpeg packetiser plugin based on rfc 2435? regards, ~Ajit. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From marc.leeman at gmail.com Mon Sep 22 10:12:54 2008 From: marc.leeman at gmail.com (Marc Leeman) Date: Mon, 22 Sep 2008 10:12:54 +0200 Subject: [gst-devel] flutsmux/flutsdemux Message-ID: <1f729c480809220112s39e787b9gbf8c9caa1991c95@mail.gmail.com> (resend from Friday, my ISP smtp server seems to be acting up, I didn't see this getting on the ML yet.) I am trying to grab a TS over RTP stream from the networki (10 Mbps, I-only), strip it and send it back out as ES over RTP. For some reason, I always get stuck when using flutsdemux (or around there). This is one of the latest pipelines I've been using: gst-launch-0.10 -v gstrtpbin name=rtpbin \ latency=200 gstrtpbin \ name=sendrtpbin \ latency=200 \ udpsrc multicast-group=226.255.0.9 \ port=50200 caps="application/x-rtp,clock-rate=(int)90000" ! \ rtpbin.recv_rtp_sink_0 rtpbin. caps="application/x-rtp" ! \ rtpmp2tdepay ! \ flutsdemux ! \ flutsmux ! \ rtpmp2tpay ! \ sendrtpbin.send_rtp_sink_1 \ sendrtpbin. caps="application/x-rtp/x-rtp,clock-rate=(int)90000,system=0\ ! \ udpsink host=226.255.12.0 port=5000 Quality is poor and packets get dropped (gstreamer, mplayer and vlc). Replacing "flutsdemux ! flutsmux ! rtpmp2tpay" with "flutsdemux ! rtpmpvpay" (combination I want to test/use), the visual result is the same as the above. When leaving out the flutsdemux/flutsmux; the quality is fine. Simplified, this is (same poor visual result): gst-launch udpsrc multicast-group=226.255.0.8 port=50100 \ caps="application/x-rtp,clock-rate=(int)90000" \ ! .recv_rtp_sink_0 gstrtpbin \ ! typefind \ ! rtpmp2tdepay \ ! flutsdemux \ ! rtpmpvpay \ ! udpsink host=226.255.12.0 port=5000 Again, leaving out flutsdemux shows perfect video quality: gst-launch udpsrc multicast-group=226.255.0.8 port=50100 \ caps="application/x-rtp,clock-rate=(int)90000" \ ! .recv_rtp_sink_0 gstrtpbin \ ! typefind \ ! rtpmp2tdepay \ ! rtpmp2tpay \ ! udpsink host=226.255.12.0 port=5000 I've tried playing around with queues (suggestion from IRC), rtpbins, ... to no avail. Is this a known issue, or something I missed? [mleeman at bane ~]$ dpkg -l |grep gstreamer |cut -c 4-60 gstreamer-tools 0.10.20-1 gstreamer0.10-alsa 0.10.19-2 gstreamer0.10-ffmpeg 0.10.4-3 gstreamer0.10-fluendo-mp3 0.10.7.debian-1 gstreamer0.10-fluendo-mpegdemux 0.10.15-1 gstreamer0.10-fluendo-mpegmux 0.10.4-1 gstreamer0.10-gnomevfs 0.10.20-1 gstreamer0.10-plugins-bad 0.10.7-2 gstreamer0.10-plugins-base 0.10.20-1 gstreamer0.10-plugins-farsight 0.12.8-1 gstreamer0.10-plugins-good 0.10.8-4 gstreamer0.10-plugins-ugly 0.10.8-1 gstreamer0.10-sdl 0.10.7-2 gstreamer0.10-tools 0.10.20-1 gstreamer0.10-x 0.10.20-1 libgstreamer-plugins-base0.10-0 0.10.20-1 libgstreamer-plugins-base0.10-dev 0.10.20-1 libgstreamer0.10-0 0.10.20-1 libgstreamer0.10-0-dbg 0.10.20-1 libgstreamer0.10-dev 0.10.20-1 -- br, marc. From zaheermerali at gmail.com Mon Sep 22 11:33:12 2008 From: zaheermerali at gmail.com (Zaheer Merali) Date: Mon, 22 Sep 2008 10:33:12 +0100 Subject: [gst-devel] DVB pipeline consumes a lot of CPU? In-Reply-To: References: Message-ID: <15e616860809220233m175efc6fj2611f48a70ccc601@mail.gmail.com> On Thu, Sep 18, 2008 at 11:13 AM, Tero Saarni wrote: > Hi, > > I'm seeing very high CPU usage when running pipeline for DVB-T stream. > Setting sync=false like instructed in the example in the docs makes > it consume less CPU but with the cost of loosing lipsync. > > I wonder if this is a well known problem and if there is a fix? > > Here's the pipeline: > > gst-launch \ > dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 > frequency=714000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 > hierarchy=0 pids=0:256:512:650 ! \ > flutsdemux name=demuxer \ > demuxer. ! queue max-size-buffers=0 max-size-time=0 ! > mpeg2dec ! ffmpegcolorspace ! xvimagesink sync=false \ > demuxer. ! queue max-size-buffers=0 max-size-time=0 ! mad > ! alsasink sync=false > > I'm running gstreamer 0.10.18 on Ubuntu Hardy. > > -- > Tero Hi flutsdemux has recently been optimised for speed. If you try the latest version, in gst-plugins-bad cvs directory gst/mpegdemux it should work a lot faster. Zaheer From arnabsamanta at tataelxsi.co.in Mon Sep 22 12:27:08 2008 From: arnabsamanta at tataelxsi.co.in (arnabsamanta) Date: Mon, 22 Sep 2008 15:57:08 +0530 Subject: [gst-devel] querry : copying an ordinary buffer to GstBuffer In-Reply-To: Message-ID: <00ef01c91c9d$c51487b0$26033c0a@telxsi.com> Hi , how can i copy the contents of an ordinary buffer created in C++ to the GstBuffer buffer. i have the following.... can any body tell me if its correct get_data(void *buffer, size_t frame) { GstBuffer *gst_buffer ; gst_buffer = gst_buffer_new_and_alloc (frame); memcpy(gst_buffer , buffer, frame); } where buffer is the memory segment created by the previous function by "new" i did so as i guess , gst_buffer = GST_BUFFER_CAST(buffer); or gst_buffer = GST_BUFFER_DATA(buffer) wont work because "buffer" is not a n GST type buffer. is my understanding as stated above is correct ? if wrong , can any body tell me the correct way ? regards, ~Arnab The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From bisht.sudarshan at gmail.com Mon Sep 22 12:47:07 2008 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Mon, 22 Sep 2008 16:17:07 +0530 Subject: [gst-devel] querry : copying an ordinary buffer to GstBuffer In-Reply-To: <00ef01c91c9d$c51487b0$26033c0a@telxsi.com> References: <00ef01c91c9d$c51487b0$26033c0a@telxsi.com> Message-ID: <785339900809220347q2992aef8r72bceaade9a8059a@mail.gmail.com> Hi , Try following , would work , get_data(void *buffer, size_t frame) { GstBuffer *gst_buffer ; gst_buffer = gst_buffer_new_and_alloc (frame); memcpy(GST_BUFFER_DATA(gst_buffer) , buffer, frame); } On Mon, Sep 22, 2008 at 3:57 PM, arnabsamanta wrote: > Hi , > how can i copy the contents of an ordinary buffer created in C++ to > the > GstBuffer buffer. > i have the following.... can any body tell me if its correct > > get_data(void *buffer, size_t frame) > { > GstBuffer *gst_buffer ; > gst_buffer = gst_buffer_new_and_alloc (frame); > memcpy(gst_buffer , buffer, frame); > } > > where buffer is the memory segment created by the previous function > by > "new" > > > i did so as i guess , > gst_buffer = GST_BUFFER_CAST(buffer); > or > gst_buffer = GST_BUFFER_DATA(buffer) > > wont work because "buffer" is not a n GST type buffer. > > is my understanding as stated above is correct ? > if wrong , can any body tell me the correct way ? > > regards, > ~Arnab > > > The information contained in this electronic message and any attachments to > this message are intended for the exclusive use of the addressee(s) and may > contain proprietary, confidential or privileged information. If you are not > the intended recipient, you should not disseminate, distribute or copy this > e-mail. Please notify the sender immediately and destroy all copies of this > message and any attachments contained in it. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From thiagossantos at gmail.com Mon Sep 22 13:02:57 2008 From: thiagossantos at gmail.com (thiagoss) Date: Mon, 22 Sep 2008 08:02:57 -0300 Subject: [gst-devel] FW: ffmux_asf Query In-Reply-To: <9D5E1752379A43408015F7FE9846611578276C@CHNEXVS01.VSNLXCHANGE.COM> References: <000e01c91c84$a4a35c40$37033c0a@telxsi.com> <9D5E1752379A43408015F7FE9846611578276C@CHNEXVS01.VSNLXCHANGE.COM> Message-ID: On Mon, Sep 22, 2008 at 4:37 AM, Irfan Shaikh wrote: > > Hi ved, > > As per ur suggestions i hav been trying to use ffmux_asf. I hav > been > looking into the source code of ffmpegs muxer code. > > I have been trying to mux the videotestsrc and audiotestsrc using > ffmux_asf. > but i am not able to get any output file. > Will you please help to achieve o/p ASF file. > > I used, > > gst-launch audiotestsrc ! ffenc_g726 ! queue ! mux.audio_0 > videotestsrc ! ffdec_mpeg4 ! queue ! mux.video_0 ffmux_asf name=mux ! > filesink location=/root/Desktop/vidaud.asf > This is wrong "videotestsrc ! ffdec_mpeg4", videotestsrc does not generate mpeg4 video streams. I suppose you were trying to use ffenc_mpeg4. Right? Even so, ffmux_asf caps are not compatible with ffenc_mpeg4. > > > This is also not working, Please help me regarding this pipeline if this > works i fine i will be half way through, > Since i can conclude if i use appropiate encoders for file formats needs to > be supported ffmux_asf can work properly. > > Will finish going through the source code by Evening so as to be more clear > about ffmux_asf plug-in. > > Regards, > Irfan > > > > > -----Original Message----- > From: ved kpl [mailto:ved.kpl at gmail.com ] > Sent: Friday, September 19, 2008 11:13 AM > To: irfanshaikh at tataelxsi.co.in > Cc: Discussion of the development of GStreamer > Subject: Re: ffmux_asf Query > > > HI, > > Is the dat file elementary stream or some container format?. If its > just an elementary stream then u need a parse ( h264parse) in between. > (filesrc ! 'video/x-h264' ! h264parse ! ... ) . But then its a DAT > file, so probably you need to strip off the unwanted bytes in it > before giving it to h264parse. ffmux_asf , guess does not take h264 so > add h264 decoder and supported encoder also. > > What mpeg4 data does mpeg4_mp3.asf have? Check the formats that > ffmux_asf supports. > > Ved > > On Fri, Sep 19, 2008 at 10:39 AM, irfanshaikh > wrote: > > > > Hi Ved, > > > > 1) As you replied, reading the elementary stream from file directly > (filesrc > > ! mux),wont work. The filesrc wont attach any caps and no timestamp to > the > > buffer and of course no parsed data. > > > > For the following: > > gst-launch filesrc location=/root/Desktop/audioVideo/CodecH264.dat > ! > > ffmux_asf ! filesink location=/root/Desktop/ajit1.asf > > > > Can i know the alternative ? Or else if it is the implementation issue, > can > > you suggest me where all in the plugin-code{functions} do i have to make > > changes so as i can be able to mux the audio/video data from the encoded > > audio/video files. > > > > Above issue is more critical to me. I would be thankful to you a lot. > > > > > > 2)[root at localhost ffmpeg]# gst-launch filesrc > > location=/root/Desktop/teststreams/mpeg4/mpeg4_mp3.ASF ! asfdemux > name=demux > > demux.audio_00 ! queue ! mux.audio_0 demux.video_00 ! queue ! mux.video_0 > > ffmux_asf name=mux ! filesink location=/root/Desktop/test2.asf > > > > I hav tried using asfdemux but yet i am not able to MUX the > audiodata.I am > > getting the o/p file of zero bytes. > > > > THANKING YOU, > > > > Regards, > > Irfan. > > > > > > The information contained in this electronic message and any attachments > to this message are intended for the exclusive use of the addressee(s) and > may contain proprietary, confidential or privileged information. If you are > not the intended recipient, you should not disseminate, distribute or copy > this e-mail. Please notify the sender immediately and destroy all copies of > this message and any attachments contained in it. > > > > > > > > > This message (including any attachment) is confidential and may be > legally privileged. Access to this message by anyone other than the intended > recipient(s) listed above is unauthorized. If you are not the intended > recipient you are hereby notified that any disclosure, copying, or > distribution of the message, or any action taken or omission of action by > you in reliance upon it, is prohibited and may be unlawful. Please > immediately notify the sender by reply e-mail and permanently delete all > copies of the message if you have received this message in error. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) -------------- next part -------------- An HTML attachment was scrubbed... URL: From jaredm at gmx.com Mon Sep 22 13:08:17 2008 From: jaredm at gmx.com (Jared Moore) Date: Mon, 22 Sep 2008 21:08:17 +1000 Subject: [gst-devel] bitrate in ffenc_h263p (and other encoders) not working Message-ID: <4594288b0809220408t4781b97do34c1655b724b81a0@mail.gmail.com> Hi guys, I have a question about some encoders. I am doing a project which involves sending video over RTP, simulating a low-bandwidth connection and then manually fiddling around with the encoder bitrate in order to compensate for the low bandwidth. My question is that changing the bitrate on both the theoraenc and ffenc_h263p elements doesn't seen to do anything. Here is my code which i am using for testing: gst-launch -v filesrc location=\"$1\" ! decodebin ! queue ! videorate ! ffmpegcolorspace ! ffenc_h263p bitrate=300 ! ffdec_h263 ! xvimagesink This is just meant to test what different bitrates look like, and testing whether bitrate actually works. This definitely works for h264 - setting the bitrate very low (e.g. 3) does make the video look really horrible and use less bandwidth. Unfortunately I can only really use h263 because other encodings that I've tried can't be done in real-time on my eeepc. Changing the bitrate in the above code snippet doesn't do anything visible, there's absolutely no change between bitrate=1 and bitrate=300000. Any thoughts or examples of how to use bitrate differently? Alternatively, is there a way to substantially increase the performance of the h264 encoder (with some other tradeoff, e.g. more bandwidth required, having to make a new file, whatever, the only problem is CPU usage)? Cheers, Jared From mediadevel at gmail.com Mon Sep 22 13:17:52 2008 From: mediadevel at gmail.com (john david) Date: Mon, 22 Sep 2008 16:47:52 +0530 Subject: [gst-devel] how to give pcr pid on separate pid Message-ID: Hi, I encoded one file using gstreamer. After successful encode, i open that file using some tool. In that, i saw pat, pmt, pcrpid, vpid, apid. Here vpid is equal to the pcr pid. But i want pcr pid on separate pid. For this what are the changes that i have to do in gstreamer. Regards, John david. -------------- next part -------------- An HTML attachment was scrubbed... URL: From irfanshaikh at tataelxsi.co.in Mon Sep 22 13:37:06 2008 From: irfanshaikh at tataelxsi.co.in (Irfan Shaikh) Date: Mon, 22 Sep 2008 17:07:06 +0530 Subject: [gst-devel] FW: ffmux_asf Query References: <000e01c91c84$a4a35c40$37033c0a@telxsi.com><9D5E1752379A43408015F7FE9846611578276C@CHNEXVS01.VSNLXCHANGE.COM> Message-ID: <9D5E1752379A43408015F7FE98466115782775@CHNEXVS01.VSNLXCHANGE.COM> Hi, First of all thanks for yur reply...Yaa i wanted to use ffenc_mpeg4.... But still i dont get the appropiate o/p file with ASF container... 1) Can you please help me in solving this problem....to mux the audio video file formats in ASF container format. 2) what changes do i need to do to make ffenc_mpeg4 caps compatible with ffmux_asf. Thanking you in advance, Regards, Irfan -----Original Message----- From: thiagoss [mailto:thiagossantos at gmail.com] Sent: Mon 9/22/2008 4:32 PM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] FW: ffmux_asf Query On Mon, Sep 22, 2008 at 4:37 AM, Irfan Shaikh wrote: > > Hi ved, > > As per ur suggestions i hav been trying to use ffmux_asf. I hav > been > looking into the source code of ffmpegs muxer code. > > I have been trying to mux the videotestsrc and audiotestsrc using > ffmux_asf. > but i am not able to get any output file. > Will you please help to achieve o/p ASF file. > > I used, > > gst-launch audiotestsrc ! ffenc_g726 ! queue ! mux.audio_0 > videotestsrc ! ffdec_mpeg4 ! queue ! mux.video_0 ffmux_asf name=mux ! > filesink location=/root/Desktop/vidaud.asf > This is wrong "videotestsrc ! ffdec_mpeg4", videotestsrc does not generate mpeg4 video streams. I suppose you were trying to use ffenc_mpeg4. Right? Even so, ffmux_asf caps are not compatible with ffenc_mpeg4. > > > This is also not working, Please help me regarding this pipeline if this > works i fine i will be half way through, > Since i can conclude if i use appropiate encoders for file formats needs to > be supported ffmux_asf can work properly. > > Will finish going through the source code by Evening so as to be more clear > about ffmux_asf plug-in. > > Regards, > Irfan > > > > > -----Original Message----- > From: ved kpl [mailto:ved.kpl at gmail.com ] > Sent: Friday, September 19, 2008 11:13 AM > To: irfanshaikh at tataelxsi.co.in > Cc: Discussion of the development of GStreamer > Subject: Re: ffmux_asf Query > > > HI, > > Is the dat file elementary stream or some container format?. If its > just an elementary stream then u need a parse ( h264parse) in between. > (filesrc ! 'video/x-h264' ! h264parse ! ... ) . But then its a DAT > file, so probably you need to strip off the unwanted bytes in it > before giving it to h264parse. ffmux_asf , guess does not take h264 so > add h264 decoder and supported encoder also. > > What mpeg4 data does mpeg4_mp3.asf have? Check the formats that > ffmux_asf supports. > > Ved > > On Fri, Sep 19, 2008 at 10:39 AM, irfanshaikh > wrote: > > > > Hi Ved, > > > > 1) As you replied, reading the elementary stream from file directly > (filesrc > > ! mux),wont work. The filesrc wont attach any caps and no timestamp to > the > > buffer and of course no parsed data. > > > > For the following: > > gst-launch filesrc location=/root/Desktop/audioVideo/CodecH264.dat > ! > > ffmux_asf ! filesink location=/root/Desktop/ajit1.asf > > > > Can i know the alternative ? Or else if it is the implementation issue, > can > > you suggest me where all in the plugin-code{functions} do i have to make > > changes so as i can be able to mux the audio/video data from the encoded > > audio/video files. > > > > Above issue is more critical to me. I would be thankful to you a lot. > > > > > > 2)[root at localhost ffmpeg]# gst-launch filesrc > > location=/root/Desktop/teststreams/mpeg4/mpeg4_mp3.ASF ! asfdemux > name=demux > > demux.audio_00 ! queue ! mux.audio_0 demux.video_00 ! queue ! mux.video_0 > > ffmux_asf name=mux ! filesink location=/root/Desktop/test2.asf > > > > I hav tried using asfdemux but yet i am not able to MUX the > audiodata.I am > > getting the o/p file of zero bytes. > > > > THANKING YOU, > > > > Regards, > > Irfan. > > > > > > The information contained in this electronic message and any attachments > to this message are intended for the exclusive use of the addressee(s) and > may contain proprietary, confidential or privileged information. If you are > not the intended recipient, you should not disseminate, distribute or copy > this e-mail. Please notify the sender immediately and destroy all copies of > this message and any attachments contained in it. > > > > > > > > > This message (including any attachment) is confidential and may be > legally privileged. Access to this message by anyone other than the intended > recipient(s) listed above is unauthorized. If you are not the intended > recipient you are hereby notified that any disclosure, copying, or > distribution of the message, or any action taken or omission of action by > you in reliance upon it, is prohibited and may be unlawful. Please > immediately notify the sender by reply e-mail and permanently delete all > copies of the message if you have received this message in error. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. -------------- next part -------------- An HTML attachment was scrubbed... URL: From fthiery at gmail.com Mon Sep 22 15:09:36 2008 From: fthiery at gmail.com (Florent) Date: Mon, 22 Sep 2008 15:09:36 +0200 Subject: [gst-devel] mjpeg packetiser plugin In-Reply-To: <001601c91c86$6ce37090$68033c0a@telxsi.com> References: <001601c91c86$6ce37090$68033c0a@telxsi.com> Message-ID: <1efe3a6e0809220609y3ca92875vfb197d3a862ebab5@mail.gmail.com> > > Does gstreamer have mjpeg packetiser plugin based on rfc 2435? > You mean, rtpmjpegpay/depay ? Flo -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Mon Sep 22 16:15:14 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Mon, 22 Sep 2008 17:15:14 +0300 Subject: [gst-devel] C code to gstreamer plugin conversion In-Reply-To: <006001c91965$ee171050$26033c0a@telxsi.com> References: <006001c91965$ee171050$26033c0a@telxsi.com> Message-ID: <48D7A872.4030404@hora-obscura.de> hi, arnabsamanta schrieb: > Hi > i have an application which has a number of C++ files and i need to make > the application as a gstreamer plug in. > what are the basic steps to do so ? > i know how to create a source and a sink pad. > do i need to declare all functions of all the C++ codes within the init > functions ? > Please read the plugin writers guide. Also have a look at the gst-plugins-* packages which has a few c++ based plugins (just search for *.cc) Stefan > regards, > ~Arnab > > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From irfanshaikh at tataelxsi.co.in Mon Sep 22 18:06:46 2008 From: irfanshaikh at tataelxsi.co.in (irfanshaikh) Date: Mon, 22 Sep 2008 21:36:46 +0530 Subject: [gst-devel] mjpeg packetiser plugin In-Reply-To: <1efe3a6e0809220609y3ca92875vfb197d3a862ebab5@mail.gmail.com> Message-ID: <004f01c91ccd$37602110$37033c0a@telxsi.com> yes i mean rtpmjpegpay -----Original Message----- From: Florent [mailto:fthiery at gmail.com] Sent: Monday, September 22, 2008 6:40 PM To: ajitjohn at tataelxsi.co.in; Discussion of the development of GStreamer Subject: Re: [gst-devel] mjpeg packetiser plugin Does gstreamer have mjpeg packetiser plugin based on rfc 2435? You mean, rtpmjpegpay/depay ? Flo The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ml_benoitfouet at purplelabs.com Fri Sep 19 09:14:29 2008 From: ml_benoitfouet at purplelabs.com (Benoit Fouet) Date: Fri, 19 Sep 2008 09:14:29 +0200 Subject: [gst-devel] testing video with SDL In-Reply-To: <20080919001704.M94981@mail2.sysu.edu.cn> References: <20080919001704.M94981@mail2.sysu.edu.cn> Message-ID: <48D35155.40001@purplelabs.com> Hi, Chen Hailiang wrote: > Hi all: > I'm using gstreamer and SDL to make a very sample vedio player.I put all > the codes in one file; video_test.c. > My question is: how can i compile it? I tried some commands like"gcc -Wall > `sdl-config --libs gstreamer-0.10` video_test.c -o video_player",it didn't > work.Can someone can help? > > $ sdl-config --libs gstreamer-0.10 -lSDL -lpthread Usage: sdl-config [--prefix[=DIR]] [--exec-prefix[=DIR]] [--version] [--cflags] [--libs] [--static-libs] you misused sdl-config command maybe you meant: `sdl-config --libs` `pkg-config gstreamer-0.10 --libs` ? and you would still maybe lack the CPPPATH -- Benoit Fouet Purple Labs S.A. www.purplelabs.com From kmaraas at broadpark.no Tue Sep 16 13:41:53 2008 From: kmaraas at broadpark.no (Kjartan Maraas) Date: Tue, 16 Sep 2008 13:41:53 +0200 Subject: [gst-devel] GStreamer tarballs and ftp.gnome.org In-Reply-To: <20080916095709.GS30241@vuntz.net> References: <20080916095709.GS30241@vuntz.net> Message-ID: <1221565313.30030.1.camel@localhost.localdomain> ti., 16.09.2008 kl. 11.57 +0200, skrev Vincent Untz: > Hi, > > See [1] for past discussion about this :-) > > The latest gst-plugins-good we have on ftp.gnome.org is 0.10.6 while the > latest version really is 0.10.10. What can we do to fix this? > Previously I've always just downloaded the latest tarball and uploaded it on f.g.o, but I guess this should be a part of the GStreamer release process or we should be using the external repo for them? Cheers Kjartan From gmane at colin.guthr.ie Mon Sep 22 10:41:11 2008 From: gmane at colin.guthr.ie (Colin Guthrie) Date: Mon, 22 Sep 2008 09:41:11 +0100 Subject: [gst-devel] Some advice Message-ID: Hi, I have been working on hacking the phonon-gstreamer backend recently and as I'm a significant degree short of even having a rudimentary knowledge of gstreamer, this was a bit of an uphill struggle. I've managed to get a few things working now, but I have a couple of questions that hopefully some of the development gurus can answer me. 1) Is it possible to find out which sink is really used when you use gconfsink, autoaudiosink or halsink? 2) When streaming data from shoutcast, it is possible to decode the StreamTitle= header via icydemux. I have hacked phonon-gstreamer backed to do this now (in a similar way to how playbin does it). However it only extracts the title. When I use the Xine backend it appears to be able to extract inline metadata out of the stream content itself (e.g. when it finds a vorbis comment header or an id3 tag). That's as much as I can tell from observation. I've hacked the phonon-gstreamer to be able to "guess" the artist name from the StreamTitle but it's really not very nice. So my main question is, can the decoders handle inline metadata decoding and if so how can I enabled this? Many thanks Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] From msmith at xiph.org Mon Sep 22 19:55:06 2008 From: msmith at xiph.org (Michael Smith) Date: Mon, 22 Sep 2008 10:55:06 -0700 Subject: [gst-devel] Some advice In-Reply-To: References: Message-ID: <3c1737210809221055l7cd42603i93d011f44461a547@mail.gmail.com> On Mon, Sep 22, 2008 at 1:41 AM, Colin Guthrie wrote: > Hi, > > I have been working on hacking the phonon-gstreamer backend recently and > as I'm a significant degree short of even having a rudimentary knowledge > of gstreamer, this was a bit of an uphill struggle. > > I've managed to get a few things working now, but I have a couple of > questions that hopefully some of the development gurus can answer me. > > > 1) Is it possible to find out which sink is really used when you use > gconfsink, autoaudiosink or halsink? These are all bins (they contain other elements); so you can just iterate (possibly recursively) over all the elements in them, looking for sinks, and then look at the type of the sink. autoaudiosink and halsink are both pretty easy, but gconfsink lets you have a more-or-less arbitrary bin, which could even contain multiple sinks (though that would be rare). > > > 2) When streaming data from shoutcast, it is possible to decode the > StreamTitle= header via icydemux. I have hacked phonon-gstreamer backed > to do this now (in a similar way to how playbin does it). However it > only extracts the title. > > When I use the Xine backend it appears to be able to extract inline > metadata out of the stream content itself (e.g. when it finds a vorbis > comment header or an id3 tag). That's as much as I can tell from > observation. I've hacked the phonon-gstreamer to be able to "guess" the > artist name from the StreamTitle but it's really not very nice. So my > main question is, can the decoders handle inline metadata decoding and > if so how can I enabled this? > I guess it'd be useful to point us at a stream that behaves like this. 'icydemux' will extract metadata properly from shoutcast streams (so long as the source is providing this data). Vorbis streams have inline metadata that should be handled automatically, without anything special being needed. I've never seen inline id3 being streamed, nor have I ever seen software that would handle it at all. So you shouldn't need to do anything special apart from requesting the http sources to get shoutcast-style metadata. Mike From ensonic at hora-obscura.de Mon Sep 22 20:08:43 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Mon, 22 Sep 2008 21:08:43 +0300 Subject: [gst-devel] Building gstreamer+sdl program In-Reply-To: <20080918090832.M22120@mail2.sysu.edu.cn> References: <20080918090832.M22120@mail2.sysu.edu.cn> Message-ID: <48D7DF2B.2030904@hora-obscura.de> Chen Hailiang schrieb: > Hi all: > I'm using gstreamer and sdl to make a very sample vedio player.I put all > the codes in one file; video_test.c. > My question is: how can i compile it? I tried some commands like"gcc -Wall > `sdl-config --libs gstreamer-0.10` video_test.c -o video_player",it didn't > work.Can someone can help? > gcc -Wall `pkg-config --cflags --libs gstreamer-0.10` video_test.c -o video_player and also show us the errors, how else should we know what went wrong? Stefan > Chen > -- > Best regards > > > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From jpuydt at free.fr Mon Sep 22 21:44:15 2008 From: jpuydt at free.fr (Julien Puydt) Date: Mon, 22 Sep 2008 21:44:15 +0200 Subject: [gst-devel] Video input with various systems Message-ID: <48D7F58F.30903@free.fr> Hi, I have noticed that for linux-based systems, there are v4l2src and dv1394src : I added support to them in ekiga (well, in a personal post3 branch...). For win32, it seems ksvideosrc is what I want (except it doesn't have property probe). For osx, there's a directory sys/osxvideo in gst-plugins-bad and gst-plugins, but it doesn't have sources... at least not in my checkout, and gst-plugins-good's sys/osxvideo has only a sink. For *BSD: I have no clue what they have. Thanks, Snark on freenode&gimpnet From jeff_barish at earthlink.net Tue Sep 23 05:13:06 2008 From: jeff_barish at earthlink.net (Jeffrey Barish) Date: Mon, 22 Sep 2008 21:13:06 -0600 Subject: [gst-devel] Lame quality vs. bitrate Message-ID: Does anyone know of a chart that gives approximate bitrates for the various quality factors and vbr/cbr? -- Jeffrey Barish From jag.lnx at gmail.com Tue Sep 23 08:43:35 2008 From: jag.lnx at gmail.com (jagadees r) Date: Tue, 23 Sep 2008 12:13:35 +0530 Subject: [gst-devel] C code to gstreamer plugin conversion In-Reply-To: <48D7A872.4030404@hora-obscura.de> References: <006001c91965$ee171050$26033c0a@telxsi.com> <48D7A872.4030404@hora-obscura.de> Message-ID: <49ba0cd10809222343ic69359cy2e324267678d108c@mail.gmail.com> Please have a look at this if you have not seen already: http://code.google.com/p/gst-player/downloads/list -Jagadees On Mon, Sep 22, 2008 at 7:45 PM, Stefan Kost wrote: > hi, > > arnabsamanta schrieb: >> Hi >> i have an application which has a number of C++ files and i need to make >> the application as a gstreamer plug in. >> what are the basic steps to do so ? >> i know how to create a source and a sink pad. >> do i need to declare all functions of all the C++ codes within the init >> functions ? >> > Please read the plugin writers guide. Also have a look at the > gst-plugins-* packages which has a few c++ based plugins (just search > for *.cc) > > Stefan > >> regards, >> ~Arnab >> >> >> >> The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's challenge >> Build the coolest Linux based applications with Moblin SDK & win great prizes >> Grand prize is a trip for two to an Open Source event anywhere in the world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ajitjohn at tataelxsi.co.in Tue Sep 23 09:55:31 2008 From: ajitjohn at tataelxsi.co.in (ajitjohn) Date: Tue, 23 Sep 2008 13:25:31 +0530 Subject: [gst-devel] Query regarding "rtpsend" element Message-ID: <005401c91d51$c1077e20$68033c0a@telxsi.com> Hii all, Am facing a small problem regarding "rtpsend" element in "gst-plugins-farsight-0.12.1" package. There is no "ip" property in this element. Can anybody please tell me in which package does this element exist which has the "ip" property. Kindly help. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From ensonic at hora-obscura.de Tue Sep 23 14:46:39 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 23 Sep 2008 15:46:39 +0300 Subject: [gst-devel] C code to gstreamer plugin conversion In-Reply-To: <003101c91d2e$337a3d40$26033c0a@telxsi.com> References: <003101c91d2e$337a3d40$26033c0a@telxsi.com> Message-ID: <48D8E52F.9050005@hora-obscura.de> hi, please leave the discussion on the list. arnabsamanta schrieb: > Hi Stefan, > thanks a lot for your reply. > i have gone through the pwg. and i know the basics of plug ins. > i just have a small confusion. when i already have a C++ code with a number > of functions calling each other , so in the _init() function do i need to > initialize only the main calling function of the code ? in that case can i > name and treat the function as the chain function ? > All gstreamer elements are gobjects. You want to read the first chapter in gobject api docs as well. a gobject has a lifecycle. The _init() function is called once for each new instance. A _chain() function is used to process chunks of data. You need to implement a gobject and call your c++ functions at the appropriate place. In _init() you would e.g. do a new() of the c++ class and store the instance in the gobject instance. But as I said, please check the existing c++ plugins. Stefan > thanks and regards, > ~Arnab > > > -----Original Message----- > From: Stefan Kost [mailto:ensonic at hora-obscura.de] > Sent: Monday, September 22, 2008 7:45 PM > To: arnabsamanta at tataelxsi.co.in; Discussion of the development of > GStreamer > Subject: Re: [gst-devel] C code to gstreamer plugin conversion > > > hi, > > arnabsamanta schrieb: > >> Hi >> i have an application which has a number of C++ files and i need to make >> the application as a gstreamer plug in. >> what are the basic steps to do so ? >> i know how to create a source and a sink pad. >> do i need to declare all functions of all the C++ codes within the init >> functions ? >> >> > Please read the plugin writers guide. Also have a look at the > gst-plugins-* packages which has a few c++ based plugins (just search > for *.cc) > > Stefan > > >> regards, >> ~Arnab >> >> >> >> The information contained in this electronic message and any attachments >> > to this message are intended for the exclusive use of the addressee(s) and > may contain proprietary, confidential or privileged information. If you are > not the intended recipient, you should not disseminate, distribute or copy > this e-mail. Please notify the sender immediately and destroy all copies of > this message and any attachments contained in it. > >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> > challenge > >> Build the coolest Linux based applications with Moblin SDK & win great >> > prizes > >> Grand prize is a trip for two to an Open Source event anywhere in the >> > world > >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. > From andreas.wagner at fh-campuswien.ac.at Tue Sep 23 14:40:08 2008 From: andreas.wagner at fh-campuswien.ac.at (Andreas Wagner) Date: Tue, 23 Sep 2008 14:40:08 +0200 Subject: [gst-devel] udpsink qos-dscp Message-ID: <48D8E3A8.9030400@fh-campuswien.ac.at> I am using gst-plugins-good-0.10.10. When I start a pipeline like: gst-launch fakesrc ! udpsink host="192.168.0.14" port=16666 qos-dscp=13 following error occurs: (gst-launch-0.10:7214): GLib-GObject-CRITICAL **: g_value_get_uint: assertion `G_VALUE_HOLDS_UINT (value)' failed could someone give me a reason, or check if it is a bug. The Error occurs because of qos-dscp=13, but i need dcsp for my application. Thanks and cheers Andy -------------- next part -------------- A non-text attachment was scrubbed... Name: andreas_wagner.vcf Type: text/x-vcard Size: 350 bytes Desc: not available URL: From jeff_barish at earthlink.net Tue Sep 23 18:35:58 2008 From: jeff_barish at earthlink.net (Jeffrey Barish) Date: Tue, 23 Sep 2008 10:35:58 -0600 Subject: [gst-devel] Lame Message-ID: If the vbr property is set to 4 (so vbr is on), does setting the bitrate property have any effect? It seems to me that it should not because the bitrate property is related to cbr mode. Also, is it true that the quality property applies only to cbr mode and the vbr-quality property applies to vbr mode? -- Jeffrey Barish From wim.taymans at gmail.com Tue Sep 23 19:36:55 2008 From: wim.taymans at gmail.com (Wim Taymans) Date: Tue, 23 Sep 2008 19:36:55 +0200 Subject: [gst-devel] udpsink qos-dscp In-Reply-To: <48D8E3A8.9030400@fh-campuswien.ac.at> References: <48D8E3A8.9030400@fh-campuswien.ac.at> Message-ID: <1222191416.7301.22.camel@metal> On Tue, 2008-09-23 at 14:40 +0200, Andreas Wagner wrote: > I am using gst-plugins-good-0.10.10. > When I start a pipeline like: > > gst-launch fakesrc ! udpsink host="192.168.0.14" port=16666 qos-dscp=13 > > following error occurs: > > (gst-launch-0.10:7214): GLib-GObject-CRITICAL **: g_value_get_uint: > assertion `G_VALUE_HOLDS_UINT (value)' failed > > could someone give me a reason, or check if it is a bug. It was a bug that is now fixed in CVS. Wim > > The Error occurs because of qos-dscp=13, but i need dcsp for my application. > > Thanks and cheers > Andy > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From vlyamtsev at gmail.com Tue Sep 23 19:42:27 2008 From: vlyamtsev at gmail.com (Victor lyamtsev) Date: Tue, 23 Sep 2008 13:42:27 -0400 Subject: [gst-devel] application feasibility question Message-ID: <76224b100809231042q5e04cedfmcede63f27922903e@mail.gmail.com> Hello, I am working on H264 video streaming server. The main feature - it has to be able to manipulate x264 codec parameters ( quantization step, specifically... ) at run time. Can Gstreamer possibly be used as platform for this app? Thank you, -vl -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Tue Sep 23 19:55:50 2008 From: wim.taymans at gmail.com (Wim Taymans) Date: Tue, 23 Sep 2008 19:55:50 +0200 Subject: [gst-devel] application feasibility question In-Reply-To: <76224b100809231042q5e04cedfmcede63f27922903e@mail.gmail.com> References: <76224b100809231042q5e04cedfmcede63f27922903e@mail.gmail.com> Message-ID: <1222192550.7301.25.camel@metal> On Tue, 2008-09-23 at 13:42 -0400, Victor lyamtsev wrote: > Hello, > I am working on H264 video streaming server. The main feature - it has > to be able to manipulate x264 codec parameters ( quantization step, > specifically... ) at run time. Can Gstreamer possibly be used as > platform for this app? yes Wim > Thank you, > -vl > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From julien.isorce at gmail.com Wed Sep 24 00:30:22 2008 From: julien.isorce at gmail.com (Julien Isorce) Date: Wed, 24 Sep 2008 00:30:22 +0200 Subject: [gst-devel] G_LOG_DOMAIN, GST_CAT_DEFAULT, vc8 (vs2005) Message-ID: <180a127d0809231530l633b4903ha809afba9ed5600@mail.gmail.com> Hi, Some news about the problem I was talking on IRC yesterday. About G_LOG_DOMAIN ( with vc8 ) which was always set to 0 (same about GST_CAT_DEFAULT) and so I could not use severals domains or categories. After some investigations it seems that since 11-Mar-2008, the file * glibconfig.h* introduces: /* varargs macros available since msvc8 (vs2005) */ # if _MSC_VER >= 1400 # define G_HAVE_ISO_VARARGS 1 # endif Ok so then I have upgraded my win32 glib dev package and then I can use several G_LOG_DOMAINs. So that's cool. But it caused some "error C2065: '__func__' : undeclared identifier" for GST_DEBUGs. So I have added: #define __func__ __FUNCTION__ in *glibconfig.h*: /* varargs macros available since msvc8 (vs2005) */ # if _MSC_VER >= 1400 # define G_HAVE_ISO_VARARGS 1 # define __func__ __FUNCTION__ # endif Then I can also use several gst custom categories. ( Waw, I could not, with vc8, for a long time) But I am not sure to define __func__ as __FUNCTION__ in the best place (glibconfig.h). And so I would like to know how the gstreamer developpers do on win32 with vc8 ? Sincerely Julien -------------- next part -------------- An HTML attachment was scrubbed... URL: From sachinpandhare at gmail.com Wed Sep 24 06:00:22 2008 From: sachinpandhare at gmail.com (Sachin Pandhare) Date: Wed, 24 Sep 2008 09:30:22 +0530 Subject: [gst-devel] Fwd: changing volume using alsamixer In-Reply-To: <72cf309c0809190505q55ed719bn59c438135a0519f@mail.gmail.com> References: <72cf309c0809190505q55ed719bn59c438135a0519f@mail.gmail.com> Message-ID: <72cf309c0809232100j433bda4n89121dd8f3744eab@mail.gmail.com> I think this email didn't reach the list. so sending again. ---------- Forwarded message ---------- From: Sachin Pandhare Date: Fri, Sep 19, 2008 at 5:35 PM Subject: changing volume using alsamixer To: gstreamer-devel Hi, i executed following commands gst-launch audiotestsrc ! alsasink & amixer set Master 10%+ it changed the volume for the test source playback. Does amixer use alsamixer plugin internally? is there any command for gst-launch to include alsamixer in the chain with some preset volume? can this volume be changed dynamically? thanks, Sachin -------------- next part -------------- An HTML attachment was scrubbed... URL: From irfanshaikh at tataelxsi.co.in Wed Sep 24 06:41:56 2008 From: irfanshaikh at tataelxsi.co.in (irfanshaikh) Date: Wed, 24 Sep 2008 10:11:56 +0530 Subject: [gst-devel] ffmpeg's ffmux_asf In-Reply-To: <1221493366.6848.75.camel@metal> Message-ID: <005a01c91dff$e0687d60$37033c0a@telxsi.com> Hi all, I am trying to mux the various audio/video elementry streams using ffmux_asf. I used following pipeline. 1)gst-launch filesrc location=/root/Desktop/teststreams/mpeg4/mpeg4_mp3.ASF ! ffdemux_asf name=demux demux.audio_ 00 ! queue ! mux.audio_0 demux.video_00 ! queue ! mux.video_0 ffmux_asf name=mux ! filesink location=/root/Desktop/test2.asf ffmpeg (FFmpeg MJPEG decoder) mp3lib (mp3lib MPEG layer-2, layer-3 2)gst-launch audiotestsrc ! ffenc_mp3 ! queue ! mux.audio_0 videotestsrc ! ffenc_mpeg4 ! queue ! mux.video_0 ffmux_asf name=mux ! filesink location=/root/Desktop/Muxed.asf 3)gst-launch filesrc location=/root/Desktop/teststreams/mpeg4/mpeg4_wma.ASF ! ffdemux_asf ! ffmux_asf ! filesink location=/root/Desktop/test4.asf ffmpeg (FFmpeg M$ MPEG-4 v2) ffmpeg (DivX audio v2 (FFmpeg)) I think there is something wrong in my pipeline structure. Can any one help me to find out how does ffmux_asf works for both audo, video elementry streams {videotestsrc and audiotestsrc}. I would be very thankful to you all if u explaing me with videotestsrc and audiotestsrc how does ffmux_asf works. Regards, Irfan. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From liangzhihong1984 at 126.com Wed Sep 24 09:50:39 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Wed, 24 Sep 2008 15:50:39 +0800 (CST) Subject: [gst-devel] Problems with more xvimagesink Message-ID: <4809545.294741222242639502.JavaMail.coremail@bj126app62.126.com> I played two local .ts files with xvimagesink, when the first pipeline started, it worked in order. However, when i start the second pipeline, the output window didn't appear, and error messages are like this: gst_xvimagesink_get_xv_support():/pipeline0/xvimagesink0:No port available It means that I could not open two xv play windows at the same time, then what should I do? Any of your help would be appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien at moutte.net Wed Sep 24 10:06:20 2008 From: julien at moutte.net (Julien Moutte) Date: Wed, 24 Sep 2008 10:06:20 +0200 Subject: [gst-devel] Problems with more xvimagesink In-Reply-To: <4809545.294741222242639502.JavaMail.coremail@bj126app62.126.com> References: <4809545.294741222242639502.JavaMail.coremail@bj126app62.126.com> Message-ID: <48D9F4FC.1050903@moutte.net> If you only have a single XV port you should just use ximagesink when you one the XV port is in use. The most sensible way of doing that is to use autovideosink. Best regards, Julien Moutte, FLUENDO S.A. liangzhihong1984 wrote: > I played two local .ts files with xvimagesink, when the first > pipeline started, it worked in order. However, when i start the second > pipeline, the output window didn't appear, and error messages are like this: > gst_xvimagesink_get_xv_support():/pipeline0/xvimagesink0:No port available > > It means that I could not open two xv play windows at the same time, > then what should I do? > Any of your help would be appreciated. > > > > > > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From dragos.cirjan at gmail.com Wed Sep 24 10:53:34 2008 From: dragos.cirjan at gmail.com (Dragos Cirjan) Date: Wed, 24 Sep 2008 11:53:34 +0300 Subject: [gst-devel] need help for debugging a plugin Message-ID: <2f0090140809240153g69d0eb27u38b2511e570d246d@mail.gmail.com> Hi guys. I need some help with a plugin I wrote. It's a CURL based source plugin. I know I'm reinventing the wheel but, I really need to finish this plugin and use it. I can add a link in my next reply, but I need to know if anyone is willing to help me. The plugin works just fine, but if I need to make it repeat the request the buffer just goes crazy. Here is the code that might interest you for a start (just ignore the g_print() ): static GstFlowReturn gst_curlsrc_create (GstPushSrc * psrc, GstBuffer ** outbuf) { GstCUrlSrc *src; src = GST_CURLSRC (psrc); g_print("curl_create_start\n"); if (src->buf_total) { if (!src->repeatread) { return GST_FLOW_UNEXPECTED; } //* else { src->buf_size = 0; return GST_FLOW_OK; }//*/ } src->outbuf = outbuf; if (src->curl) { curl_easy_perform(src->curl); } else return GST_FLOW_ERROR; if (src->repeatread) gst_buffer_ref(src->outbuf[0]); g_print("curl_create_end\n"); return GST_FLOW_OK; } -- ----------------------------------------------------------------- Cristian - Dragos, Cirjan ----------------------------------------------------------------- Email: dragos.cirjan at yahoo.com Email: dragos.cirjan at itmediaconnect.ro, doru at bocancul-literar.ro Telefon: +40726355762 -------------- next part -------------- An HTML attachment was scrubbed... URL: From liangzhihong1984 at 126.com Wed Sep 24 11:16:37 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Wed, 24 Sep 2008 17:16:37 +0800 (CST) Subject: [gst-devel] Can I play more than one video with xvimagesink on one computer? Message-ID: <33545899.354251222247797619.JavaMail.coremail@bj126app84.126.com> Can anyone give me some help? -------------- next part -------------- An HTML attachment was scrubbed... URL: From fthiery at gmail.com Wed Sep 24 11:18:58 2008 From: fthiery at gmail.com (Florent) Date: Wed, 24 Sep 2008 11:18:58 +0200 Subject: [gst-devel] Fwd: changing volume using alsamixer In-Reply-To: <72cf309c0809232100j433bda4n89121dd8f3744eab@mail.gmail.com> References: <72cf309c0809190505q55ed719bn59c438135a0519f@mail.gmail.com> <72cf309c0809232100j433bda4n89121dd8f3744eab@mail.gmail.com> Message-ID: <1efe3a6e0809240218g6a8452aeh76903cfee542af91@mail.gmail.com> Hi Hi, > i executed following commands > gst-launch audiotestsrc ! alsasink & > amixer set Master 10%+ > > it changed the volume for the test source playback. > > Does amixer use alsamixer plugin internally? > is there any command for gst-launch to include alsamixer in the chain with > some preset volume? > can this volume be changed dynamically? > I'd rather use the following pipeline, programatically: audiotestsrc ! volume ! alsasink And play with volume's "volume" property, example in python: vol = gst.element_factory_make("volume") vol.set_property("volume", 0.8) Cheers Florent -------------- next part -------------- An HTML attachment was scrubbed... URL: From liangzhihong1984 at 126.com Wed Sep 24 11:38:03 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Wed, 24 Sep 2008 17:38:03 +0800 (CST) Subject: [gst-devel] How can I get number of the xv port? Message-ID: <29388467.368461222249083866.JavaMail.coremail@bj126app84.126.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: From sameer at nextbitcpu.com Wed Sep 24 12:56:58 2008 From: sameer at nextbitcpu.com (Sameer Naik) Date: Wed, 24 Sep 2008 16:26:58 +0530 Subject: [gst-devel] How can I get number of the xv port? In-Reply-To: <29388467.368461222249083866.JavaMail.coremail@bj126app84.126.com> References: <29388467.368461222249083866.JavaMail.coremail@bj126app84.126.com> Message-ID: <200809241626.58880.sameer@nextbitcpu.com> This is how: [user at localhost ~]$ xvinfo | grep ports number of ports: 16 Regards ~Sameer From sameer at nextbitcpu.com Wed Sep 24 13:18:18 2008 From: sameer at nextbitcpu.com (Sameer Naik) Date: Wed, 24 Sep 2008 16:48:18 +0530 Subject: [gst-devel] How can I get number of the xv port? Message-ID: <200809241648.18251.sameer.subscriptions@damagehead.com> This is how: [user at localhost ~]$ xvinfo | grep ports number of ports: 16 Regards ~Sameer From ensonic at hora-obscura.de Wed Sep 24 13:54:58 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Wed, 24 Sep 2008 14:54:58 +0300 Subject: [gst-devel] Using GstAppSink In-Reply-To: <48D5ED39.8010104@free.fr> References: <48D5558A.4010300@free.fr> <48D5ED39.8010104@free.fr> Message-ID: <48DA2A92.8080802@hora-obscura.de> Julien Puydt schrieb: > Julien Puydt a ?crit : > >> Hi, >> >> I'm trying to use GstAppSink in my ekiga code -- however I don't know >> exactly how to make those pipelines work : >> >> gst-launch videotestsrc ! appsink max_buffers=2 drop=true >> caps=video/x-raw-yuv,format=\(fourcc\)I420,width=320,height=240,framerate=30 >> name=ekiga_sink >> >> gst-launch v4lsrc ! appsink max_buffers=2 drop=true >> caps=video/x-raw-yuv,format=\(fourcc\)I420,width=320,height=240,framerate=30 >> name=ekiga_sink >> >> I'm pretty sure I lack some magic in-between my source and my sink... >> > > It seems the "framerate=30" is the problem... > yep, its a fraction and should be given as: caps=...,framerate='(fraction)'30/1 Stefan > Snark on #gstreamer > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ensonic at hora-obscura.de Wed Sep 24 13:52:41 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Wed, 24 Sep 2008 14:52:41 +0300 Subject: [gst-devel] Seeking multifilesrc In-Reply-To: <1221755299.16314.22.camel@matroskin> References: <1221755299.16314.22.camel@matroskin> Message-ID: <48DA2A09.5040107@hora-obscura.de> Jorge schrieb: > Hi to all. > > I'm trying to write a stop-motion creation app, in the vein of > http://developer.skolelinux.no/info/studentgrupper/2005-hig-stopmotion/index.php > but using python, gtk, and well, gstreamer. It's my first using all > three technologies, tough :p > What i'm stump on is on how step frame by frame forward and backwards in > a video loaded trough multifilesrc. I was trying to seek using > gst.FORMAT_BUFFERS as to move one frame at a time, but it doesn't work. > When i attempt a query_position on the pipeline, i get a "query failed". > Can you try |GST_FORMAT_DEFAULT? Stefan | > My pipeline is > > multifilesrc ! jpegdec ! cspvideo !videosink. > > (multifilesrc has a "image/jpeg,framerate=15/1" cap for correct playing) > > I've tried a queue right after multifilesrc, but no difference. Is there > something i have to do for a stream be seekable? Also, the idea is being > able to keep adding image frames without having to reprocess everything > each time... > > Any pointer? > > Jorge > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ensonic at hora-obscura.de Wed Sep 24 13:59:19 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Wed, 24 Sep 2008 14:59:19 +0300 Subject: [gst-devel] querry : copying an ordinary buffer to GstBuffer In-Reply-To: <785339900809220347q2992aef8r72bceaade9a8059a@mail.gmail.com> References: <00ef01c91c9d$c51487b0$26033c0a@telxsi.com> <785339900809220347q2992aef8r72bceaade9a8059a@mail.gmail.com> Message-ID: <48DA2B97.2050901@hora-obscura.de> hi, sudarshan bisht schrieb: > > Hi , > Try following , would work , > > get_data(void *buffer, size_t frame) > { > GstBuffer *gst_buffer ; > gst_buffer = gst_buffer_new_and_alloc (frame); > memcpy(GST_BUFFER_DATA(gst_buffer) , buffer, frame); > } you should also try to avoid the copy if you can (if you can take ownership and if the buffer can be freed with g_free()): gst_buffer = gst_buffer_new() GST_BUFFER_MALLOCDATA(gst_buffer) = GST_BUFFER_DATA(gst_buffer) = buffer; GST_BUFFER_SIZE(gst_buffer) = frame; Stefan > > > > On Mon, Sep 22, 2008 at 3:57 PM, arnabsamanta > > > wrote: > > Hi , > how can i copy the contents of an ordinary buffer created > in C++ to the > GstBuffer buffer. > i have the following.... can any body tell me if its correct > > get_data(void *buffer, size_t frame) > { > GstBuffer *gst_buffer ; > gst_buffer = gst_buffer_new_and_alloc (frame); > memcpy(gst_buffer , buffer, frame); > } > > where buffer is the memory segment created by the previous > function by > "new" > > > i did so as i guess , > gst_buffer = GST_BUFFER_CAST(buffer); > or > gst_buffer = GST_BUFFER_DATA(buffer) > > wont work because "buffer" is not a n GST type buffer. > > is my understanding as stated above is correct ? > if wrong , can any body tell me the correct way ? > > regards, > ~Arnab > > > The information contained in this electronic message and any > attachments to this message are intended for the exclusive use of > the addressee(s) and may contain proprietary, confidential or > privileged information. If you are not the intended recipient, you > should not disseminate, distribute or copy this e-mail. Please > notify the sender immediately and destroy all copies of this > message and any attachments contained in it. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win > great prizes > Grand prize is a trip for two to an Open Source event anywhere in > the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > Regards, > > Sudarshan Bisht > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ensonic at hora-obscura.de Wed Sep 24 14:03:22 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Wed, 24 Sep 2008 15:03:22 +0300 Subject: [gst-devel] Video input with various systems In-Reply-To: <48D7F58F.30903@free.fr> References: <48D7F58F.30903@free.fr> Message-ID: <48DA2C8A.1040201@hora-obscura.de> hi, Julien Puydt schrieb: > Hi, > > I have noticed that for linux-based systems, there are v4l2src and > dv1394src : I added support to them in ekiga (well, in a personal post3 > branch...). > > For win32, it seems ksvideosrc is what I want (except it doesn't have > property probe). > > For osx, there's a directory sys/osxvideo in gst-plugins-bad and > gst-plugins, but it doesn't have sources... at least not in my checkout, > and gst-plugins-good's sys/osxvideo has only a sink. > > For *BSD: I have no clue what they have. > It would totally rock if you could have a go at writing a autovideosrc that does this automatically. Have a look at gst-plugins-good/gst/autodetect/ Stefan > Thanks, > > Snark on freenode&gimpnet > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From Vinod_Nanjaiah at mindtree.com Wed Sep 24 14:22:58 2008 From: Vinod_Nanjaiah at mindtree.com (Vinod Nanjaiah) Date: Wed, 24 Sep 2008 17:52:58 +0530 Subject: [gst-devel] using mulaw decoder Message-ID: <174230991E95D743B0C91DF471EF44E84275B46D0C@MTW02MSG02.mindtree.com> Hi, I am trying to playback a mulaw audio file using the following pipeline construct: gst-launch-0.10 filesrc location=$1 ! mulawdec ! osssink But, I am getting this error: ERROR default mulaw-decode.c:199:gst_mulawdec_chain: no format negotiated Is there any other element that I need to include in the pipeline? I tried using audioconvert (after mulawdec), but got the same error. what could be the problem? BTW, I am trying this on Da-vinci platform. Best Regards, Vinod Pura ________________________________ This message (including attachment if any) is confidential and may be privileged. If you have received this message by mistake please notify the sender by return e-mail and delete this message from your system. Any unauthorized use or dissemination of this message in whole or in part is strictly prohibited. E-mail may contain viruses. Before opening attachments please check them for viruses and defects. While MindTree Limited (MindTree) has put in place checks to minimize the risks, MindTree will not be responsible for any viruses or defects or any forwarded attachments emanating either from within MindTree or outside. Please note that e-mails are susceptible to change and MindTree shall not be liable for any improper, untimely or incomplete transmission. MindTree reserves the right to monitor and review the content of all messages sent to or from MindTree e-mail address. Messages sent to or from this e-mail address may be stored on the MindTree e-mail system or else where. -------------- next part -------------- An HTML attachment was scrubbed... URL: From thiagossantos at gmail.com Wed Sep 24 14:58:19 2008 From: thiagossantos at gmail.com (thiagoss) Date: Wed, 24 Sep 2008 09:58:19 -0300 Subject: [gst-devel] using mulaw decoder In-Reply-To: <174230991E95D743B0C91DF471EF44E84275B46D0C@MTW02MSG02.mindtree.com> References: <174230991E95D743B0C91DF471EF44E84275B46D0C@MTW02MSG02.mindtree.com> Message-ID: Maybe adding capsfilter after filesrc would work (or typefind) On Wed, Sep 24, 2008 at 9:22 AM, Vinod Nanjaiah wrote: > Hi, > > I am trying to playback a mulaw audio file using the following pipeline > construct: > > > > *gst-launch-0.10 filesrc location=$1 ! mulawdec ! osssink * > > > > But, I am getting this error: > > > > *ERROR default mulaw-decode.c:199:gst_mulawdec_chain: no format > negotiated* > > > > Is there any other element that I need to include in the pipeline? > > I tried using audioconvert (after mulawdec), but got the same error. > > > > what could be the problem? > > BTW, I am trying this on Da-vinci platform. > > > > Best Regards, > > Vinod Pura > > > > > > ------------------------------ > This message (including attachment if any) is confidential and may be > privileged. If you have received this message by mistake please notify the > sender by return e-mail and delete this message from your system. Any > unauthorized use or dissemination of this message in whole or in part is > strictly prohibited. E-mail may contain viruses. Before opening attachments > please check them for viruses and defects. While MindTree Limited (MindTree) > has put in place checks to minimize the risks, MindTree will not be > responsible for any viruses or defects or any forwarded attachments > emanating either from within MindTree or outside. > > Please note that e-mails are susceptible to change and MindTree shall not > be liable for any improper, untimely or incomplete transmission. > > MindTree reserves the right to monitor and review the content of all > messages sent to or from MindTree e-mail address. Messages sent to or from > this e-mail address may be stored on the MindTree e-mail system or else > where. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) -------------- next part -------------- An HTML attachment was scrubbed... URL: From liangzhihong1984 at 126.com Wed Sep 24 15:19:35 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Wed, 24 Sep 2008 21:19:35 +0800 (CST) Subject: [gst-devel] About xv ports Message-ID: <32294104.453141222262375493.JavaMail.coremail@bj126app46.126.com> There is only one xv port on my PC, does this related with my video card or something else? -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Wed Sep 24 15:33:29 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Wed, 24 Sep 2008 16:33:29 +0300 Subject: [gst-devel] About xv ports In-Reply-To: <32294104.453141222262375493.JavaMail.coremail@bj126app46.126.com> References: <32294104.453141222262375493.JavaMail.coremail@bj126app46.126.com> Message-ID: <48DA41A9.9050101@hora-obscura.de> hi, yes, your video card has just one. E.g. the ati chip on laptop has 1 only, my nvidea card in my desktop has 10. If I use xgl it emulates 32 via opengl. Julien gave you a good advise, use autovideosink. Stefan liangzhihong1984 schrieb: > There is only one xv port on my PC, does this related with my video > card or something else? > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From mediadevel at gmail.com Wed Sep 24 17:50:43 2008 From: mediadevel at gmail.com (john david) Date: Wed, 24 Sep 2008 21:20:43 +0530 Subject: [gst-devel] osssrc using gstreamer Message-ID: Hi, gst-launch-0.10 -v osssrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... ERROR: from element /pipeline0/osssrc0: Could not get/set settings from/on resource. Additional debug info: gstosssrc.c(439): gst_oss_src_prepare (): /pipeline0/osssrc0: Unable to set param SETFRAGMENT: Invalid argument ERROR: pipeline doesn't want to preroll. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... FREEING pipeline ... pls tell what is the solution. -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Wed Sep 24 19:46:33 2008 From: msmith at xiph.org (Michael Smith) Date: Wed, 24 Sep 2008 10:46:33 -0700 Subject: [gst-devel] Can I play more than one video with xvimagesink on one computer? In-Reply-To: <33545899.354251222247797619.JavaMail.coremail@bj126app84.126.com> References: <33545899.354251222247797619.JavaMail.coremail@bj126app84.126.com> Message-ID: <3c1737210809241046nf0b87fbn58cefb20c0e49e37@mail.gmail.com> On Wed, Sep 24, 2008 at 2:16 AM, liangzhihong1984 wrote: > > Can anyone give me some help? If you ask more specific questions, you'll get more useful answers. Anyway: xvimagesink uses the Xv extension. Some hardware only provides a single Xv port (running 'xvinfo' will tell you how many you have), in which case you can only use one xvimagesink at a time. Most modern hardware supports many Xv ports (16 or more), but not all (it's mostly driver-dependent). Mike From dbourgoyne at austin.rr.com Wed Sep 24 20:50:53 2008 From: dbourgoyne at austin.rr.com (dbourgoyne at austin.rr.com) Date: Wed, 24 Sep 2008 13:50:53 -0500 Subject: [gst-devel] GPP + DSPs Message-ID: <19265608.432581222282253669.JavaMail.root@hrndva-web15-z02> Hi all, I'm considering using GStreamer for an application is which GStreamer will span a GPP and multiple DSPs. The GPP and DSPs are connected via PCI. So GStreamer will have to proxy buffers and messages across the PCI bus. Can you point me to documentation, info, case studies, etc. that will help me to architect this solution? Thanks in advance. From vlyamtsev at gmail.com Wed Sep 24 22:56:33 2008 From: vlyamtsev at gmail.com (Victor lyamtsev) Date: Wed, 24 Sep 2008 16:56:33 -0400 Subject: [gst-devel] installation trouble Message-ID: <76224b100809241356h2103ecaagf5b69c2d03f3b433@mail.gmail.com> What's the "core" RPMs I should install? I am trying to build source distrubution ( didn't have much luck with rpms on RedHat EL 4), but have following error: Any idea how to fix it? Thanks, -vl make[4]: Entering directory `/home/mzeal/gstreamer-0.8.10/gst/autoplug' /bin/sh ../../libtool --mode=link --tag=CC gcc -g -O2 -o spidertest spidertest-spidertest.o ../../gst/libgstreamer-0.8.la gcc -g -O2 -o .libs/spidertest spidertest-spidertest.o ../../gst/.libs/ libgstreamer-0.8.so -L/usr/lib/valgrind/x86-linux /usr/lib/libxml2.so -lpthread -lz -lgobject-2.0 -lgthread-2.0 -lgmodule-2.0 -ldl -lglib-2.0 /usr/lib/libpopt.so -lcoregrind -lvex -lgcc -lm -Wl,--rpath -Wl,/usr/local/lib ../../gst/.libs/libgstreamer-0.8.so: undefined reference to `vgPlain_tool_info' -------------- next part -------------- An HTML attachment was scrubbed... URL: From liangzhihong1984 at 126.com Thu Sep 25 07:27:20 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Thu, 25 Sep 2008 13:27:20 +0800 (CST) Subject: [gst-devel] About multi-thread in GStreamer Message-ID: <18749687.205481222320440119.JavaMail.coremail@bj126app33.126.com> I captured video stream from a USB camera, now want to send it to other users with udpsink, and at the same time playback the video stream locally, how should I implement this through gst-launch command line, as well as C language? -------------- next part -------------- An HTML attachment was scrubbed... URL: From irfanshaikh at tataelxsi.co.in Thu Sep 25 07:41:15 2008 From: irfanshaikh at tataelxsi.co.in (Irfan Shaikh) Date: Thu, 25 Sep 2008 11:11:15 +0530 Subject: [gst-devel] About multi-thread in GStreamer References: <18749687.205481222320440119.JavaMail.coremail@bj126app33.126.com> Message-ID: <9D5E1752379A43408015F7FE98466115782795@CHNEXVS01.VSNLXCHANGE.COM> hi, You can try the following approches, 1)Encode the raw elementry stream, use a ASF muxer, try streaming it using udpsink. 2)Or else packetixe the encoded file stream, use rtppayload packetizer and send it using rtpsend element. Regards, Irfan -----Original Message----- From: liangzhihong1984 [mailto:liangzhihong1984 at 126.com] Sent: Thu 9/25/2008 10:57 AM To: gstreamer-devel at lists.sourceforge.net Subject: [gst-devel] About multi-thread in GStreamer I captured video stream from a USB camera, now want to send it to other users with udpsink, and at the same time playback the video stream locally, how should I implement this through gst-launch command line, as well as C language? This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mediadevel at gmail.com Thu Sep 25 09:09:50 2008 From: mediadevel at gmail.com (john david) Date: Thu, 25 Sep 2008 12:39:50 +0530 Subject: [gst-devel] RE : how to set/get settings from/on resource Message-ID: Hi, I want to record sound, from my sound card using OSS and encode it to an Ogg/Vorbis file using gstreamer. I used btaudio driver. *Command* that used : gst-launch-0.10 -v osssrc device=/dev/dsp ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg For this, i got *error message*, from element /pipeline0/osssrc0: Could not get/set settings from/on resource. and the detailed error messages are: Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... ERROR: from element /pipeline0/osssrc0: Could not get/set settings from/on resource. Additional debug info: gstosssrc.c(439): gst_oss_src_prepare (): /pipeline0/osssrc0: Unable to set param SETFRAGMENT: Invalid argument ERROR: pipeline doesn't want to preroll. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... FREEING pipeline ... and how to set/get settings from/on resource. But i 'm able to capture audio data using (audio device /dev/dsp for oss ) some other application tools and play it without any errors. Regards, John david. -------------- next part -------------- An HTML attachment was scrubbed... URL: From volter619 at 163.com Thu Sep 25 09:57:20 2008 From: volter619 at 163.com (Volter Yen) Date: Thu, 25 Sep 2008 15:57:20 +0800 (CST) Subject: [gst-devel] gstreamer dvb a/v sync problem Message-ID: <22575222.271371222329440264.JavaMail.coremail@bj163app93.163.com> hi all, when using gstreamer to play dvb signal, if I set the osssink's sync to true, the pipelie give 'Unexpected discontinuity in audio timestamps of more than half a second(...) resyncing ' and stalled there....but it is said that 'sync=true ' should work , my gstreamer core is 0.10.14,fluendo-mpegdmuex-0.10.15, and by the way my pipelie is: gst-launch-0.10 dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 freq=474000000 code-rate-lp=2/3 code-rate-hp=1/2 guard=8 hierarchy=NONE pids=630:512:128 ! flutsdemux es-pids=650:512 name=demuxer demuxer. ! queue ! mad ! audioconvert ! osssink and if i add 'sync=false' to ossink , it works but when play with video the audio is heavy lagged. I have referrenced to this link https://core.fluendo.com/gstreamer/trac/ticket/46, it seems that these problem should have been resolved. anybody know the way out? thank you Volter -------------- next part -------------- An HTML attachment was scrubbed... URL: From irfanshaikh at tataelxsi.co.in Thu Sep 25 10:52:04 2008 From: irfanshaikh at tataelxsi.co.in (Irfan Shaikh) Date: Thu, 25 Sep 2008 14:22:04 +0530 Subject: [gst-devel] check the fuction References: <22575222.271371222329440264.JavaMail.coremail@bj163app93.163.com> Message-ID: <9D5E1752379A43408015F7FE9846611578279A@CHNEXVS01.VSNLXCHANGE.COM> Hi, What does the following function does? As per my understanding Buffer to hold the output(ffmpegmux->context->filename). but i am not so clear about it g_snprintf (ffmpegmux->context->filename, sizeof (ffmpegmux->context->filename), "gstreamer://%p", ffmpegmux->srcpad); This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. -------------- next part -------------- An HTML attachment was scrubbed... URL: From yongchen at arcsoft.com.cn Thu Sep 25 11:15:30 2008 From: yongchen at arcsoft.com.cn (=?gb2312?B?Q2hlbiBZb25nKEV2ZXIpW7PC08Jd?=) Date: Thu, 25 Sep 2008 17:15:30 +0800 Subject: [gst-devel] How to implement trick mode Message-ID: <5559508B2BD31A44B1C267F4E6EF471D974130@hz-email05.apac.arcsoft.corp> Hi, all, I am developing a plug-in which obtains MPEG2 source and demux functionality. I want to add forward scan and backward scan functionality. When rate is low(< 4.0), we can process it in decoder, but I want to implement scan with high rate, maybe 32x and later. If so, this plug-in need start a new-segment and send to all down-streaming plug-ins, and send data I picture by I picture. But in my side, only 1st I picture is showed, why? The detail is: 1. Application send a segment to pipeline with rate 32.0 and start time is 5 seconds 2. When my plug-in receive this segment event, I find the correspond I picture in 5 second time point. 3. After sent flush-stop event, push this I picture with a BUFFER DISCONT, time stamp is 5 second REMARK: this picture can be shown. 4. Then I push 2nd I picture with a BUFFER DISCONT, time stamp is 30 second, but this picture does not present. REMARK: I using mpeg2dec as my mpeg2 video decoder, I trace it, found it gst_push_pad this 1st picture and not return, why? Any friends have experience about trick mode can teach me, thanks very much. Best Regards, Ever Chen -------------- next part -------------- An HTML attachment was scrubbed... URL: From yongchen at arcsoft.com.cn Thu Sep 25 11:23:38 2008 From: yongchen at arcsoft.com.cn (=?gb2312?B?Q2hlbiBZb25nKEV2ZXIpW7PC08Jd?=) Date: Thu, 25 Sep 2008 17:23:38 +0800 Subject: [gst-devel] video freeze while playing a file without audio Message-ID: <5559508B2BD31A44B1C267F4E6EF471D974136@hz-email05.apac.arcsoft.corp> Hi, all, I am developing a plug-in which integrate source with demux functions, and it always have two src pads, one for video, another for audio, in my pipeline, we connected 2 branches to video sink and audio sink. It is ok to playback a file with audio and video, but if there is no audio data in this file, video freezed. I trace gstreamer code, found audiosink is prerolled status. What I need to do to solve it? Best Regards Ever Chen -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Thu Sep 25 12:38:33 2008 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Thu, 25 Sep 2008 16:08:33 +0530 Subject: [gst-devel] video freeze while playing a file without audio In-Reply-To: <5559508B2BD31A44B1C267F4E6EF471D974136@hz-email05.apac.arcsoft.corp> References: <5559508B2BD31A44B1C267F4E6EF471D974136@hz-email05.apac.arcsoft.corp> Message-ID: <785339900809250338u64261d4atb3bb269af7873ee2@mail.gmail.com> Hi , When u don't have audio data in your container file then that time don't create pad for that stream in plugin code i.e. dont try to push audio data to next element . In your case what is happening is that you dont have audio data even then you are trying to push some junk data to next element and in that case next element is returning some other return value than GST_FLOW_OK from its chain function . To debug this just see return value of gst_pad_push( ) fucntion , this value should be GST_FLOW_OK . 2008/9/25 Chen Yong(Ever)[??] > Hi, all, > > I am developing a plug-in which integrate source with demux > functions, and it always have two src pads, one for video, another for > audio, in my pipeline, we connected 2 branches to video sink and audio sink. > It is ok to playback a file with audio and video, but if there is no audio > data in this file, video freezed. I trace gstreamer code, found audiosink is > prerolled status. What I need to do to solve it? > > > > Best Regards > > Ever Chen > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From kuo-lang.tseng at intel.com Thu Sep 25 02:50:11 2008 From: kuo-lang.tseng at intel.com (Tseng, Kuo-Lang) Date: Wed, 24 Sep 2008 17:50:11 -0700 Subject: [gst-devel] Use of gst_pipeline_set_new_stream_time function Message-ID: <63FEFD5869C1DB49B69DCC57DF8F21574267A320@orsmsx505.amr.corp.intel.com> Hi, I am trying to stream/render content over rtp using gstreamer on two Ubuntu machines. I kept getting the following error and warning on receiver side if I start the receiver side pipeline first. The video will stop rendering after a while: Unexpected discontinuity in audio timestamps of more than half a second (0:00:00.512000000), resyncing GstMessageWarning, gerror=(GstGError)(NULL), debug=(string)"gstbaseaudiosink.c\(1365\):\ gst_base_audio_sink_render\ \(\):\ /pipe/autoaudiosink1/autoaudiosink1-actual-sink-alsa:\012Unexpected\ discontinuity\ in\ audio\ timestamps\ of\ more\ than\ half\ a\ second\ \(0:00:00.512000000\)\,\ resyncing"; bus_cb[WARNING]: from element /pipe/autoaudiosink1/autoaudiosink1-actual-sink-alsa: Compensating for audio synchronisation problems Additional debug info: gstbaseaudiosink.c(1365): gst_base_audio_sink_render (): /pipe/autoaudiosink1/autoaudiosink1-actual-sink-alsa: My sender side command is: gst-launch -v gstrtpbin name=rtpbin filesrc location=file.ts ! \ queue2 max-size-buffers=65535 ! \ mpegtsparse ! \ rtpmp2tpay ! \ queue2 max-size-buffers=65535 ! \ rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=10.3.66.53 port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink host=10.3.66.53 port=5001 sync=false async=false \ udpsrc port=5002 ! rtpbin.recv_rtcp_sink_0 My receiver side is a program which mimics following pipeline: gst-launch -v gstrtpbin name=rtpbin udpsrc port=5000 caps="application/x-rtp,media=video,clock-rate=90000,encoding-name=mpegts" ! \ rtpbin.recv_rtp_sink_0 rtpbin. ! \ rtpmp2tdepay ! \ flutsdemux name=demuxer \ demuxer. ! queue2 max-size-buffers=0 max-size-time=0 ! a52dec ! audioconvert ! volume volume=10 ! autoaudiosink \ demuxer. ! queue2 max-size-buffers=0 max-size-time=0 ! mpeg2dec ! autovideosink \ udpsrc port=5001 ! \ rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! \ udpsink host=10.3.66.198 port=5002 sync=false async=false -t I found that if I added a call to gst_pipeline_set_new_stream_time(bin, 0) in the on-new-ssrc callback function in the program, the problem won't show and video/audio redering works fine (I still started the receiver side program first before the sender side command). What is the consequence of calling this function and is it required to use this function? Any help is appreciated. Thanks Kuo From Ankur_Sharma06 at infosys.com Thu Sep 25 10:13:28 2008 From: Ankur_Sharma06 at infosys.com (Ankur Sharma) Date: Thu, 25 Sep 2008 13:43:28 +0530 Subject: [gst-devel] gstreamer dvb a/v sync problem In-Reply-To: <22575222.271371222329440264.JavaMail.coremail@bj163app93.163.com> References: <22575222.271371222329440264.JavaMail.coremail@bj163app93.163.com> Message-ID: <9C45FA1DEBF6DA448E8492D0A00BFE951882946165@BLRKECMBX05.ad.infosys.com> Hello, I am facing a similar problem. After some debugging, the exact problem I found was: 1. The resynchronization of the Demux Clock using PCR is not happening. Is there any problem with this part of the gstClock code? I am not sure if the add_clock_observation function works fine? Could anyone throw some light on this? Regards Ankur Sharma ________________________________ From: Volter Yen [mailto:volter619 at 163.com] Sent: Thursday, September 25, 2008 1:27 PM To: gstreamer-devel at lists.sourceforge.net Subject: [gst-devel] gstreamer dvb a/v sync problem hi all, when using gstreamer to play dvb signal, if I set the osssink's sync to true, the pipelie give 'Unexpected discontinuity in audio timestamps of more than half a second(...) resyncing ' and stalled there....but it is said that 'sync=true ' should work , my gstreamer core is 0.10.14,fluendo-mpegdmuex-0.10.15, and by the way my pipelie is: gst-launch-0.10 dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 freq=474000000 code-rate-lp=2/3 code-rate-hp=1/2 guard=8 hierarchy=NONE pids=630:512:128 ! flutsdemux es-pids=650:512 name=demuxer demuxer. ! queue ! mad ! audioconvert ! osssink and if i add 'sync=false' to ossink , it works but when play with video the audio is heavy lagged. I have referrenced to this link https://core.fluendo.com/gstreamer/trac/ticket/46, it seems that these problem should have been resolved. anybody know the way out? thank you Volter ________________________________ [??] ???????????????"???"????? **************** CAUTION - Disclaimer ***************** This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful. This e-mail may contain viruses. Infosys has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachment. Infosys reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infosys e-mail system. ***INFOSYS******** End of Disclaimer ********INFOSYS*** -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ankur_Sharma06 at infosys.com Thu Sep 25 12:51:47 2008 From: Ankur_Sharma06 at infosys.com (Ankur Sharma) Date: Thu, 25 Sep 2008 16:21:47 +0530 Subject: [gst-devel] gstreamer dvb a/v sync problem Message-ID: <9C45FA1DEBF6DA448E8492D0A00BFE95188294633A@BLRKECMBX05.ad.infosys.com> Hello, I am facing a similar problem. After some debugging, the exact problem I found was: 1. The resynchronization of the Demux Clock using PCR is not happening. Is there any problem with this part of the gstClock code? I am not sure if the add_clock_observation function works fine? Could anyone throw some light on this? Regards Ankur Sharma ________________________________ From: Volter Yen [mailto:volter619 at 163.com] Sent: Thursday, September 25, 2008 1:27 PM To: gstreamer-devel at lists.sourceforge.net Subject: [gst-devel] gstreamer dvb a/v sync problem hi all, when using gstreamer to play dvb signal, if I set the osssink's sync to true, the pipelie give 'Unexpected discontinuity in audio timestamps of more than half a second(...) resyncing ' and stalled there....but it is said that 'sync=true ' should work , my gstreamer core is 0.10.14,fluendo-mpegdmuex-0.10.15, and by the way my pipelie is: gst-launch-0.10 dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 freq=474000000 code-rate-lp=2/3 code-rate-hp=1/2 guard=8 hierarchy=NONE pids=630:512:128 ! flutsdemux es-pids=650:512 name=demuxer demuxer. ! queue ! mad ! audioconvert ! osssink and if i add 'sync=false' to ossink , it works but when play with video the audio is heavy lagged. I have referrenced to this link https://core.fluendo.com/gstreamer/trac/ticket/46, it seems that these problem should have been resolved. anybody know the way out? thank you Volter ________________________________ [??] ???????????????"???"????? **************** CAUTION - Disclaimer ***************** This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful. This e-mail may contain viruses. Infosys has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachment. Infosys reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infosys e-mail system. ***INFOSYS******** End of Disclaimer ********INFOSYS*** -------------- next part -------------- An HTML attachment was scrubbed... URL: From felipe.contreras at gmail.com Thu Sep 25 14:23:38 2008 From: felipe.contreras at gmail.com (Felipe Contreras) Date: Thu, 25 Sep 2008 15:23:38 +0300 Subject: [gst-devel] GPP + DSPs In-Reply-To: <19265608.432581222282253669.JavaMail.root@hrndva-web15-z02> References: <19265608.432581222282253669.JavaMail.root@hrndva-web15-z02> Message-ID: <94a0d4530809250523j221d5f02u3a98b462bb7496f@mail.gmail.com> On Wed, Sep 24, 2008 at 9:50 PM, wrote: > Hi all, > > I'm considering using GStreamer for an application is which GStreamer will span a GPP and multiple DSPs. The GPP and DSPs are connected via PCI. So GStreamer will have to proxy buffers and messages across the PCI bus. > > Can you point me to documentation, info, case studies, etc. that will help me to architect this solution? What kind of DSP/GPP? We can propose different ways to do it at GStreamer level, but ultimately you would have to communicate with the hardware, is there code that is doing that already? -- Felipe Contreras From liangzhihong1984 at 126.com Thu Sep 25 14:52:35 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Thu, 25 Sep 2008 20:52:35 +0800 (CST) Subject: [gst-devel] How can I capture my screen with GStreamer? Message-ID: <15101986.424291222347155941.JavaMail.coremail@bj126app88.126.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: From ved.kpl at gmail.com Thu Sep 25 15:14:13 2008 From: ved.kpl at gmail.com (ved kpl) Date: Thu, 25 Sep 2008 18:44:13 +0530 Subject: [gst-devel] How can I capture my screen with GStreamer? In-Reply-To: <15101986.424291222347155941.JavaMail.coremail@bj126app88.126.com> References: <15101986.424291222347155941.JavaMail.coremail@bj126app88.126.com> Message-ID: <7496c23f0809250614k5f890d41t6974ecee6be368d3@mail.gmail.com> gst-launch ximagesrc num-buffers=1 ! ffmpegcolorspace ! jpegenc ! filesink location=~/Desktop/cap.jpg -v 2008/9/25 liangzhihong1984 : > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From liangzhihong1984 at 126.com Thu Sep 25 15:38:10 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Thu, 25 Sep 2008 21:38:10 +0800 (CST) Subject: [gst-devel] About screen capture In-Reply-To: References: Message-ID: <30186681.436591222349890965.JavaMail.coremail@bj126app88.126.com> Sorry, exactly ,I mean how to capture and streaming it? ?2008-09-25 21:14:53?gstreamer-devel-request at lists.sourceforge.net ??? >Send gstreamer-devel mailing list submissions to > gstreamer-devel at lists.sourceforge.net > >To subscribe or unsubscribe via the World Wide Web, visit > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >or, via email, send a message with subject or body 'help' to > gstreamer-devel-request at lists.sourceforge.net > >You can reach the person managing the list at > gstreamer-devel-owner at lists.sourceforge.net > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of gstreamer-devel digest..." > > >Today's Topics: > > 1. Re: gstreamer dvb a/v sync problem (Ankur Sharma) > 2. Re: GPP + DSPs (Felipe Contreras) > 3. How can I capture my screen with GStreamer? (liangzhihong1984) > 4. Re: How can I capture my screen with GStreamer? (ved kpl) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 25 Sep 2008 16:21:47 +0530 >From: Ankur Sharma > >**************** CAUTION - Disclaimer ***************** >This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely >for the use of the addressee(s). If you are not the intended recipient, please >notify the sender by e-mail and delete the original message. Further, you are not >to copy, disclose, or distribute this e-mail or its contents to any other person and >any such actions are unlawful. This e-mail may contain viruses. Infosys has taken >every reasonable precaution to minimize this risk, but is not liable for any damage >you may sustain as a result of any virus in this e-mail. You should carry out your >own virus checks before opening the e-mail or attachment. Infosys reserves the >right to monitor and review the content of all messages sent to or from this e-mail >address. Messages sent to or from this e-mail address may be stored on the >Infosys e-mail system. >***INFOSYS******** End of Disclaimer ********INFOSYS*** >-------------- next part -------------- >An HTML attachment was scrubbed... > >------------------------------ > >Message: 2 >Date: Thu, 25 Sep 2008 15:23:38 +0300 >From: "Felipe Contreras" -------------- next part -------------- An HTML attachment was scrubbed... URL: From liangzhihong1984 at 126.com Thu Sep 25 15:43:23 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Thu, 25 Sep 2008 21:43:23 +0800 (CST) Subject: [gst-devel] How can I capture my screen and streaming it? Message-ID: <5775554.438021222350203231.JavaMail.coremail@bj126app88.126.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: From zaheermerali at gmail.com Thu Sep 25 19:03:51 2008 From: zaheermerali at gmail.com (Zaheer Merali) Date: Thu, 25 Sep 2008 18:03:51 +0100 Subject: [gst-devel] gstreamer dvb a/v sync problem In-Reply-To: <22575222.271371222329440264.JavaMail.coremail@bj163app93.163.com> References: <22575222.271371222329440264.JavaMail.coremail@bj163app93.163.com> Message-ID: <15e616860809251003t7a5d7481gbf1988be6efe8585@mail.gmail.com> 2008/9/25 Volter Yen : > hi all, > when using gstreamer to play dvb signal, if I set the osssink's sync to > true, the pipelie give 'Unexpected discontinuity in audio timestamps of > more than half a second(...) resyncing ' and stalled there....but it is said > that 'sync=true ' should work , my gstreamer core is > 0.10.14,fluendo-mpegdmuex-0.10.15, and by the way my pipelie is: > gst-launch-0.10 dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 > freq=474000000 code-rate-lp=2/3 code-rate-hp=1/2 guard=8 hierarchy=NONE > pids=630:512:128 ! flutsdemux es-pids=650:512 name=demuxer demuxer. ! queue > ! mad ! audioconvert ! osssink > and if i add 'sync=false' to ossink , it works but when play with video > the audio is heavy lagged. > > I have referrenced to this link > https://core.fluendo.com/gstreamer/trac/ticket/46, it seems that these > problem should have been resolved. > anybody know the way out? thank you > > Volter You have a few problems with your pipeline. i) es-pids should not be used with flutsdemux. If you have the pmt pid and the elementary stream pids coming out of dvbsrc then there is no issue. Using es-pids like you are makes it ignore the pcr values contained in the transport stream. ii) The queues you should be using should have max-size-buffers=0 and max-size-time=0 set. You should also be using a more recent version of gstreamer core. Zaheer From kuo-lang.tseng at intel.com Thu Sep 25 19:51:28 2008 From: kuo-lang.tseng at intel.com (Tseng, Kuo-Lang) Date: Thu, 25 Sep 2008 10:51:28 -0700 Subject: [gst-devel] Use of gst_pipeline_set_new_stream_time function Message-ID: <63FEFD5869C1DB49B69DCC57DF8F21574267A877@orsmsx505.amr.corp.intel.com> Hi, I am trying to stream/render content over rtp using gstreamer on two Ubuntu machines. I kept getting the following error and warning on receiver side if I start the receiver side pipeline first. The video will stop rendering after a while: Unexpected discontinuity in audio timestamps of more than half a second (0:00:00.512000000), resyncing GstMessageWarning, gerror=(GstGError)(NULL), debug=(string)"gstbaseaudiosink.c\(1365\):\ gst_base_audio_sink_render\ \(\):\ /pipe/autoaudiosink1/autoaudiosink1-actual-sink-alsa:\012Unexpected\ discontinuity\ in\ audio\ timestamps\ of\ more\ than\ half\ a\ second\ \(0:00:00.512000000\)\,\ resyncing"; bus_cb[WARNING]: from element /pipe/autoaudiosink1/autoaudiosink1-actual-sink-alsa: Compensating for audio synchronisation problems Additional debug info: gstbaseaudiosink.c(1365): gst_base_audio_sink_render (): /pipe/autoaudiosink1/autoaudiosink1-actual-sink-alsa: My sender side command is: gst-launch -v gstrtpbin name=rtpbin filesrc location=file.ts ! \ queue2 max-size-buffers=65535 ! \ mpegtsparse ! \ rtpmp2tpay ! \ queue2 max-size-buffers=65535 ! \ rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=10.3.66.53 port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink host=10.3.66.53 port=5001 sync=false async=false \ udpsrc port=5002 ! rtpbin.recv_rtcp_sink_0 My receiver side is a program which mimics following pipeline: gst-launch -v gstrtpbin name=rtpbin udpsrc port=5000 caps="application/x-rtp,media=video,clock-rate=90000,encoding-name=mpegts" ! \ rtpbin.recv_rtp_sink_0 rtpbin. ! \ rtpmp2tdepay ! \ flutsdemux name=demuxer \ demuxer. ! queue2 max-size-buffers=0 max-size-time=0 ! a52dec ! audioconvert ! volume volume=10 ! autoaudiosink \ demuxer. ! queue2 max-size-buffers=0 max-size-time=0 ! mpeg2dec ! autovideosink \ udpsrc port=5001 ! \ rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! \ udpsink host=10.3.66.198 port=5002 sync=false async=false -t I found that if I added a call to gst_pipeline_set_new_stream_time(bin, 0) in the on-new-ssrc callback function in the program, the problem won't show and video/audio redering works fine (I still started the receiver side program first before the sender side command). What is the consequence of calling this function and is it required to use this function? Any help is appreciated. Thanks Kuo From yongchen at arcsoft.com.cn Fri Sep 26 02:45:21 2008 From: yongchen at arcsoft.com.cn (=?gb2312?B?Q2hlbiBZb25nKEV2ZXIpW7PC08Jd?=) Date: Fri, 26 Sep 2008 08:45:21 +0800 Subject: [gst-devel] video freeze while playing a file without audio In-Reply-To: Message-ID: <5559508B2BD31A44B1C267F4E6EF471D974180@hz-email05.apac.arcsoft.corp> Thanks your reply. In my example, we don't push any audio data to down-streaming element, only push video data, but video is freeezed. >Hi , > > When u don't have audio data in your container file then that time >don't create pad for that stream in plugin code i.e. dont try to push audio >data to next element . > In your case what is happening is that you dont have audio data even >then you are trying to push some junk data to next element and in that case >next element is returning some other return value than GST_FLOW_OK from its >chain function . > To debug this just see return value of gst_pad_push( ) fucntion , >this value should be GST_FLOW_OK . From liangzhihong1984 at 126.com Fri Sep 26 02:59:54 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Fri, 26 Sep 2008 08:59:54 +0800 (CST) Subject: [gst-devel] How can I capture my screen and streaming it? Message-ID: <30297099.509371222390794949.JavaMail.coremail@bj126app62.126.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Fri Sep 26 03:10:01 2008 From: msmith at xiph.org (Michael Smith) Date: Thu, 25 Sep 2008 18:10:01 -0700 Subject: [gst-devel] How can I capture my screen and streaming it? In-Reply-To: <30297099.509371222390794949.JavaMail.coremail@bj126app62.126.com> References: <30297099.509371222390794949.JavaMail.coremail@bj126app62.126.com> Message-ID: <3c1737210809251810k4dc4a9c0r1229cba2935c99f7@mail.gmail.com> Stop sending your question multiple times. You shouldn't expect instant responses to your questions. You've now asked the same question three times, with no specifics about what you want to do. There's no way anyone could answer you usefully even if they wanted to. If you want people to help you, then you should be polite and helpful - we can't guess all the information you didn't tell us. Mike From volter619 at 163.com Fri Sep 26 05:27:35 2008 From: volter619 at 163.com (Volter Yen) Date: Fri, 26 Sep 2008 11:27:35 +0800 (CST) Subject: [gst-devel] gstreamer dvb a/v sync problem In-Reply-To: <15e616860809251003t7a5d7481gbf1988be6efe8585@mail.gmail.com> References: <15e616860809251003t7a5d7481gbf1988be6efe8585@mail.gmail.com> <22575222.271371222329440264.JavaMail.coremail@bj163app93.163.com> Message-ID: <33271139.760981222399655632.JavaMail.coremail@bj163app125.163.com> Hi zaheer, Thank you for your reply first. I have adjusted my pipeline to the followings: gst-launch-0.10 dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 freq=474000000 code-rate-lp=2/3 code-rate-hp=1/2 guard=8 hierarchy=NONE pids=650:512:256:128 ! flutsdemux name=demuxer demuxer. ! queue max-size-buffers=0 max-size-time=0 ! mad ! audioconvert ! osssink the pids string contained audio-pid:video-pid:pmt-pid:pcr-pid, but it still report the similar problems : Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... 0:00:04.409004000 995 0x15088 WARN pipeline gstpipeline.c:491:gst_pipeline_change_state: failed to query pipeline latency New clock: FluTSClock 0:00:04.574164000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:46.765607000 observation 0:00:00.037777777 pcr: 4360223573 base_pcr: 4360220173pid: 128 0:00:04.615655000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:46.807092000 observation 0:00:00.076044444 pcr: 4360227017 base_pcr: 4360220173pid: 128 0:00:04.645182000 995 0x67e38 WARN mad gstmad.c:1385:gst_mad_chain: mad_header_decode had an error: lost synchronization 0:00:04.648290000 995 0x67e38 WARN mad gstmad.c:1385:gst_mad_chain: mad_header_decode had an error: lost synchronization 0:00:04.657230000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:46.848686000 observation 0:00:00.114100000 pcr: 4360230442 base_pcr: 4360220173pid: 128 0:00:04.675449000 995 0x67e38 WARN mad gstmad.c:1385:gst_mad_chain: mad_header_decode had an error: lost synchronization 0:00:04.692057000 995 0x67e38 ERROR oss gstosshelper.c:265:gst_oss_helper_rate_probe_check: Driver bug recognized (driver does not round rates correctly). Please file a bug report. 0:00:04.699159000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:46.890615000 observation 0:00:00.151877777 pcr: 4360233842 base_pcr: 4360220173pid: 128 0:00:04.717075000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:46.908528000 observation 0:00:00.189855555 pcr: 4360237260 base_pcr: 4360220173pid: 128 0:00:04.752799000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:46.944243000 observation 0:00:00.227844444 pcr: 4360240679 base_pcr: 4360220173pid: 128 0:00:04.793547000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:46.985002000 observation 0:00:00.265622222 pcr: 4360244079 base_pcr: 4360220173pid: 128 0:00:04.834954000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.026418000 observation 0:00:00.303811111 pcr: 4360247516 base_pcr: 4360220173pid: 128 0:00:04.876851000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.068275000 observation 0:00:00.342011111 pcr: 4360250954 base_pcr: 4360220173pid: 128 0:00:04.916750000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.107569000 observation 0:00:00.379855555 pcr: 4360254360 base_pcr: 4360220173pid: 128 0:00:04.956533000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.147965000 observation 0:00:00.417766666 pcr: 4360257772 base_pcr: 4360220173pid: 128 0:00:05.006744000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.198204000 observation 0:00:00.455888888 pcr: 4360261203 base_pcr: 4360220173pid: 128 0:00:05.038447000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.229892000 observation 0:00:00.493877777 pcr: 4360264622 base_pcr: 4360220173pid: 128 0:00:05.065626000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.257086000 observation 0:00:00.531866666 pcr: 4360268041 base_pcr: 4360220173pid: 128 0:00:05.098700000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.290139000 observation 0:00:00.570055555 pcr: 4360271478 base_pcr: 4360220173pid: 128 0:00:05.131209000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.322674000 observation 0:00:00.607977777 pcr: 4360274891 base_pcr: 4360220173pid: 128 0:00:05.173427000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.364868000 observation 0:00:00.645955555 pcr: 4360278309 base_pcr: 4360220173pid: 128 0:00:05.213681000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.405120000 observation 0:00:00.684077777 pcr: 4360281740 base_pcr: 4360220173pid: 128 0:00:05.253960000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.445400000 observation 0:00:00.722411111 pcr: 4360285190 base_pcr: 4360220173pid: 128 0:00:05.296089000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.487409000 observation 0:00:00.760533333 pcr: 4360288621 base_pcr: 4360220173pid: 128 0:00:05.337196000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.528626000 observation 0:00:00.798455555 pcr: 4360292034 base_pcr: 4360220173pid: 128 0:00:05.378814000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.570257000 observation 0:00:00.836511111 pcr: 4360295459 base_pcr: 4360220173pid: 128 0:00:05.419167000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.610611000 observation 0:00:00.874077777 pcr: 4360298840 base_pcr: 4360220173pid: 128 0:00:05.451952000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.643388000 observation 0:00:00.912477777 pcr: 4360302296 base_pcr: 4360220173pid: 128 0:00:05.471591000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.663034000 observation 0:00:00.950188888 pcr: 4360305690 base_pcr: 4360220173pid: 128 0:00:05.519283000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.710669000 observation 0:00:00.988244444 pcr: 4360309115 base_pcr: 4360220173pid: 128 0:00:05.557732000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.749193000 observation 0:00:01.026300000 pcr: 4360312540 base_pcr: 4360220173pid: 128 0:00:05.589757000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.781208000 observation 0:00:01.064144444 pcr: 4360315946 base_pcr: 4360220173pid: 128 0:00:05.630612000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.822074000 observation 0:00:01.102333333 pcr: 4360319383 base_pcr: 4360220173pid: 128 0:00:05.670879000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.862288000 observation 0:00:01.140388888 pcr: 4360322808 base_pcr: 4360220173pid: 128 0:00:05.717881000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.909333000 observation 0:00:01.178577777 pcr: 4360326245 base_pcr: 4360220173pid: 128 0:00:05.770904000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:47.962369000 observation 0:00:01.216844444 pcr: 4360329689 base_pcr: 4360220173pid: 128 0:00:05.827346000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.018787000 observation 0:00:01.254488888 pcr: 4360333077 base_pcr: 4360220173pid: 128 0:00:05.883173000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.074045000 observation 0:00:01.292677777 pcr: 4360336514 base_pcr: 4360220173pid: 128 0:00:05.932232000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.123661000 observation 0:00:01.330733333 pcr: 4360339939 base_pcr: 4360220173pid: 128 0:00:05.981189000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.172626000 observation 0:00:01.368444444 pcr: 4360343333 base_pcr: 4360220173pid: 128 0:00:06.022916000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.214368000 observation 0:00:01.406766666 pcr: 4360346782 base_pcr: 4360220173pid: 128 0:00:06.058763000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.250219000 observation 0:00:01.444344444 pcr: 4360350164 base_pcr: 4360220173pid: 128 0:00:06.093449000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.284892000 observation 0:00:01.482600000 pcr: 4360353607 base_pcr: 4360220173pid: 128 0:00:06.123607000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.315043000 observation 0:00:01.520444444 pcr: 4360357013 base_pcr: 4360220173pid: 128 0:00:06.152111000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.343540000 observation 0:00:01.559133333 pcr: 4360360495 base_pcr: 4360220173pid: 128 0:00:06.178283000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.369747000 observation 0:00:01.596633333 pcr: 4360363870 base_pcr: 4360220173pid: 128 0:00:06.204352000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.395800000 observation 0:00:01.634611111 pcr: 4360367288 base_pcr: 4360220173pid: 128 0:00:06.226999000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.418438000 observation 0:00:01.672600000 pcr: 4360370707 base_pcr: 4360220173pid: 128 0:00:06.253321000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.444755000 observation 0:00:01.710722222 pcr: 4360374138 base_pcr: 4360220173pid: 128 0:00:06.287208000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.478570000 observation 0:00:01.748844444 pcr: 4360377569 base_pcr: 4360220173pid: 128 0:00:06.328169000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.519617000 observation 0:00:01.787044444 pcr: 4360381007 base_pcr: 4360220173pid: 128 0:00:06.369345000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.560683000 observation 0:00:01.825022222 pcr: 4360384425 base_pcr: 4360220173pid: 128 0:00:06.403399000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.594802000 observation 0:00:01.862866666 pcr: 4360387831 base_pcr: 4360220173pid: 128 0:00:06.439097000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.630498000 observation 0:00:01.900788888 pcr: 4360391244 base_pcr: 4360220173pid: 128 0:00:06.466664000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.658119000 observation 0:00:01.938766666 pcr: 4360394662 base_pcr: 4360220173pid: 128 0:00:06.504782000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.696231000 observation 0:00:01.976966666 pcr: 4360398100 base_pcr: 4360220173pid: 128 0:00:06.547100000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.738536000 observation 0:00:02.015022222 pcr: 4360401525 base_pcr: 4360220173pid: 128 0:00:06.577145000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.768577000 observation 0:00:02.053077777 pcr: 4360404950 base_pcr: 4360220173pid: 128 0:00:06.617597000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.808995000 observation 0:00:02.091133333 pcr: 4360408375 base_pcr: 4360220173pid: 128 0:00:06.653936000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.845400000 observation 0:00:02.128911111 pcr: 4360411775 base_pcr: 4360220173pid: 128 0:00:06.691507000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.882946000 observation 0:00:02.167166666 pcr: 4360415218 base_pcr: 4360220173pid: 128 0:00:06.728782000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.920238000 observation 0:00:02.205011111 pcr: 4360418624 base_pcr: 4360220173pid: 128 0:00:06.768842000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:48.960276000 observation 0:00:02.243066666 pcr: 4360422049 base_pcr: 4360220173pid: 128 0:00:06.812396000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.003835000 observation 0:00:02.281466666 pcr: 4360425505 base_pcr: 4360220173pid: 128 0:00:06.849304000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.040759000 observation 0:00:02.319177777 pcr: 4360428899 base_pcr: 4360220173pid: 128 0:00:06.894183000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.085636000 observation 0:00:02.357022222 pcr: 4360432305 base_pcr: 4360220173pid: 128 0:00:06.934070000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.125530000 observation 0:00:02.395288888 pcr: 4360435749 base_pcr: 4360220173pid: 128 0:00:06.974929000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.166381000 observation 0:00:02.433277777 pcr: 4360439168 base_pcr: 4360220173pid: 128 0:00:07.022311000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.213777000 observation 0:00:02.471255555 pcr: 4360442586 base_pcr: 4360220173pid: 128 0:00:07.037682000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.229149000 observation 0:00:02.509311111 pcr: 4360446011 base_pcr: 4360220173pid: 128 0:00:07.081660000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.273117000 observation 0:00:02.547711111 pcr: 4360449467 base_pcr: 4360220173pid: 128 0:00:07.135488000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.326926000 observation 0:00:02.585211111 pcr: 4360452842 base_pcr: 4360220173pid: 128 0:00:07.174099000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.365470000 observation 0:00:02.623200000 pcr: 4360456261 base_pcr: 4360220173pid: 128 0:00:07.189351000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.380792000 observation 0:00:02.661466666 pcr: 4360459705 base_pcr: 4360220173pid: 128 0:00:07.232429000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.423832000 observation 0:00:02.699233333 pcr: 4360463104 base_pcr: 4360220173pid: 128 0:00:07.272664000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.464124000 observation 0:00:02.737500000 pcr: 4360466548 base_pcr: 4360220173pid: 128 0:00:07.306085000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.497433000 observation 0:00:02.775411111 pcr: 4360469960 base_pcr: 4360220173pid: 128 0:00:07.341311000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.532218000 observation 0:00:02.813466666 pcr: 4360473385 base_pcr: 4360220173pid: 128 0:00:07.371949000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.563399000 observation 0:00:02.851455555 pcr: 4360476804 base_pcr: 4360220173pid: 128 0:00:07.422800000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.614263000 observation 0:00:02.889855555 pcr: 4360480260 base_pcr: 4360220173pid: 128 0:00:07.453956000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.645423000 observation 0:00:02.927633333 pcr: 4360483660 base_pcr: 4360220173pid: 128 0:00:07.497989000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.689439000 observation 0:00:02.965833333 pcr: 4360487098 base_pcr: 4360220173pid: 128 0:00:07.529698000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.721151000 observation 0:00:03.004233333 pcr: 4360490554 base_pcr: 4360220173pid: 128 0:00:07.571903000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.763335000 observation 0:00:03.041588888 pcr: 4360493916 base_pcr: 4360220173pid: 128 0:00:07.614566000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.806023000 observation 0:00:03.079711111 pcr: 4360497347 base_pcr: 4360220173pid: 128 0:00:07.641538000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.832975000 observation 0:00:03.117700000 pcr: 4360500766 base_pcr: 4360220173pid: 128 0:00:07.692544000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.883977000 observation 0:00:03.155966666 pcr: 4360504210 base_pcr: 4360220173pid: 128 0:00:07.721539000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.912982000 observation 0:00:03.193600000 pcr: 4360507597 base_pcr: 4360220173pid: 128 0:00:07.759673000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.951109000 observation 0:00:03.231788888 pcr: 4360511034 base_pcr: 4360220173pid: 128 0:00:07.795630000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:49.987072000 observation 0:00:03.269777777 pcr: 4360514453 base_pcr: 4360220173pid: 128 0:00:07.830727000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.022196000 observation 0:00:03.307966666 pcr: 4360517890 base_pcr: 4360220173pid: 128 0:00:07.868586000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.060046000 observation 0:00:03.346100000 pcr: 4360521322 base_pcr: 4360220173pid: 128 0:00:07.921394000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.112836000 observation 0:00:03.383800000 pcr: 4360524715 base_pcr: 4360220173pid: 128 0:00:07.958325000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.149715000 observation 0:00:03.422066666 pcr: 4360528159 base_pcr: 4360220173pid: 128 0:00:07.994094000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.185553000 observation 0:00:03.460122222 pcr: 4360531584 base_pcr: 4360220173pid: 128 0:00:08.031076000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.222521000 observation 0:00:03.497966666 pcr: 4360534990 base_pcr: 4360220173pid: 128 0:00:08.058683000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.250107000 observation 0:00:03.535955555 pcr: 4360538409 base_pcr: 4360220173pid: 128 0:00:08.107609000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.299060000 observation 0:00:03.573866666 pcr: 4360541821 base_pcr: 4360220173pid: 128 0:00:08.145280000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.336718000 observation 0:00:03.612344444 pcr: 4360545284 base_pcr: 4360220173pid: 128 0:00:08.177014000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.368446000 observation 0:00:03.649977777 pcr: 4360548671 base_pcr: 4360220173pid: 128 0:00:08.218102000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.409557000 observation 0:00:03.688522222 pcr: 4360552140 base_pcr: 4360220173pid: 128 0:00:08.251214000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.442674000 observation 0:00:03.726088888 pcr: 4360555521 base_pcr: 4360220173pid: 128 0:00:08.292924000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.484360000 observation 0:00:03.763933333 pcr: 4360558927 base_pcr: 4360220173pid: 128 0:00:08.324351000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.515805000 observation 0:00:03.802055555 pcr: 4360562358 base_pcr: 4360220173pid: 128 0:00:08.370270000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.561706000 observation 0:00:03.840244444 pcr: 4360565795 base_pcr: 4360220173pid: 128 0:00:08.405390000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.596825000 observation 0:00:03.878022222 pcr: 4360569195 base_pcr: 4360220173pid: 128 0:00:08.449019000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.640465000 observation 0:00:03.916155555 pcr: 4360572627 base_pcr: 4360220173pid: 128 0:00:08.495403000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.686831000 observation 0:00:03.954411111 pcr: 4360576070 base_pcr: 4360220173pid: 128 0:00:08.548830000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.740262000 observation 0:00:03.992122222 pcr: 4360579464 base_pcr: 4360220173pid: 128 0:00:08.598858000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.790305000 observation 0:00:04.030111111 pcr: 4360582883 base_pcr: 4360220173pid: 128 0:00:08.643605000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.835046000 observation 0:00:04.068155555 pcr: 4360586307 base_pcr: 4360220173pid: 128 0:00:08.688080000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.879513000 observation 0:00:04.106700000 pcr: 4360589776 base_pcr: 4360220173pid: 128 0:00:08.728006000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.919458000 observation 0:00:04.144344444 pcr: 4360593164 base_pcr: 4360220173pid: 128 0:00:08.764482000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.955930000 observation 0:00:04.182322222 pcr: 4360596582 base_pcr: 4360220173pid: 128 0:00:08.797103000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:50.988535000 observation 0:00:04.220244444 pcr: 4360599995 base_pcr: 4360220173pid: 128 0:00:08.824707000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.016135000 observation 0:00:04.258644444 pcr: 4360603451 base_pcr: 4360220173pid: 128 0:00:08.850001000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.041451000 observation 0:00:04.296766666 pcr: 4360606882 base_pcr: 4360220173pid: 128 0:00:08.872805000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.064232000 observation 0:00:04.334477777 pcr: 4360610276 base_pcr: 4360220173pid: 128 0:00:08.912113000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.103016000 observation 0:00:04.372388888 pcr: 4360613688 base_pcr: 4360220173pid: 128 0:00:09.182485000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.373921000 observation 0:00:04.410511111 pcr: 4360617119 base_pcr: 4360220173pid: 128 0:00:09.206356000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.397785000 observation 0:00:04.448711111 pcr: 4360620557 base_pcr: 4360220173pid: 128 0:00:09.229181000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.420620000 observation 0:00:04.486411111 pcr: 4360623950 base_pcr: 4360220173pid: 128 0:00:09.254370000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.445819000 observation 0:00:04.525088888 pcr: 4360627431 base_pcr: 4360220173pid: 128 0:00:09.281855000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.473286000 observation 0:00:04.562455555 pcr: 4360630794 base_pcr: 4360220173pid: 128 0:00:09.313371000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.504827000 observation 0:00:04.600711111 pcr: 4360634237 base_pcr: 4360220173pid: 128 0:00:09.350106000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.541543000 observation 0:00:04.638633333 pcr: 4360637650 base_pcr: 4360220173pid: 128 0:00:09.391508000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.582959000 observation 0:00:04.676755555 pcr: 4360641081 base_pcr: 4360220173pid: 128 0:00:09.435974000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.627289000 observation 0:00:04.714744444 pcr: 4360644500 base_pcr: 4360220173pid: 128 0:00:09.484366000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.675804000 observation 0:00:04.752588888 pcr: 4360647906 base_pcr: 4360220173pid: 128 0:00:09.528966000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.720386000 observation 0:00:04.790644444 pcr: 4360651331 base_pcr: 4360220173pid: 128 0:00:09.573197000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.764631000 observation 0:00:04.828622222 pcr: 4360654749 base_pcr: 4360220173pid: 128 0:00:09.612957000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.804380000 observation 0:00:04.866822222 pcr: 4360658187 base_pcr: 4360220173pid: 128 0:00:09.646685000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.838118000 observation 0:00:04.904733333 pcr: 4360661599 base_pcr: 4360220173pid: 128 0:00:09.677000000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.868453000 observation 0:00:04.943133333 pcr: 4360665055 base_pcr: 4360220173pid: 128 0:00:09.705090000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.896526000 observation 0:00:04.981055555 pcr: 4360668468 base_pcr: 4360220173pid: 128 0:00:09.729455000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.920887000 observation 0:00:05.018833333 pcr: 4360671868 base_pcr: 4360220173pid: 128 0:00:09.753245000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.944703000 observation 0:00:05.057022222 pcr: 4360675305 base_pcr: 4360220173pid: 128 0:00:09.771724000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.963148000 observation 0:00:05.094944444 pcr: 4360678718 base_pcr: 4360220173pid: 128 0:00:09.789507000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.980944000 observation 0:00:05.132922222 pcr: 4360682136 base_pcr: 4360220173pid: 128 0:00:09.804228000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:51.995659000 observation 0:00:05.171044444 pcr: 4360685567 base_pcr: 4360220173pid: 128 0:00:09.819578000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.011007000 observation 0:00:05.209033333 pcr: 4360688986 base_pcr: 4360220173pid: 128 0:00:09.837411000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.028852000 observation 0:00:05.246944444 pcr: 4360692398 base_pcr: 4360220173pid: 128 0:00:09.852385000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.043831000 observation 0:00:05.285000000 pcr: 4360695823 base_pcr: 4360220173pid: 128 0:00:09.871863000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.063281000 observation 0:00:05.322988888 pcr: 4360699242 base_pcr: 4360220173pid: 128 0:00:09.894087000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.085514000 observation 0:00:05.361044444 pcr: 4360702667 base_pcr: 4360220173pid: 128 0:00:10.051501000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.242930000 observation 0:00:05.399166666 pcr: 4360706098 base_pcr: 4360220173pid: 128 0:00:10.113472000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.304888000 observation 0:00:05.437077777 pcr: 4360709510 base_pcr: 4360220173pid: 128 0:00:10.197019000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.388447000 observation 0:00:05.475211111 pcr: 4360712942 base_pcr: 4360220173pid: 128 0:00:10.271922000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.463282000 observation 0:00:05.513188888 pcr: 4360716360 base_pcr: 4360220173pid: 128 0:00:10.342440000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.533889000 observation 0:00:05.551244444 pcr: 4360719785 base_pcr: 4360220173pid: 128 0:00:10.429176000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.620027000 observation 0:00:05.589166666 pcr: 4360723198 base_pcr: 4360220173pid: 128 0:00:10.511633000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.703050000 observation 0:00:05.627222222 pcr: 4360726623 base_pcr: 4360220173pid: 128 0:00:10.592263000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.783616000 observation 0:00:05.665411111 pcr: 4360730060 base_pcr: 4360220173pid: 128 0:00:10.667845000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.859261000 observation 0:00:05.703533333 pcr: 4360733491 base_pcr: 4360220173pid: 128 0:00:10.730790000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.922164000 observation 0:00:05.741455555 pcr: 4360736904 base_pcr: 4360220173pid: 128 0:00:10.760748000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.951574000 observation 0:00:05.779366666 pcr: 4360740316 base_pcr: 4360220173pid: 128 0:00:10.816072000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:52.997264000 observation 0:00:05.817488888 pcr: 4360743747 base_pcr: 4360220173pid: 128 0:00:10.860156000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:53.051584000 observation 0:00:05.855544444 pcr: 4360747172 base_pcr: 4360220173pid: 128 0:00:10.917036000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:53.108467000 observation 0:00:05.894088888 pcr: 4360750641 base_pcr: 4360220173pid: 128 0:00:10.960515000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:53.151938000 observation 0:00:05.931788888 pcr: 4360754034 base_pcr: 4360220173pid: 128 0:00:11.009135000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:53.200554000 observation 0:00:05.969711111 pcr: 4360757447 base_pcr: 4360220173pid: 128 0:00:11.068780000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:53.260203000 observation 0:00:06.007555555 pcr: 4360760853 base_pcr: 4360220173pid: 128 0:00:11.130503000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:53.321929000 observation 0:00:06.045466666 pcr: 4360764265 base_pcr: 4360220173pid: 128 0:00:11.191709000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:53.383145000 observation 0:00:06.084077777 pcr: 4360767740 base_pcr: 4360220173pid: 128 0:00:11.264307000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:53.455680000 observation 0:00:06.121788888 pcr: 4360771134 base_pcr: 4360220173pid: 128 0:00:11.319882000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:53.511311000 observation 0:00:06.159633333 pcr: 4360774540 base_pcr: 4360220173pid: 128 0:00:11.372402000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:53.563297000 observation 0:00:06.197622222 pcr: 4360777959 base_pcr: 4360220173pid: 128 0:00:11.453440000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:53.644796000 observation 0:00:06.235744444 pcr: 4360781390 base_pcr: 4360220173pid: 128 0:00:11.465217000 995 0x67e38 WARN baseaudiosink gstbaseaudiosink.c:696:gst_base_audio_sink_render: warning: Compensating for audio synchronisation problems 0:00:11.468943000 995 0x67e38 WARN baseaudiosink gstbaseaudiosink.c:696:gst_base_audio_sink_render: warning: Unexpected discontinuity in audio timestamps of more than half a second (0:00:04.541812500), resyncing WARNING: from element /pipeline0/osssink0: Compensating for audio synchronisation problems Additional debug info: gstbaseaudiosink.c(696): gst_base_audio_sink_render (): /pipeline0/osssink0: Unexpected discontinuity in audio timestamps of more than half a second (0:00:04.541812500), resyncing 0:00:12.056207000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:54.247465000 observation 0:00:06.273733333 pcr: 4360784809 base_pcr: 4360220173pid: 128 0:00:12.087686000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:54.279119000 observation 0:00:06.311988888 pcr: 4360788252 base_pcr: 4360220173pid: 128 0:00:12.117722000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:54.309095000 observation 0:00:06.349977777 pcr: 4360791671 base_pcr: 4360220173pid: 128 0:00:12.150405000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:54.341837000 observation 0:00:06.388100000 pcr: 4360795102 base_pcr: 4360220173pid: 128 0:00:12.180920000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:54.372344000 observation 0:00:06.425877777 pcr: 4360798502 base_pcr: 4360220173pid: 128 0:00:12.215942000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:54.407259000 observation 0:00:06.463788888 pcr: 4360801914 base_pcr: 4360220173pid: 128 0:00:12.252148000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:54.443580000 observation 0:00:06.501777777 pcr: 4360805333 base_pcr: 4360220173pid: 128 0:00:12.289305000 995 0x68030 ERROR flutsdemux gstmpegtsdemux.c:1562:gst_fluts_demux_parse_adaptation_field: internal 337465:33:54.480742000 observation 0:00:06.540111111 pcr: 4360808783 base_pcr: 4360220173pid: 128 ........ And according to your last advice, I should update my gstreamer core to a more recent one,could you tell me which version of gstreamer that you used to have fixed these bugs , thank you. Best regards ! Votler ?2008-09-26?"Zaheer Merali" ??? >2008/9/25 Volter Yen >> hi all, >> when using gstreamer to play dvb signal, if I set the osssink's sync to >> true, the pipelie give 'Unexpected discontinuity in audio timestamps of >> more than half a second(...) resyncing ' and stalled there....but it is said >> that 'sync=true ' should work , my gstreamer core is >> 0.10.14,fluendo-mpegdmuex-0.10.15, and by the way my pipelie is: >> gst-launch-0.10 dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 >> freq=474000000 code-rate-lp=2/3 code-rate-hp=1/2 guard=8 hierarchy=NONE >> pids=630:512:128 ! flutsdemux es-pids=650:512 name=demuxer demuxer. ! queue >> ! mad ! audioconvert ! osssink >> and if i add 'sync=false' to ossink , it works but when play with video >> the audio is heavy lagged. >> >> I have referrenced to this link >> https://core.fluendo.com/gstreamer/trac/ticket/46, it seems that these >> problem should have been resolved. >> anybody know the way out? thank you >> >> Volter > >You have a few problems with your pipeline. > >i) es-pids should not be used with flutsdemux. If you have the pmt pid >and the elementary stream pids coming out of dvbsrc then there is no >issue. Using es-pids like you are makes it ignore the pcr values >contained in the transport stream. > >ii) The queues you should be using should have max-size-buffers=0 and >max-size-time=0 set. > >You should also be using a more recent version of gstreamer core. > >Zaheer > >------------------------------------------------------------------------- >This SF.Net email is sponsored by the Moblin Your Move Developer's challenge >Build the coolest Linux based applications with Moblin SDK & win great prizes >Grand prize is a trip for two to an Open Source event anywhere in the world >http://moblin-contest.org/redirect.php?banner_id=100&url=/ >_______________________________________________ >gstreamer-devel mailing list >gstreamer-devel at lists.sourceforge.net >https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From liangzhihong1984 at 126.com Fri Sep 26 06:54:15 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Fri, 26 Sep 2008 12:54:15 +0800 (CST) Subject: [gst-devel] Sorry for my multiple questions Message-ID: <13238094.661121222404855607.JavaMail.coremail@bj126app78.126.com> Sorry, I really didn't mean to do that. Any disturbances, I would like to apologize. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Fri Sep 26 07:06:02 2008 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Fri, 26 Sep 2008 10:36:02 +0530 Subject: [gst-devel] video freeze while playing a file without audio In-Reply-To: <5559508B2BD31A44B1C267F4E6EF471D974180@hz-email05.apac.arcsoft.corp> References: <5559508B2BD31A44B1C267F4E6EF471D974180@hz-email05.apac.arcsoft.corp> Message-ID: <785339900809252206y30480e08ve07b971c4a6cc5cf@mail.gmail.com> Hi , Can u please tell me how you are running pipeline ? using gst-launch or using some application written by you ? On Fri, Sep 26, 2008 at 6:15 AM, Chen Yong(Ever)[??] < yongchen at arcsoft.com.cn> wrote: > Thanks your reply. In my example, we don't push any audio data to > down-streaming element, only push video data, but video is freeezed. > > > >Hi , > > > > When u don't have audio data in your container file then that time > >don't create pad for that stream in plugin code i.e. dont try to push > audio > >data to next element . > > In your case what is happening is that you dont have audio data even > >then you are trying to push some junk data to next element and in that > case > >next element is returning some other return value than GST_FLOW_OK from > its > >chain function . > > To debug this just see return value of gst_pad_push( ) fucntion , > >this value should be GST_FLOW_OK . > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From jpuydt at free.fr Fri Sep 26 07:55:48 2008 From: jpuydt at free.fr (Julien Puydt) Date: Fri, 26 Sep 2008 07:55:48 +0200 Subject: [gst-devel] Video input with various systems In-Reply-To: <48DA2C8A.1040201@hora-obscura.de> References: <48D7F58F.30903@free.fr> <48DA2C8A.1040201@hora-obscura.de> Message-ID: <48DC7964.3090904@free.fr> Stefan Kost a ?crit : > Julien Puydt schrieb: >> I have noticed that for linux-based systems, there are v4l2src and >> dv1394src : I added support to them in ekiga (well, in a personal post3 >> branch...). >> >> For win32, it seems ksvideosrc is what I want (except it doesn't have >> property probe). >> >> For osx, there's a directory sys/osxvideo in gst-plugins-bad and >> gst-plugins, but it doesn't have sources... at least not in my checkout, >> and gst-plugins-good's sys/osxvideo has only a sink. >> >> For *BSD: I have no clue what they have. >> > It would totally rock if you could have a go at writing a autovideosrc > that does this automatically. Have a look at > gst-plugins-good/gst/autodetect/ Sigh. I have no idea how to do that... and notice that such a plugin would need to implement property probe so the devices user-friendly names could be gotten. Snark on freenode&gimpnet From linguang_wang at astrocom.cn Fri Sep 26 09:46:09 2008 From: linguang_wang at astrocom.cn (=?gb2312?B?zfXB1rni?=) Date: Fri, 26 Sep 2008 15:46:09 +0800 Subject: [gst-devel] how to set "max-size-time" property of queue ? Message-ID: <200809261546081198321@astrocom.cn> Hi, all developers! When I seted the "max-size-time" property of queue in my application, it printed out one error such as: GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstQueue' has no property named `\x80?^@\u0001' I didn't know what is the matter. In my codes, the setting codes is: g_object_set(G_OBJECT(aQueue), "max-size-time", 0, NULL); But if I used gst-launch like this: gst-launch neonhttpsrc uri=$ ! qtdemux ! queue max-size-buffers=0 max-size-time=0 ! faad ! osssink sync=false it is ok.Is there anyone to tell me some information? Looking forward for answers. Thanks in advance! 2008-09-26 linguang_wang at astrocom.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 802 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 43 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... 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Name: not available Type: image/gif Size: 64 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 344 bytes Desc: not available URL: From yongchen at arcsoft.com.cn Fri Sep 26 10:48:09 2008 From: yongchen at arcsoft.com.cn (=?gb2312?B?Q2hlbiBZb25nKEV2ZXIpW7PC08Jd?=) Date: Fri, 26 Sep 2008 16:48:09 +0800 Subject: [gst-devel] video freeze while playing a file without audio In-Reply-To: Message-ID: <5559508B2BD31A44B1C267F4E6EF471D97424C@hz-email05.apac.arcsoft.corp> Hi, We write an application to create a pipeline, add connect all pads to all sinks, and simply set pipeline status to playing. In fact, this problem also occur in other case, maybe we playback a mpeg file, but while forward scan with high-speed, we should send video only(I picture) , video is free zed, too. >From: "sudarshan bisht" >Subject: Re: [gst-devel] video freeze while playing a file without audio >To: "Discussion of the development of GStreamer" >Message-ID: <785339900809252206y30480e08ve07b971c4a6cc5cf at mail.gmail.com> >Content-Type: text/plain; charset="gb2312" >Hi , Can u please tell me how you are running pipeline ? using >gst-launch or using some application written by you ? >On Fri, Sep 26, 2008 at 6:15 AM, Chen Yong(Ever)[??] < >yongchen at arcsoft.com.cn> wrote: > Thanks your reply. In my example, we don't push any audio data to > down-streaming element, only push video data, but video is freeezed. > > > >Hi , > > > > When u don't have audio data in your container file then that time > >don't create pad for that stream in plugin code i.e. dont try to push > audio > >data to next element . > > In your case what is happening is that you dont have audio data even > >then you are trying to push some junk data to next element and in that > case > >next element is returning some other return value than GST_FLOW_OK from > its > >chain function . > > To debug this just see return value of gst_pad_push( ) fucntion , > >this value should be GST_FLOW_OK . > > From wim.taymans at gmail.com Fri Sep 26 11:10:01 2008 From: wim.taymans at gmail.com (Wim Taymans) Date: Fri, 26 Sep 2008 11:10:01 +0200 Subject: [gst-devel] how to set "max-size-time" property of queue ? In-Reply-To: <200809261546081198321@astrocom.cn> References: <200809261546081198321@astrocom.cn> Message-ID: <1222420201.30312.47.camel@metal> On Fri, 2008-09-26 at 15:46 +0800, ??? wrote: max-size-time is a guint64 property so it needs to be cast to this in a vararg function like g_object_set(). Like this: g_object_set (G_OBJECT(aQueue), "max-size-time", (guint64)0, NULL); Wim > > > > > > Hi, all developers! > When I seted the > "max-size-time" property of queue in my application, it printed out one error such as: > GLib-GObject-WARNING > **: > IA__g_object_set_valist: object class `GstQueue' has no property named `\x80?^@\u0001' > I didn't know what is > the matter. In my > codes, the setting > codes is: > g_object_set(G_OBJECT(aQueue), "max-size-time", 0, NULL); > > But if I used > gst-launch like this: > gst-launch neonhttpsrc > uri=$ ! qtdemux ! queue > max-size-buffers=0 > max-size-time=0 ! > faad ! osssink > sync=false > it is ok.Is there > anyone to tell me some > information? Looking > forward for answers. > Thanks in advance! > > 2008-09-26 > > _______________________ > linguang_wang at astrocom.cn > > > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From zaheermerali at gmail.com Fri Sep 26 12:41:17 2008 From: zaheermerali at gmail.com (Zaheer Merali) Date: Fri, 26 Sep 2008 11:41:17 +0100 Subject: [gst-devel] gstreamer dvb a/v sync problem In-Reply-To: <33271139.760981222399655632.JavaMail.coremail@bj163app125.163.com> References: <22575222.271371222329440264.JavaMail.coremail@bj163app93.163.com> <15e616860809251003t7a5d7481gbf1988be6efe8585@mail.gmail.com> <33271139.760981222399655632.JavaMail.coremail@bj163app125.163.com> Message-ID: <15e616860809260341t3700b287xa4c8ed89bf876dc6@mail.gmail.com> > And according to your last advice, I should update my gstreamer core to a > more recent one,could you tell me which version of gstreamer that you used > to have fixed these bugs , thank you. > > > > Best regards ! > > Votler Good practise is to always use the latest stable release. In today's case gstreamer 0.10.20 gst-plugins-base 0.10.20 and gst-plugins-bad 0.10.8. Zaheer From gstmediaplayer at gmail.com Fri Sep 26 13:03:15 2008 From: gstmediaplayer at gmail.com (John Doe) Date: Fri, 26 Sep 2008 14:03:15 +0300 Subject: [gst-devel] Problem with GStreamer tag information Message-ID: Hi, I've following problem with GStreamer tags, when pipeline is on playing state all tag information(artists,title,duration etc.) is parsed without any problem by using bus message subsystem. How is it possible to get same information from file without playing? And how is it to get tag_image from file(both of cases, when playing and ready states)? I've get information from tag by using gst_tag_list_get_string function but what's the way with image case? ps. thanks for helping code: src = gst_element_factory_make("filesrc", "source"); g_object_set(src, location, "path", NULL); gst_message_parse_tag(gst_message_new_tag(GST_OBJECT(src),tag_list), &tag_list); //Is this the way to get information without playing state??? -JD -------------- next part -------------- An HTML attachment was scrubbed... URL: From ml_benoitfouet at purplelabs.com Fri Sep 26 14:47:56 2008 From: ml_benoitfouet at purplelabs.com (Benoit Fouet) Date: Fri, 26 Sep 2008 14:47:56 +0200 Subject: [gst-devel] Problem with GStreamer tag information In-Reply-To: References: Message-ID: <48DCD9FC.8050609@purplelabs.com> Hi, John Doe wrote: > Hi, I've following problem with GStreamer tags, when pipeline is on playing state > all tag information(artists,title,duration etc.) is parsed without any problem by using bus message subsystem. How is it possible to get same information from file without playing? > And how is it to get tag_image from file(both of cases, when playing and ready states)? > > to parse a file and retrieve its information, you need to have access to its data. to have access to data, you at least need to be prerolling, so in PAUSED state -- Benoit Fouet Purple Labs S.A. www.purplelabs.com From gmane at colin.guthr.ie Fri Sep 26 16:19:24 2008 From: gmane at colin.guthr.ie (Colin Guthrie) Date: Fri, 26 Sep 2008 15:19:24 +0100 Subject: [gst-devel] Some advice In-Reply-To: <3c1737210809221055l7cd42603i93d011f44461a547@mail.gmail.com> References: <3c1737210809221055l7cd42603i93d011f44461a547@mail.gmail.com> Message-ID: (Sorry for the late reply... I'm not subscribed and post via Gmane, but hoped that after having my first message validated manually my subsequent messages would pass. Alas that didn't happen, so I've now subscribed officially and disabled delivery so gmane can still be used :)) Michael Smith wrote: >> 1) Is it possible to find out which sink is really used when you use >> gconfsink, autoaudiosink or halsink? > > These are all bins (they contain other elements); so you can just > iterate (possibly recursively) over all the elements in them, looking > for sinks, and then look at the type of the sink. autoaudiosink and > halsink are both pretty easy, but gconfsink lets you have a > more-or-less arbitrary bin, which could even contain multiple sinks > (though that would be rare). OK, thanks for the info. I'll look into doing that :) >> 2) When streaming data from shoutcast, it is possible to decode the >> StreamTitle= header via icydemux. I have hacked phonon-gstreamer backed >> to do this now (in a similar way to how playbin does it). However it >> only extracts the title. >> >> When I use the Xine backend it appears to be able to extract inline >> metadata out of the stream content itself (e.g. when it finds a vorbis >> comment header or an id3 tag). That's as much as I can tell from >> observation. I've hacked the phonon-gstreamer to be able to "guess" the >> artist name from the StreamTitle but it's really not very nice. So my >> main question is, can the decoders handle inline metadata decoding and >> if so how can I enabled this? >> > > I guess it'd be useful to point us at a stream that behaves like this. Oh, erm, yeah that would help wouldn't it. I've been playing with this one: http://87.117.200.136:8010 (it's not my musical taste: http://www.last.fm/user/coling) It seems to be an AAC stream and the xine backend for phonon can update the artist and title streams properly from this it seems. > 'icydemux' will extract metadata properly from shoutcast streams (so > long as the source is providing this data). Vorbis streams have inline > metadata that should be handled automatically, without anything > special being needed. I've never seen inline id3 being streamed, nor > have I ever seen software that would handle it at all. Yeah I think I inferred too much from my initial observations for which I apologise :) I see now with more testing that the xine metadata extraction seems to be using ORGANIZATION for ARTIST and ALBUM fields on streams. The above mentioned stream has it's artist/album data extracted properly with xine, but it doesn't seem to happen with gstreamer (at least I don't get any TAG messages on the bus). I guess this is a shortcoming of the faad implementation? > So you shouldn't need to do anything special apart from requesting the > http sources to get shoutcast-style metadata. From what I can tell icydebux only parses the StreamTitle= header and sets the TITLE metadata accordingly. This is useful but inline metadata extraction is nicer. As you've said, vorbis comments should work fine, so I'll have to find some streams that do this so I can test. Thanks for your help Mike :) Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] From irfanshaikh at tataelxsi.co.in Sat Sep 27 11:04:44 2008 From: irfanshaikh at tataelxsi.co.in (Irfan Shaikh) Date: Sat, 27 Sep 2008 14:34:44 +0530 Subject: [gst-devel] asfdemux Message-ID: <9D5E1752379A43408015F7FE984661157827A4@CHNEXVS01.VSNLXCHANGE.COM> Hi all, In the following path /home/GStreamer/gst-plugins-ugly-0.10.9/gst/asfdemux/ We have gstasfmux.c and gstasfmux.h, i think which provides ASF muxer functionalitiy. When ever i use, { gst-inspeact | grep asf } ,I get only asdemux as a available plug-in element {which is an ASF demuxer}. Why i dont get an ASF muxer element on ASF when i use gst-inspect. How do i use gstasfmux.c and gstasfmux.h to make as gstreamer plug-in element and see the muxer element when i do gst-inspect. Regards, Irfan.... This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Sat Sep 27 11:36:47 2008 From: bilboed at gmail.com (Edward Hervey) Date: Sat, 27 Sep 2008 11:36:47 +0200 Subject: [gst-devel] asfdemux In-Reply-To: <9D5E1752379A43408015F7FE984661157827A4@CHNEXVS01.VSNLXCHANGE.COM> References: <9D5E1752379A43408015F7FE984661157827A4@CHNEXVS01.VSNLXCHANGE.COM> Message-ID: <1222508207.2544.8.camel@putamadre> The asf muxer code in that directory is code that hasn't been ported to gstreamer 0.10. It is therefore not built. Porting it to 0.10 would be much appreciated though :) Edward On Sat, 2008-09-27 at 14:34 +0530, Irfan Shaikh wrote: > > Hi all, > > In the following > path /home/GStreamer/gst-plugins-ugly-0.10.9/gst/asfdemux/ > > We have gstasfmux.c and gstasfmux.h, i think which provides ASF muxer > functionalitiy. > > When ever i use, { gst-inspeact | grep asf } ,I get only asdemux as a > available plug-in element {which is an ASF demuxer}. > > Why i dont get an ASF muxer element on ASF when i use gst-inspect. > > How do i use gstasfmux.c and gstasfmux.h to make as gstreamer plug-in > element and see the muxer element when i do gst-inspect. > > > Regards, > Irfan.... > > > > This message (including any attachment) is confidential and may be > legally privileged. Access to this message by anyone other than the > intended recipient(s) listed above is unauthorized. If you are not the > intended recipient you are hereby notified that any disclosure, > copying, or distribution of the message, or any action taken or > omission of action by you in reliance upon it, is prohibited and may > be unlawful. Please immediately notify the sender by reply e-mail and > permanently delete all copies of the message if you have received this > message in error. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From irfanshaikh at tataelxsi.co.in Sat Sep 27 11:40:48 2008 From: irfanshaikh at tataelxsi.co.in (Irfan Shaikh) Date: Sat, 27 Sep 2008 15:10:48 +0530 Subject: [gst-devel] asfdemux References: <9D5E1752379A43408015F7FE984661157827A4@CHNEXVS01.VSNLXCHANGE.COM> <1222508207.2544.8.camel@putamadre> Message-ID: <9D5E1752379A43408015F7FE984661157827A7@CHNEXVS01.VSNLXCHANGE.COM> Hi, Can you please suggest me what all steps are required so as to port gstasfmux.c on Gstreamer. I would port it to gstreamer as soon as possible, if you provide me some hints. Regards, Irfan -----Original Message----- From: Edward Hervey [mailto:bilboed at gmail.com] Sent: Sat 9/27/2008 3:06 PM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] asfdemux The asf muxer code in that directory is code that hasn't been ported to gstreamer 0.10. It is therefore not built. Porting it to 0.10 would be much appreciated though :) Edward On Sat, 2008-09-27 at 14:34 +0530, Irfan Shaikh wrote: > > Hi all, > > In the following > path /home/GStreamer/gst-plugins-ugly-0.10.9/gst/asfdemux/ > > We have gstasfmux.c and gstasfmux.h, i think which provides ASF muxer > functionalitiy. > > When ever i use, { gst-inspeact | grep asf } ,I get only asdemux as a > available plug-in element {which is an ASF demuxer}. > > Why i dont get an ASF muxer element on ASF when i use gst-inspect. > > How do i use gstasfmux.c and gstasfmux.h to make as gstreamer plug-in > element and see the muxer element when i do gst-inspect. > > > Regards, > Irfan.... > > > > This message (including any attachment) is confidential and may be > legally privileged. Access to this message by anyone other than the > intended recipient(s) listed above is unauthorized. If you are not the > intended recipient you are hereby notified that any disclosure, > copying, or distribution of the message, or any action taken or > omission of action by you in reliance upon it, is prohibited and may > be unlawful. Please immediately notify the sender by reply e-mail and > permanently delete all copies of the message if you have received this > message in error. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. -------------- next part -------------- An HTML attachment was scrubbed... URL: From peter at cs.upt.ro Sat Sep 27 12:09:53 2008 From: peter at cs.upt.ro (Petre Mierlutiu) Date: Sat, 27 Sep 2008 13:09:53 +0300 (EEST) Subject: [gst-devel] EOS on rtpbin's send_rtcp_src pad? Message-ID: Hello everyone, I am using a simple pipeline that delivers h263 encoded video over rtp using gstrtpbin. My pipeline is something like this: (videosrc)->ffenc_h263p->rtph263ppay->gstrtpbin-[send_rtp_src]->udpsink1 \-[send_rtcp_src]->udpsink2 I also have a udpsrc--[recv_rtcp_sink]-->gstrtpbin. All works fine except that the pipelink doesn't finish due to udpsink2 not seeing EOS. I think RtpSession should push the EOS event on this pad also, somewhere at gstrtpsession.c in gst_rtp_session_event_send_rtp_sink() function. But I'm not sure how to do this because just sending the EOS would prevent RtpSession to send out the RTCP BYE message which is scheduled for later. So my simple modification from below does not work correctly, although it causes the pipeline to end. Any advice, please? I am using gst-plugins-bad-0.10.8, gst-plugins-base-0.10.20, gst-plugins-good-0.10.10 and gstreamer-0.10.20. Thank you, Peter --------------------------------------------------- --- gst-plugins-bad-0.10.8/gst/rtpmanager/gstrtpsession.c 2008-07-19 16:18:23.000000000 +0300 +++ build/gst-plugins-bad-0.10.8/gst/rtpmanager/gstrtpsession.c 2008-09-27 13:09:02.000000000 +0300 @@ -1556,10 +1556,12 @@ gst_rtp_session_event_send_rtp_sink (Gst case GST_EVENT_EOS:{ GstClockTime current_time; + gst_event_ref(event); ret = gst_pad_push_event (rtpsession->send_rtp_src, event); current_time = gst_clock_get_time (rtpsession->priv->sysclock); rtp_session_send_bye (rtpsession->priv->session, "End of stream", current_time); + ret = gst_pad_push_event (rtpsession->send_rtcp_src, event) && ret; break; } default: ------------------------------------------------------------------ From liangzhihong1984 at 126.com Sun Sep 28 03:20:55 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Sun, 28 Sep 2008 09:20:55 +0800 (CST) Subject: [gst-devel] About capture audio from Microphone? Message-ID: <20950614.40501222564856000.JavaMail.coremail@bj126app43.126.com> Hi, how can I capture audio from a microphone and stream it to other machines in the network? By the way, how can I capture my screen and broadcast it? Thank you in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: From liangzhihong1984 at 126.com Sun Sep 28 08:51:15 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Sun, 28 Sep 2008 14:51:15 +0800 (CST) Subject: [gst-devel] About A/V synchronization Message-ID: <3079502.198691222584675019.JavaMail.coremail@bj126app56.126.com> I'm working on a A/V chat tool. Now can capture the video stream from a webcam and audio from the microphone. Video is h264 encoded and audio PCMU. Then send them to other clients in the network. But I don't know how to do the A/V synchronization. Can someone give me some help. It would be appreciated. Thanks a lot. -------------- next part -------------- An HTML attachment was scrubbed... URL: From Michael.MIelewczik at linux-technical.info Mon Sep 29 00:17:03 2008 From: Michael.MIelewczik at linux-technical.info (Michael Mielewczik) Date: Mon, 29 Sep 2008 00:17:03 +0200 Subject: [gst-devel] gst-plugins-base de.po modification Message-ID: <200809290017.03796.Michael.MIelewczik@linux-technical.info> I just added some additional german translations to the last CVS/SVN version of gst-plugins-base. Is this usefull? What ist the workflow to commit such translation patches? I additionally made some modifications to the de.po files for the other packages (gstreamer and gst-plugins-good) Michael -------------- next part -------------- A non-text attachment was scrubbed... Name: de.po Type: text/x-gettext-translation Size: 16300 bytes Desc: not available URL: From thaytan at noraisin.net Mon Sep 29 01:36:35 2008 From: thaytan at noraisin.net (Jan Schmidt) Date: Mon, 29 Sep 2008 00:36:35 +0100 Subject: [gst-devel] New pre-releases of Core/Base uploaded - 0.10.20.3. In-Reply-To: <1221658286.6090.20.camel@fancy-ubuntu> References: <1220949484.1673.5.camel@fancy-ubuntu> <1221071491.1673.29.camel@fancy-ubuntu> <1221658286.6090.20.camel@fancy-ubuntu> Message-ID: <1222644995.29073.6.camel@fancy-ubuntu> Hi all, New pre-release tarballs of Core & Base are available - 0.10.20.4 of each. I expect these to be the last pre-releases, with the release itself on Thursday. See the ChangeLog files for details of the changes since the 0.10.20.3 pre-releases. The tarballs are at: http://gstreamer.freedesktop.org/data/src/gstreamer/pre/gstreamer-0.10.20.4.tar.bz2 http://gstreamer.freedesktop.org/data/src/gst-plugins-base/pre/gst-plugins-base-0.10.20.4.tar.bz2 and http://gstreamer.freedesktop.org/data/src/gst-python/pre/gst-python-0.10.12.2.tar.bz2 Cheers, Jan. On Wed, 2008-09-17 at 14:31 +0100, Jan Schmidt wrote: > Hi all, > > I've just uploaded new pre-release tarballs of Core & Base, which are > now at 0.10.20.3. gst-python is still at 0.10.12.2 > > The tarballs are at: > > http://gstreamer.freedesktop.org/data/src/gstreamer/pre/gstreamer-0.10.20.3.tar.bz2 > > http://gstreamer.freedesktop.org/data/src/gst-plugins-base/pre/gst-plugins-base-0.10.20.3.tar.bz2 > and > http://gstreamer.freedesktop.org/data/src/gst-python/pre/gst-python-0.10.12.2.tar.bz2 > > The release is scheduled for next Monday, 22nd September, however there > are 2 unresolved blocker bugs in Core relating to BaseTransform change > that may delay it. > > Please test, and file bugs in http://bugzilla.gnome.org > > Cheers, > Jan. > > On Wed, 2008-09-10 at 19:31 +0100, Jan Schmidt wrote: > > Pre-releases of GStreamer Core 0.10.20.2, Base 0.10.20.2 and Python > > bindings 0.10.12.2 are now available: > > > > http://gstreamer.freedesktop.org/data/src/gstreamer/pre/gstreamer-0.10.20.2.tar.bz2 > > > > http://gstreamer.freedesktop.org/data/src/gst-plugins-base/pre/gst-plugins-base-0.10.20.2.tar.bz2 > > and > > http://gstreamer.freedesktop.org/data/src/gst-python/pre/gst-python-0.10.12.2.tar.bz2 > > > > Please test them out, and file bugs in http://bugzilla.gnome.org/ > > > > New pre-releases Friday or Saturday as needed. > > > > Cheers, > > Jan. > > > > On Tue, 2008-09-09 at 09:38 +0100, Jan Schmidt wrote: > > > Freezing Core/Base/Python to make 0.10.20.2, 0.10.20.2 and 0.10.12.2 > > > respectively. > > > > > > For details, see the release schedule: > > > http://gstreamer.freedesktop.org/wiki/ReleasePlanning2008 > > > > > > Cheers, > > > Jan. -- Jan Schmidt From olivier.crete at collabora.co.uk Mon Sep 29 02:11:47 2008 From: olivier.crete at collabora.co.uk (Olivier =?ISO-8859-1?Q?Cr=EAte?=) Date: Sun, 28 Sep 2008 20:11:47 -0400 Subject: [gst-devel] New pre-releases of Core/Base uploaded - 0.10.20.3. In-Reply-To: <1222644995.29073.6.camel@fancy-ubuntu> References: <1220949484.1673.5.camel@fancy-ubuntu> <1221071491.1673.29.camel@fancy-ubuntu> <1221658286.6090.20.camel@fancy-ubuntu> <1222644995.29073.6.camel@fancy-ubuntu> Message-ID: <1222647107.7666.1.camel@TesterBox.tester.ca> On Mon, 2008-09-29 at 00:36 +0100, Jan Schmidt wrote: > Hi all, > > New pre-release tarballs of Core & Base are available - 0.10.20.4 of > each. I expect these to be the last pre-releases, with the release > itself on Thursday. > > The tarballs are at: > http://gstreamer.freedesktop.org/data/src/gstreamer/pre/gstreamer-0.10.20.4.tar.bz2 This tarball is still missing the patch for bug #548764 -- Olivier Cr?te olivier.crete at collabora.co.uk Collabora Ltd -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: This is a digitally signed message part URL: From manish.rana at gmail.com Mon Sep 29 14:28:56 2008 From: manish.rana at gmail.com (Manish Rana) Date: Mon, 29 Sep 2008 17:58:56 +0530 Subject: [gst-devel] How is queue2 as compared to queue Message-ID: <8c192ddd0809290528x438f3968v51bc7f153d804eba@mail.gmail.com> Hi All, Can i know what is the difference between the two queue elements present in Gstreamer. Thanks in advance Manish -------------- next part -------------- An HTML attachment was scrubbed... URL: From ved.kpl at gmail.com Mon Sep 29 14:46:26 2008 From: ved.kpl at gmail.com (ved kpl) Date: Mon, 29 Sep 2008 18:16:26 +0530 Subject: [gst-devel] How is queue2 as compared to queue In-Reply-To: <8c192ddd0809290528x438f3968v51bc7f153d804eba@mail.gmail.com> References: <8c192ddd0809290528x438f3968v51bc7f153d804eba@mail.gmail.com> Message-ID: <7496c23f0809290546x29c0231cnab3e8a86b2240d7e@mail.gmail.com> Hi, gst-inspect on both the queues. Queue2 has buffering support. It can post messages(if enabled) on the bus when it is buffering. Queue also emits signals depending on the buffer status, but then I guess it is in the streaming thread. Ved On Mon, Sep 29, 2008 at 5:58 PM, Manish Rana wrote: > Hi All, > > Can i know what is the difference between the two queue elements present in > Gstreamer. > > Thanks in advance > Manish > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From dominicchen at via-telecom.com Sat Sep 27 12:12:43 2008 From: dominicchen at via-telecom.com (Dominic Chen) Date: Sat, 27 Sep 2008 18:12:43 +0800 Subject: [gst-devel] link error when X overlaying using gst_x_overlay_set_xwindow_id Message-ID: Dear subscribers I am working on an application related to video playback. And we choose gstreamer as mediaengine. But when I use overlay2sink element, there comes a problem: undefined reference to `gst_x_overlay_get_type' Undefined reference to `gst_x_overlay_set_xwindow_id' I don't know why! I will be very appreciated if you give me some suggestion. Thanks! Dominic Chen 2008/09/27 -------------- next part -------------- An HTML attachment was scrubbed... URL: From ke at suse.de Mon Sep 29 16:31:22 2008 From: ke at suse.de (Karl Eichwalder) Date: Mon, 29 Sep 2008 16:31:22 +0200 Subject: [gst-devel] [Translation-team-de] gst-plugins-base de.po modification In-Reply-To: <200809290017.03796.Michael.MIelewczik@linux-technical.info> (Michael Mielewczik's message of "Mon, 29 Sep 2008 00:17:03 +0200") References: <200809290017.03796.Michael.MIelewczik@linux-technical.info> Message-ID: Michael Mielewczik writes: > I just added some additional german translations to the last CVS/SVN > version of gst-plugins-base. > Is this usefull? What ist the workflow to commit such translation patches? > > I additionally made some modifications to the de.po files for the other > packages (gstreamer and gst-plugins-good) Better submit to the Translation Project at http://translationproject.org/html/translators.html where I'm listed as the translator (http://translationproject.org/domain/gst-plugins-base.html). But I do not remember me that I actually wanted to work on this file. -- Karl Eichwalder R&D / Documentation SUSE LINUX Products GmbH, GF: Markus Rex, HRB 16746 (AG Nuernberg) From irfanshaikh at tataelxsi.co.in Mon Sep 29 16:59:48 2008 From: irfanshaikh at tataelxsi.co.in (Irfan Shaikh) Date: Mon, 29 Sep 2008 20:29:48 +0530 Subject: [gst-devel] ffenc_h264 plug-in ??? Message-ID: <9D5E1752379A43408015F7FE984661157827B5@CHNEXVS01.VSNLXCHANGE.COM> Hi All, Can any one please tell me in which package can i find ffmpeg's H264 encoder plug-in ffenc_h264. Regards, Irfan This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Mon Sep 29 17:49:10 2008 From: wim.taymans at gmail.com (Wim Taymans) Date: Mon, 29 Sep 2008 17:49:10 +0200 Subject: [gst-devel] ffenc_h264 plug-in ??? In-Reply-To: <9D5E1752379A43408015F7FE984661157827B5@CHNEXVS01.VSNLXCHANGE.COM> References: <9D5E1752379A43408015F7FE984661157827B5@CHNEXVS01.VSNLXCHANGE.COM> Message-ID: <1222703351.7431.36.camel@metal> On Mon, 2008-09-29 at 20:29 +0530, Irfan Shaikh wrote: > > > Hi All, > > Can any one please tell me in which package can i find ffmpeg's > H264 encoder plug-in ffenc_h264. Are you making up element names? That element does not exist and neither does the 'ffmpeg h264 encoder'. Regards, Wim > > Regards, > Irfan > > > This message (including any attachment) is confidential and may be > legally privileged. Access to this message by anyone other than the > intended recipient(s) listed above is unauthorized. If you are not the > intended recipient you are hereby notified that any disclosure, > copying, or distribution of the message, or any action taken or > omission of action by you in reliance upon it, is prohibited and may > be unlawful. Please immediately notify the sender by reply e-mail and > permanently delete all copies of the message if you have received this > message in error. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bilboed at gmail.com Mon Sep 29 18:19:59 2008 From: bilboed at gmail.com (Edward Hervey) Date: Mon, 29 Sep 2008 18:19:59 +0200 Subject: [gst-devel] ffenc_h264 plug-in ??? In-Reply-To: <1222703351.7431.36.camel@metal> References: <9D5E1752379A43408015F7FE984661157827B5@CHNEXVS01.VSNLXCHANGE.COM> <1222703351.7431.36.camel@metal> Message-ID: <1222705199.16208.0.camel@putamadre> On Mon, 2008-09-29 at 17:49 +0200, Wim Taymans wrote: > On Mon, 2008-09-29 at 20:29 +0530, Irfan Shaikh wrote: > > > > > > Hi All, > > > > Can any one please tell me in which package can i find ffmpeg's > > H264 encoder plug-in ffenc_h264. > > Are you making up element names? That element does not exist and neither > does the 'ffmpeg h264 encoder'. Indeed. I'm guessing he means the h264 encoder used by ffmpeg ergo... x264. And for that we have a plugin : x264enc. Edward > > Regards, > Wim > > > > Regards, > > Irfan > > > > > > This message (including any attachment) is confidential and may be > > legally privileged. Access to this message by anyone other than the > > intended recipient(s) listed above is unauthorized. If you are not the > > intended recipient you are hereby notified that any disclosure, > > copying, or distribution of the message, or any action taken or > > omission of action by you in reliance upon it, is prohibited and may > > be unlawful. Please immediately notify the sender by reply e-mail and > > permanently delete all copies of the message if you have received this > > message in error. > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > > Build the coolest Linux based applications with Moblin SDK & win great prizes > > Grand prize is a trip for two to an Open Source event anywhere in the world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From thaytan at noraisin.net Mon Sep 29 18:23:45 2008 From: thaytan at noraisin.net (Jan Schmidt) Date: Mon, 29 Sep 2008 17:23:45 +0100 Subject: [gst-devel] New pre-releases of Core/Base uploaded - 0.10.20.3. In-Reply-To: <1222647107.7666.1.camel@TesterBox.tester.ca> References: <1220949484.1673.5.camel@fancy-ubuntu> <1221071491.1673.29.camel@fancy-ubuntu> <1221658286.6090.20.camel@fancy-ubuntu> <1222644995.29073.6.camel@fancy-ubuntu> <1222647107.7666.1.camel@TesterBox.tester.ca> Message-ID: <1222705425.29073.9.camel@fancy-ubuntu> On Sun, 2008-09-28 at 20:11 -0400, Olivier Cr?te wrote: > On Mon, 2008-09-29 at 00:36 +0100, Jan Schmidt wrote: > > Hi all, > > > > New pre-release tarballs of Core & Base are available - 0.10.20.4 of > > each. I expect these to be the last pre-releases, with the release > > itself on Thursday. > > > > The tarballs are at: > > http://gstreamer.freedesktop.org/data/src/gstreamer/pre/gstreamer-0.10.20.4.tar.bz2 > > This tarball is still missing the patch for bug #548764 Good call... it sort of looks like Wtay added a comment on the bug indicating that he'd committed the patch, but didn't actually commit it? I can't see the ChangeLog entry he quoted in the bug, or the patch itself in CVS. J. -- Jan Schmidt From tristan at sat.qc.ca Mon Sep 29 18:51:09 2008 From: tristan at sat.qc.ca (Tristan Matthews) Date: Mon, 29 Sep 2008 12:51:09 -0400 Subject: [gst-devel] link error when X overlaying using gst_x_overlay_set_xwindow_id In-Reply-To: References: Message-ID: <48E1077D.1000201@sat.qc.ca> Hi, Dominic Chen wrote: > Dear subscribers > I am working on an application related to video playback. And we choose gstreamer as mediaengine. But when I use overlay2sink element, there comes a problem: undefined reference to `gst_x_overlay_get_type' > Undefined reference to `gst_x_overlay_set_xwindow_id' I think you have to link against libgstinterfaces-0.10.so Best, Tristan > > I don?t know why! I will be very appreciated if you give me some suggestion. > > Thanks! > > Dominic Chen > 2008/09/27 > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Tristan Matthews Soci?t? des arts technologiques [SAT] email: tristan at sat.qc.ca web: http://www.music.mcgill.ca/~tmatthews From acassis at gmail.com Mon Sep 29 19:57:33 2008 From: acassis at gmail.com (Alan Carvalho de Assis) Date: Mon, 29 Sep 2008 14:57:33 -0300 Subject: [gst-devel] Converting a gst-launch pipeline to C code Message-ID: <37367b3a0809291057j57bf0408jfad9a283ee4f3f47@mail.gmail.com> Hi, I am converting a gst-launch command to C code but it is not working. This is the line I am trying to convert (this is working fine): gst-launch-0.10 -v filesrc location=potter.avi ! avidemux name=demux demux.video_00 ! {queue ! ffdec_h264 ! xvimagesink} demux.audio_00 ! {queue ! mad ! alsasink} I am basing on manual Ogg playback example, but I can't get video and audio working at the same time. When I try to do that I see a window stopped at first video frame and no audio is played. Please find below my video/audio player. Notice I don't want to use playbin, I really want to fix this code to understand what I am doing wrong. Best Regards, Alan #include #include #include static void on_pad_added (GstElement *element, GstPad *pad, gpointer data) { GstPad *sinkpad; GstElement *decoder = (GstElement *) data; /* We can now link this pad with the vorbis-decoder sink pad */ g_print ("Dynamic pad created, linking demuxer/decoder\n"); sinkpad = gst_element_get_static_pad (decoder, "sink"); gst_pad_link (pad, sinkpad); gst_object_unref (sinkpad); } static gboolean bus_call (GstBus *bus, GstMessage *msg, gpointer data) { GMainLoop *loop = (GMainLoop *) data; switch (GST_MESSAGE_TYPE (msg)) { case GST_MESSAGE_EOS: g_print ("End of stream\n"); g_main_loop_quit (loop); break; case GST_MESSAGE_ERROR: { gchar *debug; GError *error; gst_message_parse_error (msg, &error, &debug); g_free (debug); g_printerr ("Error: %s\n", error->message); g_error_free (error); g_main_loop_quit (loop); break; } default: break; } return TRUE; } int main (int argc, char *argv[]) { GMainLoop *loop; GstElement *pipeline, *source, *demuxer, *decvd, *decad, *vdqueue, *adqueue, *vdsink, *adsink; GstBus *bus; /* Initialisation */ gst_init (&argc, &argv); loop = g_main_loop_new (NULL, FALSE); /* Check input arguments */ if (argc != 2) { g_printerr ("Usage: %s \n", argv[0]); return -1; } /* Create gstreamer elements */ pipeline = gst_pipeline_new ("media-player"); source = gst_element_factory_make ("filesrc", "file-source"); demuxer = gst_element_factory_make ("avidemux", "avi-demuxer"); decvd = gst_element_factory_make ("ffdec_h264", "h264-decoder"); decad = gst_element_factory_make ("mad", "mp3-decoder"); vdsink = gst_element_factory_make ("autovideosink", "video-sink"); vdqueue = gst_element_factory_make ("queue", "video-queue"); adqueue = gst_element_factory_make ("queue", "audio-queue"); adsink = gst_element_factory_make ("alsasink", "audio-sink"); if (!pipeline || !source || !demuxer || !decvd || !decad || !vdsink || !vdqueue || !adqueue || !adsink) { g_printerr ("One element could not be created. Exiting.\n"); return -1; } /* Set up the pipeline */ /* we set the input filename to the source element */ g_object_set (G_OBJECT (source), "location", argv[1], NULL); /* we add a message handler */ bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline)); gst_bus_add_watch (bus, bus_call, loop); gst_object_unref (bus); /* we add all elements into the pipeline */ /* file-source | ogg-demuxer | vorbis-decoder | converter | alsa-output */ gst_bin_add_many (GST_BIN (pipeline), source, demuxer, decvd, decad, vdsink, vdqueue, adqueue, adsink, NULL); //gst_bin_add_many (GST_BIN (pipeline), // source, demuxer, decvd, vdqueue, vdsink, NULL); /* we link the elements together */ /* file-source -> ogg-demuxer ~> vorbis-decoder -> converter -> alsa-output */ gst_element_link (source, demuxer); gst_element_link (decvd, vdqueue); gst_element_link (vdqueue, vdsink); gst_element_link (decad, adqueue); gst_element_link (adqueue, adsink); //gst_element_link_many (decvd, vdqueue, vdsink, NULL); g_signal_connect (demuxer, "pad-added", G_CALLBACK (on_pad_added), decvd); /* note that the demuxer will be linked to the decoder dynamically. The reason is that Ogg may contain various streams (for example audio and video). The source pad(s) will be created at run time, by the demuxer when it detects the amount and nature of streams. Therefore we connect a callback function which will be executed when the "pad-added" is emitted.*/ /* Set the pipeline to "playing" state*/ g_print ("Now playing: %s\n", argv[1]); gst_element_set_state (pipeline, GST_STATE_PLAYING); /* Iterate */ g_print ("Running...\n"); g_main_loop_run (loop); /* Out of the main loop, clean up nicely */ g_print ("Returned, stopping playback\n"); gst_element_set_state (pipeline, GST_STATE_NULL); g_print ("Deleting pipeline\n"); gst_object_unref (GST_OBJECT (pipeline)); return 0; } From thijsvermeir at gmail.com Mon Sep 29 20:21:36 2008 From: thijsvermeir at gmail.com (Thijs Vermeir) Date: Mon, 29 Sep 2008 20:21:36 +0200 Subject: [gst-devel] Converting a gst-launch pipeline to C code In-Reply-To: <37367b3a0809291057j57bf0408jfad9a283ee4f3f47@mail.gmail.com> References: <37367b3a0809291057j57bf0408jfad9a283ee4f3f47@mail.gmail.com> Message-ID: <2ecbfb170809291121o4b90a647o881c18329e2d4ba2@mail.gmail.com> Hi, On Mon, Sep 29, 2008 at 7:57 PM, Alan Carvalho de Assis wrote: > Hi, > > I am converting a gst-launch command to C code but it is not working. > > This is the line I am trying to convert (this is working fine): > gst-launch-0.10 -v filesrc location=potter.avi ! avidemux name=demux > demux.video_00 ! {queue ! ffdec_h264 ! xvimagesink} demux.audio_00 ! > {queue ! mad ! alsasink} > > I am basing on manual Ogg playback example, but I can't get video and > audio working at the same time. When I try to do that I see a window > stopped at first video frame and no audio is played. Avidemux is adding 2 pads so it calls on_pad_added twice, so you should check if the new pad is a audio/video pad and connect to the correct decoder element. Gr, Thijs > Please find below my video/audio player. Notice I don't want to use > playbin, I really want to fix this code to understand what I am doing > wrong. > > Best Regards, > > Alan > > > > > #include > #include > #include > > static void > on_pad_added (GstElement *element, > GstPad *pad, > gpointer data) > { > GstPad *sinkpad; > GstElement *decoder = (GstElement *) data; > > /* We can now link this pad with the vorbis-decoder sink pad */ > g_print ("Dynamic pad created, linking demuxer/decoder\n"); > > sinkpad = gst_element_get_static_pad (decoder, "sink"); > > gst_pad_link (pad, sinkpad); > > gst_object_unref (sinkpad); > } > > static gboolean > bus_call (GstBus *bus, > GstMessage *msg, > gpointer data) > { > GMainLoop *loop = (GMainLoop *) data; > > switch (GST_MESSAGE_TYPE (msg)) { > > case GST_MESSAGE_EOS: > g_print ("End of stream\n"); > g_main_loop_quit (loop); > break; > > case GST_MESSAGE_ERROR: { > gchar *debug; > GError *error; > > gst_message_parse_error (msg, &error, &debug); > g_free (debug); > > g_printerr ("Error: %s\n", error->message); > g_error_free (error); > > g_main_loop_quit (loop); > break; > } > default: > break; > } > > return TRUE; > } > > int > main (int argc, > char *argv[]) > { > GMainLoop *loop; > > GstElement *pipeline, *source, *demuxer, *decvd, *decad, *vdqueue, > *adqueue, *vdsink, *adsink; > GstBus *bus; > > /* Initialisation */ > gst_init (&argc, &argv); > > loop = g_main_loop_new (NULL, FALSE); > > > /* Check input arguments */ > if (argc != 2) { > g_printerr ("Usage: %s \n", argv[0]); > return -1; > } > > > /* Create gstreamer elements */ > pipeline = gst_pipeline_new ("media-player"); > source = gst_element_factory_make ("filesrc", "file-source"); > demuxer = gst_element_factory_make ("avidemux", "avi-demuxer"); > decvd = gst_element_factory_make ("ffdec_h264", "h264-decoder"); > decad = gst_element_factory_make ("mad", "mp3-decoder"); > vdsink = gst_element_factory_make ("autovideosink", "video-sink"); > vdqueue = gst_element_factory_make ("queue", "video-queue"); > adqueue = gst_element_factory_make ("queue", "audio-queue"); > adsink = gst_element_factory_make ("alsasink", "audio-sink"); > > if (!pipeline || !source || !demuxer || !decvd || !decad || !vdsink > || !vdqueue || !adqueue || !adsink) { > g_printerr ("One element could not be created. Exiting.\n"); > return -1; > } > > /* Set up the pipeline */ > > /* we set the input filename to the source element */ > g_object_set (G_OBJECT (source), "location", argv[1], NULL); > > /* we add a message handler */ > bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline)); > gst_bus_add_watch (bus, bus_call, loop); > gst_object_unref (bus); > > /* we add all elements into the pipeline */ > /* file-source | ogg-demuxer | vorbis-decoder | converter | alsa-output */ > gst_bin_add_many (GST_BIN (pipeline), > source, demuxer, decvd, decad, vdsink, vdqueue, > adqueue, adsink, NULL); > //gst_bin_add_many (GST_BIN (pipeline), > // source, demuxer, decvd, vdqueue, vdsink, NULL); > > /* we link the elements together */ > /* file-source -> ogg-demuxer ~> vorbis-decoder -> converter -> alsa-output */ > gst_element_link (source, demuxer); > gst_element_link (decvd, vdqueue); > gst_element_link (vdqueue, vdsink); > gst_element_link (decad, adqueue); > gst_element_link (adqueue, adsink); > //gst_element_link_many (decvd, vdqueue, vdsink, NULL); > > g_signal_connect (demuxer, "pad-added", G_CALLBACK (on_pad_added), decvd); > > /* note that the demuxer will be linked to the decoder dynamically. > The reason is that Ogg may contain various streams (for example > audio and video). The source pad(s) will be created at run time, > by the demuxer when it detects the amount and nature of streams. > Therefore we connect a callback function which will be executed > when the "pad-added" is emitted.*/ > > /* Set the pipeline to "playing" state*/ > g_print ("Now playing: %s\n", argv[1]); > gst_element_set_state (pipeline, GST_STATE_PLAYING); > > > /* Iterate */ > g_print ("Running...\n"); > g_main_loop_run (loop); > > > /* Out of the main loop, clean up nicely */ > g_print ("Returned, stopping playback\n"); > gst_element_set_state (pipeline, GST_STATE_NULL); > > g_print ("Deleting pipeline\n"); > gst_object_unref (GST_OBJECT (pipeline)); > > return 0; > } > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From tzicatl at gmail.com Tue Sep 30 01:02:06 2008 From: tzicatl at gmail.com (Noe Nieto) Date: Mon, 29 Sep 2008 16:02:06 -0700 Subject: [gst-devel] Problem with clockoverlay Message-ID: Hi. An app that displays MJPEG Streams from webcam cameras looks like the following. gst-launch-0.10 gnomevfssrc location=$CAMERA ! decodebin ! ffmpegcolorspace ! videoscale ! ximagesink force-aspect-ratio="true" But when I add a clockoverlay element between the videoscale and ximagesink the video gets frozen and just displays one frame. I am not understanding something, can someone please, tell me what I am doing wrong? Regards. Noe Nieto. -------------- next part -------------- An HTML attachment was scrubbed... URL: From acassis at gmail.com Tue Sep 30 01:21:54 2008 From: acassis at gmail.com (Alan Carvalho de Assis) Date: Mon, 29 Sep 2008 20:21:54 -0300 Subject: [gst-devel] Converting a gst-launch pipeline to C code In-Reply-To: <2ecbfb170809291121o4b90a647o881c18329e2d4ba2@mail.gmail.com> References: <37367b3a0809291057j57bf0408jfad9a283ee4f3f47@mail.gmail.com> <2ecbfb170809291121o4b90a647o881c18329e2d4ba2@mail.gmail.com> Message-ID: <37367b3a0809291621o50bd35e4j24b6db8d485a082e@mail.gmail.com> Hi Thijs, 2008/9/29 Thijs Vermeir : > Hi, >> I am basing on manual Ogg playback example, but I can't get video and >> audio working at the same time. When I try to do that I see a window >> stopped at first video frame and no audio is played. > > Avidemux is adding 2 pads so it calls on_pad_added twice, so you should > check if the new pad is a audio/video pad and connect to the correct > decoder element. > I will test it and let you know if it works. Thank you very much. > Gr, > Thijs > Best Regards, Alan From airmind at gmail.com Tue Sep 30 02:21:12 2008 From: airmind at gmail.com (Alexandre) Date: Mon, 29 Sep 2008 21:21:12 -0300 Subject: [gst-devel] ffenc_h264 plug-in ??? In-Reply-To: <1222705199.16208.0.camel@putamadre> References: <9D5E1752379A43408015F7FE984661157827B5@CHNEXVS01.VSNLXCHANGE.COM> <1222703351.7431.36.camel@metal> <1222705199.16208.0.camel@putamadre> Message-ID: <48f4838d0809291721y7649ab66nb79a37c95c63207d@mail.gmail.com> On Mon, Sep 29, 2008 at 1:19 PM, Edward Hervey wrote: > On Mon, 2008-09-29 at 17:49 +0200, Wim Taymans wrote: > > On Mon, 2008-09-29 at 20:29 +0530, Irfan Shaikh wrote: > > > > > > > > > Hi All, > > > > > > Can any one please tell me in which package can i find ffmpeg's > > > H264 encoder plug-in ffenc_h264. > > > > Are you making up element names? That element does not exist and neither > > does the 'ffmpeg h264 encoder'. > > Indeed. I'm guessing he means the h264 encoder used by ffmpeg ergo... > x264. And for that we have a plugin : x264enc. Or he meant ffenc_h264 or ffdec_h264, both of which can be found in the ffmpeg plugin. > > > Edward > > > > > Regards, > > Wim > > > > > > Regards, > > > Irfan > > > > > > > > > This message (including any attachment) is confidential and may be > > > legally privileged. Access to this message by anyone other than the > > > intended recipient(s) listed above is unauthorized. If you are not the > > > intended recipient you are hereby notified that any disclosure, > > > copying, or distribution of the message, or any action taken or > > > omission of action by you in reliance upon it, is prohibited and may > > > be unlawful. Please immediately notify the sender by reply e-mail and > > > permanently delete all copies of the message if you have received this > > > message in error. > > > > > > > ------------------------------------------------------------------------- > > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Alexandre Rosenfeld EngComp 06 - USP S?o Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: From tzicatl at gmail.com Tue Sep 30 02:56:21 2008 From: tzicatl at gmail.com (Noe Nieto) Date: Mon, 29 Sep 2008 17:56:21 -0700 Subject: [gst-devel] Help setting frame rate with videorate In-Reply-To: <35662.69.17.64.250.1198126715.squirrel@webmail.8bh.com> References: <35662.69.17.64.250.1198126715.squirrel@webmail.8bh.com> Message-ID: Hi All I am experiencing the same problem in a similar situation. On this case the MJPEG source is at 25 fps. Here is the example: gnomevfssrc location=$CAMERA ! typefind ! multipartdemux ! image/jpeg, framerate=25/1 ! jpegdec ! ffmpegcolorspace ! theoraenc ! oggmux ! filesink = location=test0.ogg The first problem here is that past the jpegdec element, the framerate is ignored and falls back to 0/1. The second problem is that If I try to force the framerate, say, right before ffmpegcolorspace, then the pipeline will not construct. The third problem occurs if I use the videorate element. The video accumulates frames, but totem only displays a still picture instead of a video. Can anyone advice? Regards. Noe Nieto. 2007/12/19 Aaron Lindsey > With the help from Edward and quite a bit of searching on my own I was > able to solve my problem. The frame rates seemed to be ignored and 1/1 > was always assumed until I used the multipartdemux element. Here is the > pipeline that finally worked: > > gnomevfssrc location=http://url?resolution=320x240\&fps=2! > multipartdemux ! image/jpeg,framerate=2/1 ! jpegdec ! theoraenc ! oggmux > ! filesink location=test.ogg > > After looking at the docs for multipartdemux, I'm not sure why I got any > video at all before. I guess jpegdec was figuring out some way to parse > the multipart stream into jpegs with the side effect that the frame rate > from the caps filter was ignored....or something. Haven't dug deep enough > into gstreamer to really know. Can anyone shed some light? > > Aaron > > > Hi, > > > > Add a capsfilter informing the content is jpeg data at Xfps. > > > > Ex : gnomevfssrc ! image/jpeg,framerate=2/1 ! jpegdec .... > > > > Edward > > > > > ------------------------------------------------------------------------- > SF.Net email is sponsored by: > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services > for just about anything Open Source. > > http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From irfanshaikh at tataelxsi.co.in Tue Sep 30 06:08:10 2008 From: irfanshaikh at tataelxsi.co.in (Irfan Shaikh) Date: Tue, 30 Sep 2008 09:38:10 +0530 Subject: [gst-devel] ffenc_h264 plug-in ??? Message-ID: <9D5E1752379A43408015F7FE984661157827B8@CHNEXVS01.VSNLXCHANGE.COM> Hi Wim, According to your reply ffenc_h264 element does not exist and neither does the 'ffmpeg h264 encoder'. But just going through net i found some pipelines which used ffenc_h264. Earlier i was also not very sure about exsiting ffmpeg h264 encoder.But needed some clarification. Can i know in which package can i find h264 encoder used by ffmpeg ergo..... x264. As Edward told that we we have a plugin : x264enc...Please help me to find the package Pipelines found on net which used ffenc_h264: 1)gst-launch -v filesrc location=/home/kgupta/testflv/limca.flv ! decodebin name=d ! queue ! videorate ! ffmpegcolorspace ! videoscale ! video/x-raw-yuv,height=144,width=176,framerate=\(fraction\)15 1 ! ffenc_h264 ! flutsmux name=mux ! filesink location=limca_aac_h264_faacadts.ts d. ! queue ! audioconvert ! audioresample ! audio/x-raw-int, endianness=1234, signed=\(boolean\)true, width=\(int\)16, depth=\(int\)16, rate=(int\)16000,channels=(int\)1 ! faac bitrate=16000 ! mux. 2)gst-launch cameracapture prio=90 ! queue ! ffmpegcolorspace prio=50 ! queue ! ffenc_h264 prio=50 ! ... And there are many other to Thanks for your reply, Regards, Irfan This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ajitjohn at tataelxsi.co.in Tue Sep 30 08:45:27 2008 From: ajitjohn at tataelxsi.co.in (ajitjohn) Date: Tue, 30 Sep 2008 12:15:27 +0530 Subject: [gst-devel] Query regarding FU-B support in rtph264pay plugin of gstreamer Message-ID: <007901c922c8$203ef410$68033c0a@telxsi.com> Hii all, I am using the rtph264pay plugin of gstreamer which supports only FU-A mode but there is reference of FU-B mode in RFC 3984 ,is this mode (FU-B) support required for streaming purpose . Regards, Ajit. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From gmane at colin.guthr.ie Tue Sep 30 10:09:34 2008 From: gmane at colin.guthr.ie (Colin Guthrie) Date: Tue, 30 Sep 2008 09:09:34 +0100 Subject: [gst-devel] ffenc_h264 plug-in ??? In-Reply-To: <48f4838d0809291721y7649ab66nb79a37c95c63207d@mail.gmail.com> References: <9D5E1752379A43408015F7FE984661157827B5@CHNEXVS01.VSNLXCHANGE.COM> <1222703351.7431.36.camel@metal> <1222705199.16208.0.camel@putamadre> <48f4838d0809291721y7649ab66nb79a37c95c63207d@mail.gmail.com> Message-ID: Alexandre wrote: > Or he meant ffenc_h264 or ffdec_h264, both of which can be found in the > ffmpeg plugin. Many distros disable h264 codecs in their official builds due to patent issues IIRC. I know Mandriva do. Perhaps the OPs build is affected in this way? On Mandriva you can use the third party PLF repositories to install identical packages to the official ones but with the "questionable" bits enabled. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] From henrique.ferreiro at gmail.com Tue Sep 30 11:41:44 2008 From: henrique.ferreiro at gmail.com (Henrique Ferreiro =?ISO-8859-1?Q?Garc=EDa?=) Date: Tue, 30 Sep 2008 11:41:44 +0200 Subject: [gst-devel] muxing streams Message-ID: <1222767704.18136.1.camel@macbook> Hi all! I have extracted the video and audio streams from a matroska file getting: mkv.0: JVT NAL sequence, H.264 video @ L 31 mkv.1: ATSC A/52 aka AC-3 aka Dolby Digital stream, 48 kHz,, complete main (CM) 3 front/2 rear, LFE on,, 448 kbit/s reserved Dolby Surround mode Afterwards, I have run the following: gst-launch-0.10 filesrc location=mkv.0 ! queue ! mux.video_0 avimux name=mux ! queue ! filesink location=house.avi filesrc location=mkv.1 ! queue ! mux.audio_0 But the end result is a unplayable file. This is the output from mplayer: AVI file format detected. [aviheader] Video stream found, -vid 0 [aviheader] Audio stream found, -aid 1 Detected NON-INTERLEAVED AVI file format. VIDEO: [] 0x0 0bpp nan fps -17179870.0 kbps (-2097152.0 kbyte/s) FPS not specified in the header or invalid, use the -fps option. ========================================================================== Forced audio codec: mad Opening audio decoder: [pcm] Uncompressed PCM audio decoder Unknown/missing audio format -> no sound ADecoder init failed :( Cannot find codec for audio format 0x0. Read DOCS/HTML/en/codecs.html! Audio: no sound Video: no video Am I doing something wrong? -- Henrique Ferreiro Garc?a From kekko84 at gmail.com Tue Sep 30 11:59:52 2008 From: kekko84 at gmail.com (Francesco Argese) Date: Tue, 30 Sep 2008 11:59:52 +0200 Subject: [gst-devel] problems sending a theora video through rtp streaming Message-ID: Hi all. I have problems sending a theora flow in streaming through rtp. My two pipeline are the following: Sender 1)gst-launch gstrtpbin name=rtpbin dv1394src ! dvdemux name=demux ! dvdec ! ffmpegcolorspace ! theoraenc ! rtptheorapay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink port=5000 host=172.16.1.56 rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=172.16.1.56 sync=false async=false udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 Receiver 2)gst-launch-0.10 gstrtpbin name=rtpbin udpsrc port=5000 caps="application/x-rtp,media=video,clock-rate=90000,encoding-name=THEORA,profile-level-id=1,payload=96" ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtptheoradepay ! theoradec ! ffmpegcolorspace ! video/x-raw-rgb,bpp=32,endianness=4321,depth=24,red_mask=65280,green_mask=16711680,blue_mask=-16777216,width=720,height=576 ! fakesink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink host=172.16.100.277 port=5005 sync=false async=false The sender seems to works well but at the receiver i obtain the following error: New clock: GstSystemClock ERROR: from element /pipeline0/udpsrc0: Internal data flow error. Additional debug info: gstbasesrc.c(2240): gst_base_src_loop (): /pipeline0/udpsrc0: streaming task paused, reason not-negotiated (-4) Any suggestion? What could be the problem? A pipeline very similar using MPEG4 works well. Regards Francesco Argese From wim.taymans at gmail.com Tue Sep 30 12:22:38 2008 From: wim.taymans at gmail.com (Wim Taymans) Date: Tue, 30 Sep 2008 12:22:38 +0200 Subject: [gst-devel] problems sending a theora video through rtp streaming In-Reply-To: References: Message-ID: <1222770158.7431.48.camel@metal> On Tue, 2008-09-30 at 11:59 +0200, Francesco Argese wrote: > Hi all. > > I have problems sending a theora flow in streaming through rtp. > > My two pipeline are the following: > Sender > 1)gst-launch gstrtpbin name=rtpbin dv1394src ! dvdemux name=demux ! > dvdec ! ffmpegcolorspace ! theoraenc ! rtptheorapay ! > rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink port=5000 > host=172.16.1.56 rtpbin.send_rtcp_src_0 ! udpsink port=5001 > host=172.16.1.56 sync=false async=false udpsrc port=5005 ! > rtpbin.recv_rtcp_sink_0 > > Receiver > 2)gst-launch-0.10 gstrtpbin name=rtpbin udpsrc port=5000 > caps="application/x-rtp,media=video,clock-rate=90000,encoding-name=THEORA,profile-level-id=1,payload=96" > ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtptheoradepay ! theoradec ! > ffmpegcolorspace ! > video/x-raw-rgb,bpp=32,endianness=4321,depth=24,red_mask=65280,green_mask=16711680,blue_mask=-16777216,width=720,height=576 > ! fakesink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 > rtpbin.send_rtcp_src_0 ! udpsink host=172.16.100.277 port=5005 > sync=false async=false > > The sender seems to works well but at the receiver i obtain the following error: > New clock: GstSystemClock > ERROR: from element /pipeline0/udpsrc0: Internal data flow error. > Additional debug info: > gstbasesrc.c(2240): gst_base_src_loop (): /pipeline0/udpsrc0: > streaming task paused, reason not-negotiated (-4) The receiver needs the sampling, width, height and configuration strings. Run the sender pipeline with -v, then copy the caps on the RTP udpsink to the receiver udpsrc caps (and yes, they are huge and they are needed). Wim > > Any suggestion? What could be the problem? A pipeline very similar > using MPEG4 works well. > > Regards > Francesco Argese > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From manish.rana at gmail.com Tue Sep 30 14:40:03 2008 From: manish.rana at gmail.com (Manish Rana) Date: Tue, 30 Sep 2008 18:10:03 +0530 Subject: [gst-devel] Problem in gstrtpbin and AV Transfer---Urgent Message-ID: <8c192ddd0809300540s15390f64j58e9488b146ee19c@mail.gmail.com> Hi All.. I am facing some other problem in RTP Streaming... I am sending the rtp data over the UDP and playing it on the other end. I am recording the data from v4l2src @ 10 fps and voice data from alsasrc to send it across. The transmission starts well as well as i can see the video on the other end. Now in between i starts to take the data from a file. I decode the data and encode again to send it using the same rtpbin and udpsink. Enc, Pay gstrtpbin and udpsink are the element kept common...means i will use these elements in both the cases. the packets goes well and on the other end i can see the AV fine in both the cases.... Now when i again switch back to v4l2src the packets on the receiver end come but the decoder (ffdec_mpeg4) gives error.......and there is no updates on display........ The file i am using is @25fps...... also when i increase the FPS of v4l2src the whole system works well..... at 20fps..... Please let me know if i am missing something or i am doing something wrong way.... Also i tried using different enc and pay elements in both the case..but i i unlink and link to gstrtpbin, there is no data at all..even in the first switch to file transfer.............. :( Please help...i have tried every possible thing to fix this issue....but i am stuck with the same for last couple of days now..... Any input will be appreciated from bottum of heart....... BR Manish Tue, Sep 30, 2008 at 3:52 PM, Wim Taymans wrote: > On Tue, 2008-09-30 at 11:59 +0200, Francesco Argese wrote: > > Hi all. > > > > I have problems sending a theora flow in streaming through rtp. > > > > My two pipeline are the following: > > Sender > > 1)gst-launch gstrtpbin name=rtpbin dv1394src ! dvdemux name=demux ! > > dvdec ! ffmpegcolorspace ! theoraenc ! rtptheorapay ! > > rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink port=5000 > > host=172.16.1.56 rtpbin.send_rtcp_src_0 ! udpsink port=5001 > > host=172.16.1.56 sync=false async=false udpsrc port=5005 ! > > rtpbin.recv_rtcp_sink_0 > > > > Receiver > > 2)gst-launch-0.10 gstrtpbin name=rtpbin udpsrc port=5000 > > > caps="application/x-rtp,media=video,clock-rate=90000,encoding-name=THEORA,profile-level-id=1,payload=96" > > ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtptheoradepay ! theoradec ! > > ffmpegcolorspace ! > > > video/x-raw-rgb,bpp=32,endianness=4321,depth=24,red_mask=65280,green_mask=16711680,blue_mask=-16777216,width=720,height=576 > > ! fakesink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 > > rtpbin.send_rtcp_src_0 ! udpsink host=172.16.100.277 port=5005 > > sync=false async=false > > > > The sender seems to works well but at the receiver i obtain the following > error: > > New clock: GstSystemClock > > ERROR: from element /pipeline0/udpsrc0: Internal data flow error. > > Additional debug info: > > gstbasesrc.c(2240): gst_base_src_loop (): /pipeline0/udpsrc0: > > streaming task paused, reason not-negotiated (-4) > > The receiver needs the sampling, width, height and configuration > strings. Run the sender pipeline with -v, then copy the caps on the RTP > udpsink to the receiver udpsrc caps (and yes, they are huge and they are > needed). > > Wim > > > > > Any suggestion? What could be the problem? A pipeline very similar > > using MPEG4 works well. > > > > Regards > > Francesco Argese > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Vikas.Patel at imgtec.com Tue Sep 30 15:32:01 2008 From: Vikas.Patel at imgtec.com (Vikas Patel) Date: Tue, 30 Sep 2008 19:02:01 +0530 Subject: [gst-devel] Recall: Problem in gstrtpbin and AV Transfer---Urgent Message-ID: Vikas Patel would like to recall the message, "[gst-devel] Problem in gstrtpbin and AV Transfer---Urgent". - This message is subject to Imagination Technologies' e-mail terms: http://www.imgtec.com/e-mail.htm - -------------- next part -------------- An HTML attachment was scrubbed... URL: From Vikas.Patel at imgtec.com Tue Sep 30 15:18:41 2008 From: Vikas.Patel at imgtec.com (Vikas Patel) Date: Tue, 30 Sep 2008 18:48:41 +0530 Subject: [gst-devel] Videoesink Message-ID: Hi, I am creating an elementary video player by using Gstreamer plugins. I am using imagesink for rendering the video. But Facing some problem because I am not able to configure the imagesink window location. Is there any way to configure the window to synchronize with GUI location? Or Is there any other plugin available for video sink? Regards Vikas. - This message is subject to Imagination Technologies' e-mail terms: http://www.imgtec.com/e-mail.htm - -------------- next part -------------- An HTML attachment was scrubbed... URL: From Vikas.Patel at imgtec.com Tue Sep 30 15:18:42 2008 From: Vikas.Patel at imgtec.com (Vikas Patel) Date: Tue, 30 Sep 2008 18:48:42 +0530 Subject: [gst-devel] Problem in gstrtpbin and AV Transfer---Urgent References: <8c192ddd0809300540s15390f64j58e9488b146ee19c@mail.gmail.com> Message-ID: ________________________________ From: Manish Rana [mailto:manish.rana at gmail.com] Sent: Tuesday, September 30, 2008 6:10 PM To: Discussion of the development of GStreamer Subject: [gst-devel] Problem in gstrtpbin and AV Transfer---Urgent Hi All.. I am facing some other problem in RTP Streaming... I am sending the rtp data over the UDP and playing it on the other end. I am recording the data from v4l2src @ 10 fps and voice data from alsasrc to send it across. The transmission starts well as well as i can see the video on the other end. Now in between i starts to take the data from a file. I decode the data and encode again to send it using the same rtpbin and udpsink. Enc, Pay gstrtpbin and udpsink are the element kept common...means i will use these elements in both the cases. the packets goes well and on the other end i can see the AV fine in both the cases.... Now when i again switch back to v4l2src the packets on the receiver end come but the decoder (ffdec_mpeg4) gives error.......and there is no updates on display........ The file i am using is @25fps...... also when i increase the FPS of v4l2src the whole system works well..... at 20 fps..... Please let me know if i am missing something or i am doing something wrong way.... Also i tried using different enc and pay elements in both the case..but i i unlink and link to gstrtpbin, there is no data at all..even in the first switch to file transfer.............. :( Please help...i have tried every possible thing to fix this issue....but i am stuck with the same for last couple of days now..... Any input will be appreciated from bottum of heart....... BR Manish Tue, Sep 30, 2008 at 3:52 PM, Wim Taymans wrote: On Tue, 2008-09-30 at 11:59 +0200, Francesco Argese wrote: > Hi all. > > I have problems sending a theora flow in streaming through rtp. > > My two pipeline are the following: > Sender > 1)gst-launch gstrtpbin name=rtpbin dv1394src ! dvdemux name=demux ! > dvdec ! ffmpegcolorspace ! theoraenc ! rtptheorapay ! > rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink port=5000 > host=172.16.1.56 rtpbin.send_rtcp_src_0 ! udpsink port=5001 > host=172.16.1.56 sync=false async=false udpsrc port=5005 ! > rtpbin.recv_rtcp_sink_0 > > Receiver > 2)gst-launch-0.10 gstrtpbin name=rtpbin udpsrc port=5000 > caps="application/x-rtp,media=video,clock-rate=90000,encoding-name=THEOR A,profile-level-id=1,payload=96" > ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtptheoradepay ! theoradec ! > ffmpegcolorspace ! > video/x-raw-rgb,bpp=32,endianness=4321,depth=24,red_mask=65280,green_mas k=16711680,blue_mask=-16777216,width=720,height=576 > ! fakesink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 > rtpbin.send_rtcp_src_0 ! udpsink host=172.16.100.277 port=5005 > sync=false async=false > > The sender seems to works well but at the receiver i obtain the following error: > New clock: GstSystemClock > ERROR: from element /pipeline0/udpsrc0: Internal data flow error. > Additional debug info: > gstbasesrc.c(2240): gst_base_src_loop (): /pipeline0/udpsrc0: > streaming task paused, reason not-negotiated (-4) The receiver needs the sampling, width, height and configuration strings. Run the sender pipeline with -v, then copy the caps on the RTP udpsink to the receiver udpsrc caps (and yes, they are huge and they are needed). Wim > > Any suggestion? What could be the problem? A pipeline very similar > using MPEG4 works well. > > Regards > Francesco Argese > > ------------------------------------------------------------------------ - > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------ - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel - This message is subject to Imagination Technologies' e-mail terms: http://www.imgtec.com/e-mail.htm - -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Tue Sep 30 15:48:43 2008 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 30 Sep 2008 16:48:43 +0300 Subject: [gst-devel] asfdemux In-Reply-To: <9D5E1752379A43408015F7FE984661157827A7@CHNEXVS01.VSNLXCHANGE.COM> References: <9D5E1752379A43408015F7FE984661157827A4@CHNEXVS01.VSNLXCHANGE.COM> <1222508207.2544.8.camel@putamadre> <9D5E1752379A43408015F7FE984661157827A7@CHNEXVS01.VSNLXCHANGE.COM> Message-ID: <48E22E3B.6070803@hora-obscura.de> hi, Irfan Shaikh schrieb: > > Hi, > > Can you please suggest me what all steps are required so as to port > gstasfmux.c on Gstreamer. > I would port it to gstreamer as soon as possible, if you provide me > some hints. > These links have some info http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/chapter-porting.html http://gstreamer.freedesktop.org/data/doc/gstreamer/head/pwg/html/chapter-porting.html Stefan > > > Regards, > Irfan > > > -----Original Message----- > From: Edward Hervey [mailto:bilboed at gmail.com] > Sent: Sat 9/27/2008 3:06 PM > To: Discussion of the development of GStreamer > Subject: Re: [gst-devel] asfdemux > > > The asf muxer code in that directory is code that hasn't been ported > to gstreamer 0.10. It is therefore not built. > > Porting it to 0.10 would be much appreciated though :) > > > Edward > > On Sat, 2008-09-27 at 14:34 +0530, Irfan Shaikh wrote: > > > > Hi all, > > > > In the following > > path /home/GStreamer/gst-plugins-ugly-0.10.9/gst/asfdemux/ > > > > We have gstasfmux.c and gstasfmux.h, i think which provides ASF muxer > > functionalitiy. > > > > When ever i use, { gst-inspeact | grep asf } ,I get only asdemux as a > > available plug-in element {which is an ASF demuxer}. > > > > Why i dont get an ASF muxer element on ASF when i use gst-inspect. > > > > How do i use gstasfmux.c and gstasfmux.h to make as gstreamer plug-in > > element and see the muxer element when i do gst-inspect. > > > > > > Regards, > > Irfan.... > > > > > > > > This message (including any attachment) is confidential and may be > > legally privileged. Access to this message by anyone other than the > > intended recipient(s) listed above is unauthorized. If you are not the > > intended recipient you are hereby notified that any disclosure, > > copying, or distribution of the message, or any action taken or > > omission of action by you in reliance upon it, is prohibited and may > > be unlawful. Please immediately notify the sender by reply e-mail and > > permanently delete all copies of the message if you have received this > > message in error. > > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win > great prizes > > Grand prize is a trip for two to an Open Source event anywhere in > the world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > _______________________________________________ gstreamer-devel > mailing list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the > world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > This message (including any attachment) is confidential and may be > legally privileged. Access to this message by anyone other than the > intended recipient(s) listed above is unauthorized. If you are not the > intended recipient you are hereby notified that any disclosure, > copying, or distribution of the message, or any action taken or > omission of action by you in reliance upon it, is prohibited and may > be unlawful. Please immediately notify the sender by reply e-mail and > permanently delete all copies of the message if you have received this > message in error. > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From felipe.contreras at gmail.com Tue Sep 30 15:53:51 2008 From: felipe.contreras at gmail.com (Felipe Contreras) Date: Tue, 30 Sep 2008 16:53:51 +0300 Subject: [gst-devel] Videoesink In-Reply-To: References: Message-ID: <94a0d4530809300653v1caf255fg2ba341f63fe2a5dc@mail.gmail.com> On Tue, Sep 30, 2008 at 4:18 PM, Vikas Patel wrote: > Hi, > > > > I am creating an elementary video player by using Gstreamer plugins. I am > using imagesink for rendering the video. > > But Facing some problem because I am not able to configure the imagesink > window location. > > Is there any way to configure the window to synchronize with GUI location? > > Or Is there any other plugin available for video sink? You can check my gst-player for reference: http://code.google.com/p/gst-player/ gst-backend.c:backend_play -- Felipe Contreras From kekko84 at gmail.com Tue Sep 30 16:01:32 2008 From: kekko84 at gmail.com (Francesco Argese) Date: Tue, 30 Sep 2008 16:01:32 +0200 Subject: [gst-devel] ffenc_h264 plug-in ??? In-Reply-To: <9D5E1752379A43408015F7FE984661157827B8@CHNEXVS01.VSNLXCHANGE.COM> References: <9D5E1752379A43408015F7FE984661157827B8@CHNEXVS01.VSNLXCHANGE.COM> Message-ID: In order to have that plugin you have to install Gst-ffmpeg (http://gstreamer.freedesktop.org/modules/gst-ffmpeg.html). For this reason it is better if you install gstreamer and all the necessary plugins from source in orer to have more control on the plugins intalled. As written from Colin Guthrie, in some Linux distribution these plugins aren't packetized so you are obliged to install from source (I don't know if Mandriva is one of that). Installing from source those versions I have ffenc_h263 together to other useful patented codec such as mpeg4 and others. Hovewer, if you are working on a raw video created by you (for example captured from camera), you could use theora to encode the video (the name of the GstElement to do that is theoraenc), vorbis to encode music and speex to encode voice. They work well and, probably, you have it already installed. Hi Francesco Argese From mypuppy9999 at gmail.com Tue Sep 30 16:44:33 2008 From: mypuppy9999 at gmail.com (puppy little) Date: Tue, 30 Sep 2008 22:44:33 +0800 Subject: [gst-devel] problem with mpeg-1 av synchronization and oss gstosshelper.c:246 error Message-ID: <18ec06590809300744l30069ea8m951b18b587240306@mail.gmail.com> HI all, I develop a media player based gstreamer on S3C2440 board. my gstreamer version is latest . when i construct a pipeline: gst-launch filesrc location=/tmp/MPEG-1.mpg ! mpegdemux name=t ! mad ! queue2 ! audioconvert ! audioresample ! queue2 ! osssink sync=0 t. ! mpeg2dec ! queue2 ! ffmpegcolorspace ! fbdevsink sync=0 *the video could correct playback ,but the audio is great lagged. * and there is a error report: GST_DEBUG=1 gst-launch filesrc location=/tmp/MPEG-1.mpg ! mpegdemux name=t ! mad ! queue2 ! audioconvert ! audioresample ! queue2 ! osssink sync=0 t. ! mpeg2dec ! queue2 ! ffmpegcolorspace ! fbdevsink sync=0 Setting pipeline to PAUSED ... Pipeline is PREROLLING ... 0:00:14.091585000 1584 0xc43a8 ERROR oss gstosshelper.c:246:gst_oss_helper_rate_probe_check: Driver bug recognized (driver does not round rates correctly). Please file a bug report. Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstAudioSinkClock besides , when i use pipeline like this : gst-launch filesrc location=/tmp/MPEG-1.mpg ! mpegdemux ! queue ! mad ! audioconvert ! audioresample ! osssink it is reported: WARNING: erroneous pipeline: could not link queue0 to mad0 Regards! -------------- next part -------------- An HTML attachment was scrubbed... URL: From manish.rana at gmail.com Tue Sep 30 17:13:32 2008 From: manish.rana at gmail.com (Manish Rana) Date: Tue, 30 Sep 2008 20:43:32 +0530 Subject: [gst-devel] Recall: Problem in gstrtpbin and AV Transfer---Urgent In-Reply-To: References: Message-ID: <8c192ddd0809300813o43c0d4eex88dc122fb2068fc3@mail.gmail.com> Hey.... I didnt get what is the problem in the email.....??? Please let me know... Thanks Manish On Tue, Sep 30, 2008 at 7:02 PM, Vikas Patel wrote: > Vikas Patel would like to recall the message, "[gst-devel] Problem in > gstrtpbin and AV Transfer---Urgent". > > > > - > This message is subject to Imagination Technologies' e-mail terms: > http://www.imgtec.com/e-mail.htm > - > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From vlyamtsev at gmail.com Tue Sep 30 17:45:17 2008 From: vlyamtsev at gmail.com (Victor lyamtsev) Date: Tue, 30 Sep 2008 11:45:17 -0400 Subject: [gst-devel] RedHat EL4 installation - no registry problem Message-ID: <76224b100809300845k66abf16fy3c06141cf486d88a@mail.gmail.com> Hello, I installed gst on RedHat EL4... Everything is built but when I try to run samples runtime cannot find registry, or scheduler. What is the default location? How can I force alternative one ( form configure ) ?? Thanks, -V -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien.isorce at gmail.com Tue Sep 30 22:15:06 2008 From: julien.isorce at gmail.com (Julien Isorce) Date: Tue, 30 Sep 2008 22:15:06 +0200 Subject: [gst-devel] asfdemux In-Reply-To: <48E22E3B.6070803@hora-obscura.de> References: <9D5E1752379A43408015F7FE984661157827A4@CHNEXVS01.VSNLXCHANGE.COM> <1222508207.2544.8.camel@putamadre> <9D5E1752379A43408015F7FE984661157827A7@CHNEXVS01.VSNLXCHANGE.COM> <48E22E3B.6070803@hora-obscura.de> Message-ID: <180a127d0809301315ja39ef69v55006144658c062@mail.gmail.com> Hi, I am also interested to use the asfmux element. With the current ffmpeg code (libavcodec around september), the ffmux_asf crash (log: gst-launch fakesrc ! ffmux_asf ! fakesink -> Gstreamer developpers were too lazy to handle that error etc... something like that.) ( and after doing some fixes in gst-ffmpeg because libavcodec api has changed a little bit) But I cannot do the porting 0.8 to 0.10 because a lack of time and I do not know a lot about muxers. Julien 2008/9/30 Stefan Kost > hi, > > > Irfan Shaikh schrieb: > > > > Hi, > > > > Can you please suggest me what all steps are required so as to port > > gstasfmux.c on Gstreamer. > > I would port it to gstreamer as soon as possible, if you provide me > > some hints. > > > These links have some info > > http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/chapter-porting.html > > http://gstreamer.freedesktop.org/data/doc/gstreamer/head/pwg/html/chapter-porting.html > > Stefan > > > > > > Regards, > > Irfan > > > > > > -----Original Message----- > > From: Edward Hervey [mailto:bilboed at gmail.com] > > Sent: Sat 9/27/2008 3:06 PM > > To: Discussion of the development of GStreamer > > Subject: Re: [gst-devel] asfdemux > > > > > > The asf muxer code in that directory is code that hasn't been ported > > to gstreamer 0.10. It is therefore not built. > > > > Porting it to 0.10 would be much appreciated though :) > > > > > > Edward > > > > On Sat, 2008-09-27 at 14:34 +0530, Irfan Shaikh wrote: > > > > > > Hi all, > > > > > > In the following > > > path /home/GStreamer/gst-plugins-ugly-0.10.9/gst/asfdemux/ > > > > > > We have gstasfmux.c and gstasfmux.h, i think which provides ASF muxer > > > functionalitiy. > > > > > > When ever i use, { gst-inspeact | grep asf } ,I get only asdemux as a > > > available plug-in element {which is an ASF demuxer}. > > > > > > Why i dont get an ASF muxer element on ASF when i use gst-inspect. > > > > > > How do i use gstasfmux.c and gstasfmux.h to make as gstreamer plug-in > > > element and see the muxer element when i do gst-inspect. > > > > > > > > > Regards, > > > Irfan.... > > > > > > > > > > > > This message (including any attachment) is confidential and may be > > > legally privileged. Access to this message by anyone other than the > > > intended recipient(s) listed above is unauthorized. If you are not the > > > intended recipient you are hereby notified that any disclosure, > > > copying, or distribution of the message, or any action taken or > > > omission of action by you in reliance upon it, is prohibited and may > > > be unlawful. Please immediately notify the sender by reply e-mail and > > > permanently delete all copies of the message if you have received this > > > message in error. > > > > > > > > ------------------------------------------------------------------------- > > > This SF.Net email is sponsored by the Moblin Your Move Developer's > > challenge > > > Build the coolest Linux based applications with Moblin SDK & win > > great prizes > > > Grand prize is a trip for two to an Open Source event anywhere in > > the world > > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > > > _______________________________________________ gstreamer-devel > > mailing list gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > > challenge > > Build the coolest Linux based applications with Moblin SDK & win great > > prizes > > Grand prize is a trip for two to an Open Source event anywhere in the > > world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > This message (including any attachment) is confidential and may be > > legally privileged. Access to this message by anyone other than the > > intended recipient(s) listed above is unauthorized. If you are not the > > intended recipient you are hereby notified that any disclosure, > > copying, or distribution of the message, or any action taken or > > omission of action by you in reliance upon it, is prohibited and may > > be unlawful. Please immediately notify the sender by reply e-mail and > > permanently delete all copies of the message if you have received this > > message in error. > > > > ------------------------------------------------------------------------ > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > > Build the coolest Linux based applications with Moblin SDK & win great > prizes > > Grand prize is a trip for two to an Open Source event anywhere in the > world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From wingo at pobox.com Tue Sep 30 21:34:42 2008 From: wingo at pobox.com (Andy Wingo) Date: Tue, 30 Sep 2008 21:34:42 +0200 Subject: [gst-devel] problem with mpeg-1 av synchronization and oss gstosshelper.c:246 error In-Reply-To: <18ec06590809300744l30069ea8m951b18b587240306@mail.gmail.com> (puppy little's message of "Tue, 30 Sep 2008 22:44:33 +0800") References: <18ec06590809300744l30069ea8m951b18b587240306@mail.gmail.com> Message-ID: 'Sup dog, On Tue 30 Sep 2008 16:44, "puppy little" writes: > osssink sync=0 > > the video could correct playback ,but the audio is great lagged. Well, you're not synchronizing on the audio. > 0:00:14.091585000 1584 0xc43a8 ERROR oss > gstosshelper.c:246:gst_oss_helper_rate_probe_check: Driver bug recognized > (driver does not round rates correctly). Please file a bug report. Sounds like your kernel driver has problems, so the bug report should be with them. But OSS is so old, yo -- have you tried ALSA? Word, Andy -- http://wingolog.org/ From xxopxe at gmail.com Thu Sep 25 17:37:01 2008 From: xxopxe at gmail.com (Jorge) Date: Thu, 25 Sep 2008 12:37:01 -0300 Subject: [gst-devel] Seeking multifilesrc In-Reply-To: <48DA2A09.5040107@hora-obscura.de> References: <1221755299.16314.22.camel@matroskin> <48DA2A09.5040107@hora-obscura.de> Message-ID: <1222357021.10003.12.camel@matroskin> Hi. Trying GST_FORMAT_DEFAULT i get the same behaviour When querying position (hope it's ok posting python code): pos = self.player.pipeline.query_position(gst.FORMAT_DEFAULT, None)[0] gst.QueryError: query failed When seeking: no errors, but nothing happens. I can play and pause. Something i've found is that when you pause the pipeline the multifilesrc's "index" property is valued as the current frame. If i change its value and resume it starts playing from the new value. I guess i could manipulate this value to advance frame by frame, but then i should have to feed a single frame by playing and pausing inmediately... It would be much uglier that proper seek. Jorge On Wed, 2008-09-24 at 14:52 +0300, Stefan Kost wrote: > Jorge schrieb: > > Hi to all. > > > > I'm trying to write a stop-motion creation app, in the vein of > > http://developer.skolelinux.no/info/studentgrupper/2005-hig-stopmotion/index.php > > but using python, gtk, and well, gstreamer. It's my first using all > > three technologies, tough :p > > What i'm stump on is on how step frame by frame forward and backwards in > > a video loaded trough multifilesrc. I was trying to seek using > > gst.FORMAT_BUFFERS as to move one frame at a time, but it doesn't work. > > When i attempt a query_position on the pipeline, i get a "query failed". > > > Can you try |GST_FORMAT_DEFAULT? From duane.mckinney at gmail.com Thu Sep 25 18:10:56 2008 From: duane.mckinney at gmail.com (Duane McKinney) Date: Thu, 25 Sep 2008 09:10:56 -0700 Subject: [gst-devel] GPP + DSPs In-Reply-To: <94a0d4530809250523j221d5f02u3a98b462bb7496f@mail.gmail.com> References: <19265608.432581222282253669.JavaMail.root@hrndva-web15-z02> <94a0d4530809250523j221d5f02u3a98b462bb7496f@mail.gmail.com> Message-ID: <48dbb812.0913c00a.57fb.779a@mx.google.com> I am doing something similar, I created a live source to stream data out of the DSP using existing functions. What would be the other options for doing this? -----Original Message----- From: Felipe Contreras [mailto:felipe.contreras at gmail.com] Sent: Thursday, September 25, 2008 5:24 AM To: gstreamer-devel at lists.sourceforge.net Subject: Re: [gst-devel] GPP + DSPs On Wed, Sep 24, 2008 at 9:50 PM, wrote: > Hi all, > > I'm considering using GStreamer for an application is which GStreamer will span a GPP and multiple DSPs. The GPP and DSPs are connected via PCI. So GStreamer will have to proxy buffers and messages across the PCI bus. > > Can you point me to documentation, info, case studies, etc. that will help me to architect this solution? What kind of DSP/GPP? We can propose different ways to do it at GStreamer level, but ultimately you would have to communicate with the hardware, is there code that is doing that already? -- Felipe Contreras ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From manish.rana at gmail.com Thu Sep 25 19:05:56 2008 From: manish.rana at gmail.com (Manish Rana) Date: Thu, 25 Sep 2008 22:35:56 +0530 Subject: [gst-devel] About multi-thread in GStreamer In-Reply-To: <18749687.205481222320440119.JavaMail.coremail@bj126app33.126.com> References: <18749687.205481222320440119.JavaMail.coremail@bj126app33.126.com> Message-ID: <8c192ddd0809251005m1bff88e0w32f3781b70af3c97@mail.gmail.com> Hi, You can try following: 1. Take the data for the camera. 2. Split the stream in to tow using the Tee element 3. Now use queue in front of both the src pads of the Tee 4. Connect one queue to display for Preview 5. Connect the other src pad of the queue to encoder->pay->rtpbin->udpsink And enjoy the Streaming using Gstreamer..... :) Manish On Thu, Sep 25, 2008 at 10:57 AM, liangzhihong1984 wrote: > I captured video stream from a USB camera, now want to send it to other > users with udpsink, > and at the same time playback the video stream locally, how should I > implement this through gst-launch > command line, as well as C language? > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gstelzz at yahoo.fr Thu Sep 25 21:48:02 2008 From: gstelzz at yahoo.fr (Aurelien Grimaud) Date: Thu, 25 Sep 2008 21:48:02 +0200 Subject: [gst-devel] About multi-thread in GStreamer In-Reply-To: <18749687.205481222320440119.JavaMail.coremail@bj126app33.126.com> References: <18749687.205481222320440119.JavaMail.coremail@bj126app33.126.com> Message-ID: <48DBEAF2.5090303@yahoo.fr> Hi, You want to split stream into two parts. Use tee element for that. This pipeline will duplicate stream. v4l2src ! tee name=t t. ! encoder ! packetizer ! udpsink t. ! xvimagesink Note : there is only one thread running in this pipeline. To prevent one branch from being slowed by the other one, you have to run one in a thread. For this use queue element : v4l2src ! tee name=t t. ! queue ! encoder ! packetizer ! udpsink t. ! xvimagesink Aurelien liangzhihong1984 a ??crit : > I captured video stream from a USB camera, now want to send it to > other users with udpsink, > and at the same time playback the video stream locally, how should I > implement this through gst-launch > command line, as well as C language? > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From gmane at colin.guthr.ie Thu Sep 25 23:18:50 2008 From: gmane at colin.guthr.ie (Colin Guthrie) Date: Thu, 25 Sep 2008 22:18:50 +0100 Subject: [gst-devel] Some advice In-Reply-To: <3c1737210809221055l7cd42603i93d011f44461a547@mail.gmail.com> References: <3c1737210809221055l7cd42603i93d011f44461a547@mail.gmail.com> Message-ID: (Sorry for the late reply... I'm not subscribed and post via Gmane, but hoped that after having my first message validated manually my subsequent messages would pass. Alas that didn't happen, so I've now subscribed officially and disabled delivery so gmane can still be used :)) Michael Smith wrote: >> 1) Is it possible to find out which sink is really used when you use >> gconfsink, autoaudiosink or halsink? > > These are all bins (they contain other elements); so you can just > iterate (possibly recursively) over all the elements in them, looking > for sinks, and then look at the type of the sink. autoaudiosink and > halsink are both pretty easy, but gconfsink lets you have a > more-or-less arbitrary bin, which could even contain multiple sinks > (though that would be rare). OK, thanks for the info. I'll look into doing that :) >> 2) When streaming data from shoutcast, it is possible to decode the >> StreamTitle= header via icydemux. I have hacked phonon-gstreamer backed >> to do this now (in a similar way to how playbin does it). However it >> only extracts the title. >> >> When I use the Xine backend it appears to be able to extract inline >> metadata out of the stream content itself (e.g. when it finds a vorbis >> comment header or an id3 tag). That's as much as I can tell from >> observation. I've hacked the phonon-gstreamer to be able to "guess" the >> artist name from the StreamTitle but it's really not very nice. So my >> main question is, can the decoders handle inline metadata decoding and >> if so how can I enabled this? >> > > I guess it'd be useful to point us at a stream that behaves like this. Oh, erm, yeah that would help wouldn't it. I've been playing with this one: http://87.117.200.136:8010 (it's not my musical taste: http://www.last.fm/user/coling) It seems to be an AAC stream and the xine backend for phonon can update the artist and title streams properly from this it seems. > 'icydemux' will extract metadata properly from shoutcast streams (so > long as the source is providing this data). Vorbis streams have inline > metadata that should be handled automatically, without anything > special being needed. I've never seen inline id3 being streamed, nor > have I ever seen software that would handle it at all. Yeah I think I inferred too much from my initial observations for which I apologise :) I see now with more testing that the xine metadata extraction seems to be using ORGANIZATION for ARTIST and ALBUM fields on streams. The above mentioned stream has it's artist/album data extracted properly with xine, but it doesn't seem to happen with gstreamer (at least I don't get any TAG messages on the bus). I guess this is a shortcoming of the faad implementation? > So you shouldn't need to do anything special apart from requesting the > http sources to get shoutcast-style metadata. From what I can tell icydebux only parses the StreamTitle= header and sets the TITLE metadata accordingly. This is useful but inline metadata extraction is nicer. As you've said, vorbis comments should work fine, so I'll have to find some streams that do this so I can test. Thanks for your help Mike :) Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] From dbourgoyne at austin.rr.com Fri Sep 26 04:35:16 2008 From: dbourgoyne at austin.rr.com (David Bourgoyne) Date: Thu, 25 Sep 2008 21:35:16 -0500 Subject: [gst-devel] GPP + DSPs In-Reply-To: <94a0d4530809250523j221d5f02u3a98b462bb7496f@mail.gmail.com> References: <19265608.432581222282253669.JavaMail.root@hrndva-web15-z02> <94a0d4530809250523j221d5f02u3a98b462bb7496f@mail.gmail.com> Message-ID: <48DC4A64.7060102@austin.rr.com> Felipe Contreras wrote: > On Wed, Sep 24, 2008 at 9:50 PM, wrote: > >> Hi all, >> >> I'm considering using GStreamer for an application is which GStreamer will span a GPP and multiple DSPs. The GPP and DSPs are connected via PCI. So GStreamer will have to proxy buffers and messages across the PCI bus. >> >> Can you point me to documentation, info, case studies, etc. that will help me to architect this solution? >> > > What kind of DSP/GPP? > > We can propose different ways to do it at GStreamer level, but > ultimately you would have to communicate with the hardware, is there > code that is doing that already? > > Assume an OMAP DSP with gst-openmax. A GPP running Linux uses v4l as a gstreamer src and needs to send buffers to the OMAP DSP to be encoded which then sends buffers back to the GPP for transmission over ethernet. There needs to be a gstreamer proxy (marshalling) that connects the GPP GStreamer Bin to the OMAP DSP GStreamer Bin. From irfanshaikh at tataelxsi.co.in Fri Sep 26 06:51:09 2008 From: irfanshaikh at tataelxsi.co.in (Irfan Shaikh) Date: Fri, 26 Sep 2008 10:21:09 +0530 Subject: [gst-devel] Issue in muxing g726 is ASF using following pipelines Message-ID: <9D5E1752379A43408015F7FE984661157827A0@CHNEXVS01.VSNLXCHANGE.COM> Hi all, I am able to mux the audiotestsrc using following and get a o/p file generated in ASF container format, gst-launch audiotestsrc ! audioconvert ! ffenc_wmav1 ! ffmux_asf ! filesink location=/root/Desktop/abc.ASF But same thing fails is i use ffenc_g726, and shows following warning. gst-launch audiotestsrc ! audioconvert ! ffenc_g726 ! ffmux_asf ! filesink location=/root/Desktop/abc.ASF WARNING: erroneous pipeline: could not link ffenc_g7260 to ffmux_asf0 I have also tried it using, gst-launch audiotestsrc ! audioconvert ! ffenc_g726 ! ffdec_g726 ! ffenc_wmav1 ! ffmux_asf ! filesink location=/root/Desktop/abc.ASF But nothing works, So the question is how do i mux g726 audio in ASF. Am i doing any mistake ?? Thanks in Advance, Regards, Irfan This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ved.kpl at gmail.com Fri Sep 26 09:07:47 2008 From: ved.kpl at gmail.com (ved kpl) Date: Fri, 26 Sep 2008 12:37:47 +0530 Subject: [gst-devel] video freeze while playing a file without audio In-Reply-To: <785339900809252206y30480e08ve07b971c4a6cc5cf@mail.gmail.com> References: <5559508B2BD31A44B1C267F4E6EF471D974180@hz-email05.apac.arcsoft.corp> <785339900809252206y30480e08ve07b971c4a6cc5cf@mail.gmail.com> Message-ID: <7496c23f0809260007m7fc6992vab9c355bc0b242a3@mail.gmail.com> Hi, If you are not pushing any audiodata, then with what element is the audiosink element linked? Looks like, you add the audiosink in the pipeline right at the start. So when set the pipeline is set to playing and if the file has only video data, then audiosink will never complete the state change to PAUSED. (wont preroll) You need to add the elements after the demux (decoders, sinks), later., based on the new pads fired by the demuxer. Ved 2008/9/26 sudarshan bisht : > Hi , > Can u please tell me how you are running pipeline ? using > gst-launch or using some application written by you ? > > On Fri, Sep 26, 2008 at 6:15 AM, Chen Yong(Ever)[??] > wrote: >> >> Thanks your reply. In my example, we don't push any audio data to >> down-streaming element, only push video data, but video is freeezed. >> >> >> >Hi , >> > >> > When u don't have audio data in your container file then that >> > time >> >don't create pad for that stream in plugin code i.e. dont try to push >> > audio >> >data to next element . >> > In your case what is happening is that you dont have audio data >> > even >> >then you are trying to push some junk data to next element and in that >> > case >> >next element is returning some other return value than GST_FLOW_OK from >> > its >> >chain function . >> > To debug this just see return value of gst_pad_push( ) fucntion , >> >this value should be GST_FLOW_OK . >> >> >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> challenge >> Build the coolest Linux based applications with Moblin SDK & win great >> prizes >> Grand prize is a trip for two to an Open Source event anywhere in the >> world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > -- > Regards, > > Sudarshan Bisht > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From yongchen at arcsoft.com.cn Fri Sep 26 10:11:18 2008 From: yongchen at arcsoft.com.cn (=?gb2312?B?Q2hlbiBZb25nKEV2ZXIpW7PC08Jd?=) Date: Fri, 26 Sep 2008 16:11:18 +0800 Subject: [gst-devel] video freeze at Trick-mode high-speed Message-ID: <5559508B2BD31A44B1C267F4E6EF471D97423B@hz-email05.apac.arcsoft.corp> Hi, all, I am developing an element to implement source + demux. It have two pads (video + audio). When it playback a file with audio and video at 1.0 speed, it is ok, while it is trick-mode forward scan mode(32x speed), it only send video data, but video will be freezed, why and how to resolve? Best regards, Ever Chen -------------- next part -------------- An HTML attachment was scrubbed... URL: From liangzhihong1984 at 126.com Fri Sep 26 10:53:55 2008 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Fri, 26 Sep 2008 16:53:55 +0800 (CST) Subject: [gst-devel] About theoraenc plugin Message-ID: <194810.799311222419235949.JavaMail.coremail@bj126app43.126.com> Where can I get the theora elements? -------------- next part -------------- An HTML attachment was scrubbed... URL: From linguang_wang at astrocom.cn Fri Sep 26 10:56:47 2008 From: linguang_wang at astrocom.cn (=?gb2312?B?zfXB1rni?=) Date: Fri, 26 Sep 2008 16:56:47 +0800 Subject: [gst-devel] Subject: how to set "max-size-time" property of queue ? Message-ID: <200809261656460627925@astrocom.cn> Hi,all I have done it rightly, just like this: g_object_set(G_OBJECT(aQueue), "max-size-time", (guint64)0, NULL); -------------------------------------------------------------------------------------------- Hi, all developers! When I seted the "max-size-time" property of queue in my application, it printed out one error such as: GLib-GObject-WARNING **: IA__g_object_set_valist: object class `GstQueue' has no property named `\x80?^@\u0001' I didn't know what is the matter. In my codes, the setting codes is: g_object_set(G_OBJECT(aQueue), "max-size-time", 0, NULL); But if I used gst-launch like this: gst-launch neonhttpsrc uri=$ ! qtdemux ! queue max-size-buffers=0 max-size-time=0 ! faad ! osssink sync=false it is ok.Is there anyone to tell me some information? Looking forward for answers. Thanks in advance! 2008-09-26 2008-09-26 linguang_wang at astrocom.cn -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/gif Size: 344 bytes Desc: not available URL: From bisht.sudarshan at gmail.com Fri Sep 26 14:06:57 2008 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Fri, 26 Sep 2008 17:36:57 +0530 Subject: [gst-devel] video freeze while playing a file without audio In-Reply-To: <5559508B2BD31A44B1C267F4E6EF471D97424C@hz-email05.apac.arcsoft.corp> References: <5559508B2BD31A44B1C267F4E6EF471D97424C@hz-email05.apac.arcsoft.corp> Message-ID: <785339900809260506h4433feb1re3eaafbc2c8eb8bf@mail.gmail.com> Hi , In your application you have to check if any stream is not present then dont connect related elemet to the pipeline . You might be using some callback funtion to connect demuxer src pad's with decoder's sink pad . There you check it u get a valid audio src pad from demuxer then only link those element with the main bin or pipeline . On Fri, Sep 26, 2008 at 2:18 PM, Chen Yong(Ever)[??] < yongchen at arcsoft.com.cn> wrote: > Hi, > We write an application to create a pipeline, add connect all pads > to all sinks, > and simply set pipeline status to playing. In fact, this problem also occur > in other case, maybe we playback a mpeg file, but while forward scan with > high-speed, we should send video only(I picture) , video is free zed, too. > > >From: "sudarshan bisht" > >Subject: Re: [gst-devel] video freeze while playing a file without > audio > >To: "Discussion of the development of GStreamer" > > >Message-ID: > <785339900809252206y30480e08ve07b971c4a6cc5cf at mail.gmail.com> > >Content-Type: text/plain; charset="gb2312" > > >Hi , > Can u please tell me how you are running pipeline ? using > >gst-launch or using some application written by you ? > > >On Fri, Sep 26, 2008 at 6:15 AM, Chen Yong(Ever)[??] < > >yongchen at arcsoft.com.cn> wrote: > > > Thanks your reply. In my example, we don't push any audio data to > > down-streaming element, only push video data, but video is freeezed. > > > > > > >Hi , > > > > > > When u don't have audio data in your container file then that > time > > >don't create pad for that stream in plugin code i.e. dont try to push > > audio > > >data to next element . > > > In your case what is happening is that you dont have audio data > even > > >then you are trying to push some junk data to next element and in that > > case > > >next element is returning some other return value than GST_FLOW_OK from > > its > > >chain function . > > > To debug this just see return value of gst_pad_push( ) fucntion , > > >this value should be GST_FLOW_OK . > > > > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From ambilyn at gmail.com Sat Sep 27 19:32:11 2008 From: ambilyn at gmail.com (Ambily N) Date: Sat, 27 Sep 2008 23:02:11 +0530 Subject: [gst-devel] reg:non-conventional encoder/decoder plugin development Message-ID: <9b285a9f0809271032l387ab7d9j47ff062a937908e6@mail.gmail.com> Hi We are trying to develop gstreamer plugins for one hardware H.264 encoder and decoder. One issue we are facing is, the encoder library doesnt provide an interface for providing an input buffer for accepting uncompressed video. Instead it captures from HDMI/Component video/DVI interface,for which there exists an option for selection. Similary, the decoder doesnt provide a decoded output buffer. Instead it sends output to HDMI/DVI. So our understanding is , due to these restrictions,the plugins cant be developed as normal trasnsform plugins. Instead the encoder plugin need to be developed as a source plugin which generates encoded data and the decoder plugin can be developed as sink plugin which handles both decoding and rendering. Is this understanding correct and is this a right approach? could experts please comment on this? Regards Ambily -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Sun Sep 28 17:06:21 2008 From: bilboed at gmail.com (Edward Hervey) Date: Sun, 28 Sep 2008 17:06:21 +0200 Subject: [gst-devel] PiTiVi pre-release Message-ID: <1222614381.2575.6.camel@putamadre> Hi everyone, I've just finished merging the SoC branches into trunk and doing some needed cleanups and finally done a pre-release available here: http://ftp.gnome.org/pub/GNOME/sources/pitivi/0.11/ The major features are a revamped advanced timeline (now the default) and some webcam/networkstream capture support. In order to use the capture support you'll need the dbus python bindings and HAL running on your machine. Bug reports and comments welcome as usual. Unless a big issue comes up, I expect to do the release Monday the 6th of October. Edward From henrique.ferreiro at gmail.com Mon Sep 29 00:19:17 2008 From: henrique.ferreiro at gmail.com (Henrique Ferreiro =?ISO-8859-1?Q?Garc=EDa?=) Date: Mon, 29 Sep 2008 00:19:17 +0200 Subject: [gst-devel] muxing streams Message-ID: <1222640357.8588.6.camel@macbook> Hi all! I have extracted the video and audio streams from a matroska file getting: mkv.0: JVT NAL sequence, H.264 video @ L 31 mkv.1: ATSC A/52 aka AC-3 aka Dolby Digital stream, 48 kHz,, complete main (CM) 3 front/2 rear, LFE on,, 448 kbit/s reserved Dolby Surround mode Afterwards, I have run the following: gst-launch-0.10 filesrc location=mkv.0 ! queue ! mux.video_0 avimux name=mux ! queue ! filesink location=house.avi filesrc location=mkv.1 ! queue ! mux.audio_0 But the end result is a unplayable file. This is the output from mplayer: AVI file format detected. [aviheader] Video stream found, -vid 0 [aviheader] Audio stream found, -aid 1 Detected NON-INTERLEAVED AVI file format. VIDEO: [] 0x0 0bpp nan fps -17179870.0 kbps (-2097152.0 kbyte/s) FPS not specified in the header or invalid, use the -fps option. ========================================================================== Forced audio codec: mad Opening audio decoder: [pcm] Uncompressed PCM audio decoder Unknown/missing audio format -> no sound ADecoder init failed :( Cannot find codec for audio format 0x0. Read DOCS/HTML/en/codecs.html! Audio: no sound Video: no video Am I doing something wrong? -- Henrique Ferreiro Garc?a From mediadevel at gmail.com Mon Sep 29 17:15:17 2008 From: mediadevel at gmail.com (john david) Date: Mon, 29 Sep 2008 20:45:17 +0530 Subject: [gst-devel] Re : how to set mixer settings Message-ID: Hi, I want to record sound, from my sound card using OSS and encode it to an Ogg/Vorbis file using gstreamer. I used btaudio driver. *Command* that used : gst-launch-0.10 -v osssrc device=/dev/dsp ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg For this, i got *error message*, from element /pipeline0/osssrc0: Could not get/set settings from/on resource. and the detailed error messages are: Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... ERROR: from element /pipeline0/osssrc0: Could not get/set settings from/on resource. Additional debug info: gstosssrc.c(439): gst_oss_src_prepare (): /pipeline0/osssrc0: Unable to set param SETFRAGMENT: Invalid argument ERROR: pipeline doesn't want to preroll. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... FREEING pipeline ... *and how to set mixer settings, and is there any packages i have to install. If then pls tell which packages i have to install. * But i 'm able to capture audio data using (audio device /dev/dsp for oss ) some other application tools and play it without any errors. Regards, John david. -------------- next part -------------- An HTML attachment was scrubbed... URL: From acassis at gmail.com Tue Sep 30 04:04:57 2008 From: acassis at gmail.com (Alan Carvalho de Assis) Date: Mon, 29 Sep 2008 23:04:57 -0300 Subject: [gst-devel] Converting a gst-launch pipeline to C code In-Reply-To: <37367b3a0809291621o50bd35e4j24b6db8d485a082e@mail.gmail.com> References: <37367b3a0809291057j57bf0408jfad9a283ee4f3f47@mail.gmail.com> <2ecbfb170809291121o4b90a647o881c18329e2d4ba2@mail.gmail.com> <37367b3a0809291621o50bd35e4j24b6db8d485a082e@mail.gmail.com> Message-ID: <37367b3a0809291904p6498d38ie61fc72952b5b81d@mail.gmail.com> Hi, 2008/9/29 Alan Carvalho de Assis : > Hi Thijs, > I will test it and let you know if it works. > I got it working but using an ugly/dirt/newbee/dumb hack: static GstElement *decoder, *decvd, *decad; static int i; static void on_pad_added (GstElement *element, GstPad *pad, gpointer data) { GstPad *sinkpad; //GstElement *decoder = (GstElement *) data; if(i==0) decoder = decvd; else decoder = decad; i++; /* We can now link this pad with the vorbis-decoder sink pad */ g_print ("Dynamic pad created\n"); sinkpad = gst_element_get_static_pad (decoder, "sink"); gst_pad_link (pad, sinkpad); gst_object_unref (sinkpad); } I can't figure out how can on_pad_added get access to two decoders? currently I can just pass one: g_signal_connect (demuxer, "pad-added", G_CALLBACK (on_pad_added), decvd); Please, let me know the right way to on_pad_added get the decvd in the first call and get the decad in the second call. I am ashamed because this code, but anyway I am here to learn. Thank you, Alan From bilboed at gmail.com Tue Sep 30 08:15:25 2008 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 30 Sep 2008 08:15:25 +0200 Subject: [gst-devel] ffenc_h264 plug-in ??? In-Reply-To: <48f4838d0809291721y7649ab66nb79a37c95c63207d@mail.gmail.com> References: <9D5E1752379A43408015F7FE984661157827B5@CHNEXVS01.VSNLXCHANGE.COM> <1222703351.7431.36.camel@metal> <1222705199.16208.0.camel@putamadre> <48f4838d0809291721y7649ab66nb79a37c95c63207d@mail.gmail.com> Message-ID: <1222755325.2528.4.camel@putamadre> On Mon, 2008-09-29 at 21:21 -0300, Alexandre wrote: > On Mon, Sep 29, 2008 at 1:19 PM, Edward Hervey > wrote: > On Mon, 2008-09-29 at 17:49 +0200, Wim Taymans wrote: > > On Mon, 2008-09-29 at 20:29 +0530, Irfan Shaikh wrote: > > > > > > > > > Hi All, > > > > > > Can any one please tell me in which package can i > find ffmpeg's > > > H264 encoder plug-in ffenc_h264. > > > > Are you making up element names? That element does not exist > and neither > > does the 'ffmpeg h264 encoder'. > > > Indeed. I'm guessing he means the h264 encoder used by ffmpeg > ergo... > x264. And for that we have a plugin : x264enc. > > Or he meant ffenc_h264 or ffdec_h264, both of which can be found in > the ffmpeg plugin. If you have ffenc_h264 on your distro, that means they're using the system wide ffmpeg, most likely compiled with everything-under-the-sun support (which includes x264). So instead of only having one level of abstraction for using x264 (gstplugin-x264enc <=> x264), you have two (gstffmpeg <=> libavcodec(ffmpeg) <=> x264). Bad for efficiency. Oh, and as you might know.. we don't support gst-ffmpeg built with system-wide ffmpeg. Just use x264enc, please. Edward, > > > > Edward > > > > > > Regards, > > Wim > > > > > > Regards, > > > Irfan > > > > > > > > > This message (including any attachment) is confidential > and may be > > > legally privileged. Access to this message by anyone other > than the > > > intended recipient(s) listed above is unauthorized. If you > are not the > > > intended recipient you are hereby notified that any > disclosure, > > > copying, or distribution of the message, or any action > taken or > > > omission of action by you in reliance upon it, is > prohibited and may > > > be unlawful. Please immediately notify the sender by reply > e-mail and > > > permanently delete all copies of the message if you have > received this > > > message in error. > > > > > > > ------------------------------------------------------------------------- > > > This SF.Net email is sponsored by the Moblin Your Move > Developer's challenge > > > Build the coolest Linux based applications with Moblin SDK > & win great prizes > > > Grand prize is a trip for two to an Open Source event > anywhere in the world > > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move > Developer's challenge > > Build the coolest Linux based applications with Moblin SDK & > win great prizes > > Grand prize is a trip for two to an Open Source event > anywhere in the world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move > Developer's challenge > Build the coolest Linux based applications with Moblin SDK & > win great prizes > Grand prize is a trip for two to an Open Source event anywhere > in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > Alexandre Rosenfeld > > EngComp 06 - USP S?o Carlos > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ved.kpl at gmail.com Tue Sep 30 08:20:40 2008 From: ved.kpl at gmail.com (ved kpl) Date: Tue, 30 Sep 2008 11:50:40 +0530 Subject: [gst-devel] ffenc_h264 plug-in ??? In-Reply-To: <9D5E1752379A43408015F7FE984661157827B5@CHNEXVS01.VSNLXCHANGE.COM> References: <9D5E1752379A43408015F7FE984661157827B5@CHNEXVS01.VSNLXCHANGE.COM> Message-ID: <7496c23f0809292320m29593e3cn774fa66bd2686d5e@mail.gmail.com> You need to install libx264-dev first and then install gst-plugins-bad package that contains x264enc. On Mon, Sep 29, 2008 at 8:29 PM, Irfan Shaikh wrote: > > > Hi All, > > Can any one please tell me in which package can i find ffmpeg's H264 > encoder plug-in ffenc_h264. > > Regards, > Irfan > > This message (including any attachment) is confidential and may be legally > privileged. Access to this message by anyone other than the intended > recipient(s) listed above is unauthorized. If you are not the intended > recipient you are hereby notified that any disclosure, copying, or > distribution of the message, or any action taken or omission of action by > you in reliance upon it, is prohibited and may be unlawful. Please > immediately notify the sender by reply e-mail and permanently delete all > copies of the message if you have received this message in error. > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From irfanshaikh at tataelxsi.co.in Tue Sep 30 10:45:41 2008 From: irfanshaikh at tataelxsi.co.in (Irfan Shaikh) Date: Tue, 30 Sep 2008 14:15:41 +0530 Subject: [gst-devel] rtpmp4gpay & rtpmp4vpay Message-ID: <9D5E1752379A43408015F7FE984661157827C0@CHNEXVS01.VSNLXCHANGE.COM> Hi all, I am facing difficulty to choose the correct payloader for my requirement. My requirement is to hav a live capture through video camera and stream in on Client QT player.I am very confused to differentiate between the payload packetizers to use. Earlier i had a understanding that rtpmp4vpay is for MPEG4 simple profile {layer 0, 1, 2, 3} And rtpmp4gpay is for MPEG4 advanced profile. But i feel my understanding is not so correct. Please make clear about the usage of both payload packetizers for MPEG4.Which one should i prefer ?? rtpmp4vpay: RTP MPEG-4 Video packet payloader Payload MPEG-4 video as RTP packets (RFC 3016) media: video encoding-name: MP4V-ES rtpmp4gpay: RTP packet payloader Codec/Payloader/Network Payload MPEG4 elementary streams as RTP packets (RFC 3640) media: { video, audio, application } encoding-name: MPEG4-GENERIC Regards, Irfan This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ved.kpl at gmail.com Tue Sep 30 14:34:29 2008 From: ved.kpl at gmail.com (ved kpl) Date: Tue, 30 Sep 2008 18:04:29 +0530 Subject: [gst-devel] muxing streams In-Reply-To: <1222767704.18136.1.camel@macbook> References: <1222767704.18136.1.camel@macbook> Message-ID: <7496c23f0809300534j61a0a9bw868b06af5f0aa611@mail.gmail.com> Hi, You probably need the elementary stream parser elements after the filesrc. The muxer is unware of the data format that is getting. Ved On Tue, Sep 30, 2008 at 3:11 PM, Henrique Ferreiro Garc?a wrote: > Hi all! > > I have extracted the video and audio streams from a matroska file > getting: > > mkv.0: JVT NAL sequence, H.264 video @ L 31 > mkv.1: ATSC A/52 aka AC-3 aka Dolby Digital stream, 48 kHz,, complete > main (CM) 3 front/2 rear, LFE on,, 448 kbit/s reserved Dolby Surround > mode > > Afterwards, I have run the following: > > gst-launch-0.10 filesrc location=mkv.0 ! queue ! mux.video_0 avimux > name=mux ! queue ! filesink location=house.avi filesrc location=mkv.1 ! > queue ! mux.audio_0 > > But the end result is a unplayable file. This is the output from > mplayer: > > AVI file format detected. > [aviheader] Video stream found, -vid 0 > [aviheader] Audio stream found, -aid 1 > Detected NON-INTERLEAVED AVI file format. > VIDEO: [] 0x0 0bpp nan fps -17179870.0 kbps (-2097152.0 kbyte/s) > FPS not specified in the header or invalid, use the -fps option. > ========================================================================== > Forced audio codec: mad > Opening audio decoder: [pcm] Uncompressed PCM audio decoder > Unknown/missing audio format -> no sound > ADecoder init failed :( > Cannot find codec for audio format 0x0. > Read DOCS/HTML/en/codecs.html! > Audio: no sound > Video: no video > > Am I doing something wrong? > > -- > Henrique Ferreiro Garc?a > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From felipe.contreras at gmail.com Tue Sep 30 16:50:22 2008 From: felipe.contreras at gmail.com (Felipe Contreras) Date: Tue, 30 Sep 2008 17:50:22 +0300 Subject: [gst-devel] ffenc_h264 plug-in ??? In-Reply-To: <9D5E1752379A43408015F7FE984661157827B8@CHNEXVS01.VSNLXCHANGE.COM> References: <9D5E1752379A43408015F7FE984661157827B8@CHNEXVS01.VSNLXCHANGE.COM> Message-ID: <94a0d4530809300750n30c0797buf3bb2906767f3dae@mail.gmail.com> On Tue, Sep 30, 2008 at 7:08 AM, Irfan Shaikh wrote: > Hi Wim, > > According to your reply ffenc_h264 element does not exist and neither > does the 'ffmpeg h264 encoder'. But just going through net i found some > pipelines which used ffenc_h264. > Earlier i was also not very sure about exsiting ffmpeg h264 encoder.But > needed some clarification. > > Can i know in which package can i find h264 encoder used by ffmpeg ergo..... > x264. > As Edward told that we we have a plugin : x264enc...Please help me to find > the package FFmpeg has some encoders/decoders on it's own, but on on other cases it's using an external library. That is the case with the H.264 encoder: it's using libx264. When it does that it uses the name of the library, as opposed to the name of the codec. In gst-ffmpeg you get the same name, so you have ffenc_libx264. There's a wrapper for libx264 in GStreamer, but it's in gst-plugins-bad, which means it's probably not in a great state. Last time I tried gst's x264enc produced clips with not so good quality (bad defaults?) and ffenc_libx264 worked better. If anyone is interested in proper libx264 support maybe we can work together as I'm interested too. Cheers. -- Felipe Contreras From gstelzz at yahoo.fr Tue Sep 30 22:37:50 2008 From: gstelzz at yahoo.fr (Aurelien Grimaud) Date: Tue, 30 Sep 2008 22:37:50 +0200 Subject: [gst-devel] [Bulk] Problem in gstrtpbin and AV Transfer---Urgent In-Reply-To: <8c192ddd0809300540s15390f64j58e9488b146ee19c@mail.gmail.com> References: <8c192ddd0809300540s15390f64j58e9488b146ee19c@mail.gmail.com> Message-ID: <48E28E1E.1050508@yahoo.fr> Can you describe how you switch from one source to another ? Using input-selector ? Do you change pipeline or bin state when switching ? Is it a receiver problem or a sender problem ? Are rtp packets really sent from one side to the other ? Can you post the errors ? Using dynamic pipeline is tricky because of buffer timestamp and new segment management. Using GST_DEBUG=5 when starting your pipeline will provide huge log, in which you will find for sure where exactly the problem occurs. Aurelien Manish Rana a ?crit : > Hi All.. > > I am facing some other problem in RTP Streaming... > > I am sending the rtp data over the UDP and playing it on the other end. > I am recording the data from v4l2src @ 10 fps and voice data from > alsasrc to send it across. The transmission starts well as well as i > can see the video on the other end. > Now in between i starts to take the data from a file. > I decode the data and encode again to send it using the same rtpbin > and udpsink. > > Enc, Pay gstrtpbin and udpsink are the element kept common...means i > will use these elements in both the cases. > > the packets goes well and on the other end i can see the AV fine in > both the cases.... > > Now when i again switch back to v4l2src the packets on the receiver > end come but the decoder (ffdec_mpeg4) gives error.......and there is > no updates on display........ > > The file i am using is @25fps...... > also when i increase the FPS of v4l2src the whole system works > well..... at 20 fps..... > > Please let me know if i am missing something or i am doing something > wrong way.... > > Also i tried using different enc and pay elements in both the > case..but i i unlink and link to gstrtpbin, there is no data at > all..even in the first switch to file transfer.............. :( > > Please help...i have tried every possible thing to fix this > issue....but i am stuck with the same for last couple of days now..... > > > Any input will be appreciated from bottum of heart....... > > BR > Manish > > > Tue, Sep 30, 2008 at 3:52 PM, Wim Taymans > wrote: > > On Tue, 2008-09-30 at 11:59 +0200, Francesco Argese wrote: > > Hi all. > > > > I have problems sending a theora flow in streaming through rtp. > > > > My two pipeline are the following: > > Sender > > 1)gst-launch gstrtpbin name=rtpbin dv1394src ! dvdemux name=demux ! > > dvdec ! ffmpegcolorspace ! theoraenc ! rtptheorapay ! > > rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink port=5000 > > host=172.16.1.56 rtpbin.send_rtcp_src_0 ! > udpsink port=5001 > > host=172.16.1.56 sync=false async=false > udpsrc port=5005 ! > > rtpbin.recv_rtcp_sink_0 > > > > Receiver > > 2)gst-launch-0.10 gstrtpbin name=rtpbin udpsrc port=5000 > > > caps="application/x-rtp,media=video,clock-rate=90000,encoding-name=THEORA,profile-level-id=1,payload=96" > > ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtptheoradepay ! theoradec ! > > ffmpegcolorspace ! > > > video/x-raw-rgb,bpp=32,endianness=4321,depth=24,red_mask=65280,green_mask=16711680,blue_mask=-16777216,width=720,height=576 > > ! fakesink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 > > rtpbin.send_rtcp_src_0 ! udpsink host=172.16.100.277 port=5005 > > sync=false async=false > > > > The sender seems to works well but at the receiver i obtain the > following error: > > New clock: GstSystemClock > > ERROR: from element /pipeline0/udpsrc0: Internal data flow error. > > Additional debug info: > > gstbasesrc.c(2240): gst_base_src_loop (): /pipeline0/udpsrc0: > > streaming task paused, reason not-negotiated (-4) > > The receiver needs the sampling, width, height and configuration > strings. Run the sender pipeline with -v, then copy the caps on > the RTP > udpsink to the receiver udpsrc caps (and yes, they are huge and > they are > needed). > > Wim > > > > > Any suggestion? What could be the problem? A pipeline very similar > > using MPEG4 works well. > > > > Regards > > Francesco Argese > > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move > Developer's challenge > > Build the coolest Linux based applications with Moblin SDK & win > great prizes > > Grand prize is a trip for two to an Open Source event anywhere > in the world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win > great prizes > Grand prize is a trip for two to an Open Source event anywhere in > the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From aniruddhachourasia at tataelxsi.co.in Mon Sep 29 16:10:05 2008 From: aniruddhachourasia at tataelxsi.co.in (Aniruddha) Date: Mon, 29 Sep 2008 19:40:05 +0530 Subject: [gst-devel] Porting gst-plugins-good on DaVinCi Message-ID: <009301c9223d$12969e20$3f033c0a@telxsi.com> Hi All, How to port gst-plugins-good on TI DaVinCi Platform? 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