<div class="gmail_quote">On Fri, Jul 3, 2009 at 10:10 PM, <<a href="mailto:gstreamer-devel-request@lists.sourceforge.net">gstreamer-devel-request@lists.sourceforge.net</a>> wrote: <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> Send gstreamer-devel mailing list submissions to <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> To subscribe or unsubscribe via the World Wide Web, visit <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> or, via email, send a message with subject or body 'help' to <a href="mailto:gstreamer-devel-request@lists.sourceforge.net">gstreamer-devel-request@lists.sourceforge.net</a> You can reach the person managing the list at <a href="mailto:gstreamer-devel-owner@lists.sourceforge.net">gstreamer-devel-owner@lists.sourceforge.net</a> When replying, please edit your Subject line so it is more specific than "Re: Contents of gstreamer-devel digest..." Today's Topics: 1. low framerate with udp (Mehta, Neel) 2. Re: analogue of gst-launch with caps (Tim-Philipp M?ller) 3. Re: analogue of gst-launch with caps (Tim-Philipp M?ller) 4. Re: Need help with first gstreamer program. (Ognen Bezanov) 5. A question regarding to dynamic pipeline change (Zheng, Huan) 6. Re: Save a stream to different file every hour (Juraj Kubelka) ---------------------------------------------------------------------- Message: 1 Date: Fri, 3 Jul 2009 16:52:29 +0530 From: "Mehta, Neel" <<a href="mailto:x0112149@ti.com">x0112149@ti.com</a>> Subject: [gst-devel] low framerate with udp To: Discussion of the development of GStreamer <<a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>> Message-ID: <<a href="mailto:E0D41E29EB0DAC4E9F3FF173962E9E94024ED84AE4@dbde02.ent.ti.com">E0D41E29EB0DAC4E9F3FF173962E9E94024ED84AE4@dbde02.ent.ti.com</a>> Content-Type: text/plain; charset="us-ascii" Hi I am currently streaming MPEG4 video via UDP (resolution 720P/30 fps). Currently using the Live555, and RTSP, the video can be played at 30 fps. Using Gstreamer, I am able to get only 21 fps via UDP (for VGA, I do get 30 fps). I am using gstreamer for transmission only. The encoding process is same. I am using appsrc to get the encoded data from application buffer. I am attaching the code that I am using currently for transmission using Gstreamer. Please help me out to increase the framerate. The pipeline is basically: appsrc is-live=true name=source caps=" video/mpeg, mpegversion=(int)4, framerate=(fraction)1001/30000, width=(int)1280, height=(int)720" ! rtpmp4vpay send-config=TRUE ! udpsink host=172.24.136.232 port=5000 Note: I am using UDP instead of rtsp, as I am unable to write the code completely between appsrc and gst-rtsp-server. I have posted on this forum for help titled "appsrc and gst-rtsp-server" on 29th June, Monday, but I haven't received a help yet. Someone please help me out with this too. Please fell free to ask for more information. Thanks in advance. Regards, Neel. -------------- next part -------------- An HTML attachment was scrubbed... -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: appsrc-udp.txt ------------------------------ Message: 2 Date: Fri, 03 Jul 2009 12:30:24 +0100 From: Tim-Philipp M?ller <<a href="mailto:t.i.m@zen.co.uk">t.i.m@zen.co.uk</a>> Subject: Re: [gst-devel] analogue of gst-launch with caps To: <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> Message-ID: <1246620624.5472.2.camel@zingle> Content-Type: text/plain; charset="UTF-8" On Thu, 2009-07-02 at 17:52 +0530, Mehta, Neel wrote: > Can anyone suggest a substitute of gst_parse_launch (string, NULL), so > that I have the negotiated caps as the part of the output. > > i.e. I want an analogue of gst-launch-0.10 ?v. A quick look at gst-launch.c reveals that -v boils down to: g_signal_connect (pipeline, "deep-notify", G_CALLBACK (gst_object_default_deep_notify), NULL); Cheers -Tim ------------------------------ Message: 3 Date: Fri, 03 Jul 2009 12:31:34 +0100 From: Tim-Philipp M?ller <<a href="mailto:t.i.m@zen.co.uk">t.i.m@zen.co.uk</a>> Subject: Re: [gst-devel] analogue of gst-launch with caps To: <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> Message-ID: <1246620694.5472.3.camel@zingle> Content-Type: text/plain; charset="UTF-8" On Thu, 2009-07-02 at 17:52 +0530, Mehta, Neel wrote: > Can anyone suggest a substitute of gst_parse_launch (string, NULL), so > that I have the negotiated caps as the part of the output. > > i.e. I want an analogue of gst-launch-0.10 ?v. A quick look at gst-launch.c reveals that -v boils down to: g_signal_connect (pipeline, "deep-notify", G_CALLBACK (gst_object_default_deep_notify), NULL); Cheers -Tim ------------------------------ Message: 4 Date: Fri, 03 Jul 2009 13:00:29 +0100 From: Ognen Bezanov <<a href="mailto:ognen.bezanov@wai.co.uk">ognen.bezanov@wai.co.uk</a>> Subject: Re: [gst-devel] Need help with first gstreamer program. To: <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> Message-ID: <1246622429.9434.17.camel@PC133> Content-Type: text/plain No-one has any idea? ------------------------------ Message: 5 Date: Wed, 1 Jul 2009 16:20:44 +0800 From: "Zheng, Huan" <<a href="mailto:huan.zheng@intel.com">huan.zheng@intel.com</a>> Subject: [gst-devel] A question regarding to dynamic pipeline change To: gstreamer-devel <<a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>> Message-ID: <<a href="mailto:BB1F052FCDB1EA468BD99786C8B1ED2C05870A1A07@PDSMSX501.ccr.corp.intel.com">BB1F052FCDB1EA468BD99786C8B1ED2C05870A1A07@PDSMSX501.ccr.corp.intel.com</a>> Content-Type: text/plain; charset="us-ascii" Hi, dear developers If I dynamically replace one element inside the pipeline, will the negotiation of the *whole pipeline* be performed again when the pipeline restarted next time? Thanks a lot for your answering! Best Regards, Zheng, Huan(ZBT) OTC/SSD/SSG Intel Asia-Pacific Research & Developement Ltd Tel: 021-6116 6435 Inet: 8821 6435 Cub: 3W035 -------------- next part -------------- An HTML attachment was scrubbed... ------------------------------ Message: 6 Date: Fri, 3 Jul 2009 18:40:21 +0200 From: Juraj Kubelka <<a href="mailto:juraj.kubelka@googlemail.com">juraj.kubelka@googlemail.com</a>> Subject: Re: [gst-devel] Save a stream to different file every hour To: Discussion of the development of GStreamer <<a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>> Message-ID: <<a href="mailto:f54511ec0907030940r4db3acf6k41015c82ee931f91@mail.gmail.com">f54511ec0907030940r4db3acf6k41015c82ee931f91@mail.gmail.com</a>> Content-Type: text/plain; charset="utf-8" Hopefully I got it and my piece of code works :) So, if anyone is interested in saving a stream to a new file every hour, download here <a href="http://files.getdropbox.com/u/710942/audio-2009-07-03_18-32.tar.gz" target="_blank">http://files.getdropbox.com/u/710942/audio-2009-07-03_18-32.tar.gz</a> Best regards, Jura On Fri, Jul 3, 2009 at 11:42 AM, Juraj Kubelka <<a href="mailto:juraj.kubelka@googlemail.com">juraj.kubelka@googlemail.com</a> > wrote: > Hi! > > As I have played with it, it seams it does not work properly all the time. > Can you, please, check my source code in Python? There are TODOs which tag > lines with possible wrong gstreamer handling (as I suppose). I will > appreciate any suggestions. > > the file (pipelineholder.py): <a href="http://cesta.pastebin.com/m3b0c7ad" target="_blank">http://cesta.pastebin.com/m3b0c7ad</a> > other two files witch work together: <a href="http://cesta.pastebin.com/m46860962" target="_blank">http://cesta.pastebin.com/m46860962</a> > <a href="http://cesta.pastebin.com/m5afca033" target="_blank">http://cesta.pastebin.com/m5afca033</a> > full sourcecode (3 files, run ./pipelinemanager.py, pipelineholder.py is > the main one): <a href="http://files.getdropbox.com/u/710942/audio.tar.gz" target="_blank">http://files.getdropbox.com/u/710942/audio.tar.gz</a> > > Thank you in advance. > Jura > > > On Tue, Jun 23, 2009 at 4:34 PM, Juraj Kubelka < > <a href="mailto:juraj.kubelka@googlemail.com">juraj.kubelka@googlemail.com</a>> wrote: > >> On Thu, Jun 18, 2009 at 8:46 AM, Arnout Vandecappelle <<a href="mailto:arnout@mind.be">arnout@mind.be</a>>wrote: >> >>> On Wednesday 17 June 2009 10:29:43 Viraj Karandikar wrote: >>> > And after every hour you will have to: pause the pipeline, create new >>> > filesink, disconnect old filesink, connect new filesink and close old >>> > filesink. >>> >>> Actually, no. Pausing the pipeline will not (immediately) stop the >>> threads >>> in the source and queues. In addition, once it's paused any new data >>> coming >>> in on the live source will be dropped until it plays again, which >>> probably is >>> not the intention. >>> >>> Instead, you have to use blocking. Below there's a piece of code that I >>> use. >>> See also docs/design/part-block.txt. There should be high-level API for >>> that, but nobody has bothered up to now to write it. >>> >>> You need to replace the decoder and muxer as well, so the proper headers >>> are >>> created in the new file. To simplify all that I put it in a bin (which >>> can >>> be unreffed as a whole). >>> >>> > You can do above things in another independent thread which will >>> > just wait till 1 hour is complete. >>> >>> Or you can use g_timeout_add_seconds (3600, ...) in your main loop. >>> >>> Regards, >>> Arnout >>> >>> -- >>> >>> static gboolean destroy_bin_cb (gpointer user_data) >>> { >>> GstElement *oldbin = user_data; >>> >>> log_message("Destroying old bin.\n"); >>> if (pipeline) >>> { >>> gst_element_set_state (oldbin, GST_STATE_NULL); >>> gst_bin_remove (GST_BIN(pipeline), oldbin); >>> } >>> return FALSE; >>> } >>> >>> >>> static void replace_filesink_blocked_cb (GstPad *pad, gboolean blocked, >>> gpointer user_data) >>> { >>> if (blocked) >>> { >>> GstElement *oldbin; >>> GstPad *sinkpad; >>> >>> log_message("Blocked filesink queue.\n"); >>> g_static_rec_mutex_lock(&mutex); >>> log_message("Locked.\n"); >>> oldbin = filesinkbin; >>> sinkpad = gst_element_get_static_pad(oldbin, "sink"); >>> if (!sinkpad) >>> { >>> log_message("replace_filesink_blocked_cb: oldbin doesn't have >>> sink >>> pad.\n"); >>> goto fail; >>> } >>> gst_pad_unlink(queuesrcpad, sinkpad); >>> log_message("Unlinked.\n"); >>> >>> /* Finalize the old tail. */ >>> /* Sending EOS should be done from here (we're in the queue >>> thread). >>> */ >>> gst_pad_send_event(sinkpad, gst_event_new_eos()); >>> log_message("Sent event.\n"); >>> >>> log_message("Checked sinks.\n"); >>> /* Setting state should be done from the main thread. Do it before >>> the >>> pipeline has been set to PLAYING - implicitly, because that is >>> also >>> done in an idle callback, but one that is started after this >>> one. >>> */ >>> g_idle_add(destroy_bin_cb, oldbin); >>> >>> create_filesink_tail(); >>> log_message("Created filesink.\n"); >>> g_static_rec_mutex_unlock(&mutex); >>> log_message("Unlocked mutex.\n"); >>> >>> /* And unblock again. */ >>> gst_pad_set_blocked_async(queuesrcpad, FALSE, >>> replace_filesink_blocked_cb, NULL); >>> >>> log_message("Done replacing filesink.\n"); >>> } >>> else >>> { >>> log_message("Unblocked filesink queue.\n"); >>> } >>> return; >>> >>> fail: >>> pipeline_terminate_async(); >>> g_static_rec_mutex_unlock(&mutex); >>> } >>> >>> static void replace_filesink() >>> { >>> if (queuesrcpad == NULL) >>> log_message("replace_filesink while no queuesrcpad yet, waiting >>> for it >>> to appear.\n"); >>> else >>> gst_pad_set_blocked_async(queuesrcpad, TRUE, >>> replace_filesink_blocked_cb, NULL); >>> } >>> >>> >> Hi, >> >> thank you a lot for your help. I did it in Python and it seams it works. >> Source code is on pastebin: <a href="http://pastebin.com/f3a80ed09" target="_blank">http://pastebin.com/f3a80ed09</a> >> (pipelineholder.py <<a href="http://pastebin.com/f3a80ed09%28pipelineholder.py" target="_blank">http://pastebin.com/f3a80ed09%28pipelineholder.py</a>>) >> and <a href="http://pastebin.com/f4ee27d6d" target="_blank">http://pastebin.com/f4ee27d6d</a> (pipelinemanager). Any comments are >> appreciated. >> >> Best regards, >> Jura >> > > -------------- next part -------------- An HTML attachment was scrubbed... ------------------------------ ------------------------------------------------------------------------------ ------------------------------ _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> End of gstreamer-devel Digest, Vol 38, Issue 7 ********************************************** </blockquote></div> hi I recorded the audio and video with out any encoding and decoding.Video is coming from the camera as mpeg4 and audio in the format of aac.My pipeline looks like this, gst-launch -e rtspsrc location="rtsp://<a href="http://user:carinov123@10.0.0.104/live.sdp">user:carinov123@10.0.0.104/live.sdp</a>" name=rtsp ! rtpmp4vdepay ! mpeg4videoparse ! qtmux name=qt ! filesink location=a.mp4 rtsp. ! rtpmp4gdepay ! aacparse ! qt. But when i tried to play it with mplayer,i got the video only, the out put of "mplayer a.mp4 " is, libavformat file format detected. [lavf] Audio stream found, -aid 0 [lavf] Video stream found, -vid 1 VIDEO: [mp4v] 320x240 24bpp 30.333 fps 0.0 kbps ( 0.0 kbyte/s) ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family Selected video codec: [ffodivx] vfm: ffmpeg (FFmpeg MPEG-4) ========================================================================== ========================================================================== Opening audio decoder: [faad] AAC (MPEG2/4 Advanced Audio Coding) FAAD: compressed input bitrate missing, assuming 128kbit/s! AUDIO: 32000 Hz, 2 ch, s16le, 128.0 kbit/12.50% (ratio: 16000->128000) Selected audio codec: [faad] afm: faad (FAAD AAC (MPEG-2/MPEG-4 Audio)) ========================================================================== [AO OSS] audio_setup: Can't open audio device /dev/dsp: Device or resource busy AO: [alsa] 32000Hz 2ch s16le (2 bytes per sample) Starting playback... FAAD: Failed to decode frame: Maximum number of scalefactor bands exceeded -- Sreerenj B <a href="http://sreerenj.livejournal.com">http://sreerenj.livejournal.com</a> <a href="mailto:bsreerenj@gmail.com">bsreerenj@gmail.com</a> mob: +91 9995377714