<html> <head> <meta content="text/html; charset=ISO-8859-1" http-equiv="Content-Type"> </head> <body bgcolor="#FFFFFF" text="#000000"> Hi Kris, You are correct when you say that you did not explain why my program didn't work - but I tried your advice and now have 2 programs which don't work. I use a gst_parse_launch, passing it a string such as <blockquote>"v4l2src norm=PAL ! videorate ! 'video/x-raw,format=(string)I420,width=320,height=240,framerate=(fraction)25/1' ! queue ! mux. alsasrc ! audioconvert ! 'audio/x-raw,rate=44100,channels=2' ! queue ! mux. avimux name=mux ! filesink location=file.avi" </blockquote> but when I run the program I get a message GStreamer-CRITICAL **: gst_element_make_from_uri: assertion `gst_uri_is_valid (uri)' failed I spent a good while chasing wild geese, asuming that the problem must be with the file name I passed in for the filesink - but then studied the debug trace more closely to see this extract <blockquote>(SBCRecord2:2968): GStreamer-CRITICAL **: gst_element_make_from_uri: assertion `gst_uri_is_valid (uri)' failed 0:00:00.172690420 [335m 2968[00m 0xc5f830 [31;01mERROR [00m [00;01;37;41m GST_PIPELINE ./grammar.y:827:priv_gst_parse_yyparse:[00m no source element for URI "/x-raw,rate=44100,channels=2'" 0:00:00.172703486 [335m 2968[00m 0xc5f830 [31;01mERROR [00m [00;01;37;41m GST_PIPELINE ./grammar.y:744:priv_gst_parse_yyparse:[00m link without source element 0:00:00.172715265 [335m 2968[00m 0xc5f830 [31;01mERROR [00m [00;01;37;41m GST_PIPELINE ./grammar.y:750:priv_gst_parse_yyparse:[00m link without sink element (SBCRecord2:2968): GStreamer-CRITICAL **: gst_element_make_from_uri: assertion `gst_uri_is_valid (uri)' failed 0:00:00.172740072 [335m 2968[00m 0xc5f830 [31;01mERROR [00m [00;01;37;41m GST_PIPELINE ./grammar.y:827:priv_gst_parse_yyparse:[00m no source element for URI "/x-raw,format=(string)I420,width=320,height=240,framerate=(fraction)25/1'" 0:00:00.172753256 [335m 2968[00m 0xc5f830 [31;01mERROR [00m [00;01;37;41m GST_PIPELINE ./grammar.y:744:priv_gst_parse_yyparse:[00m link without source element 0:00:00.172764467 [335m 2968[00m 0xc5f830 [31;01mERROR [00m [00;01;37;41m GST_PIPELINE ./grammar.y:750:priv_gst_parse_yyparse:[00m link without sink element </blockquote> For some reason, the caps I am passing in of " 'audio/x-raw,rate=44100,channels=2'" has been pulled apart and then found to be broken. So copy-and-paste from a good command line into the C program did not have 100% success. Any help to get one (or both) versions of my program working would be appreciated. Ian <div class="moz-cite-prefix">On 05/01/2013 22:36, Krzysztof Konopko wrote: </div> <blockquote cite="mid:50E8AAD6.6010301@youview.com" type="cite"> <pre wrap="">Hi Ian, I don't know what particular problem in your app is, but in a situation when a pipeline works with gst-launch-1.0, but a C app doesn't, my stock advise is: copy your command line pipeline and use [1]. You can start writing something dead simple and than introduce whatever complexity you need ensuring it still works at each step. In your particular case IMO you've introduced too much complexity. BTW, by using [1] your error checking reduces to something like this (pilfered/inspired from/by gst-launch.c): if (!pipeline) { g_printerr ("ERROR: pipeline could not be constructed: %s\n", error ? GST_STR_NULL (error->message) : "(unknown error)"); goto untergang; } else if (error) { g_printerr ("Erroneous pipeline: %s\n", GST_STR_NULL (error->message)); goto untergang; } [1] <a class="moz-txt-link-freetext" href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstParse.html#gst-parse-launch">http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstParse.html#gst-parse-launch</a> Kris On 05/01/13 17:06, Ian Davidson wrote: </pre> <blockquote type="cite"> <pre wrap="">If I use the following command, I can record Audio and Video into an AVI File. gst-launch-1.0 -e v4l2src norm=PAL ! videorate ! 'video/x-raw,format=(string)I420,width=320,height=240,framerate=(fraction)25/1' ! queue ! mux. alsasrc ! audioconvert ! 'audio/x-raw,rate=44100,channels=2' ! queue ! mux. avimux name=mux ! filesink location=script-test.avi I would like to replicate that in a program and my program is below. However, the video does not make it through - the Audio plays OK when I play the avi file. It had been suggested that videorate and audiorate might be beneficial for me. In my program, if I included audiorate (as indicated by the commented out lines), I just heard a brief burst of sound at the beginning of the clip, whereas without audiorate, I heard the full audio. Including/excluding videorate does not seem to make any difference at the moment. I tried to set a value for "norm" on the v4l2src as PAL, but the compiler did not recognise the ENUM value. The only way I could set it was to use the value indicated by gst-inspect. I assume that I have done something stupid again, but I cannot see what it is. Thanks for your help. Ian #include <gst/gst.h> #include <glib.h> 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; } 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); } int main (int argc, char *argv[]) { GMainLoop *loop; GstElement *pipeline, *vsource, *vcapsfilter, *vidrate, *queue1; GstElement *asource, *aconv, *audrate, *acapsfilter, *queue2; GstElement *mux, *sink; GstBus *bus; guint bus_watch_id; GstCaps *caps; /* Initialisation */ gst_init (&argc, &argv); loop = g_main_loop_new (NULL, FALSE); /* Check input arguments */ if (argc != 2) { g_printerr ("Usage: %s <AVI filename>\n", argv[0]); return -1; } /* Create gstreamer elements */ pipeline = gst_pipeline_new ("av-recorder"); vsource = gst_element_factory_make ("v4l2src", "vid-source"); vcapsfilter = gst_element_factory_make ("capsfilter", "vid-caps"); vidrate = gst_element_factory_make ("videorate", "vidrate"); queue1 = gst_element_factory_make ("queue", "queue1"); asource = gst_element_factory_make ("alsasrc", "alsa-source"); aconv = gst_element_factory_make ("audioconvert", "audio-conv"); acapsfilter = gst_element_factory_make ("capsfilter", "audio-caps"); audrate = gst_element_factory_make ("audiorate", "audrate"); queue2 = gst_element_factory_make ("queue", "queue2"); mux = gst_element_factory_make ("avimux", "avi-mux"); sink = gst_element_factory_make ("filesink", "file-output"); if (!pipeline || !vsource || !vcapsfilter || !vidrate || !queue1 || !asource || !aconv || !audrate || !acapsfilter || !queue2 || !mux || !sink) { g_printerr ("One element could not be created. Exiting.\n"); return -1; } /* Set up the pipeline */ /* we set the output filename to the sink element */ g_object_set (G_OBJECT (sink), "location", argv[1], NULL); /* we set the video capabilities on the vidcaps element */ caps = gst_caps_from_string("video/x-raw,format=(string)I420,width=320,height=240,framerate=(fraction)25/1"); g_object_set (G_OBJECT (vcapsfilter), "caps", caps, NULL); gst_caps_unref (caps); /* we set the audio capabilities on the audiocaps element */ caps = gst_caps_from_string("audio/x-raw,rate=44100,channels=2"); g_object_set (G_OBJECT (acapsfilter), "caps", caps, NULL); gst_caps_unref (caps); g_object_set (G_OBJECT (vsource), "norm", 255, NULL); /* we add a message handler */ bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline)); bus_watch_id = gst_bus_add_watch (bus, bus_call, loop); gst_object_unref (bus); /* we add all elements into the pipeline */ /* vsource, vcapsfilter, vidrate, queue1, asource, aconv, acapsfilter, queue2, mux, sink */ gst_bin_add_many (GST_BIN (pipeline), vsource, vcapsfilter, queue1, /* vsource, vidrate, vcapsfilter, queue1,*/ asource, aconv, acapsfilter, queue2, /* asource, aconv, audrate, acapsfilter, queue2,*/ mux, sink, NULL); /* we link the elements together */ /* vsource -> vcapsfilter -> vidrate -> queue1 -> avimux asource -> aconv -> acapsfilter -> queue2 -> avimux mux -> sink */ gst_element_link_many (vsource, vcapsfilter, queue1, mux, NULL); /* gst_element_link_many (vsource, vcapsfilter, vidrate, queue1, mux, NULL);*/ gst_element_link_many (asource, aconv, acapsfilter, queue2, mux, NULL); /* gst_element_link_many (asource, aconv, audrate, acapsfilter, queue2, mux, NULL);*/ gst_element_link_many (mux, sink, NULL); /* Set the pipeline to "playing" state*/ g_print ("Now recording: %s\n", argv[1]); gst_element_set_state (pipeline, GST_STATE_PLAYING); /* Iterate */ g_print ("Recording (or not!)...\n"); g_main_loop_run (loop); /* Out of the main loop, clean up nicely */ g_print ("Returned, stopping recording\n"); gst_element_set_state (pipeline, GST_STATE_NULL); g_print ("Deleting pipeline\n"); gst_object_unref (GST_OBJECT (pipeline)); g_source_remove (bus_watch_id); g_main_loop_unref (loop); return 0; } -- -- Ian Davidson -- </pre> </blockquote> </blockquote> <div class="moz-signature">-- -- Ian Davidson -- Facts used in this message may or may not reflect an underlying objective reality. Facts are supplied for personal use only. Recipients quoting supplied information do so at their own risk. Facts supplied may vary in whole or part from widely accepted standards. While painstakingly researched, facts may or may not be indicative of actually occurring events or natural phenomena. The author accepts no responsibility for personal loss or injury resulting from memorisation and subsequent use. </div> </body> </html>