[gst-devel] Converting a gst-launch pipeline to C code

[Thijs Vermeir]

Hi,

On Mon, Sep 29, 2008 at 7:57 PM, Alan Carvalho de Assis
<acassis at gmail.com> 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 <gst/gst.h>
> #include <glib.h>
> #include <string.h>
>
> 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 <Ogg/Vorbis filename>\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;
> }
>
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