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

[Alan Carvalho de Assis]

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 <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;
}