[gst-devel] C code for rtp h264 decoding. I can't find how to solve the error. Read is insteresting

[giorgino]

Hi All. 
I have two pc where one is the sender and the other the receiver. In the
sender I use this pipe

gst-launch -v gstrtpbin name=rtpbin v4l2src !
video/x-raw-yuv,width=320,height=240! queue ! videorate ! ffmpegcolorspace !
x264enc byte-stream=true bitrate=300 ! rtph264pay ! rtpbin.send_rtp_sink_0
rtpbin.send_rtp_src_0 ! udpsink port=5000 host=192.168.100.196 ts-offset=0
name=vrtpsink rtpbin.send_rtcp_src_0 ! udpsink port=5001
host=192.168.100.196 sync=false async=false name=vrtcpsink udpsrc port=5005
name=vrtpsrc ! rtpbin.recv_rtcp_sink_0

and it works well. If in the receiver I use the following pipe the receiver
works well

st-launch -v gstrtpbin name=rtpbin latency=200 udpsrc
caps=application/x-rtp,media=video,clock-rate=90000,encoding-name=H264
port=5000 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtph264depay ! ffdec_h264 !
ffmpegcolorspace ! autovideosink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0
rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=127.0.0.1 sync=false
async=false

Now I need to map this pipe into a C file and I use the following source

#include <string.h>
#include <math.h>

#include <gst/gst.h>

/* the caps of the sender RTP stream. This is usually negotiated out of band
with
 * SDP or RTSP. */
#define VIDEO_CAPS
"application/x-rtp,media=(string)video,clock-rate=(int)9000,encoding-name=(string)H264"
//#define VIDEO_CAPS
"application/x-rtp,media=video,clock-rate=9000,encoding-name=H264"

#define VIDEO_DEPAY "rtph264depay"
#define VIDEO_DEC   "ffdec_h264"
#define VIDEO_SINK  "autovideosink"

/* the destination machine to send RTCP to. This is the address of the
sender and
 * is used to send back the RTCP reports of this receiver. If the data is
sent
 * from another machine, change this address. */
#define DEST_HOST "127.0.0.1"

/* print the stats of a source */
static void print_source_stats (GObject * source) {
  GstStructure *stats;
  gchar *str;

  g_return_if_fail (source != NULL);

  /* get the source stats */
  g_object_get (source, "stats", &stats, NULL);

  /* simply dump the stats structure */
  str = gst_structure_to_string (stats);
  g_print ("source stats: %s\n", str);

  gst_structure_free (stats);
  g_free (str);
}

/* will be called when gstrtpbin signals on-ssrc-active. It means that an
RTCP
 * packet was received from another source. */
static void on_ssrc_active_cb (GstElement * rtpbin, guint sessid, guint
ssrc, GstElement * depay) {

    GObject *session, *isrc, *osrc;
    g_print ("got RTCP from session %u, SSRC %u\n", sessid, ssrc);

  /* get the right session */
  g_signal_emit_by_name (rtpbin, "get-internal-session", sessid, &session);

  /* get the internal source (the SSRC allocated to us, the receiver */
  g_object_get (session, "internal-source", &isrc, NULL);
  print_source_stats (isrc);

  /* get the remote source that sent us RTCP */
  g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &osrc);
  print_source_stats (osrc);
}

/* will be called when rtpbin has validated a payload that we can depayload
*/
static void
pad_added_cb (GstElement * rtpbin, GstPad * new_pad, GstElement * depay)
{
  GstPad *sinkpad;
  GstPadLinkReturn lres;

  g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad));

  sinkpad = gst_element_get_static_pad (depay, "sink");
  g_assert (sinkpad);

  lres = gst_pad_link (new_pad, sinkpad);
  g_assert (lres == GST_PAD_LINK_OK);
  gst_object_unref (sinkpad);

}


int main (int argc, char *argv[])
{
  GstElement *rtpbin, *rtpsrc, *rtcpsrc, *rtcpsink;
  GstElement *videodepay,
	     *videodec,
	     //*videores,
	     *videoconv,
	     *videosink;

  GstElement *pipeline;
  GMainLoop *loop;
  GstCaps *caps;
  gboolean res;
  GstPadLinkReturn lres;
  GstPad *srcpad, *sinkpad;

  /* always init first */
  gst_init (&argc, &argv);

  /* the pipeline to hold everything */
  pipeline = gst_pipeline_new (NULL);
  g_assert (pipeline);

  /* the udp src and source we will use for RTP and RTCP */
  rtpsrc = gst_element_factory_make ("udpsrc", "rtpsrc");
  g_assert (rtpsrc);
  g_object_set (rtpsrc, "port", 5000, NULL);
  /* we need to set caps on the udpsrc for the RTP data */
  caps = gst_caps_from_string (VIDEO_CAPS);
  g_object_set (rtpsrc, "caps", caps, NULL);
  gst_caps_unref (caps);

  rtcpsrc = gst_element_factory_make ("udpsrc", "rtcpsrc");
  g_assert (rtcpsrc);
  g_object_set (rtcpsrc, "port", 5001, NULL);

  rtcpsink = gst_element_factory_make ("udpsink", "rtcpsink");
  g_assert (rtcpsink);
  g_object_set (rtcpsink, "port", 5005, "host", DEST_HOST, NULL);
  /* no need for synchronisation or preroll on the RTCP sink */
  g_object_set (rtcpsink, "async", FALSE, "sync", FALSE, NULL);

  gst_bin_add_many (GST_BIN (pipeline), rtpsrc, rtcpsrc, rtcpsink, NULL);

  /* the depayloading and decoding */
  videodepay = gst_element_factory_make (VIDEO_DEPAY, "videodepay");
  g_assert (videodepay);
  videodec = gst_element_factory_make (VIDEO_DEC, "videodec");
  g_assert (videodec);
  /* the audio playback and format conversion */
  videoconv = gst_element_factory_make ("ffmpegcolorspace", "videoconv");
  g_assert (videoconv);
/*
  audiores = gst_element_factory_make ("audioresample", "audiores");
  g_assert (audiores);
*/
  videosink = gst_element_factory_make (VIDEO_SINK, "videosink");
  g_assert (videosink);

  /* add depayloading and playback to the pipeline and link */
  gst_bin_add_many (GST_BIN (pipeline), videodepay, videodec, videoconv,
/*videores,*/ videosink, NULL);

  res = gst_element_link_many (videodepay, videodec, videoconv,
/*videores,*/videosink, NULL);
  g_assert (res == TRUE);

  /* the rtpbin element */
  rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
  g_assert (rtpbin);

  g_object_set (G_OBJECT (rtpbin),"latency",200,NULL);

  gst_bin_add (GST_BIN (pipeline), rtpbin);

  /* now link all to the rtpbin, start by getting an RTP sinkpad for session
0 */
  srcpad = gst_element_get_static_pad (rtpsrc, "src");
  sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0");
  lres = gst_pad_link (srcpad, sinkpad);
  g_assert (lres == GST_PAD_LINK_OK);
  gst_object_unref (srcpad);

  /* get an RTCP sinkpad in session 0 */
  srcpad = gst_element_get_static_pad (rtcpsrc, "src");
  sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtcp_sink_0");
  lres = gst_pad_link (srcpad, sinkpad);
  g_assert (lres == GST_PAD_LINK_OK);
  gst_object_unref (srcpad);
  gst_object_unref (sinkpad);

  /* get an RTCP srcpad for sending RTCP back to the sender */
  srcpad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0");
  sinkpad = gst_element_get_static_pad (rtcpsink, "sink");
  lres = gst_pad_link (srcpad, sinkpad);
  g_assert (lres == GST_PAD_LINK_OK);
  gst_object_unref (sinkpad);

  /* the RTP pad that we have to connect to the depayloader will be created
   * dynamically so we connect to the pad-added signal, pass the depayloader
as
   * user_data so that we can link to it. */
  g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb),
videodepay);

  /* give some stats when we receive RTCP */
  //g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK
(on_ssrc_active_cb),videodepay);

  /* set the pipeline to playing */
  g_print ("starting receiver pipeline\n");
  gst_element_set_state (pipeline, GST_STATE_PLAYING);

  /* we need to run a GLib main loop to get the messages */
  loop = g_main_loop_new (NULL, FALSE);
  g_main_loop_run (loop);

  g_print ("stopping receiver pipeline\n");
  gst_element_set_state (pipeline, GST_STATE_NULL);

  gst_object_unref (pipeline);

  return 0;
}


When I launch it I receive the following error
ERROR:rtpclient.c::pad_added_cb: assertion failed: (lres == GST_PAD_LINK_OK)

How I can solve the problem? Do you have any ideas?

G.


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