Gstreamer application for receiving audio and video over RTP streaming.I have written an app for expanding the RTP reciever sample code from audio player to audio adn video playerWhile running the code below I am getting the Error given below pad_added_cb: assertion failed: (lres == GST_PAD_LINK_OK) Can you please see the code given below for reference?I am suing queues with pipeline for spereating audio and video. DO we need to link the queues? #include <string.h> #include <math.h>#include <gst/gst.h> #define AV 1#define VIDEO_CAPS "application/x-rtp,payload=96,encoding-name=H264,clock-rate=90000" #define VIDEO_DEPAY "rtph264depay" #define VIDEO_DEC "ffdec_h264" #define VIDEO_PARSE "h264parse" #define VIDEO_COLORSPACE "ffmpegcolorspace" #define VIDEO_DEI "deinterlace" #define VIDEO_SINK "ximagesink" #define VIDEO_PORT 9700 #define AUDIO_CAPS "application/x-rtp,media=(string)audio,clock-rate=(int)16000,encoding-name=(string)AMR-WB,encoding-params=(string)1,octet-align=(string)1" #define AUDIO_DEPAY "rtpamrdepay" #define AUDIO_DEC "amrwbdec" #define AUDIO_SINK "alsasink" #define AUDIO_PORT 1200 /* 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 "107.108.198.131"/* 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); } /* build a pipeline equivalent to: * * gst-launch -v gstrtpbin name=rtpbin udpsrc caps="application/x-rtp,payload=96,encoding-name=H264,clock-rate=90000" port=9200 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtph264depay ! * h264parse ! ffdec_h264 ! ffmpegcolorspace ! deinterlace ! ximagesink udpsrc port=42000 caps="application/x-rtp,media=(string)audio,payload=(int)101,clock-rate=(int)16000,encoding-name= * (string)AMR-WB,encoding-params=(string)1,octet-align=(string)1" port=42000 ! rtpbin.recv_rtp_sink_1 rtpbin. ! rtpamrdepay ! amrwbdec ! audioresample ! audioconvert ! alsasink sync=false */ int main (void)//av_play { GstElement *rtpbin; GstElement *vrtpsrc, *vrtcpsrc, *vrtcpsink; GstElement *artpsrc, *artcpsrc, *artcpsink; GstElement *videodepay, *videodec, *videoparse, *colorspace, *deinterlacer,*videosink;//h264parse ! ffdec_h264 ! ffmpegcolorspace ! deinterlace GstElement *audiodepay, *audiodec, *audiores, *audioconv, *audiosink; GstElement *pipeline; GstElement *queueA,* *queueV; GMainLoop *loop; GstCaps *vcaps,*acaps; gboolean res,res1; GstPadLinkReturn lres,lres1; GstPad *asrcpad, *asinkpad,*vsrcpad, *vsinkpad; /* always init first */ gst_init (NULL, NULL); g_printerr("Client App for Audio and Video\n"); /* 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 */ vrtpsrc = gst_element_factory_make ("udpsrc", "vrtpsrc"); g_assert (vrtpsrc); g_object_set (vrtpsrc, "port", VIDEO_PORT, NULL); /* we need to set caps on the udpsrc for the RTP data */ vcaps = gst_caps_from_string (VIDEO_CAPS); g_object_set (vrtpsrc, "caps", vcaps, NULL); gst_caps_unref (vcaps); #if AV artpsrc = gst_element_factory_make ("udpsrc", "artpsrc"); g_assert (artpsrc); g_object_set (artpsrc, "port", AUDIO_PORT, NULL); /* we need to set caps on the udpsrc for the RTP data */ acaps = gst_caps_from_string (AUDIO_CAPS); g_object_set (artpsrc, "caps", acaps, NULL); gst_caps_unref (acaps); #endif vrtcpsrc = gst_element_factory_make ("udpsrc", "vrtcpsrc"); g_assert (vrtcpsrc); g_object_set (vrtcpsrc, "port", 5003, NULL); vrtcpsink = gst_element_factory_make ("udpsink", "vrtcpsink"); g_assert (vrtcpsink); g_object_set (vrtcpsink, "port", 5007, "host", DEST_HOST, NULL); /* no need for synchronisation or preroll on the RTCP sink */ g_object_set (vrtcpsink, "async", FALSE, "sync", FALSE, NULL); #if AV artcpsrc = gst_element_factory_make ("udpsrc", "artcpsrc"); g_assert (artcpsrc); g_object_set (artcpsrc, "port", 5003, NULL); artcpsink = gst_element_factory_make ("udpsink", "rtcpsink"); g_assert (artcpsink); g_object_set (artcpsink, "port", 5007, "host", DEST_HOST, NULL); /* no need for synchronisation or preroll on the RTCP sink */ g_object_set (artcpsink, "async", FALSE, "sync", FALSE, NULL); #endif /* 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); #if AV /* the depayloading and decoding */ audiodepay = gst_element_factory_make (AUDIO_DEPAY, "audiodepay"); g_assert (audiodepay); audiodec = gst_element_factory_make (AUDIO_DEC, "audiodec"); g_assert (audiodec); #endif //*videoparse, *colorspace, *deinterlacer,*videosink; /* the audio playback and format conversion */ videoparse = gst_element_factory_make (VIDEO_PARSE, "videoparse"); g_assert (videoparse); colorspace = gst_element_factory_make (VIDEO_COLORSPACE, "colorspace"); g_assert (colorspace); deinterlacer = gst_element_factory_make (VIDEO_DEI, "deinterlacer"); g_assert (deinterlacer); videosink = gst_element_factory_make (VIDEO_SINK, "videosink"); g_assert (videosink); #if AV /* the audio playback and format conversion */ audioconv = gst_element_factory_make ("audioconvert", "audioconv"); g_assert (audioconv); audiores = gst_element_factory_make ("audioresample", "audiores"); g_assert (audiores); audiosink = gst_element_factory_make (AUDIO_SINK, "audiosink"); g_assert (audiosink); g_object_set (audiosink,"sync", FALSE, NULL); #endif queueA = gst_element_factory_make("queue", "queue-audio"); queueV = gst_element_factory_make("queue", "queue-video"); /* add depayloading and playback to the pipeline and link */ #if AV gst_bin_add_many (GST_BIN (pipeline), vrtpsrc,videodepay,videodec,videoparse,queueV,colorspace,deinterlacer,videosink,artpsrc, artcpsrc, artcpsink, \ audiodepay,audiodec,queueA,audioconv,audiores,audiosink, NULL); res = gst_element_link_many (audiodepay, audiodec, audioconv, audiores, audiosink, NULL); if(!res) g_printerr("Linking audio elements failed\n"); res1 = gst_element_link_many (videodepay, videodec, colorspace,deinterlacer, videosink, NULL); if(!res1) g_printerr("Linking video elements failed\n"); #else gst_bin_add_many (GST_BIN (pipeline), audiodepay, audiodec, audioconv, audiores, audiosink, NULL); res = gst_element_link_many (audiodepay, audiodec, audioconv, audiores, audiosink, NULL); g_assert (res == TRUE);#endif rtpbin = gst_element_factory_make("gstrtpbin", "rtpbin"); if ( !rtpbin ) { g_printerr("Failed to create gstrtpbin\n"); g_assert (rtpbin); return 0; } gst_bin_add (GST_BIN (pipeline), rtpbin); /* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */ vsrcpad = gst_element_get_static_pad (vrtpsrc, "src"); vsinkpad = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0"); lres = gst_pad_link (vsrcpad, vsinkpad); g_assert (lres == GST_PAD_LINK_OK); gst_object_unref (vsrcpad); #if AV asrcpad = gst_element_get_static_pad (artpsrc, "src"); asinkpad = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_1"); lres1 = gst_pad_link (asrcpad, asinkpad); g_assert (lres1 == GST_PAD_LINK_OK); gst_object_unref (asrcpad); #endif#if 1 /* get an RTCP sinkpad in session 0 */ asrcpad = gst_element_get_static_pad (artcpsrc, "src"); asinkpad = gst_element_get_request_pad (rtpbin, "recv_rtcp_sink_0"); lres = gst_pad_link (asrcpad, asinkpad); g_assert (lres == GST_PAD_LINK_OK); gst_object_unref (asrcpad); gst_object_unref (asinkpad); /* get an RTCP srcpad for sending RTCP back to the sender */ asrcpad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0"); asinkpad = gst_element_get_static_pad (artcpsink, "sink"); lres = gst_pad_link (asrcpad, asinkpad); g_assert (lres == GST_PAD_LINK_OK); gst_object_unref (asinkpad); #if AV /* get an RTCP sinkpad in session 1 */ vsrcpad = gst_element_get_static_pad (vrtcpsrc, "src"); vsinkpad = gst_element_get_request_pad (rtpbin, "recv_rtcp_sink_1"); lres1 = gst_pad_link (vsrcpad, vsinkpad); g_assert (lres == GST_PAD_LINK_OK); gst_object_unref (vsrcpad); gst_object_unref (vsinkpad); /* get an RTCP srcpad for sending RTCP back to the sender */ vsrcpad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_1"); vsinkpad = gst_element_get_static_pad (vrtcpsink, "sink"); lres1 = gst_pad_link (vsrcpad, vsinkpad); g_assert (lres == GST_PAD_LINK_OK); gst_object_unref (vsinkpad); #endif #endif /* 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), audiodepay); g_print ("pad added cb for audio\n"); #if AV g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb), videodepay); g_print ("pad added cb for video\n"); #endif /* give some stats when we receive RTCP */ #if 0 g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK (on_ssrc_active_cb), audiodepay); #endif /* 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;} <div> </div><div> </div><div> </div><div> </div> Cheers! Deepth