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 player
While 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;
}
Cheers! Deepth
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