[gst-devel] C code for rtp h264 decoding. I can't find how to solve the error. Read is insteresting
Marco Ballesio
gibrovacco at gmail.com
Thu Jan 27 18:28:50 CET 2011
Hi,
On Thu, Jan 27, 2011 at 1:34 PM, giorgino <giorgio9 at libero.it> wrote:
..snip..
> #define VIDEO_CAPS
> "application/x-rtp,media=(string)video,clock-rate=(int)9000,encoding-name=(string)H264"
typo, s/9000/90000 or, that is the same:
"application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264"
Regards
> //#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.
>
>
> --
> View this message in context: http://gstreamer-devel.966125.n4.nabble.com/C-code-for-rtp-h264-decoding-I-can-t-find-how-to-solve-the-error-Read-is-insteresting-tp3242017p3242017.html
> Sent from the GStreamer-devel mailing list archive at Nabble.com.
>
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