im newbie in gstreamer and simple try to wath rtsp video flow from Dlink 2103 camera.
When i trying it (just video):
gst-launch rtspsrc location=rtsp://192.168.0.20/live1.sdp ! \
rtph264depay ! \
h264parse ! capsfilter caps="video/x-h264,width=1280,height=800,framerate=(fraction)25/1" !
ffdec_h264 ! ffmpegcolorspace ! autovideosink
Its ok.
When i trying it (just audio):
gst-launch rtspsrc location=rtsp://192.168.0.20/live1.sdp ! \
rtpg726depay ! ffdec_g726 ! audioconvert ! audioresample ! autoaudiosink
Its also ok.
Next i try play both audio and video. gst-launch man page was used for generate something like this:
gst-launch-0.10 -m -vvv -e rtspsrc location=rtsp://192.168.0.20/live1.sdp latency=1000 ! \
gstrtpptdemux name=demuxer demuxer. ! \
queue ! \
rtph264depay ! h264parse ! capsfilter caps="video/x-h264,width=1280,height=800,framerate=(fraction)25/1" ! \
ffdec_h264 ! ffmpegcolorspace ! autovideosink demuxer. ! \
queue !
rtpg726depay ! ffdec_g726 ! audioconvert ! audioresample ! autoaudiosink
But video freez with first frame. I also try this classic way using decodebin (both 1 and 2 ver):
gst-launch-0.10 -v souphttpsrc rtspsrc location=rtsp://192.168.0.20/live1.sdp !
decodebin name=decoder decoder. ! queue ! audioconvert ! audioresample !
autoaudiosink decoder. ! \
ffmpegcolorspace ! autovideosink
BUT it also freez on first frame.
ONE way i have success it using playbin...
gst-launch-0.10 playbin2 uri=rtsp://192.168.0.20/live1.sdp
IS IT my bad pipeline or something wrong with dlink camera? Can you tell me key-word that i should to learn more?
thanks in advance !
Solution 1 (Tested)
Ok I made my own RTSP server to test
I created a RTSP server using video and audio test srcs using the following info ( http://www.ip-sense.com/linuxsense/how-to-develop-a-rtsp-server-in-linux-using-gstreamer/ )
/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
* Copyright (c) 2012 enthusiasticgeek <enthusiasticgeek@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
//Edited by: enthusiasticgeek (c) 2012 for Stack Overflow Sept 11, 2012
//###########################################################################
//Important
//###########################################################################
//On ubuntu: sudo apt-get install libgstrtspserver-0.10-0 libgstrtspserver-0.10-dev
//Play with VLC
//rtsp://localhost:8554/test
//video decode only: gst-launch -v rtspsrc location="rtsp://localhost:8554/test" ! rtph264depay ! ffdec_h264 ! autovideosink
//audio and video:
//gst-launch -v rtspsrc location="rtsp://localhost:8554/test" name=demux demux. ! queue ! rtph264depay ! ffdec_h264 ! ffmpegcolorspace ! autovideosink sync=false demux. ! queue ! rtppcmadepay ! alawdec ! autoaudiosink
//###########################################################################
#include <gst/gst.h>
#include <gst/rtsp-server/rtsp-server.h>
/* define this if you want the resource to only be available when using
* user/admin as the password */
#undef WITH_AUTH
/* this timeout is periodically run to clean up the expired sessions from the
* pool. This needs to be run explicitly currently but might be done
* automatically as part of the mainloop. */
static gboolean
timeout (GstRTSPServer * server, gboolean ignored)
{
GstRTSPSessionPool *pool;
pool = gst_rtsp_server_get_session_pool (server);
gst_rtsp_session_pool_cleanup (pool);
g_object_unref (pool);
return TRUE;
}
int
main (int argc, char *argv[])
{
GMainLoop *loop;
GstRTSPServer *server;
GstRTSPMediaMapping *mapping;
GstRTSPMediaFactory *factory;
#ifdef WITH_AUTH
GstRTSPAuth *auth;
gchar *basic;
#endif
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
/* create a server instance */
server = gst_rtsp_server_new ();
/* get the mapping for this server, every server has a default mapper object
* that be used to map uri mount points to media factories */
mapping = gst_rtsp_server_get_media_mapping (server);
#ifdef WITH_AUTH
/* make a new authentication manager. it can be added to control access to all
* the factories on the server or on individual factories. */
auth = gst_rtsp_auth_new ();
basic = gst_rtsp_auth_make_basic ("user", "admin");
gst_rtsp_auth_set_basic (auth, basic);
g_free (basic);
/* configure in the server */
gst_rtsp_server_set_auth (server, auth);
#endif
/* make a media factory for a test stream. The default media factory can use
* gst-launch syntax to create pipelines.
* any launch line works as long as it contains elements named pay%d. Each
* element with pay%d names will be a stream */
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_launch (factory, "( "
"videotestsrc ! video/x-raw-yuv,width=320,height=240,framerate=10/1 ! "
"x264enc ! queue ! rtph264pay name=pay0 pt=96 ! audiotestsrc ! audio/x-raw-int,rate=8000 ! alawenc ! rtppcmapay name=pay1 pt=97 "")");
/* attach the test factory to the /test url */
gst_rtsp_media_mapping_add_factory (mapping, "/test", factory);
/* don't need the ref to the mapper anymore */
g_object_unref (mapping);
/* attach the server to the default maincontext */
if (gst_rtsp_server_attach (server, NULL) == 0)
goto failed;
/* add a timeout for the session cleanup */
g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
/* start serving, this never stops */
g_main_loop_run (loop);
return 0;
/* ERRORS */
failed:
{
g_print ("failed to attach the server\n");
return -1;
}
}
Makefile
# Copyright (c) 2012 enthusiasticgeek
# RTSP demo for Stack Overflow
sample:
gcc -Wall -I/usr/include/gstreamer-0.10 rtsp.c -o rtsp `pkg-config --libs --cflags gstreamer-0.10 gstreamer-rtsp-0.10` -lglib-2.0 -lgstrtspserver-0.10 -lgstreamer-0.10
Tested the decoding pipeline. It works fine!
gst-launch -v rtspsrc location="rtsp://localhost:8554/test" name=demux demux. ! queue ! rtph264depay ! ffdec_h264 ! ffmpegcolorspace ! autovideosink sync=false demux. ! queue ! rtppcmadepay ! alawdec ! autoaudiosink
solution 2 (Tested)
Try using mux/demux combination
`gst-launch-1.0 -e rtspsrc location='rtsp://localhost:554' latency=0 name=d d. ! queue ! capsfilter caps="application/x-rtp,media=video" ! rtph264depay ! mpegtsmux name=mux ! filesink location=file.ts d. ! queue ! capsfilter caps="application/x-rtp,media=audio" ! decodebin ! audioconvert ! audioresample ! lamemp3enc ! mux.`
To decode the pipeline
gst-launch filesrc location=file.ts ! typefind ! mpegtsdemux name=demux demux. ! queue ! h264parse ! ffdec_h264 ! autovideosink demux. ! queue ! mp3parse ! ffdec_mp3 ! audioconvert ! autoaudiosink demux.
Solution 3 (Untested)
Try using a Tee based approach. Also run gst-launch-0.10 -v playbin2 uri=rtsp://192.168.0.20/live1.sdp. Notice the verbose option. This will give you a lot of hints on how to construct the pipeline.
Have a common source to Tee bin -> fork this into two pipelines one for audio decode and one for video decode.
src -> tee (fork into two branches - sub pipelines) -> (branch 1 will have audio demux -> audio decoder -> audio sink) and (branch 2 will have video demux -> video decoder -> video sink)
Give the following a shot (untested). You may have to tweak this pipeline a bit to get it to work but you will get an idea.
gst-launch rtspsrc location=rtsp://192.168.0.20/live1.sdp ! queue ! tee name=t !\
rtph264depay t. ! \
h264parse t. ! capsfilter caps="video/x-h264,width=1280,height=800,framerate=(fraction)25/1" t. !
ffdec_h264 t. ! ffmpegcolorspace t. ! autovideosink t. ! queue ! \
rtpg726depay ! ffdec_g726 ! audioconvert ! audioresample ! autoaudiosink