Gstreamer rtsp播放(带声音)

Question

我是新手在gstreamer和简单的尝试rtsp视频流从Dlink 2103相机。

当我尝试它(仅仅是视频)时：

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

没问题。

当我尝试时(仅音频)：

gst-launch rtspsrc location=rtsp://192.168.0.20/live1.sdp ! \
rtpg726depay !  ffdec_g726 !  audioconvert ! audioresample ! autoaudiosink

这也没问题。

接下来，我尝试播放音频和视频。gst-launch手册页用于生成如下内容：

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

但视频freez与第一帧。我也尝试了这个经典的方法，使用decodebin (1和2版本)：

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

但它在第一帧也是freez。

我成功使用playbin的一种方法是...

gst-launch-0.10 playbin2 uri=rtsp://192.168.0.20/live1.sdp

是我的管道坏了还是dlink摄像头出了问题？你能告诉我应该学习更多的关键字吗？

提前感谢！

Answer 1

解决方案1(测试)

好的，我做了我自己的RTSP服务器来测试

我使用以下信息创建了一个使用视频和音频测试srcs的RTSP服务器( 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

已测试解码管道。它工作得很好！

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

解决方案2(测试)

尝试使用mux/demux组合

 `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.`

对管道进行解码

gst-launch filesrc location=file.ts ! typefind ! mpegtsdemux name=demux demux. ! queue ! h264parse ! ffdec_h264 ! autovideosink demux. ! queue ! mp3parse ! ffdec_mp3 ! audioconvert ! autoaudiosink demux.

解决方案3(未经测试)

尝试使用基于Tee的方法。还要运行gst-launch-0.10 -v playbin2 uri=rtsp://192.168.0.20/live1.sdp。请注意详细选项。这将给你很多关于如何构建管道的提示。

有一个通向Tee bin ->的公共源代码，将其分成两条管道，一条用于音频解码，另一条用于视频解码。

源->三通(分支到两个分支-子管道) -> (分支1将具有音频解复用器->音频解码器->音频接收器)和(分支2将具有视频解复用器->视频解码器->视频接收器)

尝试以下内容(未经测试)。您可能需要对此管道进行一些调整才能使其正常工作，但您会有所了解的。

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