Jetson 学习笔记(十二)：CSI摄像头实现rtsp流的传输并对动态获取多路流进行探索

文章目录

需求
安装依赖
下载、编译gst-rtsp-server
测试
源代码介绍
动态获取多路流的rtsp server
错误
参考

需求

gcc 版本7.5.0
g++ 版本7.5.0
ubuntu 版本18.04
gst-rtsp-server 版本1.8.0

安装依赖

sudo apt-get install gtk-doc-tools 
sudo apt-get install libgstreamer1.0-0 gstreamer1.0-plugins-base 
sudo apt-get install gstreamer1.0-plugins-good gstreamer1.0-plugins-bad gstreamer1.0-plugins-ugly 
sudo apt-get install gstreamer1.0-libav gstreamer1.0-doc gstreamer1.0-tools 
sudo apt-get install gstreamer1.0-x gstreamer1.0-alsa gstreamer1.0-gl gstreamer1.0-gtk3 gstreamer1.0-qt5 gstreamer1.0-pulseaudio
sudo apt-get install libgstreamer1.0-dev libgstreamer-plugins-base1.0-dev

下载、编译gst-rtsp-server

git clone git://anongit.freedesktop.org/gstreamer/gst-rtsp-server
或者
wget https://github.com/GStreamer/gst-rtsp-server/archive/1.8.zip

cd gst-rtsp-server
git checkout remotes/origin/1.8 or git clone https://github.com/GStreamer/common.git
./autogen.sh
make -j4 
sudo make install

//进入demo
cd example

测试

1、切换到examples目录：
cd examples
2、搭建Rtsp Server：
./test-launch "( videotestsrc ! x264enc ! rtph264pay name=pay0 pt=96 )"


直接读取摄像头（笔记本电脑一般自带摄像头，台式机请插入USB摄像头）视频的命令就是它了：
$ ./test-launch "( v4l2src ! video/x-raw-yuv,format='fourcc'YUY2,width=640,height=480 ! ffmpegcolorspace ! x264enc ! rtph264pay name=pay0 pt=96 )"

直接读取CSI摄像头
./test-launch "nvarguscamerasrc ! nvvidconv ! clockoverlay ! omxh264enc ! rtph264pay pt=96 name=pay0"---成功

./test-launch "(  mfw_v4lsrc device=/dev/video0 ! queue ! vpuenc codec=6 ! rtph264pay name=pay0 pt=96 )"---失败


3、播放rtsp流：
gst-launch-1.0 playbin uri=rtsp://127.0.0.1:8554/test
或者直接通过VLC打开以及通过opencv也可以打开

# 发送1
gst-launch-1.0 videotestsrc ! video/x-raw,format=I420 ! omxh264enc ! video/x-h264,stream-format=byte-stream ! rtph264pay mtu=1400 ! udpsink host=127.0.0.1 port=5000

# 接收1
gst-launch-1.0 udpsrc port=5000 ! 'application/x-rtp,encoding-name=(string)H264' ! rtph264depay ! h264parse ! omxh264dec ! nvoverlaysink sync=false async=false 

# 发送2
./test-launch "(videotestsrc ! video/x-raw,format=I420,framerate= 25/1 !  x264enc ! video/x-h264,stream-format=byte-stream ! rtph264pay name=pay0 pt=96)"

# 接收2
gst-launch-1.0 rtspsrc location=rtsp://127.0.01:8554/test ! rtph264depay ! h264parse ! omxh264dec ! nvoverlaysink sync=false async=false

源代码介绍

test-readme.c

#include 
#include 
int main (int argc, char *argv[]) {
 	//声明相关对象
  	GMainLoop *loop;
  	GstRTSPServer *server;
  	GstRTSPMountPoints *mounts;
  	GstRTSPMediaFactory *factory;

	//构建 rtsp 服务器
  	gst_init (&argc, &argv);
  	loop = g_main_loop_new (NULL, FALSE);					// 创建 rtsp 服务器的主消息循环，也是默认的消息循环。
  	server = gst_rtsp_server_new ();						// 创建 rtsp 服务器对象
  	mounts = gst_rtsp_server_get_mount_points (server);		// 获取 rtsp 服务器的装载点集合的引用
  															// 装载点集合 mounts 是服务器 server 的属性
  	factory = gst_rtsp_media_factory_new ();				// 创建媒体工厂，用来产生媒体数据流 
  	gst_rtsp_media_factory_set_launch (factory, "( videotestsrc is-live=1 ! x264enc ! rtph264pay name=pay0 pt=96 )");
  	gst_rtsp_media_factory_set_shared (factory, TRUE);
  	gst_rtsp_mount_points_add_factory (mounts, "/test", factory);	// 把媒体工厂添加到装载点集合
  	g_object_unref (mounts);
  	gst_rtsp_server_attach (server, NULL);					// 把服务器附加到默认的消息循环。

  	//运行 rtsp 服务器
  	g_print ("stream ready at rtsp://127.0.0.1:8554/testn");
  	g_main_loop_run (loop);

  	return 0;
}

服务器管理另外四个对象： GstRTSPSessionPool、GstRTSPMountPoints、 GstRTSPAuth 和 GstRTSPThreadPool。

GstRTSPSessionPool 是一个跟踪服务器中所有活动会话的对象。通常会为每个为某个媒体流执行设置请求的客户机保留一个会话。它包含客户端与服务器协商以接收特定流的配置，即UDP使用的传输和端口对以及流的状态。会话池的默认实现通常就足够了，但服务器可以使用替代实现。

GstRTSPMountPoints 对象更有趣，在服务器对象有用之前需要更多的配置。此对象管理从请求URL到特定流的映射及其配置。我们将在下一个主题中解释如何配置此对象。

GstRTSPAuth是对用户进行身份验证并授权用户执行的操作的对象。默认情况下，服务器没有 GstRTSPAuth 对象，因此不会尝试执行任何身份验证或授权。

GstRTSPThreadPool 管理用于客户端连接和媒体管道的线程。服务器有一个线程池的默认实现，在大多数情况下应该足够了。

动态获取多路流的rtsp server

类似example里面的C文件，将你所命名的文件名(C文件)，加在Makefile，Makefile.am,Makefile.in对应位置，直接通过sudo make -j8即可完成修改，将生成对应可执行文件直接./运行即可。

代码1

#include 
 
#include 
#include 
 
void* testFun(void *args)
{
  GMainLoop *loop = (GMainLoop *) args;
  g_main_loop_run (loop);
}
 
int
main (int argc, char *argv[])
{
  GMainLoop *loop;
  GstRTSPServer *server;
  GstRTSPMountPoints *mounts;
  GstRTSPMediaFactory *factory;
 
  gst_init (&argc, &argv);
 
  loop = g_main_loop_new (NULL, FALSE);
  pthread_t tTest;
  pthread_create(&tTest, NULL, testFun, loop);
 
  /* create a server instance */
  server = gst_rtsp_server_new ();
  gst_rtsp_server_set_service (server, "8555");
 
  /* get the mount points for this server, every server has a default object
   * that be used to map uri mount points to media factories */
  mounts = gst_rtsp_server_get_mount_points (server);
 
  /* 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 is-live=1 ! x264enc ! rtph264pay name=pay0 pt=96 )");
 
  gst_rtsp_media_factory_set_shared (factory, TRUE);
 
  /* attach the test factory to the /test url */
  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
 
  /* don't need the ref to the mapper anymore */
  g_object_unref (mounts);
 
  /* attach the server to the default maincontext */
  gst_rtsp_server_attach (server, NULL);
 
  /* start serving */
  g_print ("stream ready at rtsp://127.0.0.1:8554/testn");
  //g_main_loop_run (loop);
  pthread_join(tTest, NULL);
 
  return 0;
}

代码2：

#include 
#include 
 
const char* port = "10001";
 
static void
handle_client (GstRTSPClient * client, GstRTSPContext * ctx,
    GstRTSPServer * server, gpointer user_data)
{
  GstRTSPClientClass *klass;
  GstRTSPMountPoints *mounts;
  GstRTSPMediaFactory *factory;
  GstRTSPUrl *uri;
  gchar *path;
  gchar *launch = "( videotestsrc is-live=1 ! x264enc ! rtph264pay name=pay0 pt=96 )";
 
  uri = ctx->uri;
 
  if (!uri)
    return;
 
  klass = GST_RTSP_CLIENT_GET_CLASS (client);
  path = klass->make_path_from_uri (client, uri);
 
  mounts = gst_rtsp_server_get_mount_points (server);
  factory = gst_rtsp_mount_points_match (mounts, path, NULL);
  if (!factory)
  {
    factory = gst_rtsp_media_factory_new ();
    gst_rtsp_media_factory_set_launch (factory, launch);
    gst_rtsp_media_factory_set_shared (factory, TRUE);
 
    //g_signal_connect (factory, "media-constructed", (GCallback)
      //                                              media_constructed, NULL);
 
    gst_rtsp_mount_points_add_factory (mounts, path, factory);
    g_print ("new factory: %sn", launch);
  }
  else
  {
    g_object_unref (factory);
  }
  g_object_unref (mounts);
  g_free (path);
  //g_free (launch);
}
 
static void
client_connected (GstRTSPServer * server,
    GstRTSPClient * client, gpointer user_data)
{
  g_signal_connect_object (client, "options-request", (GCallback)
      handle_client, server, G_CONNECT_AFTER);
}
 
static gboolean
timeout (GstRTSPServer * server)
{
  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;
  GError *error = NULL;
 
  gst_init (&argc, &argv);
 
  loop = g_main_loop_new (NULL, FALSE);
 
  /* create a server instance */
  server = gst_rtsp_server_new ();
  gst_rtsp_server_set_service (server, port);
 
  /* attach the server to the default maincontext */
  gst_rtsp_server_attach (server, NULL);
 
  g_signal_connect (server, "client-connected", (GCallback)
      client_connected, NULL);
 
  g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
 
  g_object_unref (server);
 
  /* start serving */
  g_print ("rtsp://127.0.0.1:%s/n", port);
  g_main_loop_run (loop);
 
  return 0;
}

错误

Error generated. /dvs/git/dirty/git-master_linux/multimedia/nvgstreamer/gst-nvarguscamera/gstnvargus
解决办法：sudo systemctl restart nvargus-daemon (可以重启nvargus-daemon，重启后程序可以正常运行

)

参考

基于树莓派板子

使用树莓派专用摄像头实现rtsp流的传输，并调用Opencv显示
＜树莓派＞——树莓派通过csi摄像头实时传输rtsp流
树莓派CSI/USB摄像头使用mjpg实现网页摄像头监控
Liunx树莓派(ARM)开发篇—基于mjpg-streamer的摄像头（CSI摄像头）监控（完整流程）（附代码）
树莓派CSI摄像头实现rtsp流的传输，笔记本使用Python调用Opencv显示

基于jetson nano

gstreamer之RTSP Server编译及注意事项
下载gst_rtsp_server地址
Gstreamer——搭建RTSP服务器
记录tx2上安装配置gestermer进而使用gst-rtsp-server
Gstreamer视频传输测试gst-launch
gst-launch-1.0在Linux下的命令
linux环境下用GStreamer实现rtsp取流播放

Jetson 学习笔记(十二)：CSI摄像头实现rtsp流的传输并对动态获取多路流进行探索

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