基於gstreamer的支持動態獲取多路流的rtsp server示例

一、簡介

最近在做rtsp server相關的事情，調研了一些開源的服務器，大部分都是作爲獨立的進程啓動，有點不符合自己的場景。偶然發現gstreamer，名氣很大，但是用的人卻很少。粗略百度了下相關資料--很少。大部分示例都是提供一路流，或者事先寫死幾路，沒法根據自己播放訪問時的url判斷流存不存在並動態創建。後面花了點時間稍微瞭解了下，整理出一份動態創建流的代碼，分享出來。

一共有兩種方式能達到效果，一種採用main_loop_run運行在線程裏，而創建factory等操作可在有需要的時候動態創建。代碼基於gst自帶的示例test-readme.c改造，編譯同gst示例

二、方法一代碼

#include <gst/gst.h>

#include <gst/rtsp-server/rtsp-server.h>
#include <pthread.h>

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/test\n");
  //g_main_loop_run (loop);
  pthread_join(tTest, NULL);

  return 0;
}

三、方法二的代碼

#include <gst/gst.h>
#include <gst/rtsp-server/rtsp-server.h>

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: %s\n", 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;
}

由於偷懶，handle_client函數裏直接用的teat-readme裏的代碼。如果想獲取自定義的流，可以將改函數裏的內容替換爲test-appsrc相關代碼

四、下載鏈接

https://download.csdn.net/download/yingyemin/11016282