Gstreamer rtspsrc 停止不发送teardown 命令

最新推荐文章于 2024-08-20 16:53:54 发布

心动随你

最新推荐文章于 2024-08-20 16:53:54 发布

阅读量2.6k

点赞数

CC 4.0 BY-SA版权

分类专栏：流媒体传输 Gstreamer 文章标签： linux Gstreamer

本文链接：https://blog.youkuaiyun.com/jintaofu/article/details/51596119

Gstreamer 同时被 2 个专栏收录

4 篇文章

订阅专栏

流媒体传输

3 篇文章

订阅专栏

本文档详细介绍了针对GStreamer RTSP源插件中的状态转换逻辑进行的一系列优化措施。主要内容包括：对发送命令的处理流程进行改进，避免不必要的请求冲突；在不同状态变化时，如从暂停到准备状态，调整关闭命令的发送方式，确保资源能够正确释放。

摘要生成于 C知道，由 DeepSeek-R1 满血版支持，前往体验 >

修改 gstrtspsrc.c

参考加粗字体

 <pre name="code" class="cpp">gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask)
{
  gint old;
  gboolean flushed = FALSE;

  /* start new request */
  gst_rtspsrc_loop_start_cmd (src, cmd);

  GST_DEBUG_OBJECT (src, "sending cmd %s", cmd_to_string (cmd));

  GST_OBJECT_LOCK (src);
  old = src->pending_cmd;
  if (old == CMD_RECONNECT) {
    GST_DEBUG_OBJECT (src, "ignore, we were reconnecting");
    cmd = CMD_RECONNECT;
  }
<strong><strong>  // vvv ----- Patch----
  else if (old == CMD_CLOSE) {
	  /* our CMD_CLOSE might have interrutped CMD_LOOP. gst_rtspsrc_loop
	   * will send a CMD_WAIT which would cancel our pending CMD_CLOSE (if
	    * still pending). We just avoid it here by making sure CMD_CLOSE is
	    * * still the pending command. */
	  GST_DEBUG_OBJECT (src, "ignore, we were closing");
  }
  //if (old != CMD_WAIT) {
  else if (old != CMD_WAIT) {
 // ^^^  ---- Patch ---</strong></strong>

    src->pending_cmd = CMD_WAIT;
    GST_OBJECT_UNLOCK (src);
    /* cancel previous request */
    GST_DEBUG_OBJECT (src, "cancel previous request %s", cmd_to_string (old));
    gst_rtspsrc_loop_cancel_cmd (src, old);
    GST_OBJECT_LOCK (src);
  }

gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
{
  GstRTSPSrc *rtspsrc;
  GstStateChangeReturn ret;

  rtspsrc = GST_RTSPSRC (element);

  switch (transition) {
    case GST_STATE_CHANGE_NULL_TO_READY:
      if (!gst_rtspsrc_start (rtspsrc))
        goto start_failed;
      break;
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      /* init some state */
      rtspsrc->cur_protocols = rtspsrc->protocols;
      /* first attempt, don't ignore timeouts */
      rtspsrc->ignore_timeout = FALSE;
      rtspsrc->open_error = FALSE;
      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
      break;
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
      set_manager_buffer_mode (rtspsrc);
      /* fall-through */
    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
      /* unblock the tcp tasks and make the loop waiting */
      if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
        /* make sure it is waiting before we send PAUSE or PLAY below */
        GST_RTSP_STREAM_LOCK (rtspsrc);
        GST_RTSP_STREAM_UNLOCK (rtspsrc);
      }
      break;
    case GST_STATE_CHANGE_PAUSED_TO_READY:
      break;
    default:
      break;
  }

  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
  if (ret == GST_STATE_CHANGE_FAILURE)
    goto done;

  switch (transition) {
    case GST_STATE_CHANGE_NULL_TO_READY:
      ret = GST_STATE_CHANGE_SUCCESS;
      break;
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      ret = GST_STATE_CHANGE_NO_PREROLL;
      break;
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
      ret = GST_STATE_CHANGE_SUCCESS;
      break;
    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
      /* send pause request and keep the idle task around */
      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
      ret = GST_STATE_CHANGE_NO_PREROLL;
      break;
    case GST_STATE_CHANGE_PAUSED_TO_READY:
  <strong>  	<strong>// vvv ----patch
		// gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
		gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_ALL);
		/* make sure teardown request is not interrupted by stop below */
		GST_RTSP_STREAM_LOCK (rtspsrc);
		GST_RTSP_STREAM_UNLOCK (rtspsrc);
        // ^^^ ---- patch</strong></strong>
      ret = GST_STATE_CHANGE_SUCCESS;
      break;