gstreamer -- appsrc 到 rtsp推流

经常用到将appsrc中填充数据后，以rtsp流的形式推出。和rtmp不同的是，rtmp需要用ngnix做为server，而rtsp就是通过代码自己搭建一个server，代码如下

static int need_data_rtsp (GstElement * appsrc, guint unused, data * ptr)
{


if(*(ptr->Stop))
{
cout<<"stop pushing"<<endl;
return 0;
}
  GstBuffer *buffer;
  guint size;
  GstFlowReturn ret =GST_FLOW_OK;


  size = (ptr->Input->size().width) * (ptr->Input->size().height) * 3;
   
  buffer = gst_buffer_new_allocate (NULL, size, NULL);
  
  GstMapInfo map;
  
  gst_buffer_map(buffer,&map,GST_MAP_READ);

  ptr->Mux->lock();
  memcpy(map.data,ptr->Input->data,size);
  ptr->Mux->unlock();
 
  guint64 s = ( getTickCount() - ptr->t) * GST_SECOND / getTickFrequency();
  
  ptr->t = getTickCount();


  GST_BUFFER_PTS (buffer) = ptr->Timestamp;

  GST_BUFFER_DURATION (buffer) = s;
 
  ptr->Timestamp += GST_BUFFER_DURATION (buffer) ;

  g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
  
  gst_buffer_unmap(buffer,&map);
  std::cout<<"pushing"<<std::endl;
 
  gst_buffer_unref (buffer);


   if ( *(ptr->Stop)  ) {
    /* somet