问题描述
在实现gstreamer插件以在Android上播放RTP视频时,我遇到了一些问题。我有以下代码(可以正常工作):
full_pipeline_description = g_strdup_printf("playbin3 uri=%s",uri);
gub_log_pipeline(pipeline,"Using pipeline: %s",full_pipeline_description);
pipeline->pipeline = gst_parse_launch(full_pipeline_description,&err);
g_free(full_pipeline_description);
if (err) {
gub_log_pipeline(pipeline,"Failed to create pipeline: %s",err->message);
return;
}
vsink = gst_parse_bin_from_description(gub_get_video_branch_description(),TRUE,NULL);
gub_log_pipeline(pipeline,"Using video sink: %s",gub_get_video_branch_description());
g_object_set(pipeline->pipeline,"video-sink",vsink,NULL);
g_object_set(pipeline->pipeline,"flags",0x0003,NULL);
bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline->pipeline));
gst_bus_add_signal_watch(bus);
gst_object_unref(bus);
g_signal_connect(bus,"message",G_CALLBACK(message_received),pipeline);
if (vsink) {
// Plant a pad probe to answer context queries
GstElement *sink;
sink = gst_bin_get_by_name(GST_BIN(vsink),"sink");
if (sink) {
GstPad *pad = gst_element_get_static_pad(sink,"sink");
if (pad) {
gulong id = gst_pad_add_probe(pad,GST_PAD_PROBE_TYPE_QUERY_DOWNSTREAM,pad_probe,pipeline,NULL);
gst_object_unref(pad);
}
gst_object_unref(sink);
}
}
并不能将相同的代码用于另一个管道(基于udpsrc而不是playbin3)。我在这种情况下使用的管道是:
udpsrc端口= 53512! application / x-rtp,media = video,clock-rate = 90000,encoding-name = H264,payload = 96! rtph264depay! encodebin3! glupload! glcolorconvert! video / x-raw(内存:GLMemory),format = RGBA,texture-target = 2D! fakesink sync = 0 qos = 1 name = sink
代码如下:
full_pipeline_description = g_strdup_printf("%s",pipeline_cmd);
gub_log_pipeline(pipeline,full_pipeline_description);
pipeline->pipeline = gst_parse_launch(full_pipeline_description,&err);
g_free(full_pipeline_description);
if (err) {
gub_log_pipeline(pipeline,err->message);
return;
}
vsink = gst_parse_bin_from_description(gub_get_video_branch_description(),NULL);
gub_log_pipeline(pipeline,gub_get_video_branch_description());
g_object_set(pipeline->pipeline,"sink",NULL);
g_object_set(pipeline->pipeline,NULL);
bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline->pipeline));
gst_bus_add_signal_watch(bus);
gst_object_unref(bus);
g_signal_connect(bus,pipeline);
// Plant a pad probe to answer context queries
GstElement *sink;
sink = gst_bin_get_by_name(GST_BIN(vsink),"sink");
if (sink) {
GstPad *pad = gst_element_get_static_pad(sink,"sink");
if (pad) {
gulong id = gst_pad_add_probe(pad,NULL);
gst_object_unref(pad);
}
gst_object_unref(sink);
}
基本上,在这种情况下,我只看到空白窗口(具有不同的颜色)。执行方面的唯一区别是,使用 playbin3 时会调用 pad_probe ,而使用udpsrc时不会调用。这是我看到的添加日志的唯一区别。我想了解为什么在使用udpsrc时,如果我错过了某些东西或使用了错误的东西,却没有调用此回调。
使用gstreamer-1.14.4和1.16.2版本都遇到相同的问题。任何提示都值得欢迎。
解决方法
g_object_set(pipeline->pipeline,"sink",vsink,NULL);
实际上什么也没做; GstPipeline没有“接收器”属性(不同于播放器)。通常,它会吐出一个日志警告,准确地说明这一点。
要将接收器添加到管道中,您需要像在GStreamer应用程序中正常进行的那样进行操作:找到需要连接的源焊盘,或者等到正确的源焊盘链接以显示为“添加了键盘”的信号(例如,发生在decodebin
中)。
最后,在进行了一些调查之后,基于此线程Gstreamer devel,我找到了问题的根本原因。基本上,我怀疑使用 udpsrc 时不会调用pad探测的回调,它仅在使用 playbin3 时才起作用。结果,没有提供图形上下文并且视频不能正确再现。为了解决此问题,我必须添加逻辑以处理总线上的消息以正确回答GST_MESSAGE_NEED_CONTEXT请求。为此,首先必须连接一个回调来处理总线消息,如下所示:
g_signal_connect(总线,“消息”,G_CALLBACK(消息已接收),管道);
然后在 message_received 函数中,添加了以下代码。
static void message_received(GstBus *bus,GstMessage *message,GUBPipeline *pipeline) {
switch (GST_MESSAGE_TYPE(message)) {
...
case GST_MESSAGE_NEED_CONTEXT:
{
const gchar *context_type;
GstContext *context = NULL;
gst_message_parse_context_type (message,&context_type);
context = gub_provide_graphic_context(pipeline->graphic_context,context_type);
if (context)
{
gst_element_set_context (GST_ELEMENT (message->src),context);
gst_context_unref (context);
}
break;
}
...
}
通过这些修改,我现在可以正确接收和再现视频了。使用ffmpeg testsrc工具模拟RTP视频流:
ffmpeg -f lavfi -i testsrc -vf scale=1280:960 -vcodec libx264 -profile:v baseline -pix_fmt yuv420p -f rtp rtp://YOUR_IP:PORT