void GstEnginePipeline::BufferingMessageReceived(GstMessage* msg) {
// Only handle buffering messages from the queue2 element in audiobin - not
// the one that's created automatically by uridecodebin.
if (GST_ELEMENT(GST_MESSAGE_SRC(msg)) != queue_) {
return;
}
int percent = 0;
gst_message_parse_buffering(msg, &percent);
const GstState current_state = state();
if (percent == 0 && current_state == GST_STATE_PLAYING && !buffering_) {
buffering_ = true;
emit BufferingStarted();
SetState(GST_STATE_PAUSED);
} else if (percent == 100 && buffering_) {
buffering_ = false;
emit BufferingFinished();
SetState(GST_STATE_PLAYING);
} else if (buffering_) {
emit BufferingProgress(percent);
}
}
void GstEnginePipeline::BufferingMessageReceived(GstMessage* msg) {
// Only handle buffering messages from the queue2 element in audiobin - not
// the one that's created automatically by uridecodebin.
if (GST_ELEMENT(GST_MESSAGE_SRC(msg)) != queue_) {
return;
}
// If we are loading new next track, we don't have to pause the playback.
// The buffering is for the next track and not the current one.
if (emit_track_ended_on_stream_start_) {
qLog(Debug) << "Buffering next track";
return;
}
int percent = 0;
gst_message_parse_buffering(msg, &percent);
const GstState current_state = state();
if (percent == 0 && current_state == GST_STATE_PLAYING && !buffering_) {
buffering_ = true;
emit BufferingStarted();
SetState(GST_STATE_PAUSED);
} else if (percent == 100 && buffering_) {
buffering_ = false;
emit BufferingFinished();
SetState(GST_STATE_PLAYING);
} else if (buffering_) {
emit BufferingProgress(percent);
}
}