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rtpjitterbuffer: adjust latency in buffer mode
When we are in buffer mode, adjust the buffering low/high thresholds based on the total configured latency. If we don't and there is a huge queue or element with a big latency downstream we might drain the complete queue immediately and start buffering again.
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1 changed files with 20 additions and 0 deletions
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@ -933,6 +933,7 @@ gst_rtp_jitter_buffer_change_state (GstElement * element,
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priv->blocked = TRUE;
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/* reset skew detection initialy */
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rtp_jitter_buffer_reset_skew (priv->jbuf);
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rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
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JBUF_UNLOCK (priv);
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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@ -987,6 +988,25 @@ gst_rtp_jitter_buffer_src_event (GstPad * pad, GstEvent * event)
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GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_LATENCY:
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{
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GstClockTime latency;
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gst_event_parse_latency (event, &latency);
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JBUF_LOCK (priv);
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/* adjust the overall buffer delay to the total pipeline latency in
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* buffering mode because if downstream consumes too fast (because of
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* large latency or queues, we would start rebuffering again. */
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if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
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RTP_JITTER_BUFFER_MODE_BUFFER) {
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rtp_jitter_buffer_set_delay (priv->jbuf, latency);
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}
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JBUF_UNLOCK (priv);
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ret = gst_pad_push_event (priv->sinkpad, event);
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break;
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}
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default:
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ret = gst_pad_push_event (priv->sinkpad, event);
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break;
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