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decklinkaudiosink: Drop late buffers
Asking decklink to render audio data seems to be based entirely on the sample counts which completely disregards the timestamps we pass to decklink. As a result, we need to explicitly check for late buffers and drop them ourselves.
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1 changed files with 28 additions and 7 deletions
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@ -609,7 +609,7 @@ gst_decklink_audio_sink_render (GstBaseSink * bsink, GstBuffer * buffer)
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GstClockTime buffered_time;
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guint32 written = 0;
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GstClock *clock;
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GstClockTime clock_ahead;
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GstClockTimeDiff clock_ahead;
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if (GST_BASE_SINK_CAST (self)->flushing) {
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flow_ret = GST_FLOW_FLUSHING;
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@ -664,14 +664,13 @@ gst_decklink_audio_sink_render (GstBaseSink * bsink, GstBuffer * buffer)
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else
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clock_now = 0;
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if (clock_now < running_time)
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clock_ahead = running_time - clock_now;
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clock_ahead = running_time - clock_now;
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}
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}
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GST_DEBUG_OBJECT (self,
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"Ahead %" GST_TIME_FORMAT " of the clock running time",
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GST_TIME_ARGS (clock_ahead));
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"Ahead %" GST_STIME_FORMAT " of the clock running time",
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GST_STIME_ARGS (clock_ahead));
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if (self->output->
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output->GetBufferedAudioSampleFrameCount (&buffered_samples) != S_OK)
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@ -683,13 +682,35 @@ gst_decklink_audio_sink_render (GstBaseSink * bsink, GstBuffer * buffer)
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GST_DEBUG_OBJECT (self,
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"Buffered %" GST_TIME_FORMAT " in the driver (%u samples)",
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GST_TIME_ARGS (buffered_time), buffered_samples);
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{
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GstClockTimeDiff buffered_ahead_of_clock_ahead = GST_CLOCK_DIFF (clock_ahead, buffered_time);
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GST_DEBUG_OBJECT (self, "driver is %" GST_STIME_FORMAT " ahead of the "
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"expected clock", GST_STIME_ARGS (buffered_ahead_of_clock_ahead));
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/* we don't want to store too much data in the driver as decklink
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* doesn't seem to actually use our provided timestamps to perform its
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* own synchronisation. It seems to count samples instead. */
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/* FIXME: do we need to split buffers? */
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if (buffered_ahead_of_clock_ahead > 0 &&
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buffered_ahead_of_clock_ahead > gst_base_sink_get_max_lateness (bsink)) {
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GST_DEBUG_OBJECT (self, "Dropping buffer that is %" GST_STIME_FORMAT
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" too late", GST_STIME_ARGS (buffered_ahead_of_clock_ahead));
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if (self->resampler)
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gst_audio_resampler_reset (self->resampler);
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flow_ret = GST_FLOW_OK;
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break;
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}
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}
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// We start waiting once we have more than buffer-time buffered
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if (buffered_time > self->buffer_time || clock_ahead > self->buffer_time) {
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if (((GstClockTime) clock_ahead) > self->buffer_time) {
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GstClockReturn clock_ret;
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GstClockTime wait_time = running_time;
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GST_DEBUG_OBJECT (self,
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"Buffered enough, wait for preroll or the clock or flushing");
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"Buffered enough, wait for preroll or the clock or flushing. "
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"Configured buffer time: %" GST_TIME_FORMAT, GST_TIME_ARGS (self->buffer_time));
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if (wait_time < self->buffer_time)
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wait_time = 0;
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