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audiomixer: If getting a timeout before having caps, just advance our position

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This can happen if this is a live pipeline and no source produced any buffer
and sent no caps until the an output buffer should've been produced according
to the latency.
This commit is contained in:
Sebastian Dröge 2014-12-23 12:24:48 +01:00
parent eefea80dae
commit cd256acf03
1 changed files with 30 additions and 8 deletions
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38
gst/audiomixer/gstaudiomixer.c
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@ -707,7 +707,11 @@ gst_audiomixer_sink_event (GstAggregator * agg, GstAggregatorPad * aggpad,
static void static void
gst_audiomixer_reset (GstAudioMixer * audiomixer) gst_audiomixer_reset (GstAudioMixer * audiomixer)
{ {
GstAggregator *agg = GST_AGGREGATOR (audiomixer);
audiomixer->offset = 0; audiomixer->offset = 0;
agg->segment.position = -1;
gst_caps_replace (&audiomixer->current_caps, NULL); gst_caps_replace (&audiomixer->current_caps, NULL);
gst_buffer_replace (&audiomixer->current_buffer, NULL); gst_buffer_replace (&audiomixer->current_buffer, NULL);
} }
@ -1375,9 +1379,32 @@ gst_audiomixer_aggregate (GstAggregator * agg, gboolean timeout)
audiomixer = GST_AUDIO_MIXER (agg); audiomixer = GST_AUDIO_MIXER (agg);
/* this is fatal */ /* Update position from the segment start/stop if needed */
if (G_UNLIKELY (audiomixer->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) if (agg->segment.position == -1) {
goto not_negotiated; if (agg->segment.rate > 0.0)
agg->segment.position = agg->segment.start;
else
agg->segment.position = agg->segment.stop;
}
if (G_UNLIKELY (audiomixer->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) {
if (timeout) {
GST_DEBUG_OBJECT (audiomixer,
"Got timeout before receiving any caps, don't output anything");
/* Advance position */
if (agg->segment.rate > 0.0)
agg->segment.position += audiomixer->output_buffer_duration;
else if (agg->segment.position > audiomixer->output_buffer_duration)
agg->segment.position -= audiomixer->output_buffer_duration;
else
agg->segment.position = 0;
return GST_FLOW_OK;
} else {
goto not_negotiated;
}
}
blocksize = blocksize =
gst_util_uint64_scale (audiomixer->output_buffer_duration, gst_util_uint64_scale (audiomixer->output_buffer_duration,
@ -1387,11 +1414,6 @@ gst_audiomixer_aggregate (GstAggregator * agg, gboolean timeout)
if (audiomixer->send_caps) { if (audiomixer->send_caps) {
gst_aggregator_set_src_caps (agg, audiomixer->current_caps); gst_aggregator_set_src_caps (agg, audiomixer->current_caps);
if (agg->segment.rate > 0.0)
agg->segment.position = agg->segment.start;
else
agg->segment.position = agg->segment.stop;
audiomixer->offset = gst_util_uint64_scale (agg->segment.position, audiomixer->offset = gst_util_uint64_scale (agg->segment.position,
GST_AUDIO_INFO_RATE (&audiomixer->info), GST_SECOND); GST_AUDIO_INFO_RATE (&audiomixer->info), GST_SECOND);