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webrtcdsp: Synchronize with delays

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Until now, we were synchronizing both DSP and Probe adapter by
waiting and clipping the probe adapter data. This increases the CPU
usage, can cause copies if the audio is not 10ms aligned and the worst
is that it prevents the processing from compensating for inaccurate
latency. This is also a step forward toward supporting playback
filters.
This commit is contained in:
Nicolas Dufresne 2016-06-22 21:54:13 -04:00
parent fb8662eb5c
commit 86aa3b5f9c
1 changed files with 5 additions and 21 deletions
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26
ext/webrtcdsp/gstwebrtcdsp.cpp
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@ -176,6 +176,7 @@ struct _GstWebrtcDsp
GstClockTime timestamp;
GstAdapter *adapter;
webrtc::AudioProcessing * apm;
gint delay_ms;
/* Protected by the object lock */
gchar *probe_name;
@ -288,28 +289,11 @@ gst_webrtc_dsp_sync_reverse_stream (GstWebrtcDsp * self,
probe_timestamp += gst_util_uint64_scale_int (distance / probe->info.bpf,
GST_SECOND, probe->info.rate);
probe_timestamp += probe->latency;
diff = GST_CLOCK_DIFF (probe_timestamp, self->timestamp);
if (diff < 0) {
GST_TRACE_OBJECT (self,
"Echo cancellation will start in in %" GST_TIME_FORMAT,
GST_TIME_ARGS (-diff));
return FALSE;
}
distance = gst_util_uint64_scale_int ((guint64) diff,
probe->info.rate * probe->info.bpf, GST_SECOND);
if (gst_adapter_available (probe->adapter) < distance) {
GST_TRACE_OBJECT (self, "Not enough data to synchronize for now.");
return FALSE;
}
gst_adapter_flush (probe->adapter, (gsize) distance);
probe->synchronized = TRUE;
self->delay_ms = (probe->latency - diff) / GST_MSECOND;
GST_DEBUG_OBJECT (probe, "Echo Probe is now synchronized");
probe->synchronized = TRUE;
return TRUE;
}
@ -387,8 +371,7 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self)
memcpy (frame.data_, info.data, self->period_size);
/* We synchronize in GStreamer */
apm->set_stream_delay_ms (5);
apm->set_stream_delay_ms (self->delay_ms);
if ((err = apm->ProcessStream (&frame)) < 0) {
GST_WARNING_OBJECT (self, "Failed to filter the audio: %s.",
@ -505,6 +488,7 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
GST_OBJECT_LOCK (self);
gst_adapter_clear (self->adapter);
self->delay_ms = 0;
self->info = *info;
apm = self->apm;