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webrtcdsp: Propagate VAD to audio level meta

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Whenever the voice activity changed on the stream, update or create an
AudioLevelMeta and associate it to the corresponding buffer.

Fixes #1073

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2170>
This commit is contained in:
Philippe Normand 2021-04-18 13:49:59 +01:00 committed by GStreamer Marge Bot
parent c2635c154d
commit 8b1051cdea
1 changed files with 15 additions and 2 deletions
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17
ext/webrtcdsp/gstwebrtcdsp.cpp
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@ -442,12 +442,24 @@ done:
}
static void
gst_webrtc_vad_post_message (GstWebrtcDsp *self, GstClockTime timestamp,
gst_webrtc_vad_post_activity (GstWebrtcDsp *self, GstBuffer *buffer,
gboolean stream_has_voice)
{
GstClockTime timestamp = GST_BUFFER_PTS (buffer);
GstBaseTransform *trans = GST_BASE_TRANSFORM_CAST (self);
GstStructure *s;
GstClockTime stream_time;
GstAudioLevelMeta *meta;
guint8 level;
level = self->apm->level_estimator ()->RMS ();
meta = gst_buffer_get_audio_level_meta (buffer);
if (meta) {
meta->voice_activity = stream_has_voice;
meta->level = level;
} else {
gst_buffer_add_audio_level_meta (buffer, level, stream_has_voice);
}
stream_time = gst_segment_to_stream_time (&trans->segment, GST_FORMAT_TIME,
timestamp);
@ -502,7 +514,7 @@ gst_webrtc_dsp_process_stream (GstWebrtcDsp * self,
gboolean stream_has_voice = apm->voice_detection ()->stream_has_voice ();
if (stream_has_voice != self->stream_has_voice)
gst_webrtc_vad_post_message (self, GST_BUFFER_PTS (buffer), stream_has_voice);
gst_webrtc_vad_post_activity (self, buffer, stream_has_voice);
self->stream_has_voice = stream_has_voice;
}
@ -716,6 +728,7 @@ gst_webrtc_dsp_setup (GstAudioFilter * filter, const GstAudioInfo * info)
apm->voice_detection ()->set_likelihood (self->voice_detection_likelihood);
apm->voice_detection ()->set_frame_size_ms (
self->voice_detection_frame_size_ms);
apm->level_estimator ()->Enable (true);
}
GST_OBJECT_UNLOCK (self);