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authorPhilippe Normand <philn@igalia.com>2021-04-18 13:49:59 +0100
committerGStreamer Marge Bot <gitlab-merge-bot@gstreamer-foundation.org>2021-04-19 15:51:32 +0000
commit8b1051cdea243e5d5079c4218067d9e7a585385b (patch)
tree99d56f74858e7ee64f1359f63cbb1885b3396d82 /ext/webrtcdsp
parentc2635c154de7bc88967c0155f322d471ed97938c (diff)
webrtcdsp: Propagate VAD to audio level meta
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>
Diffstat (limited to 'ext/webrtcdsp')
-rw-r--r--ext/webrtcdsp/gstwebrtcdsp.cpp17
1 files changed, 15 insertions, 2 deletions
diff --git a/ext/webrtcdsp/gstwebrtcdsp.cpp b/ext/webrtcdsp/gstwebrtcdsp.cpp
index 27b286fa5..7ee09488f 100644
--- a/ext/webrtcdsp/gstwebrtcdsp.cpp
+++ b/ext/webrtcdsp/gstwebrtcdsp.cpp
@@ -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);