Calculate the signal level on the media stream local audio track

content/renderer/media/webrtc_local_audio_track.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc_local_audio_track.h"
 
 #include "content/public/renderer/media_stream_audio_sink.h"
+#include "content/renderer/media/media_stream_audio_level_calculator.h"
 #include "content/renderer/media/media_stream_audio_sink_owner.h"
 #include "content/renderer/media/media_stream_audio_track_sink.h"
 #include "content/renderer/media/peer_connection_audio_sink_owner.h"
 #include "content/renderer/media/webaudio_capturer_source.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 
 namespace content {
 
 WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
@@ skipping 21 matching lines @@
   // Users might not call Stop() on the track.
   Stop();
 }
 
 void WebRtcLocalAudioTrack::Capture(const int16* audio_data,
                                     base::TimeDelta delay,
                                     int volume,
                                     bool key_pressed,
                                     bool need_audio_processing) {
   DCHECK(capture_thread_checker_.CalledOnValidThread());
+
+  // Calculate the signal level regardless if the track is disabled or enabled.
+  int signal_level = level_calculator_->Calculate(
+      audio_data, audio_parameters_.channels(),
+      audio_parameters_.frames_per_buffer());
+  adapter_->SetSignalLevel(signal_level);
+
   scoped_refptr<WebRtcAudioCapturer> capturer;
   SinkList::ItemList sinks;
   SinkList::ItemList sinks_to_notify_format;
   {
     base::AutoLock auto_lock(lock_);
     capturer = capturer_;
     sinks = sinks_.Items();
     sinks_.RetrieveAndClearTags(&sinks_to_notify_format);
   }
 
@@ skipping 32 matching lines @@
 
 void WebRtcLocalAudioTrack::OnSetFormat(
     const media::AudioParameters& params) {
   DVLOG(1) << "WebRtcLocalAudioTrack::OnSetFormat()";
   // If the source is restarted, we might have changed to another capture
   // thread.
   capture_thread_checker_.DetachFromThread();
   DCHECK(capture_thread_checker_.CalledOnValidThread());
 
   audio_parameters_ = params;
+  level_calculator_.reset(new MediaStreamAudioLevelCalculator());
 
   base::AutoLock auto_lock(lock_);
   // Remember to notify all sinks of the new format.
   sinks_.TagAll();
 }
 
 void WebRtcLocalAudioTrack::AddSink(MediaStreamAudioSink* sink) {
   DCHECK(main_render_thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcLocalAudioTrack::AddSink()";
   base::AutoLock auto_lock(lock_);
@@ skipping 115 matching lines @@
 
   for (SinkList::ItemList::const_iterator it = sinks.begin();
        it != sinks.end();
        ++it){
     (*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateEnded);
     (*it)->Reset();
   }
 }
 
 }  // namespace content