Wire up AGC to the MediaStreamAudioProcessor

content/renderer/media/webrtc_audio_capturer.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_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
@@ skipping 421 matching lines @@
   TrackList::ItemList tracks_to_notify_format;
   int current_volume = 0;
   base::TimeDelta audio_delay;
   scoped_refptr<MediaStreamAudioProcessor> audio_processor;
   bool need_audio_processing = true;
   {
     base::AutoLock auto_lock(lock_);
     if (!running_)
       return;
 
     // Map internal volume range of [0.0, 1.0] into [0, 255] used by the

[ajm, 2014/01/22 21:50:55]

FYI, I'd like to change the AudioProcessing interface to take a float.

[no longer working on chromium, 2014/01/23 12:46:08]

I will be happy to do the chromium change once APM accepts float.

     // webrtc::VoiceEngine. webrtc::VoiceEngine will handle the case when the
     // volume is higher than 255.
     volume_ = static_cast<int>((volume * MaxVolume()) + 0.5);
     current_volume = volume_;
     audio_delay = base::TimeDelta::FromMilliseconds(audio_delay_milliseconds);
     audio_delay_ = audio_delay;
     key_pressed_ = key_pressed;
     tracks = tracks_.Items();
     tracks_.RetrieveAndClearTags(&tracks_to_notify_format);
     audio_processor = audio_processor_;
@@ skipping 18 matching lines @@
        it != tracks_to_notify_format.end(); ++it) {
     (*it)->OnSetFormat(output_params);
   }
 
   // Push the data to the processor for processing.
   audio_processor->PushCaptureData(audio_source);
 
   // Process and consume the data in the processor until there is not enough
   // data in the processor.
   int16* output = NULL;
+  int new_volume = 0;
   while (audio_processor->ProcessAndConsumeData(
-      audio_delay, current_volume, key_pressed, &output)) {
+      audio_delay, current_volume, key_pressed, &new_volume, &output)) {
     // Feed the post-processed data to the tracks.
     for (TrackList::ItemList::const_iterator it = tracks.begin();
          it != tracks.end(); ++it) {
       (*it)->Capture(output, audio_delay, current_volume, key_pressed,
                      need_audio_processing);
     }
-    // TODO(xians): Apply the new volume after AGC is working with the
-    // MediaStreamAudioProcessor.
+
+    // TODO(xians): Check with ajm@ if we should set curret_volume to

[tommi (sloooow) - chröme, 2014/01/21 15:06:20]

current_volume

[no longer working on chromium, 2014/01/23 12:46:08]

comment was removed.

+    // new_volume to avoid setting the volume twice.

[ajm, 2014/01/22 21:50:55]

Yes, you should update current_volume.

[no longer working on chromium, 2014/01/23 12:46:08]

Done.

+    if (new_volume)
+      SetVolume(new_volume);
   }
 }
 
 void WebRtcAudioCapturer::OnCaptureError() {
   NOTIMPLEMENTED();
 }
 
 media::AudioParameters WebRtcAudioCapturer::source_audio_parameters() const {
   base::AutoLock auto_lock(lock_);
   return audio_processor_ ?
@@ skipping 74 matching lines @@
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::AudioParameters params) {
   // Create a new audio stream as source which uses the new source.
   SetCapturerSource(source, params.channel_layout(),
                     static_cast<float>(params.sample_rate()),
                     params.effects(),
                     constraints_);
 }
 
 }  // namespace content