When an audio track is disabled, still pass the data to webrtc for audio processing.

content/renderer/media/webrtc_local_audio_track.cc

Index: content/renderer/media/webrtc_local_audio_track.cc
diff --git a/content/renderer/media/webrtc_local_audio_track.cc b/content/renderer/media/webrtc_local_audio_track.cc
index 37846335e8531270c5c6c66658c400137b82bcf4..0cac639eb7dc96ad10d8717dfc2a47af61b33a71 100644
--- a/content/renderer/media/webrtc_local_audio_track.cc
+++ b/content/renderer/media/webrtc_local_audio_track.cc
@@ -154,14 +154,10 @@ void WebRtcLocalAudioTrack::Capture(media::AudioBus* audio_source,
   int number_of_frames = 0;
   SinkList sinks;
   scoped_refptr<ConfiguredBuffer> current_buffer;
+  bool disabled = false;
   {
     base::AutoLock auto_lock(lock_);
-    // When the track is disabled, we simply return here.
-    // TODO(xians): Figure out if we should feed zero to sinks instead, in
-    // order to inject VAD data in such case.
-    if (!enabled())
-      return;
-
+    disabled = !enabled();
     capturer = capturer_;
     voe_channels = voe_channels_;
     current_buffer = buffer_;
@@ -177,7 +173,7 @@ void WebRtcLocalAudioTrack::Capture(media::AudioBus* audio_source,
   // well as there is no correct delay value.
   bool need_audio_processing = need_audio_processing_ ?
       need_audio_processing_ : (audio_delay_milliseconds != 0);
-  int current_volume = volume;
+  int current_volume = disabled ? 0 : volume;

[no longer working on chromium, 2013/09/27 20:22:52]

Note that you are fooling the AGC, please test this CL in your local machine and
make sure AGC won't try to set the volume to a value closed to 0 when changing
the track from disable to enable state. 

[ajm, 2013/09/27 22:54:40]

This doesn't look good. You shouldn't be messing around with the volume you
provide, even if disabled.

   while (current_buffer->Consume()) {
     // Feed the data to the sinks.
     for (SinkList::const_iterator it = sinks.begin(); it != sinks.end(); ++it) {
@@ -190,7 +186,7 @@ void WebRtcLocalAudioTrack::Capture(media::AudioBus* audio_source,
                                           current_volume,
                                           need_audio_processing,
                                           key_pressed);
-      if (new_volume != 0 && capturer.get()) {
+      if (!disabled && new_volume != 0 && capturer.get()) {
         // Feed the new volume to WebRtc while changing the volume on the
         // browser.
         capturer->SetVolume(new_volume);