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

content/renderer/media/webrtc_local_audio_renderer.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_renderer.h"
 
 #include "base/debug/trace_event.h"
 #include "base/logging.h"
 #include "base/message_loop/message_loop_proxy.h"
 #include "base/metrics/histogram.h"
@@ skipping 51 matching lines @@
 int WebRtcLocalAudioRenderer::CaptureData(const std::vector<int>& channels,
                                           const int16* audio_data,
                                           int sample_rate,
                                           int number_of_channels,
                                           int number_of_frames,
                                           int audio_delay_milliseconds,
                                           int current_volume,
                                           bool need_audio_processing,
                                           bool key_pressed) {
   TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::CaptureData");
+  if (!current_volume)

[no longer working on chromium, 2013/09/27 12:31:52]

remove

+    return 0;
+
   base::AutoLock auto_lock(thread_lock_);
   if (!playing_ || !volume_)
     return 0;
 
   // Push captured audio to FIFO so it can be read by a local sink.
   if (loopback_fifo_) {
     if (loopback_fifo_->frames() + number_of_frames <=
         loopback_fifo_->max_frames()) {
       scoped_ptr<media::AudioBus> audio_source = media::AudioBus::Create(
           number_of_channels, number_of_frames);
@@ skipping 199 matching lines @@
     return;
 
   sink_->Start();
   sink_started_ = true;
 
   UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates",
                             kSinkStarted, kSinkStatesMax);
 }
 
 }  // namespace content