Detect AudioInputStream muting and propagate to MediaStreamAudioSource.

content/renderer/media/webrtc/processed_local_audio_source.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_
 
 #include "base/atomicops.h"
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
@@ skipping 77 matching lines @@
   void EnsureSourceIsStopped() final;
 
   // AudioCapturerSource::CaptureCallback implementation.
   // Called on the AudioCapturerSource audio thread.
   void OnCaptureStarted() override;
   void Capture(const media::AudioBus* audio_source,
                int audio_delay_milliseconds,
                double volume,
                bool key_pressed) override;
   void OnCaptureError(const std::string& message) override;
+  void OnCaptureMuted(bool is_muted) override;
 
  private:
   // Helper function to get the source buffer size based on whether audio
   // processing will take place.
   int GetBufferSize(int sample_rate) const;
 
   // The RenderFrame that will consume the audio data. Used when creating
   // AudioCapturerSources.
   const int consumer_render_frame_id_;
 
@@ skipping 27 matching lines @@
   MediaStreamAudioLevelCalculator level_calculator_;
 
   bool allow_invalid_render_frame_id_for_testing_;
 
   DISALLOW_COPY_AND_ASSIGN(ProcessedLocalAudioSource);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_