Added audio mixer and removed audio device module in AudioState::Config.

webrtc/api/call/audio_state.h

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #ifndef WEBRTC_API_CALL_AUDIO_STATE_H_
 #define WEBRTC_API_CALL_AUDIO_STATE_H_
 
+#include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 
 namespace webrtc {
 
 class AudioDeviceModule;

[the sun, 2016/11/09 10:51:44]

Please remove - not needed anymore.

[aleloi, 2016/11/09 13:10:15]

Done in https://codereview.webrtc.org/2491483002/!

 class VoiceEngine;
 
 // WORK IN PROGRESS
 // This class is under development and is not yet intended for for use outside
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
 
 // AudioState holds the state which must be shared between multiple instances of
 // webrtc::Call for audio processing purposes.
 class AudioState : public rtc::RefCountInterface {
  public:
   struct Config {
     // VoiceEngine used for audio streams and audio/video synchronization.
     // AudioState will tickle the VoE refcount to keep it alive for as long as
     // the AudioState itself.
     VoiceEngine* voice_engine = nullptr;
 
-    // The AudioDeviceModule associated with the Calls.
-    AudioDeviceModule* audio_device_module = nullptr;
+    // The audio mixer connected to active receive streams. One per
+    // AudioState.
+    rtc::scoped_refptr<AudioMixer> audio_mixer;
   };
 
   // TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.
   static rtc::scoped_refptr<AudioState> Create(
       const AudioState::Config& config);
 
   virtual ~AudioState() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_CALL_AUDIO_STATE_H_