Remove usage of VoEAudioProcessing from WVoE/MC.

webrtc/media/engine/webrtcvoiceengine.h

 /*
  *  Copyright (c) 2004 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.
  */
 
@@ skipping 12 matching lines @@
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/call.h"
 #include "webrtc/config.h"
 #include "webrtc/media/base/rtputils.h"
 #include "webrtc/media/engine/webrtccommon.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/pc/channel.h"
 
+namespace webrtc {
+namespace voe {
+class TransmitMixer;
+}  // namespace voe
+}  // namespace webrtc
+
 namespace cricket {
 
 class AudioDeviceModule;
 class AudioMixer;
 class AudioSource;
 class VoEWrapper;
 class WebRtcVoiceMediaChannel;
 
 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
 // It uses the WebRtc VoiceEngine library for audio handling.
@@ skipping 25 matching lines @@
   const std::vector<AudioCodec>& send_codecs() const;
   const std::vector<AudioCodec>& recv_codecs() const;
   RtpCapabilities GetCapabilities() const;
 
   // For tracking WebRtc channels. Needed because we have to pause them
   // all when switching devices.
   // May only be called by WebRtcVoiceMediaChannel.
   void RegisterChannel(WebRtcVoiceMediaChannel* channel);
   void UnregisterChannel(WebRtcVoiceMediaChannel* channel);
 
-  // Called by WebRtcVoiceMediaChannel to set a gain offset from
-  // the default AGC target level.
-  bool AdjustAgcLevel(int delta);
-
   VoEWrapper* voe() { return voe_wrapper_.get(); }
   int GetLastEngineError();
 
   // Starts AEC dump using an existing file. A maximum file size in bytes can be
   // specified. When the maximum file size is reached, logging is stopped and
   // the file is closed. If max_size_bytes is set to <= 0, no limit will be
   // used.
   bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
 
   // Stops AEC dump.
@@ skipping 10 matching lines @@
   bool ApplyOptions(const AudioOptions& options);
   void SetDefaultDevices();
 
   // webrtc::TraceCallback:
   void Print(webrtc::TraceLevel level, const char* trace, int length) override;
 
   void StartAecDump(const std::string& filename);
   int CreateVoEChannel();
   webrtc::AudioDeviceModule* adm();
   webrtc::AudioProcessing* apm();
+  webrtc::voe::TransmitMixer* transmit_mixer();
 
   AudioCodecs CollectRecvCodecs() const;
 
   rtc::ThreadChecker signal_thread_checker_;
   rtc::ThreadChecker worker_thread_checker_;
 
   // The audio device manager.
   rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_;
   rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_;
   // Reference to the APM, owned by VoE.
   webrtc::AudioProcessing* apm_ = nullptr;
+  // Reference to the TransmitMixer, owned by VoE.
+  webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr;
   // The primary instance of WebRtc VoiceEngine.
   std::unique_ptr<VoEWrapper> voe_wrapper_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::vector<AudioCodec> send_codecs_;
   std::vector<AudioCodec> recv_codecs_;
   std::vector<WebRtcVoiceMediaChannel*> channels_;
   webrtc::VoEBase::ChannelConfig channel_config_;
   bool is_dumping_aec_ = false;
 
   webrtc::AgcConfig default_agc_config_;
@@ skipping 143 matching lines @@
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_