Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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 21 matching lines @@
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/output_mixer.h"
 #include "webrtc/voice_engine/statistics.h"
-#include "webrtc/voice_engine/transmit_mixer.h"
 #include "webrtc/voice_engine/utility.h"
 
 namespace webrtc {
 namespace voe {
 
 namespace {
 
 constexpr int64_t kMaxRetransmissionWindowMs = 1000;
 constexpr int64_t kMinRetransmissionWindowMs = 30;
 
@@ skipping 848 matching lines @@
                       // random offset
       ntp_estimator_(Clock::GetRealTimeClock()),
       playout_timestamp_rtp_(0),
       playout_delay_ms_(0),
       send_sequence_number_(0),
       rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
       capture_start_rtp_time_stamp_(-1),
       capture_start_ntp_time_ms_(-1),
       _engineStatisticsPtr(NULL),
       _outputMixerPtr(NULL),
-      _transmitMixerPtr(NULL),
       _moduleProcessThreadPtr(NULL),
       _audioDeviceModulePtr(NULL),
       _voiceEngineObserverPtr(NULL),
       _callbackCritSectPtr(NULL),
       _transportPtr(NULL),
       _sendFrameType(0),
       _mixFileWithMicrophone(false),
       input_mute_(false),
       previous_frame_muted_(false),
       _panLeft(1.0f),
@@ skipping 190 matching lines @@
                      codec.pltype, codec.plfreq);
       }
     }
   }
 
   return 0;
 }
 
 int32_t Channel::SetEngineInformation(Statistics& engineStatistics,
                                       OutputMixer& outputMixer,
-                                      voe::TransmitMixer& transmitMixer,
                                       ProcessThread& moduleProcessThread,
                                       AudioDeviceModule& audioDeviceModule,
                                       VoiceEngineObserver* voiceEngineObserver,
                                       rtc::CriticalSection* callbackCritSect) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetEngineInformation()");
   _engineStatisticsPtr = &engineStatistics;
   _outputMixerPtr = &outputMixer;
-  _transmitMixerPtr = &transmitMixer,
   _moduleProcessThreadPtr = &moduleProcessThread;
   _audioDeviceModulePtr = &audioDeviceModule;
   _voiceEngineObserverPtr = voiceEngineObserver;
   _callbackCritSectPtr = callbackCritSect;
   return 0;
 }
 
 int32_t Channel::UpdateLocalTimeStamp() {
   _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
   return 0;
@@ skipping 1925 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc