Remove the VoiceEngineObserver callback interface.

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 755 matching lines @@
       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),
       _moduleProcessThreadPtr(NULL),
       _audioDeviceModulePtr(NULL),
-      _voiceEngineObserverPtr(NULL),
       _callbackCritSectPtr(NULL),
       _transportPtr(NULL),
       input_mute_(false),
       previous_frame_muted_(false),
       _outputGain(1.0f),
       _includeAudioLevelIndication(false),
       transport_overhead_per_packet_(0),
       rtp_overhead_per_packet_(0),
       _outputSpeechType(AudioFrame::kNormalSpeech),
       rtcp_observer_(new VoERtcpObserver(this)),
@@ skipping 155 matching lines @@
   // De-register modules in process thread
   if (_moduleProcessThreadPtr)
     _moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get());
 
   // End of modules shutdown
 }
 
 int32_t Channel::SetEngineInformation(Statistics& engineStatistics,
                                       ProcessThread& moduleProcessThread,
                                       AudioDeviceModule& audioDeviceModule,
-                                      VoiceEngineObserver* voiceEngineObserver,
                                       rtc::CriticalSection* callbackCritSect,
                                       rtc::TaskQueue* encoder_queue) {
   RTC_DCHECK(encoder_queue);
   RTC_DCHECK(!encoder_queue_);
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetEngineInformation()");
   _engineStatisticsPtr = &engineStatistics;
   _moduleProcessThreadPtr = &moduleProcessThread;
   _audioDeviceModulePtr = &audioDeviceModule;
-  _voiceEngineObserverPtr = voiceEngineObserver;
   _callbackCritSectPtr = callbackCritSect;
   encoder_queue_ = encoder_queue;
   return 0;
 }
 
 void Channel::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   rtc::CritScope cs(&_callbackCritSect);
   audio_sink_ = std::move(sink);
 }
 
@@ skipping 144 matching lines @@
   audio_coding_->SetEncoder(std::move(encoder));
   codec_manager_.UnsetCodecInst();
   return true;
 }
 
 void Channel::ModifyEncoder(
     rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
   audio_coding_->ModifyEncoder(modifier);
 }
 
-int32_t Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) {
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::RegisterVoiceEngineObserver()");
-  rtc::CritScope cs(&_callbackCritSect);
-
-  if (_voiceEngineObserverPtr) {
-    _engineStatisticsPtr->SetLastError(
-        VE_INVALID_OPERATION, kTraceError,
-        "RegisterVoiceEngineObserver() observer already enabled");
-    return -1;
-  }
-  _voiceEngineObserverPtr = &observer;
-  return 0;
-}
-
-int32_t Channel::DeRegisterVoiceEngineObserver() {
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::DeRegisterVoiceEngineObserver()");
-  rtc::CritScope cs(&_callbackCritSect);
-
-  if (!_voiceEngineObserverPtr) {
-    _engineStatisticsPtr->SetLastError(
-        VE_INVALID_OPERATION, kTraceWarning,
-        "DeRegisterVoiceEngineObserver() observer already disabled");
-    return 0;
-  }
-  _voiceEngineObserverPtr = NULL;
-  return 0;
-}
-
 int32_t Channel::GetSendCodec(CodecInst& codec) {
   if (cached_send_codec_) {
     codec = *cached_send_codec_;
     return 0;
   } else {
     const CodecInst* send_codec = codec_manager_.GetCodecInst();
     if (send_codec) {
       codec = *send_codec;
       return 0;
     }
@@ skipping 862 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