Remove the VoiceEngineObserver callback interface.

voice_engine/channel.h

 /*
  *  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 37 matching lines @@
 class RateLimiter;
 class ReceiveStatistics;
 class RemoteNtpTimeEstimator;
 class RtcEventLog;
 class RTPPayloadRegistry;
 class RTPReceiverAudio;
 class RtpPacketReceived;
 class RtpRtcp;
 class RtpTransportControllerSendInterface;
 class TelephoneEventHandler;
-class VoiceEngineObserver;
 
 struct SenderInfo;
 
 struct CallStatistics {
   unsigned short fractionLost;
   unsigned int cumulativeLost;
   unsigned int extendedMax;
   unsigned int jitterSamples;
   int64_t rttMs;
   size_t bytesSent;
@@ skipping 85 matching lines @@
                                uint32_t instanceId,
                                const VoEBase::ChannelConfig& config);
   Channel(int32_t channelId,
           uint32_t instanceId,
           const VoEBase::ChannelConfig& config);
   int32_t Init();
   void Terminate();
   int32_t SetEngineInformation(Statistics& engineStatistics,
                                ProcessThread& moduleProcessThread,
                                AudioDeviceModule& audioDeviceModule,
-                               VoiceEngineObserver* voiceEngineObserver,
                                rtc::CriticalSection* callbackCritSect,
                                rtc::TaskQueue* encoder_queue);
 
   void SetSink(std::unique_ptr<AudioSinkInterface> sink);
 
   // TODO(ossu): Don't use! It's only here to confirm that the decoder factory
   // passed into AudioReceiveStream is the same as the one set when creating the
   // ADM. Once Channel creation is moved into Audio{Send,Receive}Stream this can
   // go.
   const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const;
 
   void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
 
   // Send using this encoder, with this payload type.
   bool SetEncoder(int payload_type, std::unique_ptr<AudioEncoder> encoder);
   void ModifyEncoder(
       rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier);
 
   // API methods
 
   // VoEBase
   int32_t StartPlayout();
   int32_t StopPlayout();
   int32_t StartSend();
   void StopSend();
-  int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
-  int32_t DeRegisterVoiceEngineObserver();
 
   // Codecs
   int32_t GetSendCodec(CodecInst& codec);
   int32_t GetRecCodec(CodecInst& codec);
   int32_t SetSendCodec(const CodecInst& codec);
   void SetBitRate(int bitrate_bps, int64_t probing_interval_ms);
   bool EnableAudioNetworkAdaptor(const std::string& config_string);
   void DisableAudioNetworkAdaptor();
   void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                    int max_frame_length_ms);
@@ skipping 218 matching lines @@
   // The rtp timestamp of the first played out audio frame.
   int64_t capture_start_rtp_time_stamp_;
   // The capture ntp time (in local timebase) of the first played out audio
   // frame.
   int64_t capture_start_ntp_time_ms_ RTC_GUARDED_BY(ts_stats_lock_);
 
   // uses
   Statistics* _engineStatisticsPtr;
   ProcessThread* _moduleProcessThreadPtr;
   AudioDeviceModule* _audioDeviceModulePtr;
-  VoiceEngineObserver* _voiceEngineObserverPtr;  // owned by base
   rtc::CriticalSection* _callbackCritSectPtr;    // owned by base
   Transport* _transportPtr;  // WebRtc socket or external transport
   RmsLevel rms_level_ RTC_ACCESS_ON(encoder_queue_);
   bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_);
   bool previous_frame_muted_ RTC_ACCESS_ON(encoder_queue_);
   float _outputGain RTC_GUARDED_BY(volume_settings_critsect_);
   // VoeRTP_RTCP
   // TODO(henrika): can today be accessed on the main thread and on the
   // task queue; hence potential race.
   bool _includeAudioLevelIndication;
@@ skipping 29 matching lines @@
 
   bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_lock_) = false;
 
   rtc::TaskQueue* encoder_queue_ = nullptr;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // VOICE_ENGINE_CHANNEL_H_