Remove VoEExternalMedia interface.

webrtc/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 ProcessThread;
 class RateLimiter;
 class ReceiveStatistics;
 class RemoteNtpTimeEstimator;
 class RtcEventLog;
 class RTPPayloadRegistry;
 class RtpReceiver;
 class RTPReceiverAudio;
 class RtpRtcp;
 class TelephoneEventHandler;
-class VoEMediaProcess;
 class VoERTPObserver;
 class VoiceEngineObserver;
 
 struct CallStatistics;
 struct ReportBlock;
 struct SenderInfo;
 
 namespace voe {
 
 class OutputMixer;
 class RtcEventLogProxy;
 class RtcpRttStatsProxy;
 class RtpPacketSenderProxy;
 class Statistics;
 class StatisticsProxy;
 class TransportFeedbackProxy;
 class TransmitMixer;
 class TransportSequenceNumberProxy;
 class VoERtcpObserver;
 
 // Helper class to simplify locking scheme for members that are accessed from
 // multiple threads.
 // Example: a member can be set on thread T1 and read by an internal audio
 // thread T2. Accessing the member via this class ensures that we are
 // safe and also avoid TSan v2 warnings.
 class ChannelState {
  public:
   struct State {
-    bool input_external_media = false;
     bool output_file_playing = false;
     bool input_file_playing = false;
     bool playing = false;
     bool sending = false;
   };
 
   ChannelState() {}
   virtual ~ChannelState() {}
 
   void Reset() {
     rtc::CritScope lock(&lock_);
     state_ = State();
   }
 
   State Get() const {
     rtc::CritScope lock(&lock_);
     return state_;
   }
 
-  void SetInputExternalMedia(bool enable) {
-    rtc::CritScope lock(&lock_);
-    state_.input_external_media = enable;
-  }
-
   void SetOutputFilePlaying(bool enable) {
     rtc::CritScope lock(&lock_);
     state_.output_file_playing = enable;
   }
 
   void SetInputFilePlaying(bool enable) {
     rtc::CritScope lock(&lock_);
     state_.input_file_playing = enable;
   }
 
@@ skipping 124 matching lines @@
                                    int stopPosition,
                                    const CodecInst* codecInst);
   int StopPlayingFileAsMicrophone();
   int IsPlayingFileAsMicrophone() const;
   int StartRecordingPlayout(const char* fileName, const CodecInst* codecInst);
   int StartRecordingPlayout(OutStream* stream, const CodecInst* codecInst);
   int StopRecordingPlayout();
 
   void SetMixWithMicStatus(bool mix);
 
-  // VoEExternalMediaProcessing
-  int RegisterExternalMediaProcessing(ProcessingTypes type,
-                                      VoEMediaProcess& processObject);
-  int DeRegisterExternalMediaProcessing(ProcessingTypes type);
-  int SetExternalMixing(bool enabled);
-
   // VoEVolumeControl
   int GetSpeechOutputLevel(uint32_t& level) const;
   int GetSpeechOutputLevelFullRange(uint32_t& level) const;
   int SetInputMute(bool enable);
   bool InputMute() const;
   int SetOutputVolumePan(float left, float right);
   int GetOutputVolumePan(float& left, float& right) const;
   int SetChannelOutputVolumeScaling(float scaling);
   int GetChannelOutputVolumeScaling(float& scaling) const;
 
@@ skipping 113 matching lines @@
   void RecordFileEnded(int32_t id) override;
 
   uint32_t InstanceId() const { return _instanceId; }
   int32_t ChannelId() const { return _channelId; }
   bool Playing() const { return channel_state_.Get().playing; }
   bool Sending() const { return channel_state_.Get().sending; }
   bool ExternalTransport() const {
     rtc::CritScope cs(&_callbackCritSect);
     return _externalTransport;
   }
-  bool ExternalMixing() const { return _externalMixing; }
   RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); }
   int8_t OutputEnergyLevel() const { return _outputAudioLevel.Level(); }
   uint32_t Demultiplex(const AudioFrame& audioFrame);
   // Demultiplex the data to the channel's |_audioFrame|. The difference
   // between this method and the overloaded method above is that |audio_data|
   // does not go through transmit_mixer and APM.
   void Demultiplex(const int16_t* audio_data,
                    int sample_rate,
                    size_t number_of_frames,
                    size_t number_of_channels);
@@ skipping 70 matching lines @@
   AudioFrame _audioFrame;
   // Downsamples to the codec rate if necessary.
   PushResampler<int16_t> input_resampler_;
   std::unique_ptr<FilePlayer> input_file_player_;
   std::unique_ptr<FilePlayer> output_file_player_;
   std::unique_ptr<FileRecorder> output_file_recorder_;
   int _inputFilePlayerId;
   int _outputFilePlayerId;
   int _outputFileRecorderId;
   bool _outputFileRecording;
-  bool _outputExternalMedia;
-  VoEMediaProcess* _inputExternalMediaCallbackPtr;
-  VoEMediaProcess* _outputExternalMediaCallbackPtr;
   uint32_t _timeStamp;
 
   RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_);
 
   // Timestamp of the audio pulled from NetEq.
   rtc::Optional<uint32_t> jitter_buffer_playout_timestamp_;
   uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_);
   uint32_t playout_timestamp_rtcp_;
   uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_);
   uint32_t _numberOfDiscardedPackets;
@@ skipping 14 matching lines @@
   OutputMixer* _outputMixerPtr;
   TransmitMixer* _transmitMixerPtr;
   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_;
   int32_t _sendFrameType;  // Send data is voice, 1-voice, 0-otherwise
   // VoEBase
-  bool _externalMixing;
   bool _mixFileWithMicrophone;
   // VoEVolumeControl
   bool input_mute_ GUARDED_BY(volume_settings_critsect_);
   bool previous_frame_muted_;  // Only accessed from PrepareEncodeAndSend().
   float _panLeft GUARDED_BY(volume_settings_critsect_);
   float _panRight GUARDED_BY(volume_settings_critsect_);
   float _outputGain GUARDED_BY(volume_settings_critsect_);
   // VoeRTP_RTCP
   uint32_t _lastLocalTimeStamp;
   int8_t _lastPayloadType;
@@ skipping 20 matching lines @@
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_