Remove voe::Statistics.

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 72 matching lines @@
   uint32_t interarrival_jitter;
   uint32_t last_SR_timestamp;
   uint32_t delay_since_last_SR;
 };
 
 namespace voe {
 
 class RtcEventLogProxy;
 class RtcpRttStatsProxy;
 class RtpPacketSenderProxy;
-class Statistics;
 class TransportFeedbackProxy;
 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 {
@@ skipping 46 matching lines @@
   virtual ~Channel();
   static int32_t CreateChannel(Channel*& channel,
                                int32_t channelId,
                                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,
+  int32_t SetEngineInformation(ProcessThread& moduleProcessThread,
                                AudioDeviceModule& audioDeviceModule,
-                               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;
 
@@ skipping 16 matching lines @@
   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);
 
   // Network
-  int32_t RegisterExternalTransport(Transport* transport);
-  int32_t DeRegisterExternalTransport();
+  void RegisterTransport(Transport* transport);
   // TODO(nisse, solenberg): Delete when VoENetwork is deleted.
   int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
   void OnRtpPacket(const RtpPacketReceived& packet);
 
   // Muting, Volume and Level.
   void SetInputMute(bool enable);
   void SetChannelOutputVolumeScaling(float scaling);
   int GetSpeechOutputLevel() const;
   int GetSpeechOutputLevelFullRange() const;
   // See description of "totalAudioEnergy" in the WebRTC stats spec:
@@ skipping 67 matching lines @@
   AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
       int sample_rate_hz,
       AudioFrame* audio_frame);
 
   int PreferredSampleRate() const;
 
   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;
-  }
   RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); }
   int8_t OutputEnergyLevel() const { return _outputAudioLevel.Level(); }
 
   // ProcessAndEncodeAudio() creates an audio frame copy and posts a task
   // on the shared encoder task queue, wich in turn calls (on the queue)
   // ProcessAndEncodeAudioOnTaskQueue() where the actual processing of the
   // audio takes place. The processing mainly consists of encoding and preparing
   // the result for sending by adding it to a send queue.
   // The main reason for using a task queue here is to release the native,
   // OS-specific, audio capture thread as soon as possible to ensure that it
@@ skipping 85 matching lines @@
   std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
   std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
   std::unique_ptr<RtpReceiver> rtp_receiver_;
   TelephoneEventHandler* telephone_event_handler_;
   std::unique_ptr<RtpRtcp> _rtpRtcpModule;
   std::unique_ptr<AudioCodingModule> audio_coding_;
   acm2::CodecManager codec_manager_;
   acm2::RentACodec rent_a_codec_;
   std::unique_ptr<AudioSinkInterface> audio_sink_;
   AudioLevel _outputAudioLevel;
-  bool _externalTransport;
   // Downsamples to the codec rate if necessary.
   PushResampler<int16_t> input_resampler_;
   uint32_t _timeStamp RTC_ACCESS_ON(encoder_queue_);
 
   RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_);
 
   // Timestamp of the audio pulled from NetEq.
   rtc::Optional<uint32_t> jitter_buffer_playout_timestamp_;
 
   rtc::CriticalSection video_sync_lock_;
   uint32_t playout_timestamp_rtp_ RTC_GUARDED_BY(video_sync_lock_);
   uint32_t playout_delay_ms_ RTC_GUARDED_BY(video_sync_lock_);
   uint16_t send_sequence_number_;
 
   rtc::CriticalSection ts_stats_lock_;
 
   std::unique_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
   // 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;
-  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;
   size_t transport_overhead_per_packet_
@@ skipping 28 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_