| Index: voice_engine/channel.h
|
| diff --git a/voice_engine/channel.h b/voice_engine/channel.h
|
| index 8c163657ec90cf04ea4ed56cfffad056df0e1673..696dd70a4141b96af940f55cfee57154cba506eb 100644
|
| --- a/voice_engine/channel.h
|
| +++ b/voice_engine/channel.h
|
| @@ -90,7 +90,6 @@ namespace voe {
|
| class RtcEventLogProxy;
|
| class RtcpRttStatsProxy;
|
| class RtpPacketSenderProxy;
|
| -class Statistics;
|
| class TransportFeedbackProxy;
|
| class TransportSequenceNumberProxy;
|
| class VoERtcpObserver;
|
| @@ -157,10 +156,8 @@ class Channel
|
| 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);
|
| @@ -197,8 +194,7 @@ class Channel
|
| 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);
|
| @@ -286,10 +282,6 @@ class Channel
|
| 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(); }
|
|
|
| @@ -395,7 +387,6 @@ class Channel
|
| 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_);
|
| @@ -420,10 +411,8 @@ class Channel
|
| 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_);
|
|
|