Index: webrtc/voice_engine/channel_proxy.h |
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h |
index d19c0092a715f791eb7af3aac5b33812b86b4d89..be48abdc2a57bb6e429770117c74be7144d85b30 100644 |
--- a/webrtc/voice_engine/channel_proxy.h |
+++ b/webrtc/voice_engine/channel_proxy.h |
@@ -65,34 +65,31 @@ class ChannelProxy { |
virtual void RegisterReceiverCongestionControlObjects( |
PacketRouter* packet_router); |
virtual void ResetCongestionControlObjects(); |
- |
virtual CallStatistics GetRTCPStatistics() const; |
virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const; |
virtual NetworkStatistics GetNetworkStatistics() const; |
virtual AudioDecodingCallStats GetDecodingCallStatistics() const; |
virtual int32_t GetSpeechOutputLevelFullRange() const; |
virtual uint32_t GetDelayEstimate() const; |
- |
virtual bool SetSendTelephoneEventPayloadType(int payload_type); |
virtual bool SendTelephoneEventOutband(int event, int duration_ms); |
virtual void SetBitrate(int bitrate_bps); |
virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink); |
virtual void SetInputMute(bool muted); |
- |
virtual void RegisterExternalTransport(Transport* transport); |
virtual void DeRegisterExternalTransport(); |
virtual bool ReceivedRTPPacket(const uint8_t* packet, |
size_t length, |
const PacketTime& packet_time); |
virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length); |
- |
virtual const rtc::scoped_refptr<AudioDecoderFactory>& |
GetAudioDecoderFactory() const; |
- |
virtual void SetChannelOutputVolumeScaling(float scaling); |
- |
virtual void SetRtcEventLog(RtcEventLog* event_log); |
- |
+ virtual void EnableAudioNetworkAdaptor(const std::string& config_string); |
+ virtual void DisableAudioNetworkAdaptor(); |
+ virtual void SetReceiverFrameLengthRange(int min_frame_length_ms, |
+ int max_frame_length_ms); |
virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo( |
int sample_rate_hz, |
AudioFrame* audio_frame); |