| Index: webrtc/voice_engine/channel_proxy.h
|
| diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
|
| index 829b094cc686686d3ffc85dda52e9fa7611c04e0..50c1e5f75b0be96d1fd01d2c36e3420dd176f2ab 100644
|
| --- a/webrtc/voice_engine/channel_proxy.h
|
| +++ b/webrtc/voice_engine/channel_proxy.h
|
| @@ -58,6 +58,8 @@ class ChannelProxy {
|
|
|
| virtual bool SetEncoder(int payload_type,
|
| std::unique_ptr<AudioEncoder> encoder);
|
| + virtual void ModifyEncoder(
|
| + rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier);
|
|
|
| virtual void SetRTCPStatus(bool enable);
|
| virtual void SetLocalSSRC(uint32_t ssrc);
|
| @@ -98,10 +100,6 @@ class ChannelProxy {
|
| 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);
|
| @@ -115,13 +113,6 @@ class ChannelProxy {
|
| virtual void SetMinimumPlayoutDelay(int delay_ms);
|
| virtual void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
|
| virtual bool GetRecCodec(CodecInst* codec_inst) const;
|
| - virtual bool GetSendCodec(CodecInst* codec_inst) const;
|
| - virtual bool SetVADStatus(bool enable);
|
| - virtual bool SetCodecFECStatus(bool enable);
|
| - virtual bool SetOpusDtx(bool enable);
|
| - virtual bool SetOpusMaxPlaybackRate(int frequency_hz);
|
| - virtual bool SetSendCodec(const CodecInst& codec_inst);
|
| - virtual bool SetSendCNPayloadType(int type, PayloadFrequencies frequency);
|
| virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
|
| virtual void OnRecoverableUplinkPacketLossRate(
|
| float recoverable_packet_loss_rate);
|
|
|