webrtc/media/engine/webrtcvoiceengine.h - Issue 1841083008: WebRtcVoiceEngine refactoring - moving codec configuration code into Send/RecvStreams

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Unified Diff: webrtc/media/engine/webrtcvoiceengine.h

Issue 1841083008: WebRtcVoiceEngine refactoring - moving codec configuration code into Send/RecvStreams (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Created 4 years, 9 months ago

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Index: webrtc/media/engine/webrtcvoiceengine.h

diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h

index c9798cbc9567a5f32193a38661c9f80df340e390..e84a7c248b96b8af5bedc0564116b79cccba58d9 100644

--- a/webrtc/media/engine/webrtcvoiceengine.h

+++ b/webrtc/media/engine/webrtcvoiceengine.h

@@ -215,9 +215,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,

bool SetOptions(const AudioOptions& options);

bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);

bool SetSendCodecs(const std::vector<AudioCodec>& codecs);

- bool SetSendCodecs(int channel);

- void SetNack(int channel, bool nack_enabled);

- bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);

bool SetMaxSendBandwidth(int bps);

bool SetLocalSource(uint32_t ssrc, AudioSource* source);

bool MuteStream(uint32_t ssrc, bool mute);

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