webrtc/media/base/mediachannel.h - Issue 1847353004: Allow applications to control audio send bitrate through RtpParameters.

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Unified Diff: webrtc/media/base/mediachannel.h

Issue 1847353004: Allow applications to control audio send bitrate through RtpParameters. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Redesigned to keep the logic inside WebRtcVoiceMediaChannel Created 4 years, 8 months ago

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Index: webrtc/media/base/mediachannel.h

diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h

index 424572de2f07b9491e49a6ca20d09c0f6a545c40..4166a60dd3171917773a4d1f88a3cebee178b4ee 100644

--- a/webrtc/media/base/mediachannel.h

+++ b/webrtc/media/base/mediachannel.h

@@ -904,6 +904,9 @@ class VoiceMediaChannel : public MediaChannel {

virtual ~VoiceMediaChannel() {}

virtual bool SetSendParameters(const AudioSendParameters& params) = 0;

virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0;

+ virtual webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const = 0;

+ virtual bool SetRtpParameters(uint32_t ssrc,

+ const webrtc::RtpParameters& parameters) = 0;

// Starts or stops playout of received audio.

virtual bool SetPlayout(bool playout) = 0;

// Starts or stops sending (and potentially capture) of local audio.

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