webrtc/api/peerconnection.h - Issue 2046173002: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.

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Unified Diff: webrtc/api/peerconnection.h

Issue 2046173002: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Comment formatting. Created 4 years, 6 months ago

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Index: webrtc/api/peerconnection.h

diff --git a/webrtc/api/peerconnection.h b/webrtc/api/peerconnection.h

index aba595c4bbf517d121e592f3b6b1dc5919f732a2..fc754b9c838761402f2b1629c5a09295419c5c00 100644

--- a/webrtc/api/peerconnection.h

+++ b/webrtc/api/peerconnection.h

@@ -174,7 +174,6 @@ class PeerConnection : public PeerConnectionInterface,

void CreateVideoReceiver(MediaStreamInterface* stream,

const std::string& track_id,

uint32_t ssrc);

- void StopReceivers(cricket::MediaType media_type);

void DestroyReceiver(const std::string& track_id);

void DestroyAudioSender(MediaStreamInterface* stream,

AudioTrackInterface* audio_track,

@@ -325,7 +324,9 @@ class PeerConnection : public PeerConnectionInterface,

void OnSctpDataChannelClosed(DataChannel* channel);

// Notifications from WebRtcSession relating to BaseChannels.

+ void OnVoiceChannelCreated();

void OnVoiceChannelDestroyed();

+ void OnVideoChannelCreated();

void OnVideoChannelDestroyed();

void OnDataChannelCreated();

void OnDataChannelDestroyed();

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