Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1015)

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

Issue 1311533009: - Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel) (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@lj_remove_typingmonitor_files
Patch Set: typo in comment Created 5 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « talk/media/webrtc/webrtcvideoengine2.cc ('k') | talk/media/webrtc/webrtcvoiceengine.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: talk/media/webrtc/webrtcvoiceengine.h
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 4b35ba7db256a444c199f5a6ac133b4d8ad0765a..951c878161585b2ef4738457929fc7aa0ba38ea1 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -302,12 +302,13 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
bool SetSend(SendFlags send) override;
bool PauseSend();
bool ResumeSend();
+ bool SetAudioSend(uint32 ssrc, bool mute, const AudioOptions* options,
+ AudioRenderer* renderer) override;
bool AddSendStream(const StreamParams& sp) override;
bool RemoveSendStream(uint32 ssrc) override;
bool AddRecvStream(const StreamParams& sp) override;
bool RemoveRecvStream(uint32 ssrc) override;
bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) override;
- bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) override;
bool GetActiveStreams(AudioInfo::StreamList* actives) override;
int GetOutputLevel() override;
int GetTimeSinceLastTyping() override;
@@ -328,7 +329,6 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
void OnRtcpReceived(rtc::Buffer* packet,
const rtc::PacketTime& packet_time) override;
void OnReadyToSend(bool ready) override {}
- bool MuteStream(uint32 ssrc, bool on) override;
bool SetMaxSendBandwidth(int bps) override;
bool GetStats(VoiceMediaInfo* info) override;
// Gets last reported error from WebRtc voice engine. This should be only
@@ -359,6 +359,8 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
void SetCall(webrtc::Call* call);
private:
+ bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer);
+ bool MuteStream(uint32 ssrc, bool mute);
WebRtcVoiceEngine* engine() { return engine_; }
int GetLastEngineError() { return engine()->GetLastEngineError(); }
int GetOutputLevel(int channel);
« no previous file with comments | « talk/media/webrtc/webrtcvideoengine2.cc ('k') | talk/media/webrtc/webrtcvoiceengine.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698