Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(811)

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

Issue 1327933002: Full impl of NnChannel::SetSendParameters and NnChannel::SetRecvParameters (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase 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
Index: talk/media/webrtc/webrtcvoiceengine.h
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 951c878161585b2ef4738457929fc7aa0ba38ea1..2332b4c36db1fb44935a99db70f233841bd8dccf 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -280,7 +280,8 @@ class WebRtcVoiceEngine
class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
public webrtc::Transport {
public:
- explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine);
+ WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine,
+ const AudioOptions& options);
~WebRtcVoiceMediaChannel() override;
int voe_channel() const { return voe_channel_; }
@@ -289,13 +290,6 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
bool SetSendParameters(const AudioSendParameters& params) override;
bool SetRecvParameters(const AudioRecvParameters& params) override;
- bool SetOptions(const AudioOptions& options) override;
- bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) override;
- bool SetSendCodecs(const std::vector<AudioCodec>& codecs) override;
- bool SetRecvRtpHeaderExtensions(
- const std::vector<RtpHeaderExtension>& extensions) override;
- bool SetSendRtpHeaderExtensions(
- const std::vector<RtpHeaderExtension>& extensions) override;
bool SetPlayout(bool playout) override;
bool PausePlayout();
bool ResumePlayout();
@@ -329,7 +323,6 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
void OnRtcpReceived(rtc::Buffer* packet,
const rtc::PacketTime& packet_time) override;
void OnReadyToSend(bool ready) override {}
- bool SetMaxSendBandwidth(int bps) override;
bool GetStats(VoiceMediaInfo* info) override;
// Gets last reported error from WebRtc voice engine. This should be only
// called in response a failure.
@@ -359,8 +352,17 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
void SetCall(webrtc::Call* call);
private:
+ bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
+ bool SetSendRtpHeaderExtensions(
+ const std::vector<RtpHeaderExtension>& extensions);
+ bool SetOptions(const AudioOptions& options);
+ bool SetMaxSendBandwidth(int bps);
+ bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
+ bool SetRecvRtpHeaderExtensions(
+ const std::vector<RtpHeaderExtension>& extensions);
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);

Powered by Google App Engine
This is Rietveld 408576698