| Index: talk/media/webrtc/webrtcvoiceengine.h
|
| diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
|
| index 9c7f7bb525be79d0883b06c2750cc9f80a5431c7..57c9439efca75fa89603062ed8dea429e4cd51a1 100644
|
| --- a/talk/media/webrtc/webrtcvoiceengine.h
|
| +++ b/talk/media/webrtc/webrtcvoiceengine.h
|
| @@ -57,8 +57,10 @@ class WebRtcVoiceMediaChannel;
|
| // It uses the WebRtc VoiceEngine library for audio handling.
|
| class WebRtcVoiceEngine final : public webrtc::TraceCallback {
|
| friend class WebRtcVoiceMediaChannel;
|
| -
|
| public:
|
| + // Exposed for the WVoE/MC unit test.
|
| + static bool ToCodecInst(const AudioCodec& in, webrtc::CodecInst* out);
|
| +
|
| WebRtcVoiceEngine();
|
| // Dependency injection for testing.
|
| explicit WebRtcVoiceEngine(VoEWrapper* voe_wrapper);
|
| @@ -78,9 +80,6 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback {
|
| int GetInputLevel();
|
|
|
| const std::vector<AudioCodec>& codecs();
|
| - bool FindCodec(const AudioCodec& codec);
|
| - bool FindWebRtcCodec(const AudioCodec& codec, webrtc::CodecInst* gcodec);
|
| -
|
| const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
|
|
|
| // For tracking WebRtc channels. Needed because we have to pause them
|
| @@ -114,8 +113,6 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback {
|
|
|
| private:
|
| void Construct();
|
| - void ConstructCodecs();
|
| - bool GetVoeCodec(int index, webrtc::CodecInst* codec);
|
| bool InitInternal();
|
| // Every option that is "set" will be applied. Every option not "set" will be
|
| // ignored. This allows us to selectively turn on and off different options
|
| @@ -258,7 +255,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
|
| }
|
| bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
|
| bool SetSendBitrateInternal(int bps);
|
| - bool SetRecvCodecsInternal(const std::vector<AudioCodec>& new_codecs);
|
|
|
| rtc::ThreadChecker worker_thread_checker_;
|
|
|
|
|