| Index: webrtc/media/base/fakemediaengine.h
|
| diff --git a/webrtc/media/base/fakemediaengine.h b/webrtc/media/base/fakemediaengine.h
|
| index 2b774b2bb98ecad54a32ead95f44da6308173cf5..d9a79ccca299f9c500bc7f13f043de3481a7ad90 100644
|
| --- a/webrtc/media/base/fakemediaengine.h
|
| +++ b/webrtc/media/base/fakemediaengine.h
|
| @@ -304,7 +304,7 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
|
| };
|
| explicit FakeVoiceMediaChannel(FakeVoiceEngine* engine,
|
| const AudioOptions& options)
|
| - : engine_(engine), time_since_last_typing_(-1), max_bps_(-1) {
|
| + : engine_(engine), max_bps_(-1) {
|
| output_scalings_[0] = 1.0; // For default channel.
|
| SetOptions(options);
|
| }
|
| @@ -366,11 +366,6 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
|
|
|
| virtual bool GetActiveStreams(AudioInfo::StreamList* streams) { return true; }
|
| virtual int GetOutputLevel() { return 0; }
|
| - void set_time_since_last_typing(int ms) { time_since_last_typing_ = ms; }
|
| - virtual int GetTimeSinceLastTyping() { return time_since_last_typing_; }
|
| - virtual void SetTypingDetectionParameters(
|
| - int time_window, int cost_per_typing, int reporting_threshold,
|
| - int penalty_decay, int type_event_delay) {}
|
|
|
| virtual bool CanInsertDtmf() {
|
| for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin();
|
| @@ -488,7 +483,6 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
|
| std::vector<AudioCodec> send_codecs_;
|
| std::map<uint32_t, double> output_scalings_;
|
| std::vector<DtmfInfo> dtmf_info_queue_;
|
| - int time_since_last_typing_;
|
| AudioOptions options_;
|
| std::map<uint32_t, VoiceChannelAudioSink*> local_sinks_;
|
| std::unique_ptr<webrtc::AudioSinkInterface> sink_;
|
|
|