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_; |