Remove unused methods in WebRtcVoiceEngine and VoiceMediaChannel.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 2387 matching lines @@
 
 int WebRtcVoiceMediaChannel::GetOutputLevel() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   int highest = 0;
   for (const auto& ch : recv_streams_) {
     highest = std::max(GetOutputLevel(ch.second->channel()), highest);
   }
   return highest;
 }
 
-int WebRtcVoiceMediaChannel::GetTimeSinceLastTyping() {
-  int ret;
-  if (engine()->voe()->processing()->TimeSinceLastTyping(ret) == -1) {
-    // In case of error, log the info and continue
-    LOG_RTCERR0(TimeSinceLastTyping);
-    ret = -1;
-  } else {
-    ret *= 1000;  // We return ms, webrtc returns seconds.

[ossu, 2017/02/09 15:49:03]

This is the most confusing comment I've seen in a while. I thought this _was_
WebRTC.

-  }
-  return ret;
-}
-
-void WebRtcVoiceMediaChannel::SetTypingDetectionParameters(int time_window,
-    int cost_per_typing, int reporting_threshold, int penalty_decay,
-    int type_event_delay) {
-  if (engine()->voe()->processing()->SetTypingDetectionParameters(
-          time_window, cost_per_typing,
-          reporting_threshold, penalty_decay, type_event_delay) == -1) {
-    // In case of error, log the info and continue
-    LOG_RTCERR5(SetTypingDetectionParameters, time_window,
-                cost_per_typing, reporting_threshold, penalty_decay,
-                type_event_delay);
-  }
-}
-
 bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (ssrc == 0) {
     default_recv_volume_ = volume;
     if (default_recv_ssrc_ == -1) {
       return true;
     }
     ssrc = static_cast<uint32_t>(default_recv_ssrc_);
   }
   const auto it = recv_streams_.find(ssrc);
@@ skipping 282 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE