- Clean up unused voice engine DTMF code.

webrtc/media/engine/fakewebrtccall.cc

 /*
  *  Copyright (c) 2015 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 21 matching lines @@
 void FakeAudioSendStream::SetStats(
     const webrtc::AudioSendStream::Stats& stats) {
   stats_ = stats;
 }
 
 FakeAudioSendStream::TelephoneEvent
     FakeAudioSendStream::GetLatestTelephoneEvent() const {
   return latest_telephone_event_;
 }
 
-bool FakeAudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
-                                             uint32_t duration_ms) {
+bool FakeAudioSendStream::SendTelephoneEvent(int payload_type, int event,
+                                             int duration_ms) {
   latest_telephone_event_.payload_type = payload_type;
   latest_telephone_event_.event_code = event;
   latest_telephone_event_.duration_ms = duration_ms;
   return true;
 }
 
 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const {
   return stats_;
 }
 
@@ skipping 362 matching lines @@
 }
 
 void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
   network_state_ = state;
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
 }
 }  // namespace cricket