- Clean up unused voice engine DTMF code.

webrtc/voice_engine/channel_proxy.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 130 matching lines @@
 uint32_t ChannelProxy::GetDelayEstimate() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return channel()->GetDelayEstimate();
 }
 
 bool ChannelProxy::SetSendTelephoneEventPayloadType(int payload_type) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return channel()->SetSendTelephoneEventPayloadType(payload_type) == 0;
 }
 
-bool ChannelProxy::SendTelephoneEventOutband(uint8_t event,
-                                             uint32_t duration_ms) {
+bool ChannelProxy::SendTelephoneEventOutband(int event, int duration_ms) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  return
-      channel()->SendTelephoneEventOutband(event, duration_ms, 10, false) == 0;
+  return channel()->SendTelephoneEventOutband(event, duration_ms) == 0;
 }
 
 void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   channel()->SetSink(std::move(sink));
 }
 
 Channel* ChannelProxy::channel() const {
   RTC_DCHECK(channel_owner_.channel());
   return channel_owner_.channel();
 }
 
 }  // namespace voe
 }  // namespace webrtc