- Clean up unused voice engine DTMF code.

webrtc/audio/audio_send_stream.h

 /*
  *  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 22 matching lines @@
                   CongestionController* congestion_controller);
   ~AudioSendStream() override;
 
   // webrtc::SendStream implementation.
   void Start() override;
   void Stop() override;
   void SignalNetworkState(NetworkState state) override;
   bool DeliverRtcp(const uint8_t* packet, size_t length) override;
 
   // webrtc::AudioSendStream implementation.
-  bool SendTelephoneEvent(int payload_type, uint8_t event,
-                          uint32_t duration_ms) override;
+  bool SendTelephoneEvent(int payload_type, int event,
+                          int duration_ms) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
   const webrtc::AudioSendStream::Config& config() const;
 
  private:
   VoiceEngine* voice_engine() const;
 
   rtc::ThreadChecker thread_checker_;
   const webrtc::AudioSendStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_