Relanding https://codereview.webrtc.org/1715883002/ in pieces.

webrtc/voice_engine/channel.h

 /*
  *  Copyright (c) 2012 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 276 matching lines @@
   uint32_t GetDelayEstimate() const;
   int LeastRequiredDelayMs() const;
   int SetMinimumPlayoutDelay(int delayMs);
   int GetPlayoutTimestamp(unsigned int& timestamp);
   int SetInitTimestamp(unsigned int timestamp);
   int SetInitSequenceNumber(short sequenceNumber);
 
   // VoEVideoSyncExtended
   int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
 
-  // VoEDtmf
+  // DTMF
   int SendTelephoneEventOutband(int event, int duration_ms);
-  int SetSendTelephoneEventPayloadType(unsigned char type);
+  int SetSendTelephoneEventPayloadType(int payload_type);
 
   // VoEAudioProcessingImpl
   int UpdateRxVadDetection(AudioFrame& audioFrame);
   int RegisterRxVadObserver(VoERxVadCallback& observer);
   int DeRegisterRxVadObserver();
   int VoiceActivityIndicator(int& activity);
 #ifdef WEBRTC_VOICE_ENGINE_AGC
   int SetRxAgcStatus(bool enable, AgcModes mode);
   int GetRxAgcStatus(bool& enabled, AgcModes& mode);
   int SetRxAgcConfig(AgcConfig config);
@@ skipping 265 matching lines @@
   PacketRouter* packet_router_ = nullptr;
   std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
   std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
   std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_