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.
  */
 
 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_
 
 #include <memory>
 
 #include "webrtc/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/file_player.h"
 #include "webrtc/modules/utility/include/file_recorder.h"
-#include "webrtc/voice_engine/dtmf_inband.h"
-#include "webrtc/voice_engine/dtmf_inband_queue.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 #include "webrtc/voice_engine/level_indicator.h"
 #include "webrtc/voice_engine/network_predictor.h"
 #include "webrtc/voice_engine/shared_data.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace rtc {
 
 class TimestampWrapAroundHandler;
@@ skipping 251 matching lines @@
   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
   int SendTelephoneEventOutband(int event, int duration_ms);
-  int SendTelephoneEventInband(unsigned char eventCode,
-                               int lengthMs,
-                               int attenuationDb,
-                               bool playDtmfEvent);
   int SetSendTelephoneEventPayloadType(unsigned char type);
-  int GetSendTelephoneEventPayloadType(unsigned char& 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 137 matching lines @@
   bool ReceivePacket(const uint8_t* packet,
                      size_t packet_length,
                      const RTPHeader& header,
                      bool in_order);
   bool HandleRtxPacket(const uint8_t* packet,
                        size_t packet_length,
                        const RTPHeader& header);
   bool IsPacketInOrder(const RTPHeader& header) const;
   bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
   int ResendPackets(const uint16_t* sequence_numbers, int length);
-  int InsertInbandDtmfTone();
   int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
   int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
   void UpdatePlayoutTimestamp(bool rtcp);
   void UpdatePacketDelay(uint32_t timestamp, uint16_t sequenceNumber);
   void RegisterReceiveCodecsToRTPModule();
 
   int SetRedPayloadType(int red_payload_type);
   int SetSendRtpHeaderExtension(bool enable,
                                 RTPExtensionType type,
                                 unsigned char id);
@@ skipping 25 matching lines @@
   AudioFrame _audioFrame;
   // Downsamples to the codec rate if necessary.
   PushResampler<int16_t> input_resampler_;
   FilePlayer* _inputFilePlayerPtr;
   FilePlayer* _outputFilePlayerPtr;
   FileRecorder* _outputFileRecorderPtr;
   int _inputFilePlayerId;
   int _outputFilePlayerId;
   int _outputFileRecorderId;
   bool _outputFileRecording;
-  DtmfInbandQueue _inbandDtmfQueue;
-  DtmfInband _inbandDtmfGenerator;
   bool _outputExternalMedia;
   VoEMediaProcess* _inputExternalMediaCallbackPtr;
   VoEMediaProcess* _outputExternalMediaCallbackPtr;
   uint32_t _timeStamp;
-  uint8_t _sendTelephoneEventPayloadType;

[tlegrand-webrtc, 2016/03/11 13:49:17]

I wonder if this should remain? It's used for the out of band DTMF tones sent on
a separate payload type, right?

[the sun, 2016/03/11 13:56:24]

Correct, but it is only used here to support retrieving the payload type again
(which isn't supported through the new API). The payload type is propagated down
to the RtpRtcpModule where it is also stored for stamping outgoing packets. See
SetSendTelephoneEventPayloadType().

[tlegrand-webrtc, 2016/03/14 08:24:50]

Acknowledged.

 
   RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_);
 
   // Timestamp of the audio pulled from NetEq.
   uint32_t jitter_buffer_playout_timestamp_;
   uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_);
   uint32_t playout_timestamp_rtcp_;
   uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_);
   uint32_t _numberOfDiscardedPackets;
   uint16_t send_sequence_number_;
@@ skipping 25 matching lines @@
   // VoEBase
   bool _externalMixing;
   bool _mixFileWithMicrophone;
   // VoEVolumeControl
   bool _mute;
   float _panLeft;
   float _panRight;
   float _outputGain;
   // VoEDtmf
   bool _playOutbandDtmfEvent;
-  bool _playInbandDtmfEvent;
   // VoeRTP_RTCP
   uint32_t _lastLocalTimeStamp;
   int8_t _lastPayloadType;
   bool _includeAudioLevelIndication;
   // VoENetwork
   AudioFrame::SpeechType _outputSpeechType;
   // VoEVideoSync
   rtc::CriticalSection video_sync_lock_;
   uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_);
   uint32_t _previousTimestamp;
@@ skipping 14 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_