- Clean up unused voice engine DTMF code.

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 249 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
-  int SendTelephoneEventOutband(unsigned char eventCode,
-                                int lengthMs,
-                                int attenuationDb,
-                                bool playDtmfEvent);
-  int SendTelephoneEventInband(unsigned char eventCode,
-                               int lengthMs,
-                               int attenuationDb,
-                               bool playDtmfEvent);
-  int SetSendTelephoneEventPayloadType(unsigned char type);
-  int GetSendTelephoneEventPayloadType(unsigned char& type);
+  // DTMF
+  int SendTelephoneEventOutband(int event, int duration_ms);
+  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 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;
 
   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 23 matching lines @@
   int32_t _oldVadDecision;
   int32_t _sendFrameType;  // Send data is voice, 1-voice, 0-otherwise
   // 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_