Remove the unused and untested functions from VoERTP_RTCP.

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 52 matching lines @@
 struct ReportBlock;
 struct SenderInfo;
 
 namespace voe {
 
 class OutputMixer;
 class RtcEventLogProxy;
 class RtcpRttStatsProxy;
 class RtpPacketSenderProxy;
 class Statistics;
-class StatisticsProxy;
 class TransportFeedbackProxy;
 class TransmitMixer;
 class TransportSequenceNumberProxy;
 class VoERtcpObserver;
 
 // Helper class to simplify locking scheme for members that are accessed from
 // multiple threads.
 // Example: a member can be set on thread T1 and read by an internal audio
 // thread T2. Accessing the member via this class ensures that we are
 // safe and also avoid TSan v2 warnings.
@@ skipping 93 matching lines @@
   // go.
   const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const;
 
   // API methods
 
   // VoEBase
   int32_t StartPlayout();
   int32_t StopPlayout();
   int32_t StartSend();
   int32_t StopSend();
-  void ResetDiscardedPacketCount();
   int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
   int32_t DeRegisterVoiceEngineObserver();
 
   // VoECodec
   int32_t GetSendCodec(CodecInst& codec);
   int32_t GetRecCodec(CodecInst& codec);
   int32_t SetSendCodec(const CodecInst& codec);
   void SetBitRate(int bitrate_bps, int64_t probing_interval_ms);
   int32_t SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX);
   int32_t GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX);
@@ skipping 94 matching lines @@
 
   // VoEAudioProcessingImpl
   int VoiceActivityIndicator(int& activity);
 
   // VoERTP_RTCP
   int SetLocalSSRC(unsigned int ssrc);
   int GetLocalSSRC(unsigned int& ssrc);
   int GetRemoteSSRC(unsigned int& ssrc);
   int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
   int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
-  int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
-  int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
   void EnableSendTransportSequenceNumber(int id);
   void EnableReceiveTransportSequenceNumber(int id);
 
   void RegisterSenderCongestionControlObjects(
       RtpPacketSender* rtp_packet_sender,
       TransportFeedbackObserver* transport_feedback_observer,
       PacketRouter* packet_router,
       RtcpBandwidthObserver* bandwidth_observer);
   void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
   void ResetCongestionControlObjects();
 
   void SetRTCPStatus(bool enable);
   int GetRTCPStatus(bool& enabled);
   int SetRTCP_CNAME(const char cName[256]);
   int GetRemoteRTCP_CNAME(char cName[256]);
-  int GetRemoteRTCPData(unsigned int& NTPHigh,
-                        unsigned int& NTPLow,
-                        unsigned int& timestamp,
-                        unsigned int& playoutTimestamp,
-                        unsigned int* jitter,
-                        unsigned short* fractionLost);
   int SendApplicationDefinedRTCPPacket(unsigned char subType,
                                        unsigned int name,
                                        const char* data,
                                        unsigned short dataLengthInBytes);
-  int GetRTPStatistics(unsigned int& averageJitterMs,
-                       unsigned int& maxJitterMs,
-                       unsigned int& discardedPackets);
   int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks);
   int GetRTPStatistics(CallStatistics& stats);
   int SetCodecFECStatus(bool enable);
   bool GetCodecFECStatus();
   void SetNACKStatus(bool enable, int maxNumberOfPackets);
 
   // From AudioPacketizationCallback in the ACM
   int32_t SendData(FrameType frameType,
                    uint8_t payloadType,
                    uint32_t timeStamp,
@@ skipping 114 matching lines @@
   int32_t _channelId;
 
   ChannelState channel_state_;
 
   std::unique_ptr<voe::RtcEventLogProxy> event_log_proxy_;
   std::unique_ptr<voe::RtcpRttStatsProxy> rtcp_rtt_stats_proxy_;
 
   std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
   std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
   std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
-  std::unique_ptr<StatisticsProxy> statistics_proxy_;
   std::unique_ptr<RtpReceiver> rtp_receiver_;
   TelephoneEventHandler* telephone_event_handler_;
   std::unique_ptr<RtpRtcp> _rtpRtcpModule;
   std::unique_ptr<AudioCodingModule> audio_coding_;
   acm2::CodecManager codec_manager_;
   acm2::RentACodec rent_a_codec_;
   std::unique_ptr<AudioSinkInterface> audio_sink_;
   AudioLevel _outputAudioLevel;
   bool _externalTransport;
   AudioFrame _audioFrame;
   // Downsamples to the codec rate if necessary.
   PushResampler<int16_t> input_resampler_;
   std::unique_ptr<FilePlayer> input_file_player_;
   std::unique_ptr<FilePlayer> output_file_player_;
   std::unique_ptr<FileRecorder> output_file_recorder_;
   int _inputFilePlayerId;
   int _outputFilePlayerId;
   int _outputFileRecorderId;
   bool _outputFileRecording;
   bool _outputExternalMedia;
   VoEMediaProcess* _inputExternalMediaCallbackPtr;
   VoEMediaProcess* _outputExternalMediaCallbackPtr;
   uint32_t _timeStamp;
 
   RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_);
 
   // Timestamp of the audio pulled from NetEq.
   rtc::Optional<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_;
   uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes];
 
   rtc::CriticalSection ts_stats_lock_;
 
   std::unique_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
   // The rtp timestamp of the first played out audio frame.
   int64_t capture_start_rtp_time_stamp_;
   // The capture ntp time (in local timebase) of the first played out audio
   // frame.
@@ skipping 45 matching lines @@
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_