Remove VoERTP_RTCP

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 38 matching lines @@
 class RateLimiter;
 class ReceiveStatistics;
 class RemoteNtpTimeEstimator;
 class RtcEventLog;
 class RTPPayloadRegistry;
 class RTPReceiverAudio;
 class RtpPacketReceived;
 class RtpRtcp;
 class RtpTransportControllerSendInterface;
 class TelephoneEventHandler;
-class VoERTPObserver;
 class VoiceEngineObserver;
 
-struct CallStatistics;
-struct ReportBlock;
 struct SenderInfo;
 
+struct CallStatistics {
+  unsigned short fractionLost;
+  unsigned int cumulativeLost;
+  unsigned int extendedMax;
+  unsigned int jitterSamples;
+  int64_t rttMs;
+  size_t bytesSent;
+  int packetsSent;
+  size_t bytesReceived;
+  int packetsReceived;
+  // The capture ntp time (in local timebase) of the first played out audio
+  // frame.
+  int64_t capture_start_ntp_time_ms_;
+};
+
+// See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details.
+struct ReportBlock {
+  uint32_t sender_SSRC;  // SSRC of sender
+  uint32_t source_SSRC;
+  uint8_t fraction_lost;
+  uint32_t cumulative_num_packets_lost;
+  uint32_t extended_highest_sequence_number;
+  uint32_t interarrival_jitter;
+  uint32_t last_SR_timestamp;
+  uint32_t delay_since_last_SR;
+};
+
 namespace voe {
 
 class OutputMixer;
 class RtcEventLogProxy;
 class RtcpRttStatsProxy;
 class RtpPacketSenderProxy;
 class Statistics;
 class TransportFeedbackProxy;
 class TransportSequenceNumberProxy;
 class VoERtcpObserver;
@@ skipping 132 matching lines @@
   // Audio+Video Sync.
   uint32_t GetDelayEstimate() const;
   int SetMinimumPlayoutDelay(int delayMs);
   int GetPlayoutTimestamp(unsigned int& timestamp);
   int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
 
   // DTMF.
   int SendTelephoneEventOutband(int event, int duration_ms);
   int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency);
 
-  // VoERTP_RTCP
+  // RTP+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);
   void EnableSendTransportSequenceNumber(int id);
   void EnableReceiveTransportSequenceNumber(int id);
 
   void RegisterSenderCongestionControlObjects(
       RtpTransportControllerSendInterface* transport,
       RtcpBandwidthObserver* bandwidth_observer);
   void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
   void ResetSenderCongestionControlObjects();
   void ResetReceiverCongestionControlObjects();
   void SetRTCPStatus(bool enable);
-  int GetRTCPStatus(bool& enabled);
   int SetRTCP_CNAME(const char cName[256]);
-  int GetRemoteRTCP_CNAME(char cName[256]);
-  int SendApplicationDefinedRTCPPacket(unsigned char subType,
-                                       unsigned int name,
-                                       const char* data,
-                                       unsigned short dataLengthInBytes);
   int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks);
   int GetRTPStatistics(CallStatistics& stats);
   void SetNACKStatus(bool enable, int maxNumberOfPackets);
 
   // From AudioPacketizationCallback in the ACM
   int32_t SendData(FrameType frameType,
                    uint8_t payloadType,
                    uint32_t timeStamp,
                    const uint8_t* payloadData,
                    size_t payloadSize,
@@ skipping 85 matching lines @@
 
   void OnRecoverableUplinkPacketLossRate(float recoverable_packet_loss_rate);
 
   std::vector<RtpSource> GetSources() const {
     return rtp_receiver_->GetSources();
   }
 
  private:
   class ProcessAndEncodeAudioTask;
 
+  int GetRemoteSSRC(unsigned int& ssrc);
   void OnUplinkPacketLossRate(float packet_loss_rate);
   bool InputMute() const;
   bool OnRtpPacketWithHeader(const uint8_t* received_packet,
                              size_t length,
                              RTPHeader *header);
   bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length);
 
   bool ReceivePacket(const uint8_t* packet,
                      size_t packet_length,
                      const RTPHeader& header,
@@ skipping 112 matching lines @@
 
   bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_lock_) = false;
 
   rtc::TaskQueue* encoder_queue_ = nullptr;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // VOICE_ENGINE_CHANNEL_H_