Add RtcpRttStats to AudioStream

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 49 matching lines @@
 class VoiceEngineObserver;
 
 struct CallStatistics;
 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.
@@ skipping 330 matching lines @@
     rtc::CritScope lock(&assoc_send_channel_lock_);
     associate_send_channel_ = channel;
   }
 
   // Disassociate a send channel if it was associated.
   void DisassociateSendChannel(int channel_id);
 
   // Set a RtcEventLog logging object.
   void SetRtcEventLog(RtcEventLog* event_log);
 
+  void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
+
  protected:
   void OnIncomingFractionLoss(int fraction_lost);
 
  private:
   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,
@@ skipping 15 matching lines @@
 
   rtc::CriticalSection _fileCritSect;
   rtc::CriticalSection _callbackCritSect;
   rtc::CriticalSection volume_settings_critsect_;
   uint32_t _instanceId;
   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_;
@@ skipping 81 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_