Attach TransportFeedbackPacketLossTracker to ANA (PLR only)

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 15 matching lines @@
 #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/voice_engine/audio_level.h"
 #include "webrtc/voice_engine/file_player.h"
 #include "webrtc/voice_engine/file_recorder.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 #include "webrtc/voice_engine/shared_data.h"
+#include "webrtc/voice_engine/transport_feedback_packet_loss_tracker.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace rtc {
 class TimestampWrapAroundHandler;
 }
 
 namespace webrtc {
 
 class AudioDeviceModule;
 class FileWrapper;
@@ skipping 323 matching lines @@
 
   // Set a RtcEventLog logging object.
   void SetRtcEventLog(RtcEventLog* event_log);
 
   void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
   void SetTransportOverhead(size_t transport_overhead_per_packet);
 
   // From OverheadObserver in the RTP/RTCP module
   void OnOverheadChanged(size_t overhead_bytes_per_packet) override;
 
+  // Note: The existence of this function alongside OnUplinkPacketLossRate is
+  // a compromise. We want the encoder to be agnostic of the PLR source, but
+  // we also don't want it to receive conflicting information from TWCC and
+  // from RTCP-XR.
+  void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
+
  protected:
-  void OnIncomingFractionLoss(int fraction_lost);
+  void OnUplinkPacketLossRate(float packet_loss_rate);
 
  private:
   bool InputMute() const;
   bool OnRtpPacketWithHeader(const uint8_t* received_packet,
                              size_t length,
                              RTPHeader *header);
   bool ReceivePacket(const uint8_t* packet,
                      size_t packet_length,
                      const RTPHeader& header,
                      bool in_order);
@@ skipping 106 matching lines @@
 
   bool pacing_enabled_;
   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_;
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+
+  const bool use_twcc_plr_for_ana_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_