Attach TransportFeedbackPacketLossTracker to ANA (PLR only)

webrtc/audio/audio_send_stream.h

 /*
  *  Copyright (c) 2015 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_AUDIO_AUDIO_SEND_STREAM_H_
 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 
 #include <memory>
+#include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/bitrate_allocator.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/voice_engine/transport_feedback_packet_loss_tracker.h"
 
 namespace webrtc {
 class SendSideCongestionController;
 class VoiceEngine;
 class RtcEventLog;
 class RtcpBandwidthObserver;
 class RtcpRttStats;
 class PacketRouter;
 
 namespace voe {
 class ChannelProxy;
 }  // namespace voe
 
 namespace internal {
 class AudioSendStream final : public webrtc::AudioSendStream,
-                              public webrtc::BitrateAllocatorObserver {
+                              public webrtc::BitrateAllocatorObserver,
+                              public webrtc::PacketFeedbackObserver {
  public:
   AudioSendStream(const webrtc::AudioSendStream::Config& config,
                   const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
                   rtc::TaskQueue* worker_queue,
                   PacketRouter* packet_router,
                   SendSideCongestionController* send_side_cc,
                   BitrateAllocator* bitrate_allocator,
                   RtcEventLog* event_log,
                   RtcpRttStats* rtcp_rtt_stats);
   ~AudioSendStream() override;
 
   // webrtc::AudioSendStream implementation.
   void Start() override;
   void Stop() override;
   bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
                           int duration_ms) override;
   void SetMuted(bool muted) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
   void SignalNetworkState(NetworkState state);
   bool DeliverRtcp(const uint8_t* packet, size_t length);
 
   // Implements BitrateAllocatorObserver.
   uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
                             uint8_t fraction_loss,
                             int64_t rtt,
                             int64_t probing_interval_ms) override;
 
+  // From PacketFeedbackObserver.
+  void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override;
+  void OnPacketFeedbackVector(
+      const std::vector<PacketFeedback>& packet_feedback_vector) override;
+
   const webrtc::AudioSendStream::Config& config() const;
   void SetTransportOverhead(int transport_overhead_per_packet);
 
  private:
   VoiceEngine* voice_engine() const;
 
   bool SetupSendCodec();
 
-  rtc::ThreadChecker thread_checker_;
+  rtc::ThreadChecker worker_thread_checker_;
+  rtc::ThreadChecker pacer_thread_checker_;
   rtc::TaskQueue* worker_queue_;
   const webrtc::AudioSendStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   BitrateAllocator* const bitrate_allocator_;
   SendSideCongestionController* const send_side_cc_;
   std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
 
+  rtc::CriticalSection packet_loss_tracker_cs_;
+  TransportFeedbackPacketLossTracker packet_loss_tracker_
+      GUARDED_BY(&packet_loss_tracker_cs_);
+
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_