Attach TransportFeedbackPacketLossTracker to ANA (PLR only)

webrtc/audio/audio_send_stream.h

Index: webrtc/audio/audio_send_stream.h
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index 436c49824cc2325921c560e59f2f56bc3d439f99..f7ab1af3f986aab88714db2f6e59612290fd41ef 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -18,6 +18,8 @@
 #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;

[minyue-webrtc, 2017/03/22 07:51:39]

just a note: please try rebasing in a separate patch set. usually, such a patch
set will require minimum review.

[elad.alon_webrtc.org, 2017/03/22 09:36:30]

Sure, will do.

@@ -33,7 +35,8 @@ class ChannelProxy;
 
 namespace internal {
 class AudioSendStream final : public webrtc::AudioSendStream,
-                              public webrtc::BitrateAllocatorObserver {
+                              public webrtc::BitrateAllocatorObserver,
+                              public webrtc::TransportFeedbackAdapterObserver {
  public:
   AudioSendStream(const webrtc::AudioSendStream::Config& config,
                   const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
@@ -62,6 +65,11 @@ class AudioSendStream final : public webrtc::AudioSendStream,
                             int64_t rtt,
                             int64_t probing_interval_ms) override;
 
+  // From TransportFeedbackAdapterObserver
+  void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override;
+  void OnNewTransportFeedbackVector(
+      const std::vector<PacketFeedback>& packet_feedback_vector) override;
+
   const webrtc::AudioSendStream::Config& config() const;
   void SetTransportOverhead(int transport_overhead_per_packet);
 
@@ -70,6 +78,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
 
   bool SetupSendCodec();
 
+  const Clock* const clock_;
   rtc::ThreadChecker thread_checker_;
   rtc::TaskQueue* worker_queue_;
   const webrtc::AudioSendStream::Config config_;
@@ -80,6 +89,10 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   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