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 5ee49da91a7a1469285d1fcf675c39c53a43ac5c..87d2a3b2e3a952adc47a09c196e418ad30357e22 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -12,12 +12,14 @@
 #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"
 
 namespace webrtc {
 class CongestionController;
@@ -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 +65,10 @@ 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 OnTransportFeedback(const rtcp::TransportFeedback& feedback) override;
+
   const webrtc::AudioSendStream::Config& config() const;
   void SetTransportOverhead(int transport_overhead_per_packet);
 
@@ -70,6 +77,7 @@ 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 +88,10 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   CongestionController* const congestion_controller_;
   std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
 
+  rtc::CriticalSection packets_sent_since_last_feedback_cs_;
+  std::vector<SentTransportPacketRecord> packets_sent_since_last_feedback_

[minyue-webrtc, 2017/03/15 10:54:13]

I don't understand why we need this buffer given that the packet loss tracker
has its internal buffer.

[elad.alon_webrtc.org, 2017/03/16 18:37:35]

Because of the threading issue. Please see my other comment about this, in the
.cc file.

+      GUARDED_BY(&packets_sent_since_last_feedback_cs_);
+
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal