Attach TransportFeedbackPacketLossTracker to ANA (PLR only)

webrtc/audio/audio_send_stream.cc

Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 438d1cc78a5aca5d7657b6368bfbac03fa5aed8e..79afdf6579a1a7300755841deb4f47c4e11d513d 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -40,6 +40,11 @@ bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
 }  // namespace
 
 namespace internal {
+// TODO(elad.alon): Subsequent CL will make these values experiment-dependent.
+constexpr size_t kPacketLossTrackerMaxWindowSizeMs = 15000;
+constexpr size_t kPlrMinNumAckedPackets = 50;
+constexpr size_t kRplrMinNumAckedPairs = 40;
+
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
@@ -49,11 +54,15 @@ AudioSendStream::AudioSendStream(
     BitrateAllocator* bitrate_allocator,
     RtcEventLog* event_log,
     RtcpRttStats* rtcp_rtt_stats)
-    : worker_queue_(worker_queue),
+    : clock_(Clock::GetRealTimeClock()),
+      worker_queue_(worker_queue),
       config_(config),
       audio_state_(audio_state),
       bitrate_allocator_(bitrate_allocator),
-      congestion_controller_(congestion_controller) {
+      congestion_controller_(congestion_controller),
+      packet_loss_tracker_(kPacketLossTrackerMaxWindowSizeMs,
+                           kPlrMinNumAckedPackets,
+                           kRplrMinNumAckedPairs) {
   LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
@@ -72,6 +81,7 @@ AudioSendStream::AudioSendStream(
                                 config_.rtp.nack.rtp_history_ms / 20);
 
   channel_proxy_->RegisterExternalTransport(config.send_transport);
+  congestion_controller_->RegisterTransportFeedbackAdapterObserver(this);
 
   for (const auto& extension : config.rtp.extensions) {
     if (extension.uri == RtpExtension::kAudioLevelUri) {
@@ -96,6 +106,7 @@ AudioSendStream::AudioSendStream(
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
+  congestion_controller_->DeRegisterTransportFeedbackAdapterObserver(this);
   channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
@@ -247,6 +258,23 @@ uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
   return 0;
 }
 
+void AudioSendStream::OnPacketAdded(uint32_t ssrc, uint16_t seq_num) {
+  // Only packets that belong to this stream are of interest.
+  if (ssrc == config_.rtp.ssrc) {
+    rtc::CritScope lock(&packet_loss_tracker_cs_);
+    packet_loss_tracker_.OnPacketAdded(seq_num, clock_->TimeInMilliseconds());
+  }
+}
+
+void AudioSendStream::OnTransportFeedback(
+    const rtcp::TransportFeedback& feedback) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  rtc::CritScope lock(&packet_loss_tracker_cs_);
+  packet_loss_tracker_.OnReceivedTransportFeedback(feedback);
+  channel_proxy_->OnTwccBasedUplinkPacketLossRate(
+      packet_loss_tracker_.GetPacketLossRate());
+}
+
 const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return config_;