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 d8bc84858a6a3e33936a89dd1dfa6c6c173d68b9..be7672d7cbb6e55a16d9f77649fa39bf19919867 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -11,6 +11,8 @@
 #include "webrtc/audio/audio_send_stream.h"
 
 #include <string>
+#include <vector>
+#include <utility>
 
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
@@ -18,6 +20,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/mod_ops.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
@@ -76,6 +79,7 @@ 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) {
@@ -95,6 +99,7 @@ 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);
@@ -253,6 +258,51 @@ uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
   return 0;
 }
 
+void AudioSendStream::OnPacketAdded(uint32_t ssrc,
+                                    uint16_t transport_sequence_number) {
+  if (ssrc != config_.rtp.ssrc)
+    return;
+
+  // To make sure everything happens on the same thread, we'll buffer
+  // this information and pass it down with the first OnTransportFeedback,
+  // which is called on the thread which channel_proxy_ mostly works on.
+  rtc::CritScope lock(&packets_sent_since_last_feedback_cs_);
+
+  // Prevent unbounded memory consumption if OnTransportFeedback ends up
+  // never being called. Messages which are 0x8000 (or more) sequence numbers
+  // away from the newest message will end up having no effect, so we can
+  // discard those.
+  if (!packets_sent_since_last_feedback_.empty() &&
+      ForwardDiff(packets_sent_since_last_feedback_[0],
+                  transport_sequence_number) >= 0x8000) {
+    // The element are ordered (circularly), so we can batch-remove. No need
+    // for binary search, because we expect to usually find the edge in the
+    // beginning of the container.
+    auto it = packets_sent_since_last_feedback_.cbegin();
+    while (it != packets_sent_since_last_feedback_.cend() &&
+           ForwardDiff(*it, transport_sequence_number) >= 0x8000) {
+      ++it;
+    }
+    packets_sent_since_last_feedback_.erase(
+        packets_sent_since_last_feedback_.cbegin(), it);
+  }
+
+  packets_sent_since_last_feedback_.push_back(transport_sequence_number);
+}
+
+void AudioSendStream::OnTransportFeedback(
+    const rtcp::TransportFeedback& feedback) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  std::vector<uint16_t> packets_sent_since_last_feedback;
+  {
+    rtc::CritScope lock(&packets_sent_since_last_feedback_cs_);
+    std::swap(packets_sent_since_last_feedback_,
+              packets_sent_since_last_feedback);
+  }
+  channel_proxy_->HandleTransportFeedback(
+      packets_sent_since_last_feedback, feedback);
+}
+
 const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return config_;