Attach TransportFeedbackPacketLossTracker to ANA (PLR only)

webrtc/audio/audio_send_stream.cc

 /*
  *  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.
  */
 
@@ skipping 22 matching lines @@
 namespace {
 
 constexpr char kOpusCodecName[] = "opus";
 
 bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
   return (STR_CASE_CMP(codec.plname, ref_name) == 0);
 }
 }  // 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,
     rtc::TaskQueue* worker_queue,
     PacketRouter* packet_router,
     CongestionController* congestion_controller,
     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());
   RTC_DCHECK(congestion_controller);
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->SetRtcEventLog(event_log);
   channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
   channel_proxy_->SetRTCPStatus(true);
   channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
   channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
   // TODO(solenberg): Config NACK history window (which is a packet count),
   // using the actual packet size for the configured codec.
   channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0,
                                 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) {
       channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
     } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
       channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
       congestion_controller->EnablePeriodicAlrProbing(true);
       bandwidth_observer_.reset(congestion_controller->GetBitrateController()
                                     ->CreateRtcpBandwidthObserver());
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
   }
   channel_proxy_->RegisterSenderCongestionControlObjects(
       congestion_controller->pacer(), congestion_controller, packet_router,
       bandwidth_observer_.get());
   if (!SetupSendCodec()) {
     LOG(LS_ERROR) << "Failed to set up send codec state.";
   }
 }
 
 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);
   channel_proxy_->SetRtcpRttStats(nullptr);
 }
 
 void AudioSendStream::Start() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config_.max_bitrate_bps, config_.min_bitrate_bps);
@@ skipping 131 matching lines @@
   if (bitrate_bps > max_bitrate_bps)
     bitrate_bps = max_bitrate_bps;
 
   channel_proxy_->SetBitrate(bitrate_bps, probing_interval_ms);
 
   // The amount of audio protection is not exposed by the encoder, hence
   // always returning 0.
   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::OnNewTransportFeedbacks(
+    const std::vector<PacketFeedback>& packet_feedbacks) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  rtc::CritScope lock(&packet_loss_tracker_cs_);
+  packet_loss_tracker_.OnNewTransportFeedbacks(packet_feedbacks);
+  channel_proxy_->OnTwccBasedUplinkPacketLossRate(
+      packet_loss_tracker_.GetPacketLossRate());
+}
+
 const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return config_;
 }
 
 void AudioSendStream::SetTransportOverhead(int transport_overhead_per_packet) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   congestion_controller_->SetTransportOverhead(transport_overhead_per_packet);
   channel_proxy_->SetTransportOverhead(transport_overhead_per_packet);
 }
@@ skipping 113 matching lines @@
         LOG(LS_WARNING) << "SetVADStatus() failed.";
         return false;
       }
     }
   }
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc