Attach TransportFeedbackPacketLossTracker to ANA (PLR only)

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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 15 matching lines @@
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/output_mixer.h"
 #include "webrtc/voice_engine/statistics.h"
 #include "webrtc/voice_engine/utility.h"
 
 namespace webrtc {
 namespace voe {
 
 namespace {
 
 constexpr int64_t kMaxRetransmissionWindowMs = 1000;
 constexpr int64_t kMinRetransmissionWindowMs = 30;
 
+// 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;
+
 }  // namespace
 
 const int kTelephoneEventAttenuationdB = 10;
 
 class RtcEventLogProxy final : public webrtc::RtcEventLog {
  public:
   RtcEventLogProxy() : event_log_(nullptr) {}
 
   bool StartLogging(const std::string& file_name,
                     int64_t max_size_bytes) override {
@@ skipping 177 matching lines @@
   }
 
   void SetTransportFeedbackObserver(
       TransportFeedbackObserver* feedback_observer) {
     RTC_DCHECK(thread_checker_.CalledOnValidThread());
     rtc::CritScope lock(&crit_);
     feedback_observer_ = feedback_observer;
   }
 
   // Implements TransportFeedbackObserver.
-  void AddPacket(uint16_t sequence_number,
+  void AddPacket(uint32_t ssrc,
+                 uint16_t sequence_number,
                  size_t length,
                  const PacedPacketInfo& pacing_info) override {
     RTC_DCHECK(pacer_thread_.CalledOnValidThread());
     rtc::CritScope lock(&crit_);
     if (feedback_observer_)
-      feedback_observer_->AddPacket(sequence_number, length, pacing_info);
+      feedback_observer_->AddPacket(ssrc, sequence_number, length, pacing_info);
   }
 
   void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override {
     RTC_DCHECK(network_thread_.CalledOnValidThread());
     rtc::CritScope lock(&crit_);
     if (feedback_observer_)
       feedback_observer_->OnTransportFeedback(feedback);
   }
   std::vector<PacketFeedback> GetTransportFeedbackVector() const override {
     RTC_NOTREACHED();
@@ skipping 639 matching lines @@
       _outputSpeechType(AudioFrame::kNormalSpeech),
       restored_packet_in_use_(false),
       rtcp_observer_(new VoERtcpObserver(this)),
       associate_send_channel_(ChannelOwner(nullptr)),
       pacing_enabled_(config.enable_voice_pacing),
       feedback_observer_proxy_(new TransportFeedbackProxy()),
       seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
       rtp_packet_sender_proxy_(new RtpPacketSenderProxy()),
       retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(),
                                                    kMaxRetransmissionWindowMs)),
-      decoder_factory_(config.acm_config.decoder_factory) {
+      decoder_factory_(config.acm_config.decoder_factory),
+      // TODO(elad.alon): Subsequent CL experiments with PLR source.
+      use_twcc_plr_for_ana_(false),
+      packet_loss_tracker_(kPacketLossTrackerMaxWindowSizeMs,
+                           kPlrMinNumAckedPackets,
+                           kRplrMinNumAckedPairs) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::Channel() - ctor");
   AudioCodingModule::Config acm_config(config.acm_config);
   acm_config.id = VoEModuleId(instanceId, channelId);
   acm_config.neteq_config.enable_muted_state = true;
   audio_coding_.reset(AudioCodingModule::Create(acm_config));
 
   _outputAudioLevel.Clear();
 
   RtpRtcp::Configuration configuration;
@@ skipping 370 matching lines @@
   audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
     if (*encoder) {
       (*encoder)->OnReceivedUplinkBandwidth(
           bitrate_bps, rtc::Optional<int64_t>(probing_interval_ms));
     }
   });
   retransmission_rate_limiter_->SetMaxRate(bitrate_bps);
 }
 
 void Channel::OnIncomingFractionLoss(int fraction_lost) {
+  if (use_twcc_plr_for_ana_)
+    return;
   audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
     if (*encoder)
       (*encoder)->OnReceivedUplinkPacketLossFraction(fraction_lost / 255.0f);
   });
 }
 
 int32_t Channel::SetVADStatus(bool enableVAD,
                               ACMVADMode mode,
                               bool disableDTX) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
@@ skipping 1407 matching lines @@
   UpdateOverheadForEncoder();
 }
 
 // TODO(solenberg): Make AudioSendStream an OverheadObserver instead.
 void Channel::OnOverheadChanged(size_t overhead_bytes_per_packet) {
   rtc::CritScope cs(&overhead_per_packet_lock_);
   rtp_overhead_per_packet_ = overhead_bytes_per_packet;
   UpdateOverheadForEncoder();
 }
 
+void Channel::HandleTransportFeedback(
+    const std::vector<SentTransportPacketRecord>&
+        packets_sent_since_last_feedback,
+    const rtcp::TransportFeedback& feedback) {
+  for (const auto& sent_packet : packets_sent_since_last_feedback) {
+    packet_loss_tracker_.OnPacketAdded(sent_packet.sequence_number,
+                                       sent_packet.sent_time_ms);
+  }
+
+  // TODO(elad.alon): Needed even when !use_twcc_plr_for_ana_, because
+  // subsequent CLs will introduce pushing RPLR down to ANA, too, and that
+  // would happen unconditionally.
+  // (Rationale for unconditionality: it's cheap enough, and once
+  // experimentation is over, we either always do it or never do it, in which
+  // case the code involved will just be deleted.)
+  packet_loss_tracker_.OnReceivedTransportFeedback(feedback);
+
+  if (use_twcc_plr_for_ana_) {
+    rtc::Optional<float> plr = packet_loss_tracker_.GetPacketLossRate();
+    if (plr) {
+      audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+        if (*encoder)
+          (*encoder)->OnReceivedUplinkPacketLossFraction(*plr);
+      });
+    }
+  }
+}
+
 int Channel::GetNetworkStatistics(NetworkStatistics& stats) {
   return audio_coding_->GetNetworkStatistics(&stats);
 }
 
 void Channel::GetDecodingCallStatistics(AudioDecodingCallStats* stats) const {
   audio_coding_->GetDecodingCallStatistics(stats);
 }
 
 uint32_t Channel::GetDelayEstimate() const {
   rtc::CritScope lock(&video_sync_lock_);
@@ skipping 266 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc