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 229 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 122 matching lines @@
 
       extended_max_sequence_number_[block_it->sourceSSRC] =
           block_it->extendedHighSeqNum;
     }
     int weighted_fraction_lost = 0;
     if (total_number_of_packets > 0) {
       weighted_fraction_lost =
           (fraction_lost_aggregate + total_number_of_packets / 2) /
           total_number_of_packets;
     }
-    owner_->OnIncomingFractionLoss(weighted_fraction_lost);
+    owner_->OnUplinkPacketLossRate(weighted_fraction_lost / 255.0f);
   }
 
  private:
   Channel* owner_;
   // Maps remote side ssrc to extended highest sequence number received.
   std::map<uint32_t, uint32_t> extended_max_sequence_number_;
   rtc::CriticalSection crit_;
   RtcpBandwidthObserver* bandwidth_observer_ GUARDED_BY(crit_);
 };
 
@@ skipping 486 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) {
   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 366 matching lines @@
                "Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps);
   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) {
+void Channel::OnTwccBasedUplinkPacketLossRate(
+    const rtc::Optional<float>& packet_loss_rate) {
+  if (!use_twcc_plr_for_ana_)
+    return;
   audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
     if (*encoder)
       (*encoder)->OnReceivedUplinkPacketLossFraction(
-          rtc::Optional<float>(fraction_lost / 255.0f));
+          rtc::Optional<float>(packet_loss_rate));
   });
 }
 
+void Channel::OnUplinkPacketLossRate(float packet_loss_rate) {
+  if (use_twcc_plr_for_ana_)
+    return;
+  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+    if (*encoder) {
+      (*encoder)->OnReceivedUplinkPacketLossFraction(
+          rtc::Optional<float>(packet_loss_rate));
+    }
+  });
+}
+
 int32_t Channel::SetVADStatus(bool enableVAD,
                               ACMVADMode mode,
                               bool disableDTX) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetVADStatus(mode=%d)", mode);
   RTC_DCHECK(!(disableDTX && enableVAD));  // disableDTX mode is deprecated.
   if (!codec_manager_.SetVAD(enableVAD, mode) ||
       !codec_manager_.MakeEncoder(&rent_a_codec_, audio_coding_.get())) {
     _engineStatisticsPtr->SetLastError(VE_AUDIO_CODING_MODULE_ERROR,
                                        kTraceError,
@@ skipping 1682 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