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.
  */
 
 #include "webrtc/audio/audio_send_stream.h"
 
 #include <string>
 
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/task_queue.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/transmit_mixer.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 
 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 kPacketLossRateMinNumAckedPackets = 50;
+constexpr size_t kRecoverablePacketLossRateMinNumAckedPairs = 40;
+
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     rtc::TaskQueue* worker_queue,
     PacketRouter* packet_router,
     SendSideCongestionController* send_side_cc,
     BitrateAllocator* bitrate_allocator,
     RtcEventLog* event_log,
     RtcpRttStats* rtcp_rtt_stats)
     : worker_queue_(worker_queue),
       config_(config),
       audio_state_(audio_state),
       bitrate_allocator_(bitrate_allocator),
-      send_side_cc_(send_side_cc) {
+      send_side_cc_(send_side_cc),
+      packet_loss_tracker_(kPacketLossTrackerMaxWindowSizeMs,
+                           kPacketLossRateMinNumAckedPackets,
+                           kRecoverablePacketLossRateMinNumAckedPairs) {
   LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
   RTC_DCHECK(send_side_cc);
 
   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);
+  send_side_cc_->RegisterPacketFeedbackObserver(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);
       send_side_cc->EnablePeriodicAlrProbing(true);
       bandwidth_observer_.reset(
           send_side_cc->GetBitrateController()->CreateRtcpBandwidthObserver());
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
   }
   channel_proxy_->RegisterSenderCongestionControlObjects(
       send_side_cc->pacer(), send_side_cc, packet_router,
       bandwidth_observer_.get());
   if (!SetupSendCodec()) {
     LOG(LS_ERROR) << "Failed to set up send codec state.";
   }
+
+  pacer_thread_checker_.DetachFromThread();
 }
 
 AudioSendStream::~AudioSendStream() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
+  send_side_cc_->DeRegisterPacketFeedbackObserver(this);
   channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
   channel_proxy_->SetRtcpRttStats(nullptr);
 }
 
 void AudioSendStream::Start() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_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);
     rtc::Event thread_sync_event(false /* manual_reset */, false);
     worker_queue_->PostTask([this, &thread_sync_event] {
       bitrate_allocator_->AddObserver(this, config_.min_bitrate_bps,
                                       config_.max_bitrate_bps, 0, true);
       thread_sync_event.Set();
     });
     thread_sync_event.Wait(rtc::Event::kForever);
   }
 
   ScopedVoEInterface<VoEBase> base(voice_engine());
   int error = base->StartSend(config_.voe_channel_id);
   if (error != 0) {
     LOG(LS_ERROR) << "AudioSendStream::Start failed with error: " << error;
   }
 }
 
 void AudioSendStream::Stop() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   rtc::Event thread_sync_event(false /* manual_reset */, false);
   worker_queue_->PostTask([this, &thread_sync_event] {
     bitrate_allocator_->RemoveObserver(this);
     thread_sync_event.Set();
   });
   thread_sync_event.Wait(rtc::Event::kForever);
 
   ScopedVoEInterface<VoEBase> base(voice_engine());
   int error = base->StopSend(config_.voe_channel_id);
   if (error != 0) {
     LOG(LS_ERROR) << "AudioSendStream::Stop failed with error: " << error;
   }
 }
 
 bool AudioSendStream::SendTelephoneEvent(int payload_type,
                                          int payload_frequency, int event,
                                          int duration_ms) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type,
                                                           payload_frequency) &&
          channel_proxy_->SendTelephoneEventOutband(event, duration_ms);
 }
 
 void AudioSendStream::SetMuted(bool muted) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   channel_proxy_->SetInputMute(muted);
 }
 
 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   webrtc::AudioSendStream::Stats stats;
   stats.local_ssrc = config_.rtp.ssrc;
 
   webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
   stats.bytes_sent = call_stats.bytesSent;
   stats.packets_sent = call_stats.packetsSent;
   // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
   // returns 0 to indicate an error value.
   if (call_stats.rttMs > 0) {
     stats.rtt_ms = call_stats.rttMs;
@@ skipping 43 matching lines @@
       audio_processing_stats.residual_echo_likelihood_recent_max;
 
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   stats.typing_noise_detected = audio_state->typing_noise_detected();
 
   return stats;
 }
 
 void AudioSendStream::SignalNetworkState(NetworkState state) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
 }
 
 bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   // TODO(solenberg): Tests call this function on a network thread, libjingle
   // calls on the worker thread. We should move towards always using a network
   // thread. Then this check can be enabled.
-  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
+  // RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread());
   return channel_proxy_->ReceivedRTCPPacket(packet, length);
 }
 
 uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
                                            uint8_t fraction_loss,
                                            int64_t rtt,
                                            int64_t probing_interval_ms) {
   RTC_DCHECK_GE(bitrate_bps,
                 static_cast<uint32_t>(config_.min_bitrate_bps));
   // The bitrate allocator might allocate an higher than max configured bitrate
   // if there is room, to allow for, as example, extra FEC. Ignore that for now.
   const uint32_t max_bitrate_bps = config_.max_bitrate_bps;
   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) {
+  RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
+  // Only packets that belong to this stream are of interest.
+  if (ssrc == config_.rtp.ssrc) {
+    rtc::CritScope lock(&packet_loss_tracker_cs_);
+    // TODO(elad.alon): This function call could potentially reset the window,
+    // setting both PLR and RPLR to unknown. Consider (during upcoming
+    // refactoring) passing an indication of such an event.
+    packet_loss_tracker_.OnPacketAdded(seq_num, rtc::TimeMillis());
+  }
+}
+
+void AudioSendStream::OnPacketFeedbackVector(
+    const std::vector<PacketFeedback>& packet_feedback_vector) {
+  // TODO(elad.alon): This fails in UT; fix and uncomment.
+  // RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  rtc::Optional<float> plr;
+  {
+    rtc::CritScope lock(&packet_loss_tracker_cs_);
+    packet_loss_tracker_.OnPacketFeedbackVector(packet_feedback_vector);
+    plr = packet_loss_tracker_.GetPacketLossRate();
+  }
+  // TODO(elad.alon): If PLR goes back to unknown, no indication is given that
+  // the previously sent value is no longer relevant. This will be taken care
+  // of with some refactoring which is now being done.
+  if (plr) {
+    channel_proxy_->OnTwccBasedUplinkPacketLossRate(*plr);
+  }
+}
+
 const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return config_;
 }
 
 void AudioSendStream::SetTransportOverhead(int transport_overhead_per_packet) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   send_side_cc_->SetTransportOverhead(transport_overhead_per_packet);
   channel_proxy_->SetTransportOverhead(transport_overhead_per_packet);
 }
 
 VoiceEngine* AudioSendStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
@@ skipping 106 matching lines @@
         LOG(LS_WARNING) << "SetVADStatus() failed.";
         return false;
       }
     }
   }
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc