Replace the implementation of GetContributingSources() on the audio side.

This change replaces the `ContributingSources`-implementation of `GetContributingSources()` and `GetSynchronizationSources()` on the audio side with the spec-compliant `SourceTracker`-implementation.

The most noticeable impact is that the per-frame dictionaries are now updated when frames are delivered to the RTCRtpReceiver's MediaStreamTrack rather than when RTP packets are received on the network.

This change is almost identical to the previous video side change at: https://webrtc-review.googlesource.com/c/src/+/143177

Bug: webrtc:10545
Change-Id: Ife7f08ee8ca1346099b7466837a3756947085fc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144422
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28459}

audio/audio_receive_stream.cc

@@ -113,7 +113,9 @@ AudioReceiveStream::AudioReceiveStream(
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     webrtc::RtcEventLog* event_log,
     std::unique_ptr<voe::ChannelReceiveInterface> channel_receive)
-    : audio_state_(audio_state), channel_receive_(std::move(channel_receive)) {
+    : audio_state_(audio_state),
+      channel_receive_(std::move(channel_receive)),
+      source_tracker_(clock) {
   RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.rtp.remote_ssrc;
   RTC_DCHECK(config.decoder_factory);
   RTC_DCHECK(config.rtcp_send_transport);
@@ -267,13 +269,18 @@ int AudioReceiveStream::GetBaseMinimumPlayoutDelayMs() const {
 
 std::vector<RtpSource> AudioReceiveStream::GetSources() const {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
-  return channel_receive_->GetSources();
+  return source_tracker_.GetSources();
 }
 
 AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
     int sample_rate_hz,
     AudioFrame* audio_frame) {
-  return channel_receive_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
+  AudioMixer::Source::AudioFrameInfo audio_frame_info =
+      channel_receive_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
+  if (audio_frame_info != AudioMixer::Source::AudioFrameInfo::kError) {
+    source_tracker_.OnFrameDelivered(audio_frame->packet_infos_);
+  }
+  return audio_frame_info;
 }
 
 int AudioReceiveStream::Ssrc() const {

audio/audio_receive_stream.h

@@ -19,6 +19,7 @@
 #include "audio/audio_state.h"
 #include "call/audio_receive_stream.h"
 #include "call/syncable.h"
+#include "modules/rtp_rtcp/source/source_tracker.h"
 #include "rtc_base/constructor_magic.h"
 #include "rtc_base/thread_checker.h"
 #include "system_wrappers/include/clock.h"
@@ -107,6 +108,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   webrtc::AudioReceiveStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   const std::unique_ptr<voe::ChannelReceiveInterface> channel_receive_;
+  SourceTracker source_tracker_;
   AudioSendStream* associated_send_stream_ = nullptr;
 
   bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false;

audio/channel_receive.cc

@@ -30,7 +30,6 @@
 #include "modules/rtp_rtcp/include/receive_statistics.h"
 #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "modules/rtp_rtcp/source/contributing_sources.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
@@ -155,8 +154,6 @@ class ChannelReceive : public ChannelReceiveInterface,
   // Used for obtaining RTT for a receive-only channel.
   void SetAssociatedSendChannel(const ChannelSendInterface* channel) override;
 
-  std::vector<RtpSource> GetSources() const override;
-
   // TODO(sukhanov): Return const pointer. It requires making media transport
   // getters like GetLatestTargetTransferRate to be also const.
   MediaTransportInterface* media_transport() const {
@@ -213,16 +210,13 @@ class ChannelReceive : public ChannelReceiveInterface,
   std::unique_ptr<RtpRtcp> _rtpRtcpModule;
   const uint32_t remote_ssrc_;
 
-  // Info for GetSources and GetSyncInfo is updated on network or worker thread,
-  // queried on the worker thread.
-  rtc::CriticalSection rtp_sources_lock_;
-  ContributingSources contributing_sources_ RTC_GUARDED_BY(&rtp_sources_lock_);
+  // Info for GetSyncInfo is updated on network or worker thread, and queried on
+  // the worker thread.
+  rtc::CriticalSection sync_info_lock_;
   absl::optional<uint32_t> last_received_rtp_timestamp_
-      RTC_GUARDED_BY(&rtp_sources_lock_);
+      RTC_GUARDED_BY(&sync_info_lock_);
   absl::optional<int64_t> last_received_rtp_system_time_ms_
-      RTC_GUARDED_BY(&rtp_sources_lock_);
-  absl::optional<uint8_t> last_received_rtp_audio_level_
-      RTC_GUARDED_BY(&rtp_sources_lock_);
+      RTC_GUARDED_BY(&sync_info_lock_);
 
   std::unique_ptr<AudioCodingModule> audio_coding_;
   AudioSinkInterface* audio_sink_ = nullptr;
@@ -555,24 +549,6 @@ absl::optional<std::pair<int, SdpAudioFormat>>
   return audio_coding_->ReceiveCodec();
 }
 
-std::vector<webrtc::RtpSource> ChannelReceive::GetSources() const {
-  RTC_DCHECK(worker_thread_checker_.IsCurrent());
-  int64_t now_ms = rtc::TimeMillis();
-  std::vector<RtpSource> sources;
-  {
-    rtc::CritScope cs(&rtp_sources_lock_);
-    sources = contributing_sources_.GetSources(now_ms);
-    if (last_received_rtp_system_time_ms_ >=
-        now_ms - ContributingSources::kHistoryMs) {
-      RTC_DCHECK(last_received_rtp_timestamp_.has_value());
-      sources.emplace_back(*last_received_rtp_system_time_ms_, remote_ssrc_,
-                           RtpSourceType::SSRC, last_received_rtp_audio_level_,
-                           *last_received_rtp_timestamp_);
-    }
-  }
-  return sources;
-}
-
 void ChannelReceive::SetReceiveCodecs(
     const std::map<int, SdpAudioFormat>& codecs) {
   RTC_DCHECK(worker_thread_checker_.IsCurrent());
@@ -586,22 +562,11 @@ void ChannelReceive::SetReceiveCodecs(
 // May be called on either worker thread or network thread.
 void ChannelReceive::OnRtpPacket(const RtpPacketReceived& packet) {
   int64_t now_ms = rtc::TimeMillis();
-  uint8_t audio_level;
-  bool voice_activity;
-  bool has_audio_level =
-      packet.GetExtension<::webrtc::AudioLevel>(&voice_activity, &audio_level);
 
   {
-    rtc::CritScope cs(&rtp_sources_lock_);
+    rtc::CritScope cs(&sync_info_lock_);
     last_received_rtp_timestamp_ = packet.Timestamp();
     last_received_rtp_system_time_ms_ = now_ms;
-    if (has_audio_level)
-      last_received_rtp_audio_level_ = audio_level;
-    std::vector<uint32_t> csrcs = packet.Csrcs();
-    contributing_sources_.Update(
-        now_ms, csrcs,
-        has_audio_level ? absl::optional<uint8_t>(audio_level) : absl::nullopt,
-        packet.Timestamp());
   }
 
   // Store playout timestamp for the received RTP packet
@@ -875,7 +840,7 @@ absl::optional<Syncable::Info> ChannelReceive::GetSyncInfo() const {
     return absl::nullopt;
   }
   {
-    rtc::CritScope cs(&rtp_sources_lock_);
+    rtc::CritScope cs(&sync_info_lock_);
     if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) {
       return absl::nullopt;
     }

audio/channel_receive.h

@@ -135,8 +135,6 @@ class ChannelReceiveInterface : public RtpPacketSinkInterface {
   // Used for obtaining RTT for a receive-only channel.
   virtual void SetAssociatedSendChannel(
       const ChannelSendInterface* channel) = 0;
-
-  virtual std::vector<RtpSource> GetSources() const = 0;
 };
 
 std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(