Store/restore RTP state for audio streams with same SSRC within a call

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 287 matching lines @@
   };
   std::map<uint32_t, ReceiveRtpConfig> receive_rtp_config_
       GUARDED_BY(receive_crit_);
 
   std::unique_ptr<RWLockWrapper> send_crit_;
   // Audio and Video send streams are owned by the client that creates them.
   std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_ GUARDED_BY(send_crit_);
   std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_);
   std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_);
 
-  VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
+  using RtpStateMap = std::map<uint32_t, RtpState>;
+  RtpStateMap suspended_audio_send_ssrcs_
+      GUARDED_BY(configuration_thread_checker_);
+  RtpStateMap suspended_video_send_ssrcs_
+      GUARDED_BY(configuration_thread_checker_);
+
   webrtc::RtcEventLog* event_log_;
 
   // The following members are only accessed (exclusively) from one thread and
   // from the destructor, and therefore doesn't need any explicit
   // synchronization.
   RateCounter received_bytes_per_second_counter_;
   RateCounter received_audio_bytes_per_second_counter_;
   RateCounter received_video_bytes_per_second_counter_;
   RateCounter received_rtcp_bytes_per_second_counter_;
 
@@ skipping 66 matching lines @@
       received_video_bytes_per_second_counter_(clock_, nullptr, true),
       received_rtcp_bytes_per_second_counter_(clock_, nullptr, true),
       min_allocated_send_bitrate_bps_(0),
       configured_max_padding_bitrate_bps_(0),
       estimated_send_bitrate_kbps_counter_(clock_, nullptr, true),
       pacer_bitrate_kbps_counter_(clock_, nullptr, true),
       receive_side_cc_(clock_, transport_send->packet_router()),
       video_send_delay_stats_(new SendDelayStats(clock_)),
       start_ms_(clock_->TimeInMilliseconds()),
       worker_queue_("call_worker_queue") {
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   RTC_DCHECK(config.event_log != nullptr);
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
                 config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
                   config.bitrate_config.start_bitrate_bps);
   }
   Trace::CreateTrace();
   transport_send->send_side_cc()->RegisterNetworkObserver(this);
@@ skipping 13 matching lines @@
                                          RTC_FROM_HERE);
   pacer_thread_->RegisterModule(transport_send_->send_side_cc()->pacer(),
                                 RTC_FROM_HERE);
   pacer_thread_->RegisterModule(
       receive_side_cc_.GetRemoteBitrateEstimator(true), RTC_FROM_HERE);
 
   pacer_thread_->Start();
 }
 
 Call::~Call() {
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
 
   RTC_CHECK(audio_send_ssrcs_.empty());
   RTC_CHECK(video_send_ssrcs_.empty());
   RTC_CHECK(video_send_streams_.empty());
   RTC_CHECK(audio_receive_streams_.empty());
   RTC_CHECK(video_receive_streams_.empty());
 
   pacer_thread_->Stop();
   pacer_thread_->DeRegisterModule(transport_send_->send_side_cc()->pacer());
   pacer_thread_->DeRegisterModule(
@@ skipping 106 matching lines @@
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps",
                                 recv_bytes_per_sec.average * 8 / 1000);
     LOG(LS_INFO) << "WebRTC.Call.BitrateReceivedInBps, "
                  << recv_bytes_per_sec.ToStringWithMultiplier(8);
   }
 }
 
 PacketReceiver* Call::Receiver() {
   // TODO(solenberg): Some test cases in EndToEndTest use this from a different
   // thread. Re-enable once that is fixed.
-  // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  // RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   return this;
 }
 
 webrtc::AudioSendStream* Call::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   event_log_->LogAudioSendStreamConfig(CreateRtcLogStreamConfig(config));
+
+  rtc::Optional<RtpState> suspended_rtp_state;
+  {
+    const auto& iter = suspended_audio_send_ssrcs_.find(config.rtp.ssrc);
+    if (iter != suspended_audio_send_ssrcs_.end()) {
+      suspended_rtp_state.emplace(iter->second);
+    }
+  }
+
   AudioSendStream* send_stream = new AudioSendStream(
       config, config_.audio_state, &worker_queue_, transport_send_.get(),
-      bitrate_allocator_.get(), event_log_, call_stats_->rtcp_rtt_stats());
+      bitrate_allocator_.get(), event_log_, call_stats_->rtcp_rtt_stats(),
+      suspended_rtp_state);
   {
     WriteLockScoped write_lock(*send_crit_);
     RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
                audio_send_ssrcs_.end());
     audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
   }
   {
     ReadLockScoped read_lock(*receive_crit_);
     for (AudioReceiveStream* stream : audio_receive_streams_) {
       if (stream->config().rtp.local_ssrc == config.rtp.ssrc) {
         stream->AssociateSendStream(send_stream);
       }
     }
   }
   send_stream->SignalNetworkState(audio_network_state_);
   UpdateAggregateNetworkState();
   return send_stream;
 }
 
 void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyAudioSendStream");
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   RTC_DCHECK(send_stream != nullptr);
 
   send_stream->Stop();
 
   webrtc::internal::AudioSendStream* audio_send_stream =
       static_cast<webrtc::internal::AudioSendStream*>(send_stream);
-  uint32_t ssrc = audio_send_stream->config().rtp.ssrc;
+  const uint32_t ssrc = audio_send_stream->config().rtp.ssrc;
+  suspended_audio_send_ssrcs_[ssrc] = audio_send_stream->GetRtpState();
   {
     WriteLockScoped write_lock(*send_crit_);
     size_t num_deleted = audio_send_ssrcs_.erase(ssrc);
     RTC_DCHECK_EQ(1, num_deleted);
   }
   {
     ReadLockScoped read_lock(*receive_crit_);
     for (AudioReceiveStream* stream : audio_receive_streams_) {
       if (stream->config().rtp.local_ssrc == ssrc) {
         stream->AssociateSendStream(nullptr);
       }
     }
   }
   UpdateAggregateNetworkState();
   delete audio_send_stream;
 }
 
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   event_log_->LogAudioReceiveStreamConfig(CreateRtcLogStreamConfig(config));
   AudioReceiveStream* receive_stream =
       new AudioReceiveStream(transport_send_->packet_router(), config,
                              config_.audio_state, event_log_);
   {
     WriteLockScoped write_lock(*receive_crit_);
     audio_rtp_demuxer_.AddSink(config.rtp.remote_ssrc, receive_stream);
     receive_rtp_config_[config.rtp.remote_ssrc] =
         ReceiveRtpConfig(config.rtp.extensions, UseSendSideBwe(config));
     audio_receive_streams_.insert(receive_stream);
 
     ConfigureSync(config.sync_group);
   }
   {
     ReadLockScoped read_lock(*send_crit_);
     auto it = audio_send_ssrcs_.find(config.rtp.local_ssrc);
     if (it != audio_send_ssrcs_.end()) {
       receive_stream->AssociateSendStream(it->second);
     }
   }
   receive_stream->SignalNetworkState(audio_network_state_);
   UpdateAggregateNetworkState();
   return receive_stream;
 }
 
 void Call::DestroyAudioReceiveStream(
     webrtc::AudioReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyAudioReceiveStream");
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   RTC_DCHECK(receive_stream != nullptr);
   webrtc::internal::AudioReceiveStream* audio_receive_stream =
       static_cast<webrtc::internal::AudioReceiveStream*>(receive_stream);
   {
     WriteLockScoped write_lock(*receive_crit_);
     const AudioReceiveStream::Config& config = audio_receive_stream->config();
     uint32_t ssrc = config.rtp.remote_ssrc;
     receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config))
         ->RemoveStream(ssrc);
     size_t num_deleted = audio_rtp_demuxer_.RemoveSink(audio_receive_stream);
     RTC_DCHECK(num_deleted == 1);
     audio_receive_streams_.erase(audio_receive_stream);
     const std::string& sync_group = audio_receive_stream->config().sync_group;
     const auto it = sync_stream_mapping_.find(sync_group);
     if (it != sync_stream_mapping_.end() &&
         it->second == audio_receive_stream) {
       sync_stream_mapping_.erase(it);
       ConfigureSync(sync_group);
     }
     receive_rtp_config_.erase(ssrc);
   }
   UpdateAggregateNetworkState();
   delete audio_receive_stream;
 }
 
 webrtc::VideoSendStream* Call::CreateVideoSendStream(
     webrtc::VideoSendStream::Config config,
     VideoEncoderConfig encoder_config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
 
   video_send_delay_stats_->AddSsrcs(config);
   for (size_t ssrc_index = 0; ssrc_index < config.rtp.ssrcs.size();
        ++ssrc_index) {
     event_log_->LogVideoSendStreamConfig(
         CreateRtcLogStreamConfig(config, ssrc_index));
   }
 
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
@@ skipping 15 matching lines @@
   }
   send_stream->SignalNetworkState(video_network_state_);
   UpdateAggregateNetworkState();
 
   return send_stream;
 }
 
 void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
   RTC_DCHECK(send_stream != nullptr);
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
 
   send_stream->Stop();
 
   VideoSendStream* send_stream_impl = nullptr;
   {
     WriteLockScoped write_lock(*send_crit_);
     auto it = video_send_ssrcs_.begin();
     while (it != video_send_ssrcs_.end()) {
       if (it->second == static_cast<VideoSendStream*>(send_stream)) {
         send_stream_impl = it->second;
@@ skipping 14 matching lines @@
     suspended_video_send_ssrcs_[it->first] = it->second;
   }
 
   UpdateAggregateNetworkState();
   delete send_stream_impl;
 }
 
 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
     webrtc::VideoReceiveStream::Config configuration) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
 
   VideoReceiveStream* receive_stream =
       new VideoReceiveStream(num_cpu_cores_, transport_send_->packet_router(),
                              std::move(configuration),
                              module_process_thread_.get(), call_stats_.get());
 
   const webrtc::VideoReceiveStream::Config& config = receive_stream->config();
   ReceiveRtpConfig receive_config(config.rtp.extensions,
                                   UseSendSideBwe(config));
   {
@@ skipping 13 matching lines @@
   }
   receive_stream->SignalNetworkState(video_network_state_);
   UpdateAggregateNetworkState();
   event_log_->LogVideoReceiveStreamConfig(CreateRtcLogStreamConfig(config));
   return receive_stream;
 }
 
 void Call::DestroyVideoReceiveStream(
     webrtc::VideoReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   RTC_DCHECK(receive_stream != nullptr);
   VideoReceiveStream* receive_stream_impl =
       static_cast<VideoReceiveStream*>(receive_stream);
   const VideoReceiveStream::Config& config = receive_stream_impl->config();
   {
     WriteLockScoped write_lock(*receive_crit_);
     // Remove all ssrcs pointing to a receive stream. As RTX retransmits on a
     // separate SSRC there can be either one or two.
     size_t num_deleted = video_rtp_demuxer_.RemoveSink(receive_stream_impl);
     RTC_DCHECK_GE(num_deleted, 1);
     receive_rtp_config_.erase(config.rtp.remote_ssrc);
     if (config.rtp.rtx_ssrc) {
       receive_rtp_config_.erase(config.rtp.rtx_ssrc);
     }
     video_receive_streams_.erase(receive_stream_impl);
     ConfigureSync(config.sync_group);
   }
 
   receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config))
       ->RemoveStream(config.rtp.remote_ssrc);
 
   UpdateAggregateNetworkState();
   delete receive_stream_impl;
 }
 
 FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
     const FlexfecReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateFlexfecReceiveStream");
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
 
   RecoveredPacketReceiver* recovered_packet_receiver = this;
   FlexfecReceiveStreamImpl* receive_stream = new FlexfecReceiveStreamImpl(
       config, recovered_packet_receiver, call_stats_->rtcp_rtt_stats(),
       module_process_thread_.get());
 
   {
     WriteLockScoped write_lock(*receive_crit_);
     video_rtp_demuxer_.AddSink(config.remote_ssrc, receive_stream);
 
     for (auto ssrc : config.protected_media_ssrcs)
       video_rtp_demuxer_.AddSink(ssrc, receive_stream);
 
     RTC_DCHECK(receive_rtp_config_.find(config.remote_ssrc) ==
                receive_rtp_config_.end());
     receive_rtp_config_[config.remote_ssrc] =
         ReceiveRtpConfig(config.rtp_header_extensions, UseSendSideBwe(config));
   }
 
   // TODO(brandtr): Store config in RtcEventLog here.
 
   return receive_stream;
 }
 
 void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyFlexfecReceiveStream");
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
 
   RTC_DCHECK(receive_stream != nullptr);
   // There exist no other derived classes of FlexfecReceiveStream,
   // so this downcast is safe.
   FlexfecReceiveStreamImpl* receive_stream_impl =
       static_cast<FlexfecReceiveStreamImpl*>(receive_stream);
   {
     WriteLockScoped write_lock(*receive_crit_);
 
     const FlexfecReceiveStream::Config& config =
         receive_stream_impl->GetConfig();
     uint32_t ssrc = config.remote_ssrc;
     receive_rtp_config_.erase(ssrc);
 
     // Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be
     // destroyed.
     video_rtp_demuxer_.RemoveSink(receive_stream_impl);
     receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config))
         ->RemoveStream(ssrc);
   }
 
   delete receive_stream_impl;
 }
 
 Call::Stats Call::GetStats() const {
   // TODO(solenberg): Some test cases in EndToEndTest use this from a different
   // thread. Re-enable once that is fixed.
-  // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  // RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   Stats stats;
   // Fetch available send/receive bitrates.
   uint32_t send_bandwidth = 0;
   transport_send_->send_side_cc()->GetBitrateController()->AvailableBandwidth(
       &send_bandwidth);
   std::vector<unsigned int> ssrcs;
   uint32_t recv_bandwidth = 0;
   receive_side_cc_.GetRemoteBitrateEstimator(false)->LatestEstimate(
       &ssrcs, &recv_bandwidth);
   stats.send_bandwidth_bps = send_bandwidth;
   stats.recv_bandwidth_bps = recv_bandwidth;
   stats.pacer_delay_ms =
       transport_send_->send_side_cc()->GetPacerQueuingDelayMs();
   stats.rtt_ms = call_stats_->rtcp_rtt_stats()->LastProcessedRtt();
   {
     rtc::CritScope cs(&bitrate_crit_);
     stats.max_padding_bitrate_bps = configured_max_padding_bitrate_bps_;
   }
   return stats;
 }
 
 void Call::SetBitrateConfig(
     const webrtc::Call::Config::BitrateConfig& bitrate_config) {
   TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
   if (bitrate_config.max_bitrate_bps != -1)
     RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
   if (config_.bitrate_config.min_bitrate_bps ==
           bitrate_config.min_bitrate_bps &&
       (bitrate_config.start_bitrate_bps <= 0 ||
        config_.bitrate_config.start_bitrate_bps ==
            bitrate_config.start_bitrate_bps) &&
       config_.bitrate_config.max_bitrate_bps ==
           bitrate_config.max_bitrate_bps) {
     // Nothing new to set, early abort to avoid encoder reconfigurations.
     return;
   }
   config_.bitrate_config.min_bitrate_bps = bitrate_config.min_bitrate_bps;
   // Start bitrate of -1 means we should keep the old bitrate, which there is
   // no point in remembering for the future.
   if (bitrate_config.start_bitrate_bps > 0)
     config_.bitrate_config.start_bitrate_bps = bitrate_config.start_bitrate_bps;
   config_.bitrate_config.max_bitrate_bps = bitrate_config.max_bitrate_bps;
   RTC_DCHECK_NE(bitrate_config.start_bitrate_bps, 0);
   transport_send_->send_side_cc()->SetBweBitrates(
       bitrate_config.min_bitrate_bps, bitrate_config.start_bitrate_bps,
       bitrate_config.max_bitrate_bps);
 }
 
 void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   switch (media) {
     case MediaType::AUDIO:
       audio_network_state_ = state;
       break;
     case MediaType::VIDEO:
       video_network_state_ = state;
       break;
     case MediaType::ANY:
     case MediaType::DATA:
       RTC_NOTREACHED();
@@ skipping 41 matching lines @@
     case MediaType::ANY:
     case MediaType::DATA:
       RTC_NOTREACHED();
       break;
   }
 }
 
 // TODO(honghaiz): Add tests for this method.
 void Call::OnNetworkRouteChanged(const std::string& transport_name,
                                  const rtc::NetworkRoute& network_route) {
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   // Check if the network route is connected.
   if (!network_route.connected) {
     LOG(LS_INFO) << "Transport " << transport_name << " is disconnected";
     // TODO(honghaiz): Perhaps handle this in SignalChannelNetworkState and
     // consider merging these two methods.
     return;
   }
 
   // Check whether the network route has changed on each transport.
   auto result =
@@ skipping 16 matching lines @@
                  << " bps.";
     RTC_DCHECK_GT(config_.bitrate_config.start_bitrate_bps, 0);
     transport_send_->send_side_cc()->OnNetworkRouteChanged(
         network_route, config_.bitrate_config.start_bitrate_bps,
         config_.bitrate_config.min_bitrate_bps,
         config_.bitrate_config.max_bitrate_bps);
   }
 }
 
 void Call::UpdateAggregateNetworkState() {
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
 
   bool have_audio = false;
   bool have_video = false;
   {
     ReadLockScoped read_lock(*send_crit_);
     if (audio_send_ssrcs_.size() > 0)
       have_audio = true;
     if (video_send_ssrcs_.size() > 0)
       have_video = true;
   }
@@ skipping 266 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc