Move RemoteBitrateEstimator::RemoveStream calls from receive streams to 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 42 matching lines @@
 #include "webrtc/video/send_delay_stats.h"
 #include "webrtc/video/stats_counter.h"
 #include "webrtc/video/video_receive_stream.h"
 #include "webrtc/video/video_send_stream.h"
 #include "webrtc/video/vie_remb.h"
 
 namespace webrtc {
 
 const int Call::Config::kDefaultStartBitrateBps = 300000;
 
+namespace {
+
+// TODO(nisse): This really begs for a shared context struct.
+bool UseSendSideBwe(const std::vector<RtpExtension>& extensions,
+                    bool transport_cc) {
+  if (!transport_cc)
+    return false;
+  for (const auto& extension : extensions) {
+    if (extension.uri == RtpExtension::kTransportSequenceNumberUri)
+      return true;
+  }
+  return false;
+}
+
+bool UseSendSideBwe(const VideoReceiveStream::Config& config) {
+  return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc);
+}
+
+bool UseSendSideBwe(const AudioReceiveStream::Config& config) {
+  return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc);
+}
+
+bool UseSendSideBwe(const FlexfecReceiveStream::Config& config) {
+  return UseSendSideBwe(config.rtp_header_extensions, config.transport_cc);
+}
+
+}  // namespace
+
 namespace internal {
 
 class Call : public webrtc::Call,
              public PacketReceiver,
              public RecoveredPacketReceiver,
              public CongestionController::Observer,
              public BitrateAllocator::LimitObserver {
  public:
   explicit Call(const Call::Config& config);
   virtual ~Call();
@@ skipping 119 matching lines @@
 
   // This extra map is used for receive processing which is
   // independent of media type.
 
   // TODO(nisse): In the RTP transport refactoring, we should have a
   // single mapping from ssrc to a more abstract receive stream, with
   // accessor methods for all configuration we need at this level.
   struct ReceiveRtpConfig {
     ReceiveRtpConfig() = default;  // Needed by std::map
     ReceiveRtpConfig(const std::vector<RtpExtension>& extensions,
-                     bool transport_cc)
-        : extensions(extensions), transport_cc(transport_cc) {}
+                     bool use_send_side_bwe)
+        : extensions(extensions), use_send_side_bwe(use_send_side_bwe) {}
 
     // Registered RTP header extensions for each stream. Note that RTP header
     // extensions are negotiated per track ("m= line") in the SDP, but we have
     // no notion of tracks at the Call level. We therefore store the RTP header
     // extensions per SSRC instead, which leads to some storage overhead.
     RtpHeaderExtensionMap extensions;
-    // Set if the RTCP feedback message needed for send side BWE was negotiated.
-    bool transport_cc = false;
+    // Set if both RTP extension the RTCP feedback message needed for
+    // send side BWE are negotiated.
+    bool use_send_side_bwe = false;
   };
   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_);
 
@@ skipping 296 matching lines @@
   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());
   event_log_->LogAudioReceiveStreamConfig(config);
   AudioReceiveStream* receive_stream = new AudioReceiveStream(
-      &packet_router_,
-      congestion_controller_->GetRemoteBitrateEstimator(true), config,
+      &packet_router_, config,
       config_.audio_state, event_log_);
   {
     WriteLockScoped write_lock(*receive_crit_);
     RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
                audio_receive_ssrcs_.end());
     audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
     receive_rtp_config_[config.rtp.remote_ssrc] =
-        ReceiveRtpConfig(config.rtp.extensions, config.rtp.transport_cc);
+        ReceiveRtpConfig(config.rtp.extensions, UseSendSideBwe(config));
 
     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(receive_stream != nullptr);
   webrtc::internal::AudioReceiveStream* audio_receive_stream =
       static_cast<webrtc::internal::AudioReceiveStream*>(receive_stream);
   {
     WriteLockScoped write_lock(*receive_crit_);
-    uint32_t ssrc = audio_receive_stream->config().rtp.remote_ssrc;
-
+    const AudioReceiveStream::Config& config = audio_receive_stream->config();
+    uint32_t ssrc = config.rtp.remote_ssrc;
+    congestion_controller_->GetRemoteBitrateEstimator(UseSendSideBwe(config))
+        ->RemoveStream(ssrc);
     size_t num_deleted = audio_receive_ssrcs_.erase(ssrc);
     RTC_DCHECK(num_deleted == 1);
     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);
@@ skipping 77 matching lines @@
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   bool protected_by_flexfec = false;
   {
     ReadLockScoped read_lock(*receive_crit_);
     protected_by_flexfec =
         flexfec_receive_ssrcs_media_.find(configuration.rtp.remote_ssrc) !=
         flexfec_receive_ssrcs_media_.end();
   }
   VideoReceiveStream* receive_stream = new VideoReceiveStream(
-      num_cpu_cores_, protected_by_flexfec, congestion_controller_.get(),
+      num_cpu_cores_, protected_by_flexfec,
       &packet_router_, std::move(configuration), module_process_thread_.get(),
       call_stats_.get(), &remb_);
 
   const webrtc::VideoReceiveStream::Config& config = receive_stream->config();
   ReceiveRtpConfig receive_config(config.rtp.extensions,
-                                  config.rtp.transport_cc);
+                                  UseSendSideBwe(config));
   {
     WriteLockScoped write_lock(*receive_crit_);
     RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
                video_receive_ssrcs_.end());
     video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
     if (config.rtp.rtx_ssrc) {
       video_receive_ssrcs_[config.rtp.rtx_ssrc] = receive_stream;
       // We record identical config for the rtx stream as for the main
       // stream. Since the transport_cc negotiation is per payload
       // type, we may get an incorrect value for the rtx stream, but
@@ skipping 29 matching lines @@
         receive_rtp_config_.erase(it->first);
         it = video_receive_ssrcs_.erase(it);
       } else {
         ++it;
       }
     }
     video_receive_streams_.erase(receive_stream_impl);
     RTC_CHECK(receive_stream_impl != nullptr);
     ConfigureSync(receive_stream_impl->config().sync_group);
   }
+  const VideoReceiveStream::Config& config = receive_stream_impl->config();
+
+  congestion_controller_->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());
 
   RecoveredPacketReceiver* recovered_packet_receiver = this;
@@ skipping 11 matching lines @@
     for (auto ssrc : config.protected_media_ssrcs)
       flexfec_receive_ssrcs_media_.insert(std::make_pair(ssrc, receive_stream));
 
     RTC_DCHECK(flexfec_receive_ssrcs_protection_.find(config.remote_ssrc) ==
                flexfec_receive_ssrcs_protection_.end());
     flexfec_receive_ssrcs_protection_[config.remote_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, config.transport_cc);
+        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(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_);
 
-    uint32_t ssrc = receive_stream_impl->GetConfig().remote_ssrc;
+    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.
     auto prot_it = flexfec_receive_ssrcs_protection_.begin();
     while (prot_it != flexfec_receive_ssrcs_protection_.end()) {
       if (prot_it->second == receive_stream_impl)
         prot_it = flexfec_receive_ssrcs_protection_.erase(prot_it);
       else
         ++prot_it;
     }
     auto media_it = flexfec_receive_ssrcs_media_.begin();
     while (media_it != flexfec_receive_ssrcs_media_.end()) {
       if (media_it->second == receive_stream_impl)
         media_it = flexfec_receive_ssrcs_media_.erase(media_it);
       else
         ++media_it;
     }
 
+    congestion_controller_->GetRemoteBitrateEstimator(UseSendSideBwe(config))
+        ->RemoveStream(ssrc);
+
     flexfec_receive_streams_.erase(receive_stream_impl);
   }
 
   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());
@@ skipping 429 matching lines @@
   ReadLockScoped read_lock(*receive_crit_);
   auto it = video_receive_ssrcs_.find(ssrc);
   if (it == video_receive_ssrcs_.end())
     return false;
   return it->second->OnRecoveredPacket(packet, length);
 }
 
 void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
                                      MediaType media_type) {
   auto it = receive_rtp_config_.find(packet.Ssrc());
-  bool transport_cc =
-      (it != receive_rtp_config_.end()) && it->second.transport_cc;
+  bool use_send_side_bwe =
+      (it != receive_rtp_config_.end()) && it->second.use_send_side_bwe;
 
   RTPHeader header;
   packet.GetHeader(&header);
 
-  if (!transport_cc && header.extension.hasTransportSequenceNumber) {
+  if (!use_send_side_bwe && header.extension.hasTransportSequenceNumber) {
     // Inconsistent configuration of send side BWE. Do nothing.
     // TODO(nisse): Without this check, we may produce RTCP feedback
     // packets even when not negotiated. But it would be cleaner to
     // move the check down to RTCPSender::SendFeedbackPacket, which
     // would also help the PacketRouter to select an appropriate rtp
     // module in the case that some, but not all, have RTCP feedback
     // enabled.
     return;
   }
   // For audio, we only support send side BWE.
   // TODO(nisse): Tests passes MediaType::ANY, see
   // FakeNetworkPipe::Process. We need to treat that as video. Tests
   // should be fixed to use the same MediaType as the production code.
   if (media_type != MediaType::AUDIO ||
-      (transport_cc && header.extension.hasTransportSequenceNumber)) {
+      (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     congestion_controller_->OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 }  // namespace webrtc