RtpPacketReceiver base class and OnRtpPacket, with a pre-parsed RTP packet.

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 64 matching lines @@
   DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
                              size_t length);
   DeliveryStatus DeliverRtp(MediaType media_type,
                             const uint8_t* packet,
                             size_t length,
                             const PacketTime& packet_time);
   void ConfigureSync(const std::string& sync_group)
       EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
 
   void NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
-                                 MediaType media_type)
-      SHARED_LOCKS_REQUIRED(receive_crit_);
-
-  rtc::Optional<RtpPacketReceived> ParseRtpPacket(const uint8_t* packet,
-                                                  size_t length,
-                                                  const PacketTime& packet_time)
+                                 MediaType media_type,
+                                 bool use_send_side_bwe)
       SHARED_LOCKS_REQUIRED(receive_crit_);
 
   void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
   void UpdateReceiveHistograms();
   void UpdateHistograms();
   void UpdateAggregateNetworkState();
 
   Clock* const clock_;
 
   const int num_cpu_cores_;
@@ skipping 20 matching lines @@
   // streams.
   std::multimap<uint32_t, FlexfecReceiveStreamImpl*>
       flexfec_receive_ssrcs_media_ GUARDED_BY(receive_crit_);
   std::map<uint32_t, FlexfecReceiveStreamImpl*>
       flexfec_receive_ssrcs_protection_ GUARDED_BY(receive_crit_);
   std::set<FlexfecReceiveStreamImpl*> flexfec_receive_streams_
       GUARDED_BY(receive_crit_);
   std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
       GUARDED_BY(receive_crit_);
 
-  // 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 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 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_);
 
   VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
   webrtc::RtcEventLog* event_log_;
 
   // The following members are only accessed (exclusively) from one thread and
@@ skipping 133 matching lines @@
   {
     rtc::CritScope lock(&bitrate_crit_);
     UpdateSendHistograms();
   }
   UpdateReceiveHistograms();
   UpdateHistograms();
 
   Trace::ReturnTrace();
 }
 
-rtc::Optional<RtpPacketReceived> Call::ParseRtpPacket(
-    const uint8_t* packet,
-    size_t length,
-    const PacketTime& packet_time) {
-  RtpPacketReceived parsed_packet;
-  if (!parsed_packet.Parse(packet, length))
-    return rtc::Optional<RtpPacketReceived>();
-
-  auto it = receive_rtp_config_.find(parsed_packet.Ssrc());
-  if (it != receive_rtp_config_.end())
-    parsed_packet.IdentifyExtensions(it->second.extensions);
-
-  int64_t arrival_time_ms;
-  if (packet_time.timestamp != -1) {
-    arrival_time_ms = (packet_time.timestamp + 500) / 1000;
-  } else {
-    arrival_time_ms = clock_->TimeInMilliseconds();
-  }
-  parsed_packet.set_arrival_time_ms(arrival_time_ms);
-
-  return rtc::Optional<RtpPacketReceived>(std::move(parsed_packet));
-}
-
 void Call::UpdateHistograms() {
   RTC_HISTOGRAM_COUNTS_100000(
       "WebRTC.Call.LifetimeInSeconds",
       (clock_->TimeInMilliseconds() - start_ms_) / 1000);
 }
 
 void Call::UpdateSendHistograms() {
   if (first_packet_sent_ms_ == -1)
     return;
   int64_t elapsed_sec =
@@ skipping 123 matching lines @@
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   event_log_->LogAudioReceiveStreamConfig(config);
   AudioReceiveStream* receive_stream = new AudioReceiveStream(
       &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, 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_);
     const AudioReceiveStream::Config& config = audio_receive_stream->config();
     uint32_t ssrc = config.rtp.remote_ssrc;
-    congestion_controller_->GetRemoteBitrateEstimator(UseSendSideBwe(config))
+    congestion_controller_->GetRemoteBitrateEstimator(
+        audio_receive_stream->rtp_config().use_send_side_bwe)
         ->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);
   }
   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());
@@ skipping 72 matching lines @@
     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,
       &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,
-                                  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
-      // that is unlikely to matter in practice.
-      receive_rtp_config_[config.rtp.rtx_ssrc] = receive_config;
     }
-    receive_rtp_config_[config.rtp.remote_ssrc] = receive_config;
     video_receive_streams_.insert(receive_stream);
     ConfigureSync(config.sync_group);
   }
   receive_stream->SignalNetworkState(video_network_state_);
   UpdateAggregateNetworkState();
   event_log_->LogVideoReceiveStreamConfig(config);
   return receive_stream;
 }
 
 void Call::DestroyVideoReceiveStream(
     webrtc::VideoReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(receive_stream != nullptr);
   VideoReceiveStream* receive_stream_impl = nullptr;
   {
     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.
     auto it = video_receive_ssrcs_.begin();
     while (it != video_receive_ssrcs_.end()) {
       if (it->second == static_cast<VideoReceiveStream*>(receive_stream)) {
         if (receive_stream_impl != nullptr)
           RTC_DCHECK(receive_stream_impl == it->second);
         receive_stream_impl = it->second;
-        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))
+  congestion_controller_->GetRemoteBitrateEstimator(
+      receive_stream_impl->rtp_config().use_send_side_bwe)
       ->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;
   FlexfecReceiveStreamImpl* receive_stream = new FlexfecReceiveStreamImpl(
       config, recovered_packet_receiver, call_stats_->rtcp_rtt_stats(),
       module_process_thread_.get());
 
   {
     WriteLockScoped write_lock(*receive_crit_);
 
     RTC_DCHECK(flexfec_receive_streams_.find(receive_stream) ==
                flexfec_receive_streams_.end());
     flexfec_receive_streams_.insert(receive_stream);
 
     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, 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_);
 
     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))
+    congestion_controller_->GetRemoteBitrateEstimator(
+        receive_stream_impl->rtp_config().use_send_side_bwe)
         ->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
@@ skipping 337 matching lines @@
 
   return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
                                                 const uint8_t* packet,
                                                 size_t length,
                                                 const PacketTime& packet_time) {
   TRACE_EVENT0("webrtc", "Call::DeliverRtp");
 
+  RtpPacketReceived parsed_packet;
+  if (!parsed_packet.Parse(packet, length))
+    return DELIVERY_PACKET_ERROR;
+  uint32_t ssrc = parsed_packet.Ssrc();
+
   ReadLockScoped read_lock(*receive_crit_);
-  // TODO(nisse): We should parse the RTP header only here, and pass
-  // on parsed_packet to the receive streams.
-  rtc::Optional<RtpPacketReceived> parsed_packet =
-      ParseRtpPacket(packet, length, packet_time);
 
-  if (!parsed_packet)
-    return DELIVERY_PACKET_ERROR;
-
-  NotifyBweOfReceivedPacket(*parsed_packet, media_type);
-
-  uint32_t ssrc = parsed_packet->Ssrc();
+  // Look up receiver, so we can parse extensions properly.
+  RtpPacketReceiver* receiver = nullptr;
+  bool pass_to_flexfec = false;
 
   if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
     auto it = audio_receive_ssrcs_.find(ssrc);
     if (it != audio_receive_ssrcs_.end()) {
       received_bytes_per_second_counter_.Add(static_cast<int>(length));
       received_audio_bytes_per_second_counter_.Add(static_cast<int>(length));
-      auto status = it->second->DeliverRtp(packet, length, packet_time)
-                        ? DELIVERY_OK
-                        : DELIVERY_PACKET_ERROR;
-      if (status == DELIVERY_OK)
-        event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
-      return status;
+      receiver = it->second;
     }
   }
-  if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
+  if (!receiver &&
+      (media_type == MediaType::ANY || media_type == MediaType::VIDEO)) {
     auto it = video_receive_ssrcs_.find(ssrc);
     if (it != video_receive_ssrcs_.end()) {
       received_bytes_per_second_counter_.Add(static_cast<int>(length));
       received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
-      // TODO(brandtr): Notify the BWE of received media packets here.
-      auto status = it->second->DeliverRtp(packet, length, packet_time)
-                        ? DELIVERY_OK
-                        : DELIVERY_PACKET_ERROR;
-      // Deliver media packets to FlexFEC subsystem. RTP header extensions need
-      // not be parsed, as FlexFEC is oblivious to the semantic meaning of the
-      // packet contents beyond the 12 byte RTP base header. The BWE is fed
-      // information about these media packets from the regular media pipeline.
-      if (parsed_packet) {
-        auto it_bounds = flexfec_receive_ssrcs_media_.equal_range(ssrc);
-        for (auto it = it_bounds.first; it != it_bounds.second; ++it)
-          it->second->AddAndProcessReceivedPacket(*parsed_packet);
-      }
-      if (status == DELIVERY_OK)
-        event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
-      return status;
-    }
-  }
-  if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
-    auto it = flexfec_receive_ssrcs_protection_.find(ssrc);
-    if (it != flexfec_receive_ssrcs_protection_.end()) {
-      if (parsed_packet) {
-        auto status = it->second->AddAndProcessReceivedPacket(*parsed_packet)
-                          ? DELIVERY_OK
-                          : DELIVERY_PACKET_ERROR;
-        if (status == DELIVERY_OK)
-          event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
-        return status;
+      receiver = it->second;
+      pass_to_flexfec = true;

[brandtr, 2017/02/09 14:51:55]

maybe_pass_to_flexfec

+    } else {
+      auto it = flexfec_receive_ssrcs_protection_.find(ssrc);
+      if (it != flexfec_receive_ssrcs_protection_.end()) {
+        receiver = it->second;
+        // TODO(nisse): Update received_bytes_per_second_counter_ ?

[brandtr, 2017/02/09 14:51:55]

Yes! This is an omission, the counters should be updated for FlexFEC as well.

[nisse-webrtc, 2017/02/10 08:09:23]

That's what you're fixing in
https://codereview.webrtc.org/2684243002/, right?

       }
     }
   }
-  return DELIVERY_UNKNOWN_SSRC;
+  if (!receiver)
+    return DELIVERY_UNKNOWN_SSRC;
+
+  parsed_packet.IdentifyExtensions(receiver->rtp_config().extensions);
+  int64_t arrival_time_ms;
+  if (packet_time.timestamp != -1) {
+    arrival_time_ms = (packet_time.timestamp + 500) / 1000;
+  } else {
+    arrival_time_ms = clock_->TimeInMilliseconds();
+  }
+  parsed_packet.set_arrival_time_ms(arrival_time_ms);
+
+  NotifyBweOfReceivedPacket(parsed_packet, media_type,
+                            receiver->rtp_config().use_send_side_bwe);
+
+  bool success = receiver->OnRtpPacket(parsed_packet);
+  if (success)
+    event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
+
+  if (pass_to_flexfec) {
+    // Deliver media packets to FlexFEC subsystem. RTP header extensions need
+    // not be parsed, as FlexFEC is oblivious to the semantic meaning of the
+    // packet contents beyond the 12 byte RTP base header. The BWE is fed
+    // information about these media packets from the regular media pipeline.
+    auto it_bounds = flexfec_receive_ssrcs_media_.equal_range(ssrc);
+    for (auto it = it_bounds.first; it != it_bounds.second; ++it)
+      it->second->OnRtpPacket(parsed_packet);
+  }
+
+  return success ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverPacket(
     MediaType media_type,
     const uint8_t* packet,
     size_t length,
     const PacketTime& packet_time) {
   // 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(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 // TODO(brandtr): Update this member function when we support protecting
 // audio packets with FlexFEC.
 bool Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {
   uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
   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 use_send_side_bwe =
-      (it != receive_rtp_config_.end()) && it->second.use_send_side_bwe;
-
+                                     MediaType media_type,
+                                     bool use_send_side_bwe) {
   RTPHeader header;
   packet.GetHeader(&header);
 
   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 ||
       (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