Reland of Always call RemoteBitrateEstimator::IncomingPacket from 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 91 matching lines @@
 
   // Implements PacketReceiver.
   DeliveryStatus DeliverPacket(MediaType media_type,
                                const uint8_t* packet,
                                size_t length,
                                const PacketTime& packet_time) override;
 
   // Implements RecoveredPacketReceiver.
   bool OnRecoveredPacket(const uint8_t* packet, size_t length) override;
 
-  void NotifyBweOfReceivedPacket(const RtpPacketReceived& packet);
-
   void SetBitrateConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
 
   void SignalChannelNetworkState(MediaType media, NetworkState state) override;
 
   void OnTransportOverheadChanged(MediaType media,
                                   int transport_overhead_per_packet) override;
 
   void OnNetworkRouteChanged(const std::string& transport_name,
                              const rtc::NetworkRoute& network_route) override;
@@ skipping 14 matching lines @@
  private:
   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)
       SHARED_LOCKS_REQUIRED(receive_crit_);
 
   void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
   void UpdateReceiveHistograms();
   void UpdateHistograms();
   void UpdateAggregateNetworkState();
 
@@ skipping 23 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_);
 
-  // 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.
-  std::map<uint32_t, RtpHeaderExtensionMap> received_rtp_header_extensions_
+  // 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) {}
+
+    // 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;
+  };
+  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_;
@@ skipping 143 matching lines @@
 }
 
 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 = received_rtp_header_extensions_.find(parsed_packet.Ssrc());
-  if (it != received_rtp_header_extensions_.end())
-    parsed_packet.IdentifyExtensions(it->second);
+  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));
@@ skipping 129 matching lines @@
   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_,
-      // TODO(nisse): Used only when UseSendSideBwe(config) is true.
       congestion_controller_->GetRemoteBitrateEstimator(true), 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);
+
     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_);
-    size_t num_deleted = audio_receive_ssrcs_.erase(
-        audio_receive_stream->config().rtp.remote_ssrc);
+    uint32_t ssrc = audio_receive_stream->config().rtp.remote_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, congestion_controller_.get(),
       &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);
   {
     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)
+    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;
-        video_receive_ssrcs_.erase(it++);
+        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);
   }
   UpdateAggregateNetworkState();
   delete receive_stream_impl;
@@ skipping 16 matching lines @@
                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(received_rtp_header_extensions_.find(config.remote_ssrc) ==
-               received_rtp_header_extensions_.end());
-    RtpHeaderExtensionMap rtp_header_extensions(config.rtp_header_extensions);
-    received_rtp_header_extensions_[config.remote_ssrc] = rtp_header_extensions;
+    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);
   }
 
   // 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;
-    received_rtp_header_extensions_.erase(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;
     }
@@ skipping 352 matching lines @@
     event_log_->LogRtcpPacket(kIncomingPacket, media_type, packet, length);
 
   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");
-  // Minimum RTP header size.
-  if (length < 12)
+
+  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;
 
-  uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
-  ReadLockScoped read_lock(*receive_crit_);
+  NotifyBweOfReceivedPacket(*parsed_packet, media_type);
+
+  uint32_t ssrc = parsed_packet->Ssrc();
+
   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;
     }
   }
   if (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.
-      rtc::Optional<RtpPacketReceived> parsed_packet =
-          ParseRtpPacket(packet, length, packet_time);
       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()) {
-      rtc::Optional<RtpPacketReceived> parsed_packet =
-          ParseRtpPacket(packet, length, packet_time);
       if (parsed_packet) {
-        NotifyBweOfReceivedPacket(*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;
       }
     }
   }
   return DELIVERY_UNKNOWN_SSRC;
@@ skipping 18 matching lines @@
 // 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) {
+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;
+
   RTPHeader header;
   packet.GetHeader(&header);
-  congestion_controller_->OnReceivedPacket(packet.arrival_time_ms(),
-                                           packet.payload_size(), header);
+
+  if (!transport_cc && header.extension.hasTransportSequenceNumber)

[brandtr, 2017/02/03 09:50:29]

Add {}.

[nisse-webrtc, 2017/02/03 10:19:47]

Done.

+    // 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))

[brandtr, 2017/02/03 09:50:29]

{}

[nisse-webrtc, 2017/02/03 10:19:47]

Done.

+    congestion_controller_->OnReceivedPacket(packet.arrival_time_ms(),
+                                             packet.payload_size(), header);
 }
 
 }  // namespace internal
 }  // namespace webrtc