Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call.

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 6aa564e95e96694eadb562aa93fd18917f9fe93e..37b5c6a4fbf5fef7683d9b6eea3f6e08c1b7fff6 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -109,8 +109,6 @@
   // 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;
 
@@ -144,6 +142,9 @@
                             const PacketTime& packet_time);
   void ConfigureSync(const std::string& sync_group)
       EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
+
+  void NotifyBweOfReceivedPacket(const RtpPacketReceived& packet)
+      SHARED_LOCKS_REQUIRED(receive_crit_);
 
   rtc::Optional<RtpPacketReceived> ParseRtpPacket(const uint8_t* packet,
                                                   size_t length,
@@ -188,12 +189,27 @@
   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_;
@@ -357,9 +373,9 @@
   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) {
@@ -509,7 +525,6 @@
   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_);
   {
@@ -517,6 +532,9 @@
     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);
   }
   {
@@ -540,8 +558,9 @@
       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);
@@ -550,6 +569,7 @@
       sync_stream_mapping_.erase(it);
       ConfigureSync(sync_group);
     }
+    receive_rtp_config_.erase(ssrc);
   }
   UpdateAggregateNetworkState();
   delete audio_receive_stream;
@@ -642,13 +662,22 @@
       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);
   }
@@ -674,7 +703,8 @@
         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;
       }
@@ -711,10 +741,10 @@
                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.
@@ -735,7 +765,7 @@
     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.
@@ -1108,12 +1138,20 @@
                                                 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);
+
+  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()) {
@@ -1140,8 +1178,6 @@
       // 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)
@@ -1155,10 +1191,7 @@
   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;
@@ -1198,8 +1231,21 @@
 }
 
 void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet) {
+  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);
+
+  // transport_cc represents the negotiation of the RTCP feedback
+  // message used for send side BWE. If it was negotiated but the
+  // corresponding RTP header extension is not present, or vice versa,
+  // bandwidth estimation is not correctly configured.
+  if (transport_cc != header.extension.hasTransportSequenceNumber) {
+    LOG(LS_ERROR) << "Inconsistent configuration of send side BWE.";
+    return;
+  }
   congestion_controller_->OnReceivedPacket(packet.arrival_time_ms(),

[nisse-webrtc, 2017/02/02 09:53:39]

webrtc_perf_tests crashes in the first test
(CallPerfTest.PlaysOutAudioAndVideoInSyncWithVideoNtpDrift), because this call
is directed to RemoteBitrateEstimatorAbsSendTime::IncomingPacket, where the
DCHECK

  RTC_DCHECK(network_thread_.CalledOnValidThread());

fails.

                                            packet.payload_size(), header);
 }