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

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index e21b0762fe8429f21b87b38916c5c5b772ddb119..919bec54f713a2e6e91c8375b980e9420f2adafe 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -60,34 +60,6 @@ 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,
@@ -172,12 +144,8 @@ class Call : public webrtc::Call,
       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_);
@@ -218,30 +186,6 @@ class Call : public webrtc::Call,
   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_);
@@ -395,29 +339,6 @@ Call::~Call() {
   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",
@@ -561,8 +482,6 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     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);
   }
@@ -589,7 +508,8 @@ void Call::DestroyAudioReceiveStream(
     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);
@@ -600,7 +520,6 @@ void Call::DestroyAudioReceiveStream(
       sync_stream_mapping_.erase(it);
       ConfigureSync(sync_group);
     }
-    receive_rtp_config_.erase(ssrc);
   }
   UpdateAggregateNetworkState();
   delete audio_receive_stream;
@@ -693,8 +612,6 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
       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) ==
@@ -702,13 +619,7 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
     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);
   }
@@ -734,7 +645,6 @@ void Call::DestroyVideoReceiveStream(
         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;
@@ -746,7 +656,8 @@ void Call::DestroyVideoReceiveStream(
   }
   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();
@@ -776,11 +687,6 @@ FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
     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.
@@ -803,7 +709,6 @@ void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
     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.
@@ -822,7 +727,8 @@ void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
         ++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);
@@ -1180,69 +1086,71 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
                                                 const PacketTime& packet_time) {
   TRACE_EVENT0("webrtc", "Call::DeliverRtp");
 
-  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)
+  RtpPacketReceived parsed_packet;
+  if (!parsed_packet.Parse(packet, length))
     return DELIVERY_PACKET_ERROR;
+  uint32_t ssrc = parsed_packet.Ssrc();
 
-  NotifyBweOfReceivedPacket(*parsed_packet, media_type);
+  ReadLockScoped read_lock(*receive_crit_);
 
-  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);
+      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?

       }
-      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;
-      }
-    }
+  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 DELIVERY_UNKNOWN_SSRC;
+
+  return success ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverPacket(
@@ -1272,11 +1180,8 @@ bool Call::OnRecoveredPacket(const uint8_t* packet, size_t 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);