Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket.

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 119 matching lines @@
 
   Stats GetStats() const override;
 
   // 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 OnRecoveredPacket(const uint8_t* packet, size_t length) override;
 
   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,
@@ skipping 21 matching lines @@
                             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)
+                                                  const PacketTime* packet_time)
       SHARED_LOCKS_REQUIRED(receive_crit_);
 
   void UpdateSendHistograms(int64_t first_sent_packet_ms)
       EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
   void UpdateReceiveHistograms();
   void UpdateHistograms();
   void UpdateAggregateNetworkState();
 
   Clock* const clock_;
 
@@ skipping 209 matching lines @@
   }
   UpdateReceiveHistograms();
   UpdateHistograms();
 
   Trace::ReturnTrace();
 }
 
 rtc::Optional<RtpPacketReceived> Call::ParseRtpPacket(
     const uint8_t* packet,
     size_t length,
-    const PacketTime& packet_time) {
+    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;
+  if (packet_time && 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(
@@ skipping 748 matching lines @@
                                                 size_t length,
                                                 const PacketTime& packet_time) {
   TRACE_EVENT0("webrtc", "Call::DeliverRtp");
 
   RTC_DCHECK(media_type == MediaType::AUDIO || media_type == MediaType::VIDEO);
 
   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);
+      ParseRtpPacket(packet, length, &packet_time);
 
   if (!parsed_packet)
     return DELIVERY_PACKET_ERROR;
 
   NotifyBweOfReceivedPacket(*parsed_packet, media_type);
 
   uint32_t ssrc = parsed_packet->Ssrc();
 
   if (media_type == MediaType::AUDIO) {
     auto it = audio_receive_ssrcs_.find(ssrc);
@@ skipping 45 matching lines @@
   // 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]);

[brandtr, 2017/05/10 14:27:29]

Nice to get rid of this!

+void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {
   ReadLockScoped read_lock(*receive_crit_);
-  auto it = video_receive_ssrcs_.find(ssrc);
+  rtc::Optional<RtpPacketReceived> parsed_packet =
+      ParseRtpPacket(packet, length, nullptr);
+  if (!parsed_packet)
+    return;
+
+  parsed_packet->set_recovered(true);
+
+  auto it = video_receive_ssrcs_.find(parsed_packet->Ssrc());
   if (it == video_receive_ssrcs_.end())
-    return false;
-  return it->second->OnRecoveredPacket(packet, length);
+    return;
+
+  it->second->OnRtpPacket(*parsed_packet);
 }
 
 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;
 
   RTPHeader header;
   packet.GetHeader(&header);
@@ skipping 13 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc