Inform jitter buffer about FlexFEC protection.

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.
  */
 
+#include <stdint.h>
 #include <string.h>
+
 #include <algorithm>
 #include <map>
 #include <memory>
 #include <set>
 #include <utility>
 #include <vector>
 
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
@@ skipping 37 matching lines @@
 
 namespace webrtc {
 
 const int Call::Config::kDefaultStartBitrateBps = 300000;
 
 namespace internal {
 
 class Call : public webrtc::Call,
              public PacketReceiver,
              public RecoveredPacketReceiver,
+             public FlexfecProtectionMediator,
              public CongestionController::Observer,
              public BitrateAllocator::LimitObserver {
  public:
   explicit Call(const Call::Config& config);
   virtual ~Call();
 
   // Implements webrtc::Call.
   PacketReceiver* Receiver() override;
 
   webrtc::AudioSendStream* CreateAudioSendStream(
@@ skipping 24 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;
 
+  // Implements FlexfecProtectionMediator.
+  bool IsProtectedByFlexfec(uint32_t ssrc) 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;
 
@@ skipping 515 matching lines @@
   delete send_stream_impl;
 }
 
 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
     webrtc::VideoReceiveStream::Config configuration) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   VideoReceiveStream* receive_stream = new VideoReceiveStream(
       num_cpu_cores_, congestion_controller_.get(), &packet_router_,
       std::move(configuration), voice_engine(), module_process_thread_.get(),
-      call_stats_.get(), &remb_);
+      call_stats_.get(), &remb_, this /* flexfec_protection_mediator */);
 
   const webrtc::VideoReceiveStream::Config& config = receive_stream->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;
     // TODO(pbos): Configure different RTX payloads per receive payload.
     VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it =
         config.rtp.rtx.begin();
@@ skipping 538 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);
 }
 
+bool Call::IsProtectedByFlexfec(uint32_t ssrc) {
+  ReadLockScoped read_lock(*receive_crit_);
+  return flexfec_receive_ssrcs_media_.find(ssrc) !=
+         flexfec_receive_ssrcs_media_.end();
+}
+
 void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet) {
   RTPHeader header;
   packet.GetHeader(&header);
   congestion_controller_->OnReceivedPacket(packet.arrival_time_ms(),
                                            packet.payload_size(), header);
 }
 
 }  // namespace internal
 }  // namespace webrtc