Integrate FlexfecReceiveStream with 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.
  */
 
 #include <string.h>
 #include <algorithm>
 #include <map>
 #include <memory>
+#include <utility>
 #include <vector>
 
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call.h"
 #include "webrtc/call/bitrate_allocator.h"
+#include "webrtc/call/flexfec_receive_stream.h"
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/clock.h"
@@ skipping 18 matching lines @@
 
 class Call : public webrtc::Call,
              public PacketReceiver,
              public RecoveredPacketReceiver,
              public CongestionController::Observer,
              public BitrateAllocator::LimitObserver {
  public:
   explicit Call(const Call::Config& config);
   virtual ~Call();
 
+  // Implements webrtc::Call.
   PacketReceiver* Receiver() override;
 
   webrtc::AudioSendStream* CreateAudioSendStream(
       const webrtc::AudioSendStream::Config& config) override;
   void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
 
   webrtc::AudioReceiveStream* CreateAudioReceiveStream(
       const webrtc::AudioReceiveStream::Config& config) override;
   void DestroyAudioReceiveStream(
       webrtc::AudioReceiveStream* receive_stream) override;
 
   webrtc::VideoSendStream* CreateVideoSendStream(
       webrtc::VideoSendStream::Config config,
       VideoEncoderConfig encoder_config) override;
   void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
 
   webrtc::VideoReceiveStream* CreateVideoReceiveStream(
       webrtc::VideoReceiveStream::Config configuration) override;
   void DestroyVideoReceiveStream(
       webrtc::VideoReceiveStream* receive_stream) override;
 
+  webrtc::FlexfecReceiveStream* CreateFlexfecReceiveStream(
+      webrtc::FlexfecReceiveStream::Config configuration) override;
+  void DestroyFlexfecReceiveStream(
+      webrtc::FlexfecReceiveStream* receive_stream) override;
+
   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 SetBitrateConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
@@ skipping 51 matching lines @@
   NetworkState video_network_state_;
 
   std::unique_ptr<RWLockWrapper> receive_crit_;
   // Audio and Video receive streams are owned by the client that creates them.
   std::map<uint32_t, AudioReceiveStream*> audio_receive_ssrcs_
       GUARDED_BY(receive_crit_);
   std::map<uint32_t, VideoReceiveStream*> video_receive_ssrcs_
       GUARDED_BY(receive_crit_);
   std::set<VideoReceiveStream*> video_receive_streams_
       GUARDED_BY(receive_crit_);
+  std::multimap<uint32_t, FlexfecReceiveStream*> flexfec_receive_ssrcs_media_
+      GUARDED_BY(receive_crit_);
+  std::map<uint32_t, FlexfecReceiveStream*> flexfec_receive_ssrcs_protection_
+      GUARDED_BY(receive_crit_);
+  std::set<FlexfecReceiveStream*> flexfec_receive_streams_
+      GUARDED_BY(receive_crit_);
   std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
       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_;
@@ skipping 398 matching lines @@
       }
     }
     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;
 }
 
+webrtc::FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
+    webrtc::FlexfecReceiveStream::Config configuration) {
+  TRACE_EVENT0("webrtc", "Call::CreateFlexfecReceiveStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  FlexfecReceiveStream* receive_stream =
+      new FlexfecReceiveStream(std::move(configuration), this);
+
+  const webrtc::FlexfecReceiveStream::Config& config = receive_stream->config();
+  {
+    WriteLockScoped write_lock(*receive_crit_);
+    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.flexfec_ssrc) ==
+               flexfec_receive_ssrcs_protection_.end());
+    flexfec_receive_ssrcs_protection_[config.flexfec_ssrc] = receive_stream;
+    flexfec_receive_streams_.insert(receive_stream);
+  }
+  // TODO(brandtr): Store config in RtcEventLog here.
+  return receive_stream;
+}
+
+void Call::DestroyFlexfecReceiveStream(
+    webrtc::FlexfecReceiveStream* receive_stream) {
+  TRACE_EVENT0("webrtc", "Call::DestroyFlexfecReceiveStream");
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(receive_stream != nullptr);
+  FlexfecReceiveStream* receive_stream_impl = nullptr;
+  {
+    WriteLockScoped write_lock(*receive_crit_);
+    // Remove all ssrcs pointing to a receive stream.

[stefan-webrtc, 2016/10/18 18:13:29]

"...pointing to the FlexFec receive stream to be destroyed", right?

[brandtr, 2016/10/19 08:14:31]

Yep, fixed.

+    auto media_it = flexfec_receive_ssrcs_media_.begin();
+    while (media_it != flexfec_receive_ssrcs_media_.end()) {
+      if (media_it->second ==
+          static_cast<FlexfecReceiveStream*>(receive_stream)) {
+        if (receive_stream_impl != nullptr)
+          RTC_DCHECK(receive_stream_impl == media_it->second);
+        receive_stream_impl = media_it->second;
+        flexfec_receive_ssrcs_media_.erase(media_it++);
+      } else {
+        ++media_it;
+      }
+    }
+    auto prot_it = flexfec_receive_ssrcs_protection_.begin();
+    while (prot_it != flexfec_receive_ssrcs_protection_.end()) {
+      if (prot_it->second ==
+          static_cast<FlexfecReceiveStream*>(receive_stream)) {
+        if (receive_stream_impl != nullptr)
+          RTC_DCHECK(receive_stream_impl == prot_it->second);
+        receive_stream_impl = prot_it->second;
+        flexfec_receive_ssrcs_protection_.erase(prot_it++);
+      } else {
+        ++prot_it;
+      }
+    }
+    flexfec_receive_streams_.erase(receive_stream_impl);

[stefan-webrtc, 2016/10/18 18:13:29]

If flexfec_receive_streams_ own the receive stream and has unique_ptrs then I
think we can remove the delete below, or am I missing something?

[brandtr, 2016/10/19 08:14:31]

Yes, but Call will not own the created FlexfecReceiveStream. This is the same as
how VideoReceiveStream is handled; it is owned by WebRtcVideoChannel2, but the
Call functions are responsible for creating and destroying the objects. See
e.g.:
https://cs.chromium.org/chromium/src/third_party/webrtc/call/call.cc?l=518
https://cs.chromium.org/chromium/src/third_party/webrtc/call/call.cc?l=546
https://cs.chromium.org/chromium/src/third_party/webrtc/media/engine/webrtcvi...
https://cs.chromium.org/chromium/src/third_party/webrtc/media/engine/webrtcvi...

I think it would make sense to change all of the streams to use unique_ptr's, in
order to make the ownerships clear, but that is probably better to do in a
another CL.

+    RTC_CHECK(receive_stream_impl != nullptr);
+  }
+  delete receive_stream_impl;
+}
+
 Call::Stats Call::GetStats() const {
   // TODO(solenberg): Some test cases in EndToEndTest use this from a different
   // thread. Re-enable once that is fixed.
   // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   Stats stats;
   // Fetch available send/receive bitrates.
   uint32_t send_bandwidth = 0;
   congestion_controller_->GetBitrateController()->AvailableBandwidth(
       &send_bandwidth);
   std::vector<unsigned int> ssrcs;
@@ skipping 325 matching lines @@
     }
   }
   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));
       auto status = it->second->DeliverRtp(packet, length, packet_time)
                         ? DELIVERY_OK
                         : DELIVERY_PACKET_ERROR;
+      auto flexfec_lower_it = flexfec_receive_ssrcs_media_.lower_bound(ssrc);
+      auto flexfec_upper_it = flexfec_receive_ssrcs_media_.upper_bound(ssrc);
+      while (flexfec_lower_it != flexfec_upper_it) {
+        flexfec_lower_it->second->AddAndProcessReceivedPacket(packet, length);
+        ++flexfec_lower_it;
+      }
+      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()) {
+      auto status = it->second->AddAndProcessReceivedPacket(packet, length)
+                        ? DELIVERY_OK
+                        : DELIVERY_PACKET_ERROR;
       if (status == DELIVERY_OK)
         event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
       return status;
     }
   }
   return DELIVERY_UNKNOWN_SSRC;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverPacket(
     MediaType media_type,
@@ skipping 16 matching lines @@
   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 it->second->OnRecoveredPacket(packet, length);
   return false;
 }
 
 }  // namespace internal
 }  // namespace webrtc