Move RtcEventLog object from inside VoiceEngine to 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.
  */
 
@@ skipping 77 matching lines @@
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
 
   void SignalChannelNetworkState(MediaType media, NetworkState state) override;
 
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
   // Implements BitrateObserver.
   void OnNetworkChanged(uint32_t bitrate_bps, uint8_t fraction_loss,
                         int64_t rtt_ms) override;
 
+  bool StartEventLog(rtc::PlatformFile log_file,
+                     int64_t max_size_bytes) override {
+    return event_log_->StartLogging(log_file, max_size_bytes);
+  }
+
+  void StopEventLog() override { event_log_->StopLogging(); }
+
  private:
   DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
                              size_t length);
   DeliveryStatus DeliverRtp(MediaType media_type,
                             const uint8_t* packet,
                             size_t length,
                             const PacketTime& packet_time);
 
   void ConfigureSync(const std::string& sync_group)
       EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
@@ skipping 36 matching lines @@
       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_;
 
-  RtcEventLog* event_log_ = nullptr;
+  std::unique_ptr<webrtc::RtcEventLog> event_log_;
 
   // The following members are only accessed (exclusively) from one thread and
   // from the destructor, and therefore doesn't need any explicit
   // synchronization.
   int64_t received_video_bytes_;
   int64_t received_audio_bytes_;
   int64_t received_rtcp_bytes_;
   int64_t first_rtp_packet_received_ms_;
   int64_t last_rtp_packet_received_ms_;
   int64_t first_packet_sent_ms_;
@@ skipping 24 matching lines @@
       module_process_thread_(
           rtc::ScopedToUnique(ProcessThread::Create("ModuleProcessThread"))),
       pacer_thread_(rtc::ScopedToUnique(ProcessThread::Create("PacerThread"))),
       call_stats_(new CallStats(clock_)),
       bitrate_allocator_(new BitrateAllocator()),
       config_(config),
       audio_network_state_(kNetworkUp),
       video_network_state_(kNetworkUp),
       receive_crit_(RWLockWrapper::CreateRWLock()),
       send_crit_(RWLockWrapper::CreateRWLock()),
+      event_log_(RtcEventLog::Create()),
       received_video_bytes_(0),
       received_audio_bytes_(0),
       received_rtcp_bytes_(0),
       first_rtp_packet_received_ms_(-1),
       last_rtp_packet_received_ms_(-1),
       first_packet_sent_ms_(-1),
       estimated_send_bitrate_sum_kbits_(0),
       pacer_bitrate_sum_kbits_(0),
       num_bitrate_updates_(0),
       remb_(clock_),
-      congestion_controller_(new CongestionController(clock_, this, &remb_)) {
+      congestion_controller_(
+          new CongestionController(clock_, this, &remb_, event_log_.get())) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
                 config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
                   config.bitrate_config.start_bitrate_bps);
   }
-  if (config.audio_state.get()) {
-    ScopedVoEInterface<VoECodec> voe_codec(voice_engine());
-    event_log_ = voe_codec->GetEventLog();
-  }
 
   Trace::CreateTrace();
   call_stats_->RegisterStatsObserver(congestion_controller_.get());
 
   congestion_controller_->SetBweBitrates(
       config_.bitrate_config.min_bitrate_bps,
       config_.bitrate_config.start_bitrate_bps,
       config_.bitrate_config.max_bitrate_bps);
-  congestion_controller_->GetBitrateController()->SetEventLog(event_log_);
 
   module_process_thread_->Start();
   module_process_thread_->RegisterModule(call_stats_.get());
   module_process_thread_->RegisterModule(congestion_controller_.get());
   pacer_thread_->RegisterModule(congestion_controller_->pacer());
   pacer_thread_->RegisterModule(
       congestion_controller_->GetRemoteBitrateEstimator(true));
   pacer_thread_->Start();
 }
 
@@ skipping 107 matching lines @@
     RTC_DCHECK(num_deleted == 1);
   }
   UpdateAggregateNetworkState();
   delete audio_send_stream;
 }
 
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
-  AudioReceiveStream* receive_stream = new AudioReceiveStream(
-      congestion_controller_.get(), config, config_.audio_state);
+  AudioReceiveStream* receive_stream =
+      new AudioReceiveStream(congestion_controller_.get(), config,
+                             config_.audio_state, event_log_.get());
   {
     WriteLockScoped write_lock(*receive_crit_);
     RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
                audio_receive_ssrcs_.end());
     audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
     ConfigureSync(config.sync_group);
   }
   receive_stream->SignalNetworkState(audio_network_state_);
   UpdateAggregateNetworkState();
   return receive_stream;
@@ skipping 38 matching lines @@
   {
     WriteLockScoped write_lock(*send_crit_);
     for (uint32_t ssrc : config.rtp.ssrcs) {
       RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
       video_send_ssrcs_[ssrc] = send_stream;
     }
     video_send_streams_.insert(send_stream);
   }
   send_stream->SignalNetworkState(video_network_state_);
   UpdateAggregateNetworkState();
-  if (event_log_)
-    event_log_->LogVideoSendStreamConfig(config);
+  event_log_->LogVideoSendStreamConfig(config);
   return send_stream;
 }
 
 void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
   RTC_DCHECK(send_stream != nullptr);
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   send_stream->Stop();
 
@@ skipping 36 matching lines @@
     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();
     if (it != config.rtp.rtx.end())
       video_receive_ssrcs_[it->second.ssrc] = receive_stream;
     video_receive_streams_.insert(receive_stream);
-
     ConfigureSync(config.sync_group);
   }
   receive_stream->SignalNetworkState(video_network_state_);
   UpdateAggregateNetworkState();
-  if (event_log_)
-    event_log_->LogVideoReceiveStreamConfig(config);
+  event_log_->LogVideoReceiveStreamConfig(config);
   return receive_stream;
 }
 
 void Call::DestroyVideoReceiveStream(
     webrtc::VideoReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(receive_stream != nullptr);
   VideoReceiveStream* receive_stream_impl = nullptr;
   {
@@ skipping 226 matching lines @@
   //             Do NOT broadcast! Also make sure it's a valid packet.
   //             Return DELIVERY_UNKNOWN_SSRC if it can be determined that
   //             there's no receiver of the packet.
   received_rtcp_bytes_ += length;
   bool rtcp_delivered = false;
   if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
     ReadLockScoped read_lock(*receive_crit_);
     for (VideoReceiveStream* stream : video_receive_streams_) {
       if (stream->DeliverRtcp(packet, length)) {
         rtcp_delivered = true;
-        if (event_log_)
-          event_log_->LogRtcpPacket(kIncomingPacket, media_type, packet,
-                                    length);
+        event_log_->LogRtcpPacket(kIncomingPacket, media_type, packet, length);
       }
     }
   }
   if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
     ReadLockScoped read_lock(*send_crit_);
     for (VideoSendStream* stream : video_send_streams_) {
       if (stream->DeliverRtcp(packet, length)) {
         rtcp_delivered = true;
-        if (event_log_)
-          event_log_->LogRtcpPacket(kIncomingPacket, media_type, packet,
-                                    length);
+        event_log_->LogRtcpPacket(kIncomingPacket, media_type, packet, length);
       }
     }
   }
   return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
                                                 const uint8_t* packet,
                                                 size_t length,
                                                 const PacketTime& packet_time) {
   TRACE_EVENT0("webrtc", "Call::DeliverRtp");
   // Minimum RTP header size.
   if (length < 12)
     return DELIVERY_PACKET_ERROR;
 
   last_rtp_packet_received_ms_ = clock_->TimeInMilliseconds();
   if (first_rtp_packet_received_ms_ == -1)
     first_rtp_packet_received_ms_ = last_rtp_packet_received_ms_;
 
   uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
   ReadLockScoped read_lock(*receive_crit_);
   if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
     auto it = audio_receive_ssrcs_.find(ssrc);
     if (it != audio_receive_ssrcs_.end()) {
       received_audio_bytes_ += length;
       auto status = it->second->DeliverRtp(packet, length, packet_time)
                         ? DELIVERY_OK
                         : DELIVERY_PACKET_ERROR;
-      if (status == DELIVERY_OK && event_log_)
+      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 = video_receive_ssrcs_.find(ssrc);
     if (it != video_receive_ssrcs_.end()) {
       received_video_bytes_ += length;
       auto status = it->second->DeliverRtp(packet, length, packet_time)
                         ? DELIVERY_OK
                         : DELIVERY_PACKET_ERROR;
-      if (status == DELIVERY_OK && event_log_)
+      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,
     const uint8_t* packet,
     size_t length,
     const PacketTime& packet_time) {
   // TODO(solenberg): Tests call this function on a network thread, libjingle
   // calls on the worker thread. We should move towards always using a network
   // 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);
 }
 
 }  // namespace internal
 }  // namespace webrtc