Revert of Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index cbf9da7494f808fe568bcfdf20218abf5909d1f5..a6a79789418626dd275556f14bd7616d212fd2f6 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -35,7 +35,6 @@
 #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"
 #include "webrtc/system_wrappers/include/cpu_info.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/metrics.h"
@@ -108,13 +107,6 @@
   void OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
                                  uint32_t max_padding_bitrate_bps) 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);
@@ -170,7 +162,7 @@
 
   VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
 
-  std::unique_ptr<webrtc::RtcEventLog> event_log_;
+  RtcEventLog* event_log_ = nullptr;
 
   // The following members are only accessed (exclusively) from one thread and
   // from the destructor, and therefore doesn't need any explicit
@@ -218,7 +210,6 @@
       video_network_state_(kNetworkUp),
       receive_crit_(RWLockWrapper::CreateRWLock()),
       send_crit_(RWLockWrapper::CreateRWLock()),
-      event_log_(RtcEventLog::Create(webrtc::Clock::GetRealTimeClock())),
       received_video_bytes_(0),
       received_audio_bytes_(0),
       received_rtcp_bytes_(0),
@@ -230,8 +221,7 @@
       min_allocated_send_bitrate_bps_(0),
       num_bitrate_updates_(0),
       remb_(clock_),
-      congestion_controller_(
-          new CongestionController(clock_, this, &remb_, event_log_.get())),
+      congestion_controller_(new CongestionController(clock_, this, &remb_)),
       video_send_delay_stats_(new SendDelayStats(clock_)) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
@@ -241,6 +231,10 @@
     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());
@@ -249,6 +243,7 @@
       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());
@@ -376,9 +371,8 @@
     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, event_log_.get());
+  AudioReceiveStream* receive_stream = new AudioReceiveStream(
+      congestion_controller_.get(), config, config_.audio_state);
   {
     WriteLockScoped write_lock(*receive_crit_);
     RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
@@ -427,8 +421,8 @@
   VideoSendStream* send_stream = new VideoSendStream(
       num_cpu_cores_, module_process_thread_.get(), call_stats_.get(),
       congestion_controller_.get(), bitrate_allocator_.get(),
-      video_send_delay_stats_.get(), &remb_, event_log_.get(), config,
-      encoder_config, suspended_video_send_ssrcs_);
+      video_send_delay_stats_.get(), &remb_, event_log_, config, encoder_config,
+      suspended_video_send_ssrcs_);
   {
     WriteLockScoped write_lock(*send_crit_);
     for (uint32_t ssrc : config.rtp.ssrcs) {
@@ -439,7 +433,8 @@
   }
   send_stream->SignalNetworkState(video_network_state_);
   UpdateAggregateNetworkState();
-  event_log_->LogVideoSendStreamConfig(config);
+  if (event_log_)
+    event_log_->LogVideoSendStreamConfig(config);
   return send_stream;
 }
 
@@ -498,11 +493,13 @@
     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();
-  event_log_->LogVideoReceiveStreamConfig(config);
+  if (event_log_)
+    event_log_->LogVideoReceiveStreamConfig(config);
   return receive_stream;
 }
 
@@ -830,7 +827,7 @@
       auto status = it->second->DeliverRtp(packet, length, packet_time)
                         ? DELIVERY_OK
                         : DELIVERY_PACKET_ERROR;
-      if (status == DELIVERY_OK)
+      if (status == DELIVERY_OK && event_log_)
         event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
       return status;
     }
@@ -842,7 +839,7 @@
       auto status = it->second->DeliverRtp(packet, length, packet_time)
                         ? DELIVERY_OK
                         : DELIVERY_PACKET_ERROR;
-      if (status == DELIVERY_OK)
+      if (status == DELIVERY_OK && event_log_)
         event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
       return status;
     }