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 41ac328eab14f44249a20d7ac0dba85a36bb9d1d..feb120397ef65ad75418fbeab1a52b57fe47820c 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -94,6 +94,8 @@ class Call : public webrtc::Call, public PacketReceiver,
   void OnNetworkChanged(uint32_t bitrate_bps, uint8_t fraction_loss,
                         int64_t rtt_ms) override;
 
+  webrtc::RtcEventLog* RtcEventLog() override { return event_log_.get(); }
+
  private:
   DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
                              size_t length);
@@ -148,7 +150,7 @@ class Call : public webrtc::Call, public PacketReceiver,
 
   VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
 
-  RtcEventLog* event_log_ = nullptr;
+  rtc::scoped_ptr<webrtc::RtcEventLog> event_log_ = RtcEventLog::Create();

[the sun, 2016/03/03 09:25:13]

Is there a thread checker in RtcEventLog::Create()? We want to enforce
construction/destruction is on the same thread.

[stefan-webrtc, 2016/03/03 10:08:25]

Use std::unique_ptr. I'd also prefer if you create the log in the initializer
list.

[ivoc, 2016/03/10 13:15:36]

Done.

[ivoc, 2016/03/10 13:15:36]

I moved the initialization to the initializer list (although I was under the
impression that this style was preferred now). I think switching to
std::unique_ptr is a good idea, but I would prefer to do that in a separate CL
if that's okay with you.

 
   // The following members are only accessed (exclusively) from one thread and
   // from the destructor, and therefore doesn't need any explicit
@@ -210,10 +212,6 @@ Call::Call(const Call::Config& config)
     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());
@@ -222,7 +220,7 @@ Call::Call(const Call::Config& config)
       config_.bitrate_config.min_bitrate_bps,
       config_.bitrate_config.start_bitrate_bps,
       config_.bitrate_config.max_bitrate_bps);
-  congestion_controller_->GetBitrateController()->SetEventLog(event_log_);
+  congestion_controller_->GetBitrateController()->SetEventLog(event_log_.get());

[the sun, 2016/03/03 09:25:12]

Could we pass the log* in the CongestionController ctor?

[stefan-webrtc, 2016/03/03 10:08:25]

That would be nicer.

[ivoc, 2016/03/10 13:15:36]

Good point, done.

 
   module_process_thread_->Start();
   module_process_thread_->RegisterModule(call_stats_.get());
@@ -351,6 +349,7 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   AudioReceiveStream* receive_stream = new AudioReceiveStream(
       congestion_controller_.get(), config, config_.audio_state);
+  receive_stream->SetRtcEventLog(event_log_.get());

[stefan-webrtc, 2016/03/03 10:08:25]

Isn't it possible to pass this via the constructor instead?

[ivoc, 2016/03/10 13:15:36]

Done.

   {
     WriteLockScoped write_lock(*receive_crit_);
     RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
@@ -407,8 +406,7 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
   }
   video_send_streams_.insert(send_stream);
 
-  if (event_log_)
-    event_log_->LogVideoSendStreamConfig(config);
+  event_log_->LogVideoSendStreamConfig(config);
 
   return send_stream;
 }
@@ -471,8 +469,7 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
   if (!network_enabled_)
     receive_stream->SignalNetworkState(kNetworkDown);
 
-  if (event_log_)
-    event_log_->LogVideoReceiveStreamConfig(config);
+  event_log_->LogVideoReceiveStreamConfig(config);
 
   return receive_stream;
 }
@@ -672,9 +669,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
     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);
       }
     }
   }
@@ -683,9 +678,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
     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);
       }
     }
   }
@@ -714,7 +707,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
       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;
     }
@@ -726,7 +719,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
       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;
     }