 Chromium Code Reviews
 Chromium Code Reviews Issue 1748403002:
  Move RtcEventLog object from inside VoiceEngine to Call.  (Closed) 
  Base URL: https://chromium.googlesource.com/external/webrtc.git@master
    
  
    Issue 1748403002:
  Move RtcEventLog object from inside VoiceEngine to Call.  (Closed) 
  Base URL: https://chromium.googlesource.com/external/webrtc.git@master| Index: webrtc/call/call.cc | 
| diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc | 
| index 3fd7a931d0cecc88cff0b30a47557db5a21ed586..20f73815eada75b812f1e1a0d10acfb30bd887ec 100644 | 
| --- a/webrtc/call/call.cc | 
| +++ b/webrtc/call/call.cc | 
| @@ -94,6 +94,13 @@ class Call : public webrtc::Call, public PacketReceiver, | 
| 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); | 
| @@ -148,7 +155,7 @@ class Call : public webrtc::Call, public PacketReceiver, | 
| VideoSendStream::RtpStateMap suspended_video_send_ssrcs_; | 
| - RtcEventLog* event_log_ = nullptr; | 
| + rtc::scoped_ptr<webrtc::RtcEventLog> event_log_; | 
| 
the sun
2016/03/23 10:26:45
use unique_ptr
 
ivoc
2016/03/23 14:27:26
Done.
 | 
| // The following members are only accessed (exclusively) from one thread and | 
| // from the destructor, and therefore doesn't need any explicit | 
| @@ -192,6 +199,7 @@ Call::Call(const Call::Config& config) | 
| network_enabled_(true), | 
| receive_crit_(RWLockWrapper::CreateRWLock()), | 
| send_crit_(RWLockWrapper::CreateRWLock()), | 
| + event_log_(RtcEventLog::Create()), | 
| received_video_bytes_(0), | 
| received_audio_bytes_(0), | 
| received_rtcp_bytes_(0), | 
| @@ -202,7 +210,8 @@ Call::Call(const Call::Config& config) | 
| 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, | 
| @@ -211,10 +220,6 @@ 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()); | 
| @@ -223,7 +228,6 @@ 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_); | 
| module_process_thread_->Start(); | 
| module_process_thread_->RegisterModule(call_stats_.get()); | 
| @@ -350,8 +354,9 @@ 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) == | 
| @@ -408,8 +413,7 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream( | 
| } | 
| video_send_streams_.insert(send_stream); | 
| - if (event_log_) | 
| - event_log_->LogVideoSendStreamConfig(config); | 
| + event_log_->LogVideoSendStreamConfig(config); | 
| return send_stream; | 
| } | 
| @@ -472,8 +476,7 @@ 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; | 
| } | 
| @@ -673,9 +676,7 @@ 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); | 
| } | 
| } | 
| } | 
| @@ -684,9 +685,7 @@ 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); | 
| } | 
| } | 
| } | 
| @@ -715,7 +714,7 @@ 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; | 
| } | 
| @@ -727,7 +726,7 @@ 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; | 
| } |