Chromium Code Reviews| 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; |
| } |