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; |
} |