Index: webrtc/logging/rtc_event_log/rtc_event_log.cc |
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.cc b/webrtc/logging/rtc_event_log/rtc_event_log.cc |
index 976ff2321c08106dcea319a19c63f50619ebb0af..96d6e56d653fcd6633f5c10ff9d7391d291cc336 100644 |
--- a/webrtc/logging/rtc_event_log/rtc_event_log.cc |
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.cc |
@@ -77,6 +77,8 @@ class RtcEventLogImpl final : public RtcEventLog { |
void LogBwePacketLossEvent(int32_t bitrate, |
uint8_t fraction_loss, |
int32_t total_packets) override; |
+ void LogEncoderRuntimeConfig( |
+ const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override; |
private: |
void StoreEvent(std::unique_ptr<rtclog::Event>* event); |
@@ -437,6 +439,28 @@ void RtcEventLogImpl::LogBwePacketLossEvent(int32_t bitrate, |
StoreEvent(&event); |
} |
+void RtcEventLogImpl::LogEncoderRuntimeConfig( |
+ const AudioNetworkAdaptor::EncoderRuntimeConfig& config) { |
+ std::unique_ptr<rtclog::Event> event(new rtclog::Event()); |
+ event->set_timestamp_us(clock_->TimeInMicroseconds()); |
+ event->set_type(rtclog::Event::AUDIO_NETWORK_ADAPTOR_EVENT); |
minyue-webrtc
2016/12/20 10:39:29
Slightly confusing is that the function is LogEnco
michaelt
2016/12/20 13:41:41
Done.
minyue-webrtc
2016/12/20 14:11:05
Almost, but we cannot drop "Audio" in the function
michaelt
2016/12/20 14:34:03
Right :)
Done
|
+ auto ana_event = event->mutable_encoder_runtime_config(); |
+ if (config.bitrate_bps) |
minyue-webrtc
2016/12/20 10:39:29
how about making 448 ~ 460 a utility function, whi
michaelt
2016/12/20 13:41:40
Done.
|
+ ana_event->set_bitrate_bps(*config.bitrate_bps); |
+ if (config.frame_length_ms) |
+ ana_event->set_frame_length_ms(*config.frame_length_ms); |
+ if (config.uplink_packet_loss_fraction) |
+ ana_event->set_uplink_packet_loss_fraction( |
+ *config.uplink_packet_loss_fraction); |
+ if (config.enable_fec) |
+ ana_event->set_enable_fec(*config.enable_fec); |
+ if (config.enable_dtx) |
+ ana_event->set_enable_dtx(*config.enable_dtx); |
+ if (config.num_channels) |
+ ana_event->set_num_channels(*config.num_channels); |
+ StoreEvent(&event); |
+} |
+ |
void RtcEventLogImpl::StoreEvent(std::unique_ptr<rtclog::Event>* event) { |
if (!event_queue_.Insert(event)) { |
LOG(LS_ERROR) << "WebRTC event log queue full. Dropping event."; |