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 65ee7d8881e6c413301e85fd02667878d29defbe..21ca5e127de2f8d3b135cc966cdcf9d8dd80cd6d 100644 |
--- a/webrtc/logging/rtc_event_log/rtc_event_log.cc |
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.cc |
@@ -77,8 +77,6 @@ |
void LogBwePacketLossEvent(int32_t bitrate, |
uint8_t fraction_loss, |
int32_t total_packets) override; |
- void LogAudioNetworkAdaptation( |
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override; |
private: |
void StoreEvent(std::unique_ptr<rtclog::Event>* event); |
@@ -436,29 +434,6 @@ |
StoreEvent(&event); |
} |
-void RtcEventLogImpl::LogAudioNetworkAdaptation( |
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) { |
- std::unique_ptr<rtclog::Event> event(new rtclog::Event()); |
- event->set_timestamp_us(rtc::TimeMicros()); |
- event->set_type(rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT); |
- auto audio_network_adaptation = event->mutable_audio_network_adaptation(); |
- if (config.bitrate_bps) |
- audio_network_adaptation->set_bitrate_bps(*config.bitrate_bps); |
- if (config.frame_length_ms) |
- audio_network_adaptation->set_frame_length_ms(*config.frame_length_ms); |
- if (config.uplink_packet_loss_fraction) { |
- audio_network_adaptation->set_uplink_packet_loss_fraction( |
- *config.uplink_packet_loss_fraction); |
- } |
- if (config.enable_fec) |
- audio_network_adaptation->set_enable_fec(*config.enable_fec); |
- if (config.enable_dtx) |
- audio_network_adaptation->set_enable_dtx(*config.enable_dtx); |
- if (config.num_channels) |
- audio_network_adaptation->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."; |