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 00314e619d086d13fa4a20510d47ba0e1f73dcbc..976ff2321c08106dcea319a19c63f50619ebb0af 100644 |
--- a/webrtc/logging/rtc_event_log/rtc_event_log.cc |
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.cc |
@@ -62,6 +62,9 @@ class RtcEventLogImpl final : public RtcEventLog { |
void LogVideoReceiveStreamConfig( |
const VideoReceiveStream::Config& config) override; |
void LogVideoSendStreamConfig(const VideoSendStream::Config& config) override; |
+ void LogAudioReceiveStreamConfig( |
+ const AudioReceiveStream::Config& config) override; |
+ void LogAudioSendStreamConfig(const AudioSendStream::Config& config) override; |
void LogRtpHeader(PacketDirection direction, |
MediaType media_type, |
const uint8_t* header, |
@@ -292,6 +295,46 @@ void RtcEventLogImpl::LogVideoSendStreamConfig( |
StoreEvent(&event); |
} |
+void RtcEventLogImpl::LogAudioReceiveStreamConfig( |
+ const AudioReceiveStream::Config& config) { |
+ std::unique_ptr<rtclog::Event> event(new rtclog::Event()); |
+ event->set_timestamp_us(clock_->TimeInMicroseconds()); |
+ event->set_type(rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT); |
+ |
+ rtclog::AudioReceiveConfig* receiver_config = |
+ event->mutable_audio_receiver_config(); |
+ receiver_config->set_remote_ssrc(config.rtp.remote_ssrc); |
+ receiver_config->set_local_ssrc(config.rtp.local_ssrc); |
+ |
+ for (const auto& e : config.rtp.extensions) { |
+ rtclog::RtpHeaderExtension* extension = |
+ receiver_config->add_header_extensions(); |
+ extension->set_name(e.uri); |
+ extension->set_id(e.id); |
+ } |
+ StoreEvent(&event); |
+} |
+ |
+void RtcEventLogImpl::LogAudioSendStreamConfig( |
+ const AudioSendStream::Config& config) { |
+ std::unique_ptr<rtclog::Event> event(new rtclog::Event()); |
+ event->set_timestamp_us(clock_->TimeInMicroseconds()); |
+ event->set_type(rtclog::Event::AUDIO_SENDER_CONFIG_EVENT); |
+ |
+ rtclog::AudioSendConfig* sender_config = event->mutable_audio_sender_config(); |
+ |
+ sender_config->set_ssrc(config.rtp.ssrc); |
+ |
+ for (const auto& e : config.rtp.extensions) { |
+ rtclog::RtpHeaderExtension* extension = |
+ sender_config->add_header_extensions(); |
+ extension->set_name(e.uri); |
+ extension->set_id(e.id); |
+ } |
+ |
+ StoreEvent(&event); |
+} |
+ |
void RtcEventLogImpl::LogRtpHeader(PacketDirection direction, |
MediaType media_type, |
const uint8_t* header, |