Index: webrtc/logging/rtc_event_log/rtc_event_log_parser.cc |
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc |
index ce55a4fb39b740a422eda7b874a8a000f88e4601..3b808b2e43d451fac6915f3aa2a891c5685a2160 100644 |
--- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc |
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc |
@@ -79,6 +79,8 @@ ParsedRtcEventLog::EventType GetRuntimeEventType( |
return ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT; |
case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT: |
return ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT; |
+ case rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT: |
+ return ParsedRtcEventLog::EventType::AUDIO_NETWORK_ADAPTATION_EVENT; |
} |
RTC_NOTREACHED(); |
return ParsedRtcEventLog::EventType::UNKNOWN_EVENT; |
@@ -454,4 +456,29 @@ void ParsedRtcEventLog::GetBwePacketLossEvent(size_t index, |
} |
} |
+void ParsedRtcEventLog::GetAudioNetworkAdaptation( |
+ size_t index, |
+ AudioNetworkAdaptor::EncoderRuntimeConfig* config) const { |
+ RTC_CHECK_LT(index, GetNumberOfEvents()); |
+ const rtclog::Event& event = events_[index]; |
+ RTC_CHECK(event.has_type()); |
+ RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT); |
+ RTC_CHECK(event.has_audio_network_adaptation()); |
+ const rtclog::AudioNetworkAdaptation& ana_event = |
+ event.audio_network_adaptation(); |
+ if (ana_event.has_bitrate_bps()) |
+ config->bitrate_bps = rtc::Optional<int>(ana_event.bitrate_bps()); |
+ if (ana_event.has_enable_fec()) |
+ config->enable_fec = rtc::Optional<bool>(ana_event.enable_fec()); |
+ if (ana_event.has_enable_dtx()) |
+ config->enable_dtx = rtc::Optional<bool>(ana_event.enable_dtx()); |
+ if (ana_event.has_frame_length_ms()) |
+ config->frame_length_ms = rtc::Optional<int>(ana_event.frame_length_ms()); |
+ if (ana_event.has_num_channels()) |
+ config->num_channels = rtc::Optional<size_t>(ana_event.num_channels()); |
+ if (ana_event.has_uplink_packet_loss_fraction()) |
+ config->uplink_packet_loss_fraction = |
+ rtc::Optional<float>(ana_event.uplink_packet_loss_fraction()); |
+} |
+ |
} // namespace webrtc |