| 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 3b808b2e43d451fac6915f3aa2a891c5685a2160..ce55a4fb39b740a422eda7b874a8a000f88e4601 100644
|
| --- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
|
| +++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
|
| @@ -79,8 +79,6 @@
|
| 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;
|
| @@ -456,29 +454,4 @@
|
| }
|
| }
|
|
|
| -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
|
|
|