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 |