| Index: webrtc/logging/rtc_event_log/rtc_event_log.proto
|
| diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.proto b/webrtc/logging/rtc_event_log/rtc_event_log.proto
|
| index a6d169579659e496379394de86cb9b4fac220b13..7230177f699eabc6a1f17594c51dc4d519c8c877 100644
|
| --- a/webrtc/logging/rtc_event_log/rtc_event_log.proto
|
| +++ b/webrtc/logging/rtc_event_log/rtc_event_log.proto
|
| @@ -2,6 +2,8 @@ syntax = "proto2";
|
| option optimize_for = LITE_RUNTIME;
|
| package webrtc.rtclog;
|
|
|
| +import "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto";
|
| +
|
| enum MediaType {
|
| ANY = 0;
|
| AUDIO = 1;
|
| @@ -37,6 +39,7 @@ message Event {
|
| VIDEO_SENDER_CONFIG_EVENT = 9;
|
| AUDIO_RECEIVER_CONFIG_EVENT = 10;
|
| AUDIO_SENDER_CONFIG_EVENT = 11;
|
| + AUDIO_NETWORK_ADAPTOR_EVENT = 12;
|
| }
|
|
|
| // required - Indicates the type of this event
|
| @@ -65,6 +68,10 @@ message Event {
|
|
|
| // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT
|
| optional AudioSendConfig audio_sender_config = 11;
|
| +
|
| + // optional - but required if type == AUDIO_NETWORK_ADAPTOR_EVENT
|
| + optional webrtc.audio_network_adaptor.debug_dump.EncoderRuntimeConfig
|
| + encoder_runtime_config = 12;
|
| }
|
|
|
| message RtpPacket {
|
|
|