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 { |