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..e80772297997ce2b088404d8ff83395b71fab4f8 100644 |
--- a/webrtc/logging/rtc_event_log/rtc_event_log.proto |
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.proto |
@@ -37,6 +37,7 @@ message Event { |
VIDEO_SENDER_CONFIG_EVENT = 9; |
AUDIO_RECEIVER_CONFIG_EVENT = 10; |
AUDIO_SENDER_CONFIG_EVENT = 11; |
+ AUDIO_NETWORK_ADAPTATION_EVENT = 16; |
} |
// required - Indicates the type of this event |
@@ -65,6 +66,9 @@ message Event { |
// optional - but required if type == AUDIO_SENDER_CONFIG_EVENT |
optional AudioSendConfig audio_sender_config = 11; |
+ |
+ // optional - but required if type == AUDIO_NETWORK_ADAPTATION_EVENT |
+ optional AudioNetworkAdaptation audio_network_adaptation = 16; |
} |
message RtpPacket { |
@@ -227,3 +231,24 @@ message AudioSendConfig { |
// RTP header extensions used for the outgoing audio stream. |
repeated RtpHeaderExtension header_extensions = 2; |
} |
+ |
+message AudioNetworkAdaptation { |
+ // Bit rate that the audio encoder is operating at. |
+ optional int32 bitrate_bps = 1; |
+ |
+ // Frame length that each encoded audio packet consists of. |
+ optional int32 frame_length_ms = 2; |
+ |
+ // Packet loss fraction that the encoder's forward error correction (FEC) is |
+ // optimized for. |
+ optional float uplink_packet_loss_fraction = 3; |
+ |
+ // Whether forward error correction (FEC) is turned on or off. |
+ optional bool enable_fec = 4; |
+ |
+ // Whether discontinuous transmission (DTX) is turned on or off. |
+ optional bool enable_dtx = 5; |
+ |
+ // Number of audio channels that each encoded packet consists of. |
+ optional uint32 num_channels = 6; |
+} |