| 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;
|
| +}
|
|
|