| Index: webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
|
| diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
|
| index f4252449987a341f2ceae6ecc81297f81b2a2923..2ffee52432cedb9b763764f92bb8e8a3b23000fe 100644
|
| --- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
|
| +++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
|
| @@ -7,11 +7,14 @@ message NetworkMetrics {
|
| optional float uplink_packet_loss_fraction = 2;
|
| optional int32 target_audio_bitrate_bps = 3;
|
| optional int32 rtt_ms = 4;
|
| + optional int32 uplink_recoverable_packet_loss_fraction = 5;
|
| }
|
|
|
| message EncoderRuntimeConfig {
|
| optional int32 bitrate_bps = 1;
|
| optional int32 frame_length_ms = 2;
|
| + // Note: This is what tell the encoder. It doesn't have to reflect
|
| + // the actual NetworkMetrics; it's subject to our decision.
|
| optional float uplink_packet_loss_fraction = 3;
|
| optional bool enable_fec = 4;
|
| optional bool enable_dtx = 5;
|
|
|