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..3f9275a05ce1700095c83b0149c64cd6a549f0c1 100644 |
--- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto |
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto |
@@ -7,14 +7,20 @@ 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 we 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; |
+ // Some encoders can encode fewer channels than the actual input to make |
+ // better use of the bandwidth. |num_channels| sets the number of channels |
+ // to encode. |
optional uint32 num_channels = 6; |
} |