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 |
deleted file mode 100644 |
index f4252449987a341f2ceae6ecc81297f81b2a2923..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto |
+++ /dev/null |
@@ -1,30 +0,0 @@ |
-syntax = "proto2"; |
-option optimize_for = LITE_RUNTIME; |
-package webrtc.audio_network_adaptor.debug_dump; |
- |
-message NetworkMetrics { |
- optional int32 uplink_bandwidth_bps = 1; |
- optional float uplink_packet_loss_fraction = 2; |
- optional int32 target_audio_bitrate_bps = 3; |
- optional int32 rtt_ms = 4; |
-} |
- |
-message EncoderRuntimeConfig { |
- optional int32 bitrate_bps = 1; |
- optional int32 frame_length_ms = 2; |
- optional float uplink_packet_loss_fraction = 3; |
- optional bool enable_fec = 4; |
- optional bool enable_dtx = 5; |
- optional uint32 num_channels = 6; |
-} |
- |
-message Event { |
- enum Type { |
- NETWORK_METRICS = 0; |
- ENCODER_RUNTIME_CONFIG = 1; |
- } |
- required Type type = 1; |
- required uint32 timestamp = 2; |
- optional NetworkMetrics network_metrics = 3; |
- optional EncoderRuntimeConfig encoder_runtime_config = 4; |
-} |