| 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; |
| } |