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| 1 syntax = "proto2"; |
| 2 option optimize_for = LITE_RUNTIME; |
| 3 package webrtc.audio_network_adaptor.debug_dump; |
| 4 |
| 5 message NetworkMetrics { |
| 6 optional int32 uplink_bandwidth_bps = 1; |
| 7 optional float uplink_packet_loss_fraction = 2; |
| 8 optional int32 target_audio_bitrate_bps = 3; |
| 9 optional int32 rtt_ms = 4; |
| 10 } |
| 11 |
| 12 message EncoderRuntimeConfig { |
| 13 optional int32 bitrate_bps = 1; |
| 14 optional int32 frame_length_ms = 2; |
| 15 optional float uplink_packet_loss_fraction = 3; |
| 16 optional bool enable_fec = 4; |
| 17 optional bool enable_dtx = 5; |
| 18 optional uint32 num_channels = 6; |
| 19 } |
| 20 |
| 21 message Event { |
| 22 enum Type { |
| 23 NETWORK_METRICS = 0; |
| 24 ENCODER_RUNTIME_CONFIG = 1; |
| 25 } |
| 26 required Type type = 1; |
| 27 required uint32 timestamp = 2; |
| 28 optional NetworkMetrics network_metrics = 3; |
| 29 optional EncoderRuntimeConfig encoder_runtime_config = 4; |
| 30 } |
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