1syntax = "proto2"; 2option optimize_for = LITE_RUNTIME; 3package webrtc.audioproc; 4 5message Test { 6 optional int32 num_reverse_channels = 1; 7 optional int32 num_input_channels = 2; 8 optional int32 num_output_channels = 3; 9 optional int32 sample_rate = 4; 10 11 message Frame { 12 } 13 14 repeated Frame frame = 5; 15 16 optional int32 analog_level_average = 6; 17 optional int32 max_output_average = 7; 18 optional int32 has_voice_count = 9; 19 optional int32 is_saturated_count = 10; 20 21 message EchoMetrics { 22 optional float echo_return_loss = 1; 23 optional float echo_return_loss_enhancement = 2; 24 optional float divergent_filter_fraction = 3; 25 optional float residual_echo_likelihood = 4; 26 optional float residual_echo_likelihood_recent_max = 5; 27 } 28 29 repeated EchoMetrics echo_metrics = 11; 30 31 message DelayMetrics { 32 optional int32 median = 1; 33 optional int32 std = 2; 34 } 35 36 repeated DelayMetrics delay_metrics = 12; 37 38 optional float rms_dbfs_average = 13; 39 40 optional float ns_speech_probability_average = 14; 41 42 optional bool use_aec_extended_filter = 15; 43} 44 45message OutputData { 46 repeated Test test = 1; 47} 48 49