1 syntax = "proto2"; 2 option optimize_for = LITE_RUNTIME; 3 package webrtc.audioproc; 4 5 message 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 19 optional int32 has_echo_count = 8; 20 optional int32 has_voice_count = 9; 21 optional int32 is_saturated_count = 10; 22 23 message Statistic { 24 optional int32 instant = 1; 25 optional int32 average = 2; 26 optional int32 maximum = 3; 27 optional int32 minimum = 4; 28 } 29 30 message EchoMetrics { 31 optional Statistic residual_echo_return_loss = 1; 32 optional Statistic echo_return_loss = 2; 33 optional Statistic echo_return_loss_enhancement = 3; 34 optional Statistic a_nlp = 4; 35 } 36 37 optional EchoMetrics echo_metrics = 11; 38 39 message DelayMetrics { 40 optional int32 median = 1; 41 optional int32 std = 2; 42 optional float fraction_poor_delays = 3; 43 } 44 45 optional DelayMetrics delay_metrics = 12; 46 47 optional int32 rms_level = 13; 48 49 optional float ns_speech_probability_average = 14; 50 51 optional bool use_aec_extended_filter = 15; 52 } 53 54 message OutputData { 55 repeated Test test = 1; 56 } 57 58
