webrtc/modules/audio_processing/test/unittest.proto

syntax = "proto2";
option optimize_for = LITE_RUNTIME;
package webrtc.audioproc;

message Test {
  optional int32 num_reverse_channels = 1;
  optional int32 num_input_channels = 2;
  optional int32 num_output_channels = 3;
  optional int32 sample_rate = 4;

  message Frame {
  }

  repeated Frame frame = 5;

  optional int32 analog_level_average = 6;
  optional int32 max_output_average = 7;

  optional int32 has_echo_count = 8;
  optional int32 has_voice_count = 9;
  optional int32 is_saturated_count = 10;

  message Statistic {
    optional int32 instant = 1;
    optional int32 average = 2;
    optional int32 maximum = 3;
    optional int32 minimum = 4;
  }

  message EchoMetrics {
    optional Statistic residual_echo_return_loss = 1;
    optional Statistic echo_return_loss = 2;
    optional Statistic echo_return_loss_enhancement = 3;
    optional Statistic a_nlp = 4;
    optional float divergent_filter_fraction = 5;
  }

  repeated EchoMetrics echo_metrics = 11;

  message DelayMetrics {
    optional int32 median = 1;
    optional int32 std = 2;
    optional float fraction_poor_delays = 3;
  }

  repeated DelayMetrics delay_metrics = 12;

  repeated int32 rms_level = 13;

  optional float ns_speech_probability_average = 14;

  optional bool use_aec_extended_filter = 15;
}

message OutputData {
  repeated Test test = 1;
}