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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;
}
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