14b43beb7c
TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
56 lines
1.2 KiB
Protocol Buffer
56 lines
1.2 KiB
Protocol Buffer
syntax = "proto2";
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option optimize_for = LITE_RUNTIME;
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package webrtc.audioproc;
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message Test {
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optional int32 num_reverse_channels = 1;
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optional int32 num_input_channels = 2;
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optional int32 num_output_channels = 3;
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optional int32 sample_rate = 4;
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message Frame {
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}
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repeated Frame frame = 5;
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optional int32 analog_level_average = 6;
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optional int32 max_output_average = 7;
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optional int32 has_echo_count = 8;
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optional int32 has_voice_count = 9;
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optional int32 is_saturated_count = 10;
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message Statistic {
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optional int32 instant = 1;
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optional int32 average = 2;
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optional int32 maximum = 3;
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optional int32 minimum = 4;
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}
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message EchoMetrics {
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optional Statistic residual_echo_return_loss = 1;
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optional Statistic echo_return_loss = 2;
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optional Statistic echo_return_loss_enhancement = 3;
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optional Statistic a_nlp = 4;
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}
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optional EchoMetrics echo_metrics = 11;
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message DelayMetrics {
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optional int32 median = 1;
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optional int32 std = 2;
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}
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optional DelayMetrics delay_metrics = 12;
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optional int32 rms_level = 13;
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optional float ns_speech_probability_average = 14;
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}
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message OutputData {
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repeated Test test = 1;
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}
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