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
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
54message OutputData {
55  repeated Test test = 1;
56}
57
58