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The test is refitted to use the AudioProcessingStats struct to get reference data. The old metrics do not map entirely injectively to the new ones, so the reference protobuf and files are updated as well. Bug: webrtc:9535 Change-Id: I546dca2979380e03895af0077bfc77ffd24abe36 Reviewed-on: https://webrtc-review.googlesource.com/100100 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24740}
48 lines
1.1 KiB
Protocol Buffer
48 lines
1.1 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_voice_count = 9;
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optional int32 is_saturated_count = 10;
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message EchoMetrics {
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optional float echo_return_loss = 1;
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optional float echo_return_loss_enhancement = 2;
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optional float divergent_filter_fraction = 3;
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optional float residual_echo_likelihood = 4;
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optional float residual_echo_likelihood_recent_max = 5;
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}
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repeated 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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repeated DelayMetrics delay_metrics = 12;
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repeated int32 rms_level = 13;
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optional float ns_speech_probability_average = 14;
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optional bool use_aec_extended_filter = 15;
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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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