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
38 lines
746 B
Protocol Buffer
38 lines
746 B
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 Init {
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optional int32 sample_rate = 1;
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optional int32 device_sample_rate = 2;
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optional int32 num_input_channels = 3;
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optional int32 num_output_channels = 4;
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optional int32 num_reverse_channels = 5;
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}
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message ReverseStream {
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optional bytes data = 1;
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}
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message Stream {
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optional bytes input_data = 1;
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optional bytes output_data = 2;
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optional int32 delay = 3;
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optional sint32 drift = 4;
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optional int32 level = 5;
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}
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message Event {
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enum Type {
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INIT = 0;
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REVERSE_STREAM = 1;
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STREAM = 2;
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}
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required Type type = 1;
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optional Init init = 2;
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optional ReverseStream reverse_stream = 3;
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optional Stream stream = 4;
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}
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