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syntax = "proto2";
option optimize_for = LITE_RUNTIME;
package webrtc.audioproc;
message Init {
optional int32 sample_rate = 1;
optional int32 device_sample_rate = 2 [deprecated=true];
optional int32 num_input_channels = 3;
optional int32 num_output_channels = 4;
optional int32 num_reverse_channels = 5;
optional int32 reverse_sample_rate = 6;
optional int32 output_sample_rate = 7;
}
// May contain interleaved or deinterleaved data, but don't store both formats.
message ReverseStream {
// int16 interleaved data.
optional bytes data = 1;
// float deinterleaved data, where each repeated element points to a single
// channel buffer of data.
repeated bytes channel = 2;
}
// May contain interleaved or deinterleaved data, but don't store both formats.
message Stream {
// int16 interleaved data.
optional bytes input_data = 1;
optional bytes output_data = 2;
optional int32 delay = 3;
optional sint32 drift = 4;
optional int32 level = 5;
optional bool keypress = 6;
// float deinterleaved data, where each repeated element points to a single
// channel buffer of data.
repeated bytes input_channel = 7;
repeated bytes output_channel = 8;
}
message Event {
enum Type {
INIT = 0;
REVERSE_STREAM = 1;
STREAM = 2;
}
required Type type = 1;
optional Init init = 2;
optional ReverseStream reverse_stream = 3;
optional Stream stream = 4;
}
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