webrtc/modules/audio_processing/debug.proto

syntax = "proto2";
option optimize_for = LITE_RUNTIME;
package webrtc.audioproc;

message Init {
  optional int32 sample_rate = 1;
  optional int32 device_sample_rate = 2;
  optional int32 num_input_channels = 3;
  optional int32 num_output_channels = 4;
  optional int32 num_reverse_channels = 5;
  optional int32 reverse_sample_rate = 6;
}

// 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;
}