Files
webrtc-audio-processing/src/modules/audio_processing/debug.proto
Arun Raghavan 693d686b0f Update code to upstream revision r767
Just reorganisation of the audio_processing code.
2011-10-21 09:53:02 +05:30

38 lines
746 B
Protocol Buffer

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;
}
message ReverseStream {
optional bytes data = 1;
}
message Stream {
optional bytes input_data = 1;
optional bytes output_data = 2;
optional int32 delay = 3;
optional sint32 drift = 4;
optional int32 level = 5;
}
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;
}