mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-02 17:51:06 +01:00
5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
30 lines
925 B
C
30 lines
925 B
C
/*
|
|
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef MODULES_AUDIO_PROCESSING_VAD_COMMON_H_
|
|
#define MODULES_AUDIO_PROCESSING_VAD_COMMON_H_
|
|
|
|
#include <stddef.h>
|
|
|
|
static const int kSampleRateHz = 16000;
|
|
static const size_t kLength10Ms = kSampleRateHz / 100;
|
|
static const size_t kMaxNumFrames = 4;
|
|
|
|
struct AudioFeatures {
|
|
double log_pitch_gain[kMaxNumFrames];
|
|
double pitch_lag_hz[kMaxNumFrames];
|
|
double spectral_peak[kMaxNumFrames];
|
|
double rms[kMaxNumFrames];
|
|
size_t num_frames;
|
|
bool silence;
|
|
};
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_VAD_COMMON_H_
|