mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-11 08:39:49 +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.
68 lines
2.7 KiB
C++
68 lines
2.7 KiB
C++
/*
|
|
* Copyright (c) 2018 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_AGC2_RNN_VAD_COMMON_H_
|
|
#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_COMMON_H_
|
|
|
|
namespace webrtc {
|
|
namespace rnn_vad {
|
|
|
|
constexpr double kPi = 3.14159265358979323846;
|
|
|
|
constexpr size_t kSampleRate24kHz = 24000;
|
|
constexpr size_t kFrameSize10ms24kHz = kSampleRate24kHz / 100;
|
|
constexpr size_t kFrameSize20ms24kHz = kFrameSize10ms24kHz * 2;
|
|
|
|
// Pitch analysis params.
|
|
constexpr size_t kMinPitch24kHz = kSampleRate24kHz / 800; // 0.00125 s.
|
|
constexpr size_t kMaxPitch24kHz = kSampleRate24kHz / 62.5; // 0.016 s.
|
|
constexpr size_t kBufSize24kHz = kMaxPitch24kHz + kFrameSize20ms24kHz;
|
|
static_assert((kBufSize24kHz & 1) == 0, "The buffer size must be even.");
|
|
|
|
// Define a higher minimum pitch period for the initial search. This is used to
|
|
// avoid searching for very short periods, for which a refinement step is
|
|
// responsible.
|
|
constexpr size_t kInitialMinPitch24kHz = 3 * kMinPitch24kHz;
|
|
static_assert(kMinPitch24kHz < kInitialMinPitch24kHz, "");
|
|
static_assert(kInitialMinPitch24kHz < kMaxPitch24kHz, "");
|
|
|
|
// 12 kHz analysis.
|
|
constexpr size_t kSampleRate12kHz = 12000;
|
|
constexpr size_t kFrameSize10ms12kHz = kSampleRate12kHz / 100;
|
|
constexpr size_t kFrameSize20ms12kHz = kFrameSize10ms12kHz * 2;
|
|
constexpr size_t kBufSize12kHz = kBufSize24kHz / 2;
|
|
constexpr size_t kInitialMinPitch12kHz = kInitialMinPitch24kHz / 2;
|
|
constexpr size_t kMaxPitch12kHz = kMaxPitch24kHz / 2;
|
|
|
|
// 48 kHz constants.
|
|
constexpr size_t kMinPitch48kHz = kMinPitch24kHz * 2;
|
|
constexpr size_t kMaxPitch48kHz = kMaxPitch24kHz * 2;
|
|
|
|
// Sub-band frequency boundaries.
|
|
constexpr size_t kNumBands = 22;
|
|
constexpr int kBandFrequencyBoundaries[kNumBands] = {
|
|
0, 200, 400, 600, 800, 1000, 1200, 1400, 1600, 2000, 2400,
|
|
2800, 3200, 4000, 4800, 5600, 6800, 8000, 9600, 12000, 15600, 20000};
|
|
|
|
// Feature extraction parameters.
|
|
constexpr size_t kNumLowerBands = 6;
|
|
static_assert((0 < kNumLowerBands) && (kNumLowerBands < kNumBands), "");
|
|
constexpr size_t kSpectralCoeffsHistorySize = 8;
|
|
static_assert(kSpectralCoeffsHistorySize > 2,
|
|
"The history size must at least be 3 to compute first and second "
|
|
"derivatives.");
|
|
|
|
constexpr size_t kFeatureVectorSize = 42;
|
|
|
|
} // namespace rnn_vad
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_COMMON_H_
|