1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
|
/*
* Copyright (C) 2007 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#ifndef ANDROID_AUDIO_RESAMPLER_SINC_H
#define ANDROID_AUDIO_RESAMPLER_SINC_H
#include <stdint.h>
#include <sys/types.h>
#include <cutils/log.h>
#include "AudioResampler.h"
namespace android {
// ----------------------------------------------------------------------------
class AudioResamplerSinc : public AudioResampler {
public:
AudioResamplerSinc(int bitDepth, int inChannelCount, int32_t sampleRate);
~AudioResamplerSinc();
virtual void resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
private:
void init();
template<int CHANNELS>
void resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
template<int CHANNELS>
inline void filterCoefficient(
int32_t& l, int32_t& r, uint32_t phase, int16_t const *samples);
template<int CHANNELS>
inline void interpolate(
int32_t& l, int32_t& r,
int32_t const* coefs, int16_t lerp, int16_t const* samples);
template<int CHANNELS>
inline void read(int16_t*& impulse, uint32_t& phaseFraction,
int16_t const* in, size_t inputIndex);
int16_t *mState;
int16_t *mImpulse;
int16_t *mRingFull;
int32_t const * mFirCoefs;
static const int32_t mFirCoefsDown[];
static const int32_t mFirCoefsUp[];
// ----------------------------------------------------------------------------
static const int32_t RESAMPLE_FIR_NUM_COEF = 8;
static const int32_t RESAMPLE_FIR_LERP_INT_BITS = 4;
// we have 16 coefs samples per zero-crossing
static const int coefsBits = RESAMPLE_FIR_LERP_INT_BITS; // 4
static const int cShift = kNumPhaseBits - coefsBits; // 26
static const uint32_t cMask = ((1<<coefsBits)-1) << cShift; // 0xf<<26 = 3c00 0000
// and we use 15 bits to interpolate between these samples
// this cannot change because the mul below rely on it.
static const int pLerpBits = 15;
static const int pShift = kNumPhaseBits - coefsBits - pLerpBits; // 11
static const uint32_t pMask = ((1<<pLerpBits)-1) << pShift; // 0x7fff << 11
// number of zero-crossing on each side
static const unsigned int halfNumCoefs = RESAMPLE_FIR_NUM_COEF;
};
// ----------------------------------------------------------------------------
}; // namespace android
#endif /*ANDROID_AUDIO_RESAMPLER_SINC_H*/
|
