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
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
|
/*
* Copyright (C) 2013 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_DYN_H
#define ANDROID_AUDIO_RESAMPLER_DYN_H
#include <stdint.h>
#include <sys/types.h>
#include <cutils/log.h>
#include "AudioResampler.h"
namespace android {
/* AudioResamplerDyn
*
* This class template is used for floating point and integer resamplers.
*
* Type variables:
* TC = filter coefficient type (one of int16_t, int32_t, or float)
* TI = input data type (one of int16_t or float)
* TO = output data type (one of int32_t or float)
*
* For integer input data types TI, the coefficient type TC is either int16_t or int32_t.
* For float input data types TI, the coefficient type TC is float.
*/
template<typename TC, typename TI, typename TO>
class AudioResamplerDyn: public AudioResampler {
public:
AudioResamplerDyn(int inChannelCount,
int32_t sampleRate, src_quality quality);
virtual ~AudioResamplerDyn();
virtual void init();
virtual void setSampleRate(int32_t inSampleRate);
virtual void setVolume(float left, float right);
virtual size_t resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
private:
class Constants { // stores the filter constants.
public:
Constants() :
mL(0), mShift(0), mHalfNumCoefs(0), mFirCoefs(NULL)
{}
void set(int L, int halfNumCoefs,
int inSampleRate, int outSampleRate);
int mL; // interpolation phases in the filter.
int mShift; // right shift to get polyphase index
unsigned int mHalfNumCoefs; // filter half #coefs
const TC* mFirCoefs; // polyphase filter bank
};
class InBuffer { // buffer management for input type TI
public:
InBuffer();
~InBuffer();
void init();
void resize(int CHANNELS, int halfNumCoefs);
// used for direct management of the mImpulse pointer
inline TI* getImpulse() {
return mImpulse;
}
inline void setImpulse(TI *impulse) {
mImpulse = impulse;
}
template<int CHANNELS>
inline void readAgain(TI*& impulse, const int halfNumCoefs,
const TI* const in, const size_t inputIndex);
template<int CHANNELS>
inline void readAdvance(TI*& impulse, const int halfNumCoefs,
const TI* const in, const size_t inputIndex);
private:
// tuning parameter guidelines: 2 <= multiple <= 8
static const int kStateSizeMultipleOfFilterLength = 4;
// in general, mRingFull = mState + mStateSize - halfNumCoefs*CHANNELS.
TI* mState; // base pointer for the input buffer storage
TI* mImpulse; // current location of the impulse response (centered)
TI* mRingFull; // mState <= mImpulse < mRingFull
size_t mStateCount; // size of state in units of TI.
};
void createKaiserFir(Constants &c, double stopBandAtten,
int inSampleRate, int outSampleRate, double tbwCheat);
template<int CHANNELS, bool LOCKED, int STRIDE>
size_t resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider);
// define a pointer to member function type for resample
typedef size_t (AudioResamplerDyn<TC, TI, TO>::*resample_ABP_t)(TO* out,
size_t outFrameCount, AudioBufferProvider* provider);
// data - the contiguous storage and layout of these is important.
InBuffer mInBuffer;
Constants mConstants; // current set of coefficient parameters
TO __attribute__ ((aligned (8))) mVolumeSimd[2]; // must be aligned or NEON may crash
resample_ABP_t mResampleFunc; // called function for resampling
int32_t mFilterSampleRate; // designed filter sample rate.
src_quality mFilterQuality; // designed filter quality.
void* mCoefBuffer; // if a filter is created, this is not null
};
} // namespace android
#endif /*ANDROID_AUDIO_RESAMPLER_DYN_H*/
|
