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This commit is contained in:
Ken Cooke 2015-09-23 12:07:17 -07:00
parent c4bc5233b1
commit 478eab3843
1 changed files with 22 additions and 38 deletions
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60
libraries/audio/src/AudioSRC.cpp
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@ -40,7 +40,7 @@ static const int SRC_BLOCK = 1024;
//#define SRC_DITHER
#define RAND16(r) (((r) = (r) * 69069u + 1u) >> 16)
// these are performance sensitive
#define lo32(a) (((uint32_t* )&(a))[0])
#define hi32(a) (((int32_t* )&(a))[1])
@ -65,8 +65,7 @@ static const int SRC_BLOCK = 1024;
//
// Portable aligned malloc/free
//
static void* aligned_malloc(size_t size, size_t alignment)
{
static void* aligned_malloc(size_t size, size_t alignment) {
if ((alignment & (alignment-1)) == 0) {
void* p = malloc(size + sizeof(void*) + (alignment-1));
if (p) {
@ -78,8 +77,7 @@ static void* aligned_malloc(size_t size, size_t alignment)
return NULL;
}
static void aligned_free(void* ptr)
{
static void aligned_free(void* ptr) {
if (ptr) {
void* p = ((void**)ptr)[-1];
free(p);
@ -92,8 +90,7 @@ static void aligned_free(void* ptr)
// Lagrange interpolation is maximally flat near dc and well suited
// for further upsampling our heavily-oversampled prototype filter.
//
static void cubicInterpolation(const float* input, float* output, int inputSize, int outputSize, float gain)
{
static void cubicInterpolation(const float* input, float* output, int inputSize, int outputSize, float gain) {
int64_t offset = 0;
int64_t step = ((int64_t)inputSize << 32) / outputSize; // Q32
@ -123,8 +120,7 @@ static void cubicInterpolation(const float* input, float* output, int inputSize,
}
}
int AudioSRC::createPolyphaseFilter(int upFactor, int downFactor, float gain)
{
int AudioSRC::createPolyphaseFilter(int upFactor, int downFactor, float gain) {
int numPhases = upFactor;
int numCoefs = PROTOTYPE_COEFS;
int numTaps = PROTOTYPE_TAPS;
@ -135,7 +131,7 @@ int AudioSRC::createPolyphaseFilter(int upFactor, int downFactor, float gain)
//
if (downFactor > upFactor) {
numCoefs = ((int64_t)PROTOTYPE_COEFS * downFactor) / upFactor;
numTaps = (numCoefs + upFactor-1) / upFactor;
numTaps = (numCoefs + upFactor - 1) / upFactor;
gain *= (float)PROTOTYPE_COEFS / numCoefs;
}
@ -153,7 +149,7 @@ int AudioSRC::createPolyphaseFilter(int upFactor, int downFactor, float gain)
for (int j = 0; j < numTaps; j++) {
// the filter taps are reversed, so convolution is implemented as dot-product
float f = tempFilter[(numTaps-j-1) * numPhases + phase];
float f = tempFilter[(numTaps - j - 1) * numPhases + phase];
_polyphaseFilter[numTaps * phase + j] = f;
}
}
@ -172,8 +168,7 @@ int AudioSRC::createPolyphaseFilter(int upFactor, int downFactor, float gain)
return numTaps;
}
int AudioSRC::multirateFilter1(const float* input0, float* output0, int inputFrames)
{
int AudioSRC::multirateFilter1(const float* input0, float* output0, int inputFrames) {
int outputFrames = 0;
for (; hi32(_offset) < inputFrames; _offset += _step) {
@ -205,8 +200,7 @@ int AudioSRC::multirateFilter1(const float* input0, float* output0, int inputFra
return outputFrames;
}
int AudioSRC::multirateFilter2(const float* input0, const float* input1, float* output0, float* output1, int inputFrames)
{
int AudioSRC::multirateFilter2(const float* input0, const float* input1, float* output0, float* output1, int inputFrames) {
int outputFrames = 0;
for (; hi32(_offset) < inputFrames; _offset += _step) {
@ -243,8 +237,7 @@ int AudioSRC::multirateFilter2(const float* input0, const float* input1, float*
// convert int16_t to float
// deinterleave stereo samples
void AudioSRC::convertInputFromInt16(const int16_t* input, float** outputs, int numFrames)
{
void AudioSRC::convertInputFromInt16(const int16_t* input, float** outputs, int numFrames) {
for (int i = 0; i < numFrames; i++) {
for (int j = 0; j < _numChannels; j++) {
@ -256,8 +249,7 @@ void AudioSRC::convertInputFromInt16(const int16_t* input, float** outputs, int
// convert float to int16_t
// interleave stereo samples
void AudioSRC::convertOutputToInt16(float** inputs, int16_t* output, int numFrames)
{
void AudioSRC::convertOutputToInt16(float** inputs, int16_t* output, int numFrames) {
for (int i = 0; i < numFrames; i++) {
for (int j = 0; j < _numChannels; j++) {
@ -277,13 +269,12 @@ void AudioSRC::convertOutputToInt16(float** inputs, int16_t* output, int numFram
f = std::min(f, 32767.0f);
f = std::max(f, -32768.0f);
output[_numChannels*i + j] = (int16_t)f;
output[_numChannels * i + j] = (int16_t)f;
}
}
}
int AudioSRC::processFloat(float** inputs, float** outputs, int inputFrames)
{
int AudioSRC::processFloat(float** inputs, float** outputs, int inputFrames) {
int outputFrames = 0;
int nh = std::min(_numHistory, inputFrames); // number of frames from history buffer
@ -336,8 +327,7 @@ int AudioSRC::processFloat(float** inputs, float** outputs, int inputFrames)
return outputFrames;
}
AudioSRC::AudioSRC(int inputSampleRate, int outputSampleRate, int numChannels)
{
AudioSRC::AudioSRC(int inputSampleRate, int outputSampleRate, int numChannels) {
assert(inputSampleRate > 0);
assert(outputSampleRate > 0);
assert(numChannels > 0);
@ -372,14 +362,13 @@ AudioSRC::AudioSRC(int inputSampleRate, int outputSampleRate, int numChannels)
_inputBlock = std::min(SRC_BLOCK, getMaxInput(SRC_BLOCK));
// reset the state
_offset = 0x80000000 + lo32(_step)/2; // optimum phase sampling for small integer ratios
_offset = 0x80000000 + lo32(_step)/2; // optimum subset of phases for small integer ratios
memset(_history[0], 0, 2 * _numHistory * sizeof(float));
memset(_history[1], 0, 2 * _numHistory * sizeof(float));
}
AudioSRC::~AudioSRC()
{
AudioSRC::~AudioSRC() {
aligned_free(_polyphaseFilter);
delete[] _history[0];
@ -394,8 +383,7 @@ AudioSRC::~AudioSRC()
//
// This version handles input/output as interleaved int16_t
//
int AudioSRC::render(const int16_t* input, int16_t* output, int inputFrames)
{
int AudioSRC::render(const int16_t* input, int16_t* output, int inputFrames) {
int outputFrames = 0;
while (inputFrames) {
@ -419,26 +407,22 @@ int AudioSRC::render(const int16_t* input, int16_t* output, int inputFrames)
}
// the min output frames that will be produced by inputFrames
int AudioSRC::getMinOutput(int inputFrames)
{
int AudioSRC::getMinOutput(int inputFrames) {
return (int)(((int64_t)inputFrames << 32) / _step);
}
// the max output frames that will be produced by inputFrames
int AudioSRC::getMaxOutput(int inputFrames)
{
return (int)((((int64_t)inputFrames << 32) + (_step - 1)) / _step);
int AudioSRC::getMaxOutput(int inputFrames) {
return (int)((((int64_t)inputFrames << 32) + _step - 1) / _step);
}
// the min input frames that will produce at least outputFrames
int AudioSRC::getMinInput(int outputFrames)
{
int AudioSRC::getMinInput(int outputFrames) {
return (int)((outputFrames * _step + 0xffffffffu) >> 32);
}
// the max input frames that will produce at most outputFrames
int AudioSRC::getMaxInput(int outputFrames)
{
int AudioSRC::getMaxInput(int outputFrames) {
return (int)((outputFrames * _step) >> 32);
}