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Merge pull request #9273 from kencooke/audio-ambisonic-src2

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Improved audio resampler
This commit is contained in:
Andrew Meadows 2016-12-27 17:07:56 -08:00 committed by GitHub
commit b56a15b60a
3 changed files with 3164 additions and 544 deletions
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31
libraries/audio/src/AudioSRC.cpp
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@ -99,18 +99,22 @@ static void cubicInterpolation(const float* input, float* output, int inputSize,
}
}
int AudioSRC::createRationalFilter(int upFactor, int downFactor, float gain) {
int numTaps = PROTOTYPE_TAPS;
int AudioSRC::createRationalFilter(int upFactor, int downFactor, float gain, Quality quality) {
int prototypeTaps = prototypeFilterTable[quality].taps;
int prototypeCoefs = prototypeFilterTable[quality].coefs;
const float* prototypeFilter = prototypeFilterTable[quality].filter;
int numTaps = prototypeTaps;
int numPhases = upFactor;
int numCoefs = numTaps * numPhases;
int oldCoefs = numCoefs;
int prototypeCoefs = PROTOTYPE_TAPS * PROTOTYPE_PHASES;
//
// When downsampling, we can lower the filter cutoff by downFactor/upFactor using the
// time-scaling property of the Fourier transform. The gain is adjusted accordingly.
//
if (downFactor > upFactor) {
int oldCoefs = numCoefs;
numCoefs = ((int64_t)oldCoefs * downFactor) / upFactor;
numTaps = (numCoefs + upFactor - 1) / upFactor;
gain *= (float)oldCoefs / numCoefs;
@ -149,18 +153,22 @@ int AudioSRC::createRationalFilter(int upFactor, int downFactor, float gain) {
return numTaps;
}
int AudioSRC::createIrrationalFilter(int upFactor, int downFactor, float gain) {
int numTaps = PROTOTYPE_TAPS;
int AudioSRC::createIrrationalFilter(int upFactor, int downFactor, float gain, Quality quality) {
int prototypeTaps = prototypeFilterTable[quality].taps;
int prototypeCoefs = prototypeFilterTable[quality].coefs;
const float* prototypeFilter = prototypeFilterTable[quality].filter;
int numTaps = prototypeTaps;
int numPhases = upFactor;
int numCoefs = numTaps * numPhases;
int oldCoefs = numCoefs;
int prototypeCoefs = PROTOTYPE_TAPS * PROTOTYPE_PHASES;
//
// When downsampling, we can lower the filter cutoff by downFactor/upFactor using the
// time-scaling property of the Fourier transform. The gain is adjusted accordingly.
//
if (downFactor > upFactor) {
int oldCoefs = numCoefs;
numCoefs = ((int64_t)oldCoefs * downFactor) / upFactor;
numTaps = (numCoefs + upFactor - 1) / upFactor;
gain *= (float)oldCoefs / numCoefs;
@ -1405,7 +1413,8 @@ int AudioSRC::render(float** inputs, float** outputs, int inputFrames) {
return outputFrames;
}
AudioSRC::AudioSRC(int inputSampleRate, int outputSampleRate, int numChannels) {
AudioSRC::AudioSRC(int inputSampleRate, int outputSampleRate, int numChannels, Quality quality) {
assert(inputSampleRate > 0);
assert(outputSampleRate > 0);
assert(numChannels > 0);
@ -1433,9 +1442,9 @@ AudioSRC::AudioSRC(int inputSampleRate, int outputSampleRate, int numChannels) {
// create the polyphase filter
if (_step == 0) {
_numTaps = createRationalFilter(_upFactor, _downFactor, 1.0f);
_numTaps = createRationalFilter(_upFactor, _downFactor, 1.0f, quality);
} else {
_numTaps = createIrrationalFilter(_upFactor, _downFactor, 1.0f);
_numTaps = createIrrationalFilter(_upFactor, _downFactor, 1.0f, quality);
}
//printf("up=%d down=%.3f taps=%d\n", _upFactor, _downFactor + (LO32(_step)<<SRC_PHASEBITS) * Q32_TO_FLOAT, _numTaps);

14
libraries/audio/src/AudioSRC.h
View file

@ -17,7 +17,7 @@
static const int SRC_MAX_CHANNELS = 4;
// polyphase filter
static const int SRC_PHASEBITS = 8;
static const int SRC_PHASEBITS = 9;
static const int SRC_PHASES = (1 << SRC_PHASEBITS);
static const int SRC_FRACBITS = 32 - SRC_PHASEBITS;
static const uint32_t SRC_FRACMASK = (1 << SRC_FRACBITS) - 1;
@ -31,7 +31,13 @@ static const int SRC_BLOCK = 256;
class AudioSRC {
public:
AudioSRC(int inputSampleRate, int outputSampleRate, int numChannels);
enum Quality {
LOW_QUALITY,
MEDIUM_QUALITY,
HIGH_QUALITY
};
AudioSRC(int inputSampleRate, int outputSampleRate, int numChannels, Quality quality = MEDIUM_QUALITY);
~AudioSRC();
// deinterleaved float input/output (native format)
@ -70,8 +76,8 @@ private:
int64_t _offset;
int64_t _step;
int createRationalFilter(int upFactor, int downFactor, float gain);
int createIrrationalFilter(int upFactor, int downFactor, float gain);
int createRationalFilter(int upFactor, int downFactor, float gain, Quality quality);
int createIrrationalFilter(int upFactor, int downFactor, float gain, Quality quality);
int multirateFilter1(const float* input0, float* output0, int inputFrames);
int multirateFilter2(const float* input0, const float* input1, float* output0, float* output1, int inputFrames);

3663
libraries/audio/src/AudioSRCData.h
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