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Added floating-point audio resampler

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This commit is contained in:
Ken Cooke 2016-07-01 18:50:53 -07:00
parent e784d7c400
commit d3c006b5da
2 changed files with 122 additions and 8 deletions
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126
libraries/audio/src/AudioSRC.cpp
View file

@ -539,6 +539,58 @@ void AudioSRC::convertOutputToInt16(float** inputs, int16_t* output, int numFram
}
}
// deinterleave stereo
void AudioSRC::convertInputFromFloat(const float* input, float** outputs, int numFrames) {
if (_numChannels == 1) {
memcpy(outputs[0], input, numFrames * sizeof(float));
} else if (_numChannels == 2) {
int i = 0;
for (; i < numFrames - 3; i += 4) {
__m128 f0 = _mm_loadu_ps(&input[2*i + 0]);
__m128 f1 = _mm_loadu_ps(&input[2*i + 4]);
// deinterleave
_mm_storeu_ps(&outputs[0][i], _mm_shuffle_ps(f0, f1, _MM_SHUFFLE(2,0,2,0)));
_mm_storeu_ps(&outputs[1][i], _mm_shuffle_ps(f0, f1, _MM_SHUFFLE(3,1,3,1)));
}
for (; i < numFrames; i++) {
// deinterleave
outputs[0][i] = input[2*i + 0];
outputs[1][i] = input[2*i + 1];
}
}
}
// interleave stereo
void AudioSRC::convertOutputToFloat(float** inputs, float* output, int numFrames) {
if (_numChannels == 1) {
memcpy(output, inputs[0], numFrames * sizeof(float));
} else if (_numChannels == 2) {
int i = 0;
for (; i < numFrames - 3; i += 4) {
__m128 f0 = _mm_loadu_ps(&inputs[0][i]);
__m128 f1 = _mm_loadu_ps(&inputs[1][i]);
// interleave
_mm_storeu_ps(&output[2*i + 0], _mm_unpacklo_ps(f0, f1));
_mm_storeu_ps(&output[2*i + 4], _mm_unpackhi_ps(f0, f1));
}
for (; i < numFrames; i++) {
// interleave
output[2*i + 0] = inputs[0][i];
output[2*i + 1] = inputs[1][i];
}
}
}
#else
int AudioSRC::multirateFilter1(const float* input0, float* output0, int inputFrames) {
@ -738,6 +790,38 @@ void AudioSRC::convertOutputToInt16(float** inputs, int16_t* output, int numFram
}
}
// deinterleave stereo
void AudioSRC::convertInputFromFloat(const float* input, float** outputs, int numFrames) {
if (_numChannels == 1) {
memcpy(outputs[0], input, numFrames * sizeof(float));
} else if (_numChannels == 2) {
for (int i = 0; i < numFrames; i++) {
// deinterleave
outputs[0][i] = input[2*i + 0];
outputs[1][i] = input[2*i + 1];
}
}
}
// interleave stereo
void AudioSRC::convertOutputToFloat(float** inputs, float* output, int numFrames) {
if (_numChannels == 1) {
memcpy(output, inputs[0], numFrames * sizeof(float));
} else if (_numChannels == 2) {
for (int i = 0; i < numFrames; i++) {
// interleave
output[2*i + 0] = inputs[0][i];
output[2*i + 1] = inputs[1][i];
}
}
}
#endif
int AudioSRC::processFloat(float** inputs, float** outputs, int inputFrames) {
@ -749,19 +833,19 @@ int AudioSRC::processFloat(float** inputs, float** outputs, int inputFrames) {
if (_numChannels == 1) {
// refill history buffers
memcpy(_history[0] + _numHistory, _inputs[0], nh * sizeof(float));
memcpy(_history[0] + _numHistory, inputs[0], nh * sizeof(float));
// process history buffer
outputFrames += multirateFilter1(_history[0], _outputs[0], nh);
outputFrames += multirateFilter1(_history[0], outputs[0], nh);
// process remaining input
if (ni) {
outputFrames += multirateFilter1(_inputs[0], _outputs[0] + outputFrames, ni);
outputFrames += multirateFilter1(inputs[0], outputs[0] + outputFrames, ni);
}
// shift history buffers
if (ni) {
memcpy(_history[0], _inputs[0] + ni, _numHistory * sizeof(float));
memcpy(_history[0], inputs[0] + ni, _numHistory * sizeof(float));
} else {
memmove(_history[0], _history[0] + nh, _numHistory * sizeof(float));
}
@ -769,15 +853,15 @@ int AudioSRC::processFloat(float** inputs, float** outputs, int inputFrames) {
} else if (_numChannels == 2) {
// refill history buffers
memcpy(_history[0] + _numHistory, _inputs[0], nh * sizeof(float));
memcpy(_history[1] + _numHistory, _inputs[1], nh * sizeof(float));
memcpy(_history[0] + _numHistory, inputs[0], nh * sizeof(float));
memcpy(_history[1] + _numHistory, inputs[1], nh * sizeof(float));
// process history buffer
outputFrames += multirateFilter2(_history[0], _history[1], _outputs[0], _outputs[1], nh);
outputFrames += multirateFilter2(_history[0], _history[1], outputs[0], outputs[1], nh);
// process remaining input
if (ni) {
outputFrames += multirateFilter2(_inputs[0], _inputs[1], _outputs[0] + outputFrames, _outputs[1] + outputFrames, ni);
outputFrames += multirateFilter2(inputs[0], inputs[1], outputs[0] + outputFrames, outputs[1] + outputFrames, ni);
}
// shift history buffers
@ -885,6 +969,32 @@ int AudioSRC::render(const int16_t* input, int16_t* output, int inputFrames) {
return outputFrames;
}
//
// This version handles input/output as interleaved float
//
int AudioSRC::render(const float* input, float* output, int inputFrames) {
int outputFrames = 0;
while (inputFrames) {
int ni = std::min(inputFrames, _inputBlock);
convertInputFromFloat(input, _inputs, ni);
int no = processFloat(_inputs, _outputs, ni);
assert(no <= SRC_BLOCK);
convertOutputToFloat(_outputs, output, no);
input += _numChannels * ni;
output += _numChannels * no;
inputFrames -= ni;
outputFrames += no;
}
return outputFrames;
}
// the min output frames that will be produced by inputFrames
int AudioSRC::getMinOutput(int inputFrames) {
if (_step == 0) {

4
libraries/audio/src/AudioSRC.h
View file

@ -35,6 +35,7 @@ public:
~AudioSRC();
int render(const int16_t* input, int16_t* output, int inputFrames);
int render(const float* input, float* output, int inputFrames);
int getMinOutput(int inputFrames);
int getMaxOutput(int inputFrames);
@ -78,6 +79,9 @@ private:
void convertInputFromInt16(const int16_t* input, float** outputs, int numFrames);
void convertOutputToInt16(float** inputs, int16_t* output, int numFrames);
void convertInputFromFloat(const float* input, float** outputs, int numFrames);
void convertOutputToFloat(float** inputs, float* output, int numFrames);
int processFloat(float** inputs, float** outputs, int inputFrames);
};