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Replace the linear interpolation embedded in the .WAV/.RAW loader with high quality polyphase resampling.

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When downsampling 48khz to 24khz, linear interpolation creates aliasing distortion that is quite audible.  This change greatly improves the audio quality for all sound assets.
This commit is contained in:
Ken Cooke 2016-06-03 18:39:43 -07:00
parent c85923c69b
commit 54293fdad0
1 changed files with 15 additions and 40 deletions
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55
libraries/audio/src/Sound.cpp
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@ -25,6 +25,8 @@
#include "AudioRingBuffer.h"
#include "AudioLogging.h"
#include "AudioSRC.h"
#include "Sound.h"
QScriptValue soundSharedPointerToScriptValue(QScriptEngine* engine, const SharedSoundPointer& in) {
@ -89,49 +91,22 @@ void Sound::downSample(const QByteArray& rawAudioByteArray) {
// we want to convert it to the format that the audio-mixer wants
// which is signed, 16-bit, 24Khz
int numSourceSamples = rawAudioByteArray.size() / sizeof(AudioConstants::AudioSample);
int numChannels = _isStereo ? 2 : 1;
AudioSRC resampler(48000, AudioConstants::SAMPLE_RATE, numChannels);
if (_isStereo && numSourceSamples % 2 != 0){
// in the unlikely case that we have stereo audio but we seem to be missing a sample
// (the sample for one channel is missing in a set of interleaved samples)
// then drop the odd sample
--numSourceSamples;
}
// resize to max possible output
int numSourceFrames = rawAudioByteArray.size() / (numChannels * sizeof(AudioConstants::AudioSample));
int maxDestinationFrames = resampler.getMaxOutput(numSourceFrames);
int maxDestinationBytes = maxDestinationFrames * numChannels * sizeof(AudioConstants::AudioSample);
_byteArray.resize(maxDestinationBytes);
int numDestinationSamples = numSourceSamples / 2.0f;
int numDestinationFrames = resampler.render((int16_t*)rawAudioByteArray.data(),
(int16_t*)_byteArray.data(),
numSourceFrames);
if (_isStereo && numDestinationSamples % 2 != 0) {
// if this is stereo we need to make sure we produce stereo output
// which means we should have an even number of output samples
numDestinationSamples += 1;
}
int numDestinationBytes = numDestinationSamples * sizeof(AudioConstants::AudioSample);
_byteArray.resize(numDestinationBytes);
int16_t* sourceSamples = (int16_t*) rawAudioByteArray.data();
int16_t* destinationSamples = (int16_t*) _byteArray.data();
if (_isStereo) {
for (int i = 0; i < numSourceSamples; i += 4) {
if (i + 2 >= numSourceSamples) {
destinationSamples[i / 2] = sourceSamples[i];
destinationSamples[(i / 2) + 1] = sourceSamples[i + 1];
} else {
destinationSamples[i / 2] = (sourceSamples[i] + sourceSamples[i + 2]) / 2;
destinationSamples[(i / 2) + 1] = (sourceSamples[i + 1] + sourceSamples[i + 3]) / 2;
}
}
} else {
for (int i = 1; i < numSourceSamples; i += 2) {
if (i + 1 >= numSourceSamples) {
destinationSamples[(i - 1) / 2] = (sourceSamples[i - 1] + sourceSamples[i]) / 2;
} else {
destinationSamples[(i - 1) / 2] = ((sourceSamples[i - 1] + sourceSamples[i + 1]) / 4) + (sourceSamples[i] / 2);
}
}
}
// truncate to actual output
int numDestinationBytes = numDestinationFrames * numChannels * sizeof(AudioConstants::AudioSample);
_byteArray.resize(maxDestinationBytes);
}
//