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allocate 8x buffer size (for pulse audio) / more audioGain loop unrolling w/frame alignment check

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This commit is contained in:
Craig Hansen-Sturm 2014-09-08 15:28:55 -07:00
parent 04eca22a50
commit cb4aab3902
2 changed files with 81 additions and 35 deletions
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10
interface/src/Audio.cpp
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@ -403,6 +403,9 @@ void linearResampling(const int16_t* sourceSamples, int16_t* destinationSamples,
} }
} }
// $$$$$$$$$$$$$$ TEMPORARY $craighs$
float _loopbackOutputToInputRatio;
void Audio::start() { void Audio::start() {
// set up the desired audio format // set up the desired audio format
@ -431,7 +434,10 @@ void Audio::start() {
qDebug() << "Unable to set up audio output because of a problem with output format."; qDebug() << "Unable to set up audio output because of a problem with output format.";
} }
_inputFrameBuffer.initialize( _inputFormat.channelCount(), _audioInput->bufferSize() * 4 ); _loopbackOutputToInputRatio = (_outputFormat.sampleRate() / (float) _inputFormat.sampleRate())
* (_outputFormat.channelCount() / _inputFormat.channelCount());
_inputFrameBuffer.initialize( _inputFormat.channelCount(), _audioInput->bufferSize() * 8 );
_peq.initialize( _inputFormat.sampleRate() ); _peq.initialize( _inputFormat.sampleRate() );
_inputGain.initialize(); _inputGain.initialize();
_sourceGain.initialize(); _sourceGain.initialize();
@ -535,6 +541,8 @@ void Audio::handleAudioInput() {
float loopbackOutputToInputRatio = (_outputFormat.sampleRate() / (float) _inputFormat.sampleRate()) float loopbackOutputToInputRatio = (_outputFormat.sampleRate() / (float) _inputFormat.sampleRate())
* (_outputFormat.channelCount() / _inputFormat.channelCount()); * (_outputFormat.channelCount() / _inputFormat.channelCount());
assert(loopbackOutputToInputRatio == _loopbackOutputToInputRatio);
QByteArray loopBackByteArray(inputByteArray.size() * loopbackOutputToInputRatio, 0); QByteArray loopBackByteArray(inputByteArray.size() * loopbackOutputToInputRatio, 0);
linearResampling((int16_t*) inputByteArray.data(), (int16_t*) loopBackByteArray.data(), linearResampling((int16_t*) inputByteArray.data(), (int16_t*) loopBackByteArray.data(),

106
libraries/audio/src/AudioGain.h
View file

@ -55,40 +55,78 @@ public:
} }
float32_t** samples = frameBuffer.getFrameData(); float32_t** samples = frameBuffer.getFrameData();
for (uint16_t j = 0; j < frameBuffer.getChannelCount(); ++j) {
for (uint16_t i = 0; i < frameBuffer.getFrameCount(); i += 32) { bool frameAlignment16 = (frameBuffer.getFrameCount() & 0x0F) == 0;
samples[j][i + 0] *= _gain; if (frameAlignment16) {
samples[j][i + 1] *= _gain;
samples[j][i + 2] *= _gain; if (frameBuffer.getChannelCount() == 1) {
samples[j][i + 3] *= _gain;
samples[j][i + 4] *= _gain; for (uint16_t i = 0; i < frameBuffer.getFrameCount(); i += 16) {
samples[j][i + 5] *= _gain; samples[0][i + 0] *= _gain;
samples[j][i + 6] *= _gain; samples[0][i + 1] *= _gain;
samples[j][i + 7] *= _gain; samples[0][i + 2] *= _gain;
samples[j][i + 8] *= _gain; samples[0][i + 3] *= _gain;
samples[j][i + 9] *= _gain; samples[0][i + 4] *= _gain;
samples[j][i + 10] *= _gain; samples[0][i + 5] *= _gain;
samples[j][i + 11] *= _gain; samples[0][i + 6] *= _gain;
samples[j][i + 12] *= _gain; samples[0][i + 7] *= _gain;
samples[j][i + 13] *= _gain; samples[0][i + 8] *= _gain;
samples[j][i + 14] *= _gain; samples[0][i + 9] *= _gain;
samples[j][i + 15] *= _gain; samples[0][i + 10] *= _gain;
samples[j][i + 16] *= _gain; samples[0][i + 11] *= _gain;
samples[j][i + 17] *= _gain; samples[0][i + 12] *= _gain;
samples[j][i + 18] *= _gain; samples[0][i + 13] *= _gain;
samples[j][i + 19] *= _gain; samples[0][i + 14] *= _gain;
samples[j][i + 20] *= _gain; samples[0][i + 15] *= _gain;
samples[j][i + 21] *= _gain; }
samples[j][i + 22] *= _gain; }
samples[j][i + 23] *= _gain; else if (frameBuffer.getChannelCount() == 2) {
samples[j][i + 24] *= _gain;
samples[j][i + 25] *= _gain; for (uint16_t i = 0; i < frameBuffer.getFrameCount(); i += 16) {
samples[j][i + 26] *= _gain; samples[0][i + 0] *= _gain;
samples[j][i + 27] *= _gain; samples[0][i + 1] *= _gain;
samples[j][i + 28] *= _gain; samples[0][i + 2] *= _gain;
samples[j][i + 29] *= _gain; samples[0][i + 3] *= _gain;
samples[j][i + 30] *= _gain; samples[0][i + 4] *= _gain;
samples[j][i + 31] *= _gain; samples[0][i + 5] *= _gain;
samples[0][i + 6] *= _gain;
samples[0][i + 7] *= _gain;
samples[0][i + 8] *= _gain;
samples[0][i + 9] *= _gain;
samples[0][i + 10] *= _gain;
samples[0][i + 11] *= _gain;
samples[0][i + 12] *= _gain;
samples[0][i + 13] *= _gain;
samples[0][i + 14] *= _gain;
samples[0][i + 15] *= _gain;
samples[1][i + 0] *= _gain;
samples[1][i + 1] *= _gain;
samples[1][i + 2] *= _gain;
samples[1][i + 3] *= _gain;
samples[1][i + 4] *= _gain;
samples[1][i + 5] *= _gain;
samples[1][i + 6] *= _gain;
samples[1][i + 7] *= _gain;
samples[1][i + 8] *= _gain;
samples[1][i + 9] *= _gain;
samples[1][i + 10] *= _gain;
samples[1][i + 11] *= _gain;
samples[1][i + 12] *= _gain;
samples[1][i + 13] *= _gain;
samples[1][i + 14] *= _gain;
samples[1][i + 15] *= _gain;
}
}
else {
assert("unsupported channel format");
}
}
else {
for (uint16_t j = 0; j < frameBuffer.getChannelCount(); ++j) {
for (uint16_t i = 0; i < frameBuffer.getFrameCount(); i += 1) {
samples[j][i] *= _gain;
}
} }
} }
} }