Result SoundSourceWV::tryOpen(const AudioSourceConfig& audioSrcCfg) {
DEBUG_ASSERT(!m_wpc);
char msg[80]; // hold possible error message
int openFlags = OPEN_WVC | OPEN_NORMALIZE;
if ((kChannelCountMono == audioSrcCfg.channelCountHint) ||
(kChannelCountStereo == audioSrcCfg.channelCountHint)) {
openFlags |= OPEN_2CH_MAX;
}
m_wpc = WavpackOpenFileInput(
getLocalFileNameBytes().constData(), msg, openFlags, 0);
if (!m_wpc) {
qDebug() << "SSWV::open: failed to open file : " << msg;
return ERR;
}
setChannelCount(WavpackGetReducedChannels(m_wpc));
setFrameRate(WavpackGetSampleRate(m_wpc));
setFrameCount(WavpackGetNumSamples(m_wpc));
if (WavpackGetMode(m_wpc) & MODE_FLOAT) {
m_sampleScaleFactor = CSAMPLE_PEAK;
} else {
const int bitsPerSample = WavpackGetBitsPerSample(m_wpc);
const uint32_t wavpackPeakSampleValue = uint32_t(1)
<< (bitsPerSample - 1);
m_sampleScaleFactor = CSAMPLE_PEAK / CSAMPLE(wavpackPeakSampleValue);
}
return OK;
}
SINT SoundSourceWV::readSampleFrames(
SINT numberOfFrames, CSAMPLE* sampleBuffer) {
if (sampleBuffer == nullptr) {
// NOTE(uklotzde): The WavPack API does not provide any
// functions for skipping samples in the audio stream. Calling
// API functions with a nullptr buffer does not return. Since
// we don't want to read samples into a temporary buffer that
// has to be allocated we are seeking to the position after
// the skipped samples.
SINT curFrameIndexBefore = m_curFrameIndex;
SINT curFrameIndexAfter = seekSampleFrame(m_curFrameIndex + numberOfFrames);
DEBUG_ASSERT(curFrameIndexBefore <= curFrameIndexAfter);
DEBUG_ASSERT(m_curFrameIndex == curFrameIndexAfter);
return curFrameIndexAfter - curFrameIndexBefore;
}
// static assert: sizeof(CSAMPLE) == sizeof(int32_t)
SINT unpackCount = WavpackUnpackSamples(m_wpc,
reinterpret_cast<int32_t*>(sampleBuffer), numberOfFrames);
DEBUG_ASSERT(unpackCount >= 0);
if (!(WavpackGetMode(m_wpc) & MODE_FLOAT)) {
// signed integer -> float
const SINT sampleCount = frames2samples(unpackCount);
for (SINT i = 0; i < sampleCount; ++i) {
const int32_t sampleValue =
reinterpret_cast<int32_t*>(sampleBuffer)[i];
sampleBuffer[i] = CSAMPLE(sampleValue) * m_sampleScaleFactor;
}
}
m_curFrameIndex += unpackCount;
return unpackCount;
}
void FlangerEffect::processChannel(const ChannelHandle& handle,
FlangerGroupState* pState,
const CSAMPLE* pInput, CSAMPLE* pOutput,
const unsigned int numSamples,
const unsigned int sampleRate,
const EffectProcessor::EnableState enableState,
const GroupFeatureState& groupFeatures) {
Q_UNUSED(handle);
Q_UNUSED(enableState);
Q_UNUSED(groupFeatures);
Q_UNUSED(sampleRate);
CSAMPLE lfoPeriod = m_pPeriodParameter->value();
CSAMPLE lfoDepth = m_pDepthParameter->value();
// Unused in EngineFlanger
// CSAMPLE lfoDelay = m_pDelayParameter ?
// m_pDelayParameter->value().toDouble() : 0.0f;
// TODO(rryan) check ranges
// period needs to be >=0
// delay needs to be >=0
// depth is ???
CSAMPLE* delayLeft = pState->delayLeft;
CSAMPLE* delayRight = pState->delayRight;
const int kChannels = 2;
for (unsigned int i = 0; i < numSamples; i += kChannels) {
delayLeft[pState->delayPos] = pInput[i];
delayRight[pState->delayPos] = pInput[i+1];
pState->delayPos = (pState->delayPos + 1) % kMaxDelay;
pState->time++;
if (pState->time > lfoPeriod) {
pState->time = 0;
}
CSAMPLE periodFraction = CSAMPLE(pState->time) / lfoPeriod;
CSAMPLE delay = kAverageDelayLength + kLfoAmplitude * sin(M_PI * 2.0f * periodFraction);
int framePrev = (pState->delayPos - int(delay) + kMaxDelay - 1) % kMaxDelay;
int frameNext = (pState->delayPos - int(delay) + kMaxDelay ) % kMaxDelay;
CSAMPLE prevLeft = delayLeft[framePrev];
CSAMPLE nextLeft = delayLeft[frameNext];
CSAMPLE prevRight = delayRight[framePrev];
CSAMPLE nextRight = delayRight[frameNext];
CSAMPLE frac = delay - floorf(delay);
CSAMPLE delayedSampleLeft = prevLeft + frac * (nextLeft - prevLeft);
CSAMPLE delayedSampleRight = prevRight + frac * (nextRight - prevRight);
pOutput[i] = pInput[i] + lfoDepth * delayedSampleLeft;
pOutput[i+1] = pInput[i+1] + lfoDepth * delayedSampleRight;
}
}
SINT SoundSourceWV::readSampleFrames(
SINT numberOfFrames, CSAMPLE* sampleBuffer) {
// static assert: sizeof(CSAMPLE) == sizeof(int32_t)
SINT unpackCount = WavpackUnpackSamples(m_wpc,
reinterpret_cast<int32_t*>(sampleBuffer), numberOfFrames);
if (!(WavpackGetMode(m_wpc) & MODE_FLOAT)) {
// signed integer -> float
const SINT sampleCount = frames2samples(unpackCount);
for (SINT i = 0; i < sampleCount; ++i) {
const int32_t sampleValue =
reinterpret_cast<int32_t*>(sampleBuffer)[i];
sampleBuffer[i] = CSAMPLE(sampleValue) * m_sampleScaleFactor;
}
}
return unpackCount;
}
void SoundSourceFLAC::flacMetadata(const FLAC__StreamMetadata* metadata) {
// https://xiph.org/flac/api/group__flac__stream__decoder.html#ga43e2329c15731c002ac4182a47990f85
// "...one STREAMINFO block, followed by zero or more other metadata blocks."
// "...by default the decoder only calls the metadata callback for the STREAMINFO block..."
// "...always before the first audio frame (i.e. write callback)."
switch (metadata->type) {
case FLAC__METADATA_TYPE_STREAMINFO:
{
setChannelCount(metadata->data.stream_info.channels);
setSampleRate(metadata->data.stream_info.sample_rate);
initFrameIndexRangeOnce(
IndexRange::forward(
0,
metadata->data.stream_info.total_samples));
const unsigned bitsPerSample = metadata->data.stream_info.bits_per_sample;
DEBUG_ASSERT(kBitsPerSampleDefault != bitsPerSample);
if (kBitsPerSampleDefault == m_bitsPerSample) {
// not set before
m_bitsPerSample = bitsPerSample;
m_sampleScaleFactor = CSAMPLE_PEAK
/ CSAMPLE(FLAC__int32(1) << bitsPerSample);
} else {
// already set before -> check for consistency
if (bitsPerSample != m_bitsPerSample) {
kLogger.warning() << "Unexpected bits per sample:"
<< bitsPerSample << " <> " << m_bitsPerSample;
}
}
m_maxBlocksize = metadata->data.stream_info.max_blocksize;
if (0 >= m_maxBlocksize) {
kLogger.warning() << "Invalid max. blocksize" << m_maxBlocksize;
}
const SINT sampleBufferCapacity =
m_maxBlocksize * channelCount();
if (m_sampleBuffer.capacity() < sampleBufferCapacity) {
m_sampleBuffer.adjustCapacity(sampleBufferCapacity);
}
break;
}
default:
// Ignore all other metadata types
break;
}
}
ReadableSampleFrames SoundSourceWV::readSampleFramesClamped(
WritableSampleFrames writableSampleFrames) {
const SINT firstFrameIndex = writableSampleFrames.frameIndexRange().start();
if (m_curFrameIndex != firstFrameIndex) {
if (WavpackSeekSample(m_wpc, firstFrameIndex)) {
m_curFrameIndex = firstFrameIndex;
} else {
kLogger.warning()
<< "Could not seek to first frame index"
<< firstFrameIndex;
m_curFrameIndex = WavpackGetSampleIndex(m_wpc);
return ReadableSampleFrames(IndexRange::between(m_curFrameIndex, m_curFrameIndex));
}
}
DEBUG_ASSERT(m_curFrameIndex == firstFrameIndex);
const SINT numberOfFramesTotal = writableSampleFrames.frameLength();
static_assert(sizeof(CSAMPLE) == sizeof(int32_t),
"CSAMPLE and int32_t must have the same size");
CSAMPLE* pOutputBuffer = writableSampleFrames.writableData();
SINT unpackCount = WavpackUnpackSamples(m_wpc,
reinterpret_cast<int32_t*>(pOutputBuffer), numberOfFramesTotal);
DEBUG_ASSERT(unpackCount >= 0);
DEBUG_ASSERT(unpackCount <= numberOfFramesTotal);
if (!(WavpackGetMode(m_wpc) & MODE_FLOAT)) {
// signed integer -> float
const SINT sampleCount = frames2samples(unpackCount);
for (SINT i = 0; i < sampleCount; ++i) {
const int32_t sampleValue =
*reinterpret_cast<int32_t*>(pOutputBuffer);
*pOutputBuffer++ = CSAMPLE(sampleValue) * m_sampleScaleFactor;
}
}
const auto resultRange = IndexRange::forward(m_curFrameIndex, unpackCount);
m_curFrameIndex += unpackCount;
return ReadableSampleFrames(
resultRange,
SampleBuffer::ReadableSlice(
writableSampleFrames.writableData(),
frames2samples(unpackCount)));
}
void AutoPanEffect::processChannel(
const ChannelHandle& handle, AutoPanGroupState* pGroupState,
const CSAMPLE* pInput, CSAMPLE* pOutput,
const mixxx::EngineParameters& bufferParameters,
const EffectEnableState enableState,
const GroupFeatureState& groupFeatures) {
Q_UNUSED(handle);
if (enableState == EffectEnableState::Disabled) {
return;
}
AutoPanGroupState& gs = *pGroupState;
double width = m_pWidthParameter->value();
double period = m_pPeriodParameter->value();
double smoothing = 0.5 - m_pSmoothingParameter->value();
if (groupFeatures.has_beat_length_sec) {
// period is a number of beats
double beats = std::max(roundToFraction(period, 2), 0.25);
period = beats * groupFeatures.beat_length_sec * bufferParameters.sampleRate();
// TODO(xxx) sync phase
//if (groupFeatures.has_beat_fraction) {
} else {
// period is a number of seconds
period = std::max(period, 0.25) * bufferParameters.sampleRate();
}
// When the period is changed, the position of the sound shouldn't
// so time need to be recalculated
if (gs.m_dPreviousPeriod != -1.0) {
gs.time *= period / gs.m_dPreviousPeriod;
}
gs.m_dPreviousPeriod = period;
if (gs.time >= period || enableState == EffectEnableState::Enabling) {
gs.time = 0;
}
// Normally, the position goes from 0 to 1 linearly. Here we make steps at
// 0.25 and 0.75 to have the sound fully on the right or fully on the left.
// At the end, the "position" value can describe a sinusoid or a square
// curve depending of the size of those steps.
// coef of the slope
// a = (y2 - y1) / (x2 - x1)
// 1 / ( 1 - 2 * stepfrac)
float a = smoothing != 0.5f ? 1.0f / (1.0f - smoothing * 2.0f) : 1.0f;
// size of a segment of slope (controlled by the "smoothing" parameter)
float u = (0.5f - smoothing) / 2.0f;
gs.frac.setRampingThreshold(kPositionRampingThreshold);
double sinusoid = 0;
// NOTE: Assuming engine is working in stereo.
for (unsigned int i = 0; i + 1 < bufferParameters.samplesPerBuffer(); i += 2) {
CSAMPLE periodFraction = CSAMPLE(gs.time) / period;
// current quarter in the trigonometric circle
float quarter = floorf(periodFraction * 4.0f);
// part of the period fraction being a step (not in the slope)
CSAMPLE stepsFractionPart = floorf((quarter + 1.0f) / 2.0f) * smoothing;
// float inInterval = fmod( periodFraction, (period / 2.0) );
float inStepInterval = fmod(periodFraction, 0.5f);
CSAMPLE angleFraction;
if (inStepInterval > u && inStepInterval < (u + smoothing)) {
// at full left or full right
angleFraction = quarter < 2.0f ? 0.25f : 0.75f;
} else {
// in the slope (linear function)
angleFraction = (periodFraction - stepsFractionPart) * a;
}
// transforms the angleFraction into a sinusoid.
// The width parameter modulates the two limits. if width values 0.5,
// the limits will be 0.25 and 0.75. If it's 0, it will be 0.5 and 0.5
// so the sound will be stuck at the center. If it values 1, the limits
// will be 0 and 1 (full left and full right).
sinusoid = sin(M_PI * 2.0f * angleFraction) * width;
gs.frac.setWithRampingApplied((sinusoid + 1.0f) / 2.0f);
// apply the delay
gs.delay->process(&pInput[i], &pOutput[i],
-0.005 * math_clamp(((gs.frac * 2.0) - 1.0f), -1.0, 1.0) * bufferParameters.sampleRate());
double lawCoef = computeLawCoefficient(sinusoid);
pOutput[i] *= gs.frac * lawCoef;
pOutput[i+1] *= (1.0f - gs.frac) * lawCoef;
gs.time++;
while (gs.time >= period) {
// Click for debug
//pOutput[i] = 1.0f;
//pOutput[i+1] = 1.0f;
// The while loop is required in case period changes the value
gs.time -= period;
}
}
}
void AutoPanEffect::processChannel(const ChannelHandle& handle, PanGroupState* pGroupState,
const CSAMPLE* pInput,
CSAMPLE* pOutput, const unsigned int numSamples,
const unsigned int sampleRate,
const EffectProcessor::EnableState enableState,
const GroupFeatureState& groupFeatures) {
Q_UNUSED(handle);
if (enableState == EffectProcessor::DISABLED) {
return;
}
PanGroupState& gs = *pGroupState;
double width = m_pWidthParameter->value();
double period = m_pPeriodParameter->value();
double periodUnit = m_pPeriodUnitParameter->value();
double smoothing = 0.5-m_pSmoothingParameter->value();
// When the period knob is between max and max-1, the time is paused.
// Time shouldn't be paused while enabling state as the sound
// will be stuck in the middle even if the smoothing is at max.
bool timePaused = period > m_pPeriodParameter->maximum() - 1
&& enableState != EffectProcessor::ENABLING;
if (periodUnit == 1 && groupFeatures.has_beat_length) {
// floor the param on one of these values :
// 1/16, 1/8, 1/4, 1/2, 1, 2, 4, 8, 16, 32, 64, 128
int i = 0;
while (period > m_pPeriodParameter->minimum()) {
period /= 2;
i++;
}
double beats = m_pPeriodParameter->minimum();
while (i != 0.0) {
beats *= 2;
i--;
}
period = groupFeatures.beat_length * beats;
} else {
// max period is 128 seconds
period *= sampleRate;
}
// When the period is changed, the position of the sound shouldn't
// so time need to be recalculated
if (gs.m_dPreviousPeriod != -1.0) {
gs.time *= period / gs.m_dPreviousPeriod;
}
gs.m_dPreviousPeriod = period;
if (gs.time > period || enableState == EffectProcessor::ENABLING) {
gs.time = 0;
}
// Normally, the position goes from 0 to 1 linearly. Here we make steps at
// 0.25 and 0.75 to have the sound fully on the right or fully on the left.
// At the end, the "position" value can describe a sinusoid or a square
// curve depending of the size of those steps.
// coef of the slope
// a = (y2 - y1) / (x2 - x1)
// 1 / ( 1 - 2 * stepfrac)
float a = smoothing != 0.5f ? 1.0f / (1.0f - smoothing * 2.0f) : 1.0f;
// size of a segment of slope (controled by the "smoothing" parameter)
float u = (0.5f - smoothing) / 2.0f;
gs.frac.setRampingThreshold(kPositionRampingThreshold);
double sinusoid = 0;
for (unsigned int i = 0; i + 1 < numSamples; i += 2) {
CSAMPLE periodFraction = CSAMPLE(gs.time) / period;
// current quarter in the trigonometric circle
float quarter = floorf(periodFraction * 4.0f);
// part of the period fraction being a step (not in the slope)
CSAMPLE stepsFractionPart = floorf((quarter + 1.0f) / 2.0f) * smoothing;
// float inInterval = fmod( periodFraction, (period / 2.0) );
float inStepInterval = fmod(periodFraction, 0.5f);
CSAMPLE angleFraction;
if (inStepInterval > u && inStepInterval < (u + smoothing)) {
// at full left or full right
angleFraction = quarter < 2.0f ? 0.25f : 0.75f;
} else {
// in the slope (linear function)
angleFraction = (periodFraction - stepsFractionPart) * a;
}
// transforms the angleFraction into a sinusoid.
// The width parameter modulates the two limits. if width values 0.5,
// the limits will be 0.25 and 0.75. If it's 0, it will be 0.5 and 0.5
// so the sound will be stuck at the center. If it values 1, the limits
// will be 0 and 1 (full left and full right).
sinusoid = sin(M_PI * 2.0f * angleFraction) * width;
gs.frac.setWithRampingApplied((sinusoid + 1.0f) / 2.0f);
// apply the delay
gs.delay->process(&pInput[i], &pOutput[i],
-0.005 * math_clamp(((gs.frac * 2.0) - 1.0f), -1.0, 1.0) * sampleRate);
double lawCoef = computeLawCoefficient(sinusoid);
pOutput[i] *= gs.frac * lawCoef;
pOutput[i+1] *= (1.0f - gs.frac) * lawCoef;
// The time shouldn't be paused if the position has not its
// expected value due to ramping
if (!timePaused || gs.frac.ramped) {
gs.time++;
}
}
}
void SoundSourceFLAC::flacMetadata(const FLAC__StreamMetadata* metadata) {
// https://xiph.org/flac/api/group__flac__stream__decoder.html#ga43e2329c15731c002ac4182a47990f85
// "...one STREAMINFO block, followed by zero or more other metadata blocks."
// "...by default the decoder only calls the metadata callback for the STREAMINFO block..."
// "...always before the first audio frame (i.e. write callback)."
switch (metadata->type) {
case FLAC__METADATA_TYPE_STREAMINFO:
{
const SINT channelCount = metadata->data.stream_info.channels;
if (isValidChannelCount(channelCount)) {
if (hasChannelCount()) {
// already set before -> check for consistency
if (getChannelCount() != channelCount) {
qWarning() << "Unexpected channel count:"
<< channelCount << " <> " << getChannelCount();
}
} else {
// not set before
setChannelCount(channelCount);
}
} else {
qWarning() << "Invalid channel count:"
<< channelCount;
}
const SINT samplingRate = metadata->data.stream_info.sample_rate;
if (isValidSamplingRate(samplingRate)) {
if (hasSamplingRate()) {
// already set before -> check for consistency
if (getSamplingRate() != samplingRate) {
qWarning() << "Unexpected sampling rate:"
<< samplingRate << " <> " << getSamplingRate();
}
} else {
// not set before
setSamplingRate(samplingRate);
}
} else {
qWarning() << "Invalid sampling rate:"
<< samplingRate;
}
const SINT frameCount = metadata->data.stream_info.total_samples;
DEBUG_ASSERT(isValidFrameCount(frameCount));
if (isEmpty()) {
// not set before
setFrameCount(frameCount);
} else {
// already set before -> check for consistency
if (getFrameCount() != frameCount) {
qWarning() << "Unexpected frame count:"
<< frameCount << " <> " << getFrameCount();
}
}
const unsigned bitsPerSample = metadata->data.stream_info.bits_per_sample;
DEBUG_ASSERT(kBitsPerSampleDefault != bitsPerSample);
if (kBitsPerSampleDefault == m_bitsPerSample) {
// not set before
m_bitsPerSample = bitsPerSample;
m_sampleScaleFactor = CSAMPLE_PEAK
/ CSAMPLE(FLAC__int32(1) << bitsPerSample);
} else {
// already set before -> check for consistency
if (bitsPerSample != m_bitsPerSample) {
qWarning() << "Unexpected bits per sample:"
<< bitsPerSample << " <> " << m_bitsPerSample;
}
}
m_maxBlocksize = metadata->data.stream_info.max_blocksize;
if (0 >= m_maxBlocksize) {
qWarning() << "Invalid max. blocksize" << m_maxBlocksize;
}
const SINT sampleBufferCapacity =
m_maxBlocksize * getChannelCount();
m_sampleBuffer.resetCapacity(sampleBufferCapacity);
break;
}
default:
// Ignore all other metadata types
break;
}
}