void AudioBufferSourceNode::startPlaying(BufferPlaybackMode playbackMode, double when, double grainOffset, double grainDuration, ExceptionCode& ec)
{
ASSERT(isMainThread());
context()->nodeWillBeginPlayback();
if (m_playbackState != UNSCHEDULED_STATE) {
ec = INVALID_STATE_ERR;
return;
}
if (!std::isfinite(when) || (when < 0)) {
ec = INVALID_STATE_ERR;
return;
}
if (!std::isfinite(grainOffset) || (grainOffset < 0)) {
ec = INVALID_STATE_ERR;
return;
}
if (!std::isfinite(grainDuration) || (grainDuration < 0)) {
ec = INVALID_STATE_ERR;
return;
}
if (!buffer())
return;
m_isGrain = playbackMode == Partial;
if (m_isGrain) {
// Do sanity checking of grain parameters versus buffer size.
double bufferDuration = buffer()->duration();
m_grainOffset = std::min(bufferDuration, grainOffset);
double maxDuration = bufferDuration - m_grainOffset;
m_grainDuration = std::min(maxDuration, grainDuration);
} else {
m_grainOffset = 0.0;
m_grainDuration = buffer()->duration();
}
m_startTime = when;
// We call timeToSampleFrame here since at playbackRate == 1 we don't want to go through linear interpolation
// at a sub-sample position since it will degrade the quality.
// When aligned to the sample-frame the playback will be identical to the PCM data stored in the buffer.
// Since playbackRate == 1 is very common, it's worth considering quality.
if (totalPitchRate() < 0)
m_virtualReadIndex = AudioUtilities::timeToSampleFrame(m_grainOffset + m_grainDuration, buffer()->sampleRate()) - 1;
else
m_virtualReadIndex = AudioUtilities::timeToSampleFrame(m_grainOffset, buffer()->sampleRate());
m_playbackState = SCHEDULED_STATE;
}
void AudioBufferSourceNode::renderFromBuffer(AudioBus* bus, unsigned destinationFrameOffset, size_t numberOfFrames)
{
ASSERT(context()->isAudioThread());
// Basic sanity checking
ASSERT(bus);
ASSERT(buffer());
if (!bus || !buffer())
return;
unsigned numberOfChannels = this->numberOfChannels();
unsigned busNumberOfChannels = bus->numberOfChannels();
bool channelCountGood = numberOfChannels && numberOfChannels == busNumberOfChannels;
ASSERT(channelCountGood);
if (!channelCountGood)
return;
// Sanity check destinationFrameOffset, numberOfFrames.
size_t destinationLength = bus->length();
bool isLengthGood = destinationLength <= 4096 && numberOfFrames <= 4096;
ASSERT(isLengthGood);
if (!isLengthGood)
return;
bool isOffsetGood = destinationFrameOffset <= destinationLength && destinationFrameOffset + numberOfFrames <= destinationLength;
ASSERT(isOffsetGood);
if (!isOffsetGood)
return;
// Potentially zero out initial frames leading up to the offset.
if (destinationFrameOffset) {
for (unsigned i = 0; i < numberOfChannels; ++i)
memset(m_destinationChannels[i], 0, sizeof(float) * destinationFrameOffset);
}
// Offset the pointers to the correct offset frame.
unsigned writeIndex = destinationFrameOffset;
size_t bufferLength = buffer()->length();
double bufferSampleRate = buffer()->sampleRate();
// Calculate the start and end frames in our buffer that we want to play.
// If m_isGrain is true, then we will be playing a portion of the total buffer.
unsigned startFrame = m_isGrain ? AudioUtilities::timeToSampleFrame(m_grainOffset, bufferSampleRate) : 0;
// Avoid converting from time to sample-frames twice by computing
// the grain end time first before computing the sample frame.
unsigned endFrame = m_isGrain ? AudioUtilities::timeToSampleFrame(m_grainOffset + m_grainDuration, bufferSampleRate) : bufferLength;
ASSERT(endFrame >= startFrame);
if (endFrame < startFrame)
return;
unsigned deltaFrames = endFrame - startFrame;
// This is a HACK to allow for HRTF tail-time - avoids glitch at end.
// FIXME: implement tailTime for each AudioNode for a more general solution to this problem.
if (m_isGrain)
endFrame += 512;
// Do some sanity checking.
if (startFrame >= bufferLength)
startFrame = !bufferLength ? 0 : bufferLength - 1;
if (endFrame > bufferLength)
endFrame = bufferLength;
if (m_virtualReadIndex >= endFrame)
m_virtualReadIndex = startFrame; // reset to start
double pitchRate = totalPitchRate();
// Get local copy.
double virtualReadIndex = m_virtualReadIndex;
// Render loop - reading from the source buffer to the destination using linear interpolation.
int framesToProcess = numberOfFrames;
const float** sourceChannels = m_sourceChannels.get();
float** destinationChannels = m_destinationChannels.get();
// Optimize for the very common case of playing back with pitchRate == 1.
// We can avoid the linear interpolation.
if (pitchRate == 1 && virtualReadIndex == floor(virtualReadIndex)) {
unsigned readIndex = static_cast<unsigned>(virtualReadIndex);
while (framesToProcess > 0) {
int framesToEnd = endFrame - readIndex;
int framesThisTime = min(framesToProcess, framesToEnd);
framesThisTime = max(0, framesThisTime);
for (unsigned i = 0; i < numberOfChannels; ++i)
memcpy(destinationChannels[i] + writeIndex, sourceChannels[i] + readIndex, sizeof(float) * framesThisTime);
writeIndex += framesThisTime;
readIndex += framesThisTime;
framesToProcess -= framesThisTime;
// Wrap-around.
if (readIndex >= endFrame) {
readIndex -= deltaFrames;
if (renderSilenceAndFinishIfNotLooping(bus, writeIndex, framesToProcess))
break;
}
}
virtualReadIndex = readIndex;
} else {
while (framesToProcess--) {
unsigned readIndex = static_cast<unsigned>(virtualReadIndex);
double interpolationFactor = virtualReadIndex - readIndex;
// For linear interpolation we need the next sample-frame too.
unsigned readIndex2 = readIndex + 1;
if (readIndex2 >= endFrame) {
if (loop()) {
// Make sure to wrap around at the end of the buffer.
readIndex2 -= deltaFrames;
} else
readIndex2 = readIndex;
}
// Final sanity check on buffer access.
// FIXME: as an optimization, try to get rid of this inner-loop check and put assertions and guards before the loop.
if (readIndex >= bufferLength || readIndex2 >= bufferLength)
break;
// Linear interpolation.
for (unsigned i = 0; i < numberOfChannels; ++i) {
float* destination = destinationChannels[i];
const float* source = sourceChannels[i];
double sample1 = source[readIndex];
double sample2 = source[readIndex2];
double sample = (1.0 - interpolationFactor) * sample1 + interpolationFactor * sample2;
destination[writeIndex] = narrowPrecisionToFloat(sample);
}
writeIndex++;
virtualReadIndex += pitchRate;
// Wrap-around, retaining sub-sample position since virtualReadIndex is floating-point.
if (virtualReadIndex >= endFrame) {
virtualReadIndex -= deltaFrames;
if (renderSilenceAndFinishIfNotLooping(bus, writeIndex, framesToProcess))
break;
}
}
}
m_virtualReadIndex = virtualReadIndex;
}
void AudioBufferSourceNode::renderFromBuffer(AudioBus* bus, unsigned destinationFrameOffset, size_t numberOfFrames)
{
ASSERT(context()->isAudioThread());
// Basic sanity checking
ASSERT(bus);
ASSERT(buffer());
if (!bus || !buffer())
return;
unsigned numberOfChannels = this->numberOfChannels();
unsigned busNumberOfChannels = bus->numberOfChannels();
// FIXME: we can add support for sources with more than two channels, but this is not a common case.
bool channelCountGood = numberOfChannels == busNumberOfChannels && (numberOfChannels == 1 || numberOfChannels == 2);
ASSERT(channelCountGood);
if (!channelCountGood)
return;
// Get the destination pointers.
float* destinationL = bus->channel(0)->data();
ASSERT(destinationL);
if (!destinationL)
return;
float* destinationR = (numberOfChannels < 2) ? 0 : bus->channel(1)->data();
bool isStereo = destinationR;
// Sanity check destinationFrameOffset, numberOfFrames.
size_t destinationLength = bus->length();
bool isLengthGood = destinationLength <= 4096 && numberOfFrames <= 4096;
ASSERT(isLengthGood);
if (!isLengthGood)
return;
bool isOffsetGood = destinationFrameOffset <= destinationLength && destinationFrameOffset + numberOfFrames <= destinationLength;
ASSERT(isOffsetGood);
if (!isOffsetGood)
return;
// Potentially zero out initial frames leading up to the offset.
if (destinationFrameOffset) {
memset(destinationL, 0, sizeof(float) * destinationFrameOffset);
if (destinationR)
memset(destinationR, 0, sizeof(float) * destinationFrameOffset);
}
// Offset the pointers to the correct offset frame.
destinationL += destinationFrameOffset;
if (destinationR)
destinationR += destinationFrameOffset;
size_t bufferLength = buffer()->length();
double bufferSampleRate = buffer()->sampleRate();
// Calculate the start and end frames in our buffer that we want to play.
// If m_isGrain is true, then we will be playing a portion of the total buffer.
unsigned startFrame = m_isGrain ? static_cast<unsigned>(m_grainOffset * bufferSampleRate) : 0;
unsigned endFrame = m_isGrain ? static_cast<unsigned>(startFrame + m_grainDuration * bufferSampleRate) : bufferLength;
ASSERT(endFrame >= startFrame);
if (endFrame < startFrame)
return;
unsigned deltaFrames = endFrame - startFrame;
// This is a HACK to allow for HRTF tail-time - avoids glitch at end.
// FIXME: implement tailTime for each AudioNode for a more general solution to this problem.
if (m_isGrain)
endFrame += 512;
// Do some sanity checking.
if (startFrame >= bufferLength)
startFrame = !bufferLength ? 0 : bufferLength - 1;
if (endFrame > bufferLength)
endFrame = bufferLength;
if (m_virtualReadIndex >= endFrame)
m_virtualReadIndex = startFrame; // reset to start
// Get pointers to the start of the sample buffer.
float* sourceL = m_buffer->getChannelData(0)->data();
float* sourceR = m_buffer->numberOfChannels() == 2 ? m_buffer->getChannelData(1)->data() : 0;
double pitchRate = totalPitchRate();
// Get local copy.
double virtualReadIndex = m_virtualReadIndex;
// Render loop - reading from the source buffer to the destination using linear interpolation.
// FIXME: optimize for the very common case of playing back with pitchRate == 1.
// We can avoid the linear interpolation.
int framesToProcess = numberOfFrames;
while (framesToProcess--) {
unsigned readIndex = static_cast<unsigned>(virtualReadIndex);
double interpolationFactor = virtualReadIndex - readIndex;
// For linear interpolation we need the next sample-frame too.
unsigned readIndex2 = readIndex + 1;
if (readIndex2 >= endFrame) {
if (loop()) {
// Make sure to wrap around at the end of the buffer.
readIndex2 -= deltaFrames;
} else
readIndex2 = readIndex;
}
// Final sanity check on buffer access.
// FIXME: as an optimization, try to get rid of this inner-loop check and put assertions and guards before the loop.
if (readIndex >= bufferLength || readIndex2 >= bufferLength)
break;
// Linear interpolation.
double sampleL1 = sourceL[readIndex];
double sampleL2 = sourceL[readIndex2];
double sampleL = (1.0 - interpolationFactor) * sampleL1 + interpolationFactor * sampleL2;
*destinationL++ = narrowPrecisionToFloat(sampleL);
if (isStereo) {
double sampleR1 = sourceR[readIndex];
double sampleR2 = sourceR[readIndex2];
double sampleR = (1.0 - interpolationFactor) * sampleR1 + interpolationFactor * sampleR2;
*destinationR++ = narrowPrecisionToFloat(sampleR);
}
virtualReadIndex += pitchRate;
// Wrap-around, retaining sub-sample position since virtualReadIndex is floating-point.
if (virtualReadIndex >= endFrame) {
virtualReadIndex -= deltaFrames;
if (!loop()) {
// If we're not looping, then stop playing when we get to the end.
m_isPlaying = false;
if (framesToProcess > 0) {
// We're not looping and we've reached the end of the sample data, but we still need to provide more output,
// so generate silence for the remaining.
memset(destinationL, 0, sizeof(float) * framesToProcess);
if (isStereo)
memset(destinationR, 0, sizeof(float) * framesToProcess);
}
finish();
break;
}
}
}
m_virtualReadIndex = virtualReadIndex;
}
bool AudioBufferSourceNode::renderFromBuffer(AudioBus* bus, unsigned destinationFrameOffset, size_t numberOfFrames)
{
ASSERT(context()->isAudioThread());
// Basic sanity checking
ASSERT(bus);
ASSERT(buffer());
if (!bus || !buffer())
return false;
unsigned numberOfChannels = this->numberOfChannels();
unsigned busNumberOfChannels = bus->numberOfChannels();
bool channelCountGood = numberOfChannels && numberOfChannels == busNumberOfChannels;
ASSERT(channelCountGood);
if (!channelCountGood)
return false;
// Sanity check destinationFrameOffset, numberOfFrames.
size_t destinationLength = bus->length();
bool isLengthGood = destinationLength <= 4096 && numberOfFrames <= 4096;
ASSERT(isLengthGood);
if (!isLengthGood)
return false;
bool isOffsetGood = destinationFrameOffset <= destinationLength && destinationFrameOffset + numberOfFrames <= destinationLength;
ASSERT(isOffsetGood);
if (!isOffsetGood)
return false;
// Potentially zero out initial frames leading up to the offset.
if (destinationFrameOffset) {
for (unsigned i = 0; i < numberOfChannels; ++i)
memset(m_destinationChannels[i], 0, sizeof(float) * destinationFrameOffset);
}
// Offset the pointers to the correct offset frame.
unsigned writeIndex = destinationFrameOffset;
size_t bufferLength = buffer()->length();
double bufferSampleRate = buffer()->sampleRate();
// Avoid converting from time to sample-frames twice by computing
// the grain end time first before computing the sample frame.
unsigned endFrame = m_isGrain ? AudioUtilities::timeToSampleFrame(m_grainOffset + m_grainDuration, bufferSampleRate) : bufferLength;
// This is a HACK to allow for HRTF tail-time - avoids glitch at end.
// FIXME: implement tailTime for each AudioNode for a more general solution to this problem.
// https://bugs.webkit.org/show_bug.cgi?id=77224
if (m_isGrain)
endFrame += 512;
// Do some sanity checking.
if (endFrame > bufferLength)
endFrame = bufferLength;
// If the .loop attribute is true, then values of m_loopStart == 0 && m_loopEnd == 0 implies
// that we should use the entire buffer as the loop, otherwise use the loop values in m_loopStart and m_loopEnd.
double virtualEndFrame = endFrame;
double virtualDeltaFrames = endFrame;
if (loop() && (m_loopStart || m_loopEnd) && m_loopStart >= 0 && m_loopEnd > 0 && m_loopStart < m_loopEnd) {
// Convert from seconds to sample-frames.
double loopStartFrame = m_loopStart * buffer()->sampleRate();
double loopEndFrame = m_loopEnd * buffer()->sampleRate();
virtualEndFrame = std::min(loopEndFrame, virtualEndFrame);
virtualDeltaFrames = virtualEndFrame - loopStartFrame;
}
// If we're looping and the offset (virtualReadIndex) is past the end of the loop, wrap back to
// the beginning of the loop. For other cases, nothing needs to be done.
if (loop() && m_virtualReadIndex >= virtualEndFrame)
m_virtualReadIndex = (m_loopStart < 0) ? 0 : (m_loopStart * buffer()->sampleRate());
double pitchRate = totalPitchRate();
// Sanity check that our playback rate isn't larger than the loop size.
if (pitchRate > virtualDeltaFrames)
return false;
// Get local copy.
double virtualReadIndex = m_virtualReadIndex;
// Render loop - reading from the source buffer to the destination using linear interpolation.
int framesToProcess = numberOfFrames;
const float** sourceChannels = m_sourceChannels.get();
float** destinationChannels = m_destinationChannels.get();
ASSERT(virtualReadIndex >= 0);
ASSERT(virtualDeltaFrames >= 0);
ASSERT(virtualEndFrame >= 0);
// Optimize for the very common case of playing back with pitchRate == 1.
// We can avoid the linear interpolation.
if (pitchRate == 1 && virtualReadIndex == floor(virtualReadIndex)
&& virtualDeltaFrames == floor(virtualDeltaFrames)
&& virtualEndFrame == floor(virtualEndFrame)) {
unsigned readIndex = static_cast<unsigned>(virtualReadIndex);
unsigned deltaFrames = static_cast<unsigned>(virtualDeltaFrames);
endFrame = static_cast<unsigned>(virtualEndFrame);
while (framesToProcess > 0) {
int framesToEnd = endFrame - readIndex;
int framesThisTime = std::min(framesToProcess, framesToEnd);
framesThisTime = std::max(0, framesThisTime);
for (unsigned i = 0; i < numberOfChannels; ++i)
memcpy(destinationChannels[i] + writeIndex, sourceChannels[i] + readIndex, sizeof(float) * framesThisTime);
writeIndex += framesThisTime;
readIndex += framesThisTime;
framesToProcess -= framesThisTime;
// It can happen that framesThisTime is 0. Assert that we will actually exit the loop in
// this case. framesThisTime is 0 only if readIndex >= endFrame;
ASSERT(framesThisTime ? true : readIndex >= endFrame);
// Wrap-around.
if (readIndex >= endFrame) {
readIndex -= deltaFrames;
if (renderSilenceAndFinishIfNotLooping(bus, writeIndex, framesToProcess))
break;
}
}
virtualReadIndex = readIndex;
} else {
while (framesToProcess--) {
unsigned readIndex = static_cast<unsigned>(virtualReadIndex);
double interpolationFactor = virtualReadIndex - readIndex;
// For linear interpolation we need the next sample-frame too.
unsigned readIndex2 = readIndex + 1;
if (readIndex2 >= bufferLength) {
if (loop()) {
// Make sure to wrap around at the end of the buffer.
readIndex2 = static_cast<unsigned>(virtualReadIndex + 1 - virtualDeltaFrames);
} else
readIndex2 = readIndex;
}
// Final sanity check on buffer access.
// FIXME: as an optimization, try to get rid of this inner-loop check and put assertions and guards before the loop.
if (readIndex >= bufferLength || readIndex2 >= bufferLength)
break;
// Linear interpolation.
for (unsigned i = 0; i < numberOfChannels; ++i) {
float* destination = destinationChannels[i];
const float* source = sourceChannels[i];
double sample1 = source[readIndex];
double sample2 = source[readIndex2];
double sample = (1.0 - interpolationFactor) * sample1 + interpolationFactor * sample2;
destination[writeIndex] = narrowPrecisionToFloat(sample);
}
writeIndex++;
virtualReadIndex += pitchRate;
// Wrap-around, retaining sub-sample position since virtualReadIndex is floating-point.
if (virtualReadIndex >= virtualEndFrame) {
virtualReadIndex -= virtualDeltaFrames;
if (renderSilenceAndFinishIfNotLooping(bus, writeIndex, framesToProcess))
break;
}
}
}
bus->clearSilentFlag();
m_virtualReadIndex = virtualReadIndex;
return true;
}
bool AudioBufferSourceNode::renderFromBuffer(AudioBus* bus, unsigned destinationFrameOffset, size_t numberOfFrames)
{
ASSERT(context()->isAudioThread());
// Basic sanity checking
ASSERT(bus);
ASSERT(buffer());
if (!bus || !buffer())
return false;
unsigned numberOfChannels = this->numberOfChannels();
unsigned busNumberOfChannels = bus->numberOfChannels();
bool channelCountGood = numberOfChannels && numberOfChannels == busNumberOfChannels;
ASSERT(channelCountGood);
if (!channelCountGood)
return false;
// Sanity check destinationFrameOffset, numberOfFrames.
size_t destinationLength = bus->length();
bool isLengthGood = destinationLength <= 4096 && numberOfFrames <= 4096;
ASSERT(isLengthGood);
if (!isLengthGood)
return false;
bool isOffsetGood = destinationFrameOffset <= destinationLength && destinationFrameOffset + numberOfFrames <= destinationLength;
ASSERT(isOffsetGood);
if (!isOffsetGood)
return false;
// Potentially zero out initial frames leading up to the offset.
if (destinationFrameOffset) {
for (unsigned i = 0; i < numberOfChannels; ++i)
memset(m_destinationChannels[i], 0, sizeof(float) * destinationFrameOffset);
}
// Offset the pointers to the correct offset frame.
unsigned writeIndex = destinationFrameOffset;
size_t bufferLength = buffer()->length();
double bufferSampleRate = buffer()->sampleRate();
double pitchRate = totalPitchRate();
bool reverse = pitchRate < 0;
// Avoid converting from time to sample-frames twice by computing
// the grain end time first before computing the sample frame.
unsigned maxFrame;
if (m_isGrain)
maxFrame = AudioUtilities::timeToSampleFrame(m_grainOffset + m_grainDuration, bufferSampleRate);
else
maxFrame = bufferLength;
// Do some sanity checking.
if (maxFrame > bufferLength)
maxFrame = bufferLength;
if (reverse && m_virtualReadIndex <= 0)
m_virtualReadIndex = maxFrame - 1;
else if (!reverse && m_virtualReadIndex >= maxFrame)
m_virtualReadIndex = 0; // reset to start
// If the .loop attribute is true, then values of m_loopStart == 0 && m_loopEnd == 0 implies
// that we should use the entire buffer as the loop, otherwise use the loop values in m_loopStart and m_loopEnd.
double virtualMaxFrame = maxFrame;
double virtualMinFrame = 0;
double virtualDeltaFrames = maxFrame;
if (loop() && (m_loopStart || m_loopEnd) && m_loopStart >= 0 && m_loopEnd > 0 && m_loopStart < m_loopEnd) {
// Convert from seconds to sample-frames.
double loopMinFrame = m_loopStart * buffer()->sampleRate();
double loopMaxFrame = m_loopEnd * buffer()->sampleRate();
virtualMaxFrame = std::min(loopMaxFrame, virtualMaxFrame);
virtualMinFrame = std::max(loopMinFrame, virtualMinFrame);
virtualDeltaFrames = virtualMaxFrame - virtualMinFrame;
}
// Sanity check that our playback rate isn't larger than the loop size.
if (fabs(pitchRate) >= virtualDeltaFrames)
return false;
// Get local copy.
double virtualReadIndex = m_virtualReadIndex;
bool needsInterpolation = virtualReadIndex != floor(virtualReadIndex)
|| virtualDeltaFrames != floor(virtualDeltaFrames)
|| virtualMaxFrame != floor(virtualMaxFrame)
|| virtualMinFrame != floor(virtualMinFrame);
// Render loop - reading from the source buffer to the destination using linear interpolation.
int framesToProcess = numberOfFrames;
const float** sourceChannels = m_sourceChannels.get();
float** destinationChannels = m_destinationChannels.get();
// Optimize for the very common case of playing back with pitchRate == 1.
// We can avoid the linear interpolation.
if (pitchRate == 1 && !needsInterpolation) {
unsigned readIndex = static_cast<unsigned>(virtualReadIndex);
unsigned deltaFrames = static_cast<unsigned>(virtualDeltaFrames);
maxFrame = static_cast<unsigned>(virtualMaxFrame);
while (framesToProcess > 0) {
int framesToEnd = maxFrame - readIndex;
int framesThisTime = std::min(framesToProcess, framesToEnd);
framesThisTime = std::max(0, framesThisTime);
for (unsigned i = 0; i < numberOfChannels; ++i)
memcpy(destinationChannels[i] + writeIndex, sourceChannels[i] + readIndex, sizeof(float) * framesThisTime);
writeIndex += framesThisTime;
readIndex += framesThisTime;
framesToProcess -= framesThisTime;
// Wrap-around.
if (readIndex >= maxFrame) {
readIndex -= deltaFrames;
if (renderSilenceAndFinishIfNotLooping(bus, writeIndex, framesToProcess))
break;
}
}
virtualReadIndex = readIndex;
} else if (pitchRate == -1 && !needsInterpolation) {
int readIndex = static_cast<int>(virtualReadIndex);
int deltaFrames = static_cast<int>(virtualDeltaFrames);
int minFrame = static_cast<int>(virtualMinFrame) - 1;
while (framesToProcess > 0) {
int framesToEnd = readIndex - minFrame;
int framesThisTime = std::min<int>(framesToProcess, framesToEnd);
framesThisTime = std::max<int>(0, framesThisTime);
while (framesThisTime--) {
for (unsigned i = 0; i < numberOfChannels; ++i) {
float* destination = destinationChannels[i];
const float* source = sourceChannels[i];
destination[writeIndex] = source[readIndex];
}
++writeIndex;
--readIndex;
--framesToProcess;
}
// Wrap-around.
if (readIndex <= minFrame) {
readIndex += deltaFrames;
if (renderSilenceAndFinishIfNotLooping(bus, writeIndex, framesToProcess))
break;
}
}
virtualReadIndex = readIndex;
} else if (!pitchRate) {
unsigned readIndex = static_cast<unsigned>(virtualReadIndex);
for (unsigned i = 0; i < numberOfChannels; ++i)
std::fill_n(destinationChannels[i], framesToProcess, sourceChannels[i][readIndex]);
} else if (reverse) {
unsigned maxFrame = static_cast<unsigned>(virtualMaxFrame);
unsigned minFrame = static_cast<unsigned>(floorf(virtualMinFrame));
while (framesToProcess--) {
unsigned readIndex = static_cast<unsigned>(floorf(virtualReadIndex));
double interpolationFactor = virtualReadIndex - readIndex;
unsigned readIndex2 = readIndex + 1;
if (readIndex2 >= maxFrame)
readIndex2 = loop() ? minFrame : maxFrame - 1;
// Linear interpolation.
for (unsigned i = 0; i < numberOfChannels; ++i) {
float* destination = destinationChannels[i];
const float* source = sourceChannels[i];
double sample1 = source[readIndex];
double sample2 = source[readIndex2];
double sample = (1.0 - interpolationFactor) * sample1 + interpolationFactor * sample2;
destination[writeIndex] = narrowPrecisionToFloat(sample);
}
writeIndex++;
virtualReadIndex += pitchRate;
// Wrap-around, retaining sub-sample position since virtualReadIndex is floating-point.
if (virtualReadIndex < virtualMinFrame) {
virtualReadIndex += virtualDeltaFrames;
if (renderSilenceAndFinishIfNotLooping(bus, writeIndex, framesToProcess))
break;
}
}
} else {
while (framesToProcess--) {
unsigned readIndex = static_cast<unsigned>(virtualReadIndex);
double interpolationFactor = virtualReadIndex - readIndex;
// For linear interpolation we need the next sample-frame too.
unsigned readIndex2 = readIndex + 1;
if (readIndex2 >= bufferLength) {
if (loop()) {
// Make sure to wrap around at the end of the buffer.
readIndex2 = static_cast<unsigned>(virtualReadIndex + 1 - virtualDeltaFrames);
} else
readIndex2 = readIndex;
}
// Final sanity check on buffer access.
// FIXME: as an optimization, try to get rid of this inner-loop check and put assertions and guards before the loop.
if (readIndex >= bufferLength || readIndex2 >= bufferLength)
break;
// Linear interpolation.
for (unsigned i = 0; i < numberOfChannels; ++i) {
float* destination = destinationChannels[i];
const float* source = sourceChannels[i];
double sample1 = source[readIndex];
double sample2 = source[readIndex2];
double sample = (1.0 - interpolationFactor) * sample1 + interpolationFactor * sample2;
destination[writeIndex] = narrowPrecisionToFloat(sample);
}
writeIndex++;
virtualReadIndex += pitchRate;
// Wrap-around, retaining sub-sample position since virtualReadIndex is floating-point.
if (virtualReadIndex >= virtualMaxFrame) {
virtualReadIndex -= virtualDeltaFrames;
if (renderSilenceAndFinishIfNotLooping(bus, writeIndex, framesToProcess))
break;
}
}
}
bus->clearSilentFlag();
m_virtualReadIndex = virtualReadIndex;
return true;
}
