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haikuwebkit/Source/WebCore/Modules/webaudio/AudioBufferSourceNode.cpp

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/*
* Copyright (C) 2010, Google Inc. All rights reserved.
* Copyright (C) 2020, Apple Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
* DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
* DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
* ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "config.h"
#if ENABLE(WEB_AUDIO)
#include "AudioBufferSourceNode.h"
#include "AudioBuffer.h"
#include "AudioContext.h"
#include "AudioNodeOutput.h"
#include "AudioParam.h"
#include "AudioUtilities.h"
#include "FloatConversion.h"
#include "ScriptExecutionContext.h"
#include <wtf/IsoMallocInlines.h>
namespace WebCore {
WTF_MAKE_ISO_ALLOCATED_IMPL(AudioBufferSourceNode);
constexpr double DefaultGrainDuration = 0.020; // 20ms
// Arbitrary upper limit on playback rate.
// Higher than expected rates can be useful when playing back oversampled buffers
// to minimize linear interpolation aliasing.
const double MaxRate = 1024;
static float computeSampleUsingLinearInterpolation(const float* source, unsigned readIndex, unsigned readIndex2, float interpolationFactor)
{
if (readIndex == readIndex2 && readIndex >= 1) {
// We're at the end of the buffer, so just linearly extrapolate from the last two samples.
float sample1 = source[readIndex - 1];
float sample2 = source[readIndex];
return sample2 + (sample2 - sample1) * interpolationFactor;
}
float sample1 = source[readIndex];
float sample2 = source[readIndex2];
return sample1 + interpolationFactor * (sample2 - sample1);
}
ExceptionOr<Ref<AudioBufferSourceNode>> AudioBufferSourceNode::create(BaseAudioContext& context, AudioBufferSourceOptions&& options)
{
auto node = adoptRef(*new AudioBufferSourceNode(context));
node->setBufferForBindings(WTFMove(options.buffer));
node->detune().setValue(options.detune);
node->setLoopForBindings(options.loop);
node->setLoopEndForBindings(options.loopEnd);
node->setLoopStartForBindings(options.loopStart);
node->playbackRate().setValue(options.playbackRate);
return node;
}
AudioBufferSourceNode::AudioBufferSourceNode(BaseAudioContext& context)
: AudioScheduledSourceNode(context, NodeTypeAudioBufferSource)
, m_detune(AudioParam::create(context, "detune"_s, 0.0, -FLT_MAX, FLT_MAX, AutomationRate::KRate, AutomationRateMode::Fixed))
, m_playbackRate(AudioParam::create(context, "playbackRate"_s, 1.0, -FLT_MAX, FLT_MAX, AutomationRate::KRate, AutomationRateMode::Fixed))
, m_grainDuration(DefaultGrainDuration)
{
// Default to mono. A call to setBuffer() will set the number of output channels to that of the buffer.
addOutput(1);
initialize();
}
AudioBufferSourceNode::~AudioBufferSourceNode()
{
uninitialize();
}
void AudioBufferSourceNode::process(size_t framesToProcess)
{
auto& outputBus = *output(0)->bus();
if (!isInitialized()) {
outputBus.zero();
return;
}
// The audio thread can't block on this lock, so we use tryLock() instead.
if (!m_processLock.tryLock()) {
// Too bad - tryLock() failed. We must be in the middle of changing buffers and were already outputting silence anyway.
outputBus.zero();
return;
}
Locker locker { AdoptLock, m_processLock };
if (!m_buffer) {
outputBus.zero();
return;
}
// After calling setBuffer() with a buffer having a different number of channels, there can in rare cases be a slight delay
// before the output bus is updated to the new number of channels because of use of tryLocks() in the context's updating system.
// In this case, if the buffer has just been changed and we're not quite ready yet, then just output silence.
if (numberOfChannels() != m_buffer->numberOfChannels()) {
outputBus.zero();
return;
}
size_t quantumFrameOffset = 0;
size_t bufferFramesToProcess = 0;
double startFrameOffset = 0;
updateSchedulingInfo(framesToProcess, outputBus, quantumFrameOffset, bufferFramesToProcess, startFrameOffset);
if (!bufferFramesToProcess) {
outputBus.zero();
return;
}
for (unsigned i = 0; i < outputBus.numberOfChannels(); ++i)
m_destinationChannels[i] = outputBus.channel(i)->mutableData();
// Render by reading directly from the buffer.
if (!renderFromBuffer(&outputBus, quantumFrameOffset, bufferFramesToProcess, startFrameOffset)) {
outputBus.zero();
return;
}
outputBus.clearSilentFlag();
}
// Returns true if we're finished.
bool AudioBufferSourceNode::renderSilenceAndFinishIfNotLooping(AudioBus*, unsigned index, size_t framesToProcess)
{
if (!m_isLooping) {
// If we're not looping, then stop playing when we get to the end.
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.
for (unsigned i = 0; i < numberOfChannels(); ++i)
memset(m_destinationChannels[i] + index, 0, sizeof(float) * framesToProcess);
}
if (!hasFinished())
finish();
return true;
}
return false;
}
bool AudioBufferSourceNode::renderFromBuffer(AudioBus* bus, unsigned destinationFrameOffset, size_t numberOfFrames, double startFrameOffset)
{
ASSERT(context().isAudioThread());
// Basic sanity checking
ASSERT(bus);
ASSERT(m_buffer);
if (!bus || !m_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 = m_buffer->length();
double bufferSampleRate = m_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 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 (m_isLooping && (m_loopStart || m_loopEnd) && m_loopStart >= 0 && m_loopEnd > 0 && m_loopStart < m_loopEnd) {
// Convert from seconds to sample-frames.
double loopMinFrame = m_loopStart * m_buffer->sampleRate();
double loopMaxFrame = m_loopEnd * m_buffer->sampleRate();
virtualMaxFrame = std::min(loopMaxFrame, virtualMaxFrame);
virtualMinFrame = std::max(loopMinFrame, virtualMinFrame);
virtualDeltaFrames = virtualMaxFrame - virtualMinFrame;
}
// 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 (m_isLooping && m_virtualReadIndex >= virtualMaxFrame) {
m_virtualReadIndex = (m_loopStart < 0) ? 0 : (m_loopStart * m_buffer->sampleRate());
m_virtualReadIndex = std::min(m_virtualReadIndex, static_cast<double>(bufferLength - 1));
}
// Sanity check that our playback rate isn't larger than the loop size.
if (std::abs(pitchRate) > virtualDeltaFrames)
return false;
// Get local copy.
double virtualReadIndex = m_virtualReadIndex;
// Adjust the read index by the startFrameOffset (compensated by the pitch rate) because
// we always start output on a frame boundary with interpolation if necessary.
if (startFrameOffset < 0 && pitchRate)
virtualReadIndex += std::abs(startFrameOffset * pitchRate);
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] + writeIndex, 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));
float interpolationFactor = virtualReadIndex - readIndex;
unsigned readIndex2 = readIndex + 1;
if (readIndex2 >= maxFrame)
readIndex2 = m_isLooping ? minFrame : readIndex;
// Linear interpolation.
for (unsigned i = 0; i < numberOfChannels; ++i) {
float* destination = destinationChannels[i];
const float* source = sourceChannels[i];
destination[writeIndex] = computeSampleUsingLinearInterpolation(source, readIndex, readIndex2, interpolationFactor);
}
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);
float interpolationFactor = virtualReadIndex - readIndex;
// For linear interpolation we need the next sample-frame too.
unsigned readIndex2 = readIndex + 1;
if (readIndex2 >= bufferLength) {
if (m_isLooping) {
// 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];
destination[writeIndex] = computeSampleUsingLinearInterpolation(source, readIndex, readIndex2, interpolationFactor);
}
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;
}
ExceptionOr<void> AudioBufferSourceNode::setBufferForBindings(RefPtr<AudioBuffer>&& buffer)
{
ASSERT(isMainThread());
DEBUG_LOG(LOGIDENTIFIER);
// The context must be locked since changing the buffer can re-configure the number of channels that are output.
Locker contextLocker { context().graphLock() };
// This synchronizes with process().
Locker locker { m_processLock };
if (buffer && m_wasBufferSet)
return Exception { InvalidStateError, "The buffer was already set"_s };
if (buffer) {
m_wasBufferSet = true;
// Do any necesssary re-configuration to the buffer's number of channels.
unsigned numberOfChannels = buffer->numberOfChannels();
ASSERT(numberOfChannels <= AudioContext::maxNumberOfChannels);
output(0)->setNumberOfChannels(numberOfChannels);
m_sourceChannels = makeUniqueArray<const float*>(numberOfChannels);
m_destinationChannels = makeUniqueArray<float*>(numberOfChannels);
for (unsigned i = 0; i < numberOfChannels; ++i)
m_sourceChannels[i] = buffer->channelData(i)->data();
}
m_virtualReadIndex = 0;
m_buffer = WTFMove(buffer);
// In case the buffer gets set after playback has started, we need to clamp the grain parameters now.
if (m_isGrain)
adjustGrainParameters();
return { };
}
unsigned AudioBufferSourceNode::numberOfChannels()
{
return output(0)->numberOfChannels();
}
ExceptionOr<void> AudioBufferSourceNode::startLater(double when, double grainOffset, std::optional<double> grainDuration)
{
return startPlaying(when, grainOffset, grainDuration);
}
void AudioBufferSourceNode::setLoopForBindings(bool looping)
{
ASSERT(isMainThread());
Locker locker { m_processLock };
m_isLooping = looping;
}
void AudioBufferSourceNode::setLoopStartForBindings(double loopStart)
{
ASSERT(isMainThread());
Locker locker { m_processLock };
m_loopStart = loopStart;
}
void AudioBufferSourceNode::setLoopEndForBindings(double loopEnd)
{
ASSERT(isMainThread());
Locker locker { m_processLock };
m_loopEnd = loopEnd;
}
ExceptionOr<void> AudioBufferSourceNode::startPlaying(double when, double grainOffset, std::optional<double> grainDuration)
{
ASSERT(isMainThread());
ALWAYS_LOG(LOGIDENTIFIER, "when = ", when, ", offset = ", grainOffset, ", duration = ", grainDuration.value_or(0));
if (m_playbackState != UNSCHEDULED_STATE)
return Exception { InvalidStateError, "Cannot call start more than once."_s };
if (!std::isfinite(when) || (when < 0))
return Exception { RangeError, "when value should be positive"_s };
if (!std::isfinite(grainOffset) || (grainOffset < 0))
return Exception { RangeError, "offset value should be positive"_s };
if (grainDuration && (!std::isfinite(*grainDuration) || (*grainDuration < 0)))
return Exception { RangeError, "duration value should be positive"_s };
context().sourceNodeWillBeginPlayback(*this);
// This synchronizes with process().
Locker locker { m_processLock };
m_isGrain = true;
m_grainOffset = grainOffset;
m_grainDuration = grainDuration.value_or(0);
m_wasGrainDurationGiven = !!grainDuration;
m_startTime = when;
adjustGrainParameters();
m_playbackState = SCHEDULED_STATE;
return { };
}
void AudioBufferSourceNode::adjustGrainParameters()
{
ASSERT(m_processLock.isHeld());
if (!m_buffer)
return;
// Do sanity checking of grain parameters versus buffer size.
double bufferDuration = m_buffer->duration();
m_grainOffset = std::min(bufferDuration, m_grainOffset);
if (!m_wasGrainDurationGiven)
m_grainDuration = bufferDuration - m_grainOffset;
if (m_wasGrainDurationGiven && m_isLooping) {
// We're looping a grain with a grain duration specified. Schedule the loop
// to stop after grainDuration seconds after starting, possibly running the
// loop multiple times if grainDuration is larger than the buffer duration.
// The net effect is as if the user called stop(when + grainDuration).
m_grainDuration = clampTo(m_grainDuration, 0.0, std::numeric_limits<double>::infinity());
m_endTime = m_startTime + m_grainDuration;
} else
m_grainDuration = clampTo(m_grainDuration, 0.0, bufferDuration - m_grainOffset);
// 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 (playbackRate().value() < 0)
m_virtualReadIndex = AudioUtilities::timeToSampleFrame(m_grainOffset + m_grainDuration, m_buffer->sampleRate()) - 1;
else
m_virtualReadIndex = AudioUtilities::timeToSampleFrame(m_grainOffset, m_buffer->sampleRate());
}
double AudioBufferSourceNode::totalPitchRate()
{
// Incorporate buffer's sample-rate versus AudioContext's sample-rate.
// Normally it's not an issue because buffers are loaded at the AudioContext's sample-rate, but we can handle it in any case.
double sampleRateFactor = 1.0;
if (m_buffer)
sampleRateFactor = m_buffer->sampleRate() / static_cast<double>(sampleRate());
double basePitchRate = playbackRate().finalValue();
double detune = pow(2, m_detune->finalValue() / 1200);
double totalRate = sampleRateFactor * basePitchRate * detune;
totalRate = std::clamp(totalRate, -MaxRate, MaxRate);
bool isTotalRateValid = !std::isnan(totalRate) && !std::isinf(totalRate);
ASSERT(isTotalRateValid);
if (!isTotalRateValid)
totalRate = 1.0;
return totalRate;
}
bool AudioBufferSourceNode::propagatesSilence() const
{
ASSERT(context().isAudioThread());
if (!isPlayingOrScheduled() || hasFinished())
return true;
if (!m_processLock.tryLock()) {
// We weren't able to get the lock so we assume we have a buffer.
return false;
}
Locker locker { AdoptLock, m_processLock };
return !m_buffer;
}
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO)
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