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/*
* Copyright (C) 2010 Google Inc. All rights reserved.
* Copyright (C) 2017-2021 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.
* 3. Neither the name of Apple Inc. ("Apple") nor the names of
* its contributors may be used to endorse or promote products derived
* from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY APPLE 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 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 "AudioBuffer.h"
#include "AudioContext.h"
#include "AudioFileReader.h"
#include <JavaScriptCore/JSCInlines.h>
#include <JavaScriptCore/TypedArrayInlines.h>
namespace WebCore {
RefPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, LegacyPreventDetaching preventDetaching)
{
if (!BaseAudioContext::isSupportedSampleRate(sampleRate) || !numberOfChannels || numberOfChannels > AudioContext::maxNumberOfChannels || !numberOfFrames)
return nullptr;
auto buffer = adoptRef(*new AudioBuffer(numberOfChannels, numberOfFrames, sampleRate, preventDetaching));
if (!buffer->originalLength())
return nullptr;
return buffer;
}
ExceptionOr<Ref<AudioBuffer>> AudioBuffer::create(const AudioBufferOptions& options)
{
if (!options.numberOfChannels)
return Exception { NotSupportedError, "Number of channels cannot be 0."_s };
if (options.numberOfChannels > AudioContext::maxNumberOfChannels)
return Exception { NotSupportedError, "Number of channels cannot be more than max supported."_s };
if (!options.length)
return Exception { NotSupportedError, "Length must be at least 1."_s };
if (!BaseAudioContext::isSupportedSampleRate(options.sampleRate))
return Exception { NotSupportedError, "Sample rate is not in the supported range."_s };
auto buffer = adoptRef(*new AudioBuffer(options.numberOfChannels, options.length, options.sampleRate));
if (!buffer->originalLength())
return Exception { NotSupportedError, "Channel was not able to be created."_s };
return buffer;
}
RefPtr<AudioBuffer> AudioBuffer::createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float sampleRate)
{
RefPtr<AudioBus> bus = createBusFromInMemoryAudioFile(data, dataSize, mixToMono, sampleRate);
if (!bus)
return nullptr;
return adoptRef(*new AudioBuffer(*bus));
}
AudioBuffer::AudioBuffer(unsigned numberOfChannels, size_t length, float sampleRate, LegacyPreventDetaching preventDetaching)
: m_sampleRate(sampleRate)
, m_originalLength(length)
, m_isDetachable(preventDetaching == LegacyPreventDetaching::No)
{
if (static_cast<uint64_t>(m_originalLength) > s_maxLength) {
invalidate();
return;
}
m_channels.reserveCapacity(numberOfChannels);
for (unsigned i = 0; i < numberOfChannels; ++i) {
auto channelDataArray = Float32Array::tryCreate(m_originalLength);
if (!channelDataArray) {
invalidate();
break;
}
if (preventDetaching == LegacyPreventDetaching::Yes)
channelDataArray->setDetachable(false);
m_channels.append(WTFMove(channelDataArray));
}
m_channelWrappers.resize(m_channels.size());
}
AudioBuffer::AudioBuffer(AudioBus& bus)
: m_sampleRate(bus.sampleRate())
, m_originalLength(bus.length())
{
if (static_cast<uint64_t>(m_originalLength) > s_maxLength) {
invalidate();
return;
}
// Copy audio data from the bus to the Float32Arrays we manage.
unsigned numberOfChannels = bus.numberOfChannels();
m_channels.reserveCapacity(numberOfChannels);
for (unsigned i = 0; i < numberOfChannels; ++i) {
auto channelDataArray = Float32Array::tryCreate(m_originalLength);
if (!channelDataArray) {
invalidate();
break;
}
channelDataArray->setRange(bus.channel(i)->data(), m_originalLength, 0);
m_channels.append(WTFMove(channelDataArray));
}
m_channelWrappers.resize(m_channels.size());
}
void AudioBuffer::invalidate()
{
releaseMemory();
m_originalLength = 0;
}
void AudioBuffer::releaseMemory()
{
Locker locker { m_channelsLock };
m_channels.clear();
m_channelWrappers.clear();
}
ExceptionOr<JSC::JSValue> AudioBuffer::getChannelData(JSDOMGlobalObject& globalObject, unsigned channelIndex)
{
ASSERT(m_channelWrappers.size() == m_channels.size());
if (channelIndex >= m_channelWrappers.size())
return Exception { IndexSizeError, "Index must be less than number of channels."_s };
auto& channelData = m_channels[channelIndex];
auto constructJSArray = [&] {
return JSC::JSFloat32Array::create(globalObject.vm(), globalObject.typedArrayStructure(JSC::TypeFloat32), channelData.copyRef());
};
if (globalObject.worldIsNormal()) {
if (!m_channelWrappers[channelIndex])
m_channelWrappers[channelIndex] = { constructJSArray() };
return static_cast<JSC::JSValue>(m_channelWrappers[channelIndex]);
}
return constructJSArray();
}
template<typename Visitor>
void AudioBuffer::visitChannelWrappers(Visitor& visitor)
{
for (auto& channelWrapper : m_channelWrappers)
channelWrapper.visit(visitor);
}
template void AudioBuffer::visitChannelWrappers(JSC::AbstractSlotVisitor&);
template void AudioBuffer::visitChannelWrappers(JSC::SlotVisitor&);
RefPtr<Float32Array> AudioBuffer::channelData(unsigned channelIndex)
{
if (channelIndex >= m_channels.size())
return nullptr;
if (hasDetachedChannelBuffer())
return Float32Array::create(0);
return m_channels[channelIndex].copyRef();
}
float* AudioBuffer::rawChannelData(unsigned channelIndex)
{
if (channelIndex >= m_channels.size())
return nullptr;
if (hasDetachedChannelBuffer())
return nullptr;
return m_channels[channelIndex]->data();
}
ExceptionOr<void> AudioBuffer::copyFromChannel(Ref<Float32Array>&& destination, unsigned channelNumber, unsigned bufferOffset)
{
if (destination->isShared())
return Exception { TypeError, "Destination may not be a shared buffer."_s };
if (channelNumber >= m_channels.size())
return Exception { IndexSizeError, "Not a valid channelNumber."_s };
Float32Array* channelData = m_channels[channelNumber].get();
size_t dataLength = channelData->length();
if (bufferOffset >= dataLength)
return { };
size_t count = dataLength - bufferOffset;
count = std::min(destination.get().length(), count);
const float* src = channelData->data();
float* dst = destination->data();
ASSERT(src);
ASSERT(dst);
memmove(dst, src + bufferOffset, count * sizeof(*src));
return { };
}
ExceptionOr<void> AudioBuffer::copyToChannel(Ref<Float32Array>&& source, unsigned channelNumber, unsigned bufferOffset)
{
if (source->isShared())
return Exception { TypeError, "Source may not be a shared buffer."_s };
if (channelNumber >= m_channels.size())
return Exception { IndexSizeError, "Not a valid channelNumber."_s };
Float32Array* channelData = m_channels[channelNumber].get();
size_t dataLength = channelData->length();
if (bufferOffset >= dataLength)
return { };
size_t count = dataLength - bufferOffset;
count = std::min(source.get().length(), count);
const float* src = source->data();
float* dst = channelData->data();
ASSERT(src);
ASSERT(dst);
memmove(dst + bufferOffset, src, count * sizeof(*dst));
return { };
}
void AudioBuffer::zero()
{
for (auto& channel : m_channels)
channel->zeroFill();
}
size_t AudioBuffer::memoryCost() const
{
// memoryCost() may be invoked concurrently from a GC thread, and we need to be careful
// about what data we access here and how. We need to hold a lock to prevent m_channels
// from being changed while we iterate it, but calling channel->byteLength() is safe
// because it doesn't involve chasing any pointers that can be nullified while the
// AudioBuffer is alive.
Locker locker { m_channelsLock };
size_t cost = 0;
for (auto& channel : m_channels)
cost += channel->byteLength();
return cost;
}
bool AudioBuffer::hasDetachedChannelBuffer() const
{
for (auto& channel : m_channels) {
if (channel->isDetached())
return true;
}
return false;
}
bool AudioBuffer::topologyMatches(const AudioBuffer& other) const
{
return numberOfChannels() == other.numberOfChannels() && length() == other.length() && sampleRate() == other.sampleRate();
}
bool AudioBuffer::copyTo(AudioBuffer& other) const
{
if (!topologyMatches(other))
return false;
if (hasDetachedChannelBuffer() || other.hasDetachedChannelBuffer())
return false;
for (unsigned channelIndex = 0; channelIndex < numberOfChannels(); ++channelIndex)
memcpy(other.rawChannelData(channelIndex), m_channels[channelIndex]->data(), length() * sizeof(float));
return true;
}
Ref<AudioBuffer> AudioBuffer::clone(ShouldCopyChannelData shouldCopyChannelData) const
{
auto clone = AudioBuffer::create(numberOfChannels(), length(), sampleRate(), m_isDetachable ? LegacyPreventDetaching::No : LegacyPreventDetaching::Yes);
ASSERT(clone);
if (shouldCopyChannelData == ShouldCopyChannelData::Yes)
copyTo(*clone);
return clone.releaseNonNull();
}
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO)