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/*
* Copyright (C) 2010, Google Inc. All rights reserved.
* Copyright (C) 2016-2020, 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 "ConvolverNode.h"
#include "AudioBuffer.h"
#include "AudioNodeInput.h"
#include "AudioNodeOutput.h"
#include "AudioUtilities.h"
#include "Reverb.h"
#include <JavaScriptCore/TypedArrays.h>
#include <wtf/TZoneMallocInlines.h>
// Note about empirical tuning:
// The maximum FFT size affects reverb performance and accuracy.
// If the reverb is single-threaded and processes entirely in the real-time audio thread,
// it's important not to make this too high. In this case 8192 is a good value.
// But, the Reverb object is multi-threaded, so we want this as high as possible without losing too much accuracy.
// Very large FFTs will have worse phase errors. Given these constraints 32768 is a good compromise.
constexpr size_t MaxFFTSize = 32768;
namespace WebCore {
WTF_MAKE_TZONE_OR_ISO_ALLOCATED_IMPL(ConvolverNode);
static unsigned computeNumberOfOutputChannels(unsigned inputChannels, unsigned responseChannels)
{
// The number of output channels for a Convolver must be one or two. And can only be one if
// there's a mono source and a mono response buffer.
return (inputChannels == 1 && responseChannels == 1) ? 1u : 2u;
}
ExceptionOr<Ref<ConvolverNode>> ConvolverNode::create(BaseAudioContext& context, ConvolverOptions&& options)
{
auto node = adoptRef(*new ConvolverNode(context));
auto result = node->handleAudioNodeOptions(options, { 2, ChannelCountMode::ClampedMax, ChannelInterpretation::Speakers });
if (result.hasException())
return result.releaseException();
node->setNormalizeForBindings(!options.disableNormalization);
result = node->setBufferForBindings(WTFMove(options.buffer));
if (result.hasException())
return result.releaseException();
return node;
}
ConvolverNode::ConvolverNode(BaseAudioContext& context)
: AudioNode(context, NodeTypeConvolver)
{
addInput();
addOutput(1);
initialize();
}
ConvolverNode::~ConvolverNode()
{
uninitialize();
}
void ConvolverNode::process(size_t framesToProcess)
{
AudioBus* outputBus = output(0)->bus();
ASSERT(outputBus);
// Synchronize with possible dynamic changes to the impulse response.
if (!m_processLock.tryLock()) {
// Too bad - tryLock() failed. We must be in the middle of setting a new impulse response.
outputBus->zero();
return;
}
Locker locker { AdoptLock, m_processLock };
if (!isInitialized() || !m_reverb.get())
outputBus->zero();
else {
// Process using the convolution engine.
// Note that we can handle the case where nothing is connected to the input, in which case we'll just feed silence into the convolver.
// FIXME: If we wanted to get fancy we could try to factor in the 'tail time' and stop processing once the tail dies down if
// we keep getting fed silence.
m_reverb->process(input(0)->bus(), outputBus, framesToProcess);
}
}
ExceptionOr<void> ConvolverNode::setBufferForBindings(RefPtr<AudioBuffer>&& buffer)
{
ASSERT(isMainThread());
if (!buffer)
return { };
if (buffer->sampleRate() != context().sampleRate())
return Exception { ExceptionCode::NotSupportedError, "Buffer sample rate does not match the context's sample rate"_s };
unsigned numberOfChannels = buffer->numberOfChannels();
size_t bufferLength = buffer->length();
// The current implementation supports only 1-, 2-, or 4-channel impulse responses, with the
// 4-channel response being interpreted as true-stereo (see Reverb class).
bool isChannelCountGood = (numberOfChannels == 1 || numberOfChannels == 2 || numberOfChannels == 4);
if (!isChannelCountGood)
return Exception { ExceptionCode::NotSupportedError, "Buffer should have 1, 2 or 4 channels"_s };
// Wrap the AudioBuffer by an AudioBus. It's an efficient pointer set and not a memcpySpan().
// This memory is simply used in the Reverb constructor and no reference to it is kept for later use in that class.
auto bufferBus = AudioBus::create(numberOfChannels, bufferLength, false);
for (unsigned i = 0; i < numberOfChannels; ++i)
bufferBus->setChannelMemory(i, buffer->channelData(i)->typedMutableSpan().first(bufferLength));
bufferBus->setSampleRate(buffer->sampleRate());
// Create the reverb with the given impulse response.
bool useBackgroundThreads = !context().isOfflineContext();
auto reverb = makeUnique<Reverb>(bufferBus.get(), AudioUtilities::renderQuantumSize, MaxFFTSize, useBackgroundThreads, m_normalize);
{
// The context must be locked since changing the buffer can re-configure the number of channels that are output.
Locker contextLocker { context().graphLock() };
// Synchronize with process().
Locker locker { m_processLock };
m_reverb = WTFMove(reverb);
m_buffer = WTFMove(buffer);
if (m_buffer) {
// This will propagate the channel count to any nodes connected further downstream in the graph.
output(0)->setNumberOfChannels(computeNumberOfOutputChannels(input(0)->numberOfChannels(), m_buffer->numberOfChannels()));
}
}
return { };
}
AudioBuffer* ConvolverNode::bufferForBindings() WTF_IGNORES_THREAD_SAFETY_ANALYSIS
{
ASSERT(isMainThread());
return m_buffer.get();
}
void ConvolverNode::setNormalizeForBindings(bool normalize)
{
ASSERT(isMainThread());
m_normalize = normalize;
}
double ConvolverNode::tailTime() const
{
ASSERT(context().isAudioThread());
if (!m_processLock.tryLock())
return std::numeric_limits<double>::infinity();
Locker locker { AdoptLock, m_processLock };
return m_reverb ? m_reverb->impulseResponseLength() / static_cast<double>(sampleRate()) : 0;
}
double ConvolverNode::latencyTime() const
{
ASSERT(context().isAudioThread());
if (!m_processLock.tryLock())
return std::numeric_limits<double>::infinity();
Locker locker { AdoptLock, m_processLock };
return m_reverb ? m_reverb->latencyFrames() / static_cast<double>(sampleRate()) : 0;
}
bool ConvolverNode::requiresTailProcessing() const
{
// Always return true even if the tail time and latency might both be zero.
return true;
}
ExceptionOr<void> ConvolverNode::setChannelCount(unsigned count)
{
if (count > 2)
return Exception { ExceptionCode::NotSupportedError, "ConvolverNode's channel count cannot be greater than 2"_s };
return AudioNode::setChannelCount(count);
}
ExceptionOr<void> ConvolverNode::setChannelCountMode(ChannelCountMode mode)
{
if (mode == ChannelCountMode::Max)
return Exception { ExceptionCode::NotSupportedError, "ConvolverNode's channel count mode cannot be 'max'"_s };
return AudioNode::setChannelCountMode(mode);
}
void ConvolverNode::checkNumberOfChannelsForInput(AudioNodeInput* input)
{
ASSERT(context().isAudioThread() && context().isGraphOwner());
std::optional<unsigned> numberOfBufferChannels;
if (m_processLock.tryLock()) {
Locker locker { AdoptLock, m_processLock };
if (m_buffer)
numberOfBufferChannels = m_buffer->numberOfChannels();
}
if (numberOfBufferChannels) {
unsigned numberOfOutputChannels = computeNumberOfOutputChannels(input->numberOfChannels(), *numberOfBufferChannels);
if (isInitialized() && numberOfOutputChannels != output(0)->numberOfChannels()) {
// We're already initialized but the channel count has changed.
uninitialize();
}
if (!isInitialized()) {
// This will propagate the channel count to any nodes connected further
// downstream in the graph.
output(0)->setNumberOfChannels(numberOfOutputChannels);
initialize();
}
}
// Update the input's internal bus if needed.
AudioNode::checkNumberOfChannelsForInput(input);
}
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO)
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