| /* |
| * 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 <wtf/IsoMallocInlines.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_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 { 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 { 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 memcpy(). |
| // 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)->data(), 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 { 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 { 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) |