| /* |
| * Copyright (C) 2010, Google 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 "third_party/blink/renderer/modules/webaudio/convolver_node.h" |
| #include <memory> |
| #include "third_party/blink/renderer/modules/webaudio/audio_buffer.h" |
| #include "third_party/blink/renderer/modules/webaudio/audio_node_input.h" |
| #include "third_party/blink/renderer/modules/webaudio/audio_node_output.h" |
| #include "third_party/blink/renderer/modules/webaudio/convolver_options.h" |
| #include "third_party/blink/renderer/platform/audio/reverb.h" |
| #include "third_party/blink/renderer/platform/bindings/exception_state.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. |
| const size_t MaxFFTSize = 32768; |
| |
| namespace blink { |
| |
| ConvolverHandler::ConvolverHandler(AudioNode& node, float sample_rate) |
| : AudioHandler(kNodeTypeConvolver, node, sample_rate), normalize_(true) { |
| AddInput(); |
| AddOutput(1); |
| |
| // Node-specific default mixing rules. |
| channel_count_ = 2; |
| SetInternalChannelCountMode(kClampedMax); |
| SetInternalChannelInterpretation(AudioBus::kSpeakers); |
| |
| Initialize(); |
| } |
| |
| scoped_refptr<ConvolverHandler> ConvolverHandler::Create(AudioNode& node, |
| float sample_rate) { |
| return base::AdoptRef(new ConvolverHandler(node, sample_rate)); |
| } |
| |
| ConvolverHandler::~ConvolverHandler() { |
| Uninitialize(); |
| } |
| |
| void ConvolverHandler::Process(size_t frames_to_process) { |
| AudioBus* output_bus = Output(0).Bus(); |
| DCHECK(output_bus); |
| |
| // Synchronize with possible dynamic changes to the impulse response. |
| MutexTryLocker try_locker(process_lock_); |
| if (try_locker.Locked()) { |
| if (!IsInitialized() || !reverb_) { |
| output_bus->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. |
| reverb_->Process(Input(0).Bus(), output_bus, frames_to_process); |
| } |
| } else { |
| // Too bad - the tryLock() failed. We must be in the middle of setting a |
| // new impulse response. |
| output_bus->Zero(); |
| } |
| } |
| |
| void ConvolverHandler::SetBuffer(AudioBuffer* buffer, |
| ExceptionState& exception_state) { |
| DCHECK(IsMainThread()); |
| |
| if (!buffer) { |
| reverb_.reset(); |
| buffer_ = buffer; |
| return; |
| } |
| |
| if (buffer->sampleRate() != Context()->sampleRate()) { |
| exception_state.ThrowDOMException( |
| DOMExceptionCode::kNotSupportedError, |
| "The buffer sample rate of " + String::Number(buffer->sampleRate()) + |
| " does not match the context rate of " + |
| String::Number(Context()->sampleRate()) + " Hz."); |
| return; |
| } |
| |
| unsigned number_of_channels = buffer->numberOfChannels(); |
| size_t buffer_length = 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 is_channel_count_good = number_of_channels == 1 || |
| number_of_channels == 2 || |
| number_of_channels == 4; |
| |
| if (!is_channel_count_good) { |
| exception_state.ThrowDOMException( |
| DOMExceptionCode::kNotSupportedError, |
| "The buffer must have 1, 2, or 4 channels, not " + |
| String::Number(number_of_channels)); |
| return; |
| } |
| |
| // 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. |
| scoped_refptr<AudioBus> buffer_bus = |
| AudioBus::Create(number_of_channels, buffer_length, false); |
| for (unsigned i = 0; i < number_of_channels; ++i) |
| buffer_bus->SetChannelMemory(i, buffer->getChannelData(i).View()->Data(), |
| buffer_length); |
| |
| buffer_bus->SetSampleRate(buffer->sampleRate()); |
| |
| // Create the reverb with the given impulse response. |
| std::unique_ptr<Reverb> reverb = std::make_unique<Reverb>( |
| buffer_bus.get(), AudioUtilities::kRenderQuantumFrames, MaxFFTSize, |
| Context() && Context()->HasRealtimeConstraint(), normalize_); |
| |
| { |
| // The context must be locked since changing the buffer can |
| // re-configure the number of channels that are output. |
| BaseAudioContext::GraphAutoLocker context_locker(Context()); |
| |
| // Synchronize with process(). |
| MutexLocker locker(process_lock_); |
| reverb_ = std::move(reverb); |
| buffer_ = buffer; |
| if (buffer) { |
| // This will propagate the channel count to any nodes connected further |
| // downstream in the graph. |
| Output(0).SetNumberOfChannels(ComputeNumberOfOutputChannels( |
| Input(0).NumberOfChannels(), buffer_->numberOfChannels())); |
| } |
| } |
| } |
| |
| AudioBuffer* ConvolverHandler::Buffer() { |
| DCHECK(IsMainThread()); |
| return buffer_.Get(); |
| } |
| |
| bool ConvolverHandler::RequiresTailProcessing() const { |
| // Always return true even if the tail time and latency might both be zero. |
| return true; |
| } |
| |
| double ConvolverHandler::TailTime() const { |
| MutexTryLocker try_locker(process_lock_); |
| if (try_locker.Locked()) |
| return reverb_ ? reverb_->ImpulseResponseLength() / |
| static_cast<double>(Context()->sampleRate()) |
| : 0; |
| // Since we don't want to block the Audio Device thread, we return a large |
| // value instead of trying to acquire the lock. |
| return std::numeric_limits<double>::infinity(); |
| } |
| |
| double ConvolverHandler::LatencyTime() const { |
| MutexTryLocker try_locker(process_lock_); |
| if (try_locker.Locked()) |
| return reverb_ ? reverb_->LatencyFrames() / |
| static_cast<double>(Context()->sampleRate()) |
| : 0; |
| // Since we don't want to block the Audio Device thread, we return a large |
| // value instead of trying to acquire the lock. |
| return std::numeric_limits<double>::infinity(); |
| } |
| |
| unsigned ConvolverHandler::ComputeNumberOfOutputChannels( |
| unsigned input_channels, |
| unsigned response_channels) const { |
| // 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 clampTo(std::max(input_channels, response_channels), 1, 2); |
| } |
| |
| void ConvolverHandler::SetChannelCount(unsigned long channel_count, |
| ExceptionState& exception_state) { |
| DCHECK(IsMainThread()); |
| BaseAudioContext::GraphAutoLocker locker(Context()); |
| |
| // channelCount must be 2. |
| if (channel_count != 2) { |
| exception_state.ThrowDOMException( |
| DOMExceptionCode::kNotSupportedError, |
| "ConvolverNode: channelCount cannot be changed from 2"); |
| } |
| } |
| |
| void ConvolverHandler::SetChannelCountMode(const String& mode, |
| ExceptionState& exception_state) { |
| DCHECK(IsMainThread()); |
| BaseAudioContext::GraphAutoLocker locker(Context()); |
| |
| // channcelCountMode must be 'clamped-max'. |
| if (mode != "clamped-max") { |
| exception_state.ThrowDOMException( |
| DOMExceptionCode::kNotSupportedError, |
| "ConvolverNode: channelCountMode cannot be changed from 'clamped-max'"); |
| } |
| } |
| |
| void ConvolverHandler::CheckNumberOfChannelsForInput(AudioNodeInput* input) { |
| DCHECK(Context()->IsAudioThread()); |
| Context()->AssertGraphOwner(); |
| |
| DCHECK(input); |
| DCHECK_EQ(input, &this->Input(0)); |
| if (input != &this->Input(0)) |
| return; |
| |
| if (buffer_) { |
| unsigned number_of_output_channels = ComputeNumberOfOutputChannels( |
| input->NumberOfChannels(), buffer_->numberOfChannels()); |
| |
| if (IsInitialized() && |
| number_of_output_channels != 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(number_of_output_channels); |
| Initialize(); |
| } |
| } |
| |
| // Update the input's internal bus if needed. |
| AudioHandler::CheckNumberOfChannelsForInput(input); |
| } |
| // ---------------------------------------------------------------- |
| |
| ConvolverNode::ConvolverNode(BaseAudioContext& context) : AudioNode(context) { |
| SetHandler(ConvolverHandler::Create(*this, context.sampleRate())); |
| } |
| |
| ConvolverNode* ConvolverNode::Create(BaseAudioContext& context, |
| ExceptionState& exception_state) { |
| DCHECK(IsMainThread()); |
| |
| if (context.IsContextClosed()) { |
| context.ThrowExceptionForClosedState(exception_state); |
| return nullptr; |
| } |
| |
| return new ConvolverNode(context); |
| } |
| |
| ConvolverNode* ConvolverNode::Create(BaseAudioContext* context, |
| const ConvolverOptions& options, |
| ExceptionState& exception_state) { |
| ConvolverNode* node = Create(*context, exception_state); |
| |
| if (!node) |
| return nullptr; |
| |
| node->HandleChannelOptions(options, exception_state); |
| |
| // It is important to set normalize first because setting the buffer will |
| // examing the normalize attribute to see if normalization needs to be done. |
| node->setNormalize(!options.disableNormalization()); |
| if (options.hasBuffer()) |
| node->setBuffer(options.buffer(), exception_state); |
| return node; |
| } |
| |
| ConvolverHandler& ConvolverNode::GetConvolverHandler() const { |
| return static_cast<ConvolverHandler&>(Handler()); |
| } |
| |
| AudioBuffer* ConvolverNode::buffer() const { |
| return GetConvolverHandler().Buffer(); |
| } |
| |
| void ConvolverNode::setBuffer(AudioBuffer* new_buffer, |
| ExceptionState& exception_state) { |
| GetConvolverHandler().SetBuffer(new_buffer, exception_state); |
| } |
| |
| bool ConvolverNode::normalize() const { |
| return GetConvolverHandler().Normalize(); |
| } |
| |
| void ConvolverNode::setNormalize(bool normalize) { |
| GetConvolverHandler().SetNormalize(normalize); |
| } |
| |
| } // namespace blink |