third_party/blink/renderer/modules/webaudio/convolver_node.cc - chromium/src - Git at Google

 /*
  * 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(uint32_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();
   uint32_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(), audio_utilities::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 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 MakeGarbageCollected<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
	/*
	* 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(uint32_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();
	uint32_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(), audio_utilities::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 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 MakeGarbageCollected<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