Source/WebCore/Modules/webaudio/AudioNode.h

/*
 * 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
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 * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 */

#pragma once

#include "AudioBus.h"
#include "EventTarget.h"
#include "ExceptionOr.h"
#include <wtf/Forward.h>
#include <wtf/LoggerHelper.h>

#define DEBUG_AUDIONODE_REFERENCES 0

namespace WebCore {

class AudioContext;
class AudioNodeInput;
class AudioNodeOutput;
class AudioParam;

// An AudioNode is the basic building block for handling audio within an AudioContext.
// It may be an audio source, an intermediate processing module, or an audio destination.
// Each AudioNode can have inputs and/or outputs. An AudioSourceNode has no inputs and a single output.
// An AudioDestinationNode has one input and no outputs and represents the final destination to the audio hardware.
// Most processing nodes such as filters will have one input and one output, although multiple inputs and outputs are possible.

class AudioNode
    : public EventTargetWithInlineData
#if !RELEASE_LOG_DISABLED
    , private LoggerHelper
#endif
{
    WTF_MAKE_NONCOPYABLE(AudioNode);
    WTF_MAKE_ISO_ALLOCATED(AudioNode);
public:
    enum { ProcessingSizeInFrames = 128 };

    AudioNode(AudioContext&, float sampleRate);
    virtual ~AudioNode();

    AudioContext& context() { return m_context.get(); }
    const AudioContext& context() const { return m_context.get(); }

    enum NodeType {
        NodeTypeUnknown,
        NodeTypeDestination,
        NodeTypeOscillator,
        NodeTypeAudioBufferSource,
        NodeTypeMediaElementAudioSource,
        NodeTypeMediaStreamAudioDestination,
        NodeTypeMediaStreamAudioSource,
        NodeTypeJavaScript,
        NodeTypeBiquadFilter,
        NodeTypePanner,
        NodeTypeConvolver,
        NodeTypeDelay,
        NodeTypeGain,
        NodeTypeChannelSplitter,
        NodeTypeChannelMerger,
        NodeTypeAnalyser,
        NodeTypeDynamicsCompressor,
        NodeTypeWaveShaper,
        NodeTypeBasicInspector,
        NodeTypeEnd
    };

    enum ChannelCountMode {
        Max,
        ClampedMax,
        Explicit
    };

    NodeType nodeType() const { return m_nodeType; }
    void setNodeType(NodeType);

    // We handle our own ref-counting because of the threading issues and subtle nature of
    // how AudioNodes can continue processing (playing one-shot sound) after there are no more
    // JavaScript references to the object.
    enum RefType { RefTypeNormal, RefTypeConnection };

    // Can be called from main thread or context's audio thread.
    void ref(RefType refType = RefTypeNormal);
    void deref(RefType refType = RefTypeNormal);

    // Can be called from main thread or context's audio thread.  It must be called while the context's graph lock is held.
    void finishDeref(RefType refType);

    // The AudioNodeInput(s) (if any) will already have their input data available when process() is called.
    // Subclasses will take this input data and put the results in the AudioBus(s) of its AudioNodeOutput(s) (if any).
    // Called from context's audio thread.
    virtual void process(size_t framesToProcess) = 0;

    // Resets DSP processing state (clears delay lines, filter memory, etc.)
    // Called from context's audio thread.
    virtual void reset() = 0;

    // No significant resources should be allocated until initialize() is called.
    // Processing may not occur until a node is initialized.
    virtual void initialize();
    virtual void uninitialize();

    bool isInitialized() const { return m_isInitialized; }
    void lazyInitialize();

    unsigned numberOfInputs() const { return m_inputs.size(); }
    unsigned numberOfOutputs() const { return m_outputs.size(); }

    AudioNodeInput* input(unsigned);
    AudioNodeOutput* output(unsigned);

    // Called from main thread by corresponding JavaScript methods.
    virtual ExceptionOr<void> connect(AudioNode&, unsigned outputIndex, unsigned inputIndex);
    ExceptionOr<void> connect(AudioParam&, unsigned outputIndex);
    virtual ExceptionOr<void> disconnect(unsigned outputIndex);

    virtual float sampleRate() const { return m_sampleRate; }

    // processIfNecessary() is called by our output(s) when the rendering graph needs this AudioNode to process.
    // This method ensures that the AudioNode will only process once per rendering time quantum even if it's called repeatedly.
    // This handles the case of "fanout" where an output is connected to multiple AudioNode inputs.
    // Called from context's audio thread.
    void processIfNecessary(size_t framesToProcess);

    // Called when a new connection has been made to one of our inputs or the connection number of channels has changed.
    // This potentially gives us enough information to perform a lazy initialization or, if necessary, a re-initialization.
    // Called from main thread.
    virtual void checkNumberOfChannelsForInput(AudioNodeInput*);

#if DEBUG_AUDIONODE_REFERENCES
    static void printNodeCounts();
#endif

    bool isMarkedForDeletion() const { return m_isMarkedForDeletion; }

    // tailTime() is the length of time (not counting latency time) where non-zero output may occur after continuous silent input.
    virtual double tailTime() const = 0;
    // latencyTime() is the length of time it takes for non-zero output to appear after non-zero input is provided. This only applies to
    // processing delay which is an artifact of the processing algorithm chosen and is *not* part of the intrinsic desired effect. For 
    // example, a "delay" effect is expected to delay the signal, and thus would not be considered latency.
    virtual double latencyTime() const = 0;

    // propagatesSilence() should return true if the node will generate silent output when given silent input. By default, AudioNode
    // will take tailTime() and latencyTime() into account when determining whether the node will propagate silence.
    virtual bool propagatesSilence() const;
    bool inputsAreSilent();
    void silenceOutputs();

    void enableOutputsIfNecessary();
    void disableOutputsIfNecessary();

    unsigned channelCount();
    virtual ExceptionOr<void> setChannelCount(unsigned);

    String channelCountMode();
    ExceptionOr<void> setChannelCountMode(const String&);

    String channelInterpretation();
    ExceptionOr<void> setChannelInterpretation(const String&);

    ChannelCountMode internalChannelCountMode() const { return m_channelCountMode; }
    AudioBus::ChannelInterpretation internalChannelInterpretation() const { return m_channelInterpretation; }

protected:
    // Inputs and outputs must be created before the AudioNode is initialized.
    void addInput(std::unique_ptr<AudioNodeInput>);
    void addOutput(std::unique_ptr<AudioNodeOutput>);
    
    // Called by processIfNecessary() to cause all parts of the rendering graph connected to us to process.
    // Each rendering quantum, the audio data for each of the AudioNode's inputs will be available after this method is called.
    // Called from context's audio thread.
    virtual void pullInputs(size_t framesToProcess);

    // Force all inputs to take any channel interpretation changes into account.
    void updateChannelsForInputs();

#if !RELEASE_LOG_DISABLED
    const Logger& logger() const final { return m_logger.get(); }
    const void* logIdentifier() const final { return m_logIdentifier; }
    const char* logClassName() const final { return "AudioNode"; }
    WTFLogChannel& logChannel() const final;
#endif

private:
    // EventTarget
    EventTargetInterface eventTargetInterface() const override;
    ScriptExecutionContext* scriptExecutionContext() const final;

    volatile bool m_isInitialized;
    NodeType m_nodeType;
    Ref<AudioContext> m_context;
    float m_sampleRate;
    Vector<std::unique_ptr<AudioNodeInput>> m_inputs;
    Vector<std::unique_ptr<AudioNodeOutput>> m_outputs;

    double m_lastProcessingTime;
    double m_lastNonSilentTime;

    // Ref-counting
    std::atomic<int> m_normalRefCount;
    std::atomic<int> m_connectionRefCount;
    
    bool m_isMarkedForDeletion;
    bool m_isDisabled;

#if DEBUG_AUDIONODE_REFERENCES
    static bool s_isNodeCountInitialized;
    static int s_nodeCount[NodeTypeEnd];
#endif

    void refEventTarget() override { ref(); }
    void derefEventTarget() override { deref(); }

#if !RELEASE_LOG_DISABLED
    mutable Ref<const Logger> m_logger;
    const void* m_logIdentifier;
#endif

protected:
    unsigned m_channelCount;
    ChannelCountMode m_channelCountMode;
    AudioBus::ChannelInterpretation m_channelInterpretation;
};

String convertEnumerationToString(AudioNode::NodeType);

} // namespace WebCore

namespace WTF {
    
template<> struct LogArgument<WebCore::AudioNode::NodeType> {
    static String toString(WebCore::AudioNode::NodeType type) { return convertEnumerationToString(type); }
};

} // namespace WTF