root/AudioNode.h
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#ifndef AudioNode_h
#define AudioNode_h
#include "bindings/v8/ScriptWrappable.h"
#include "core/dom/EventTarget.h"
#include "core/platform/audio/AudioBus.h"
#include "wtf/Forward.h"
#include "wtf/OwnPtr.h"
#include "wtf/PassOwnPtr.h"
#include "wtf/RefPtr.h"
#include "wtf/Vector.h"
#define DEBUG_AUDIONODE_REFERENCES 0
namespace WebCore {
class AudioContext;
class AudioNodeInput;
class AudioNodeOutput;
class AudioParam;
class ExceptionState;
// 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 ScriptWrappable, public EventTarget {
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,
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 void connect(AudioNode*, unsigned outputIndex, unsigned inputIndex, ExceptionState&);
void connect(AudioParam*, unsigned outputIndex, ExceptionState&);
virtual void disconnect(unsigned outputIndex, ExceptionState&);
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 unsilenceOutputs();
void enableOutputsIfNecessary();
void disableOutputsIfNecessary();
unsigned long channelCount();
virtual void setChannelCount(unsigned long, ExceptionState&);
String channelCountMode();
void setChannelCountMode(const String&, ExceptionState&);
String channelInterpretation();
void setChannelInterpretation(const String&, ExceptionState&);
ChannelCountMode internalChannelCountMode() const { return m_channelCountMode; }
AudioBus::ChannelInterpretation internalChannelInterpretation() const { return m_channelInterpretation; }
// EventTarget
virtual const AtomicString& interfaceName() const OVERRIDE;
virtual ScriptExecutionContext* scriptExecutionContext() const OVERRIDE;
virtual EventTargetData* eventTargetData() OVERRIDE { return &m_eventTargetData; }
virtual EventTargetData* ensureEventTargetData() OVERRIDE { return &m_eventTargetData; }
protected:
// Inputs and outputs must be created before the AudioNode is initialized.
void addInput(PassOwnPtr<AudioNodeInput>);
void addOutput(PassOwnPtr<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();
private:
volatile bool m_isInitialized;
NodeType m_nodeType;
RefPtr<AudioContext> m_context;
float m_sampleRate;
Vector<OwnPtr<AudioNodeInput> > m_inputs;
Vector<OwnPtr<AudioNodeOutput> > m_outputs;
EventTargetData m_eventTargetData;
double m_lastProcessingTime;
double m_lastNonSilentTime;
// Ref-counting
volatile int m_normalRefCount;
volatile int m_connectionRefCount;
bool m_isMarkedForDeletion;
bool m_isDisabled;
#if DEBUG_AUDIONODE_REFERENCES
static bool s_isNodeCountInitialized;
static int s_nodeCount[NodeTypeEnd];
#endif
virtual void refEventTarget() OVERRIDE { ref(); }
virtual void derefEventTarget() OVERRIDE { deref(); }
protected:
unsigned m_channelCount;
ChannelCountMode m_channelCountMode;
AudioBus::ChannelInterpretation m_channelInterpretation;
};
} // namespace WebCore
#endif // AudioNode_h