Abstract This specification describes a high-level Web API for processing and synthesizing audio in web applications. The primary paradigm is of an audio routing graph, where a number of objects are connected together to define the overall audio rendering. The actual processing will primarily take place in the underlying implementation (typically optimized Assembly / C / C code), but is also supported. The section covers the motivation behind this specification.
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This API is designed to be used in conjunction with other APIs and elements on the web platform, notably: XMLHttpRequest (using the responseType and response attributes). For games and interactive applications, it is anticipated to be used with the canvas 2D and WebGL 3D graphics APIs. Status of this document. This section describes the status of this document at the time of its publication.
Other documents may supersede this document. A list of current W3C publications and the latest revision of this technical report can be found in the This document was published by the the as a Candidate Recommendation. This document is intended to become a W3C Recommendation. This document will remain a Candidate Recommendation at least until 12 December 2018 in order to ensure the opportunity for wide review. If you wish to make comments regarding this document, please or send them to (, ). A is available.
Publication as a Candidate Recommendation does not imply endorsement by the W3C Membership. This is a draft document and may be updated, replaced or obsoleted by other documents at any time.
It is inappropriate to cite this document as other than work in progress. This document was produced by a group operating under the. W3C maintains a made in connection with the deliverables of the group; that page also includes instructions for disclosing a patent. An individual who has actual knowledge of a patent which the individual believes contains must disclose the information in accordance with.
This document is governed by the. Return type: createBuffer(numberOfChannels, length, sampleRate) Creates an AudioBuffer of the given size. The audio data in the buffer will be zero-initialized (silent). A exception MUST be thrown if any of the arguments is negative, zero, or outside its nominal range.
Arguments for the method. Parameter Type Nullable Optional Description numberOfChannels unsigned long ✘ ✘ Determines how many channels the buffer will have. An implementation MUST support at least 32 channels. Length unsigned long ✘ ✘ Determines the size of the buffer in sample-frames.
This MUST be at least 1. SampleRate float ✘ ✘ Describes the sample-rate of the linear PCM audio data in the buffer in sample-frames per second. An implementation MUST support sample rates in at least the range 8000 to 96000. Return type: createIIRFilter(feedforward, feedback) Arguments for the method. Parameter Type Nullable Optional Description feedforward sequence ✘ ✘ An array of the feedforward (numerator) coefficients for the transfer function of the IIR filter. The maximum length of this array is 20. If all of the values are zero, an MUST be thrown.
A MUST be thrown if the array length is 0 or greater than 20. Feedback sequence ✘ ✘ An array of the feedback (denominator) coefficients for the transfer function of the IIR filter. The maximum length of this array is 20.
If the first element of the array is 0, an MUST be thrown. A MUST be thrown if the array length is 0 or greater than 20. When calling this method, execute these steps:. If and are not of the same length, an MUST be thrown. Let o be a new object of type. Respectively set the and parameters passed to this factory method to the attributes of the same name on o. Set the attribute on o to the value of the attribute of the constraints attribute passed to the factory method.
Construct a new p, passing the this factory method has been called on as a first argument, and o. Return p. Arguments for the method. Parameter Type Nullable Optional Description real sequence ✘ ✘ A sequence of cosine parameters.
See its constructor argument for a more detailed description. Imag sequence ✘ ✘ A sequence of sine parameters. See its constructor argument for a more detailed description. Constraints PeriodicWaveConstraints ✘ ✔ If not given, the waveform is normalized. Otherwise, the waveform is normalized according the value given by constraints. Return type: createScriptProcessor(bufferSize, numberOfInputChannels, numberOfOutputChannels) for a. This method is DEPRECATED, as it is intended to be replaced.
Creates a for direct audio processing using scripts. An exception MUST be thrown if or or are outside the valid range. It is invalid for both and to be zero. In this case an MUST be thrown. Arguments for the method. Parameter Type Nullable Optional Description bufferSize unsigned long ✘ ✔ The parameter determines the buffer size in units of sample-frames. If it’s not passed in, or if the value is 0, then the implementation will choose the best buffer size for the given environment, which will be constant power of 2 throughout the lifetime of the node.
Otherwise if the author explicitly specifies the bufferSize, it MUST be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384. This value controls how frequently the event is dispatched and how many sample-frames need to be processed each call. Lower values for will result in a lower (better).
Higher values will be necessary to avoid audio breakup. It is recommended for authors to not specify this buffer size and allow the implementation to pick a good buffer size to balance between and audio quality.
If the value of this parameter is not one of the allowed power-of-2 values listed above, an MUST be thrown. NumberOfInputChannels unsigned long ✘ ✔ This parameter determines the number of channels for this node’s input.
Values of up to 32 must be supported. A must be thrown if the number of channels is not supported. NumberOfOutputChannels unsigned long ✘ ✔ This parameter determines the number of channels for this node’s output. Values of up to 32 must be supported.
A must be thrown if the number of channels is not supported. Return type: decodeAudioData(audioData, successCallback, errorCallback) Asynchronously decodes the audio file data contained in the. The can, for example, be loaded from an XMLHttpRequest’s response attribute after setting the responseType to 'arraybuffer'. Audio file data can be in any of the formats supported by the element.
The buffer passed to has its content-type determined by sniffing, as described in. Although the primary method of interfacing with this function is via its promise return value, the callback parameters are provided for legacy reasons. The system shall ensure that the is not garbage collected before the promise is resolved or rejected and any callback function is called and completes. When queuing a decoding operation to be performed on another thread, the following steps MUST happen on a thread that is not the nor the, called the decoding thread.
Note: Multiple decoding threads can run in parallel to service multiple calls to decodeAudioData. Running a to resume an means running these steps on the:. Attempt to. Set the flag on the to running. Start. In case of failure, on the to execute the following, and abort these steps:. Reject all promises from in order, then clear.
Reject promise. on the ’s event loop, to execute these steps:. Resolve all promises from in order, then clear. Resolve promise.
If the attribute of the is not already ' ':. Set the attribute of the to ' '. to fire a simple event named statechange at the. When creating an, execute these steps:.
Set a control thread state to suspended on the. Set a rendering thread state to suspended on the.
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Let pendingResumePromises be an empty ordered list of promises. If contextOptions is given, apply the options:. Set the internal latency of this according to contextOptions., as described in. If contextOptions. Is specified, set the of this to this value. Otherwise, use the sample rate of the default output device. If the selected sample rate differs from the sample rate of the output device, this MUST resample the audio output to match the sample rate of the output device.
Note: If resampling is required, the latency of the AudioContext may be affected, possibly by a large amount. If the is not, abort these steps. Send a to start processing. Running a to close an means running these steps on the:. Attempt to. Set the to suspended. on the ’s event loop, to execute these steps:.
Resolve promise. If the attribute of the is not already ' ':.
Set the attribute of the to ' '. to fire a simple event named statechange at the. When an is closed, any s and s that were connected to an will have their output ignored. That is, these will no longer cause any output to speakers or other output devices. For more flexibility in behavior, consider using. Note: When an has been closed, implementation can choose to aggressively release more resources than when suspending.
Return type: getOutputTimestamp Returns a new instance containing two correlated context’s audio stream position values: the member contains the time of the sample frame which is currently being rendered by the audio output device (i.e., output audio stream position), in the same units and origin as context’s; the member contains the time estimating the moment when the sample frame corresponding to the stored contextTime value was rendered by the audio output device, in the same units and origin as performance.now (described in ). If the context’s rendering graph has not yet processed a block of audio, then call returns an AudioTimestamp instance with both members containing zero. After the context’s rendering graph has started processing of blocks of audio, its attribute value always exceeds the value obtained from method call. Return type: suspend Suspends the progression of 's, allows any current context processing blocks that are already processed to be played to the destination, and then allows the system to release its claim on audio hardware. This is generally useful when the application knows it will not need the for some time, and wishes to temporarily associated with the.
The promise resolves when the frame buffer is empty (has been handed off to the hardware), or immediately (with no other effect) if the context is already suspended. The promise is rejected if the context has been closed.
Running a to suspend an means running these steps on the:. Attempt to. Set the on the to suspended.
on the ’s event loop, to execute these steps:. Resolve promise. If the attribute of the is not already ' ':. Set the attribute of the to ' '.
to fire a simple event named statechange at the. While an is suspended, s will have their output ignored; that is, data will be lost by the real time nature of media streams. S will similarly have their output ignored until the system is resumed. S and s will cease to have their processing handlers invoked while suspended, but will resume when the context is resumed. For the purpose of window functions, the data is considered as a continuous stream - i.e. The resume/ suspend does not cause silence to appear in the 's stream of data.
In particular, calling functions repeatedly when a is suspended MUST return the same data. Let rendering started be an internal slot of this. Initialize this slot to false. When startRendering is called, the following steps MUST be performed on the:. If the slot on the is true, return a rejected promise with, and abort these steps. Set the slot of the to true.
Let promise be a new promise. Create a new, with a number of channels, length and sample rate equal respectively to the numberOfChannels, length and sampleRate values passed to this instance’s constructor in the contextOptions parameter. Assign this buffer to an internal slot rendered buffer in the. If an exception was thrown during the preceding constructor call, reject promise with this exception. Otherwise, in the case that the buffer was successfully constructed,. Return promise. To begin offline rendering, the following steps MUST happen on a that is created for the occasion.
Given the current connections and scheduled changes, start rendering length sample-frames of audio into. For every, check and suspend the rendering if necessary.
If a suspended context is resumed, continue to render the buffer. Once the rendering is complete, on the ’s event loop to perform the following steps:. Resolve the promise created by with. to fire an event named complete at this instance, using an instance of whose renderedBuffer property is set to. Return type: suspend(suspendTime) Schedules a suspension of the time progression in the audio context at the specified time and returns a promise. This is generally useful when manipulating the audio graph synchronously on. Note that the maximum precision of suspension is the size of the and the specified suspension time will be rounded down to the nearest boundary.
For this reason, it is not allowed to schedule multiple suspends at the same quantized frame. Also, scheduling should be done while the context is not running to ensure precise suspension.
Arguments for the method. Parameter Type Nullable Optional Description suspendTime double ✘ ✘ Schedules a suspension of the rendering at the specified time, which is quantized and rounded down to the size. If the quantized frame number. is negative or. is less than or equal to the current time or. is greater than or equal to the total render duration or.
is scheduled by another suspend for the same time, then the promise is rejected with. If any of the values in lie outside its nominal range, throw a exception and abort the following steps. Let b be a new object. Respectively assign the values of the attributes, of the passed in the constructor to the internal slots,. Set the internal slot of this to the result of calling (. ).
Note: This initializes the underlying storage to zero. Return b. Arguments for the method. Parameter Type Nullable Optional Description options AudioBufferOptions ✘ ✘ 1.4.2. Attributes duration, of type, readonly Duration of the PCM audio data in seconds.
This is computed from the and the of the by performing a division between the and the. Length, of type, readonly Length of the PCM audio data in sample-frames. This MUST return the value of. NumberOfChannels, of type, readonly The number of discrete audio channels. This MUST return the value of.
SampleRate, of type, readonly The sample-rate for the PCM audio data in samples per second. This MUST return the value of. Methods copyFromChannel(destination, channelNumber, startInChannel) The method copies the samples from the specified channel of the to the destination array. Let buffer be the with (Nb ) frames, let (Nf ) be the number of elements in the array, and (k ) be the value of. Then the number of frames copied from buffer to is ( min(Nb - k, Nf) ).
If this is less than (Nf ), then the remaining elements of are not modified. Arguments for the method. Parameter Type Nullable Optional Description destination Float32Array ✘ ✘ The array the channel data will be copied to. ChannelNumber unsigned long ✘ ✘ The index of the channel to copy the data from. If channelNumber is greater or equal than the number of channel of the, an MUST be thrown. StartInChannel unsigned long ✘ ✔ An optional offset to copy the data from. If startInChannel is greater than the length of the, an MUST be thrown.
Return type: void copyToChannel(source, channelNumber, startInChannel) The method copies the samples to the specified channel of the from the source array. A may be thrown if cannot be copied to the buffer. Let buffer be the with (Nb ) frames, let (Nf ) be the number of elements in the array, and (k ) be the value of. Then the number of frames copied from to the buffer is ( min(Nb - k, Nf) ). If this is less than (Nf ), then the remaining elements of buffer are not modified. Arguments for the method.
Parameter Type Nullable Optional Description source Float32Array ✘ ✘ The array the channel data will be copied from. ChannelNumber unsigned long ✘ ✘ The index of the channel to copy the data to. If channelNumber is greater or equal than the number of channel of the, an MUST be thrown. StartInChannel unsigned long ✘ ✔ An optional offset to copy the data to.
If startInChannel is greater than the length of the, an MUST be thrown. Enumeration description ' speakers' use. In cases where the number of channels do not match any of these basic speaker layouts, revert to ' '. ' discrete' Up-mix by filling channels until they run out then zero out remaining channels.
Down-mix by filling as many channels as possible, then dropping remaining channels. Attributes channelCount, of type is the number of channels used when up-mixing and down-mixing connections to any inputs to the node.
The default value is 2 except for specific nodes where its value is specially determined. This attribute has no effect for nodes with no inputs. If this value is set to zero or to a value greater than the implementation’s maximum number of channels the implementation MUST throw a exception. In addition, some nodes have additional channelCount constraints on the possible values for the channel count: The behavior depends on whether the destination node is the destination of an or: The channel count MUST be between 1 and o. An exception MUST be thrown for any attempt to set the count outside this range. The channel count cannot be changed. An exception MUST be thrown for any attempt to change the value.
The channel count cannot be changed, and an exception MUST be thrown for any attempt to change the value. The channel count cannot be changed, and an exception MUST be thrown for any attempt to change the value. The channel count cannot changed from two, and a exception MUST be thrown for any attempt to change the value.
The channel count cannot be greater than two, and a exception MUST be thrown for any attempt to change the to a value greater than two. The channel count cannot be greater than two, and a exception MUST be thrown for any attempt to change the to a value greater than two. The channel count cannot be changed, and an exception MUST be thrown for any attempt to change the value. The channel count cannot be greater than two, and a exception MUST be thrown for any attempt to change the to a value greater than two. See for more information on this attribute. ChannelCountMode, of type determines how channels will be counted when up-mixing and down-mixing connections to any inputs to the node.
The default value is ' '. This attribute has no effect for nodes with no inputs. In addition, some nodes have additional channelCountMode constraints on the possible values for the channel count mode: If the is the node of an, then the channel count mode cannot be changed. An exception MUST be thrown for any attempt to change the value.
The channel count mode cannot be changed from ' ' and an exception MUST be thrown for any attempt to change the value. The channel count mode cannot be changed from ' ' and an exception MUST be thrown for any attempt to change the value. The channel count mode cannot be changed from ' ', and a exception MUST be thrown for any attempt to change the value. The channel count mode cannot be set to ' ', and a exception MUST be thrown for any attempt to set it to ' '. The channel count mode cannot be set to ' ', and a exception MUST be thrown for any attempt to set it to ' '. The channel count mode cannot be changed from ' ' and an exception MUST be thrown for any attempt to change the value. The channel count mode cannot be set to ' ', and a exception MUST be thrown for any attempt to set it to ' '.
See the section for more information on this attribute. ChannelInterpretation, of type determines how individual channels will be treated when up-mixing and down-mixing connections to any inputs to the node.
The default value is ' '. This attribute has no effect for nodes with no inputs. In addition, some nodes have additional channelInterpretation constraints on the possible values for the channel interpretation: The channel intepretation can not be changed from ' ' and a exception MUST be thrown for any attempt to change the value. See for more information on this attribute. Context, of type, readonly The which owns this. NumberOfInputs, of type, readonly The number of inputs feeding into the. For source nodes, this will be 0.
This attribute is predetermined for many types, but some s, like the and the, have variable number of inputs. NumberOfOutputs, of type, readonly The number of outputs coming out of the. This attribute is predetermined for some types, but can be variable, like for the and the.
Methods connect(destinationNode, output, input) There can only be one connection between a given output of one specific node and a given input of another specific node. Multiple connections with the same termini are ignored.
For example: nodeA.connect (nodeB ); nodeA.connect (nodeB ); will have the same effect as nodeA.connect (nodeB ); This method returns destination object. Arguments for the method. Parameter Type Nullable Optional Description destinationNode The destination parameter is the to connect to.
If the destination parameter is an that has been created using another, an MUST be thrown. That is, s cannot be shared between s. Output unsigned long ✘ ✔ The output parameter is an index describing which output of the from which to connect. If this parameter is out-of-bound, an exception MUST be thrown. It is possible to connect an output to more than one input with multiple calls to connect. Thus, 'fan-out' is supported. Input The input parameter is an index describing which input of the destination to connect to.
If this parameter is out-of-bounds, an exception MUST be thrown. It is possible to connect an to another which creates a cycle: an may connect to another, which in turn connects back to the input or of the first. This is allowed only if there is at least one in the cycle or a exception MUST be thrown.
Return type: connect(destinationParam, output) Connects the to an, controlling the parameter value with an audio-rate signal. It is possible to connect an output to more than one with multiple calls to connect. Thus, 'fan-out' is supported. It is possible to connect more than one output to a single with multiple calls to connect.
Thus, 'fan-in' is supported. An will take the rendered audio data from any output connected to it and by down-mixing if it is not already mono, then mix it together with other such outputs and finally will mix with the intrinsic parameter value (the value the would normally have without any audio connections), including any timeline changes scheduled for the parameter. The down-mixing to mono is equivalent to the down-mixing for an with = 1, = ' ', and = ' '. There can only be one connection between a given output of one specific node and a specific. Multiple connections with the same termini are ignored.
For example: nodeA.connect (param ); nodeA.connect (param ); will have the same effect as nodeA.connect (param ); Arguments for the method. Parameter Type Nullable Optional Description destinationParam AudioParam ✘ ✘ The destination parameter is the to connect to. This method does not return destination object. If belongs to an that belongs to a that is different from the that has created the on which this method was called, an MUST be thrown. Output unsigned long ✘ ✔ The output parameter is an index describing which output of the from which to connect. If the parameter is out-of-bound, an exception MUST be thrown.
Return type: void disconnect(destinationNode, output) Disconnects a specific output of the from a specific input of some destination. Arguments for the method. Parameter Type Nullable Optional Description destinationNode The destinationNode parameter is the to disconnect. If there is no connection to the destinationNode from the given output, an exception MUST be thrown. Output unsigned long ✘ ✘ The output parameter is an index describing which output of the from which to disconnect. If this parameter is out-of-bound, an exception MUST be thrown.
Return type: void disconnect(destinationNode, output, input) Disconnects a specific output of the from a specific input of some destination. Arguments for the method. Parameter Type Nullable Optional Description destinationNode The destinationNode parameter is the to disconnect. If there is no connection to the destinationNode from the given output, an exception MUST be thrown. Output unsigned long ✘ ✘ The output parameter is an index describing which output of the from which to disconnect. If this parameter is out-of-bound, an exception MUST be thrown. Input The input parameter is an index describing which input of the destination to disconnect.
If this parameter is out-of-bounds, an exception MUST be thrown. Return type: void disconnect(destinationParam) Disconnects all outputs of the that go to a specific destination. The contribution of this to the computed parameter value goes to 0 when this operation takes effect. The intrinsic parameter value is not affected by this operation.
Arguments for the method. Parameter Type Nullable Optional Description destinationParam AudioParam ✘ ✘ The destinationParam parameter is the to disconnect. If there is no connection to the destinationParam, an exception MUST be thrown. Return type: void disconnect(destinationParam, output) Disconnects a specific output of the from a specific destination. The contribution of this to the computed parameter value goes to 0 when this operation takes effect. The intrinsic parameter value is not affected by this operation.
Arguments for the method. Parameter Type Nullable Optional Description destinationParam AudioParam ✘ ✘ The destinationParam parameter is the to disconnect. If there is no connection to the destinationParam, an exception MUST be thrown.
Output unsigned long ✘ ✘ The output parameter is an index describing which output of the from which to disconnect. If the parameter is out-of-bound, an exception MUST be thrown.
Let (tc ) be the value of cancelTime. Then. Let (E1 ) be the event (if any) at time (t1 ) where (t1 ) is the largest number satisfying (t1 le tc ).
Let (E2 ) be the event (if any) at time (t2 ) where (t2 ) is the smallest number satisfying (tc lt t2 ). If (E2 ) exists:.
If (E2 ) is a linear or exponential ramp,. Effectively rewrite (E2 ) to be the same kind of ramp ending at time (tc ) with an end value that would be the value of the original ramp at time (tc ). Go to step 5. Otherwise, go to step 4. If (E1 ) exists:. If (E1 ) is a setTarget event,. Implicitly insert a setValueAtTime event at time (tc ) with the value that the setTarget would have at time (tc ).
Go to step 5. If (E1 ) is a setValueCurve with a start time of (t3 ) and a duration of (d ). If (tc gt t3 + d ), go to step 5. Otherwise,. Effectively replace this event with a setValueCurve event with a start time of (t3 ) and a new duration of (tc-t3 ).
However, this is not a true replacement; this automation MUST take care to produce the same output as the original, and not one computed using a different duration. (That would cause sampling of the value curve in a slightly different way, producing different results.). Go to step 5. Remove all events with time greater than (tc ). If no events are added, then the automation value after is the constant value that the original timeline would have had at time (tc ). Arguments for the method.
Parameter Type Nullable Optional Description cancelTime double ✘ ✘ The time after which any previously scheduled parameter changes will be cancelled. It is a time in the same time coordinate system as the 's attribute. A exception MUST be thrown if cancelTime is negative or is not a finite number. If is less than, it is clamped to. Return type: cancelScheduledValues(cancelTime) Cancels all scheduled parameter changes with times greater than or equal to cancelTime. Cancelling a scheduled parameter change means removing the scheduled event from the event list.
Any active automations whose is less than cancelTime are also cancelled, and such cancellations may cause discontinuities because the original value (from before such automation) is restored immediately. Any hold values scheduled by are also removed if the hold time occurs after cancelTime. Arguments for the method. Parameter Type Nullable Optional Description cancelTime double ✘ ✘ The time after which any previously scheduled parameter changes will be cancelled.
It is a time in the same time coordinate system as the 's attribute. A exception MUST be thrown if cancelTime is negative or is not a finite number. If cancelTime is less than, it is clamped to. Return type: setValueAtTime(value, startTime) Schedules a parameter value change at the given time. If there are no more events after this SetValue event, then for (t geq T0 ), (v(t) = V ), where (T0 ) is the startTime parameter and (V ) is the value parameter. In other words, the value will remain constant. If the next event (having time (T1 )) after this SetValue event is not of type LinearRampToValue or ExponentialRampToValue, then, for (T0 leq t.
An intrinsic parameter value will be calculated at each time, which is either the value set directly to the value attribute, or, if there are any with times before or at this time, the value as calculated from these events. When read, the value attribute always returns the intrinsic value for the current time. If automation events are removed from a given time range, then the intrinsic value will remain unchanged and stay at its previous value until either the value attribute is directly set, or automation events are added for the time range. The computedValue is the sum of the intrinsic value and the value of the.
When read, the value attribute always returns the computedValue for the current time. If this is a, compute its final value with other s. The nominal range for a are the lower and higher values this parameter can effectively have. For, the is clamped to the for this parameter.
Have their final value clamped to their after having been computed from the different values they are composed of. When automation methods are used, clamping is still applied. However, the automation is run as if there were no clamping at all. Only when the automation values are to be applied to the output is the clamping done as specified above.
When this method is called, execute these steps:. If has been called on this node, or if an earlier call to has already occurred, an exception MUST be thrown. Check for any errors that must be thrown due to parameter constraints described below. to start the, including the parameter values in the messsage.
Arguments for the method. Parameter Type Nullable Optional Description when double ✘ ✔ The parameter describes at what time (in seconds) the sound should start playing. It is in the same time coordinate system as the 's attribute. When the signal emitted by the depends on the sound’s start time, the exact value of when is always used without rounding to the nearest sample frame. If 0 is passed in for this value or if the value is less than currentTime, then the sound will start playing immediately. A exception MUST be thrown if when is negative. Return type: void getByteTimeDomainData(array) Copies the (waveform data) into the passed unsigned byte array.
If the array has fewer elements than the value of, the excess elements will be dropped. If the array has more elements than, the excess elements will be ignored. The most recent frames are used in computing the byte data. The values stored in the unsigned byte array are computed in the following way. Let (xk ) be the time-domain data. Then the byte value, (bk ), is $$ bk = left lfloor 128(1 + xk) right rfloor. $$ If (bk ) lies outside the range 0 to 255, (bk ) is clipped to lie in that range.
Arguments for the method. Parameter Type Nullable Optional Description array Uint8Array ✘ ✘ This parameter is where the time-domain sample data will be copied. Return type: void getFloatFrequencyData(array) Copies the into the passed floating-point array.
If the array has fewer elements than the, the excess elements will be dropped. If the array has more elements than the, the excess elements will be ignored. The most recent frames are used in computing the frequency data.
If another call to or occurs within the same as a previous call, the is not updated with the same data. Instead, the previously computed data is returned. The frequency data are in dB units. Arguments for the method. Parameter Type Nullable Optional Description array Float32Array ✘ ✘ This parameter is where the frequency-domain analysis data will be copied. Parameter Value Notes 0 Approximately -3.4028235e38 Approximately 3.4028235e38 ' ' Has loop, of type Indicates if the region of audio data designated by and should be played continuously in a loop. The default value is false.
LoopEnd, of type An optional where looping should end if the attribute is true. Its value is exclusive of the content of the loop. Its default value is 0, and it may usefully be set to any value between 0 and the duration of the buffer. If is less than or equal to 0, or if is greater than the duration of the buffer, looping will end at the end of the buffer. LoopStart, of type An optional where looping should begin if the attribute is true.
Its default value is 0, and it may usefully be set to any value between 0 and the duration of the buffer. If is less than 0, looping will begin at 0. If is greater than the duration of the buffer, looping will begin at the end of the buffer. PlaybackRate, of type, readonly The speed at which to render the audio stream. This is a with to form a. Running a to start the means invoking the handleStart function in the which follows.
Arguments for the method. Parameter Type Nullable Optional Description when double ✘ ✔ The when parameter describes at what time (in seconds) the sound should start playing. It is in the same time coordinate system as the 's attribute. If 0 is passed in for this value or if the value is less than currentTime, then the sound will start playing immediately. A exception MUST be thrown if when is negative. Offset double ✘ ✔ The offset parameter supplies a where playback will begin. If 0 is passed in for this value, then playback will start from the beginning of the buffer.
A exception MUST be thrown if offset is negative. If offset is greater than, playback will begin at (and immediately loop to ). Offset is silently clamped to 0, duration, when startTime is reached, where duration is the value of the duration attribute of the set to the attribute of this AudioBufferSourceNode. Duration double ✘ ✔ The parameter describes the duration of sound to be played, expressed as seconds of total buffer content to be output, including any whole or partial loop iterations. The units of are independent of the effects of.
For example, a of 5 seconds with a playback rate of 0.5 will output 5 seconds of buffer content at half speed, producing 10 seconds of audible output. A exception MUST be thrown if duration is negative. Parameter Value Notes 0 Approximately -3.4028235e38 Approximately 3.4028235e38 ' ' 1.11.2. Methods setOrientation(x, y, z, xUp, yUp, zUp) This method is DEPRECATED. It is equivalent to setting.,.,.,.,.,. Directly with the given x, y, z, xUp, yUp, and zUp values, respectively. Consequently, if any of the, and s have an automation curve set using at the time this method is called, a MUST be thrown.
Describes which direction the listener is pointing in the 3D cartesian coordinate space. Both a front vector and an up vector are provided. In simple human terms, the front vector represents which direction the person’s nose is pointing. The up vector represents the direction the top of a person’s head is pointing. These two vectors are expected to be linearly independent.
For normative requirements of how these values are to be interpreted, see the. The x, y, z parameters represent a front direction vector in 3D space, with the default value being (0,0,-1).
The xUp, yUp, zUp parameters represent an up direction vector in 3D space, with the default value being (0,1,0). Arguments for the method. Parameter Type Nullable Optional Description x float ✘ ✘ y float ✘ ✘ z float ✘ ✘ xUp float ✘ ✘ yUp float ✘ ✘ zUp float ✘ ✘. Parameter Value Notes 0 Approximately -3.4028235e38 Approximately 3.4028235e38 ' ' type, of type The type of this. Its default value is 'lowpass'.
The exact meaning of the other parameters depend on the value of the attribute. Methods getFrequencyResponse(frequencyHz, magResponse, phaseResponse) Given the current filter parameter settings, synchronously calculates the frequency response for the specified frequencies. The three parameters MUST be s of the same length, or an MUST be thrown. The frequency response returned MUST be computed with the sampled for the current processing block.
Arguments for the method. Parameter Type Nullable Optional Description frequencyHz Float32Array ✘ ✘ This parameter specifies an array of frequencies at which the response values will be calculated.
MagResponse Float32Array ✘ ✘ This parameter specifies an output array receiving the linear magnitude response values. If a value in the frequencyHz parameter is not within 0; sampleRate/2, where sampleRate is the value of the property of the, the corresponding value at the same index of the magResponse array MUST be NaN. PhaseResponse Float32Array ✘ ✘ This parameter specifies an output array receiving the phase response values in radians. If a value in the frequencyHz parameter is not within 0; sampleRate/2, where sampleRate is the value of the property of the, the corresponding value at the same index of the phaseResponse array MUST be NaN. Property Value Notes see notes Defaults to 6, but is determined by, or the value specified. 1 1 Has ' ' Has ' ' reference No This interface represents an for combining channels from multiple audio streams into a single audio stream. It has a variable number of inputs (defaulting to 6), but not all of them need be connected.
There is a single output whose audio stream has a number of channels equal to the number of inputs. To merge multiple inputs into one stream, each input gets downmixed into one channel (mono) based on the specified mixing rule. An unconnected input still counts as one silent channel in the output. Changing input streams does not affect the order of output channels. Property Value Notes 1 see notes This defaults to 6, but is otherwise determined from or the value specified by or the member of the dictionary for the. Has ' ' Has ' ' Has reference No This interface represents an for accessing the individual channels of an audio stream in the routing graph.
It has a single input, and a number of 'active' outputs which equals the number of channels in the input audio stream. For example, if a stereo input is connected to an then the number of active outputs will be two (one from the left channel and one from the right). There are always a total number of N outputs (determined by the numberOfOutputs parameter to the method ), The default number is 6 if this value is not provided. Any outputs which are not 'active' will output silence and would typically not be connected to anything. The following algorithm allow determining a value for reduction gain, for each sample of input, for a render quantum of audio. Let attack and release have the values of and, respectively, sampled at the time of processing (those are parameters), mutiplied by the sample-rate of the this is with. Let detector average be the value of the slot.
Let compressor gain be the value of the slot. For each sample input of the render quantum to be processed, execute the following steps:. If the absolute value of input is less than 0.0001, let attenuation be 1.0.
Else, let shaped input be the value of applying the to the absolute value of input. Let attenuation be shaped input divided by the absolute value of input. Let releasing be true if attenuation is greater than compressor gain, false otherwise. Let detector rate be the result of applying the to attenuation. Substract detector average to attenuation, and multiply the result by detector rate. Add this new result to detector average.
Clamp detector average to a maximum of 1.0. Let envelope rate be the result of based on values of attack and release. If releasing is true, set compressor gain to the multiplication of compressor gain by envelope rate, clamped to a maximum of 1.0.
Else, if releasing is false, let gain increment to be detector average minus compressor gain. Multiply gain increment by envelope rate, and add the result to compressor gain.
Compute reduction gain to be compressor gain multiplied by the return value of. Compute metering gain to be reduction gain,. Set to compressor gain. Set to detector average. set the internal slot to the value of metering gain. Note: This step makes the metering gain update once per block, at the end of the block processing.
The makeup gain is a fixed gain stage that only depends on ratio, knee and threshold parameter of the compressor, and not on the input signal. The intent here is to increase the output level of the compressor so it is comparable to the input level. Computing the envelope rate is done by applying a function to the ratio of the compressor gain and the detector average.
User-agents are allowed to choose the shape of the envelope function. However, this function MUST respect the following constraints:. The envelope rate MUST be the calculated from the ratio of the compressor gain and the detector average. Note: When attacking, this number less than or equal to 1, when releasing, this number is strictly greater than 1. The attack curve MUST be a continuous, monotonically increasing function in the range (0, 1 ). The shape of this curve MAY be controlled by.
The release curve MUST be a continuous, monotonically decreasing function that is always greater than 1. The shape of this curve MAY be controlled. This operation returns the value computed by applying this function to the ratio of compressor gain and detector average. Applying the detector curve to the change rate when attacking or releasing allow implementing adaptive release. It is a function that MUST respect the following constraints:. The output of the function MUST be in (0,1 ).
The function MUST be monotonically increasing, continuous. Note: It is allowed, for example, to have a compressor that performs an adaptive release, that is, releasing faster the harder the compression, or to have curves for attack and release that are not of the same shape. Parameter Value Notes 440 - ' ' type, of type The shape of the periodic waveform.
It may directly be set to any of the type constant values except for 'custom'. Doing so MUST throw an exception. The method can be used to set a custom waveform, which results in this attribute being set to 'custom'.
The default value is 'sine'. When this attribute is set, the phase of the oscillator MUST be conserved. Methods setPeriodicWave(periodicWave) Sets an arbitrary custom periodic waveform given a. Arguments for the method.
Parameter Type Nullable Optional Description periodicWave PeriodicWave ✘ ✘.
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