| Interface | Description |
|---|---|
| AudioBuffer |
Read-only buffer that holds (audio) data and some meta information in the form of an
AudioFormat
instance. |
| AudioMatrix |
Read-only matrix that holds (audio) data and some meta information in the form of an
AudioFormat
instance. |
| AudioSpectrum |
Read-only representation of an (audio) spectrum.
|
| Novelty.Kernel |
Kernel.
|
| Class | Description |
|---|---|
| AbstractAudioSpectrum |
Skeleton implementation of an
AudioSpectrum. |
| AudioBufferFunctions |
Factory methods for turning
float[] functions into AudioBuffer functions. |
| AudioSignalSource |
Encapsulates an
AudioInputStream to serve as a SignalSource for
a SignalPump or a SignalPullProcessor. |
| BandSplit<T extends AudioSpectrum> |
Takes the real values of a sequence of input spectra and creates new buffers for each frequency bin.
|
| Channel |
Only lets one specified channel through.
|
| ComplexAudioBuffer |
Buffer that holds complex, time domain audio data and some meta information in the form of
an
AudioFormat instance. |
| ConstantQTransform |
Processes input using a transform created by
ConstantQTransformFactory. |
| DCT |
Transforms samples obtained with
AudioBuffer.getData()
using the DCT. |
| Decimate |
Applies an appropriate low pass filter before dropping samples.
|
| Downsample |
Drops samples in order to downsample the data.
|
| FFT |
Transforms samples obtained with
AudioBuffer.getData()
using the FFT. |
| FrameNumberFilter |
Allows to ignore
AudioBuffers, if their first frame number as returned by
AudioBuffer.getFrameNumber() does not fall into a specified
range. |
| IFFT |
Transforms samples obtained from a given
LinearFrequencySpectrum
using the inverse FFT. |
| InstantaneousFrequency |
Combines subsequent
LinearFrequencySpectrums into a InstantaneousFrequencySpectrum,
potentially providing better frequency resolution. |
| InstantaneousFrequencySpectrum |
Represents a spectrum with instantaneous frequencies based on two subsequent
LinearFrequencySpectrums. |
| InterleavedChannelJoin |
Joins multichannel audio data in an interleaved format (
LRLRLR...). |
| InterleavedChannelSplit |
Splits
AudioBuffers containing interleaved signals (for example LRLRLR) for multiple channels into
separate buffers containing the signal for one channel each. |
| Interpolate |
Adds zero samples followed by an appropriate low pass filter.
|
| LinearFrequencySpectrum |
Represents a spectrum with linear frequencies (as opposed to a
LogFrequencySpectrum)
as generated by a suitable SignalProcessor. |
| LogFrequencySpectrum |
Log frequency spectrum - possibly created by
ConstantQTransform. |
| Mel<T extends LinearFrequencySpectrum> |
Assumes that the input is linear spectral data produced by some
Transform
and sums up the powers into bins, which are spaced according to the
provided frequency boundaries. |
| MelSpectrum |
Mel spectrum typically constructed from a
LinearFrequencySpectrum created
by FFT. |
| Mono |
Averages the provided signal's interleaved channels into one (mono) channel.
|
| MultiBand<T extends AudioSpectrum> |
Assumes that the input is spectral data produced by some
Transform
and sums up the powers into bins, which are spaced according to the
provided frequency boundaries. |
| MultiBandSpectrum |
Represents an audio spectrum divided into bands as produced for example by
MultiBandSpectrum.createLogarithmicBands(float, float, int) or MultiBandSpectrum.createMidiBands(int, int). |
| Novelty<I extends AudioBuffer> |
Processes AudioBuffer features to create a novelty curve using a kernel and a distance function.
|
| Novelty.GaussianCheckerboardKernel |
Gaussian checkboard kernel.
|
| Novelty.NormGaussianCheckerboardKernel |
Normalized Gaussian checkerboard kernel.
Aggregate values are divided by the maximal possible value a Novelty.GaussianCheckerboardKernel would deliver
before they are returned, assuming the input values are between -1 and 1. |
| OLA |
Overlap-Add combines multiple overlapping windows into non-overlapping ones by
adding the overlapping parts.
|
| OnsetStrength |
Processes
AudioSpectrum to compute an onset strength signal in the form of an AudioBuffer
produced on OnsetStrength.flush().This particular onset strength is defined as the bandwise difference log(power(k,t)+1)-log(power(k,t-1)+1) for the band k and the spectrum t. |
| RealAudioBuffer |
Buffer that holds real, time domain audio data and some meta information in the form of
an
AudioFormat instance. |
| RealAudioMatrix |
AudioMatrix representing exclusively real values. |
| Resample |
Resamples mono audio data.
|
| SelfSimilarity<I extends AudioBuffer> |
Self similarity processor computes a self-similarity matrix from
AudioBuffer features. |
| SlidingWindow |
Provides overlapping frame blocks (aka slices or windows or frames) of a defined size.
|
| TimestampLimitedSignalSource<T extends AudioBuffer> |
Timestamp limited
SignalSource. |
| Upsample |
Adds zero samples (zero stuffing) in order to upsample the data.
|
| Zeropad<T extends AudioBuffer> |
Zeropads
AudioBuffers at a given Zeropad.Position,
so that a desired total size is reached. |
| Enum | Description |
|---|---|
| Zeropad.Position |
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