org.openimaj.audio.samples

Interface SampleBuffer

All Superinterfaces:

Iterable<Float>

All Known Implementing Classes:

FloatSampleBuffer, SampleBuffer16Bit, SampleBuffer8Bit

public interface SampleBuffer
extends Iterable<Float>

This class provides a consistent API for access samples of different sizes (e.g. 8-bit and 16-bit). Subclasses implement get and set methods for samples in a sample byte buffer. The getters and setters return and take floats in all cases, so it is up to the implementing class to decide how best to convert the incoming value to an appropriate value for the sample buffer. The values given and expected should be normalised between Integer.MAX_VALUE and Integer.MIN_VALUE. Despite using floats, this allows us to detect clipping. Return values are signed such that an empty signal has sample value 0. If the audio has multiple channels they will occur in the same order as the original file (likely to be multiplexed - so get(0) will be channel 1 and get(1) will be channel 2, etc.).

Author:

David Dupplaw (dpd@ecs.soton.ac.uk)

Method Summary

Methods

Modifier and Type	Method and Description
double[]	asDoubleArray() Returns the normalised (0..1) sample buffer data as a double array.
double[][]	asDoubleChannelArray() Returns the samples in their channels as normalise (0..1) sample buffer data.
float	get(int index) Get the sample at the given index.
AudioFormat	getFormat() Returns the audio format for this set of samples.
SampleChunk	getSampleChunk() Return a sample chunk that contains the data from this sample buffer.
SampleChunk	getSampleChunk(int channel) Return a sample chunk that contains the data from a specific channel in this sample buffer.
AudioTimecode	getStartTimecode() Returns the timecode of the start of this sample buffer.
float	getUnscaled(int index) Returns the sample value at the given index without scaling - that is, at it's original sample value.
void	set(int index, float sample) Set the sample value at the given index.
void	setFormat(AudioFormat af) Reset the audio format for the samples in this buffer.
int	size() Returns the size of this buffer.

Methods inherited from interface java.lang.Iterable

iterator

Method Detail

get

float get(int index)

Get the sample at the given index. The sample will be a float and have a value between Integer.MAX_VALUE and Integer.MIN_VALUE. That sample will be signed.

Parameters:

index - The index of the sample to retrieve

Returns:

The sample value as an float

set

void set(int index,
       float sample)

Set the sample value at the given index. The sample value should be scaled between Integer.MAX_VALUE and Integer.MIN_VALUE (i.e. it should be signed).

Parameters:

index - The index of the sample to set.

sample - The sample value to set.

size

int size()

Returns the size of this buffer. Divide by the number of channels to get the number of samples per channel.

Returns:

the size of this buffer.

getFormat

AudioFormat getFormat()

Returns the audio format for this set of samples.

Returns:

The AudioFormat for this set of samples.

setFormat

void setFormat(AudioFormat af)

Reset the audio format for the samples in this buffer.

Parameters:

af - The AudioFormat

getSampleChunk

SampleChunk getSampleChunk()

Return a sample chunk that contains the data from this sample buffer. Note that any timestamps will be unset in the new sample chunk.

Returns:

A SampleChunk containing data in this buffer.

getSampleChunk

SampleChunk getSampleChunk(int channel)

Return a sample chunk that contains the data from a specific channel in this sample buffer. The channel is numbered from 0 (so mono audio streams will only have 1 channel accessed with getSampleChunk(0)). Note that any timestamps will be unset in the new sample chunk.

Parameters:

channel - The channel

Returns:

A SampleChunk containing data in this buffer.

asDoubleArray

double[] asDoubleArray()

Returns the normalised (0..1) sample buffer data as a double array.

Returns:

A double array containing the normalised samples

asDoubleChannelArray

double[][] asDoubleChannelArray()

Returns the samples in their channels as normalise (0..1) sample buffer data.

Returns:

The array where the first dimension is the channels.

getUnscaled

float getUnscaled(int index)

Returns the sample value at the given index without scaling - that is, at it's original sample value. This can be useful for debugging, or if the sample buffer is being used for non-audio applications.

Parameters:

index - The index to retrieve

Returns:

The value unscaled

getStartTimecode

AudioTimecode getStartTimecode()

Returns the timecode of the start of this sample buffer. May return null if the timecode is unknown or has no meaning in this context.

Returns:

The timecode a the start of this buffer.