JUCE  v5.1.1-3-g1a0b28c73
JUCE API
ResamplingAudioSource Class Reference

A type of AudioSource that takes an input source and changes its sample rate. More...

#include <juce_ResamplingAudioSource.h>

Inheritance diagram for ResamplingAudioSource:
Collaboration diagram for ResamplingAudioSource:

Classes

struct  FilterState
 

Public Member Functions

 ResamplingAudioSource (AudioSource *inputSource, bool deleteInputWhenDeleted, int numChannels=2)
 Creates a ResamplingAudioSource for a given input source. More...
 
 ~ResamplingAudioSource ()
 Destructor. More...
 
void flushBuffers ()
 Clears any buffers and filters that the resampler is using. More...
 
void getNextAudioBlock (const AudioSourceChannelInfo &) override
 Called repeatedly to fetch subsequent blocks of audio data. More...
 
double getResamplingRatio () const noexcept
 Returns the current resampling ratio. More...
 
void prepareToPlay (int samplesPerBlockExpected, double sampleRate) override
 Tells the source to prepare for playing. More...
 
void releaseResources () override
 Allows the source to release anything it no longer needs after playback has stopped. More...
 
void setResamplingRatio (double samplesInPerOutputSample)
 Changes the resampling ratio. More...
 

Private Member Functions

void applyFilter (float *samples, int num, FilterState &fs)
 
void createLowPass (double proportionalRate)
 
void resetFilters ()
 
void setFilterCoefficients (double c1, double c2, double c3, double c4, double c5, double c6)
 

Private Attributes

AudioSampleBuffer buffer
 
int bufferPos
 
double coefficients [6]
 
HeapBlock< float * > destBuffers
 
HeapBlock< FilterStatefilterStates
 
OptionalScopedPointer< AudioSourceinput
 
double lastRatio
 
const int numChannels
 
double ratio
 
SpinLock ratioLock
 
int sampsInBuffer
 
HeapBlock< const float * > srcBuffers
 
double subSampleOffset
 

Detailed Description

A type of AudioSource that takes an input source and changes its sample rate.

See also
AudioSource, LagrangeInterpolator, CatmullRomInterpolator

Constructor & Destructor Documentation

◆ ResamplingAudioSource()

ResamplingAudioSource::ResamplingAudioSource ( AudioSource inputSource,
bool  deleteInputWhenDeleted,
int  numChannels = 2 
)

Creates a ResamplingAudioSource for a given input source.

Parameters
inputSourcethe input source to read from
deleteInputWhenDeletedif true, the input source will be deleted when this object is deleted
numChannelsthe number of channels to process

◆ ~ResamplingAudioSource()

ResamplingAudioSource::~ResamplingAudioSource ( )

Destructor.

Member Function Documentation

◆ applyFilter()

void ResamplingAudioSource::applyFilter ( float *  samples,
int  num,
FilterState fs 
)
private

◆ createLowPass()

void ResamplingAudioSource::createLowPass ( double  proportionalRate)
private

◆ flushBuffers()

void ResamplingAudioSource::flushBuffers ( )

Clears any buffers and filters that the resampler is using.

◆ getNextAudioBlock()

void ResamplingAudioSource::getNextAudioBlock ( const AudioSourceChannelInfo bufferToFill)
overridevirtual

Called repeatedly to fetch subsequent blocks of audio data.

After calling the prepareToPlay() method, this callback will be made each time the audio playback hardware (or whatever other destination the audio data is going to) needs another block of data.

It will generally be called on a high-priority system thread, or possibly even an interrupt, so be careful not to do too much work here, as that will cause audio glitches!

See also
AudioSourceChannelInfo, prepareToPlay, releaseResources

Implements AudioSource.

◆ getResamplingRatio()

double ResamplingAudioSource::getResamplingRatio ( ) const
inlinenoexcept

Returns the current resampling ratio.

This is the value that was set by setResamplingRatio().

References AudioSource::getNextAudioBlock(), AudioSource::prepareToPlay(), and AudioSource::releaseResources().

◆ prepareToPlay()

void ResamplingAudioSource::prepareToPlay ( int  samplesPerBlockExpected,
double  sampleRate 
)
overridevirtual

Tells the source to prepare for playing.

An AudioSource has two states: prepared and unprepared.

The prepareToPlay() method is guaranteed to be called at least once on an 'unpreprared' source to put it into a 'prepared' state before any calls will be made to getNextAudioBlock(). This callback allows the source to initialise any resources it might need when playing.

Once playback has finished, the releaseResources() method is called to put the stream back into an 'unprepared' state.

Note that this method could be called more than once in succession without a matching call to releaseResources(), so make sure your code is robust and can handle that kind of situation.

Parameters
samplesPerBlockExpectedthe number of samples that the source will be expected to supply each time its getNextAudioBlock() method is called. This number may vary slightly, because it will be dependent on audio hardware callbacks, and these aren't guaranteed to always use a constant block size, so the source should be able to cope with small variations.
sampleRatethe sample rate that the output will be used at - this is needed by sources such as tone generators.
See also
releaseResources, getNextAudioBlock

Implements AudioSource.

◆ releaseResources()

void ResamplingAudioSource::releaseResources ( )
overridevirtual

Allows the source to release anything it no longer needs after playback has stopped.

This will be called when the source is no longer going to have its getNextAudioBlock() method called, so it should release any spare memory, etc. that it might have allocated during the prepareToPlay() call.

Note that there's no guarantee that prepareToPlay() will actually have been called before releaseResources(), and it may be called more than once in succession, so make sure your code is robust and doesn't make any assumptions about when it will be called.

See also
prepareToPlay, getNextAudioBlock

Implements AudioSource.

◆ resetFilters()

void ResamplingAudioSource::resetFilters ( )
private

◆ setFilterCoefficients()

void ResamplingAudioSource::setFilterCoefficients ( double  c1,
double  c2,
double  c3,
double  c4,
double  c5,
double  c6 
)
private

◆ setResamplingRatio()

void ResamplingAudioSource::setResamplingRatio ( double  samplesInPerOutputSample)

Changes the resampling ratio.

(This value can be changed at any time, even while the source is running).

Parameters
samplesInPerOutputSampleif set to 1.0, the input is passed through; higher values will speed it up; lower values will slow it down. The ratio must be greater than 0

Member Data Documentation

◆ buffer

AudioSampleBuffer ResamplingAudioSource::buffer
private

◆ bufferPos

int ResamplingAudioSource::bufferPos
private

◆ coefficients

double ResamplingAudioSource::coefficients[6]
private

◆ destBuffers

HeapBlock<float*> ResamplingAudioSource::destBuffers
private

◆ filterStates

HeapBlock<FilterState> ResamplingAudioSource::filterStates
private

◆ input

OptionalScopedPointer<AudioSource> ResamplingAudioSource::input
private

◆ lastRatio

double ResamplingAudioSource::lastRatio
private

◆ numChannels

const int ResamplingAudioSource::numChannels
private

◆ ratio

double ResamplingAudioSource::ratio
private

◆ ratioLock

SpinLock ResamplingAudioSource::ratioLock
private

◆ sampsInBuffer

int ResamplingAudioSource::sampsInBuffer
private

◆ srcBuffers

HeapBlock<const float*> ResamplingAudioSource::srcBuffers
private

◆ subSampleOffset

double ResamplingAudioSource::subSampleOffset
private

The documentation for this class was generated from the following file: