Async 1.8.0
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Interpolates a lower sampling rate to a higher one. More...
#include <AsyncAudioInterpolator.h>
Public Member Functions | |
AudioInterpolator (int interpolation_factor, const float *filter_coeff, int taps) | |
Constuctor. | |
~AudioInterpolator (void) | |
Destructor. | |
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AudioProcessor (void) | |
Default constuctor. | |
virtual | ~AudioProcessor (void) |
Destructor. | |
int | writeSamples (const float *samples, int len) |
Write audio to the filter. | |
void | flushSamples (void) |
Order a flush of all samples. | |
void | resumeOutput (void) |
Resume output to the sink if previously stopped. | |
void | allSamplesFlushed (void) |
All samples have been flushed by the sink. | |
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AudioSink (void) | |
Default constuctor. | |
virtual | ~AudioSink (void) |
Destructor. | |
bool | registerSource (AudioSource *source) |
Register an audio source to provide samples to this sink. | |
void | unregisterSource (void) |
Unregister the previously registered audio source. | |
bool | isRegistered (void) const |
Check if an audio source has been registered. | |
AudioSource * | source (void) const |
Get the registered audio source. | |
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AudioSource (void) | |
Default constuctor. | |
virtual | ~AudioSource (void) |
Destructor. | |
bool | registerSink (AudioSink *sink, bool managed=false) |
Register an audio sink to provide samples to. | |
void | unregisterSink (void) |
Unregister the previously registered audio sink. | |
bool | isRegistered (void) const |
Check if an audio sink has been registered. | |
AudioSink * | sink (void) const |
Get the registered audio sink. | |
bool | sinkManaged (void) const |
Check if the sink is managed or not. | |
void | handleAllSamplesFlushed (void) |
The registered sink has flushed all samples. | |
Protected Member Functions | |
virtual void | processSamples (float *dest, const float *src, int count) |
Process incoming samples and put them into the output buffer. | |
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void | setInputOutputSampleRate (int input_rate, int output_rate) |
Set the input and output sample rates. | |
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void | sourceResumeOutput (void) |
Tell the source that we are ready to accept more samples. | |
void | sourceAllSamplesFlushed (void) |
Tell the source that all samples have been flushed. | |
bool | setHandler (AudioSink *handler) |
Setup another sink to handle the incoming audio. | |
void | clearHandler (void) |
Clear a handler that was previously setup with setHandler. | |
AudioSink * | handler (void) const |
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int | sinkWriteSamples (const float *samples, int len) |
void | sinkFlushSamples (void) |
bool | setHandler (AudioSource *handler) |
Setup another source to handle the outgoing audio. | |
AudioSource * | handler (void) const |
void | clearHandler (void) |
Clear a handler that was previously setup with setHandler. | |
Interpolates a lower sampling rate to a higher one.
This audio pipe class will interpolate an audio stream up to a higher sampling rate. Interpolation is a process where the sampling rate is increased by an integer factor. After the increase in sampling rate, a lowpass filter must be applied to avoid aliasing effects. This filter is built into this component. However, the filter coefficients (FIR) must be calculated manually.
Use this web page to calculate the coefficients: http://www.dsptutor.freeuk.com/remez/RemezFIRFilterDesign.html
This implementation is based on the multirate FAQ at dspguru.com: http://dspguru.com/info/faqs/mrfaq.htm
Definition at line 117 of file AsyncAudioInterpolator.h.
Async::AudioInterpolator::AudioInterpolator | ( | int | interpolation_factor, |
const float * | filter_coeff, | ||
int | taps ) |
Constuctor.
interpolation_factor | The factor to increase the sample rate with |
filter_coeff | The filter coefficients |
taps | The number of taps in the filter |
Async::AudioInterpolator::~AudioInterpolator | ( | void | ) |
Destructor.
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protectedvirtual |
Process incoming samples and put them into the output buffer.
dest | Destination buffer |
src | Source buffer |
count | Number of samples in the source buffer |
This function should be reimplemented by the inheriting class to do the actual processing of the incoming samples. All samples must be processed, otherwise they are lost and the output buffer will contain garbage.
Implements Async::AudioProcessor.