27 const bool deleteInputWhenDeleted,
29 : input (inputSource, deleteInputWhenDeleted),
30 numChannels (channels)
32 jassert (input !=
nullptr);
33 zeromem (coefficients,
sizeof (coefficients));
40 jassert (samplesInPerOutputSample > 0);
43 ratio = jmax (0.0, samplesInPerOutputSample);
50 auto scaledBlockSize = roundToInt (samplesPerBlockExpected * ratio);
51 input->prepareToPlay (scaledBlockSize, sampleRate * ratio);
53 buffer.
setSize (numChannels, scaledBlockSize + 32);
55 filterStates.
calloc (numChannels);
56 srcBuffers.
calloc (numChannels);
57 destBuffers.
calloc (numChannels);
58 createLowPass (ratio);
70 subSampleOffset = 0.0;
76 input->releaseResources();
77 buffer.
setSize (numChannels, 0);
91 if (lastRatio != localRatio)
93 createLowPass (localRatio);
94 lastRatio = localRatio;
97 const int sampsNeeded = roundToInt (info.
numSamples * localRatio) + 3;
101 if (bufferSize < sampsNeeded + 8)
103 bufferPos %= bufferSize;
104 bufferSize = sampsNeeded + 32;
108 bufferPos %= bufferSize;
110 int endOfBufferPos = bufferPos + sampsInBuffer;
113 while (sampsNeeded > sampsInBuffer)
115 endOfBufferPos %= bufferSize;
117 int numToDo = jmin (sampsNeeded - sampsInBuffer,
118 bufferSize - endOfBufferPos);
121 input->getNextAudioBlock (readInfo);
123 if (localRatio > 1.0001)
127 for (
int i = channelsToProcess; --i >= 0;)
128 applyFilter (buffer.
getWritePointer (i, endOfBufferPos), numToDo, filterStates[i]);
131 sampsInBuffer += numToDo;
132 endOfBufferPos += numToDo;
135 for (
int channel = 0; channel < channelsToProcess; ++channel)
141 int nextPos = (bufferPos + 1) % bufferSize;
145 jassert (sampsInBuffer > 0 && nextPos != endOfBufferPos);
147 const float alpha = (float) subSampleOffset;
149 for (
int channel = 0; channel < channelsToProcess; ++channel)
150 *destBuffers[channel]++ = srcBuffers[channel][bufferPos]
151 + alpha * (srcBuffers[channel][nextPos] - srcBuffers[channel][bufferPos]);
153 subSampleOffset += localRatio;
155 while (subSampleOffset >= 1.0)
157 if (++bufferPos >= bufferSize)
162 nextPos = (bufferPos + 1) % bufferSize;
163 subSampleOffset -= 1.0;
167 if (localRatio < 0.9999)
170 for (
int i = channelsToProcess; --i >= 0;)
173 else if (localRatio <= 1.0001 && info.numSamples > 0)
176 for (
int i = channelsToProcess; --i >= 0;)
179 FilterState& fs = filterStates[i];
183 fs.y2 = fs.x2 = *(endOfBuffer - 1);
191 fs.y1 = fs.x1 = *endOfBuffer;
195 jassert (sampsInBuffer >= 0);
198 void ResamplingAudioSource::createLowPass (
const double frequencyRatio)
200 const double proportionalRate = (frequencyRatio > 1.0) ? 0.5 / frequencyRatio
201 : 0.5 * frequencyRatio;
204 const double nSquared = n * n;
207 setFilterCoefficients (c1,
211 c1 * 2.0 * (1.0 - nSquared),
215 void ResamplingAudioSource::setFilterCoefficients (
double c1,
double c2,
double c3,
double c4,
double c5,
double c6)
217 const double a = 1.0 / c4;
225 coefficients[0] = c1;
226 coefficients[1] = c2;
227 coefficients[2] = c3;
228 coefficients[3] = c4;
229 coefficients[4] = c5;
230 coefficients[5] = c6;
233 void ResamplingAudioSource::resetFilters()
235 if (filterStates !=
nullptr)
236 filterStates.
clear ((
size_t) numChannels);
239 void ResamplingAudioSource::applyFilter (
float* samples,
int num, FilterState& fs)
243 const double in = *samples;
245 double out = coefficients[0] * in
246 + coefficients[1] * fs.x1
247 + coefficients[2] * fs.x2
248 - coefficients[4] * fs.y1
249 - coefficients[5] * fs.y2;
252 if (! (out < -1.0e-8 || out > 1.0e-8))
261 *samples++ = (float) out;
void prepareToPlay(int samplesPerBlockExpected, double sampleRate) override
Tells the source to prepare for playing.
int numSamples
The number of samples in the buffer which the callback is expected to fill with data.
void getNextAudioBlock(const AudioSourceChannelInfo &) override
Called repeatedly to fetch subsequent blocks of audio data.
void setSize(int newNumChannels, int newNumSamples, bool keepExistingContent=false, bool clearExtraSpace=false, bool avoidReallocating=false)
Changes the buffer's size or number of channels.
void releaseResources() override
Allows the source to release anything it no longer needs after playback has stopped.
const Type * getReadPointer(int channelNumber) const noexcept
Returns a pointer to an array of read-only samples in one of the buffer's channels.
void calloc(SizeType newNumElements, const size_t elementSize=sizeof(ElementType))
Allocates a specified amount of memory and clears it.
void flushBuffers()
Clears any buffers and filters that the resampler is using.
Base class for objects that can produce a continuous stream of audio.
int getNumChannels() const noexcept
Returns the number of channels of audio data that this buffer contains.
~ResamplingAudioSource() override
Destructor.
void clear(SizeType numElements) noexcept
This fills the block with zeros, up to the number of elements specified.
ResamplingAudioSource(AudioSource *inputSource, bool deleteInputWhenDeleted, int numChannels=2)
Creates a ResamplingAudioSource for a given input source.
int startSample
The first sample in the buffer from which the callback is expected to write data. ...
AudioBuffer< float > * buffer
The destination buffer to fill with audio data.
void setResamplingRatio(double samplesInPerOutputSample)
Changes the resampling ratio.
Type * getWritePointer(int channelNumber) noexcept
Returns a writeable pointer to one of the buffer's channels.
int getNumSamples() const noexcept
Returns the number of samples allocated in each of the buffer's channels.
Commonly used mathematical constants.
Used by AudioSource::getNextAudioBlock().
Automatically locks and unlocks a mutex object.
void clear() noexcept
Clears all the samples in all channels.