# Resampling an AudioSampleBuffer
**URL:** https://forum.juce.com/t/resampling-an-audiosamplebuffer/14287
**Category:** Audio Plugins
**Created:** [February 10, 2015, 2:42am UTC](https://forum.juce.com/t/resampling-an-audiosamplebuffer/14287 "2015-02-10T02:42:03Z")
**Posts on this page:** 9
**Page:** 1
### Author:  [@lapsang](https://forum.juce.com/u/lapsang)
#### Post date: [February 10, 2015, 2:42am UTC](https://forum.juce.com/t/resampling-an-audiosamplebuffer/14287/1 "2015-02-10T02:42:03Z")
Is there a "simple" way to resample an AudioSampleBuffer? I tried playing with ResamplingAudioSource but I couldn't really figure out how to use it for a fixed buffer (not live).
What should I do or where should I look?
---
### Author:  [@jules](https://forum.juce.com/u/jules)
#### Post date: [February 10, 2015, 9:16am UTC](https://forum.juce.com/t/resampling-an-audiosamplebuffer/14287/2 "2015-02-10T09:16:31Z")
There's also the LagrangeInterpolator, which is a bit more low-level, if that's what you're looking for?
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### Author:  [@lapsang](https://forum.juce.com/u/lapsang)
#### Post date: [February 10, 2015, 3:24pm UTC](https://forum.juce.com/t/resampling-an-audiosamplebuffer/14287/3 "2015-02-10T15:24:00Z")
Thanks for the reply. I basically need to load a IR sample in a AudioSampleBuffer and stretch it according to the host samplerate and a stretch factor.
I tried the LagrangeInterpolator but there's something wrong with my implementation as it doesn't do exactly what it should (or at least what I think it should do).
When then I use the new buffer for the convolution if the stretch is \< 1 it looks like the sample is slower (while it should be faster) and usually when it's \> 1 i get strange repetitions only on left channel (and sometimes a huge click/pop and no audio anymore).
I am probably overthinking here, but I just need to change the speed of this AudioSampleBuffer according to the host samplerate and a stretch factor.
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### Author:  [@lapsang](https://forum.juce.com/u/lapsang)
#### Post date: [February 15, 2015, 12:27am UTC](https://forum.juce.com/t/resampling-an-audiosamplebuffer/14287/4 "2015-02-15T00:27:33Z")
\*bump\*
Any help with this?
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### Author:  [@DaveH](https://forum.juce.com/u/DaveH)
#### Post date: [September 15, 2017, 5:41pm UTC](https://forum.juce.com/t/resampling-an-audiosamplebuffer/14287/5 "2017-09-15T17:41:12Z")
Here’s my interpretation of how to use the LagrangeInterpolator, but unfortunately it crashes with an exception on the second channel, any idea why? I really need this to work.
waveBuffer (AudioSampleBuffer) is the destination buffer that’s in the class…
```
void Buffer::fillBuffer(AudioFormatReader *reader, double currentSampleRate)
{
AudioSampleBuffer temp;
waveBuffer.clear();
temp.clear();
double ratio = currentSampleRate / reader->sampleRate;
temp.setSize((int)reader->numChannels, (int)reader->lengthInSamples);
waveBuffer.setSize((int)reader->numChannels, (int)((double)reader->lengthInSamples * ratio));
reader->read(&temp, 0, (int)reader->lengthInSamples, 0, true, true);
ScopedPointer resampler = new LagrangeInterpolator();
const float **inputs = temp.getArrayOfReadPointers();
float **outputs = waveBuffer.getArrayOfWritePointers();
for (int c = 0; c < waveBuffer.getNumChannels(); c++)
{
resampler->reset();
resampler->process(ratio, inputs[c], outputs[c], waveBuffer.getNumSamples());
}
}
```
---
### Author:  [@DaveH](https://forum.juce.com/u/DaveH)
#### Post date: [September 15, 2017, 7:35pm UTC](https://forum.juce.com/t/resampling-an-audiosamplebuffer/14287/6 "2017-09-15T19:35:18Z")
OK I’ve discovered the ratio was 1/ the correct value but it still crashes when down-sampling, it seems to work fine when up-sampling. Is the LagrangeAlgorithm code broken?
The ‘in’ gets unreadable here:-
```
else
{
for (int i = numOut; --i >= 0;)
{
while (pos < actualRatio)
{
pushInterpolationSample (lastInputSamples, *in++);
pos += 1.0;
}
pos -= actualRatio;
*out++ = InterpolatorType::valueAtOffset (lastInputSamples, jmax (0.0f, 1.0f - (float) pos));
}
}
```
---
### Author:  [@DaveH](https://forum.juce.com/u/DaveH)
#### Post date: [September 15, 2017, 8:16pm UTC](https://forum.juce.com/t/resampling-an-audiosamplebuffer/14287/7 "2017-09-15T20:16:21Z")
OK I’ve got it now, there is no filtering in the Lagrange code but ho-hum, here’s the new version:
\*Note I now divide the incoming length by the ratio.
```
AudioSampleBuffer temp;
sofar = 0;
waveBuffer.clear();
temp.clear();
double ratio = reader->sampleRate / hostSampleRate;
temp.setSize((int)reader->numChannels, (int)reader->lengthInSamples);
waveBuffer.setSize((int)reader->numChannels, (int)(((double)reader->lengthInSamples) / ratio));
reader->read(&temp, 0, (int)reader->lengthInSamples, 0, true, true);
ScopedPointer resampler = new LagrangeInterpolator();
const float **inputs = temp.getArrayOfReadPointers();
float **outputs = waveBuffer.getArrayOfWritePointers();
for (int c = 0; c < waveBuffer.getNumChannels(); c++)
{
resampler->reset();
resampler->process(ratio, inputs[c], outputs[c], waveBuffer.getNumSamples());
}
```
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### Author:  [@WilliamkWusik](https://forum.juce.com/u/WilliamkWusik)
#### Post date: [August 5, 2019, 7:52pm UTC](https://forum.juce.com/t/resampling-an-audiosamplebuffer/14287/8 "2019-08-05T19:52:54Z")
Just saw this today. 😉 The code works great, BUT, for each channel you need its own Lagrange object.
`if you're resampling multiple channels, make sure each one uses its own object.`
---
### Author:  [@DaveH](https://forum.juce.com/u/DaveH)
#### Post date: [August 10, 2019, 9:32am UTC](https://forum.juce.com/t/resampling-an-audiosamplebuffer/14287/9 "2019-08-10T09:32:59Z")
Yes of course. 🙂