rochars / wave-resampler Goto Github PK

import { resampler } from 'wave-resampler';
let newSamples = resample(samples, 48000, 44100);

This is how it works for me:

import resampler from 'wave-resampler';
let newSamples = resampler.resample(samples, 48000, 44100);

Thanks!

Hello and thanks for making this available and free to use!

Am I correct to assume that different channels should be processed using resample(...) independently, rather than resampling chunks of interleaved stereo audio? If so, it may be helpful to include specifying the correct format of input in the docs. 😄

Thanks again!

I have a Buffer with 48kHz audio as PCM signed 16-bit little-endian (FFmpeg format s16le). Then I'm doing this simple test to find out that the resample function is affecting the source data:

while (...) {
  const audioBuffer = Buffer.from(...);
  const resampled = WaveResampler.resample(audioBuffer, 48000, 16000, {
    LPF: true,
  });
  // `resampled` is not used. `audioBuffer` has been modified but it shouldn't.
  fs.appendFileSync("/tmp/test.raw", audioBuffer);
}

Now playing this audio with FFmpeg, the audio contents have been destroyed and everything is just noise, except that the original audio can be kind of noticed very subtle in the background:

ffplay -f s16le -ar 48k -ac 1 /tmp/test.raw

Setting LPF: false avoids the issue. The mistake seems to be in the downsample_() function, which saves data directly into the source buffer, when I believe it shouldn't:

samples[i]  = filter.filter(samples[i]);

As it is, no LPF downsampling can be done with this library with default settings, because the resulting audio is garbled.

All the examples show down sampling. Does this support up sampling?