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Audio Resampling samples array

Discussion in 'Audio & Video' started by video21region, Apr 5, 2020.

  1. video21region

    video21region

    Joined:
    Sep 5, 2019
    Posts:
    10
    Hi. How I can resampling my samples array in realtime? My audio file has 44100 sample rate, but Unity uses 48000 sample rate by default. I khow that I can use OnAudioFilterRead method. But I don't know which formulas I need used.

    Code (CSharp):
    1. private void OnAudioFilterRead(float[] data, int channels)
    2.     {
    3.         if (playing == false) return;
    4.  
    5.         for (var i = 0; i < data.Length; i++)
    6.         {
    7.             if (lastId > audioData.SampleData.Length - 1)
    8.             {
    9.                 lastId = 0;
    10.             }
    11.      
    12.             data[i] = audioData.SampleData[lastId];
    13.             lastId++;
    14.         }
    15.     }
     
    video21region, Apr 5, 2020
    #1

  2. cnlohr

    cnlohr

    Joined:
    Nov 25, 2020
    Posts:
    10
    I'm also running into a similar issue. If anyone finds the solution, please post. If I find a solution, I will repost here.
     
    cnlohr, Dec 26, 2020
    #2

  3. achimmihca

    achimmihca

    Joined:
    Feb 13, 2016
    Posts:
    266
    achimmihca, Jun 23, 2023
    #3

  4. SeventhString

    SeventhString

    Unity Technologies

    Joined:
    Jan 12, 2023
    Posts:
    290
    Resampling can be quite straightforward of you are going for a simple linear interpolation. The idea is basically to measure where (in time) the resampled value lands between two original samples and to set its value proportionally. Here's a likely valid ChatGPT generated example:

    Code (CSharp):
    1. using System;
    2. using System.Collections.Generic;
    3.  
    4. public class AudioResampler
    5. {
    6.     public static float[] Resample(float[] inputBuffer, int inputSampleRate, int outputSampleRate)
    7.     {
    8.         double sampleRateRatio = (double)outputSampleRate / inputSampleRate;
    9.         int outputBufferLength = (int)(inputBuffer.Length * sampleRateRatio);
    10.  
    11.         float[] outputBuffer = new float[outputBufferLength];
    12.  
    13.         for (int i = 0; i < outputBufferLength; i++)
    14.         {
    15.             double position = i / sampleRateRatio;
    16.             int leftIndex = (int)Math.Floor(position);
    17.             int rightIndex = leftIndex + 1;
    18.  
    19.             double fraction = position - leftIndex;
    20.  
    21.             if (rightIndex >= inputBuffer.Length)
    22.             {
    23.                 outputBuffer[i] = inputBuffer[leftIndex];
    24.             }
    25.             else
    26.             {
    27.                 outputBuffer[i] = (float)(inputBuffer[leftIndex] * (1 - fraction) + inputBuffer[rightIndex] * fraction);
    28.             }
    29.         }
    30.  
    31.         return outputBuffer;
    32.     }
    33. }
    More elaborate algorithms would take into account a few samples before or after to make things smoother and more reliable, but this is a good start. There is plenty of interesting resources if you Google "audio resampling" or "sample rate conversion"

    Cheers!
     
    SeventhString, Jun 27, 2023
    #4