Overview
This VI implements a basic reverberation algorithm to an audio file, rewrites the file and plays the sound effected clip.
Description
Adding a Reverb Effect to an Audio File
Reverberation is defined as the persistence of sound in a particular space after the original sound is produced. Digital reverberation can be modeled using the following equation:
Y(t) = x(t) + A * Y(t-D) Where x(t) is the original signal and A is the level of the reverberation
reverb.VI takes a dry (not effected) audio input, implements this simple reverb algorithm, writes the effected audio to file and plays sound clip. This example highlights several
important components of non real-time audio signal processing using LabVIEW including the following:
- Usage of the Simple Sound File Read, Write and Play VIs
- Reading and writing individual elements of an array using the Index Array and Replace Array Subset VIs
- Implementing a simple reverberation algorithm
Requirements
Steps to Implement or Execute Code
Additional Information or References
VI Snippet
**This document has been updated to meet the current required format for the NI Code Exchange.**
Example code from the Example Code Exchange in the NI Community is licensed with the MIT license.
Hi
Thanks for this VI, exactly what I am searching for.
Please How: "Sound file read simple.vi" read samples at rate 44.1 k sample/sec, because the sample rate depends on the wav file, not the way of reading it?????
Kindly look at the pic.
Sinan,
You are absolutely correct, that is a typo on my part. The sampling rate of the WAV file is dependent on whatever source captured the audio. The note I made there was to indicate that I was converting a time (seconds) to samples with the knowledge that the WAV file was sampled at 44.1 KHz.
Regards,
Nick
Hi Nick
As you know, the sample rate is the inverse of time interval (dt). I think it is more suiable to use 'get waveform components' to get the time interval then the sample rate.
Thanks
Sinan
University of Mosul / Iraq