Skip to main content ↑ ↓

Notice

Please note that most of the software linked on this forum is likely to be safe to use. If you are unsure, feel free to ask in the relevant topics, or send a private message to an administrator or moderator. To help curb the problems of false positives, or in the event that you do find actual malware, you can contribute through the article linked here.
Topic: One updated and one new audio processing DSP library (Read 592 times) previous topic - next topic
0 Members and 1 Guest are viewing this topic.

One updated and one new audio processing DSP library

I have made some significant improvements to the audio-stretch library and have created a sort-of complimentary library for audio resampling. Both are available on my GitHub account and both include pre-built Windows executables intended for demo purposes (although they’re perfectly usable as command-line applications):

Audio-stretch library on GitHub

Audio-resampler library on GitHub

The audio-stretch library has been up for years, but someone recently suggested a couple improvements and I also made some small performance enhancements and edge-case bug fixes. It implements time domain harmonic scaling (TDHS) to stretch (or squeeze) audio data in the time domain (i.e., it changes the duration of audio clips without changing their pitch). This is useful for slowing audio to make it more intelligible or speeding it up to more quickly digest audio books or podcasts.

Previously the stretch ratio was limited to just 0.5X - 2.0X, but by cascading two instances internally I doubled that range to 0.25X - 4.0X. The other new feature is the ability to detect silence gaps in the audio and use a different ratio for those portions. This is useful for further speeding up speech and keeping it intelligible.

The audio resampling code is a new repository, but I actually wrote it over 15 years ago for the project described in this post where it's used to oversample audio 256X. I’ve used it in several places and made improvements over the years since and thought it might be useful to others. So I wrote a command-line demo for it with an option for including a biquad lowpass if desired.

It’s a relatively simple implementation intended for real time or embedded applications while not necessarily offering the best offline quality (there are several great examples of that around). It’s all one C module (except for the biquad) and should be trivial to build and integrate with applications, and it can easily be fine-tuned for the environment with respect to CPU and memory usage (and at the higher quality settings is quite respectable). Most recently I put it in a real time ASRC used for synchronization in a wireless speaker (it includes the functionality to query the current phase used in the feedback loop).

I say that these libraries are complementary because while they are certainly useful independently, together they can provide the tempo/pitch/playback controls that media players often have. And although it’s subjective, I believe that they offer somewhat higher quality than the other free implementations I’ve tried (especially for playing slower).

Hope someone might find something useful here...   :)