ML Sample Generator Project | Phase 2 pt1

A few months ago I already explained a little bit about machine learning. This was because I started working on a project involving machine learning. Here’s a quick refresh on what I want to do and why:

Electronic music production often requires gathering audio samples from different libraries, which, depending on the library and on the platform, can be quite costly as well as time consuming. The core idea of this project was to create a simple application with as few as possible parameters, that will generate a drum sample for the end user via unsupervised machine learning. The interface’s editable parameters enable the user to control the sound of the generated sample and a drag-and-drop space could map a dragged sample’s properties to the parameters. To simplify interaction with the program as much as possible, the dataset should only be learned once and not by the end user. Thus, the application would work with the models rather than the whole algorithm. This would be a benefit as the end result should be a web application where this project is run. Taking a closer look at the machine learning process, the idea was to train the network in the experimentation phase with snare drum samples from the library noiiz. With as many different networks as possible, this would then create a decently sized batch of models from which the best one could be selected for phase 3.

So far I have worked with four different models in different variations to gather some knowledge on what works and what does not. To evaluate them I created a custom GUI.


Producing a GUI for testing purposes was pretty simple and straight-forward. Implementing a Loop Play option required the use of threads, which was a little bit of a challenge but working on the Interface was possible without any major problems thanks to the library PySimpleGUI. The application worked mostly bug free and enabled extensive testing of models and also already saving some great samples. However, as it can be seen below, this GUI is only usable for testing purposes and does not meet the specifications developed in the first phase of this project. For the final product a much simpler app should exist and instead of being standalone it should run on a web server.


An autoencoder is an unsupervised learning method where input data is encoded into a latent vector (therefore the name autoencoder). To get from the input to the latent vector multiple dense layers reduce the dimensionality of the data, creating a bottleneck layer and forcing the encoder to get rid of less important information. This results in data loss but also in a much smaller representation of input data. The latent vector can then be decoded back to produce a similar data sample to the original. While training an autoencoder, the weights and biases of individual neurons are modified to reduce data loss as much as possible.

In this project autoencoders seemed to be a valuable tool as audio samples, even though as short as only 2 seconds, can add up to a huge size. Training with an autoencoder would reduce this information down to only a latent vector with a few dimensions and the trained model itself, which seems perfect for a web application. The past semester resulted in nine different autoencoders, each containing dense layers only. All autoencoders differ from each other by either the amounts of trainable parameters, or the activation functions, or both. The chosen activation functions are rectified linear unit, hyperbolic tangent and sigmoid. These are used in all of the layers of the encoder as well as all layers of the decoder except for the last one to get back to an audio sample (where individual data points are positive and negative). 

Additionally, the autoencoders’ size (as in the amount of trainable parameters) is one of the following three: 

  • Two dense layers with units 9 and 2 (encoder) or 9 and sample length (decoder) with trainable parameters
  • Three dense layers with units 96, 24 and 2 (encoder) or 24, 96 and sample length (decoder) with trainable parameters
  • Four dense layers with units 384, 96, 24 and 2 (encoder) or 24, 96, 384 and sample length (decoder) with trainable parameters

Combining these two attributes results in nine unique models, better understandable as a 3×3 matrix as follows:

Small (2 layers)Medium (3 layers)Big (4 layers)
Rectified linear unitAe small reluAe med reluAe big relu
Hyperbolic tangentAe small tanhAe med tanhAe big tanh
SigmoidAe small sigAe med sigAe big sig

All nine of the autoencoders above have been trained on the same dataset for 700 epochs. We will take a closer look on the results in the next post.

Self Publishing Musician & Label Owner. I do everything myself including video editing, motion graphics, full stack web dev, etc. And I'm a Sound Design student as well.