Month: July 2021

Convolutional Networks

Convolutional networks include one or more convolutional layers. These layers are typically used for feature extraction. Stacking multiple on top of each other often can extract very detailed features. Depending on the input shape of the data, convolutional layers can be one- or multidimensional, but are usually 2D as they are mainly used for working with  images.  The  feature extraction can be achieved by applying filters to the input data. The image below shows a very simple black and white (or pink & white) image with a size 3 filter that can detect vertical left-sided edges. The resulting image can then be shrinked down without losing as much data as reducing the original’s dimensions would.

In this project, all models containing convolutional layers are based off of WavGAN. For this cutting the samples down to a length of 16384 was necessary, as WavGAN only works with windows of this size. In detail, the two models consist of five convolutional layers, each followed by a leaky rectified linear unit activation function and one final dense layer afterwards. Both models were again trained for 700 epochs.

Convolutional Autoencoder

The convolutional autoencoder produces samples only in the general shape of a snare drum. There is an impact and a tail but like the small autoencoders, it is clicky. In contrast to the normal autoencoders, the whole sound is not noisy though but rather a ringing sound. The latent vector does change the sound but playing the sound to a third party would not result in them guessing that this should be a snare drum.

GAN

The generative adversarial network worked much better than the autoencoder. While still being far from a snare drum sound, it produced a continuous latent space with samples resembling the shape of a snare drum. The sound itself however very closely resembles a bitcrushed version of the original samples. It would be interesting to develop this further as the current results suggest that there is just something wrong with the layers, but the network takes very long to train which might be due to the need of a custom implementation of the train function.

Variational Autoencoder

Variational autoencoders are a sub-type of autoencoders. Their big difference to a vanilla autoencoder is the encoder’s last layer, the sampling layer. With this, variational autoencoders always provide a continuous latent space, which is much better for generative models than just to sample from what has been provided. This is achieved by having the encoder output two different vectors instead of one: one for standard deviation and one for the mean. This provides a distribution rather than a single point, leading to the decoder learning that an area is responsible for a feature and not a single sample.

Training the variational autoencoder was especially troublesome as it required a custom class with it’s own train step function. The difficulty with this type of model is that the right mix between reconstruction loss and kl loss has to be found, otherwise the model produces unhelpful results. The currently trained models all have a ramp up time of 30,000 batches until full effect of the kl loss. This value gets multiplied by a different actor depending on the model. The trained versions are with a factor of 0.01 (A), 0.001(B), as well as 0.0001(C). Model A produces a snare drum like sound, but is very metallic. Additionally instead of having a continuous latent space, the sample does not change at all. Model B produces a much better sample but still does not include much changes. The main changes are the volume of the sample as well as it getting a little bit more clicky towards the edges of the y axis. Model C has much more different sounds, but the continuity is more or less not present. In some areas the sample seems to get slightly filtered over one third of the vector’s axis but then rapidly changes the sound multiple times over the next 10%. But still, out of the three variational autoencoders model C produced the best results.

Next Steps

As I briefly mentioned before, this project will ultimately run on a web server which means the next steps will be deciding how to run this app. Since all of the project has been written in python so far Django would be a good solution. But since TensorFlow offers a JavaScript Library as well this is not the only possible way to go. You will find out more about this in the next semester.

Autoencoder Results

As mentioned in the post before I have trained nine autoencoders to (re)produce snare drum samples. For easier comparison I have visualized the results below. Each image shows the location of all ~7500 input samples.

Rectified Linear Unit

All three graphics portray how the samples are mostly close together but some are very far out. A continuous representation is with all three models not possible. Reducing the latent vector’s maximum on both axes definitely helps, but even then the resulting samples are not too pleasing to hear. The small network has clicks in the beginning and generates very silent but noisy tails after the initial impact. The medium network includes some quite okay samples but moving around in the latent space often   produces   similar  but  less   pronounced issues as the small network. And the big network produces the best sounding samples but has no continuous changes.

Hyperbolic Tangent

These three networks each produce different patterns with a cluster at (0|0). The similarities between the medium and the big network lead me to believe that there is a smooth transition between random noise, to forming small clusters, to turning 45° clockwise and refining the clusters when increasing the number of trainable parameters. Just like the relu version, the reproduced audio samples of the small network contain clicks. The samples are however much better. The medium sized network is the best one out of all the trained models. It produces  mostly  good  samples  and has a continuous latent space. One issue is however that there are still some clicky areas in the latent space. The big network is the second best overall as it mostly lacks a continuous latent space as well. The produced audio samples are however very pleasing to hear and resemble the originals quite well.

Sigmoid

This group shows a clear tendency to cluster up the more trainable parameters exist. While in the above two groups the medium and the big network produced better results, in this case the small network is by far the best. The big network delivers primarily noisy audio samples and the medium network very noisy ones as well but they are better identifiable as snare drum sounds. The small network has by far the closest sounds to the originals but produces clicks at the beginning as well.

In the third part of this series we will take a closer look at the other models.

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.

The 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.

Autoencoders

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 unit	Ae small relu	Ae med relu	Ae big relu
Hyperbolic tangent	Ae small tanh	Ae med tanh	Ae big tanh
Sigmoid	Ae small sig	Ae med sig	Ae 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.