Organized Sound Spaces with Machine Learning
Website: | Hamburg Open Online University |
Kurs: | MUTOR: Artificial Intelligence for Music and Multimedia |
Buch: | Organized Sound Spaces with Machine Learning |
Gedruckt von: | Gast |
Datum: | Samstag, 23. November 2024, 10:35 |
Beschreibung
Dr. Kıvanç Tatar
Preface
Organized Sound Spaces with Machine Learning
In this lecture, we will be looking into how we can use a few different machine learning approaches to create abstract sound spaces that are organized in a way where similar sounds are brought closer to each other.
The topics that we will be showcasing in this lecture are in two parts. We will first give a brief introduction to materiality of music, to understand in which musical context the machine learning algorithms are situated.
1. Materiality of Music
Materiality of Music
Before diving into machine learning approaches, it is good to talk about the materiality of music and give a background in what we mean by music and the musical perspective that we are interested in this lecture. First, we will go through the expansion of musical material at the beginning of 20th century. and advancements in futurism and more. And then, we will look into the mid 20th century where we started describing music as organized sound.
1.1 Expansion of Musical Material
Expansion of Musical Material – Futurism and Beyond
Let's start with expansion of musical material. Let's go back to the beginning of 20th century and try to imagine what music meant back then. For example, this piece by Richard Strauss is composed in 1915, and the video below is a recent performance of that Symphony by Oslo Philharmonic (2020), which was performed in 2019.
In the beginning of 20th century we had a quite concrete understanding of what music is and what kind of sounds were musical. We had an understanding of what kind of musical instruments that we could use to make musical sounds. Of course, some composers were disagreeing with that, such as Luigi Russolo. Around the same time in 1913, Luigi Russolo released his futurist manifesto (1913). The beginning of the 20th century is an interesting time to do that, because at that time, our cities are quite different than today. The cities are loud were loud with industrial noises. We have been in the industrial revolution for quite a while at that time. We could imagine that the soundscape of our daily lives in the beginning of 20th century, were quite noisy. As Luigi Russolo mentions in the Art of Noise (1913):
Luigi Russolo suggests further that:
The video below (BBC Radio 3, 2009) is an example from Luigi Russolo's futurist intonarumori. These are devices that are made to make a variety of noises, performed in a musical way. We could call them noise instruments, and those instruments have been proposed in the beginning of 20th century as a way of expanding our sound palette for musical practices.
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This was the expansion of our musical material.
1.2.1 Music as Organized Sound
Music as Organized Sound – Varèse
Later, towards the mid 20th century, some composers started to think about what is then to make music if we can use any sound as musical material? One of those composers was Edgard Varèse, and again, I would like to read a paragraph from an article titled Liberation of Sound by Edgard Varèse (1966).
Varèse opens up the idea of music. Starting from the expansion of musical material, and thinking about, how can we call something music if any sound can be musical? And in his view, it is the organization that matters. And following Varèse's suggestion, we also have other composers coming in and giving us a more comprehensive understanding of the materiality of music:
- any sound can be used to produce music (Russolo 1913);
- music is organized sound (Varese 1966);
- relationships exist between pitch, noise, timbre, and rhythm involving multiple layers (Stockhausen 1972);
- sounds exist in a physical 3-D space (ibid.);
- the timescales of music is in multiple levels infinitesimal, subsample, sample, sound object, meso, macro, supra, and infinite (Roads 2004).
We started from Luigi Russolo's expansion of musical material. And then, we came to Varèse's understanding of music, a generalized definition of music. And then after that, Karlheinz Stockhausen proposes four criteria of electronic music, which he later expands those criterias in the late 20th century. Without getting into the details of those criterias, Stockhausen emphasizes the relationships between pitch, noise, timbre, rhythm, and how music consist of multiple musical layers. Additionally, Stockhausen emphasizes the physical 3-D space that we can use for musical composition. In the beginning of 21st century, Curtis Roads (2004) proposes the time scales of music in multiple levels.
1.2.2 Music as Organized Sound
Music as Organized Sound – Derbyshire
Without getting too much into the details of materiality of music, the video below exemplifies how a composer works with the materiality of music in music production. The composer, Delia Derbyshire is using any sounds, processing them, and making a composition out of those sound organizations and in this video. Delia Derbyshire explains what kind of musical materials she used to compose [arrange] the theme [title music] of Doctor Who.
In the video above, Derbyshire (1965) reveals how she worked with any sounds in her compositions:
In that video, Delia Derbyshire showed us how she was selecting different kinds of audio, and how they relate to each other. For example, how does a sinusoidal sound? How does a square wave sound? After selecting the musical material, we observed how Delia Derbyshire is composing by organizing the selected material in time. Like many other composers, Delia Derbyshire (1965) organized musical material consisted of any sound, to compose [arrange] the Doctor Who theme [title music] using that material:
This documentary from BBC is a great example to see how a composer works with latent audio spaces and temporal organization of sound (figure 1). The latent audio spaces are rooted in our understanding, perception, comprehension and conceptualization of audio similarity and audio dissimilarity. It also showcased how we organize sounds in time, so that we come up with musical form. Whether it is a performance or composition, We can map both of those musical organizations to machine learning approaches. For example, to organize sounds in an abstract space of similarity and dissimilarity, we can use the notion of latent space in machine learning and use various machine learning approaches to create those latent spaces. On the other hand, to organize sounds in time, we can use a variety of sequence modeling approaches or machine learning approaches for time series data.
2. Latent Spaces
Latent Spaces in Machine Learning
But what is a latent space? Let's have an example of that. For example, let's think about a latent space of colours, and let's define a colour as three values of RGB: red, green and blue. Let's think about a way of organizing those colours on a 2D surface, and let's see an example in which a machine learning algorithm generates a latent space of colors:
The machine learning algorithm in the video above is called self organizing maps, which has a predefined number of nodes that moves itself so that it takes the shape of the latent space. And in our case, it takes the shape of the colour space towards the end of the training. We can see a variety of colours and how they relate to each other, how they are similar or dissimilar to each other.
Latent Audio Spaces
Now that we have a musical perspective and background to cover latent audio spaces, we will now dive into the main topic of this lecture. We will be looking into two types of approaches to latent audio spaces: discrete approaches and continuous approaches.
Fig.: Time scaes of music by Curtis Roads (2004).
What is a discrete approach and what is a continuous approach then? Mathematically, both categories that we mention in this section are discrete approaches. However, the continuous approach is in the time scales of micro scale, whereas the discrete approach is working in the time scales of sound object and mesoscale. In the sense that, in discrete latent spaces, we are organizing audio samples that are either fraction of a second to a couple of seconds. In an abstract space, we can think those as sound objects, such as short sound gestures etc. In the continuous latent spaces, we are working in the micro scale. We are working with audio windows that are relatively short, around a few milliseconds to 50 milliseconds. Thus, by putting audio one audio window after another, we treat an audio signal as a time series data where each data point is one audio window. The continuous latent space audio space consists of an organization of audio windows, where each data point represent one audio window.
2.1 Discrete Latent Audio Spaces
Discrete Latent Audio Spaces
To create a discrete latent audio space, we need to do two things. First, we need to find a way to represent an audio sample, and second, we apply a machine learning approach to generate the latent space.
For the first step, we need to find a way to represent audio sample, because we cannot just work with audio as a signal in many of machine learning approaches. In some applications, we will see that the duration of the audio samples vary from one to another, and we need to come up with a series of numbers, that is a feature vector, and the number of numbers in that series of numbers, that is the number of features we need. We then apply a feature extraction algorithm to create the feature vectors for all audio samples in our dataset. There are a variety of ways to extract features from an audio signal. For example, we could extract spectral features.
The are quite a different types of spectral features, and we can cluster them in two main categories. The first category is Fourier transform based approaches in which we have the audio window static across different frequency bands, such as the Mel-frequency Cepstral Coefficients in the figure above. On the other hand, we have the wavelet based transforms such as constant-Q transform in the figure above, in which we have the audio window changing across different frequency bands. For example, for lower frequency bands, we have a longer audio window, and we have a shorter audio window for higher frequency bands. While the wavelet transform based spectral features are computationally heavy, they can represent relatively low frequencies while keeping low latency in the higher frequency bands.
We could use spectral features and calculate the spectrogram of an audio signal so that we could train a machine learning algorithm on those features. Alternatively, we could also use a higher level features. For example in the figure above, we see a representation of audio samples from Karlheinz Stockhausen's composition titled Kontakte. Each segment is labeled on a two dimensional effect estimation model. In that affect estimation model, we have the dimensions of arousal and valence. Arousal refers to the eventfulness of a sound, whether it is calm or it is eventful. Valance refers to the pleasantness of a sound. Whether it is, roughly, something that could initiates negative emotions versus something that could initiate positive emotions. Our previous work on this have showed that there has been a consistency in affect rankings across a variety of people (Fan et al. 2017). Using a dataset of affect ratings on audio samples, we trained the machine learning algorithm to come up with an affect estimation model that gives us a prediction on the pleasantness and eventfulness of an audio recording.
Here, the point is that we can have also higher level features, instead of audio spectrogram features, and train a machine learning algorithm on those features as well.
Let's see an example in the video above, in which we calculate higher level features on a dataset of audio samples. The video above first starts by a chaotic sound, then continuous with a low arousal/calm sound, then a more pleasant sound, then a more eventful and pleasant sound, then another chaotic sound followed by a calm sound again.
2.1.1 Musical Agents Based On SOMs
Musical Agents Based On Self-Organizing Maps
Those higher level feature calculation in the previous section was part of a system titled Musical Agents based on Self Organizing Maps (Tatar 2019).
Self-Organizing Map (SOM) is a Machine Learning algorithm to visualize, represent, and cluster high-dimensional input data with a simpler 2D topology. SOM topologies are typically square and include a finite number of nodes. Node vectors have the same number of dimensions as the input data.
SOMs organize the input data using a 2D similarity grid so that similar data clusters locate closer to each other in the topology. Moreover, SOMs cluster the input data by assigning each input vector to the closest node called the best matching unit (BMU). The figure above shows a SOM with 625 nodes for organizing RGB colors. The training is unsupervised in SOMs, but designers set the topology and the number of nodes in the topology. Each input vector is a training instance of SOM’s learning. During a training instance, a SOM also updates BMU’s neighboring model vectors using a neighborhood function. On each training instance, SOMs update their nodes using the data of an input vector. First, SOMs find the BMU of an input vector. Second, SOMs calculate the Euclidean distance between the input vector and the BMU. Third, SOMs update the BMU by this distance multiplied by the learning rate. The learning rate is a user-set global parameter in the range [0., 1.]. Lower learning rate corresponds to less adaptive and more history-depended SOMs. Depending on the neighboring rate, SOM also updates the neighbors of BMU in the direction of BMU’s update. The update amount becomes less as the neighboring node is further away from the BMU. Therefore, the BMU and its neighboring nodes move closer to input vectors on each training instance.
Without going into too much details, let's have a brief overview of how it works. Here in the image above, at the section A, we have an audio waveform which is processed through an audio segmentation algorithm. Using the audio segmentation, we find different audio segments automatically, which are visualized within rectangles in the image above. The automatic audio segmentation allows us to generate a dataset of audio samples, in which we have audio samples that are either a fraction of a second or a couple of seconds. Using that audio dataset, we extract a set of features in the section b, and then we train self-organizing maps to create a discrete latent audio space, which is the sound memory of the musical agent. The creation of sound memory is similar to the color latent space example that we have covered earlier. Using that discrete latent audio space, we go back to the original recording and we label each audio sample with their corresponding clusters in the latent space. This process gives us a symbolic representation of the audio recording. And it's very interesting that even in this short composition by Iannis Xenakis in the image above, our approach has revealed musical patterns. For example, the composition starts with a cluster pattern of (8,8,5) and then we see (8,8,5) again somewhere in the middle of the composition. Looking at such symbolic representation, we can use sequence modeling algorithms in machine learning to try to find recurring patterns in that symbolic representation. We could use this approach to perform music with it. We could generate a discrete latent audio space by using the self organizing map approach, and combine that with a sequence modeling algorithm, such as factor oracle or recurrent neural networks, to come up with a system that reacts to other performers in real time.
Here in this image above, what we see is we have a performer, who is creating an audio output in real time. This audio output is fed into the system — a musical agent. The system first carries out a feature extraction, and try to understand where the other performers current state is in the self organizing maps,that is in this abstract discrete latent audio space. And the system keeps a history of performance, and it tries to match that histroy to the time series patterns that are in the training dataset. With the patterns that the system has in its memory, patterns such as (8,8,5) as we just mentioned earlier. For example, let's assume that the current state of the audio input is the cluster 8 and then the machine knows that. After the cluster 8, there are two other clusters that the machine could play, either the cluster 8 again, or the cluster 5. After deciding on which cluster to play, choosing sounds from that cluster gives us the audio output and the reaction or the other performer.
The video above is an example performance of the musical agent system and another human performner, which is a work titled A Conversation with Artificial Intelligence (Tatar 2017). Here the machine output is in one channel. And we hear the human performer on the other channel. The video will start with the machine output in one channel, and thus we can tell the first channel that you will hear is generated by the musical agent output.
In that example, we had a feature extraction in which the musical agent were calculating 35 dimensions, that is 35 numbers as a vector to represent an audio sample. The feature set consisted of timbere features, loudness, the duration of the audio segment, and musical mention recognition features. In regards to timbre features and loudness, we were calculating the statistics on those features. Because we have a varying duration in audio samples, per future, let's say loudness, we calculate the mean and standard deviation of the feature across the whole audio sample; which gives us two values, mean and standard deviation, to be added to the feature vector. Using that feature vector, we describe or define the audio sample, to train a machine learning model.
2.1.2 Musical Agents Based On SOM's (cont'd.)
Musical Agents Based On Self-Organizing Maps (continued)
Let's see another example in the video above in which we have a relatively small self organizing maps. In that video, we have clustered audio samples, and we can hear how that clustering worked by listening to the audio samples within clusters.
There are other example systems in the literature that uses a similar approach to the architecture of Musical Agents based of Self-Organizing Maps. One of those systems is called AudioStellar by Garber and Ciccola (2019) that we can watch in the video above. The authors here organize sounds in a 2D space and each dot here will represent an audio sample.
One exciting aspect of audio stellar is the user interaction possibilities that is already available in it. You can create certain paths in the latent space, or interact with the latent space using generative approaches such as particle simulations or swarms, to use the discrete latent audio space in a musically meaningful way.
Let's have a look at the machine learning pipeline behind AudioStellar in the figure above. We have a data set of audio files, which are going through a feature extraction process that the authors refer to as preprocess. All audio is converted to a mono file and then they are calculating a spectrogram feature called Short-time Fourier Transform (STFT), so that they end up with a matrix that represents the audio. From that spectrogram representation, they first run their first machine learning algorithm, which is called principal component analysis. Using principal component analysis, we can keep the main features or the main distribution in the original data set, while representing the dataset in a lower dimensional domain, such as in three dimensions or two dimensions. After that, the authors use a stochastic visualization technique, which is called T Stochastic Neighboring Embedding (t-SNE), to come up with a visualization of the dataset in 2D domain. After running the t-SNE, we can already observe the clusters appearing. Yet, we still don't have the clusters to actually come up with the exact borders of those clusters. Hence, the author apply another machine learning approach called DBScan. And after that pipeline, we have a 2D discrete latent audio space in which we can observe clusters, which are represented as colours, in which we have circles, which are audio samples. We can play with that discrete latent audio space in a musically meaningful way.
2.2 Continuous Latent Audio Spaces
Continuous Latent Audio Spaces
Until now we have covered two examples of discrete latent audio spaces. Now, let's have a dive into the continuous latent audio spaces. As we covered earlier in the time scales of music, while the continuous latent audio spaces are not mathematically continuous, the discrete elements that we have are so perceptually short that we can approach them as if they are iknfinitely small or they are audio quanta.
In the example above, we have a frequency sweep that is starting from 220 Hertz going up to 1000 Hertz or so. If we look at the spectrogram of that sweep in the image above, we can see that it looks almost as if it is a part. The question here is, how can we come up with a more complex way in which we have a 2D representation of an audio space? And in that 2D representation, how can we represent an audio file almost like a path? This is an interesting research question because there are quite interesting musical applications that we could explore using such approach.
For example, here in the image above, we see three colours: red, green, and blue. And in those three colours, we see some small circles. The red coloured path and the blue coloured path are recordings from an audio dataset. Each circle in those paths represent one audio window. Those paths are interesting because this is actually an emerging property of a machine learning approach that I will be talking about later. The latent space in the image above is not a mathematically continuous space, it is still a grid. However, because each audio quanta or audio window is quite small, we could almost approach each audio sample as if it is a continuously changing representation.
2.2.1 Variational Autoencoders
Variational Autoencoders
To understand how we could create a continuous latent audio space, we first need to talk about a specific type of machine learning algorithm called Variational autoencoders. Variational encoders are a type of Deep Learning architectures.
Autoencoders are Deep Learning architectures for generative modelling. The architecture consists of two main modules: an encoder and a decoder. The encoder maps the input data \[ x \in R^L \] to a latent vector \[ z \in R^M \] where z = encoder(x), and M < L. The decoder aims to reconstruct the input data from its latent vector, and ideally, decoder(encoder(x)) = x. The Autoencoder architecture encodes the input data vector to a single point, that is the latent vector. Variational Autoencoder (VAE) is an improved version of the Autoencoder architecture that converts the input data vector to a stochastic distribution over the latent space. This difference is also referred to as the ``reparametrization trick'' (Kingma and Welling 2014 and 2019, Sønderby et al. 2016).
In VAE, the encoder tries to generate a latent space by approximating p(z|x) while the decoder tries to capture the true posterior p(x|z). The vanilla VAE approximates p(z|x) using \[ q(z|x) \in Q \] with the assumption that p(z|x) is in the form of a Gaussian distribution N(0,I). This approximation is referred in the literature as Variational Inference (Kingma and Welling 2014). Specifically, the encoder outputs the mean \[ \mu _M \] and the co-variance \[ \sigma _ M \] as the inputs of the Gaussian distribution function \[ N(z; \mu_M, \sigma^2 _M I) \] over a latent space with M number of dimensions. Hence, the encoder approximates p(z|x) using \[q^* (z|x) = N(z; f(x), g(x)^2 I)\] where \[\mu _M = f(x),\] \[f \in F,\] \[\sigma _ M = g(x)\] and \[g \in G.\] The decoder's input, the latent vector z is sampled from the latent distribution \[ q(z) = N(z; f(x),g(x) ^2 I) \] Hence, the loss function consists of the reconstruction loss and the regularization term of Kullback-Leibler divergence (KLD) between \[ q^* (z|x)\] and \[p^* (z)\]
\[L_{f,g} = \mathbb{E} _{q^* (z)}[logp^* (x|z)] - \alpha \cdot D_{KL}[q^* (z|x)||p^* (z)]\]
In the equation above, KLD multiplier, α is one of the training hyper-parameters of VAE architectures.
2.2.2 Latent Timbre Synthesis
Latent Timbre Synthesis
Let's reiterate that, in the sound applications of Machine Learning architectures, two types of latent spaces are utilized, discrete and continuous latent audio spaces. Continuous latent audio spaces encode audio quanta into a latent space using various Machine Learning approaches such as Variational Autoencoders (VAEs). In those types of latent audio spaces, the input is an audio window, which contains a couple of thousands of samples, and in the durations of a fraction of a second. The network can be either trained directly on the audio signal (Tatar et al. 2023), or any type of audio spectrogram as in the figure below (Tatar et al. 2020).
Latent Timbre Synthesis (Tatar et al. 20202) is a Deep Learning architecture for utilizing an abstract latent timbre space that is generated by training a VAE on the Constant-Q Transform (CQT) spectrograms of audio recordings. Latent Timbre Synthesis allow musicians, composers, and sound designers to synthesize sounds using a latent space of audio that is constrained to the timbre space of the audio recordings in the training set.
In this work titled Latent Timbre Synthesis, we are training a variational autoencoder on a specific type of spectrogram representations called Constant-Q Transform (Schörkhuber and Klapuri 2010), which is a wavelet-based spectrogram. The pipeline in the figure above starts with the calculation of Constant-Q transform (CQT) spectrograms from an audio frame. The CQT spectrogram is passed to a Variational Autoencoder, which creates a latent space of spectrograms. Those latent vectors later are passed to the decoder. Then, the decoder reconstructs the spectrograms, that is single spectrogram vectors. The Variational Autoeencoder in Latent Timbre Synthesis works in the magnitude domain, thus it is only working with real numbers. Hence, the CQT spectrograms generated by the Variational Autoencoder does not have a phase. Thus, in this last step, we are taking the magnitude CQT spectrograms, running a phase reconstruction called Griffin-Lim algorithm to predict the phase of each CQT magnitude spectrogram, and apply inverse Constant-Q Transform to generate an audio signal.
The figure above is the user interface of Latent Timbre Synthesis. We have two audio files, and using latent timbre synthesis and a continuous latent audio space, we can create interpolation curves that are changing in time. What is an interpolation curve? If we set the interpolation closer to 1, or closer to the top, the generated audio for that time point, or that audio frame, is going to be similar sounding to audio 1. As we set the interpolation curve lower and closer to -1 at that particular time point, it is going to be more similar to the audio s. And the rest of the framework is about, loading data sets and different runs in which you could have different machine learning models, and setting up the audio inputs and outputs, and selecting different sounds, and selecting different regions in those sounds to generate different interpolation curves in an interactive way.
The video above exemplifies the sound design process with Latent Timbre Synthesis.
Here on the left. What we see is a representation of the latent space that's generated by. Evaluation. Norton coder. The examples that we just heard. Was from this. Particular trend? Model. Here what we see is using T stochastic neighboring embedding TSNA as an audio stellar. We visualized the latent space generated by the variation motor encoder. One emerging property here is that even though we didn't give any information to the machine learning algorithm. That's where or which audio sample the audio frame is coming from. We can clearly see here that there are certain paths or certain squiggles appearing. If we dive into some of those giggles and zoom into them on the right. We see again. A A red part and a blue part. In those each circle is an audio frame from the original audio recording, and even though we didn't explicitly give this to the. Machine learning pipeline and emerging property was that the audio frames coming from the same audio files? Are appearing now as a part in latent space. And thus this is a continuous latent outer space. The green part that we see is an interesting one. If we take the latent vectors of each. Audio sample in blue part and the red part and if you find the middle point in between them in their original higher number of dimensions. If I'm not mistaken, here 64 dimensions. UM. If we take the middle point in each audio in each latent vector of an audio frame, here we end up with a green path and this green path. Is supposed to be the middle point in between those two audio samples, and using that region between. Red path and the blue pad and creating a variety of green paths as we saw in the previous example, we can do sound design and this is how Layton timber synthesis work. To sum up in the first part of this lecture we covered the materiality of music. We dive into musical composition and sound studies in 2010, 21st century. First, we covered the expansion of musical material starting from the beginning of 20th century with the futurism movement. And then we talked about the new approaches to music as organized. To give you an introduction to materiality of music. And to ground our latent audio space approaches in sound studies and musical practices. After that, we dived into two kinds of latent audio spaces. First, we dived into discrete latent audio spaces in which we worked with audio samples. Ranging from a fraction of a second to a couple of seconds, then we dived into continuous latent audio spaces in which we used. Audio frames that are. Ranging in 1012 milliseconds up to 50 milliseconds. Thank you so much for joining me in this lecture and I'm happy to answer your questions if you have any and feel free to contact me. Have a great day.
The continuous latent audio spaces encode audio samples as a continuous path in the latent space, where each point is encoded from one audio window. In the Figure above, the red circles represent one CQT spectrogram calculated from a single audio window with 1024 samples that are encoded from a single audio sample file, and the path-like appearance of these circles are an emerging property with the latent audio space approach using VAEs. Hence, the red path is a continuous latent audio space encoding of one audio sample recording.
Acknowledgements
Acknowledgements
This work was partially supported by the Wallenberg AI, Autonomous Systems and Software Program – Humanities and Society (WASP-HS) funded by the Marianne and Marcus Wallenberg Foundation and the Marcus and Amalia Wallenberg Foundation.
References
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