SampleMatch: Drum Sample Retrieval by Musical Context

Modern digital music production typically involves combining numerous acoustic elements to compile a piece of music. Important types of such elements are drum samples, which determine the characteristics of the percussive components of the piece. Artists must use their aesthetic judgement to assess whether a given drum sample fits the current musical context. However, selecting drum samples from a potentially large library is tedious and may interrupt the creative flow. In this work, we explore the automatic drum sample retrieval based on aesthetic principles learned from data. As a result, artists can rank the samples in their library by fit to some musical context at different stages of the production process (i.e., by fit to incomplete song mixtures). To this end, we use contrastive learning to maximize the score of drum samples originating from the same song as the mixture. We conduct a listening test to determine whether the human ratings match the automatic scoring function. We also perform objective quantitative analyses to evaluate the efficacy of our approach.Comment: 8 pages, 3 figures, 1 table; Accepted at the ISMIR conference, Bengaluru, India, 202

Lauluyhtyeen intonaation automaattinen määritys

The objective of this study is a specific music signal processing task, primarily intended to help vocal ensemble singers practice their intonation. In this case intonation is defined as deviations of pitch in relation to the note written in the score which are small, less than a semitone. These can be either intentional or unintentional. Practicing intonation is typically challenging without an external ear. The algorithm developed in this thesis combined with the presented application concept can act as the external ear, providing real-time information on intonation to support practicing. The method can be applied to the analysis of recorded material as well. The music signal generated by a vocal ensemble is polyphonic. It contains multiple simultaneous tones with partly or completely overlapping harmonic partials. We need to be able to estimate the fundamental frequency of each tone, which then indicates the pitch of each singer. Our experiments show, that the fundamental frequency estimation method based on the Fourier analysis developed in this thesis can be applied to the automatic analysis of vocal ensembles. A sufficient frequency resolution can be achieved without compromising the time resolution too much by using an adequately sized window. The accuracy and robustness can be further increased by taking advantage of solitary partials. The greatest challenge turned out to be the estimation of tones in octave and unison relationships. These intervals are fairly common in tonal music. This question requires further investigation or another type of approach.Tässä työssä tutkitaan erityistä musiikkisignaalin analysointitehtävää, jonka tarkoi- tuksena on auttaa lauluyhtyelaulajia intonaation harjoittelussa. Intonaatiolla tar- koitetaan tässä yhteydessä pieniä, alle puolen sävelaskeleen säveltasoeroja nuottiin kirjoitettuun sävelkorkeuteen nähden, jotka voivat olla joko tarkoituksenmukaisia tai tahattomia. Intonaation harjoittelu on tyypillisesti haastavaa ilman ulkopuolista korvaa. Työssä kehitetty algoritmi yhdessä esitellyn sovelluskonseptin kanssa voi toimia harjoittelutilanteessa ulkopuolisena korvana tarjoten reaaliaikaista tietoa intonaatiosta harjoittelun tueksi. Vaihtoehtoisesti menetelmää voidaan hyödyntää harjoitusäänitteiden analysointiin jälkikäteen. Lauluyhtyeen tuottama musiikki- signaali on polyfoninen. Se sisältää useita päällekkäisiä säveliä, joiden osasävelet menevät toistensa kanssa osittain tai kokonaan päällekkäin. Tästä signaalista on pystyttävä tunnistamaan kunkin sävelen perustaajuus, joka puolestaan kertoo lau- lajan laulaman sävelkorkeuden. Kokeellisten tulosten perusteella työssä kehitettyä Fourier-muunnokseen perustuvaa taajuusanalyysiä voidaan soveltaa lauluyhtyeen intonaation automaattiseen määritykseen, kun nuottiin kirjoitettua sointua hyödyn- netään analyysin lähtötietona. Sopivankokoista näyteikkunaa käyttämällä päästiin riittävään taajuusresoluutioon aikaresoluution säilyessä kohtuullisena. Yksinäisiä osasäveliä hyödyntämällä voidaan edelleen parantaa tarkkuutta ja toimintavar- muutta. Suurimmaksi haasteeksi osoittautui oktaavi- ja priimisuhteissa olevien intervallien luotettava määritys. Näitä intervallisuhteita esiintyy tonaalisessa musii- kissa erityisen paljon. Tämä kysymys vaatii vielä lisätutkimusta tai uudenlaista lähestymistapaa

Audio source separation for music in low-latency and high-latency scenarios

Aquesta tesi proposa mètodes per tractar les limitacions de les tècniques existents de separació de fonts musicals en condicions de baixa i alta latència. En primer lloc, ens centrem en els mètodes amb un baix cost computacional i baixa latència. Proposem l'ús de la regularització de Tikhonov com a mètode de descomposició de l'espectre en el context de baixa latència. El comparem amb les tècniques existents en tasques d'estimació i seguiment dels tons, que són passos crucials en molts mètodes de separació. A continuació utilitzem i avaluem el mètode de descomposició de l'espectre en tasques de separació de veu cantada, baix i percussió. En segon lloc, proposem diversos mètodes d'alta latència que milloren la separació de la veu cantada, gràcies al modelatge de components específics, com la respiració i les consonants. Finalment, explorem l'ús de correlacions temporals i anotacions manuals per millorar la separació dels instruments de percussió i dels senyals musicals polifònics complexes.Esta tesis propone métodos para tratar las limitaciones de las técnicas existentes de separación de fuentes musicales en condiciones de baja y alta latencia. En primer lugar, nos centramos en los métodos con un bajo coste computacional y baja latencia. Proponemos el uso de la regularización de Tikhonov como método de descomposición del espectro en el contexto de baja latencia. Lo comparamos con las técnicas existentes en tareas de estimación y seguimiento de los tonos, que son pasos cruciales en muchos métodos de separación. A continuación utilizamos y evaluamos el método de descomposición del espectro en tareas de separación de voz cantada, bajo y percusión. En segundo lugar, proponemos varios métodos de alta latencia que mejoran la separación de la voz cantada, gracias al modelado de componentes que a menudo no se toman en cuenta, como la respiración y las consonantes. Finalmente, exploramos el uso de correlaciones temporales y anotaciones manuales para mejorar la separación de los instrumentos de percusión y señales musicales polifónicas complejas.This thesis proposes specific methods to address the limitations of current music source separation methods in low-latency and high-latency scenarios. First, we focus on methods with low computational cost and low latency. We propose the use of Tikhonov regularization as a method for spectrum decomposition in the low-latency context. We compare it to existing techniques in pitch estimation and tracking tasks, crucial steps in many separation methods. We then use the proposed spectrum decomposition method in low-latency separation tasks targeting singing voice, bass and drums. Second, we propose several high-latency methods that improve the separation of singing voice by modeling components that are often not accounted for, such as breathiness and consonants. Finally, we explore using temporal correlations and human annotations to enhance the separation of drums and complex polyphonic music signals

Analysis and resynthesis of polyphonic music

This thesis examines applications of Digital Signal Processing to the analysis, transformation, and resynthesis of musical audio. First I give an overview of the human perception of music. I then examine in detail the requirements for a system that can analyse, transcribe, process, and resynthesise monaural polyphonic music. I then describe and compare the possible hardware and software platforms. After this I describe a prototype hybrid system that attempts to carry out these tasks using a method based on additive synthesis. Next I present results from its application to a variety of musical examples, and critically assess its performance and limitations. I then address these issues in the design of a second system based on Gabor wavelets. I conclude by summarising the research and outlining suggestions for future developments

Proceedings of the 7th Sound and Music Computing Conference

Proceedings of the SMC2010 - 7th Sound and Music Computing Conference, July 21st - July 24th 2010

Gibbs sampling with people

A core problem in cognitive science and machine learning is to understand how humans derive semantic representations from perceptual objects, such as color from an apple, pleasantness from a musical chord, or seriousness from a face. Markov Chain Monte Carlo with People (MCMCP) is a prominent method for studying such representations, in which participants are presented with binary choice trials constructed such that the decisions follow a Markov Chain Monte Carlo acceptance rule. However, while MCMCP has strong asymptotic properties, its binary choice paradigm generates relatively little information per trial, and its local proposal function makes it slow to explore the parameter space and find the modes of the distribution. Here we therefore generalize MCMCP to a continuous-sampling paradigm, where in each iteration the participant uses a slider to continuously manipulate a single stimulus dimension to optimize a given criterion such as 'pleasantness'. We formulate both methods from a utility-theory perspective, and show that the new method can be interpreted as 'Gibbs Sampling with People' (GSP). Further, we introduce an aggregation parameter to the transition step, and show that this parameter can be manipulated to flexibly shift between Gibbs sampling and deterministic optimization. In an initial study, we show GSP clearly outperforming MCMCP; we then show that GSP provides novel and interpretable results in three other domains, namely musical chords, vocal emotions, and faces. We validate these results through large-scale perceptual rating experiments. The final experiments use GSP to navigate the latent space of a state-of-the-art image synthesis network (StyleGAN), a promising approach for applying GSP to high-dimensional perceptual spaces. We conclude by discussing future cognitive applications and ethical implications

Gibbs sampling with people

A core problem in cognitive science and machine learning is to understand how humans derive semantic representations from perceptual objects, such as color from an apple, pleasantness from a musical chord, or seriousness from a face. Markov Chain Monte Carlo with People (MCMCP) is a prominent method for studying such representations, in which participants are presented with binary choice trials constructed such that the decisions follow a Markov Chain Monte Carlo acceptance rule. However, while MCMCP has strong asymptotic properties, its binary choice paradigm generates relatively little information per trial, and its local proposal function makes it slow to explore the parameter space and find the modes of the distribution. Here we therefore generalize MCMCP to a continuous-sampling paradigm, where in each iteration the participant uses a slider to continuously manipulate a single stimulus dimension to optimize a given criterion such as 'pleasantness'. We formulate both methods from a utility-theory perspective, and show that the new method can be interpreted as 'Gibbs Sampling with People' (GSP). Further, we introduce an aggregation parameter to the transition step, and show that this parameter can be manipulated to flexibly shift between Gibbs sampling and deterministic optimization. In an initial study, we show GSP clearly outperforming MCMCP; we then show that GSP provides novel and interpretable results in three other domains, namely musical chords, vocal emotions, and faces. We validate these results through large-scale perceptual rating experiments. The final experiments use GSP to navigate the latent space of a state-of-the-art image synthesis network (StyleGAN), a promising approach for applying GSP to high-dimensional perceptual spaces. We conclude by discussing future cognitive applications and ethical implications

Cross-modal and synaesthetic perception in music and vision

This thesis is concerned with the cross-modal and synaesthetic perception of musical and visual stimuli. Each of these types of perception has been researched separately, and a hypothesis is presented here that accounts for both cross-modal matching and the development of synaesthesia. This hypothesis claims that sensory information can be evaluated in another modality by using a scale of comparison in that modality. The first set of experiments examines normal subjects performing cross-modal matching with coloured circles and auditory stimuli that vary in complexity. It is shown that subjects use a variety of scales of comparison from both visual and auditory modalities to form matches. As the stimuli increase in complexity, the individual variation in cross-modal matching also increases. The second set of experiments examines matching performance using higher order stimuli, by having subjects evaluate fragments of melodies and complete melodies on affective and descriptive adjective scales. Melodies were also matched with landscape scenes to examine if subjects could form matches between two highly complex sets of stimuli. The final experiments examine synaesthetic associations with colour, evoked from music, letters, numbers, and other categorical information. Common features of synaesthesia from a population of synaesthetes are identified, and experiments performed to test the interference of the synaesthetic associations. Additional experiments are presented that explore the superior short-term memory of one synaesthete, and the role of his associations as a mnemonic device

Trennung und Schätzung der Anzahl von Audiosignalquellen mit Zeit- und Frequenzüberlappung

Everyday audio recordings involve mixture signals: music contains a mixture of instruments; in a meeting or conference, there is a mixture of human voices. For these mixtures, automatically separating or estimating the number of sources is a challenging task. A common assumption when processing mixtures in the time-frequency domain is that sources are not fully overlapped. However, in this work we consider some cases where the overlap is severe — for instance, when instruments play the same note (unison) or when many people speak concurrently ("cocktail party") — highlighting the need for new representations and more powerful models. To address the problems of source separation and count estimation, we use conventional signal processing techniques as well as deep neural networks (DNN). We first address the source separation problem for unison instrument mixtures, studying the distinct spectro-temporal modulations caused by vibrato. To exploit these modulations, we developed a method based on time warping, informed by an estimate of the fundamental frequency. For cases where such estimates are not available, we present an unsupervised model, inspired by the way humans group time-varying sources (common fate). This contribution comes with a novel representation that improves separation for overlapped and modulated sources on unison mixtures but also improves vocal and accompaniment separation when used as an input for a DNN model. Then, we focus on estimating the number of sources in a mixture, which is important for real-world scenarios. Our work on count estimation was motivated by a study on how humans can address this task, which lead us to conduct listening experiments, confirming that humans are only able to estimate the number of up to four sources correctly. To answer the question of whether machines can perform similarly, we present a DNN architecture, trained to estimate the number of concurrent speakers. Our results show improvements compared to other methods, and the model even outperformed humans on the same task. In both the source separation and source count estimation tasks, the key contribution of this thesis is the concept of “modulation”, which is important to computationally mimic human performance. Our proposed Common Fate Transform is an adequate representation to disentangle overlapping signals for separation, and an inspection of our DNN count estimation model revealed that it proceeds to find modulation-like intermediate features.Im Alltag sind wir von gemischten Signalen umgeben: Musik besteht aus einer Mischung von Instrumenten; in einem Meeting oder auf einer Konferenz sind wir einer Mischung menschlicher Stimmen ausgesetzt. Für diese Mischungen ist die automatische Quellentrennung oder die Bestimmung der Anzahl an Quellen eine anspruchsvolle Aufgabe. Eine häufige Annahme bei der Verarbeitung von gemischten Signalen im Zeit-Frequenzbereich ist, dass die Quellen sich nicht vollständig überlappen. In dieser Arbeit betrachten wir jedoch einige Fälle, in denen die Überlappung immens ist zum Beispiel, wenn Instrumente den gleichen Ton spielen (unisono) oder wenn viele Menschen gleichzeitig sprechen (Cocktailparty) —, so dass neue Signal-Repräsentationen und leistungsfähigere Modelle notwendig sind. Um die zwei genannten Probleme zu bewältigen, verwenden wir sowohl konventionelle Signalverbeitungsmethoden als auch tiefgehende neuronale Netze (DNN). Wir gehen zunächst auf das Problem der Quellentrennung für Unisono-Instrumentenmischungen ein und untersuchen die speziellen, durch Vibrato ausgelösten, zeitlich-spektralen Modulationen. Um diese Modulationen auszunutzen entwickelten wir eine Methode, die auf Zeitverzerrung basiert und eine Schätzung der Grundfrequenz als zusätzliche Information nutzt. Für Fälle, in denen diese Schätzungen nicht verfügbar sind, stellen wir ein unüberwachtes Modell vor, das inspiriert ist von der Art und Weise, wie Menschen zeitveränderliche Quellen gruppieren (Common Fate). Dieser Beitrag enthält eine neuartige Repräsentation, die die Separierbarkeit für überlappte und modulierte Quellen in Unisono-Mischungen erhöht, aber auch die Trennung in Gesang und Begleitung verbessert, wenn sie in einem DNN-Modell verwendet wird. Im Weiteren beschäftigen wir uns mit der Schätzung der Anzahl von Quellen in einer Mischung, was für reale Szenarien wichtig ist. Unsere Arbeit an der Schätzung der Anzahl war motiviert durch eine Studie, die zeigt, wie wir Menschen diese Aufgabe angehen. Dies hat uns dazu veranlasst, eigene Hörexperimente durchzuführen, die bestätigten, dass Menschen nur in der Lage sind, die Anzahl von bis zu vier Quellen korrekt abzuschätzen. Um nun die Frage zu beantworten, ob Maschinen dies ähnlich gut können, stellen wir eine DNN-Architektur vor, die erlernt hat, die Anzahl der gleichzeitig sprechenden Sprecher zu ermitteln. Die Ergebnisse zeigen Verbesserungen im Vergleich zu anderen Methoden, aber vor allem auch im Vergleich zu menschlichen Hörern. Sowohl bei der Quellentrennung als auch bei der Schätzung der Anzahl an Quellen ist ein Kernbeitrag dieser Arbeit das Konzept der “Modulation”, welches wichtig ist, um die Strategien von Menschen mittels Computern nachzuahmen. Unsere vorgeschlagene Common Fate Transformation ist eine adäquate Darstellung, um die Überlappung von Signalen für die Trennung zugänglich zu machen und eine Inspektion unseres DNN-Zählmodells ergab schließlich, dass sich auch hier modulationsähnliche Merkmale finden lassen

Space Time Exploration of Musical Instruments

Musical instruments are tools used to generate sounds for musical expression. Virtual Reality (VR) and Augmented Reality (AR) musical instruments create sounds that may be spatially disjointed from the instrument controls. Spatial audio processing can be used to position the Extended Reality (XR) musical instruments and their corresponding sounds in the same space. This dissertation investigates novel ways of combining spatial reverb models to improve the naturalness of XR musical instruments. Seven spatial reverb systems, combinations of a shoebox spatial reverb model, a raytracing spatial reverb model, and measured directional room impulse response convolution reverb, were compared in a pilot study. A novel hybrid system of synthetic early reflections and directional room impulse responses was preferred for naturalness when tested over headphones with three instruments created by the author: AR electric guitar, AR drumset, and VR Singing Kite. This research culminated in a concert, Spherical Sound Search, which showcased the preferred hybrid system, the three XR musical instruments, and four re-contextualized spatial audio effects (spatial looping, spatial delay, spatial feedback, and spatial compression). The three pieces in the concert explored different aspects of XR modalities and presented the novel system with spatial audio effects to a larger audience by rendering to an octophonic loudspeaker layout