292 research outputs found

    Recognition of Harmonic Sounds in Polyphonic Audio using a Missing Feature Approach: Extended Report

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    A method based on local spectral features and missing feature techniques is proposed for the recognition of harmonic sounds in mixture signals. A mask estimation algorithm is proposed for identifying spectral regions that contain reliable information for each sound source and then bounded marginalization is employed to treat the feature vector elements that are determined as unreliable. The proposed method is tested on musical instrument sounds due to the extensive availability of data but it can be applied on other sounds (i.e. animal sounds, environmental sounds), whenever these are harmonic. In simulations the proposed method clearly outperformed a baseline method for mixture signals

    Automatic Drum Transcription and Source Separation

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    While research has been carried out on automated polyphonic music transcription, to-date the problem of automated polyphonic percussion transcription has not received the same degree of attention. A related problem is that of sound source separation, which attempts to separate a mixture signal into its constituent sources. This thesis focuses on the task of polyphonic percussion transcription and sound source separation of a limited set of drum instruments, namely the drums found in the standard rock/pop drum kit. As there was little previous research on polyphonic percussion transcription a broad review of music information retrieval methods, including previous polyphonic percussion systems, was also carried out to determine if there were any methods which were of potential use in the area of polyphonic drum transcription. Following on from this a review was conducted of general source separation and redundancy reduction techniques, such as Independent Component Analysis and Independent Subspace Analysis, as these techniques have shown potential in separating mixtures of sources. Upon completion of the review it was decided that a combination of the blind separation approach, Independent Subspace Analysis (ISA), with the use of prior knowledge as used in music information retrieval methods, was the best approach to tackling the problem of polyphonic percussion transcription as well as that of sound source separation. A number of new algorithms which combine the use of prior knowledge with the source separation abilities of techniques such as ISA are presented. These include sub-band ISA, Prior Subspace Analysis (PSA), and an automatic modelling and grouping technique which is used in conjunction with PSA to perform polyphonic percussion transcription. These approaches are demonstrated to be effective in the task of polyphonic percussion transcription, and PSA is also demonstrated to be capable of transcribing drums in the presence of pitched instruments

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

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    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

    Automatic music transcription: challenges and future directions

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    Automatic music transcription is considered by many to be a key enabling technology in music signal processing. However, the performance of transcription systems is still significantly below that of a human expert, and accuracies reported in recent years seem to have reached a limit, although the field is still very active. In this paper we analyse limitations of current methods and identify promising directions for future research. Current transcription methods use general purpose models which are unable to capture the rich diversity found in music signals. One way to overcome the limited performance of transcription systems is to tailor algorithms to specific use-cases. Semi-automatic approaches are another way of achieving a more reliable transcription. Also, the wealth of musical scores and corresponding audio data now available are a rich potential source of training data, via forced alignment of audio to scores, but large scale utilisation of such data has yet to be attempted. Other promising approaches include the integration of information from multiple algorithms and different musical aspects

    A computational framework for sound segregation in music signals

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    Tese de doutoramento. Engenharia Electrotécnica e de Computadores. Faculdade de Engenharia. Universidade do Porto. 200

    Contributions to automatic multiple F0 detection in polyphonic music signals

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    Multiple fundamental frequency estimation, or multi-pitch estimation (MPE), is a key problem in automatic music transcription (AMT) and many other related audio processing tasks. Applications of AMT are numerous, ranging from musical genre classification to automatic piano tutoring, and these form a significant part of musical information retrieval tasks. Current AMT systems still perform considerably below human experts, and there is a consensus that the development of an automated system for full transcription of polyphonic music regardless of its complexity is still an open problem. The goal of this work is to propose contributions for the automatic detection of multiple fundamental frequencies in polyphonic music signals. A reference MPE method is chosen to be studied and implemented, and a modification is proposed to improve the performance of the system. Lastly, three refinement strategies are proposed to be incorporated into the modified method, in order to increase the quality of the results. Experimental tests reveal that such refinements improve the overall performance of the system, even if each one performs differently according to signal characteristics.Estimação de múltiplas frequências fundamentais (MPE, do inglês multipitch estimation) é um problema importante na área de transcrição musical automática (TMA) e em muitas outras tarefas relacionadas a processamento de áudio. Aplicações de TMA são diversas, desde classificação de gêneros musicais ao aprendizado automático de piano, as quais consistem em uma parcela significativa de tarefas de extração de informação musical. Métodos atuais de TMA ainda possuem um desempenho consideravelmente ruim quando comparados aos de profissionais da área, e há um consenso que o desenvolvimento de um sistema automatizado para a transcrição completa de música polifônica independentemente de sua complexidade ainda é um problema em aberto. O objetivo deste trabalho é propor contribuições para a detecção automática de múltiplas frequências fundamentais em sinais de música polifônica. Um método de referência para MPEé primeiramente escolhido para ser estudado e implementado, e uma modificação é proposta para melhorar o desempenho do sistema. Por fim, três estratégias de refinamento são propostas para serem incorporadas ao método modificado, com o objetivo de aumentar a qualidade dos resultados. Testes experimentais mostram que tais refinamentos melhoram em média o desempenho do sistema, embora cada um atue de uma maneira diferente de acordo com a natureza dos sinais

    Acoustically Inspired Probabilistic Time-domain Music Transcription and Source Separation.

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    PhD ThesisAutomatic music transcription (AMT) and source separation are important computational tasks, which can help to understand, analyse and process music recordings. The main purpose of AMT is to estimate, from an observed audio recording, a latent symbolic representation of a piece of music (piano-roll). In this sense, in AMT the duration and location of every note played is reconstructed from a mixture recording. The related task of source separation aims to estimate the latent functions or source signals that were mixed together in an audio recording. This task requires not only the duration and location of every event present in the mixture, but also the reconstruction of the waveform of all the individual sounds. Most methods for AMT and source separation rely on the magnitude of time-frequency representations of the analysed recording, i.e., spectrograms, and often arbitrarily discard phase information. On one hand, this decreases the time resolution in AMT. On the other hand, discarding phase information corrupts the reconstruction in source separation, because the phase of each source-spectrogram must be approximated. There is thus a need for models that circumvent phase approximation, while operating at sample-rate resolution. This thesis intends to solve AMT and source separation together from an unified perspective. For this purpose, Bayesian non-parametric signal processing, covariance kernels designed for audio, and scalable variational inference are integrated to form efficient and acoustically-inspired probabilistic models. To circumvent phase approximation while keeping sample-rate resolution, AMT and source separation are addressed from a Bayesian time-domain viewpoint. That is, the posterior distribution over the waveform of each sound event in the mixture is computed directly from the observed data. For this purpose, Gaussian processes (GPs) are used to define priors over the sources/pitches. GPs are probability distributions over functions, and its kernel or covariance determines the properties of the functions sampled from a GP. Finally, the GP priors and the available data (mixture recording) are combined using Bayes' theorem in order to compute the posterior distributions over the sources/pitches. Although the proposed paradigm is elegant, it introduces two main challenges. First, as mentioned before, the kernel of the GP priors determines the properties of each source/pitch function, that is, its smoothness, stationariness, and more importantly its spectrum. Consequently, the proposed model requires the design of flexible kernels, able to learn the rich frequency content and intricate properties of audio sources. To this end, spectral mixture (SM) kernels are studied, and the Mat ern spectral mixture (MSM) kernel is introduced, i.e. a modified version of the SM covariance function. The MSM kernel introduces less strong smoothness, thus it is more suitable for modelling physical processes. Second, the computational complexity of GP inference scales cubically with the number of audio samples. Therefore, the application of GP models to large audio signals becomes intractable. To overcome this limitation, variational inference is used to make the proposed model scalable and suitable for signals in the order of hundreds of thousands of data points. The integration of GP priors, kernels intended for audio, and variational inference could enable AMT and source separation time-domain methods to reconstruct sources and transcribe music in an efficient and informed manner. In addition, AMT and source separation are current challenges, because the spectra of the sources/pitches overlap with each other in intricate ways. Thus, the development of probabilistic models capable of differentiating sources/pitches in the time domain, despite the high similarity between their spectra, opens the possibility to take a step towards solving source separation and automatic music transcription. We demonstrate the utility of our methods using real and synthesized music audio datasets for various types of musical instruments
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