Real-time Sound Source Separation For Music Applications

Sound source separation refers to the task of extracting individual sound sources from some number of mixtures of those sound sources. In this thesis, a novel sound source separation algorithm for musical applications is presented. It leverages the fact that the vast majority of commercially recorded music since the 1950s has been mixed down for two channel reproduction, more commonly known as stereo. The algorithm presented in Chapter 3 in this thesis requires no prior knowledge or learning and performs the task of separation based purely on azimuth discrimination within the stereo field. The algorithm exploits the use of the pan pot as a means to achieve image localisation within stereophonic recordings. As such, only an interaural intensity difference exists between left and right channels for a single source. We use gain scaling and phase cancellation techniques to expose frequency dependent nulls across the azimuth domain, from which source separation and resynthesis is carried out. The algorithm is demonstrated to be state of the art in the field of sound source separation but also to be a useful pre-process to other tasks such as music segmentation and surround sound upmixing

Binaural to multichannel audio upmix

Audion tallennus- ja toistolaitteiden valikoiman kasvaessa on tärkeää, että kaikenlaisilla välineillä tallennettua sekä syntetisoitua audiota voidaan muokata toistettavaksi kaikenlaisilla äänentoistojärjestelmillä. Tässä diplomityössä esitellään menetelmä, jolla binauraalinen audiosignaali voidaan muokata toistettavaksi monikanavaisella kaiutinjärjestelmällä säilyttäen signaalin suuntainformaation. Tällaiselle muokkausmenetelmälle on tarvetta esimerkiksi etäläsnäolosovelluksissa keinona toistaa binauraalinen äänitys monikanavaisella kaiutinjärjestelmällä. Menetelmässä binauraalisesta signaalista estimoidaan ensin äänilähteiden suunnat käyttäen hyväksi korvien välistä aikaeroa. Signaali muokataan monofoniseksi, ja tulosuunnan estimoinnin antama tieto tallennetaan sivuinformaationa. Monofoninen signaali muokataan sen jälkeen halutulle monikanavaiselle kaiutinjärjestelmälle panoroimalla se tallennetun suuntainformaation mukaisesti. Käytännössä menetelmä siis muuntaa korvien välisen aikaeron kanavien väliseksi voimakkuuseroksi. Menetelmässä käytetään ja yhdistellään olemassaolevia tekniikoita tulosuunnan estimoinnille sekä panoroinnille. Menetelmää testattiin vapaamuotoisessa kuuntelukokeessa, sekä lisäämällä ääninäytteisiin binauraalista taustamelua ennen muokkausta ja arvioimalla sen vaikutusta muokatun signaalin laatuun. Menetelmän todettiin toimivan kelvollisesti sekä suuntainformaation säilymisen, että äänen laadun suhteen, ottaen huomioon, että sen kehitystyö on vasta aluillaan.The increasing diversity of popular audio recording and playback systems gives reasons to ensure that recordings made with any equipment, as well as any synthesised audio, can be reproduced for playback with all types of devices. In this thesis, a method is introduced for upmixing binaural audio into a multichannel format while preserving the correct spatial sensation. This type of upmix is required when a binaural recording is desired to be spatially reproduced for playback over a multichannel loudspeaker setup, a scenario typical for e.g. the prospective telepresence appliances. In the upmix method the sound source directions are estimated from the binaural signal by using the interaural time difference. The signal is then downmixed into a monophonic format and the data given by the azimuth estimation is stored as side-information. The monophonic signal is upmixed for an arbitrary multichannel loudspeaker setup by panning it on the basis of the spatial side-information. The method, thus effectively converting interaural time differences into interchannel level differences, employs and conjoins existing techniques for azimuth estimation and discrete panning. The method was tested in an informal listening test, as well as by adding spatial background noise into the samples before upmixing and evaluating its influence on the sound quality of the upmixed samples. The method was found to perform acceptably well in maintaining both the spatiality as well as the sound quality, regarding that much development work remains to be done

Investigations into the Perception of Vertical Interchannel Decorrelation in 3D Surround Sound Reproduction

The use of three-dimensional (3D) surround sound systems has seen a rapid increase over recent years. In two-dimensional (2D) loudspeaker formats (i.e. two-channel stereophony (stereo) and 5.1 Surround), horizontal interchannel decorrelation is a well-established technique for controlling the horizontal spread of a phantom image. Use of interchannel decorrelation can also be found within established two-to-five channel upmixing methods (stereo to 5.1). More recently, proprietary algorithms have been developed that perform 2D-to-3D upmixing, which presumably make use of interchannel decorrelation as well; however, it is not currently known how interchannel decorrelation is perceived in the vertical domain. From this, it is considered that formal investigations into the perception of vertical interchannel decorrelation are necessary. Findings from such experiments may contribute to the improved control of a sound source within 3D surround systems (i.e. the vertical spread), in addition to aiding the optimisation of 2D-to-3D upmixing algorithms. The current thesis presents a series of experiments that systematically assess vertical interchannel decorrelation under various conditions. Firstly, a comparison is made between horizontal and vertical interchannel decorrelation, where it is found that vertical decorrelation is weaker than horizontal decorrelation. However, it is also seen that vertical decorrelation can generate a significant increase of vertical image spread (VIS) for some conditions. Following this, vertical decorrelation is assessed for octave-band pink noise stimuli at various azimuth angles to the listener. The results demonstrate that vertical decorrelation is dependent on both frequency and presentation angle – a general relationship between the interchannel cross-correlation (ICC) and VIS is observed for the 500 Hz octave-band and above, and strongest for the 8 kHz octave-band. Objective analysis of these stimuli signals determined that spectral changes at higher frequencies appear to be associated with VIS perception – at 0° azimuth, the 8 and 16 kHz octave-bands demonstrate potential spectral cues, at ±30°, similar cues are seen in the 4, 8 and 16 kHz bands, and from ±110°, cues are featured in the 2, 4, 8 and 16 kHz bands. In the case of the 8 kHz octave-band, it seems that vertical decorrelation causes a ‘filling in’ of vertical localisation notch cues, potentially resulting in ambiguous perception of vertical extent. In contrast, the objective analysis suggests that VIS perception of the 500 Hz and 1 kHz bands may have been related to early reflections in the listening room. From the experiments above, it is demonstrated that the perception of VIS from vertical interchannel decorrelation is frequency-dependent, with high frequencies playing a particularly important role. A following experiment explores the vertical decorrelation of high frequencies only, where it is seen that decorrelation of the 500 Hz octave-band and above produces a similar perception of VIS to broadband decorrelation, whilst improving tonal quality. The results also indicate that decorrelation of the 8 kHz octave-band and above alone can significantly increase VIS, provided the source signal has sufficient high frequency energy. The final experimental chapter of the present thesis aims to provide a controlled assessment of 2D-to-3D upmixing, taking into account the findings of the previous experiments. In general, 2D-to-3D upmixing by vertical interchannel decorrelation had little impact on listener envelopment (LEV), when compared against a level-matched 2D 5.1 reference. Furthermore, amplitude-based decorrelation appeared to be marginally more effective, and ‘high-pass decorrelation’ resulted in slightly better tonal quality for sources that featured greater low frequency energy

Subjective evaluation and electroacoustic theoretical validation of a new approach to audio upmixing

Audio signal processing systems for converting two-channel (stereo) recordings to four or five channels are increasingly relevant. These audio upmixers can be used with conventional stereo sound recordings and reproduced with multichannel home theatre or automotive loudspeaker audio systems to create a more engaging and natural-sounding listening experience. This dissertation discusses existing approaches to audio upmixing for recordings of musical performances and presents specific design criteria for a system to enhance spatial sound quality. A new upmixing system is proposed and evaluated according to these criteria and a theoretical model for its behavior is validated using empirical measurements.The new system removes short-term correlated components from two electronic audio signals using a pair of adaptive filters, updated according to a frequency domain implementation of the normalized-least-means-square algorithm. The major difference of the new system with all extant audio upmixers is that unsupervised time-alignment of the input signals (typically, by up to +/-10 ms) as a function of frequency (typically, using a 1024-band equalizer) is accomplished due to the non-minimum phase adaptive filter. Two new signals are created from the weighted difference of the inputs, and are then radiated with two loudspeakers behind the listener. According to the consensus in the literature on the effect of interaural correlation on auditory image formation, the self-orthogonalizing properties of the algorithm ensure minimal distortion of the frontal source imagery and natural-sounding, enveloping reverberance (ambiance) imagery.Performance evaluation of the new upmix system was accomplished in two ways: Firstly, using empirical electroacoustic measurements which validate a theoretical model of the system; and secondly, with formal listening tests which investigated auditory spatial imagery with a graphical mapping tool and a preference experiment. Both electroacoustic and subjective methods investigated system performance with a variety of test stimuli for solo musical performances reproduced using a loudspeaker in an orchestral concert-hall and recorded using different microphone techniques.The objective and subjective evaluations combined with a comparative study with two commercial systems demonstrate that the proposed system provides a new, computationally practical, high sound quality solution to upmixing

SIGNAL TRANSFORMATIONS FOR IMPROVING INFORMATION REPRESENTATION, FEATURE EXTRACTION AND SOURCE SEPARATION

Questa tesi riguarda nuovi metodi di rappresentazione del segnale nel dominio tempo-frequenza, tali da mostrare le informazioni ricercate come dimensioni esplicite di un nuovo spazio. In particolare due trasformate sono introdotte: lo Spazio di Miscelazione Bivariato (Bivariate Mixture Space) e il Campo della Struttura Spettro-Temporale (Spectro-Temporal Structure-Field). La prima trasformata mira a evidenziare le componenti latenti di un segnale bivariato basandosi sul comportamento di ogni componente frequenziale (ad esempio a fini di separazione delle sorgenti); la seconda trasformata mira invece all'incapsulamento di informazioni relative al vicinato di un punto in R^2 in un vettore associato al punto stesso, tale da descrivere alcune propriet\ue0 topologiche della funzione di partenza. Nel dominio dell'elaborazione digitale del segnale audio, il Bivariate Mixture Space pu\uf2 essere interpretato come un modo di investigare lo spazio stereofonico per operazioni di separazione delle sorgenti o di estrazione di informazioni, mentre lo Spectro-Temporal Structure-Field pu\uf2 essere usato per ispezionare lo spazio spettro-temporale (segregare suoni percussivi da suoni intonati o tracciae modulazioni di frequenza). Queste trasformate sono studiate e testate anche in relazione allo stato del'arte in campi come la separazione delle sorgenti, l'estrazione di informazioni e la visualizzazione dei dati. Nel campo dell'informatica applicata al suono, queste tecniche mirano al miglioramento della rappresentazione del segnale nel dominio tempo-frequenza, in modo tale da rendere possibile l'esplorazione dello spettro anche in spazi alternativi, quali il panorama stereofonico o una dimensione virtuale che separa gli aspetti percussivi da quelli intonati.This thesis is about new methods of signal representation in time-frequency domain, so that required information is rendered as explicit dimensions in a new space. In particular two transformations are presented: Bivariate Mixture Space and Spectro-Temporal Structure-Field. The former transform aims at highlighting latent components of a bivariate signal based on the behaviour of each frequency base (e.g. for source separation purposes), whereas the latter aims at folding neighbourhood information of each point of a R^2 function into a vector, so as to describe some topological properties of the function. In the audio signal processing domain, the Bivariate Mixture Space can be interpreted as a way to investigate the stereophonic space for source separation and Music Information Retrieval tasks, whereas the Spectro-Temporal Structure-Field can be used to inspect spectro-temporal dimension (segregate pitched vs. percussive sounds or track pitch modulations). These transformations are investigated and tested against state-of-the-art techniques in fields such as source separation, information retrieval and data visualization. In the field of sound and music computing, these techniques aim at improving the frequency domain representation of signals such that the exploration of the spectrum can be achieved also in alternative spaces like the stereophonic panorama or a virtual percussive vs. pitched dimension

Iterative Separation of Note Events from Single-Channel Polyphonic Recordings

This thesis is concerned with the separation of audio sources from single-channel polyphonic musical recordings using the iterative estimation and separation of note events. Each event is defined as a section of audio containing largely harmonic energy identified as coming from a single sound source. Multiple events can be clustered to form separated sources. This solution is a model-based algorithm that can be applied to a large variety of audio recordings without requiring previous training stages. The proposed system embraces two principal stages. The first one considers the iterative detection and separation of note events from within the input mixture. In every iteration, the pitch trajectory of the predominant note event is automatically selected from an array of fundamental frequency estimates and used to guide the separation of the event's spectral content using two different methods: time-frequency masking and time-domain subtraction. A residual signal is then generated and used as the input mixture for the next iteration. After convergence, the second stage considers the clustering of all detected note events into individual audio sources. Performance evaluation is carried out at three different levels. Firstly, the accuracy of the note-event-based multipitch estimator is compared with that of the baseline algorithm used in every iteration to generate the initial set of pitch estimates. Secondly, the performance of the semi-supervised source separation process is compared with that of another semi-automatic algorithm. Finally, a listening test is conducted to assess the audio quality and naturalness of the separated sources when they are used to create stereo mixes from monaural recordings. Future directions for this research focus on the application of the proposed system to other music-related tasks. Also, a preliminary optimisation-based approach is presented as an alternative method for the separation of overlapping partials, and as a high resolution time-frequency representation for digital signals

Scanning Spaces: Paradigms for Spatial Sonification and Synthesis

In 1962 Karlheinz Stockhausen’s “Concept of Unity in Electronic Music” introduced a connection between the parameters of intensity, duration, pitch, and timbre using an accelerating pulse train. In 1973 John Chowning discovered that complex audio spectra could be synthesized by increasing vibrato rates past 20Hz. In both cases the notion of acceleration to produce timbre was critical to discovery. Although both composers also utilized sound spatialization in their works, spatial parameters were not unified with their synthesis techniques. This dissertation examines software studies and multimedia works involving the use of spatial and visual data to produce complex sound spectra. The culmination of these experiments, Spatial Modulation Synthesis, is introduced as a novel, mathematical control paradigm for audio-visual synthesis, providing unified control of spatialization, timbre, and visual form using high-speed sound trajectories.The unique, visual sonification and spatialization rendering paradigms of this disser- tation necessitated the development of an original audio-sample-rate graphics rendering implementation, which, unlike typical multimedia frameworks, provides an exchange of audio-visual data without downsampling or interpolation

Multichannel Music Separation with Deep Neural Networks

International audienceThis article addresses the problem of multichannel music separation. We propose a framework where the source spectra are estimated using deep neural networks and combined with spatial covariance matrices to encode the source spatial characteristics. The parameters are estimated in an iterative expectation-maximization fashion and used to derive a multichannel Wiener filter. We evaluate the proposed framework for the task of music separation on a large dataset. Experimental results show that the method we describe performs consistently well in separating singing voice and other instruments from realistic musical mixtures

Dislocations in sound design for 3-d films: sound design and the 3-d cinematic experience

Since the success of James Cameron’s Avatar (2009),1 the feature film industry has embraced 3-D feature film technology. With 3-D films now setting a new benchmark for contemporary cinemagoers, the primary focus is directed towards these new stunning visuals. Sound is often neglected until the final filmmaking process as the visuals are taking up much of the film budget. 3-D has changed the relationship between the imagery and the accompanying soundtrack, losing aspects of the cohesive union compared with 2-D film. Having designed sound effects on Australia’s first digital animated 3-D film, Legend of the Guardians: The Owls of Ga’Hoole (2010),2 and several internationally released 3-D films since, it became apparent to me that the visuals are evolving technologically and artistically at a rate far greater than the soundtrack. This is creating a dislocation between the image and the soundtrack. Although cinema sound technology companies are trialing and releasing new ‘immersive’ technologies, they are not necessarily addressing the spatial relationship between the images and soundtracks of 3-D digital films. Through first hand experience, I question many of the working methodologies currently employed within the production and creation of the soundtrack for 3-D films. There is limited documentation on sound design within the 3-D feature film context, and as such, there are no rules or standards associated with this new practice. Sound designers and film sound mixers are continuing to use previous 2-D work practices in cinema sound, with limited and cautious experimentation of spatial sound design for 3-D. Although emerging technologies are capable of providing a superior and ‘more immersive’ soundtrack than previous formats, this does not necessarily mean that they provide an ideal solution for 3-D film. Indeed the film industry and cinema managers are showing some resistance in adopting these technologies, despite the push from technology vendors. Through practice-led research, I propose to research and question the following:Does the contemporary soundtrack suit 3-D films? ; Has sound technology used in 2-D film changed with the introduction of 3-D film? If it has, is this technology an ideal solution, or are further technical developments needed to allow greater creativity and cohesiveness of 3-D film sound design? ; How might industry practices need to develop in order to accommodate the increased dimension and image depth of 3-D visuals? ; Does a language exist to describe spatial sound design in 3-D cinema? ; What is the audience awareness of emerging film technologies? And what does this mean for filmmakers and the cinema? ; Looking beyond contemporary cinema practices, is there an alternative approach to creating a soundtrack that better represents the accompanying 3-D imagery