Efficient Algorithms for Immersive Audio Rendering Enhancement

Il rendering audio immersivo è il processo di creazione di un’esperienza sonora coinvolgente e realistica nello spazio 3D. Nei sistemi audio immersivi, le funzioni di trasferimento relative alla testa (head-related transfer functions, HRTFs) vengono utilizzate per la sintesi binaurale in cuffia poiché esprimono il modo in cui gli esseri umani localizzano una sorgente sonora. Possono essere introdotti algoritmi di interpolazione delle HRTF per ridurre il numero di punti di misura e per creare un movimento del suono affidabile. La riproduzione binaurale può essere eseguita anche dagli altoparlanti. Tuttavia, il coinvolgimento di due o più gli altoparlanti causa il problema del crosstalk. In questo caso, algoritmi di cancellazione del crosstalk (CTC) sono necessari per eliminare i segnali di interferenza indesiderati. In questa tesi, partendo da un'analisi comparativa di metodi di misura delle HRTF, viene proposto un sistema di rendering binaurale basato sull'interpolazione delle HRTF per applicazioni in tempo reale. Il metodo proposto mostra buone prestazioni rispetto a una tecnica di riferimento. L'algoritmo di interpolazione è anche applicato al rendering audio immersivo tramite altoparlanti, aggiungendo un algoritmo di cancellazione del crosstalk fisso, che considera l'ascoltatore in una posizione fissa. Inoltre, un sistema di cancellazione crosstalk adattivo, che include il tracciamento della testa dell'ascoltatore, è analizzato e implementato in tempo reale. Il CTC adattivo implementa una struttura in sottobande e risultati sperimentali dimostrano che un maggiore numero di bande migliora le prestazioni in termini di errore totale e tasso di convergenza. Il sistema di riproduzione e le caratteristiche dell'ambiente di ascolto possono influenzare le prestazioni a causa della loro risposta in frequenza non ideale. L'equalizzazione viene utilizzata per livellare le varie parti dello spettro di frequenze che compongono un segnale audio al fine di ottenere le caratteristiche sonore desiderate. L'equalizzazione può essere manuale, come nel caso dell'equalizzazione grafica, dove il guadagno di ogni banda di frequenza può essere modificato dall'utente, o automatica, la curva di equalizzazione è calcolata automaticamente dopo la misurazione della risposta impulsiva della stanza. L'equalizzazione della risposta ambientale può essere applicata anche ai sistemi multicanale, che utilizzano due o più altoparlanti e la zona di equalizzazione può essere ampliata misurando le risposte impulsive in diversi punti della zona di ascolto. In questa tesi, GEQ efficienti e un sistema adattativo di equalizzazione d'ambiente. In particolare, sono proposti e approfonditi tre equalizzatori grafici a basso costo computazionale e a fase lineare e quasi lineare. Gli esperimenti confermano l'efficacia degli equalizzatori proposti in termini di accuratezza, complessità computazionale e latenza. Successivamente, una struttura adattativa in sottobande è introdotta per lo sviluppo di un sistema di equalizzazione d'ambiente multicanale. I risultati sperimentali verificano l'efficienza dell'approccio in sottobande rispetto al caso a banda singola. Infine, viene presentata una rete crossover a fase lineare per sistemi multicanale, mostrando ottimi risultati in termini di risposta in ampiezza, bande di transizione, risposta polare e risposta in fase. I sistemi di controllo attivo del rumore (ANC) possono essere progettati per ridurre gli effetti dell'inquinamento acustico e possono essere utilizzati contemporaneamente a un sistema audio immersivo. L'ANC funziona creando un'onda sonora in opposizione di fase rispetto all'onda sonora in arrivo. Il livello sonoro complessivo viene così ridotto grazie all'interferenza distruttiva. Infine, questa tesi presenta un sistema ANC utilizzato per la riduzione del rumore. L’approccio proposto implementa una stima online del percorso secondario e si basa su filtri adattativi in sottobande applicati alla stima del percorso primario che mirano a migliorare le prestazioni dell’intero sistema. La struttura proposta garantisce un tasso di convergenza migliore rispetto all'algoritmo di riferimento.Immersive audio rendering is the process of creating an engaging and realistic sound experience in 3D space. In immersive audio systems, the head-related transfer functions (HRTFs) are used for binaural synthesis over headphones since they express how humans localize a sound source. HRTF interpolation algorithms can be introduced for reducing the number of measurement points and creating a reliable sound movement. Binaural reproduction can be also performed by loudspeakers. However, the involvement of two or more loudspeakers causes the problem of crosstalk. In this case, crosstalk cancellation (CTC) algorithms are needed to delete unwanted interference signals. In this thesis, starting from a comparative analysis of HRTF measurement techniques, a binaural rendering system based on HRTF interpolation is proposed and evaluated for real-time applications. The proposed method shows good performance in comparison with a reference technique. The interpolation algorithm is also applied for immersive audio rendering over loudspeakers, by adding a fixed crosstalk cancellation algorithm, which assumes that the listener is in a fixed position. In addition, an adaptive crosstalk cancellation system, which includes the tracking of the listener's head, is analyzed and a real-time implementation is presented. The adaptive CTC implements a subband structure and experimental results prove that a higher number of bands improves the performance in terms of total error and convergence rate. The reproduction system and the characteristics of the listening room may affect the performance due to their non-ideal frequency response. Audio equalization is used to adjust the balance of different audio frequencies in order to achieve desired sound characteristics. The equalization can be manual, such as in the case of graphic equalization, where the gain of each frequency band can be modified by the user, or automatic, where the equalization curve is automatically calculated after the room impulse response measurement. The room response equalization can be also applied to multichannel systems, which employ two or more loudspeakers, and the equalization zone can be enlarged by measuring the impulse responses in different points of the listening zone. In this thesis, efficient graphic equalizers (GEQs), and an adaptive room response equalization system are presented. In particular, three low-complexity linear- and quasi-linear-phase graphic equalizers are proposed and deeply examined. Experiments confirm the effectiveness of the proposed GEQs in terms of accuracy, computational complexity, and latency. Successively, a subband adaptive structure is introduced for the development of a multichannel and multiple positions room response equalizer. Experimental results verify the effectiveness of the subband approach in comparison with the single-band case. Finally, a linear-phase crossover network is presented for multichannel systems, showing great results in terms of magnitude flatness, cutoff rates, polar diagram, and phase response. Active noise control (ANC) systems can be designed to reduce the effects of noise pollution and can be used simultaneously with an immersive audio system. The ANC works by creating a sound wave that has an opposite phase with respect to the sound wave of the unwanted noise. The additional sound wave creates destructive interference, which reduces the overall sound level. Finally, this thesis presents an ANC system used for noise reduction. The proposed approach implements an online secondary path estimation and is based on cross-update adaptive filters applied to the primary path estimation that aim at improving the performance of the whole system. The proposed structure allows for a better convergence rate in comparison with a reference algorithm

Surround by Sound: A Review of Spatial Audio Recording and Reproduction

In this article, a systematic overview of various recording and reproduction techniques for spatial audio is presented. While binaural recording and rendering is designed to resemble the human two-ear auditory system and reproduce sounds specifically for a listener’s two ears, soundfield recording and reproduction using a large number of microphones and loudspeakers replicate an acoustic scene within a region. These two fundamentally different types of techniques are discussed in the paper. A recent popular area, multi-zone reproduction, is also briefly reviewed in the paper. The paper is concluded with a discussion of the current state of the field and open problemsThe authors acknowledge National Natural Science Foundation of China (NSFC) No. 61671380 and Australian Research Council Discovery Scheme DE 150100363

Inverting non-minimum phase FIR transfer functions

The approximate solution to the inverse of a non-minimum phase finite-impulse response (FIR) transfer function is given. Such a problem arises in many areas which include room-acoustics and channel equalisation. Unlike previous work the method does not require multiple sensors or the use of all-pass transfer function networks

The SONICOM HRTF dataset

Immersive audio technologies, ranging from rendering spatialized sounds accurately to efficient room simulations, are vital to the success of augmented and virtual realities. To produce realistic sounds through headphones, the human body and head must both be taken into account. However, the measurement of the influence of the external human morphology on the sounds incoming to the ears, which is often referred to as head-related transfer function (HRTF), is expensive and time-consuming. Several datasets have been created over the years to help researcherswork on immersive audio; nevertheless, the number of individuals involved and amount of data collected is often insufficient for modern machine-learning approaches. Here, the SONICOM HRTF dataset is introduced to facilitate reproducible research in immersive audio. This dataset contains the HRTF of 120 subjects, as well as headphone transfer functions; 3D scans of ears, heads, and torsos; and depth pictures at different angles around subjects' heads

Proceedings of the EAA Spatial Audio Signal Processing symposium: SASP 2019

International audienc

Optimization and improvements in spatial sound reproduction systems through perceptual considerations

[ES] La reproducción de las propiedades espaciales del sonido es una cuestión cada vez más importante en muchas aplicaciones inmersivas emergentes. Ya sea en la reproducción de contenido audiovisual en entornos domésticos o en cines, en sistemas de videoconferencia inmersiva o en sistemas de realidad virtual o aumentada, el sonido espacial es crucial para una sensación de inmersión realista. La audición, más allá de la física del sonido, es un fenómeno perceptual influenciado por procesos cognitivos. El objetivo de esta tesis es contribuir con nuevos métodos y conocimiento a la optimización y simplificación de los sistemas de sonido espacial, desde un enfoque perceptual de la experiencia auditiva. Este trabajo trata en una primera parte algunos aspectos particulares relacionados con la reproducción espacial binaural del sonido, como son la escucha con auriculares y la personalización de la Función de Transferencia Relacionada con la Cabeza (Head Related Transfer Function - HRTF). Se ha realizado un estudio sobre la influencia de los auriculares en la percepción de la impresión espacial y la calidad, con especial atención a los efectos de la ecualización y la consiguiente distorsión no lineal. Con respecto a la individualización de la HRTF se presenta una implementación completa de un sistema de medida de HRTF y se introduce un nuevo método para la medida de HRTF en salas no anecoicas. Además, se han realizado dos experimentos diferentes y complementarios que han dado como resultado dos herramientas que pueden ser utilizadas en procesos de individualización de la HRTF, un modelo paramétrico del módulo de la HRTF y un ajuste por escalado de la Diferencia de Tiempo Interaural (Interaural Time Difference - ITD). En una segunda parte sobre reproducción con altavoces, se han evaluado distintas técnicas como la Síntesis de Campo de Ondas (Wave-Field Synthesis - WFS) o la panoramización por amplitud. Con experimentos perceptuales se han estudiado la capacidad de estos sistemas para producir sensación de distancia y la agudeza espacial con la que podemos percibir las fuentes sonoras si se dividen espectralmente y se reproducen en diferentes posiciones. Las aportaciones de esta investigación pretenden hacer más accesibles estas tecnologías al público en general, dada la demanda de experiencias y dispositivos audiovisuales que proporcionen mayor inmersión.[CA] La reproducció de les propietats espacials del so és una qüestió cada vegada més important en moltes aplicacions immersives emergents. Ja siga en la reproducció de contingut audiovisual en entorns domèstics o en cines, en sistemes de videoconferència immersius o en sistemes de realitat virtual o augmentada, el so espacial és crucial per a una sensació d'immersió realista. L'audició, més enllà de la física del so, és un fenomen perceptual influenciat per processos cognitius. L'objectiu d'aquesta tesi és contribuir a l'optimització i simplificació dels sistemes de so espacial amb nous mètodes i coneixement, des d'un criteri perceptual de l'experiència auditiva. Aquest treball tracta, en una primera part, alguns aspectes particulars relacionats amb la reproducció espacial binaural del so, com són l'audició amb auriculars i la personalització de la Funció de Transferència Relacionada amb el Cap (Head Related Transfer Function - HRTF). S'ha realitzat un estudi relacionat amb la influència dels auriculars en la percepció de la impressió espacial i la qualitat, dedicant especial atenció als efectes de l'equalització i la consegüent distorsió no lineal. Respecte a la individualització de la HRTF, es presenta una implementació completa d'un sistema de mesura de HRTF i s'inclou un nou mètode per a la mesura de HRTF en sales no anecoiques. A mès, s'han realitzat dos experiments diferents i complementaris que han donat com a resultat dues eines que poden ser utilitzades en processos d'individualització de la HRTF, un model paramètric del mòdul de la HRTF i un ajustament per escala de la Diferencià del Temps Interaural (Interaural Time Difference - ITD). En una segona part relacionada amb la reproducció amb altaveus, s'han avaluat distintes tècniques com la Síntesi de Camp d'Ones (Wave-Field Synthesis - WFS) o la panoramització per amplitud. Amb experiments perceptuals, s'ha estudiat la capacitat d'aquests sistemes per a produir una sensació de distància i l'agudesa espacial amb que podem percebre les fonts sonores, si es divideixen espectralment i es reprodueixen en diferents posicions. Les aportacions d'aquesta investigació volen fer més accessibles aquestes tecnologies al públic en general, degut a la demanda d'experiències i dispositius audiovisuals que proporcionen major immersió.[EN] The reproduction of the spatial properties of sound is an increasingly important concern in many emerging immersive applications. Whether it is the reproduction of audiovisual content in home environments or in cinemas, immersive video conferencing systems or virtual or augmented reality systems, spatial sound is crucial for a realistic sense of immersion. Hearing, beyond the physics of sound, is a perceptual phenomenon influenced by cognitive processes. The objective of this thesis is to contribute with new methods and knowledge to the optimization and simplification of spatial sound systems, from a perceptual approach to the hearing experience. This dissertation deals in a first part with some particular aspects related to the binaural spatial reproduction of sound, such as listening with headphones and the customization of the Head Related Transfer Function (HRTF). A study has been carried out on the influence of headphones on the perception of spatial impression and quality, with particular attention to the effects of equalization and subsequent non-linear distortion. With regard to the individualization of the HRTF a complete implementation of a HRTF measurement system is presented, and a new method for the measurement of HRTF in non-anechoic conditions is introduced. In addition, two different and complementary experiments have been carried out resulting in two tools that can be used in HRTF individualization processes, a parametric model of the HRTF magnitude and an Interaural Time Difference (ITD) scaling adjustment. In a second part concerning loudspeaker reproduction, different techniques such as Wave-Field Synthesis (WFS) or amplitude panning have been evaluated. With perceptual experiments it has been studied the capacity of these systems to produce a sensation of distance, and the spatial acuity with which we can perceive the sound sources if they are spectrally split and reproduced in different positions. The contributions of this research are intended to make these technologies more accessible to the general public, given the demand for audiovisual experiences and devices with increasing immersion.Gutiérrez Parera, P. (2020). Optimization and improvements in spatial sound reproduction systems through perceptual considerations [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/142696TESI

Headroom and precision requirements of fixed point audio processing in different data domains for real world content

En las técnicas de procesado de audio actuales, la señal temporal de audio se transforma en otros dominios de datos para su procesado, usando, por ejemplo, la FFT, MDCT o la QMF compleja. Cada uno de estos dominios tiene sus propiedades diferentes, lo que implica tener que satisfacer diferentes requisitos para gestionar correctamente la distorsión y el ruido en cada sistema. El objetivo del trabajo propuesto es profundizar sobre estos requisitos cuando codificamos señales reales usando técnicas de sonido inmersivo o sonido multicanal, y poder proponer pautas para conseguir el adecuado margen dinámico de protección frente a la distorsión (headroom) y la precisión necesaria en los sistemas de codificación de punto fijo. Los procesos que se aplican a las señales que van a estudiarse serán los mismo que se utilizan actualmente en los sistemas Dolby, y que incluyen implementaciones que mejoran la eficiencia tales como el uso de transformaciones de los datos en otros dominios, la reducción del número de canales de audio (como en la transformación y mezcla de las señales para la reproducción en una configuración de un altavoz), y el procesado de la sonoridad de la señal.In modern audio processing technology, the audio signal is converted to various data domains for processing by using, e.g. FFT, MDCT, or Complex QMF transforms. Each domain has different properties resulting in different requirements in headroom and precision to manage distortion and noise in such systems. The aim of the work is to investigate such requirements for immersive and multichannel real-world content and to develop guidelines for headroom and data precision requirements for processing such content. Processing blocks to study are the ones currently used in Dolby systems that can include smart implementations of data domain transforms, reduction of audio channels like rendering to a speaker configuration and down-mixing and signal loudness processing.Carbonell Tena, D. (2018). Estudio del headroom y las necesidades de precisión para el procesado de señales reales de audio en representación de punto fijo usando diferentes dominios. Universitat Politècnica de València. http://hdl.handle.net/10251/98727TFG

Measurement of head-related transfer functions : A review

A head-related transfer function (HRTF) describes an acoustic transfer function between a point sound source in the free-field and a defined position in the listener's ear canal, and plays an essential role in creating immersive virtual acoustic environments (VAEs) reproduced over headphones or loudspeakers. HRTFs are highly individual, and depend on directions and distances (near-field HRTFs). However, the measurement of high-density HRTF datasets is usually time-consuming, especially for human subjects. Over the years, various novel measurement setups and methods have been proposed for the fast acquisition of individual HRTFs while maintaining high measurement accuracy. This review paper provides an overview of various HRTF measurement systems and some insights into trends in individual HRTF measurements