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

A room acoustics measurement system using non-invasive microphone arrays

This thesis summarises research into adaptive room correction for small rooms and pre-recorded material, for example music of films. A measurement system to predict the sound at a remote location within a room, without a microphone at that location was investigated. This would allow the sound within a room to be adaptively manipulated to ensure that all listeners received optimum sound, therefore increasing their enjoyment. The solution presented used small microphone arrays, mounted on the room's walls. A unique geometry and processing system was designed, incorporating three processing stages, temporal, spatial and spectral. The temporal processing identifies individual reflection arrival times from the recorded data. Spatial processing estimates the angles of arrival of the reflections so that the three-dimensional coordinates of the reflections' origin can be calculated. The spectral processing then estimates the frequency response of the reflection. These estimates allow a mathematical model of the room to be calculated, based on the acoustic measurements made in the actual room. The model can then be used to predict the sound at different locations within the room. A simulated model of a room was produced to allow fast development of algorithms. Measurements in real rooms were then conducted and analysed to verify the theoretical models developed and to aid further development of the system. Results from these measurements and simulations, for each processing stage are presented

Flexible binaural resynthesis of room impulse responses for augmented reality research

International audienceA basic building block in audio for Augmented Reality (AR) is the use of virtual sound sources layered on top of any real sources present in an environment. In order to perceive these virtual sources as belonging to the natural scene it is important to match their acoustic parameters to those of a real source with the same characteristics, i.e. radiation properties, sound propagation and head-related impulse response (HRIR). However, it is still unclear to what extent these parameters need to be matched in order to generate plausible scenes in which virtual sound sources blend seamlessly with real sound sources. This contribution presents an auralization framework that allows protyping of augmented reality scenarios from measured multichannel room impulse responses to get a better understanding of the relevance of individual acoustic parameters.A well-established approach for binaural measurement and reproduction of sound scenes is based on capturing binaural room impulse responses (BRIR) using a head and torso simulator (HATS) and convolving these BRIRs dynamically with audio content according to the listener head orientation. However, such measurements are laborious and time consuming, requiring measuring the scene with the HATS in multiple orientations. Additionally, the HATS HRIR is inherently encoded in the BRIRs, making them unsuitable for personalization for different listeners. The approach presented here consists of the resynthesis and dynamic binaural reproduction of multichannel room impulse responses (RIR) using an arbitrary HRIR dataset. Using a compact microphone array, we obtained a pressure RIR and a set of auxiliary RIRs, and we applied the Spatial Decomposition Method (SDM) to estimate the direction-of-arrival (DOA) of the different sound events in the RIR. The DOA information was used to map sound pressure to different locations by means of an HRIR dataset, generating a binaural room impulse response (BRIR) for a specific orientation. By either rotating the DOA or the HRIR data set, BRIRs for any direction may be obtained. Auralizations using SDM are known to whiten the spectrum of late reverberation. Available alternatives such as time-frequency equalization were not feasible in this case, as a different time-frequency filter would be necessary for each direction, resulting in a non-homogeneous equalization of the BRIRs. Instead, the resynthesized BRIRs were decomposed into sub-bands and the decay slope of each sub-band was modified independently to match the reverberation time of the original pressure RIR. In this way we could apply the same reverberation correction factor to all BRIRs. In addition, we used a direction independent equalization to correct for timbral effects of equipment, HRIR, and signal processing. Real-time reproduction was achieved by means of a custom Max/MSP patch, in which the direct sound, early reflections and late reverberation were convolved separately to allow real-time changes in the time-energy properties of the BRIRs. The mixing time of the reproduced BRIRs is configurable and a single direction independent reverberation tail is used. To evaluate the quality of the resynthesis method in a real room, we conducted both objective and perceptual comparisons for a variety of source positions. The objective analysis was performed by comparing real measurements of a KEMAR mannequin with the resynthesis at the same receiver location using a simulated KEMAR HRIR. Typical room acoustic parameters of both real and resynthsized acoustics were found to be in good agreement. The perceptual validation consisted of a comparison of a loudspeaker and its resynthesized counterpart. Non-occluding headphones with individual equalization were used to ensure that listeners were able to simultaneously listen to the real and the virtual samples. Subjects were allowed to listen to the sounds for as long as they desired and freely switch between the real and virtual stimuli in real time. The integration of an Optitrack motion tracking system allowed us to present world-locked audio, accounting for head rotations.We present here the results of this listening test (N = 14) with three sections: discrimination, identification, and qualitative ratings. Preliminary analysis revealed that in these conditions listeners were generally able to discriminate between real and virtual sources and were able to consistently identify which of the presented sources was real and which was virtual. The qualitative analysis revealed that timbral differences are the most prominent cues for discrimination and identification, while spatial cues are well preserved. All the listeners reported good externalization of the binaural audio.Future work includes extending the presented validation to more environments, as well as implementing tools to arbitrarily modify BRIRs in the spatial, temporal, and frequency domains in order to study the perceptual requirements of room acoustics reproduction in AR

Messaging in mobile augmented reality audio

Monen käyttäjän välinen asynkroninen viestintä tapahtuu tyypillisesti tekstiä käyttäen. Mobiileissa käyttötilanteissa tekstinsyöttö voi kuitenkin olla hidasta ja vaivalloista. Sekä viestien kirjoittaminen että lukeminen vaatii huomion keskittämistä laitteen näyttöön. Tässä työssä kehitettiin viestintäsovellus, jossa tekstin sijaan käytetään puhetta lyhyiden viestien jakamiseen ryhmien jäsenten välillä. Näitä viestejä voidaan kuunnella heti niiden saapuessa tai niitä voi selata ja kuunnella myöhemmin. Sovellusta on tarkoitettu käytettävän mobiilin lisätyn äänitodellisuuden alustan kanssa, mikä mahdollistaa lähes häiriintymättömän ympäristön havaitsemisen samalla kun kommunikoi ääniviestien avulla. Pieni ryhmä käyttäjiä testasi sovellusta pöytätietokoneilla ja kannettavilla tietokoneilla. Yksi isoimmista eduista tekstipohjaiseen viestintään verrattuna todettiin olevan puheen mukana välittyvä ylimääräinen tieto verrattuna samaan kirjoitettuun viestiin, puheviestinnän ollessa paljon ilmeikkäämpää. Huonoja puolia verrattuna tekstipohjaiseen viestintään olivat hankaluus selata vanhojen viestien läpi sekä vaikeus osallistua useampaan keskusteluun samaan aikaan.Asynchronous multi-user communication is typically done using text. In the context of mobile use text input can, however, be slow and cumbersome, and attention on the display of the device is required both when writing and reading messages. A messaging application was developed to test the concept of sharing short messages between members of groups using recorded speech rather than text. These messages can be listened to as they arrive, or browsed through and listened to later. The application is intended to be used on a mobile augmented reality audio platform, allowing almost undisturbed perception of and interaction with the surrounding environment while communicating using audio messages. A small group of users tested the application on desktop and laptop computers. The users found one of the biggest advantages over text-based communication to be the additional information associated with a spoken message, being much more expressive than the same written message. Compared with text chats, the users thought it was difficult to quickly browse through old messages and confusing to participate in several discussions at the same time

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

Characterization of the head-related transfer function using chirp and maximum length sequence excitation signals

Both chirp (or sweep) and maximum length sequence (MLS) excitation signals are used to obtain measurements of the head-related transfer function (HRTF) for the Knowles electronic manikin for acoustic research (KEMAR), resulting in remarkably low relative error between the two measurement methods. Due to the low relative error, it is asserted that applying both methods in experiments involving system characterization can be used to confirm the accuracy of individual measurements and reduce the likelihood that erroneous measurements go unnoticed. An MLS burst-augmented excitation signal is presented to compensate for unpredictable delay added in sound-acquisition software that lacks the capability of synchronized record/playback. The HRTF measurements obtained with an external probe microphone are compared with those obtained with the standard in-ear microphones that the KEMAR is equipped with, to reveal that care should be taken to select a probe microphone that has a low susceptibility to multipath interference. An experiment is performed to investigate the feasibility of using two pre-filtered chirp responses to obtain the relative transfer function between the two locations of measurement