Improving particular components of the audio signal chain: optimising listening in the control room

In the field of audio engineering there is a constant need for optimising the listening situation. Listening to, judging and finally optimising the recorded material are essential tasks of audio engineers. The author of this contextual statement has been working in the field of audio engineering since 1993. In addition, various research projects have been undertaken in this field. A selection of three research areas and their published outputs are presented in this contextual statement: Research Area 1: Improving acoustic modules to increase efficiency in the acoustical treatment of control rooms Research Area 2: Measuring time alignment errors, testing their impact on the listening experience and providing solutions for time alignment of loudspeakers Research Area 3: Using equalisation for correcting and shaping a loudspeaker's frequency response These research areas relate to a consistent listening 'defect' that leads to a blurred and broader sound image. Measures to overcome these defects are presented and proven to be effective by built prototypes and/or products. The results of the research are published in articles and books and can be experienced in the form of hardware systems such as acoustic modules or modified loudspeakers

Modelling of the First-Order Time-Varying Filters with Periodically Variable Coefficients

The article is devoted to modelling and analysis of linear time-varying (LTV) filters with periodically variable coefficients. A transmission model of such filters has been described. Equations expressing the filter response for a given class of periodic parametric functions have been obtained and presented in a closed form. The results have been illustrated by an example

How does vintage equipment fit into a modern working process?

Contemporary students of record production, interested in the development of studio practice and seeking to discover historic techniques will find narratives that not only mythologise the works of producers such as Spector or Martin but cast historic pieces of equipment such as Pultec EQs and Fairchild limiters as mythical figures, venerated for their role in defining the sound of popular music. This chapter considers the continued use and veneration of analogue equipment, who's design first appeared over sixty years ago. As old designs are reissued by various companies copying original electrical circuits and using valves, and digital emulations are marketed as ‘tone titans’, deifying the original concepts, the question is how does this equipment fit into a modern working process and why does it survive

On the Informed Source Separation Approach for Interactive Remixing in Stereo

International audienceInformed source separation (ISS) has become a popular trend in the audio signal processing community over the past few years. Its purpose is to decompose a mixture signal into its constituent parts at the desired or the best possible quality level given some metadata. In this paper we present a comparison between two ISS systems and relate the ISS approach in various configurations with conventional coding of separate tracks for interactive remixing in stereo. The compared systems are Underdetermined Source Signal Recovery (USSR) and Enhanced Audio Object Separation (EAOS). The latter forms a part of MPEG's Spatial Audio Object Coding technology. The performance is evaluated using objective difference grades computed with PEMO-Q. The results suggest that USSR performs perceptually better than EOAS and has a lower computational complexity

Advanced automatic mixing tools for music

PhDThis thesis presents research on several independent systems that when combined together can generate an automatic sound mix out of an unknown set of multi‐channel inputs. The research explores the possibility of reproducing the mixing decisions of a skilled audio engineer with minimal or no human interaction. The research is restricted to non‐time varying mixes for large room acoustics. This research has applications in dynamic sound music concerts, remote mixing, recording and postproduction as well as live mixing for interactive scenes. Currently, automated mixers are capable of saving a set of static mix scenes that can be loaded for later use, but they lack the ability to adapt to a different room or to a different set of inputs. In other words, they lack the ability to automatically make mixing decisions. The automatic mixer research depicted here distinguishes between the engineering mixing and the subjective mixing contributions. This research aims to automate the technical tasks related to audio mixing while freeing the audio engineer to perform the fine‐tuning involved in generating an aesthetically‐pleasing sound mix. Although the system mainly deals with the technical constraints involved in generating an audio mix, the developed system takes advantage of common practices performed by sound engineers whenever possible. The system also makes use of inter‐dependent channel information for controlling signal processing tasks while aiming to maintain system stability at all times. A working implementation of the system is described and subjective evaluation between a human mix and the automatic mix is used to measure the success of the automatic mixing tools

Real-time Microphone Array Processing for Sound-field Analysis and Perceptually Motivated Reproduction

This thesis details real-time implementations of sound-field analysis and perceptually motivated reproduction methods for visualisation and auralisation purposes. For the former, various methods for visualising the relative distribution of sound energy from one point in space are investigated and contrasted; including a novel reformulation of the cross-pattern coherence (CroPaC) algorithm, which integrates a new side-lobe suppression technique. Whereas for auralisation applications, listening tests were conducted to compare ambisonics reproduction with a novel headphone formulation of the directional audio coding (DirAC) method. The results indicate that the side-lobe suppressed CroPaC method offers greater spatial selectivity in reverberant conditions compared with other popular approaches, and that the new DirAC formulation yields higher perceived spatial accuracy when compared to the ambisonics method

The Design of Audio Mixing Software Displays to Support Critical Listening

PhDThe mixing desk metaphor found in Digital Audio Workstations (DAW) is built upon a specialised and technical knowledge of signal flow and audio engineering. However, since their inception the DAW has gained a far wider and less technically specialised user-base. Furthermore, the limited screen space of laptop and tablet computers, combined with potentially limitless tracks in current DAWs has resulted in the need for complex interface navigation during mixing which may inhibit a fluid and intuitive approach to mixing. The research outlined in this thesis explores novel designs for Graphical User Interfaces (GUIs) for mixing, which acknowledge the changing role of the user, the limited space of tablet and mobile computers screens and the limitations of human perception during cross modal activities (aural and visual). The author designs and conducts several experiments using non-expert participants drawn from several music technology courses, to assess and quantify the extent to which current DAW designs might influence mixing workflow, aiming our research especially at beginner and non-expert users. The results of our studies suggest that GUIs which load visual working memory, or force the user to mentally integrate visual information across the interface, can reduce the ability to hear subtle simultaneous changes to the audio. We use the analysis of these experiments to propose novel GUI designs that are better suited to human cross-modal perceptual limitations and which take into account the specific challenges and opportunities afforded by screen-based audio mixers. By so doing, we aim to support the user in achieving a more fluid and focused interaction while mixing, where the visual feedback supports and enhances the primary goal of attending to and modifying the audio content of the mix. In turn, it is hoped this will facilitate the artistic and creative approaches required by music computer users

Responsive performance strategies with electronic feedback : shaping intrinsic behaviours

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Audio Mastering as a Musical Competency

In this dissertation, I demonstrate that audio mastering is a musical competency by elucidating the most significant, and clearly audible, facets of this competence. In fact, the mastering process impacts traditionally valued musical aspects of records, such as timbre and dynamics. By applying the emerging creative scholarship method used within the field of music production studies, this dissertation will aid scholars seeking to hear and understand audio mastering by elucidating its core practices as musical endeavours. And, in so doing, I hope to enable increased clarity and accuracy in future scholarly discussions on the topic of audio mastering, as well as the end product of the mastering process: records. Audio mastering produces a so-called master of a record, that is, a finished version of a record optimized for duplication and distribution via available formats (i.e, vinyl LP, audio cassette, compact disc, mp3, wav, and so on). This musical process plays a crucial role in determining how records finally sound, and it is not, as is so often inferred in research, the sole concern of a few technicians working in isolated rooms at a record label\u27s corporate headquarters. In fact, as Mark Cousins and Russ Hepworth-Sawyer (2013: 2) explain, nowadays “all musicians and engineers, to a lesser or greater extent, have to actively engage in the mastering process.” Thus, this dissertation clarifies the creative nature of audio mastering through an investigation of how mastering engineers hear records, and how they use technology to achieve the sonic goals they conceptualize