AES Show 2024 NY

Fourier theory is quite ubiquitous in modern audio signal processing. However, this framework is often at odds with our intuitions behind audio signals. Strictly speaking, Fourier theory is ideal to analyze periodic behaviors but when periodicities change across time it is easy to misinterpret its results. Of course, we have developed strategies around it like the Short Time Fourier Transform, yet again many times our interpretations of it falls beyond what the theory really says. This paper pushes the exact theoretical description showing examples where our interpretation of the data is incorrect. Furthermore, it shows specific instances where we incorrectly take decisions based on such paradoxical framework.

The nominal audio level is where developers of professional analog equipment design their units to have an optimal performance. Audio levels above the nominal level will at some point lead to increased harmonic distortion and eventually clipping. DSP plugins emulating such nonlinear behavior must – in the same manner as analog equipment – align to a nominal level that is simulated within the digital environment. A listening test was tailored to investigate if, or to which extent, misalignments in the audio levels compared to the simulated nominal level in analog-modelled DSP plugins are audible, thus affecting the outcome, depending on which level you choose to record at. The results of this study indicate that harmonic distortion in analog-modeled DSP plugins may become audible as the recording level increases. However, for the plugins included in this study, the immediate consequence of the harmonics added is not critical and, in most cases, not noticed by the listener.

Since the inception of electrical recording for phonograph records in 1924, records have been intentionally cut with a non-uniform frequency response to maximize the information density on a disc and to improve the signal-to-noise ratio. To reproduce a nominally flat signal within the available bandwidth, the effects of this cutting curve must be undone by applying an inverse curve on playback. Until 1953, with the introduction of what has become known as the RIAA curve, the playback curve required for any particular disc could vary by record company and over time. As a consequence, anyone seeking to hear or restore the information on a disc must have access to equipment that is capable of implementing multiple playback equalizations. This correction may be accomplished with either analog hardware or digital processing. The digital approach has the advantages of reduced cost and expanded versatility, but requires a transformation from continuous time, where the original curves are defined, to discrete time. This transformation inevitably comes with some deviations from the continuous-time response near the Nyquist frequency. There are many established methods for discretizing continuous-time filters, and these vary in performance, computational cost, and inherent latency. In this work, several methods for performing this transformation are explored in the context of phonograph playback equalization, and the performance of each approach is quantified. This work is intended as a resource for anyone developing systems for digital playback equalization or similar applications that require approximating the response of a continuous-time filter digitally.

This paper investigates using Time-Sensitive Networking (TSN) protocols, particularly from Audio Video Bridging (AVB), to support AES67 audio transport. By leveraging the IEEE 1588 Level 3 Precision Time Protocol (PTP) Media Profile, packet scheduling, and bandwidth reservation, we demonstrate that AES67 can be transported with AVB-equivalent quality guarantees while benefiting from Layer 3 networking advantages. The evolution of professional audio networking has increased the demand for high-quality, interoperable, and efficiently managed networks. AVB provides robust Layer 2 delivery guarantees but is limited by Layer 2 constraints. AES67 offers Layer 3 interoperability but lacks strict quality of service (QoS) guarantees. This paper proposes combining the strengths of both approaches by using TSN protocols to support AES67, ensuring precise audio transmission with Layer 3 flexibility. TSN extends AVB standards for time synchronization, traffic shaping, and resource reservation, ensuring low latency, low jitter, and minimal packet loss. AES67, a standard for high-performance audio over IP, leverages ubiquitous IP infrastructure for scalability and flexibility but lacks the QoS needed for professional audio. Integrating TSN protocols with AES67 achieves AVB's QoS guarantees in a Layer 3 environment. IEEE 1588 Level 3 PTP Media Profile ensures precise synchronization, packet scheduling reduces latency and jitter, and bandwidth reservation prevents congestion. Experiments show that TSN protocols enable AES67 to achieve latency, jitter, and packet loss performance on par with AVB, providing reliable audio transmission suitable for professional applications in modern, scalable networks.

Coherent sound wave interference is a persistent challenge in live sound reinforcement, where phase differences between multiple loudspeakers lead to destructive interference, resulting in inconsistent audio coverage. This review paper presents a modern solution: Diffuse Signal Processing (DiSP), which utilizes Temporally Diffuse Impulses (TDIs) to mitigate phase cancellation. Unlike traditional methods focused on phase alignment, DiSP manipulates the temporal and spectral characteristics of sound, effectively diffusing coherent wavefronts. TDIs, designed to spread acoustic energy over time, are synthesized and convolved with audio signals to reduce the likelihood of interference. This process maintains the original sound’s perceptual integrity while enhancing spatial consistency, particularly in large-scale sound reinforcement systems. Practical implementation methods are demonstrated, including a MATLAB-based workflow for generating TDIs and optimizing them for specific frequency ranges or acoustic environments. Furthermore, dynamic DiSP is introduced as a method for addressing interference caused by early reflections in small-to-medium sized rooms. This technique adapts TDIs in real-time, ensuring ongoing decorrelation in complex environments. The potential for future developments, such as integrating DiSP with immersive audio systems or creating dedicated hardware for real-time signal processing, is also discussed.