AES Milan 2018
Poster Session P05
P05 - Posters: Applications
Wednesday, May 23, 14:15 — 15:45 (Arena 2)
P05-1 Grid-Based Stage Paradigm with Equalization Extension for “Flat-Mix” Production—Jonathan Wakefield, University of Huddersfield - Huddersfield, UK; Christopher Dewey, University of Huddersfield - Huddersfield, UK; William Gale, University of Huddersfield - Huddersfield, UK
In the Stage Paradigm (SP) the visual position of each channel widget represents the channel’s level and pan position. The SP has received favorable evaluation but does not scale well to high track counts because channels with similar pan positions and level visually overlap/occlude. This paper considers a modified SP for creating a “flat-mix” that provides coarse control of channel level and pan position using a grid-based, rather than continuous, stage and extends the concept to EQ visualization. Its motivation was to convert the “overlap” deficiency of the SP into an advantage. All subjects were faster at creating audibly comparable flat-mixes with the novel SP. Subject selected satisfaction keywords were also very positive.
Convention Paper 9930 (Purchase now)
P05-2 Real Time Implementation of an Active Noise Control for Snoring Reduction—Stefania Cecchi, Universitá Politecnica della Marche - Ancona, Italy; Alessandro Terenzi, Universita Politecnica delle Marche - Ancona, Italy; Paolo Peretti, Leaff Engineering - Osimo, Italy; Ferruccio Bettarelli, Leaff Engineering - Osimo, Italy
Snoring is a well-known problem in our society. Active noise control systems can be applied to partially solve this annoyance. In this context, the presented work aims at proposing a real-time implementation of an active noise control algorithm for snoring reduction by means of a DSP embedded platform and an innovative headboard equipped with microphones and loudspeakers. Several experimental results with different snoring signals have shown the potentiality of the proposed approach in terms of computational complexity and noise reduction.
Convention Paper 9931 (Purchase now)
P05-3 Identification of Nonlinear Audio Devices Exploiting Multiple-Variance Method and Perfect Sequences—Simone Orcioni, Universita Politecnica delle Marche - Ancona, Italy; Alberto Carini, University of Urbino Carlo Bo - Urbino, Italy; Stefania Cecchi, Universitá Politecnica della Marche - Ancona, Italy; Alessandro Terenzi, Universita Politecnica delle Marche - Ancona, Italy; Francesco Piazza, Universitá Politecnica della Marche - Ancona (AN), Italy
Multiple-variance identification methods are based on the use of input signals with different powers for nonlinear system identification. They overcome the problem of the locality of the solution of traditional identification methods that well approximates the system only for inputs with approximately the same power of the identification signal. In this context, it is possible to further improve the nonlinear filter estimation exploiting as input signals the perfect periodic sequences that guarantee the orthogonality of the Wiener basis functions used for identification. Experimental results involving real measurements show that the proposed approach can accurately model nonlinear devices on a wide range of input variances. This property is particularly useful when modeling systems with high dynamic inputs, like audio amplifiers.
Convention Paper 9932 (Purchase now)
P05-4 Power Saving Audio Playback Algorithm Based on Auditory Characteristics—Mitsunori Mizumachi, Kyushu Institute of Technology - Kitakyushu, Fukuoka, Japan; Wataru Kubota, Kyushu Institute of Technology - Fukuoka, Japan; Mitsuhiro Nakagawara, Panasonic - Yokohama City, Kanagawa, Japan
Music appreciation can be achieved with a variety of manners such as smartphones, portable music players, car audio, and high-end audio systems. Power consumption is one of the important issues for portable electronic devices and electric vehicles. In this paper a power saving audio playback algorithm is proposed in restraint of perceptual distortion. An original music source is passed through filterbanks and is reconstructed after increasing or decreasing the narrow-band component in each channel. The channel-dependent manipulation is carefully done not to cause perceptual distortion. The feasibility of the proposed method is evaluated by both measuring consumption current while music playback and carrying out a listening test.
Convention Paper 9933 (Purchase now)
P05-5 Deep Neural Networks for Road Surface Roughness Classification from Acoustic Signals—Livio Ambrosini, Universita Politecnica delle Marche - Ancona, Italy; ASK Industries S.p.A. - Montecavolo di Quattro Castella (RE), Italy; Leonardo Gabrielli, Universitá Politecnica delle Marche - Ancona, Italy; Fabio Vesperini, Universita Politecnica delle Marche - Ancona, Italy; Stefano Squartini, Università Politecnica delle Marche - Ancona, Italy; Luca Cattani, Ask Industries S.p.A. - Montecavolo di Quattrocastella (RE), Italy
Vehicle noise emissions are highly dependent on the road surface roughness and materials. A classification of the road surface conditions may be useful in several regards, from driving assistance to in-car audio equalization. With the present work we exploit deep neural networks for the classification of the road surface roughness using microphones placed inside and outside the vehicle. A database is built to test our classification algorithms and results are reported, showing that the roughness classification is feasible with the proposed approach.
Convention Paper 9934 (Purchase now)
P05-6 Elicitation and Quantitative Analysis of User Requirements for Audio Mixing Interface—Christopher Dewey, University of Huddersfield - Huddersfield, UK; Jonathan Wakefield, University of Huddersfield - Huddersfield, UK
Existing Audio Mixing Interfaces (AMIs) have focused primarily on track level and pan and related visualizations. This paper places the user at the start of the AMI design process by reconsidering what are the most important aspects of an AMI’s visual feedback from a user’s perspective and also which parameters are most frequently used by users. An experiment was conducted with a novel AMI which in one mode provides the user with no visual feedback. This enabled the qualitative elicitation of the most desired visual feedback from test subjects. Additionally, logging user interactions enabled the quantitative analysis of time spent on different mix parameters. Results with music technology undergraduate students suggest that AMIs should concentrate on compression and EQ visualization.
Convention Paper 9935 (Purchase now)
P05-7 Real-Time Underwater Spatial Audio: A Feasibility Study—Symeon Delikaris-Manias, Aalto University - Helsinki, Finland; Leo McCormack, Aalto University - Espoo, Finland; Ilkka Huhtakallio, Aalto University - Espoo, Finland; Ville Pulkki, Aalto University - Espoo, Finland
In recent years, spatial audio utilizing compact microphone arrays has seen many advancements due to emerging virtual reality hardware and computational advances. These advances can be observed in three main areas of spatial audio, namely: spatial filtering, direction of arrival estimation, and sound reproduction over loudspeakers or headphones. The advantage of compact microphone arrays is their portability, which permits their use in everyday consumer applications. However, an area that has received minimal attention is the field of underwater spatial audio, using compact hydrophone arrays. Although the principles are largely the same, microphone array technologies have rarely been applied to underwater acoustic arrays. In this feasibility study we present a purpose built compact hydrophone array, which can be transported by a single diver. This study demonstrates a real-time underwater acoustic camera for underwater sound-field visualization and a parametric binaural rendering engine for auralization.
Convention Paper 9936 (Purchase now)
P05-8 Dual-Band PWM for Filterless Class-D Audio Amplification—Konstantinos Kaleris, University of Patras - Patras, Greece; Charalampos Papadakos, University of Patras - Rio, Greece; John Mourjopoulos, University of Patras - Patras, Greece
The benefits of Dual-Band Pulse Width Modulation (DBPWM) are demonstrated for filter-less audio class-D amplifier implementations. DBPWM is evaluated in terms of energy efficiency (thermal loading) and reproduction fidelity for direct driving of loudspeaker units by DBPWM signals. Detailed physical models of low-frequency (woofer) and high-frequency (tweeter) loudspeakers are employed for simulation of the coupling between the DBPWM signal and the electro-mechanical and acoustic properties of loudspeaker systems in the broadband PWM spectral range. Derived frequency responses are used to estimate the reproduced sound signal of the loudspeaker. Equivalent impedance of the speaker's voice coil is used to estimate thermal loading by the DBPWM signal's out-of-band spectral energy, comparing standard filtered PWM implementations to the proposed method.
Convention Paper 9937 (Purchase now)
P05-9 Low Cost Algorithm for Online Segmentation of Electronic Dance Music—Emanuel Aguilera, Universitat Politecnica de Valencia - Valencia, Spain; Jose J. Lopez, Universidad Politecnica de Valencia - Valencia, Spain; Pablo Gutierrez-Parera, Universitat Politecnica de Valencia - Valencia, Spain; Carlos Hernandez, Universitat Politecnica de Valencia - Valencia, Spain
Visual content animation for Electronic Dance Music (EDM) events is an emerging and demanded but costly service for the industry. In this paper an algorithm for automatic EDM structure segmentation is presented, suitable for the automation of video and 3D animation synchronized to the audio content. The algorithm implements low cost time/frequency features smoothed with a novel algorithm, resulting in a low latency output. Segmentation stage is based on a multiple threshold algorithm specifically tuned to EDM. It has been implemented in C performing in real-time and has been successfully integrated in a real-time commercial 3D graphics engine with great results over EDM music sets.
Convention Paper 10027 (Purchase now)