Academic Papers

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The Feasibility of Class D amplifiers for Active Loudspeakers Applications

Darren Rose / Genelec Oy, Iisalmi, Finland. August 2013.

In 2002 the author made a comparison of the audio quality, audio quantity and cost of some commonly available power amplifier modules. This paper will investigate the hypothesis that today the best value can obtained using Class D am- plifiers. This will be studied by comparing four topologies. The amplifiers have been measured in the same controlled conditions representing an application in an active loudspeaker. To make a fair cost comparison the amplifiers have been assembled on PCBs from components in-house rather than using complete ready power amplifier modules. In addition to the audio and cost criteria, energy efficiency will be considered.

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Swarm Synchronization for Multi-Recipient Multimedia Streaming

Mika Rautiainen, Hannu Aska, Timo Ojala, Matti Hosio, Aki Mäkivirta and Niko Haatainen - MediaTeam Oulu, Department of Electrical and Information Engineering, University of Oulu, Finland and Genelec Oy, Iisalmi, Finland. September 2009.

IP networks allow constructing versatile device configurations for multimedia streaming. However, the stochastic nature of the packet-switched data transmission may complicate IP-based implementations of some conventional applications such as analog wired transmission of synchronized multi-channel audio. This paper introduces a multimedia streaming system based on the synchronization of multiple playback clients as a ‘swarm’. The proposed ‘swarm synchronization’ mechanism is based on precise clock synchronization with the PTP protocol and adjusting the client-specific sampling rates according to the true playback rates of other clients.

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Virtualized Listening Tests for Loudspeakers

Timo Hiekkanen, Aki Mäkivirta, Matti Karjalainen - Department of Signal Processing and Acoustics, Helsinki University of Technology, Espoo, Finland and Genelec Oy, Iisalmi, Finland.
J. Audio Eng. Soc., Vol. 57, No. 4, 2009 April.

The precise location of a loudspeaker in a listening room is known to affect loudspeaker preference ratings. When multiple loudspeakers are compared, the evaluation is limited by the poor human auditory memory. To overcome these problems, a method to evaluate and compare loudspeakers using headphones is proposed. The method utilizes personal headrelated transfer functions in rendering the sound field recorded in a standard listening room with an artificial head. The equalization of circumaural headphones and the artifical-head responses for individual listeners are investigated. Formal listening tests are conducted to examine differences between the proposed binaural method and real loudspeakers in a standard listening room. Listening tests show that the virtualized loudspeakers can be nearly imperceptible from reality in many but not all cases.

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Virtualized Listening Tests for Loudspeakers

Timo Hiekkanen, Matti Karjalainen, Aki Mäkivirta - Department of Signal Processing and Acoustics, Helsinki University of Technology and Genelec Oy
AES 124th Convention, Amsterdam, The Netherlands, 2008 May 17-20

The precise location of a loudspeaker in a listening room is known to affect loudspeaker preference ratings. When multiple loudspeakers are compared the evaluation is limited by the poor human auditory memory. To overcome these problems, a method to evaluate and compare loudspeakers using headphones is proposed. The method utilizes personal head-related transfer functions in rendering the sound field recorded in a standard listening room with an artificial head. Equalization of circumaural headphones and the artificial head are investigated. Formal listening tests are conducted to examine differences between the proposed binaural method and real loudspeakers in a standard listening room. Listening tests show that the virtualized loudspeakers can be nearly imperceptible from reality in many but not in all cases.

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A Listening Test System for Measuring the Threshold of Audibility
of Temporal Decays

Andrew Goldberg
Proceedings of The Institute of Acoustics, Vol 27, 2005

A listening test system designed to measure the threshold of audibility of the decay time of low frequency resonances is described. The system employs the Parameter Estimation by Sequential Testing (PEST) technique and the listening test is conducted on calibrated headphones to remove factors associated with the listening environment. Program signal, replay level, and resonance frequency are believed to influence decay time threshold. A trial listening test shows that the system reveals realistic results but the temporal resonance modelling filter requires some adjustment to remove audible non-modal cues. Transducer limitations still affect the test at low frequencies and high replay levels. Factors for a future large-scale listening test are refined. Early indications are that temporal decay thresholds rise with reduced frequency and SPL. 

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Compensating the Acoustical Loading of Small Loudspeakers Mounted Near Desktops

Andrew Goldberg, Aki Mäkivirta and Ari Varla
The Institute of Acoustics Reproduced Sound 20 Conference, Oxford UK, October 2004 / AES 117th Convention, San Francisco, CA, USA, November 2004


In professional audio applications, small loudspeakers are often mounted on or near (within the loudspeaker’s near field region) large solid surfaces, such as mixing consoles, desktops and work surfaces. In approximately two-thirds of loudspeakers mounted in such a fashion, the magnitude response is compromised in a predictable and systematic way. An upward deviation of peak value 5.0 dB ± 1.5 dB centred on 141 Hz ± 31 Hz was observable in approximately 80% of the cases studied. An additional Room Response Control in active loudspeakers is proposed to compensate for this aberration. A statistical analysis of 89 near-field loudspeakers helps define the correction filter, and quantifies the effectiveness of the fixed filter design. Use of the proposed filter in an automated response optimisation algorithm for in-situ response equalisation is demonstrated. 

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Performance Comparison of Graphic Equalisation and Active Loudspeaker Room Response Controls

Andrew Goldberg and Aki Mäkivirta
AES 116th Convention, Berlin Germany, 2004 May 8-11

We compare the room response controls available in active loudspeakers to a third-octave graphical equaliser. The room response controls are set using an automated optimisation method presented in earlier AES publications. A third-octave ISO frequency constant-Q graphic equaliser is set to minimise the least squares deviation from linear within the passband in a smoothed acoustical response. The resulting equalisation performance of the two methods is compared using objective metrics, to show how these standard room response equalising methods perform. For all loudspeaker models pooled together, the room response controls improve the RMS deviation from a linear response from 6.1 dB to 4.7 dB (improvement 22%), whereas graphic equalisation improves the RMS deviation to 1.8 dB (improvement 70%). Both equalisation techniques achieve a similar improvement in the broadband balance, which has been shown to affect a subjective lack of colouration in sound systems. The optimisation time for a graphic equaliser is up to 48 times longer compared to that for active loudspeaker room response controls 

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Perception of Temporal Decay of Low-frequency Room Modes

Matti Karjalainen, Poju Antsalo, Aki Mäkivirta and Vesa Välimäki
AES 116th Convention, Berlin Germany, 2004 May 8-11

Modal equalization has recently been of research interest in order to improve sound reproduction in rooms that have excessively strong modes at low frequencies. Instead of acoustic treatment by expensive and space-reserving absorbing structures, modal equalization is based on DSP affecting the electric-to-acoustic reproduction chain. Several DSP-based techniques for modal equalization have been proposed recently and tested in performance. From a perceptual point of view, however, no clear picture on the importance of controlled temporal decay has been shown, although it is known that towards the lowest frequencies the human hearing becomes increasingly insensitive to temporal details. In the present study we conducted listening tests where only a single synthetic mode with increased decay time but magnitude-equalized response was used to find the JND threshold of excessive decay time. The main conclusion is that at typical listening levels and down to 100 Hz the modal decay time T60 is allowed to increase from about 0.3 seconds by 0.1 to 0.4 seconds, while at 50 Hz even decay times of up to two seconds do not make a noticeable difference. 

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An Automated In-situ Frequency Response Optimisation Algorithm for Active Loudspeakers, Including a Statistical Analysis of Its Performance

Andrew Goldberg and Aki Mäkivirta
November 2003, The Institute of Acoustics Reproduced Sound 19 Conference, Oxford, UK


This paper presents a novel method for robust automatic selection of the optimal in-situ acoustical frequency response within a discrete-valued set of responses offered by room response controls on active loudspeakers. The rationale of the room response controls for the active loudspeakers is described. The frequency response, calculated from the acquired impulse response, is used as the input for the optimisation algorithm to select the most favourable combination of room response controls. The optimisation algorithm is described and the performance of the algorithm is statistically analysed and discussed. It improves the acoustical similarity between loudspeakers in one space and performs robustly and systematically in widely varying acoustical environments. The efficiency and performance of the algorithm is discussed. The algorithm dramatically improves the speed of optimisation compared to an exhaustive search. This algorithm has been implemented and is currently in active use by specialist loudspeaker system calibrators who set up and tune studios and listening rooms. 

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Reproducing Commercial Multichannel Formats Through a Single Monitoring System

Andrew Goldberg
September 2003, IBC Conference, Amsterdam, Netherlands

A modern audio production facility must be able to supply productions in a large number of different formats. The change from mono & stereo to multi-channel reproduction has created many problems, both in converting existing production facilities to multichannel format and when building new installations. This paper examines the LFE channel across different encoding formats and presents a method to reproduce it, using a bass management system, through a single monitoring system. 

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Statistical Analysis of an Automated In-Situ Frequency Response Optimisation Algorithm for Active Loudspeakers

Andrew Goldberg and Aki Mäkivirta
May 2003, AES 23rd International Conference, Copenhagen, Denmark


This paper presents a novel method for automatically selecting the optimal in-situ acoustical frequency response of active loudspeakers within a discrete-valued set of responses offered by room response controls on active loudspeakers. The rationale of the room response controls for the active loudspeakers is explained. The frequency response, calculated from the acquired impulse response, is used as the input for the optimisation algorithm to select the most favourable combination of room response controls. The optimisation algorithm is described. The performance of the algorithm is analysed and discussed. This algorithm has been implemented and is currently in active use by specialist loudspeaker system calibrators who set up and tune studios and listening rooms. 

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Automated In-situ Frequency Response Optimisation of Active Loudspeakers

Andrew Goldberg and Aki Mäkivirta
March 2003, AES 114th Convention, Amsterdam, Netherlands

This paper presents a novel method for robust automatic selection of optimal in-situ acoustical frequency response within a discrete-valued set of responses offered by room response controls on an active loudspeaker. A frequency response measurement is used as the input data for the algorithm. The rationale of the room response control system is described. The response controls are described for each supported loudspeaker type. The optimisation algorithm is described. Examples of the optimisation process are given. The efficiency and performance of the algorithm are discussed. The algorithm dramatically improves the speed of optimisation compared to an exhaustive search. It improves the acoustical similarity between loudspeakers in one space and performs robustly and systematically in widely varying acoustical environments. The algorithm is currently in active use by specialists who set up and tune studios and listening rooms. 

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A Comparison of Modular State-of-the-Art Switch Mode and Linear Audio Power Amplifiers

Darren Rose
AES 112th Convention, Munich, Germany, May 2002

Modern commercially available, compact, low power audio power amplifiers are mostly designed around one of three main technologies. These are integrated circuit class AB, thick film hybrid class AB, and switch mode power amplifier modules. The decision to use a particular technology is not only based on idealised performance specifications, but also on the performance under realistic operating conditions, and cost-to-performance considerations. In this study, the performance of each amplifier technology is studied in ideal and realistic operating conditions with two amplifier designs for each technology category. Regulated and unregulated power supplies are used, in combination with ideal resistive and real-life complex impedance loudspeaker loads. For a fixed nominal supply voltage, the value of the different technologies with regard to noise, distortion and continuous output power is discussed. This results in an analysis of the cost effectiveness, or value, of currently competing technologies for high quality, low power, compact audio power amplifiers. 

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A survey study Of In-Situ Stereo And Multi-Channel Monitoring Conditions

Aki Mäkivirta and Christophe Anet
September 2001, AES 111th Convention, New York, USA

The in-situ responses of a total of 372 loudspeakers in 164 professional monitoring rooms around the world have been measured after acoustical calibration. All measured rooms have been equipped with factory calibrated three way monitors and acoustically calibrated with standardized apparatus. The results provide a thorough understanding of typical monitoring conditions for stereo and multi-channel rooms, distribution in room parameters and quality of reproduced audio. Results are compared to current standards and recommendations. 

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Low-Frequency Modal Equalization Of Loudspeaker-Room Responses

Aki Mäkivirta, Poju Antsalo, Matti Karjalainen and Vesa Välimäki
September 2001, AES 111th Convention, New York, USA

In a room with strong low-frequency modes the control of excessively long decays is problematic or impossible with conventional passive means. In this paper we present a systematic methodology for active modal equalization able to correct the modal decay behavior of a loudspeaker-room system. Two methods of modal equalization are proposed. The first method modifies the primary sound such that modal decays are controlled. The second method uses separate primary and secondary radiators and controls modal decays with sound fed into the secon-dary radiator. Case studies of the first method of implementation are presented. 

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Estimation of Modal Decay Parameters from Noisy Response Measurements

M. Karjalainen, P. Antsalo, A. Mäkivirta, T. Peltonen, V. Välimäki
May 2001, AES 110th Convention Amsterdam, The Netherlands

Estimation of modal decay parameters from noisy measurements of reverberant and resonating systems is a common problem in audio and acoustics, e.g., in room and concert hall measurements or musical instrument modeling. In this paper, reliable methods to estimate the initial response level, decay rate and noise floor level from noisy measurement data are studied and compared. A new method, based on nonlinear optimization of a model for exponential decay plus stationary noise floor, is presented. Comparison with traditional decay parameter estimation techniques using simulated measurement data shows that the proposed method outperforms in accuracy and robustness, especially in extreme SNR conditions. Three cases of practical applications of the method are demostrated. 

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A method for orthogonal amplitude and delay processing of subjective listening test material

Aki Mäkivirta and Jan Abildgaard Pedersen
September 2000, AES 109th Convention Los Angeles, USA


We present a method to change amplitude and delay responses in subjective listening test material independently of each other. This is necessary in subjective listening experiments to apply modern statistical methods treating simultaneously several statistical variables. A case study of producing audio test material with this method is presented. This is related to an experiment where the audibility of the amplitude response variation and the delay response variation are studied at low frequencies based on data obtained from impulse responses of a room. 

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A Design of rooms for multichannel audio monitoring

A. Varla, A. Mäkivirta, I. Martikainen, M. Pilchner, R.Schoustal, C. Anet
April 1999, AES 16th International Conference Rovaniemi, Finland

This paper presents an overview of the practical aims, methods and problems of designing monitoring rooms for multichannel audio. The main problem areas encountered in practical multichannel audio monitoring room are described with a case study of a practical high quality monitoring room and with scale model measurements. The monitoring room studied does not follow entirely the ITU-R BS 775-1 international recommendation on speaker placement. The scale model is also used to investigate the effect of structural modifications to the room suggested in literature as methods to achieve better performance in multichannel audio reproduction. The design principles and methodology applied in the industry are reviewed, and some suggestions are developed to guide the monitoring room designer. 

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