Journal Article•
Simultaneous Measurement of Impulse Response and Distortion with a Swept-Sine Technique
TL;DR: A novel measurement technique of the transfer function of weakly not-linear, approximately time-invariant systems is presented, based on an exponentially-swept sine signal, applicable to loudspeakers and other audio components, but also to room acoustics measurements.
Abstract: A novel measurement technique of the transfer function of weakly not-linear, approximately time-invariant systems is presented. The method is implemented with low-cost instrumentation; it is based on an exponentially-swept sine signal. It is applicable to loudspeakers and other audio components, but also to room acoustics measurements. The paper presents theoretical description of the method and experimental verification in comparison with MLS.
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Journal Article•
TL;DR: In this article, the transfer function measurements using sweeps as excitation signals rather than pseudonoise signals have been investigated and shown to have significantly higher immunity against distortion and time variance.
Abstract: Transfer-function measurements using sweeps as excitation signals rather than pseudonoise signals show significantly higher immunity against distortion and time variance. Capturing binaural room impulse responses for high-quality auralization purposes requires a signal-to-noise ratio (SNR) greater than 90 dB, which is unattainable with maximum-length sequence (MLS) measurements because of loudspeaker nonlinearity, but it is fairly easy to reach with sweeps due to the possibility of complete rejection of harmonic distortion. Before investigating the differences and practical problems of measurements with MLS and sweeps and arguing why sweeps are the preferable choice for the majority of measurement tasks, the existing methods of obtaining transfer functions are reviewed. The continual need to use preemphasized excitation signals in acoustical measurements is also addressed. A method to create sweeps with arbitrary spectral contents, but constant or prescribed frequency-dependent temporal envelope, is presented. Finally, the possibility of simultaneously analyzing transfer functions and harmonics is investigated.
445 citations
Cites background or methods from "Simultaneous Measurement of Impulse..."
...This possibility already anticipated by Griesinger [16] and described by Farina [15] will be further examined in chapter 6....
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...Indeed, this can be done separately for every single harmonic, as has already been proposed by Farina [15]....
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...If, however, the deconvolution is simply done with the time-inverted and amplitude-shaped stimulus, as proposed in [15], or if the correction is not feasible, as with TDS, then errors can be...
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TL;DR: An eight-channel database of head-related impulse responses and binaural room impulse responses is introduced, allowing for a realistic construction of simulated sound fields for hearing instrument research and, consequently, for a realism evaluation of hearing instrument algorithms.
Abstract: An eight-channel database of head-related impulse responses (HRIRs) and binaural room impulse responses (BRIRs) is introduced. The impulse responses (IRs) were measured with three-channel behind-the-ear (BTEs) hearing aids and an in-ear microphone at both ears of a human head and torso simulator. The database aims at providing a tool for the evaluation of multichannel hearing aid algorithms in hearing aid research. In addition to the HRIRs derived from measurements in an anechoic chamber, sets of BRIRs for multiple, realistic head and sound-source positions in four natural environments reflecting daily-life communication situations with different reverberation times are provided. For comparison, analytically derived IRs for a rigid acoustic sphere were computed at the multichannel microphone positions of the BTEs and differences to real HRIRs were examined. The scenes' natural acoustic background was also recorded in each of the real-world environments for all eight channels. Overall, the present database allows for a realistic construction of simulated sound fields for hearing instrument research and, consequently, for a realistic evaluation of hearing instrument algorithms.
299 citations
Cites background or methods from "Simultaneous Measurement of Impulse..."
...A comparison of the measurement results to an efficient method proposed by Farina [12] showed that the MIRS technique achieves competitive results in anechoic conditions with regard to signal-to-noise ratio and was better suited in public conditions (for details see [10])....
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...Furthermore, the broadband noise characteristics of MLS stimuli made them suitable for presentation in the public rather than, for example, sine-sweep stimuli-based methods [12]....
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TL;DR: This work shows how to compute the shape of a convex polyhedral room from its response to a known sound, recorded by a few microphones, and reconstructs the full 3D geometry of the room from a single sound emission, and with an arbitrary geometry ofThe microphone array.
Abstract: Imagine that you are blindfolded inside an unknown room. You snap your fingers and listen to the room’s response. Can you hear the shape of the room? Some people can do it naturally, but can we design computer algorithms that hear rooms? We show how to compute the shape of a convex polyhedral room from its response to a known sound, recorded by a few microphones. Geometric relationships between the arrival times of echoes enable us to “blindfoldedly” estimate the room geometry. This is achieved by exploiting the properties of Euclidean distance matrices. Furthermore, we show that under mild conditions, first-order echoes provide a unique description of convex polyhedral rooms. Our algorithm starts from the recorded impulse responses and proceeds by learning the correct assignment of echoes to walls. In contrast to earlier methods, the proposed algorithm reconstructs the full 3D geometry of the room from a single sound emission, and with an arbitrary geometry of the microphone array. As long as the microphones can hear the echoes, we can position them as we want. Besides answering a basic question about the inverse problem of room acoustics, our results find applications in areas such as architectural acoustics, indoor localization, virtual reality, and audio forensics.
241 citations
TL;DR: The ASR task as discussed by the authors was designed to identify keywords from sentences reverberantly mixed into audio backgrounds binaurally recorded in a busy domestic environment, and the challenge attracted thirteen submissions.
218 citations
TL;DR: An improved parametric model for a spring reverberation unit is presented, which is currently mainly used for simplified geometries or to generate reverberation impulse responses for use with a convolution method.
Abstract: The first artificial reverberation algorithms were proposed in the early 1960s, and new, improved algorithms are published regularly. These algorithms have been widely used in music production since the 1970s, and now find applications in new fields, such as game audio. This overview article provides a unified review of the various approaches to digital artificial reverberation. The three main categories have been delay networks, convolution-based algorithms, and physical room models. Delay-network and convolution techniques have been competing in popularity in the music technology field, and are often employed to produce a desired perceptual or artistic effect. In applications including virtual reality, predictive acoustic modeling, and computer-aided design of acoustic spaces, accuracy is desired, and physical models have been mainly used, although, due to their computational complexity, they are currently mainly used for simplified geometries or to generate reverberation impulse responses for use with a convolution method. With the increase of computing power, all these approaches will be available in real time. A recent trend in audio technology is the emulation of analog artificial reverberation units, such as spring reverberators, using signal processing algorithms. As a case study we present an improved parametric model for a spring reverberation unit.
218 citations
Cites background from "Simultaneous Measurement of Impulse..."
...The BBD is therefore a sampled analog system, and produces a delay given by the length of the capacitor chain and the frequency of the clocking signal....
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References
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Journal Article•
TL;DR: In this paper, the pathology of maximum length sequence (MLS) systems when there is distortion of various kinds is explored, and the properties of such a system are discussed in detail.
Abstract: A maximum length sequence (MLS) has mathematical properties that make it very useful as an excitation signal for measurement in audio and acoustics. The pathology of MLS systems when there is distortion of various kinds is explored
164 citations
"Simultaneous Measurement of Impulse..." refers background in this paper
...- 4 A mathematical explanation of the appearance of the spurious peaks in the MLS case was given in [2]....
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Journal Article•
TL;DR: In this article, a system comprising only two closely spaced loudspeakers can create very convincing virtual images around a single listener, and it is demonstrated that such a system is very robust with respect to head movement, and that the processing does not introduce any excessive artifacts.
Abstract: A system comprising only two closely spaced loudspeakers can create very convincing virtual images around a single listener. It is demonstrated that such a system is very robust with respect to head movement, and that the processing does not introduce any excessive artifacts. In practice, the loudspeakers ought to be pair matched in order to ensure accurate imaging.
121 citations
TL;DR: The aim of this work is to define a tightly standardized measurement procedure for the collection of a complete objective description of an opera house's acoustics, and some of the results obtained after measurements made in three different halls are presented.
50 citations
"Simultaneous Measurement of Impulse..." refers result in this paper
...It must be recalled that, in a previous comparative investigation among various measurement techniques [5], it was found that with the MLS technique there was no improvement in increasing the number of bits above 16, and in most cases the best results were obtained with the old MLSSA board, which is equipped with a single A/D converter with only 12 bits resolution....
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Journal Article•
TL;DR: The paper describes a suite of software tools, which have been developed as extensions to the most diffused shareware program for waveform editing on Win-32 PCs (Cool Edit 96 by David Johnston).
Abstract: The paper describes a suite of software tools, which have been developed as extensions to the most diffused shareware program for waveform editing on Win-32 PCs (Cool Edit 96 by David Johnston). For each tool a brief explanation of the underlying theory is given, along with an application example based on experimental work.
48 citations
"Simultaneous Measurement of Impulse..." refers background in this paper
...Both fast convolution and inverse filter generation are nowadays easy and cheap tasks, due to recently developed software [1,3]....
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...This means that, with MLS, the order of the shift register employed for the generation of the sequence must be high enough, depending on the reverberation time of the system: modern MLS measurement equipment can produce very high-order MLS signals [1], but previous systems occurred easily in the time-aliasing problem, which causes the late part of the reverberant tail to fold-back at the beginning of the time window containing the deconvolved h(t)....
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Journal Article•
26 citations
Additional excerpts
...Poletti [4] for linearly-swept sine signal....
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