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Methods of Measuring Impulse Responses in Architectural Acoustics
Abstract and Figures
New methods of measuring impulse responses based on carefully designed deterministic signals can further improve the performance offered by classical methods. In fact, these methods are particularly interesting when measuring long impulse responses as the ones analyzed in architectural acoustics. However, the effects of background and impulsive noise, distortion and time-variance are known rather qualitatively. For this reason, the ISO 18233 encourages to develop a deeper understanding of the theoretical bases of these techniques.
In this sense, this project presents an in depth analysis of two different methods of measuring impulse responses: the linear convolution of sweep signals with the inverse filter and the circular crosscorrelation of maximum length sequences (MLS) and inverse repeated sequences (IRS).
The results of this work reveal that the sweep technique can provide significant reduction of distortion compared to MLS/IRS technique but, unlike what is explained in the literature, sweep signals cannot reject all distortion artifacts from the causal part of the impulse response. Besides, it is proved that IRS sequences are immune to distortion of even order. On the other hand, it is confirmed that synchronous averaging procedure improves the SNR at the microphone position by 3 dB per doubling the number of averages. Alternatively, it is also proved that the noise contaminating the measured impulse response is reduced by 3 dB every time that the length of the excitation signal is doubled. In terms of impulsive noise, the sweep technique only contaminates specific frequency bands of the system's impulse response, whereas the MLS/IRS technique uniformly distributes all impulsive noise artifacts over the entire measured impulse response. Finally, it is also shown that MLS/IRS measurements are more vulnerable to time-varying systems than sweep measurements.
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... Furthermore, to avoid the division by small values, a suitable regularization method should be considered [39]. Alternatively, h(t) can be obtained by convolving the system response with the inverse of the excitation signal (x inv (t)) [40]: ...
... Except for this classical deconvolution method, several other approaches have been developed based on the properties of excitation signals to be used. As an example, the time-reversed filter and circular cross-correlation methods are particularly suitable for deconvolving some sweep signals and pseudo random sequences, respectively (see Section 2.2) [37,38,40,41]. ...
... In the literature, various methods (excitation signals and the corresponding deconvolution methods) have been proposed to measure acoustic impulse responses, e.g., impulse excitation [3,45], stepped-sine signals [37,38], sweep signals [41,46], time delay spectrometry (TDS, an early measurement method using sweep signals) [46], maximum length sequence (MLS) [47][48][49][50][51], Golay code [52], inverse repeated sequence (IRS) [53,54], random noise signals [35], etc. Most of them have been systematically described in [35,37,38,55], and a comparative analysis of several important methods can be found in [37,40,44]. Instead of using deconvolution methods, acoustic impulse responses can be estimated recursively with adaptive filtering approaches, which are widely applied to identify unknown systems [56]. ...
A head-related transfer function (HRTF) describes an acoustic transfer function between a point sound source in the free-field and a defined position in the listener’s ear canal, and plays an essential role in creating immersive virtual acoustic environments (VAEs) reproduced over headphones or loudspeakers. HRTFs are highly individual, and depend on directions and distances (near-field HRTFs). However, the measurement of high-density HRTF datasets is usually time-consuming, especially for human subjects. Over the years, various novel measurement setups and methods have been proposed for the fast acquisition of individual HRTFs while maintaining high measurement accuracy. This review paper provides an overview of various HRTF measurement systems and some insights into trends in individual HRTF measurements.
... In measurements performed for this study, we used the ESS technique [14], widely used in room impulse response (RIR) measurements [21], [18]. The ESS technique is used to estimate the feedback path IR f [n] and to investigate the nonlinear behavior of the actuator. ...
... Fortunately, this is not necessarily the case for the DIRs in fig. 9, given the good performance of the ESS technique in the presence of pink background noise, similar to the one recorded during the measurements [18], [21]. ...
Acoustic feedback is a very common problem in hearing instruments. Not only does it occur in common behindthe- ear (BTE) or in-the-ear hearing aids, it also affects bone conduction implants, middle ear implants, and more recent devices, such as the direct acoustic cochlear implant (DACI). In this paper, we present the data and analysis relating to the feedback path characterization of the CochlearTM CodacsTM DACI, performed on fresh frozen cadaver heads in four different measurement sessions (MSs). The general objectives were the following: 1) To measure and analyze the feedback path of the system and check for possible specimen-dependent variabilities; 2) To assess whether this feedback path is affected by an incorrect implantation; 3) To check for nonlinear behavior and 4) To determine differences between tissular and airborne feedback. The data analysis reveals that the feedback seems to be dependent on the specific head morphology of the implanted specimen, and that an incorrect implantation might strongly affect the feedback path; additionally, the analysis reveals that some nonlinear behavior at high stimulus levels can be expected and, finally, that the feedback path is characterized by a tissular feedback component with a rather different frequency content compared to the airborne (AB) feedback component.
The just noticeable differences (JNDs) of room acoustic parameters are important for the design of concert halls and, in general, research of room acoustics. Precise knowledge of JNDs helps the concert hall designer in assessing the impact that changes in the geometry or materials of the hall will have on its perceived acoustics. When designing a concert hall, creating an appropriate feeling of reverberance for the audience is of prime importance. The early decay time (EDT) parameter has proved to be a better predictor of the perception of reverberance than the classical reverberation time (T30), but no studies have been conducted to specifically determine the EDT JND. In the present study, the EDT JND was investigated for broadband conditions and assessed for individual frequency ranges. A subjective study was conducted with 26 subjects with musical training, in which 21 were considered reliable. The participants listened to orchestral music convolved with measured spatial room impulse responses from three concert halls. The stimuli were auralized in an anechoic chamber using third-order Ambisonic reproduction. The obtained values show that the JNDs for the broadband conditions are lower than those for the individual frequency ranges. The EDT JND for the broadband conditions was found to be approximately 18% of the EDT value.
Room Impulse Responses (RIRs) are typically measured using a set of microphones and a loudspeaker. When RIRs spanning a large volume are needed, many microphone measurements must be used to spatially sample the sound field. In order to reduce the number of microphone measurements, RIRs can be spatially interpolated. In the present study, RIR interpolation is formulated as an inverse problem. This inverse problem relies on a particular acoustic model capable of representing the measurements. Two different acoustic models are compared: the plane wave decomposition model and a novel time-domain model that consists of a collection of equivalent sources creating spherical waves. These acoustic models can both approximate any reverberant sound field created by a far field sound source. In order to produce an accurate RIR interpolation, sparsity regularization is employed when solving the inverse problem. In particular, by combining different acoustic models with different sparsity promoting regularizations, spatial sparsity, spatio-spectral sparsity and spatio-temporal sparsity are compared. The inverse problem is solved using a matrix-free large scale optimization algorithm. Simulations show that the best RIR interpolation is obtained when combining the novel time-domain acoustic model with the spatio-temporal sparsity regularization, outperforming the results of the plane wave decomposition model even when far fewer microphone measurements are available.
In comparison to using noise as excitation signal, impulse response measurements with swept sine excitation bear the advantage that harmonic distortion products are concentrated in the non-causal part of the impulse response, where they can be easily eliminated. For this reason, the achievable dynamic range in room acoustical measurements with sweeps is considerably greater than with (pseudo-random) noise excitation. However, based on theoretical simulation, some authors have cast in doubt the distortion rejection capabilities of impulse response measurements with sweeps. This work compares the real world performance of room impulse response measurements with pseudo-random noise and sweeps and also addresses some common pitfalls incurring when simulating distortion influence.
This paper investigates the determination and modeling of uncertainty contributions in measurements of room acoustic parameters, which are commonly used to de- scribe the acoustic situation of a room in an objective man- ner. Without giving the range of uncertainty and the con- fidence interval, the results remain incomparable to other measurement teams, since modern PC-based measurements still show appreciable sources of measurement errors. The Guide to the Expression of Uncertainty in Measurement (GUM) defines a unified guideline to determining uncertain- ties in all fields of measurements. Its application to measure- ments is increasingly required by modern measurement stan- dards. However, the GUM procedures have not been applied to room acoustics yet. Hence, a scalable linear approach for calculating the combined uncertainty of room acoustic pa- rameters with regard to the input quantities is proposed. In- situ measurement results of specially designed experiments show the significance of the main influence factors and are used to build the uncertainty budget.
Many frequency domain audio tests, such as measuring the frequency response, measuring frequency-dependent crosstalk, or making some kind of distortion measurements, can be performed by applying multitone test signals to the audio device under test and analyzing the output signal by means of the discrete Fourier transform. In contrast to standardized single-tone or dual-tone excitation tests, the multitone test offers significant time savings and enables realistic test conditions by approximating the frequency domain characteristics of a typical program material. But multitone testing requires the design of low-crest-factor test signals. Several computation schemes and indirect optimization methods of low-crest-factor phases, are discussed, and for the first time optimum multitone signals designed by general-purpose nonlinear optimization software are presented.
The equivalence of M-sequence matrices to Walsh-Hadamard matrices can be exploited to take advantage of the latter's fast transform algorithm. The nature of the obvious equivalence, its relation to the M-sequence and its implementation as address modifications realized by linear feedback shift-registers are discussed.
A computer-generated pulse signal for sound measurement is discussed. A pulse signal whose power spectrum is flat is generated by inverse Fourier transformation. The generation of a time-stretched pulse and its compression method are also considered. Computer-controlled measurements enable time averaging and the elimination of reflected sound is made in the computer memory by the operator's instruction monitoring acquired waveform on CRT.
A comprehensive analysis of transfer-function measurement based on maximum-length sequences (MLS) is presented. MLS methods employ efficient cross correlation between input and output to recover the periodic impulse response (PIR) of the system being measured. In the face of external noise and nonlinearities, the MLS approach is shown to be as robust as time-delay spectrometry (TDS). Like TDS, MLS methods are also capable of rejecting or selecting nonlinear (distortion) components when measuring weakly nonlinear systems. An MLS coherence function is defined that is not unlike the coherence function usually associated with dual-channel FFT analyzers. Finally, a new low-cost instrument based on the IBM-PC makes MLS measurements generally available and affordable.
A new method is presented for acquiring impulse responses in concert halls with large signal-to-noise ratios and with a high resolution. In our proposal an omnidirectional loudspeaker is used which is driven bY an amplified sweep: a signal containing all frequencies of interest smeared out in time. By using a deconvolution technique, an almost perfect pulse is obtained with a high peak pressure and a short effective duration. Measurements were made in two different concert halls to illustrate the practical implications of the new technique.