Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634114
D. Wiese
In spite of applied high sophisticated channel coding techniques [3] non correctable errors wiU occur during several conditions of reception. Conventional FM-receivers are well known for audible distortions during mobile reception in urban or montaineous regions or at the fringe of the coverage area. DAB could provide intelligent concealment techniques in the case of non correctable errors. Different methods are known to conceal burst-errors of durations up to some 20ms or even more than l a m s [4].
{"title":"Source related error concealment techniques for Digital Audio Broadcasting (DAB) considering the listeners perception","authors":"D. Wiese","doi":"10.1109/ASPAA.1991.634114","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634114","url":null,"abstract":"In spite of applied high sophisticated channel coding techniques [3] non correctable errors wiU occur during several conditions of reception. Conventional FM-receivers are well known for audible distortions during mobile reception in urban or montaineous regions or at the fringe of the coverage area. DAB could provide intelligent concealment techniques in the case of non correctable errors. Different methods are known to conceal burst-errors of durations up to some 20ms or even more than l a m s [4].","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"49 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114777104","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634143
S. Elliott, M. Tsujino
L Active control systems are often implemented as feedforward controllers, using a reference signal correlated ‘with the disturbance which is to be controlled. In many cases the disturbance is periodic. If the field to be controlled were perfectly stationary, a fixed, timeinvariant, feedforward controller could be implemented, which could be designed beforehand to give optimal reductions. In most practical situations, however, the primary disturbance is changing either in magnitude, phase, or frequency and the controller has to be made adaptive in order to track these changes. Such adaptive controllers have transient convergence propemes which, in general, it is difficult to analyse. This is because of the interaction between the dynamic behaviour of the controller and the dynamic behaviour of the physical system under control. The block diagram of a single channel adaptive feedforward controller is shown in Figure 1. Typically, the controller is implemented as an FIR digital filter and the algorithm used to adjust the filter coefficients is the fdtered-x LMS algorithm widrow and Steams, 19851, for which there is a multichannel generation known as the Multiple Error LMS algorithm Flliott et al., 19871. If it is assumed that the controller is adapting slowly compared with the delays and time coristants of the system under control, fairly conventimal methods can be used to analyse this algorithm, which are similar to those used in the analysis of the electrical LMS algorithm PVidrow and Stems, 19851. It has been observed, however, that in practice the filtered-x LMS irlgorithm is able to adapt much faster than this.
主动控制系统通常被实现为前馈控制器，使用与被控制扰动相关的参考信号。在许多情况下，这种干扰是周期性的。如果要控制的场是完全静止的，则可以实现一个固定的、时不变的前馈控制器，该控制器可以事先设计以给出最佳的缩减。然而，在大多数实际情况下，主要干扰的大小、相位或频率都在变化，为了跟踪这些变化，控制器必须自适应。这种自适应控制器具有暂态收敛性，一般情况下难以分析。这是因为控制器的动态行为和被控制的物理系统的动态行为之间的相互作用。单通道自适应前馈控制器框图如图1所示。通常，控制器被实现为FIR数字滤波器，用于调整滤波器系数的算法是fdterd -x LMS算法(widrow and Steams, 19851)，其中有一个多通道生成称为多误差LMS算法(Flliott et al.， 19871)。如果假设控制器与被控系统的延迟和时间常数相比自适应较慢，则可以使用相当常规的方法来分析该算法，类似于分析电LMS算法(PVidrow and stem, 19851)时使用的方法。然而，已经观察到，在实践中，滤过的x LMS算法能够比这更快地适应。
{"title":"The Performance Of Adaptive Feedforward And Optimal Feedback Active Control Systems","authors":"S. Elliott, M. Tsujino","doi":"10.1109/ASPAA.1991.634143","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634143","url":null,"abstract":"L Active control systems are often implemented as feedforward controllers, using a reference signal correlated ‘with the disturbance which is to be controlled. In many cases the disturbance is periodic. If the field to be controlled were perfectly stationary, a fixed, timeinvariant, feedforward controller could be implemented, which could be designed beforehand to give optimal reductions. In most practical situations, however, the primary disturbance is changing either in magnitude, phase, or frequency and the controller has to be made adaptive in order to track these changes. Such adaptive controllers have transient convergence propemes which, in general, it is difficult to analyse. This is because of the interaction between the dynamic behaviour of the controller and the dynamic behaviour of the physical system under control. The block diagram of a single channel adaptive feedforward controller is shown in Figure 1. Typically, the controller is implemented as an FIR digital filter and the algorithm used to adjust the filter coefficients is the fdtered-x LMS algorithm widrow and Steams, 19851, for which there is a multichannel generation known as the Multiple Error LMS algorithm Flliott et al., 19871. If it is assumed that the controller is adapting slowly compared with the delays and time coristants of the system under control, fairly conventimal methods can be used to analyse this algorithm, which are similar to those used in the analysis of the electrical LMS algorithm PVidrow and Stems, 19851. It has been observed, however, that in practice the filtered-x LMS irlgorithm is able to adapt much faster than this.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"99 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121496660","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634139
P. Rayner, S. Godsill
Tdep h o n e: (0223)-332 767 This paper presents recent developments in techniques for the restoration of audio material degraded by clicks. This type of degradation is associated with all forms of audio media, including CD and D.+IT, but is most characteristic of gramophone disks. The term 'click' covers a wide variety of problems ranging from loud isolated 'pops' to the relatively low level 'craclde' associated with most i s r p m records. The term 'scratch' is also used in this paper t o indicate the same type of degradation. JIost tj.pes of click can be modelled as bursts of corrupted audio samples occuring at random times and of randoni duration. For example, a poor quality isrpm record might tjyically have around 2.000 clicks per second of recorded material, with lengths ranging from less than 2Ops up t o 4ms. .A click remoi,al system is thus set the task of identifying the position and length of each indiyidual click and then replacing the click with new material in such a way that the listener is not aivare of any discontinuity. Rayner and I'aseghi designed a digital restoration system of this tj.pe in their York a t Cambridge Unii-ersity before 19S9. The techniques used are model-based, assuming that a time-varying auto-regressive (AR) model applies to the audio signal. Detection is automated , identifying significant deviations from the current AR model as clicks. The position and length of each.click is then passed t o an interpolation algorithm. This minimizes the excitation energy over the gap, resulting in a linear least squares estimator for missing samples in terms of correct samples surrounding the gap. is now possible in real-time on modern DSP hardware. The system has been rigorously tested and developed t o such an extent that click removal One limitation of restoration performance is observed when the length of a click becomes large. Visual examination of waveforms shows that the restored signal often does not have enough energy towards the centre of the gap. For many audio signals the effect is visible for scratch lengths greater than 30 samples. Fortunately, the problem is generally not audible until much longer scratches are interpolated, greater than say 100 samples. This phenomenon is a major limiting factor on the maximum number of samples which the process may successfully interpolate.
{"title":"The Detection and Correction of Artefacts in Degraded Gramophone Recordings","authors":"P. Rayner, S. Godsill","doi":"10.1109/ASPAA.1991.634139","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634139","url":null,"abstract":"Tdep h o n e: (0223)-332 767 This paper presents recent developments in techniques for the restoration of audio material degraded by clicks. This type of degradation is associated with all forms of audio media, including CD and D.+IT, but is most characteristic of gramophone disks. The term 'click' covers a wide variety of problems ranging from loud isolated 'pops' to the relatively low level 'craclde' associated with most i s r p m records. The term 'scratch' is also used in this paper t o indicate the same type of degradation. JIost tj.pes of click can be modelled as bursts of corrupted audio samples occuring at random times and of randoni duration. For example, a poor quality isrpm record might tjyically have around 2.000 clicks per second of recorded material, with lengths ranging from less than 2Ops up t o 4ms. .A click remoi,al system is thus set the task of identifying the position and length of each indiyidual click and then replacing the click with new material in such a way that the listener is not aivare of any discontinuity. Rayner and I'aseghi designed a digital restoration system of this tj.pe in their York a t Cambridge Unii-ersity before 19S9. The techniques used are model-based, assuming that a time-varying auto-regressive (AR) model applies to the audio signal. Detection is automated , identifying significant deviations from the current AR model as clicks. The position and length of each.click is then passed t o an interpolation algorithm. This minimizes the excitation energy over the gap, resulting in a linear least squares estimator for missing samples in terms of correct samples surrounding the gap. is now possible in real-time on modern DSP hardware. The system has been rigorously tested and developed t o such an extent that click removal One limitation of restoration performance is observed when the length of a click becomes large. Visual examination of waveforms shows that the restored signal often does not have enough energy towards the centre of the gap. For many audio signals the effect is visible for scratch lengths greater than 30 samples. Fortunately, the problem is generally not audible until much longer scratches are interpolated, greater than say 100 samples. This phenomenon is a major limiting factor on the maximum number of samples which the process may successfully interpolate.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"248 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122123550","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634132
M. Gerzon
Introduction of the apparent localisation of sound images than conventional 2-speaker stereo, especially when sounds must match visual images in direction. This paper describes methods of optimising subjective reproduction from larger numbers n = 3 to 5 of front-stage speakers based on theoretical models for subjective sound localisation, whereby sounds intended for reproduction via n1 stereo speakers are optimally reproduced via a larger number n2 of loudspeakers, and a compatible hierarchy of n-speaker stereo transmission standards is described.
{"title":"Hierarchical Transmission of Multispeaker Stereo","authors":"M. Gerzon","doi":"10.1109/ASPAA.1991.634132","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634132","url":null,"abstract":"Introduction of the apparent localisation of sound images than conventional 2-speaker stereo, especially when sounds must match visual images in direction. This paper describes methods of optimising subjective reproduction from larger numbers n = 3 to 5 of front-stage speakers based on theoretical models for subjective sound localisation, whereby sounds intended for reproduction via n1 stereo speakers are optimally reproduced via a larger number n2 of loudspeakers, and a compatible hierarchy of n-speaker stereo transmission standards is described.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"64 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116814293","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634104
J. Loomis, R. Golledge, R.L. Klastsky
The task of traveling safely and efficiently through a city, town or campus, either on foot or by public transportation, is far more difficult for the visually-impaired traveler than for one who possesses normal vision. The two major tasks confronting the visually impaired traveler are (1) navigation (wayfinding) and (2) avoidance of obstacles along the route. Besides the long-cane, a number of electronic devices have been develtoped to assist in obstacle avoidance; in contrast, general-purpose navigation aids for the visually impaired, similar t:o those used in aircraft, have been considered infeasible because of a lack of a means of determining the traveler's location with sufficient accuracy (on the order of meters). Recently, however, the satellite-based Global Positioning System (GPS) has radically changed the prospects for such a navigation system, for hand-held, relatively low-cost GPS receivers with adequate resolution are now becoming available. We are proposing a portable microcomputer-based personal guidance
{"title":"Personal Guidance System Employing a Virtual Auditory Display","authors":"J. Loomis, R. Golledge, R.L. Klastsky","doi":"10.1109/ASPAA.1991.634104","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634104","url":null,"abstract":"The task of traveling safely and efficiently through a city, town or campus, either on foot or by public transportation, is far more difficult for the visually-impaired traveler than for one who possesses normal vision. The two major tasks confronting the visually impaired traveler are (1) navigation (wayfinding) and (2) avoidance of obstacles along the route. Besides the long-cane, a number of electronic devices have been develtoped to assist in obstacle avoidance; in contrast, general-purpose navigation aids for the visually impaired, similar t:o those used in aircraft, have been considered infeasible because of a lack of a means of determining the traveler's location with sufficient accuracy (on the order of meters). Recently, however, the satellite-based Global Positioning System (GPS) has radically changed the prospects for such a navigation system, for hand-held, relatively low-cost GPS receivers with adequate resolution are now becoming available. We are proposing a portable microcomputer-based personal guidance","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"33 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124936575","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634130
R. Adams
Present-day digital audio systems are based on the well-known Nyquist theorem, which states that a signal may be completely reconstructed from reguarly-spaced samples of that signal as long as the highest frequency in the original signal is less than one-half of the sampling frequency. This paper will show a unique decoding algorithm that can completely reconstruct a signal based on non-uniformly spaced samples of that signal, where the non-uniformity consists of reguarly-spaced missing or incorrect samples. We will show that for this case, the signal may be completely reconstructed if the highest frequency present in the original signal is less than one-half of the "average" sample rate. This algorithm has several potential applications in digital audio systems, such as error concealment and adding a low bit-rate side channel to existing digital recorders or transmission devices. To develop this theory, we start with the following assumption. If a signal that is bandlimited to a frequency wl is applied to a FIR linear-phase lowpass filter with a cutoff frequency of w2 where w2 > wl, then the output signal equals the input signal (with delay) with an accuracy determined by the passband ripple of the low-pass filter. The response of the filter between w l and w2 does not affect the input signal, since the input signal has no energy in this frequency range. Fig. 1 shows this theory graphically. Fig. 2 shows the basic block diagram of the proposed scheme. We start with a sampling operation that is non-uniform in a regular pattern. In this example, we use a sampler that samples for 3 consecutive periods and then skips a sample. This example will be used throughout this paper, and the reader will appreziate that extending the technique to other sampling patterns is straightforward. We will assume that the input signal is bandlimited to < 3/4*(Fs/2), where Fs = l/r and T is the spacing in time between the three consecutive samples. In practice, some gaurd-band is needed to allow for filter transition bands. This non-uniformly sampled signal is then applied to a digital FIR low-pass filter. This filter is a linear-phase filter with passband ripple R and delay D. Note that the input to this filter is a continuously-sampled signal at Fs, where the missing sample has been replaced by a sample of arbitrary value or zero. The decoded output will be derived by a switching between the filtered signal …
{"title":"Reconstruction of Non-Uniformly Sampled Audio Signals","authors":"R. Adams","doi":"10.1109/ASPAA.1991.634130","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634130","url":null,"abstract":"Present-day digital audio systems are based on the well-known Nyquist theorem, which states that a signal may be completely reconstructed from reguarly-spaced samples of that signal as long as the highest frequency in the original signal is less than one-half of the sampling frequency. This paper will show a unique decoding algorithm that can completely reconstruct a signal based on non-uniformly spaced samples of that signal, where the non-uniformity consists of reguarly-spaced missing or incorrect samples. We will show that for this case, the signal may be completely reconstructed if the highest frequency present in the original signal is less than one-half of the \"average\" sample rate. This algorithm has several potential applications in digital audio systems, such as error concealment and adding a low bit-rate side channel to existing digital recorders or transmission devices. To develop this theory, we start with the following assumption. If a signal that is bandlimited to a frequency wl is applied to a FIR linear-phase lowpass filter with a cutoff frequency of w2 where w2 > wl, then the output signal equals the input signal (with delay) with an accuracy determined by the passband ripple of the low-pass filter. The response of the filter between w l and w2 does not affect the input signal, since the input signal has no energy in this frequency range. Fig. 1 shows this theory graphically. Fig. 2 shows the basic block diagram of the proposed scheme. We start with a sampling operation that is non-uniform in a regular pattern. In this example, we use a sampler that samples for 3 consecutive periods and then skips a sample. This example will be used throughout this paper, and the reader will appreziate that extending the technique to other sampling patterns is straightforward. We will assume that the input signal is bandlimited to < 3/4*(Fs/2), where Fs = l/r and T is the spacing in time between the three consecutive samples. In practice, some gaurd-band is needed to allow for filter transition bands. This non-uniformly sampled signal is then applied to a digital FIR low-pass filter. This filter is a linear-phase filter with passband ripple R and delay D. Note that the input to this filter is a continuously-sampled signal at Fs, where the missing sample has been replaced by a sample of arbitrary value or zero. The decoded output will be derived by a switching between the filtered signal …","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"22 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116487034","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634138
S. Vaseghi
L Summary Algorithms are described for the removal of scratches and impulsive noise from archived gramophone tecordhgs. Both the impulsive and the: scratch removal methods are based on a detection-interpolation filtering scheme. This is motivated by the observation that scratches and impulsive noise corrupt only a fraction of the signal, and therefore it is advantageous to detect and locally process only those signal segments which are contaminated. This avoids unnecessary modification and compromise in the quality of noiseless samples. The noise detection and the signal interpolation methods are based on linear prediction Coding (LPC) modelling of acoustic audio signals.
{"title":"Removal of Scratches and Impulsive Noise from Archive Gramophone Records","authors":"S. Vaseghi","doi":"10.1109/ASPAA.1991.634138","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634138","url":null,"abstract":"L Summary Algorithms are described for the removal of scratches and impulsive noise from archived gramophone tecordhgs. Both the impulsive and the: scratch removal methods are based on a detection-interpolation filtering scheme. This is motivated by the observation that scratches and impulsive noise corrupt only a fraction of the signal, and therefore it is advantageous to detect and locally process only those signal segments which are contaminated. This avoids unnecessary modification and compromise in the quality of noiseless samples. The noise detection and the signal interpolation methods are based on linear prediction Coding (LPC) modelling of acoustic audio signals.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"29 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116576020","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634125
J. Kates
Coherence is a measure of the degree to which the output of a system is linearly related to the system input. The signal-to-distortion ratio (S D R) , where the distortion term includes all non-linear effects and noise in the system, can be computed from the coherence. There is growing interest in using coherence to measure distortion in hearing aids and audio systems since the broadband test signal exercises intermodulation as well as harmonic distortion mechanisms in the system under test. The results of using coherence to measure distortion, however, may not be accurate due to biases in the coherence-estimation procedure. For hearing-aid measurements, where the coherence is greater than 0. 5 (S D R > 0 dB) across most of the frequency range, the major source of bias is the group delay of the system under test. The coherence is normally computed by dividing the input and output signals into segments, computing the auto-and cross-spectra for each segment, and averaging the spectra across segments. Delay in the output relative to the input, despite being a linear operation, will reduce the magnitude of the estimated cross-spectrum and thus the estimated coherence and SDR. The amount o f bias i n the coherence estimate depends on the amount of group delay as compared to the segment size. As an example, a simulated hearing-aid response is shown in Fig 1 and the associated group delay in Fig 2. The hearing aid has ideal linear gain up to the input-referred amplifier clipping level of 8 5 dB SPL. The effects of the group delay are visible in the SDR curves of Fig 3, which were computed from the coherence using speech-shaped noise as the excitation, segments of 2048 samples with Hanning windowing and a 50 percent overlap, and a total of 8192 samples at a 20-kHz sampling rate. The magnitude-squared coherence vas smoothed in the frequency domain using one-third octave bandwidths. The curve parameter in Fig 3 is the input signal level in dB SPL. The bias has reduced the SDR in the regions of high group delay, thus limiting the minimum amount of distortion that can be detected at the low input levels. At high input levels, on the other hand, the distortion causes a greater reduction in the SDR than the bias and accurate measurements are obtained. The bias effects can be reduced by using the unbiasing system shown .in Fig …
相干性是对系统输出与系统输入线性相关程度的度量。信号失真比(sdr)，其中失真项包括系统中的所有非线性效应和噪声，可以从相干性中计算出来。由于宽带测试信号在被测系统中需要互调和谐波失真机制，因此人们对使用相干性来测量助听器和音频系统中的失真越来越感兴趣。然而，由于相干估计过程中的偏差，使用相干测量失真的结果可能不准确。对于助听器测量，相干度大于0。5 (S D R > 0 dB)在大多数频率范围内，主要的偏置来源是被测系统的群延迟。相干性的计算通常是通过将输入和输出信号分成几段，计算每段的自动光谱和交叉光谱，并对各段的光谱进行平均。输出相对于输入的延迟，尽管是线性操作，但会降低估计的交叉频谱的幅度，从而降低估计的相干性和SDR。相干估计中的偏置量取决于与段大小相比的组延迟量。作为示例，模拟助听器响应如图1所示，相关组延迟如图2所示。该助听器具有理想的线性增益，可达85 dB SPL的输入参考放大器削波电平。在图3的SDR曲线中可以看到群延迟的影响，该曲线是通过使用语音形状噪声作为激励的相干性，2048个具有汉宁窗和50%重叠的采样段，以及在20 khz采样率下总共8192个样本计算得出的。幅度平方相干性在频域中使用三分之一倍频程带宽进行平滑处理。图3中的曲线参数为以dB SPL为单位的输入信号电平。偏置降低了高群延迟区域的SDR，从而限制了在低输入电平下可以检测到的最小失真量。另一方面，在高输入电平下，失真对SDR的影响比偏置更大，可以获得精确的测量结果。使用图1所示的无偏置系统可以减小偏置效应。
{"title":"Coherence Unbiasing For Hearing-aid Distortion Measurements","authors":"J. Kates","doi":"10.1109/ASPAA.1991.634125","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634125","url":null,"abstract":"Coherence is a measure of the degree to which the output of a system is linearly related to the system input. The signal-to-distortion ratio (S D R) , where the distortion term includes all non-linear effects and noise in the system, can be computed from the coherence. There is growing interest in using coherence to measure distortion in hearing aids and audio systems since the broadband test signal exercises intermodulation as well as harmonic distortion mechanisms in the system under test. The results of using coherence to measure distortion, however, may not be accurate due to biases in the coherence-estimation procedure. For hearing-aid measurements, where the coherence is greater than 0. 5 (S D R > 0 dB) across most of the frequency range, the major source of bias is the group delay of the system under test. The coherence is normally computed by dividing the input and output signals into segments, computing the auto-and cross-spectra for each segment, and averaging the spectra across segments. Delay in the output relative to the input, despite being a linear operation, will reduce the magnitude of the estimated cross-spectrum and thus the estimated coherence and SDR. The amount o f bias i n the coherence estimate depends on the amount of group delay as compared to the segment size. As an example, a simulated hearing-aid response is shown in Fig 1 and the associated group delay in Fig 2. The hearing aid has ideal linear gain up to the input-referred amplifier clipping level of 8 5 dB SPL. The effects of the group delay are visible in the SDR curves of Fig 3, which were computed from the coherence using speech-shaped noise as the excitation, segments of 2048 samples with Hanning windowing and a 50 percent overlap, and a total of 8192 samples at a 20-kHz sampling rate. The magnitude-squared coherence vas smoothed in the frequency domain using one-third octave bandwidths. The curve parameter in Fig 3 is the input signal level in dB SPL. The bias has reduced the SDR in the regions of high group delay, thus limiting the minimum amount of distortion that can be detected at the low input levels. At high input levels, on the other hand, the distortion causes a greater reduction in the SDR than the bias and accurate measurements are obtained. The bias effects can be reduced by using the unbiasing system shown .in Fig …","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"60 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126988253","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634136
R. Frenzel, M. Hennecke
In this paper, the implementation of a compensator for acoustical echoes is presented. The algorithm consists of an adaptive transversal filter, which is adjusted according to a modified version [l] of the wellknown normalized :LMS (NLMS) procedure. Decorrelation filters Rere added to improve the convergence. Beyond that, the stepsize was varied according to the noise level in order to achieve best performance in noisy environments. The paper concludes with some results of realtime measurements of the behavior in typical operating conditions. such as hands-free telephone equipment, demonstrating the performance of the system.
{"title":"A Robust Echo Compensator: Implementation And Realtime Measurements","authors":"R. Frenzel, M. Hennecke","doi":"10.1109/ASPAA.1991.634136","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634136","url":null,"abstract":"In this paper, the implementation of a compensator for acoustical echoes is presented. The algorithm consists of an adaptive transversal filter, which is adjusted according to a modified version [l] of the wellknown normalized :LMS (NLMS) procedure. Decorrelation filters Rere added to improve the convergence. Beyond that, the stepsize was varied according to the noise level in order to achieve best performance in noisy environments. The paper concludes with some results of realtime measurements of the behavior in typical operating conditions. such as hands-free telephone equipment, demonstrating the performance of the system.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"10 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115165846","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634110
J. Herre, E. Eberlein, K. Brandenburg
InjJOdWiQn Low bit rate coding of high quality digital audio uses perceptual criteria to shape the quantization noise. [ 11 is an example for such an algorithm. Modelling of the hearing process is necessary to get knowledge about the required noise shaping. Such models used to estimate the actual hearing threshold of the human ear and in this way determine the e m r limit that must not be exceeded for a transparent coding of the signal. Traditional perceptual models consider rnasking effects which state that under certain circumstances small signals cannot be detected by the listener in the presence of a 1ar;ge signal, that they have been "masked". The masking depends on the signal's spectral characteristics and its structure in time. Up to now the dependencies of some parameters are research topics. One example is the local predictability of a signal, also hown as 'tonality' ([2]) which has a strong influence on the masking ability of a signal. This paper presents a useful tool for psychoacoustic research: The Real Time Perceptual Threshold Simulator.
{"title":"A Real Time Perceptual Threshold Simulator","authors":"J. Herre, E. Eberlein, K. Brandenburg","doi":"10.1109/ASPAA.1991.634110","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634110","url":null,"abstract":"InjJOdWiQn Low bit rate coding of high quality digital audio uses perceptual criteria to shape the quantization noise. [ 11 is an example for such an algorithm. Modelling of the hearing process is necessary to get knowledge about the required noise shaping. Such models used to estimate the actual hearing threshold of the human ear and in this way determine the e m r limit that must not be exceeded for a transparent coding of the signal. Traditional perceptual models consider rnasking effects which state that under certain circumstances small signals cannot be detected by the listener in the presence of a 1ar;ge signal, that they have been \"masked\". The masking depends on the signal's spectral characteristics and its structure in time. Up to now the dependencies of some parameters are research topics. One example is the local predictability of a signal, also hown as 'tonality' ([2]) which has a strong influence on the masking ability of a signal. This paper presents a useful tool for psychoacoustic research: The Real Time Perceptual Threshold Simulator.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"30 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116178636","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
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