Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics最新文献

{"title":"Audio Compression and Echo Cancellation for Low Bit Rate VideoTeleconferencing","authors":"P. Chu","doi":"10.1109/ASPAA.1991.634116","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634116","url":null,"abstract":"The constraints for audio compression and echo cancellation in low bit rate videoteleconferencing (56 to 128 thousand bits per second) difeer in many important respects from audio-only teleconferencing. Algorithms which have been optimized for audio-only communications may be improved significantly or require significant improvement. In this talk, audio requirements will be presented, and we shall demonstrate with a tape how PictureTel has addressed these problems in their latest commercial product. A summary of the talk follows.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132870010","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}

{"title":"A real-time Spline/Wavelet Signal Analyzer","authors":"C. Chui, A. Chan","doi":"10.1109/ASPAA.1991.634107","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634107","url":null,"abstract":"Splines and Spline-Wavelets of order m differ from any other functions in that they are uniquely determined by their (mlist order derivatives that are staircase waveforms (piecewise constants) of finite length. Through this important observation, we are able to build a general purpose spline/wavelet signal analyzer. A patent application which includes this class of analyzers hais been filed with the U.S. government recently [ 11. Since the mth order 13-spline and splinew,avelet (B-wavelet) have compact supports (i.e., finite duration) andthe B--wavelets are symmetric or antisymmetric depending on m being even or odd [2,3], our signal analyzer is essentially distortion free. An input analog signal is digitized and mapped into a spline: signal space (a subspace of L2 in which signals are represented by spline functions) of specific order and sulfficiently fine grid. This mapping is done by using an FIR filter based on a local cardinal interpolation method developed in [4,5]. Accordingly, the wavelet decomposition and reconstruction algorithms [2,3,6] can be applied in parallel to separate the signal into different filequency bands for different processing purposes. The result is similar to one obtained by the multi-channel filter bank method.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125506089","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}

{"title":"Adaptive Beamformer Performance In Reverberation","authors":"J. Greenberg, P. Zurek","doi":"10.1109/ASPAA.1991.634121","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634121","url":null,"abstract":"Introduction. Recent applications of adaptive filtering to hearing aids [4-81 have shown that very simple two-microphone systems can provide large improvements in target-to-jammer ratio under anechoic conditions. Some of these studies also considered non-anechoic conditions and showed that the presence of reverberation has a strong effect on performance. Because of differences among the acoustic and signal-processing conditions of these studies, a more detailed summary cannot be given. The present study illustrates , for a particular adaptive beamformer, the effects of reverberation on performance and the interactions among reverberation, target-to-jammer ratio (TJR) and filter length (L). System Description and Methods. The system used here (Figure 1) is a modified, two-microphone version of the Griffiths-Jim (21 constnained adaptive beamformer. The modifications were developed to deal with the problems of misadjustnient and misalignment at high TJRs and do so by exploiting the fluctuations in speech to allow adaptation during intervals of low TJR [l]. The first method employs the input correlation p between bandpass-filtered microphone signals as a measure of TJR and inhibits adaptation when p exceeds a threshold. The second method includes output power in the normalization of the weight update: Aw&z + 1) = 2ay[n]d[n-k]/{L(P,,[n] + Pd[n])}, where Py and P d are running estimates of power in the system output and the adaptive filter' The study employed computer simulations of this beamformer with 7-cm spacing between microphones in free-space. Input signals were generated by convolving single-talker target and babble jammer sources with synthetic source-to-microphone impulse responses [3]. For all conditions, the target was located at 0\", broadside to the array, the jammer was at 45\", and both sources were 0.9 m from the center of the array. Output target and jammer were measured separately through use of a master and two slave processors. The master processed target and jammer summed together, while the slave systems processed the target and jammer separately using adaptive filter weights copied from the master. The performance metric, GI, is a spectrally-weighted gain in target-to-jammer ratio from input to output, measured in the steady-state [6]. Results. A sampling of rooms and source/array geometries was simulated to study the joint effects of TJR, degree of reverberation, and filter length. Condition A employed a room with dimensions 5.2 x 3.4 x 2.8 mtders and a uniform absorption coefficient of 1.0 (anechoic), 0.6, or 0.2, resulting in direct-to-reverberant energy ratios at the array of 00, 5.7,oI-2.4 dB, respectively. The …","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125913282","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}

{"title":"Determination of Musical Meter using the Method of Autocorrelation","authors":"J.C. Brown","doi":"10.1109/ASPAA.1991.634147","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634147","url":null,"abstract":"q u a n t i t i e s of d a t a and a t t e m p t i n g t o d e t e r m i n e by c o m p u t a t i o n t h e q u a n t i t i e s which a p p e a r s o e a s i l y o b t a i n e d by human b e i n g s. One s u c h q u a n t i t y i s m u s i c a l meter.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116633704","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}

{"title":"DSP-implementations of speech coding for multielectrode cochlear implants and multiband loudness correction for digital hearing aids","authors":"N. Dillier, H. Bogli, T. Frohlich, M. Kompis","doi":"10.1109/ASPAA.1991.634127","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634127","url":null,"abstract":"SUMMARY Hearing sensations can be restored for profoundly deaf patients via artificial electrical stimulation of the auditory nerve. Present electrode technology and electrophysiological constraints however allow at best a very crude and limited approximation of the normal neural excitation pattern. Signal processing for cochlear implants therefore is confronted with the problem of a severely restricted channel capacity and the necessity to select and encode a subset of the information contained in the sound signal reaching the listeners ear. With single chip digital signal processors (DSPs) incorporated in personal computers different speech coding strategies can be evaluated in relatively short laboratory experiments. In addition to the well known strategies realized with filters, amplifiers and logic circuits a DSP approach allows the implementation of much more complex algorithms such as nonlinear multiband loudness correction, speech feature contrast enhancement, adaptive noise reduction. Although many aspects of speech encoding can be efficiently studied using a laboratory digital signal processor it would be desirable to allow subjects more time for adjustment to a new coding strategy. Several days or weeks of habituation are sometimes required until a new mapping can be fully exploited. Thus for scientific as well as practical purposes the miniaturization of wearable DSPs will be of great importance. A cochlear implant digital speech processor (CIDSP) for the Nucleus 22-channel cochlear prosthesis has been implemented using a single chip digital signal processor (TMS320C25, Texas Instruments). For laboratory experiments the CIDSP is incorporated in a general purpose computer (PDP11/73) which provides interactive parameter control, graphical display of input/output and intermediate buffers and offline speech file processing facilities. In addition to the generation of stimulus parameters for the cochlear implant an acoustic signal based on a perceptive model of auditory nerve stimulation is output simultaneously. For field studies and as a take-home device for patients a wearable battery-operated unit has been built. Advantages of a DSP-implementation of speech encoding algorithms as opposed to offline prepared tcst lists are increased flexibility, controlled, reproducible and fast modifications of processing parameters, use of running speech for training and familiarization. Disadvantages are the more complex programming and numerical problems with integer arithmetic.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131384399","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}

{"title":"Narrowband Adaptive Acoustic Arrays For Directional Interference Nulling","authors":"V. DeBrunner, A. Beex","doi":"10.1109/ASPAA.1991.634120","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634120","url":null,"abstract":"Extended Summary We consider the design and performance of a restricted geometry, narrowband, adaptive 2-element (termed \"small\") acoustic array as used for directional hearing enhancement. An array is designed to be mobile and burden-free, so that the wearer is not encumbered by the harldware, while remaining usehl in noisy environments. The directionality of multi-element arrays, and thus the capability for interference rejection, is greatly superior to that possible with a single-element device. Enhanced directionality comes from the extra knowledge gained when acoustic signals are spatially sampled. Intuitively, we expect such a result since humans have 2 ears to hear with, and we do not have the \"extra\" one merely for redundancy. This increased knowledge comes at the expense of increased hardware requirements, as well as an increase in real-time computations and communications. We explore the balance between directionality improvlement and hardware requirements. Making the array adaptive is shown to enhance the array directionality above that achievable by a iixed array which we have examined previously. We examine the perf'ornnance of small, adaptive, nonlinear acoustic arrays for","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115543435","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}

{"title":"Stability Analysis of an Active Periodic Noise Control System","authors":"A. Barbosa, S. Kuo","doi":"10.1109/ASPAA.1991.634145","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634145","url":null,"abstract":"Abstract: This paper presents or stabiliry analysis of an active periodic noise control (APNC) system using an internally generated impulse train with the same fundamental frequency as the noise to be canceled as reference input. The stability analysis is made through the use of polar plots and shows how the compensation of the input to the LMS adaptation cfiltered-X) can improve stability of the APNC system. This polar plot also shows the relationship between the stability of the system and step size p andjilter order hr.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123205902","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}

{"title":"An Adaptive Feedback Equalization Algorithm For Digital Hearing Aids","authors":"Maynard Engebretson, Michael P. OConnell, Fengmin Gong","doi":"10.1109/ASPAA.1991.634124","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634124","url":null,"abstract":"A method is described for adaptively equalizing the ubiquitous feedback path of a hearing aid in order to stabilize the system. The algorithm utilizes an LMS adaptive filter and is implemented in digital form. An additional 10 to 15 dB of stable gain margin has been demonstrated. INTRODUCTION System instability is a commonly cited problem with regard to highpower hearing aids, where it is desirable to achieve high acoustic gains, and with intheear devices, where acoustical and mechanical isolation between input and output is difficult to achieve. Instability is a result of feedback due to 1) acoustic leakage around the earmold and through the vent in the earmold and 2) mechanical coupling between receiver and microphone As is well known, when the open loop gain of a system with feedback is greater than unity and has a phase which is a multiple of 2n radians, the system will oscillate [l], thereby causing a serious degradation of signal quality. In addition, if the open loop gain is close to but less than unity, the system response will be highly underdamped and will exhibit a response sharply divergent from the desired frequency-gain characteristic prescribed for the hearing-impaired patient. Current methods for reducing hearing aid instability are limited to the use of tightly fitting m o l d s . However, this is difficult to achieve without causing discomfort for the patient. A number of methods of feedback suppression have been proposed. For example, Egolf and Larson [2] have studied two methods, one, a time delay notch filter system and, two, an active feedback cancellation system. They report improvements of between 6 and 8 dB in closed loop gain margin with both approaches and, if conditions are carefully controlled, up to 15 to 20 dJ3 [3]. The algorithm described herein, is similar to the active feedback cancellation system, and stabilizes the hearing aid by adaptively cancelling its feedback path. Since the algorithm is adaptive, it can accommodate to changes in the feedback characteristic of the hearing aid. Equalization is accomplished with a Widrow LMS adaptive filter [4]. The adaptive process is driven by an internally generated pseudorandom signal presented at threshold and subthreshold levels similar to that used by Schroeder [5]. The algorithm has been refined for implementation on small digital processing structures. THE FEEDBACK EQUALIZATION MODEL The equalized hearing aid model is shown in the figure where Hm and Hr represent the microphone and receiver characteristics, respectively, Hf represents the undesirable acoustic and mechanical feedback paths, H represents a filter function that when multiplied by Hm and Hr yields the prescribed acoustic fkequencygain function for the patient, and & represents the adaptive equalization filter. X, Y, and N represent the input sound pressure at the hearing aid microphone, the sound pressure in the ear canal, and the pseudorandom probe signal, respectively. The closed-loop transfer charact","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123047254","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}

{"title":"Time-frequency And Multiple-resolution Representations In Auditory Modeling","authors":"U. Laine, M. Karjalainen, T. Altosaar","doi":"10.1109/ASPAA.1991.634095","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634095","url":null,"abstract":"The human auditory system is known to utilize different temporal and frequency re,solutions in different contexts and analysis phases. In this paper we discuss some aspects of using time-frequency representations and multiple resolutions in auditory modeling from an information and signal theoretic point of view. The first question is how to allocate resolution optimally between frequency and time. For this purpose a new method called the FAM tranSform is described. The other question is how to utilize multiple parallel and redundant resolutions to avoid some problems that are faced when using single resolution approaches.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130012203","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}

{"title":"Adaptive Prediction With Transform Domain Quantization For Low-rate Audio Coding","authors":"B. Bhaskar","doi":"10.1109/ASPAA.1991.634113","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634113","url":null,"abstract":"The technique of Adaptive Predictive Coding with Transform domain Quantization (APCTQ) has been studled for high quality coding of audio signals, at rates in the range 1.5 2.5 bit/sample. This technique is a combination of time domain predictive methods with transform domain quantization methods. Near transparent quality performance has been obtained at 5 kHz and 7.5 kHz bandwidths at rates of 24 kbit/s and 32 kbit/s respectively. In general, the APC-TQ technique is applicable over a wide range of signal bandwidths from 5 kHz to 20 kHz, for efficient transmission of high quality audio signals for applications such as direct audio broadcasing.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134229406","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}