This paper gives a summary of the MPEG/audio compression algorithm. This algorithm was developed by the Motion Picture Experts Group (MPEG), as an International Organization for Standardization standard for the high fidelity compression of digital audio. The MPEG/audio compression standard is one part of a multiple part standard that addresses the compression of video (11172-2), the compression of audio (11172-3), and the synchronization of the audio, video, and related data streams (11172-1) to an aggregate bit rate of about 1.5 Mbit/sec. The MPEG/audio standard also can be used for audio-only applications to compress high fidelity audio data at much lower bit rates. While the MPEG/audio compression algorithm is lossy, often it can provide `transparent', perceptually lossless, compression even with compression factors of 6-to-1 or more. The algorithm works by exploiting the perceptual weaknesses of the human ear. This paper also will cover the basics of psychoacoustic modeling and the methods used by the MPEG/audio algorithm to compress audio data with least perceptible degradation.
{"title":"An Overview of the MPEG/audio Compression Algorithm","authors":"D. Pan","doi":"10.1117/12.174960","DOIUrl":"https://doi.org/10.1117/12.174960","url":null,"abstract":"This paper gives a summary of the MPEG/audio compression algorithm. This algorithm was developed by the Motion Picture Experts Group (MPEG), as an International Organization for Standardization standard for the high fidelity compression of digital audio. The MPEG/audio compression standard is one part of a multiple part standard that addresses the compression of video (11172-2), the compression of audio (11172-3), and the synchronization of the audio, video, and related data streams (11172-1) to an aggregate bit rate of about 1.5 Mbit/sec. The MPEG/audio standard also can be used for audio-only applications to compress high fidelity audio data at much lower bit rates. While the MPEG/audio compression algorithm is lossy, often it can provide `transparent', perceptually lossless, compression even with compression factors of 6-to-1 or more. The algorithm works by exploiting the perceptual weaknesses of the human ear. This paper also will cover the basics of psychoacoustic modeling and the methods used by the MPEG/audio algorithm to compress audio data with least perceptible degradation.","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":"1994-05-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126514097","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 : 1993-09-01DOI: 10.1109/ASPAA.1991.634135
M. Mohan Sondhi, D. Morgan
In long‐distance telephony, echoes arise due to impedance mismatches at various points in the telephone circuit. Adaptive line echo cancelers have been used successfully for over a decade to combat this problem. Echoes also arise in teleconferencing, due to acoustic coupling between microphone and loudspeaker in each conference room. This problem is similar to the line echo problem; however, the echo paths are much longer and much more variable in this case. In this paper a further complication that arises if stereophonic transmission is used for teleconferencing is discussed: There is an inherent nonuniqueness in estimating the echo paths. It appears that the only way to resolve this nonuniqueness is by somehow decorrelating the signals in the two stereo channels. Several methods of decorrelation are discussed and how they affect adaptive echo canceller performance as well as stereophonic perception is shown.
{"title":"Acoustic Echo Cancellation for Stereophonic Teleconferencing","authors":"M. Mohan Sondhi, D. Morgan","doi":"10.1109/ASPAA.1991.634135","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634135","url":null,"abstract":"In long‐distance telephony, echoes arise due to impedance mismatches at various points in the telephone circuit. Adaptive line echo cancelers have been used successfully for over a decade to combat this problem. Echoes also arise in teleconferencing, due to acoustic coupling between microphone and loudspeaker in each conference room. This problem is similar to the line echo problem; however, the echo paths are much longer and much more variable in this case. In this paper a further complication that arises if stereophonic transmission is used for teleconferencing is discussed: There is an inherent nonuniqueness in estimating the echo paths. It appears that the only way to resolve this nonuniqueness is by somehow decorrelating the signals in the two stereo channels. Several methods of decorrelation are discussed and how they affect adaptive echo canceller performance as well as stereophonic perception is shown.","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":"1993-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132387416","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 : 1991-10-01DOI: 10.1109/ASPAA.1991.634123
C. N. Canagarajah, P. Rayner
One of the main problems of the sensorineural hearing impaired listeners is the partial or complete loss of frequency selectivity. It is now well established that most of the auditory perceptual features are very well represented in the ear by the spectrum of the incoming sound signal. Thus the loss of frequency selectivity means that it is difficult for the impaired listener to discriminate between two sounds or to understand speech in a noisy environment. This handicap is referred to as the cocktail party eflect. It is widely accepted, and proven by the experiments carried out on impaired listeners, that one of the main causes for this impairment is the broad and tilted auditory filter shapes in the damaged cochlea compared to an undamaged normal ear. As a result, in noisy surroundings these broad filters allow more noise than a normal ear making detection of signal in noise difficult. Therefore to improve intelligibility a hearing aid must, not only suppress the noise in speech but also alleviate the problems of reduced frequency selectivity. There are a few hearing aids proposed in the literature to enhance speech in noise. Most of them are based on Adaptive noise cancellation or Adaptive beamforming principles. They have proved to be very useful in situations where there are few noise sources or when there is a reference noise available. Very often the environment contains many uncorrelated noise sources effectively creating a diffusive noise source. Hence obtaining a reference noise signal that is correlated with the noise in the other inputs is impossible. In these situations the above methods produce very little speech enhancement. There are many conventional single-input systems to suppress noise but like the multi-microphone methods mentioned above, they have proved to be of very little use in increasing the intelligibility of the speech for the hearing impaired. In this paper we illustrate how a single-input system incorporating the auditory perceptual features could be employed to increase intelligibility in hearing aids. Spectral Subtraction (SS) is an efficient way of reducing noise in single-input systems. In this method an estimate of the magnitude spectrum of the noise, #(U), is obtained during nonspeech activity and is subtracted from the magnitude spectrum of the noisy speech, X(w), to obtain the enhanced speech, S(u). This performs satisfactorily when the noise source is stationary. The main drawback of this system is it does not consider the problems of the hearing impaired and as a result is of very little benefit to them. Furthermore it introduces a residual or mwacal nobe in the processed speech. It is shown in this paper that by incorporating the perceptual features like masking and excitation patterns the above problems can be eliminated. The technique proposed here, firstly transforms the power (not magnitude) spectrum of the noisy speech (X(w)) into auditory excitation patterns, E(w). The auditory system consists of a
{"title":"A Single-Input Hearing Aid Based on the Auditory Perceptual Features to Improve Speech Intelligibility in Noise","authors":"C. N. Canagarajah, P. Rayner","doi":"10.1109/ASPAA.1991.634123","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634123","url":null,"abstract":"One of the main problems of the sensorineural hearing impaired listeners is the partial or complete loss of frequency selectivity. It is now well established that most of the auditory perceptual features are very well represented in the ear by the spectrum of the incoming sound signal. Thus the loss of frequency selectivity means that it is difficult for the impaired listener to discriminate between two sounds or to understand speech in a noisy environment. This handicap is referred to as the cocktail party eflect. It is widely accepted, and proven by the experiments carried out on impaired listeners, that one of the main causes for this impairment is the broad and tilted auditory filter shapes in the damaged cochlea compared to an undamaged normal ear. As a result, in noisy surroundings these broad filters allow more noise than a normal ear making detection of signal in noise difficult. Therefore to improve intelligibility a hearing aid must, not only suppress the noise in speech but also alleviate the problems of reduced frequency selectivity. There are a few hearing aids proposed in the literature to enhance speech in noise. Most of them are based on Adaptive noise cancellation or Adaptive beamforming principles. They have proved to be very useful in situations where there are few noise sources or when there is a reference noise available. Very often the environment contains many uncorrelated noise sources effectively creating a diffusive noise source. Hence obtaining a reference noise signal that is correlated with the noise in the other inputs is impossible. In these situations the above methods produce very little speech enhancement. There are many conventional single-input systems to suppress noise but like the multi-microphone methods mentioned above, they have proved to be of very little use in increasing the intelligibility of the speech for the hearing impaired. In this paper we illustrate how a single-input system incorporating the auditory perceptual features could be employed to increase intelligibility in hearing aids. Spectral Subtraction (SS) is an efficient way of reducing noise in single-input systems. In this method an estimate of the magnitude spectrum of the noise, #(U), is obtained during nonspeech activity and is subtracted from the magnitude spectrum of the noisy speech, X(w), to obtain the enhanced speech, S(u). This performs satisfactorily when the noise source is stationary. The main drawback of this system is it does not consider the problems of the hearing impaired and as a result is of very little benefit to them. Furthermore it introduces a residual or mwacal nobe in the processed speech. It is shown in this paper that by incorporating the perceptual features like masking and excitation patterns the above problems can be eliminated. The technique proposed here, firstly transforms the power (not magnitude) spectrum of the noisy speech (X(w)) into auditory excitation patterns, E(w). The auditory system consists of a ","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":"1991-10-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121661588","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 : 1991-09-24DOI: 10.21437/Eurospeech.1991-300
Tim Aarset, B. Gold
Abstract : Two pitch perception modeling algorithms are described. The first algorithm models periodicity pitch perception, and the second algorithm models place pitch perception. The two models are now applied to various psychoacoustic stimuli. Both periodicity and place models yield results that are in general agreement with psychoacoustic measurements for the missing fundamental and for inharmonic stimuli. The place algorithm proved to be a better approximation than periodicity for processing comb-filtered noise. Periodicity was more successful for periodic pulse train stimuli.
{"title":"Models of Pitch Perception","authors":"Tim Aarset, B. Gold","doi":"10.21437/Eurospeech.1991-300","DOIUrl":"https://doi.org/10.21437/Eurospeech.1991-300","url":null,"abstract":"Abstract : Two pitch perception modeling algorithms are described. The first algorithm models periodicity pitch perception, and the second algorithm models place pitch perception. The two models are now applied to various psychoacoustic stimuli. Both periodicity and place models yield results that are in general agreement with psychoacoustic measurements for the missing fundamental and for inharmonic stimuli. The place algorithm proved to be a better approximation than periodicity for processing comb-filtered noise. Periodicity was more successful for periodic pulse train stimuli.","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":"1991-09-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127099508","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 : 1991-09-24DOI: 10.21437/Eurospeech.1991-88
H. Hermansky, L. Cox
A common wisdom in speech re-synthesis is that while the vocal tract excitation can be modified to represent the message prosody, the accurate preservation of the formants is needed in order to ensure that both the linguistic message and the speaker-dependent information is well represented in the synthesized speech. Formants are speaker-dependent. A further decomposition of the formant-based speech representation into its message-bearing and the speaker-dependent parts and the inverse problem of combining those two sources of speech information is of interest. The current paper addresses this issues.
{"title":"Perceptual Linear Predictive (PLP) Analysis-Resynthesis Technique","authors":"H. Hermansky, L. Cox","doi":"10.21437/Eurospeech.1991-88","DOIUrl":"https://doi.org/10.21437/Eurospeech.1991-88","url":null,"abstract":"A common wisdom in speech re-synthesis is that while the vocal tract excitation can be modified to represent the message prosody, the accurate preservation of the formants is needed in order to ensure that both the linguistic message and the speaker-dependent information is well represented in the synthesized speech. Formants are speaker-dependent. A further decomposition of the formant-based speech representation into its message-bearing and the speaker-dependent parts and the inverse problem of combining those two sources of speech information is of interest. The current paper addresses this issues.","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":"1991-09-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130814242","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 : 1991-09-01DOI: 10.1109/ASPAA.1991.634117
C. Grewin, T. Rydén
Subjective assessments or listening tests have always been an important part in the evaluation of audio equipment, maybe even more so today. For certain types of digital audio equipment there are no adequate methods of objective measurements available. This is certainly true for advanced bitrate reduction systems. Subjective assessments therefore play a very important role in the choice of a source coding algorithm for DAB, Digital Audio Broadcasting.
{"title":"Subjective Assessments Of Low Bit-rate Audio Codecs","authors":"C. Grewin, T. Rydén","doi":"10.1109/ASPAA.1991.634117","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634117","url":null,"abstract":"Subjective assessments or listening tests have always been an important part in the evaluation of audio equipment, maybe even more so today. For certain types of digital audio equipment there are no adequate methods of objective measurements available. This is certainly true for advanced bitrate reduction systems. Subjective assessments therefore play a very important role in the choice of a source coding algorithm for DAB, Digital Audio Broadcasting.","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":"1991-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126342060","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.1991.634142
Stuart J Fxockton
Input mise characteristics Knowledge of the geometry of the error surface is essential to the understanding of any adaptive system, and especially for one using any form of gradient descent algorithm. Most adaptive active noise control systems in the published literature use one form or another of the LMS adaptive algorithm (either in its standard non-recursive form [ 13 or in Feintuch's extension to the recursive form [2]). Because this algorithm is an approximation to a steepest descent algorithm its performance is very strongly affwted by the {gradient of the error surface and if the eigenvalues of the performance surface have substantially differing magnitudes the cmnvergence rate that can be achieved is poor. The comparative simplicity of implementation of the algorithm, however, has so far been sufficient to make it the preferred candidate i n real systems. Number of Linearity of Feedback Acoustic control transmission from reverberation channels paths secondary present sowce(s) to &tector(s) It is a problem with many real systems that the dimensionality of the error surface is so great as to make it rather difficult to perceive their character. However in many cases the essence of the system can be captured using a grossly simplified model with only a few coefficients in the adaptive system (and hence an error surface whose dimension is reasonably small). Input mise characteristics sinusoidal quasi-stationary periodic non-stationary random The following parameters may be used to separate active noise control systems into classes having different complexities. The variety of these classes is indicated in the following (rather arbitrary) table; in each case the complexity will generally increase from top to bottom of a column. Each column is of course independent of all the others so the table indicates that there are perhaps 324 significantly different complexities of active noise control system. Number of Linearity of Feedback Acoustic control transmission from reverberation channels paths secondary present sowce(s) to &tector(s)
{"title":"Error surfaces and adaption behaviour of active noise control systems","authors":"Stuart J Fxockton","doi":"10.1109/ASPAA.1991.634142","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634142","url":null,"abstract":"Input mise characteristics Knowledge of the geometry of the error surface is essential to the understanding of any adaptive system, and especially for one using any form of gradient descent algorithm. Most adaptive active noise control systems in the published literature use one form or another of the LMS adaptive algorithm (either in its standard non-recursive form [ 13 or in Feintuch's extension to the recursive form [2]). Because this algorithm is an approximation to a steepest descent algorithm its performance is very strongly affwted by the {gradient of the error surface and if the eigenvalues of the performance surface have substantially differing magnitudes the cmnvergence rate that can be achieved is poor. The comparative simplicity of implementation of the algorithm, however, has so far been sufficient to make it the preferred candidate i n real systems. Number of Linearity of Feedback Acoustic control transmission from reverberation channels paths secondary present sowce(s) to &tector(s) It is a problem with many real systems that the dimensionality of the error surface is so great as to make it rather difficult to perceive their character. However in many cases the essence of the system can be captured using a grossly simplified model with only a few coefficients in the adaptive system (and hence an error surface whose dimension is reasonably small). Input mise characteristics sinusoidal quasi-stationary periodic non-stationary random The following parameters may be used to separate active noise control systems into classes having different complexities. The variety of these classes is indicated in the following (rather arbitrary) table; in each case the complexity will generally increase from top to bottom of a column. Each column is of course independent of all the others so the table indicates that there are perhaps 324 significantly different complexities of active noise control system. Number of Linearity of Feedback Acoustic control transmission from reverberation channels paths secondary present sowce(s) to &tector(s)","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":"116378892","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.1991.634133
P. Lacombe
Multimedia is coming of age, and 16 bit, 44.1kHz audio will soon be as common as the mouse. The IRIS Indigo represents the first Advanced Computing Environment (ACE) compatible workstation (ACE consortium members at last count was greater than forty). There are still many issues to be resolved, like standard file format:;, libraries, and synchronization, etc.. However the tools exist, and the open systems environment permits the industry to leverage technology through combined efforts. This paper presents one such environment. Outline IRIS Indigo architecture, followed by brief descriptions of the Audio and Graphics libraries, concluding with two audio applications presented as a case study in software development Csoundm and Scrub. Hardware The IRIS Indigo is a MIPS R3000/R3010 based, 56001 coprocessor, graphics/audio workstation (biendiari). The audio subsystem consists of a 56001, 32K x 24 bits SRAM, 16-bit stereo 64x oversampling delta-sigma ADC, ‘18-bit stereo 8x oversampled DAC, third-order filtering, MDAC attenuator software controlled, and IEC958/AES3 digital I/O. Supported sampling rates are 29.4, 32, 44.1, 48kHz, and any of these divided by integers 2 through 8. Audio Library 1.0 The basic construct implemented in release 1.0 of the Audio Library are audio ports. Programmers open ports to listen or generate sounds. These ports have intermediate buffers to relax real-time O/S and program requirements. Audio ports may be configured to different buffer sizes, sample widths, number of channels (1,2) and two configuration management calls. Hardware state parameters control the ports sampling rate, gain and input source. The audio library is currently designed around goals similar to our early implementations of the Graphics Library. Simplicity So the programmer can quickly learn how to use it. Completeness If the hardware can do it,. the library should let you. Efficiency Close enough to the metal to be optimum, yet not stifle hardware evolution. Graphics Library 4.0 The Graphics Library has evolved over the past ten years, at least eight graphics architectures (known to the author), and a CISC to RISC migration, with major efforts placed on minimizing obsolete Csound is a trademark of MIT functions. There are currently over 300 functions ranging from drawing primitives, text, modeling transformations, and performing raster operations, to the more esoteric functions like alpha-blending, fog and haze, lighting, NURBS, stencil planes, overlays, underlays, accumulation buffer, and zbuffer support.
{"title":"IRIS Indigo Audio and Graphics Programming Environment A Case Study using Csouncl and Scrub","authors":"P. Lacombe","doi":"10.1109/ASPAA.1991.634133","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634133","url":null,"abstract":"Multimedia is coming of age, and 16 bit, 44.1kHz audio will soon be as common as the mouse. The IRIS Indigo represents the first Advanced Computing Environment (ACE) compatible workstation (ACE consortium members at last count was greater than forty). There are still many issues to be resolved, like standard file format:;, libraries, and synchronization, etc.. However the tools exist, and the open systems environment permits the industry to leverage technology through combined efforts. This paper presents one such environment. Outline IRIS Indigo architecture, followed by brief descriptions of the Audio and Graphics libraries, concluding with two audio applications presented as a case study in software development Csoundm and Scrub. Hardware The IRIS Indigo is a MIPS R3000/R3010 based, 56001 coprocessor, graphics/audio workstation (biendiari). The audio subsystem consists of a 56001, 32K x 24 bits SRAM, 16-bit stereo 64x oversampling delta-sigma ADC, ‘18-bit stereo 8x oversampled DAC, third-order filtering, MDAC attenuator software controlled, and IEC958/AES3 digital I/O. Supported sampling rates are 29.4, 32, 44.1, 48kHz, and any of these divided by integers 2 through 8. Audio Library 1.0 The basic construct implemented in release 1.0 of the Audio Library are audio ports. Programmers open ports to listen or generate sounds. These ports have intermediate buffers to relax real-time O/S and program requirements. Audio ports may be configured to different buffer sizes, sample widths, number of channels (1,2) and two configuration management calls. Hardware state parameters control the ports sampling rate, gain and input source. The audio library is currently designed around goals similar to our early implementations of the Graphics Library. Simplicity So the programmer can quickly learn how to use it. Completeness If the hardware can do it,. the library should let you. Efficiency Close enough to the metal to be optimum, yet not stifle hardware evolution. Graphics Library 4.0 The Graphics Library has evolved over the past ten years, at least eight graphics architectures (known to the author), and a CISC to RISC migration, with major efforts placed on minimizing obsolete Csound is a trademark of MIT functions. There are currently over 300 functions ranging from drawing primitives, text, modeling transformations, and performing raster operations, to the more esoteric functions like alpha-blending, fog and haze, lighting, NURBS, stencil planes, overlays, underlays, accumulation buffer, and zbuffer support.","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":"129341398","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.1991.634137
K. Gork, D. Guicking
In many situations one h i s to resolve an unknown number of overlapping and noisy echos of a signal. For this problem a numerical procedure 111 was developed which works as a modified version of the MUSIC algorithm [2,3]. The shape of the signal s’(t) must be known to estimate the number D and the t:ime delays fd of the echos. If K independent records are collected, all differing in the amplitudes mk,d of the echos qt fd) and the noise realization < k ( t ) , then the received and sampled waveforms can be written as
在许多情况下，人们需要解决未知数量的重叠和噪声回波的信号。针对这一问题，开发了一个数值过程111，作为MUSIC算法的修改版本[2,3]。必须知道信号s ' (t)的形状，才能估计回波的数量D和时间延迟fd。如果收集到K条独立的记录，回声的振幅mk,d qt fd)不同，噪声实现< K (t)，则接收到的和采样到的波形可以写成
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Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634146
H. Nam, S. Elliott
The characteristics of :most active control systems change with time. In particular, the characteristics of the transfer functions between the secondary loudspeakers and error slensors (the "secondary path") can be time-varying. In many situations, an adaptive scheme to estimate these transfer functions is needed. This is in addition to the adaptive filter implementing the controller. Most adaptive control filters have used FIR structures based on filtered-x LMS algorithms. :Recently, Eriksson er al [ 11 showed that IIR structures are more desirable for the active control of duct noise in order to remove the poles introduced by the acoustic feedback and presented an algorithm to adjust the coefficients of an IIR filter using the recursive least mean square: (RLMS) algorithm of Feintuch [2]. Since both of these approaches require knowledge of the secondary path transfer function, some adaptive algorithms which simultaneoiisly estimate the transfer function of a secondary path have been presented [1,3]. Such adaptive techniques have a tendency to diverge when the parameters vary rapidly and it is difficullt to apply them to the multiple sensor multiple speaker cases [4] because there are too many parameters to be estimated in each step. We present a new algorithm using multiple models to reduce the tendency to diverge compared with previous adaptive algorithms under time-varying conditions. Since this approach requires only a small amount of computation, it may also be used in the multiple channel case. The block diagxim of the multiple model adaptive control (MMAC) technique for noise attenuation is shown in Figure 1.
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