During the last decade, CD-quality digital audio has essentially replaced analog audio. Emerging digital audio applications for network, wireless, and multimedia computing systems face a series of constraints such as reduced channel bandwidth, limited storage capacity, and low cost. These new applications have created a demand for high-quality digital audio delivery at low bit rates. In response to this need, considerable research has been devoted to the development of algorithms for perceptually transparent coding of high-fidelity (CD-quality) digital audio. As a result, many algorithms have been proposed, and several have now become international and/or commercial product standards. This paper reviews algorithms for perceptually transparent coding of CD-quality digital audio, including both research and standardization activities. This paper is organized as follows. First, psychoacoustic principles are described, with the MPEG psychoacoustic signal analysis model 1 discussed in some detail. Next, filter bank design issues and algorithms are addressed, with a particular emphasis placed on the modified discrete cosine transform, a perfect reconstruction cosine-modulated filter bank that has become of central importance in perceptual audio coding. Then, we review methodologies that achieve perceptually transparent coding of FM- and CD-quality audio signals, including algorithms that manipulate transform components, subband signal decompositions, sinusoidal signal components, and linear prediction parameters, as well as hybrid algorithms that make use of more than one signal model. These discussions concentrate on architectures and applications of those techniques that utilize psychoacoustic models to exploit efficiently masking characteristics of the human receiver. Several algorithms that have become international and/or commercial standards receive in-depth treatment, including the ISO/IEC MPEG family (-1, -2, -4), the Lucent Technologies PAC/EPAC/MPAC, the Dolby AC-2/AC-3, and the Sony ATRAC/SDDS algorithms. Then, we describe subjective evaluation methodologies in some detail, including the ITU-R BS.1116 recommendation on subjective measurements of small impairments. This paper concludes with a discussion of future research directions.

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Perceptual coding of digital audio

A device to provide body fluid flow control in the form of a valve to be located within a duct or passageway. The device is controlled through pressure above a preselected threshold. Bulk resilience about a passageway in a valve body provides the mechanism for controlled flow. One-way valve operation may be provided through a flap or through a pressure differential on the valve body depending upon the direction of flow. A frame structure positioned within a resilient seal includes longitudinally elongate elements which may be of spring material, malleable material or heat recoverable material so as to accomplish an initial insertion state and an expanded anchoring state. A valve support transitions between the resilient seal portion and the valve body to insure that the states do not change the threshold opening pressure. Insertion devices may be employed to position and actuate a change of state of the frame in the body duct or passageway.

Body fluid flow control device

With the advent of `multimedia', digital signal processing (DSP) of sound has emerged from the shadow of bandwidth limited speech processing to become a research field of its own. To date, most research in DSP applied to sound has been concentrated on speech, which is bandwidth limited to about 4 kilohertz. Speech processing is also limited by the low fidelity typically expected in the telephone network. Today, the main applications of audio DSP are high quality audio coding and the digital generation and manipulation of music signals. They share common research topics including perceptual measurement techniques and analysis/synthesis methods. Additional important topics are hearing aids using signal processing technology and hardware architectures for digital signal processing of audio. In all these areas the last decade has seen a significant amount of application-oriented research. The frequency range of wideband audio has an upper limit of 20 kilohertz and the resulting difference in frequency range and Signal to Noise Ratio (SNR) due to sample size must be taken into account when designing DSP algorithms. There are whole classes of algorithms that the speech community is not interested in pursuing or using. These algorithms and techniques are revealed in this book. This book is suitable for advanced level courses and serves as a valuable reference for researchers in the field. Interested and informed engineers will also find the book useful in their work.

Applications of digital signal processing to audio and acoustics

A new method for the objective assessment and prediction of perceived audio quality is introduced. It represents an expansion of the speech quality measure qC, introduced by Hansen and Kollmeier, and is based on a psychoacoustically validated, quantitative model of the "effective" peripheral auditory processing by Dau et al. To evaluate the audio quality of a given distorted signal relative to a corresponding high-quality reference signal, the auditory model is employed to compute "internal representations" of the signals, which are partly assimilated in order to account for assumed cognitive aspects. The linear cross correlation coefficient of the assimilated internal representations represents the perceptual similarity measure (PSM). PSM shows good correlations with subjective quality ratings if different types of audio signals are considered separately, whereas a better accuracy of signal-independent quality prediction is achieved by a second quality measure PSMt represented by the fifth percentile of the sequence of instantaneous audio quality PSM(t). The new measures were evaluated using a large database of subjective listening tests that were originally carried out on behalf of the International Telecommunication Union (ITU) and Moving Pictures Experts Group (MPEG) for the evaluation of various low bit-rate audio codecs. Additional tests with data unknown in the development phase of the model were carried out. Except for linear distortions, the new method shows a higher prediction accuracy than the ITU-R recommendation BS.1387 ("PEAQ") for the tested data

PEMO-Q&#8212;A New Method for Objective Audio Quality Assessment Using a Model of Auditory Perception

A digital content distribution and subscription system for digital data files, which provides subscriber (64) with music or video (62) for a fee.Subscribers (64) may subscribe to a predetermined number of song slots for a period of time.The subscriber (64) may select songs from various play lists, which causes encrypted digital audio files (68) for the selected music to be downloaded to the subscriber's PC.A license key (66) is downloaded to the subscriber's PC to enable the selected songs to be decoded and displayed for a predetermined time period.At the expiration of the time period, the license key (66) expires.Each subscriber's selections are tabulated for the purpose of toyality tracking and payment of a license fee to the copyright holder.The current system provides subscribers (64) with relatively high-quality digital audio files (68) and allows subscribers (64) to select from a wide range of stored digital audio files (68) on demand while not violating copyright laws.

Digital content distribution and subscription system

A digital encoding process for transmitting and/or storing acoustical sigs and, in particular, music signals, in which scanned values of the acoustical signal are transformed by means of a transformation or a filter bank into a sequence of second scanned values, which reproduce the spectral composition of the acoustical signal, and the sequence of second scanned values is quantized in accordance with the requirements with varying precision and is partially or entirely encoded by an optimum encoder, and in which a corresponding decoding and inverse transformation takes place during the reproduction. An encoder is utilized in a manner in which the occurrence probability of the quantized spectral coefficient is correlated to the length of the code in such a way that the more frequently the spectral coefficient occurs, the shorter the code word. A code word and, if needed, a supplementary code is allocated to several elements of the sequence or to a value range in order to reduce the size of the table of the encoder. A portion of the code words of variable length are arranged in a raster, and the remaining code words are distributed in the gaps still left so that the beginning of a code word can be more easily found without completely decoding or in the event of faulty transmission.

Digital encoding process

In a method of coding a plurality of audio signals, the left and the right basic channel as well as the central channel are combined by joint stereo coding so as to obtain a jointly coded signal, which is decoded so as to provide simulated decoded signals. The simulated decoded signals and two surround channels are combined by matricization by means of a compatibility matrix so as to form compatible signals which are suitable for decoding by existing decoders. In order to avoid audible disturbances caused by excessive energy contents of the compatible signals, which would occur if joint stereo coding and decoding were carried out prior to carrying out the matricization, the compatible signals or the simulated decoded signals are dynamically weighted by means of a dynamic correction factor in such a way that the compatible signals are approximated with regard to their energy to the energy of signals which would be obtained if the two basic channels and the central channel as well as the surround channels were matricized directly.

Method of coding a plurality of audio signals

A digital adaptive transformation coding method for the transmission and/ortorage of audio signals, specifically music signals, wherein N scanned values of the audio signal are transformed into M spectral coefficients, and the coefficients are split up into frequency groups, quantized and then coded. The quantized maximum value of each frequency group is used to define the coarse variation of the spectrum. The same number of bits is assigned to all values in a frequency group. The bits are assigned to the individual frequency groups as a function of the quantized maximum value present in the particular frequency group. A multi-signal processor system is disclosed which is specifically designed for implementation of this method.

Digital adaptive transformation coding method

In a digital coding process for transmission and/or storage of acoustic signals, in particular musical signals, scanned values of the acoustic signal are transformed by means of a transformation or by a filter bank into a sequence of second scanned values which reproduce the spectral composition of the acoustic signal. The sequence of scanned values is then quantified with varying precision, depending on the requirements and partly or wholly coded by means of an optimal coder. Corresponding decoding and back-transformation is effected during reproduction. The process disclosed is characterized in that a coder is used in a manner known per se, and the probability of the occurence of the quantified spectral coefficients is correlated with the length of the code; the higher the frequency with which the spectral coefficient occurs, the shorter the code word. To reduce the tabulated quantities of the coder, a code word and possibly an additional code are associated either with several elements of the sequence or with a range of values. Some of the code words of variable length are arranged in a grid. The remaining code words are distributed to the remaining holes, so that the beginning of a code word can be found more easily without complete decoding or in the event of defective transmission.

Digital coding process

Advanced Audio Measurement System Using Psychoacoustic Properties

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