Coding Of Audio Signals Research Articles

Sub-band coding for audio signals using Matlab

This work presents the results of a research project aimed at studying the coding of audio signals in order to obtain a compact and efficient digital representation for transmission or storage purposes. A psychoacoustic model based on sub-band coding is implemented in MATLAB®, which identifies the type of audio: voice or music signal, based on a model of the human auditory system. Our focus was on measuring the relationship between reduced transmission/storage bitrate and audio quality.

Backward Adaptive and Quasi-Logarithmic Quantizer for Sub-Band Coding of Audio

This research presents an audio coding scheme, based on sub-band coding (SBC), with the implementation of quasi-logarithmic compandors. The presented coding scheme is based on signal decomposition and individual processing of the different sub-bands. The two SBC schemes for audio coding are presented, a non-adaptive and an adaptive coding scheme. The application of backward adaptation technique further improves the performance of this coding scheme, especially when using smaller compression factor values. This paper also describes the determination of an efficient bit allocation, used for coding the individual sub-bands. The results indicate that the proposed coding schemes can successfully be implemented in audio signal coding, providing a high quality output signal. DOI: http://dx.doi.org/10.5755/j01.itc.47.1.16190

The History of SMPTE and Digitization

SMPTE has been involved with the process of digitization, in one form or another, for over half the period of its existence. For more than the last four decades, the Society has been engaged actively in the process of transformation of its constituent industry segments from use of analog to use of digital technologies. SMPTE's involvement has matured from reporting on work done elsewhere in research and commercial settings, to conducting research directly, to developing technologies and standards for use by the industries it serves. The fields in which SMPTE has been engaged in the process of digitization have evolved from various control applications, to digital video and audio signal coding, to various forms of data transport, to signal data compression, to generation and use of metadata, to treatment of content as files, to all of the aspects of data workflows and content management. To deal with the very significant set of SMPTE contributions to the digitization of the motion imaging industries, this paper takes a more-or-less chronological approach. Many of the developments cited are from the author's personal recollections.

Modified BTC Algorithm for Audio Signal Coding

This paper describes modification of a well-known image coding algorithm, named Block Truncation Coding (BTC) and its application in audio signal coding. BTC algorithm was originally d ...

On perceptual distortion measures and parametric modeling

Over the past two decades, there has been much interest in incorporating human sound perception in signal processing algorithms, often in the form of perceptual distortion measure or an approximation thereof. An example of this is MDCT‐based audio coding where a good quality can be achieved at very low bit‐rates by taking masking effects into account. More recently, the same principles have been applied to parametric modeling and coding of audio signals. We discuss the inherent tradeoffs in choosing a perceptual distortion measure and a parametric model and the pros and cons of various ways of implementing such perceptual distortion measures is discussed. An important question that we seek to answer is whether perception should be taken into account in the estimation of model parameters or this should be done in a separate step.

Perceptual Coding of Audio Signals Using Adaptive Time-Frequency Transform

Wide band digital audio signals have a very high data-rate associated with them due to their complex nature and demand for high-quality reproduction. Although recent technological advancements have significantly reduced the cost of bandwidth and miniaturized storage facilities, the rapid increase in the volume of digital audio content constantly compels the need for better compression algorithms. Over the years various perceptually lossless compression techniques have been introduced, and transform-based compression techniques have made a significant impact in recent years. In this paper, we propose one such transform-based compression technique, where the joint time-frequency (TF) properties of the nonstationary nature of the audio signals were exploited in creating a compact energy representation of the signal in fewer coefficients. The decomposition coefficients were processed and perceptually filtered to retain only the relevant coefficients. Perceptual filtering (psychoacoustics) was applied in a novel way by analyzing and performing TF specific psychoacoustics experiments. An added advantage of the proposed technique is that, due to its signal adaptive nature, it does not need predetermined segmentation of audio signals for processing. Eight stereo audio signal samples of different varieties were used in the study. Subjective (mean opinion score--MOS) listening tests were performed and the subjective difference grades (SDG) were used to compare the performance of the proposed coder with MP3, AAC, and HE-AAC encoders. Compression ratios in the range of 8 to 40 were achieved by the proposed technique with subjective difference grades (SDG) ranging from ---0.53 to ---2.27.

Sparse and structured decompositions of signals with the molecular matching pursuit

This paper describes the Molecular Matching Pursuit (MMP), an extension of the popular Matching Pursuit (MP) algorithm for the decomposition of signals. The MMP is a practical solution which introduces the notion of structures within the framework of sparse overcomplete representations; these structures are based on the local dependency of significant time-frequency or time-scale atoms. We show that this algorithm is well adapted to the representation of real signals such as percussive audio signals. This is at the cost of a slight sub-optimality in terms of the rate of convergence for the approximation error, but the benefits are numerous, most notably a significant reduction in the computational cost, which facilitates the processing of long signals. Results show that this algorithm is very promising for high-quality adaptive coding of audio signals

Toward a Methodology for the Restoration of Electroacoustic Music

In recent years the evolution of digital technologies for the coding of audio signals has lead to new perspectives in the field of preservation and fruition of audio archives, even though new models for digital audio archives have not been yet developed. The most important international archives make use today of digital databases, and an increasing number of digital supports (DAT, CD-A, DVD-A). At the same time, in the field of musical philology, restored editions of electroacustic operas have never been released: these would lead to scientific problems never addressed so far, about methodologies and algorithms. We aim at applying restoration algorithms to recordings of electronic music; this repertoire has some peculiar problems: it requires a philological analysis of the piece, as many of the synthesized sounds have acoustic properties in common with noise, both impulsive and white. This work presents the results of an experiment aimed to evaluate the quality of different audio restoration algorithms based on different methods, based both time and frequency domain.

Perceptual noise shaping in the time domain via LPC prediction in the frequency domain

A method and apparatus for the perceptual coding of audio signals in which perceptual noise shaping is achieved in the time domain by performing a (linear) prediction (i.e., filtering) in the frequency domain. As a result, the temporal spread of quantization noise is reduced. Specifically, according to one illustrative embodiment of the present invention, a method comprises decomposition of the audio signal into a plurality of spectral component signals; generating a prediction signal representative of a prediction of one of said spectral component signals, said prediction based on one or more other ones of said spectral component signals; comparing the prediction signal with said one of said spectral component signals to generate a prediction error signal; coding said one of said spectral component signals based on the prediction error signal to generate a coded spectral component signal; and generating the encoded signal based on the coded spectral component signal.

Audio signal coding and decoding device

The present invention relates to an audio signal coding and decoding device for coding and decoding an audio signal using an adaptive differential coding system. The audio signal coding and decoding device according to the present invention comprises an adaptive differential pulse code modulation coding device for coding an audio signal, an adaptive differential pulse code modulation decoding device for decoding the audio signal coded, an integrator provided in the preceding stage of the coding device and for decreasing the variation between unit samples of the input audio signal of the coding device, and a differentiator provided in the succeeding stage of the decoding device and for processing the audio signal decoded by the decoding device so that the variation decreased by the integrator is restored to the original one. Since the integrator is provided in the preceding stage of the coding device, the variation between the unit samples of the input audio signal of the coding device is decreased. Therefore, a prediction error signal obtained by the coding device is decreased. As a result, quantizer step-size is decreased, resulting in a reduced quantization error.

Perceptual coding of audio signals using entropy coding and/or multiple power spectra

A technique for the masking of quantizing noise in the coding of audio signals is usable with the types of channel coding known as "noiseless" or Huffman coding and with variable radix packing. In a multichannel environment, noise masking thresholds may be determined by combining sets of power spectra for each of the channels. The stereophonic embodiment eliminates redundancies in the sum and difference signals, so that the stereo coding uses significantly less than twice the bit rate of the comparable monaural signal. The technique can be used both in transmission of signals and in recording for reproduction, particularly recording and reproduction of music. Compatibility with the ISDN transmission rates known as 1B, 2B and 3B rates has been achieved.

Multipurpose Hardware for Digital Coding of Audio Signals

We describe an adaptive coding system capable of representing audio frequency signals in a wide variety of digital code formats. To provide adaptive quantization, the step size is adjusted by voltage controlled amplifiers operating with analog control signals. Sine wave signal-to-noise ratio measurements demonstrate important relationships among the main code classifications: pcm, apcm, dpcm, adpcm. The multipurpose coder has proved a useful laboratory aid in the design of special purpose coders. After adjustment and evaluation of parameters using the multipurpose coder one proceeds to an economical, specialized design.