Small Number Of Bits Research Articles

Generalized interframe vertex-based shape encoding scheme for video sequences

The efficiency of shape coding is an important problem concerning content-based image manipulations and object-based coding of the video sequences. In order to encode the shape information of an object, the boundary is approximated by a polygon which can be encoded with the smallest number of bits for maximum allowable distortion. The conventional boundary coding schemes, however, does not successfully remove the temporal redundancy of the video sequences. This paper proposes a new boundary encoding scheme by which the temporal redundancy between two successive frames is efficiently removed, resulting in lower bit-rate than the conventional algorithms. The interframe vertex selection problem is solved by finding the path with the minimum cost in the directed acyclic graph (DAG) and its fast version using a simplified graph is introduced to reduce the computational load. The vertices were selected from both the current frame to be encoded and the previous frame already encoded, and thus, the temporal redundancy was effectively removed.

Compressing images for the internet

The World Wide Web has rapidly become the hot new mass communications medium. Content creators are using similar design and layout styles as in printed magazines, i.e., with many color images and graphics. The information is transmitted over plain telephone lines, where the speed/price trade-off is much more severe than in the case of printed media. The standard design approach is to use palettized color and to limit as much as possible the number of colors used, so that the images can be encoded with a small number of bits per pixel using the Graphics Interchange Format (GIF) file format. The World Wide Web standards contemplate a second data encoding method (JPEG) that allows color fidelity but usually performs poorly on text, which is a critical element of information communicated on this medium. We analyze the spatial compression of color images and describe a methodology for using the JPEG method in a way that allows a compact representation while preserving full color fidelity.

On temporal smoothing and speech perception

Neurophysiological and psychophysical studies provide evidence for the existence of temporal integration mechanisms in the auditory system. These may be viewed as low-pass filters, parametrized by their cutoff frequencies. It is of interest to specify these cutoffs, particularly for tasks germane to the effect of temporal smoothing on speech quality, within the context of speech-coding strategies (e.g., what is the smallest number of bits with which to represent temporally smoothed speech that is perceptually indistinguishable from the original?). In order to answer such questions, speech-processing rules have been derived so as to create synthetic speech capable of generating model auditory-nerve (AN) firing patterns temporally smoothed in a manner commensurate with that observed in neurophysiological studies of the AN (e.g., at low characteristic frequencies, neural discharges of AN fibers are phase locked to the underlying driving critical-band signal, i.e., synchrony is maintained; at high CFs, temporal information is preserved by the instantaneous average rate of the neural firing pattern, which is related to the temporal envelope of the underlying driving signal). This presentation describes the processing rules and the results of psychophysical experiments performed to measure the cutoff frequencies of these filters within the context of speech quality.

Inter-frame vertex selection scheme for object-based image sequence coder

This paper proposes a new boundary coding algorithm suitable for object-based image sequence coding system. The boundary is approximated by a polygon which can be encoded with the smallest number of bits for maximum distortion. The proposed scheme selects the vertex points considering the shape information from the past frames. The temporal redundancy between two successive frames is efficiently removed with the proposed scheme, resulting in lower bit-rate than the conventional algorithms.

Vertex-based inter-frame boundary encoding scheme

The boundary is approximated by a polygon which can be encoded with the smallest number of bits for maximum distortion. The temporal redundancy between two successive frames is efficiently removed with the proposed scheme, resulting in a lower bit rate than the conventional algorithms.

Transmission of analog signals using multicarrier modulation: a combined source-channel coding approach

We have proposed and analyzed a combined source-channel coding scheme using multicarrier modulation. By changing the power and modulation of subchannels carrying different bits of the quantized signal, the channel-induced distortion can be minimized. We derive an algorithm for the allocation of power among subchannels. Analog sources quantized by scalar and vector quantizers are considered. For a Gaussian source scalar-quantized to 8 bit, multicarrier transmission yields a 13 dB improvement of signal-to-distortion ratio over a single-carrier system. Quantizers having a smaller number of bits yield smaller improvements. We consider both full-search and binary tree-search vector quantizers using a natural binary coding scheme. For a vector quantizer using 2 b/sample, multicarrier transmission yields an improvement in signal-to-distortion ratio that lies between 0.3 and 0.8 dB, depending on the quantizer design.

Method of coding 32-kb/s audio signals

Coding digitized audio signals includes dividing an audio signal, consisting of a continuous sequence of sample values, into successive blocks of equal length, and performing overlapping windowing. The blocks are transformed into complex Fourier coefficients by means of a discrete Fourier transform, which complex Fourier coefficients are decomposed into magnitude values and phase values. The phase values are quantized with a linear quantization characteristic which becomes coarser going from low toward high frequencies. The magnitude values are combined into frequency bands which are oriented with regard to predetermined critical bands and become wider toward high frequencies. Quantization levels for each frequency band are fixed, taking into account a lower, frequency-dependent, absolute threshold of hearing, such that a range lying below the respective threshold of hearing is disregarded, and taking into account a relative threshold of hearing, such that ranges in the neighborhood of frequencies with large magnitude values are taken into account to a reduced extent. A number of sub-bands consisting of single values of the frequency bands are fixed, and the greatest single magnitude value of each sub-band is determined. Variable quantization of each frequency band is performed by quantizing the greatest magnitude value of a sub-band with a maximum available number of bits and all other magnitude values with a smaller number of bits.

Adaptively controlled 4.8 kbit/s speech coders with low distortion

AbstractAs a low bit‐rate speech coding method to realize a satisfactory quality of reconstructed speech, this paper proposes a 4.8 kbit/s pitch‐controlled multipulse coding method, which adaptively controls the speech pitch prediction interval and represents the multipulse positions for the excitation signals with a small number of bits. As a method to control the pitch adaptively, an algorithm is derived and proposed where the pitch prediction interval is adjusted adaptively for each frame so that the pulse position is determined to minimize the perceptual error power representing subjectively important difference.In addition to the forementioned method, another method also is proposed to improve the speech quality, where the pitch‐controlled multipulse coding is proposed to control adaptively the coefficients of the synthesis filter. An algorithm is proposed for the adaptive control of the coefficients of the synthesis filter, to calculate anew the optimal filter coefficients for excitation pulses derived from the adaptive control of the pitch.A simulation is executed for the two methods, and the following results are obtained as an evaluation of SNRseg. It is 7.5 dB in the conventional MPC algorithm, 8.6 dB in the proposed pitch‐controlled MPC algorithm, and 9.8 dB in the pitch‐controlled MPC algorithm including the adaptive control of the synthesis filter coefficients. In other words, the improvement in the latter two is 1.1 and 2.3 dB, respectively. The performance is satisfactory also from the viewpoint of the cepstral distance. The improvement in the reconstructed speech quality also is verified for the proposed methods by the subjective evaluation.

Discovering simple DNA sequences by the algorithmic significance method

A new method, 'algorithmic significance', is proposed as a tool for discovery of patterns in DNA sequences. The main idea is that patterns can be discovered by finding ways to encode the observed data concisely. In this sense, the method can be viewed as a formal version of the Occam's Razor principle. In this paper the method is applied to discover significantly simple DNA sequences. We define DNA sequences to be simple if they contain repeated occurrences of certain 'words' and thus can be encoded in a small number of bits. Such definition includes minisatellites and microsatellites. A standard dynamic programming algorithm for data compression is applied to compute the minimal encoding lengths of sequences in linear time. An electronic mail server for identification of simple sequences based on the proposed method has been installed at the Internet address pythia/anl.gov.

Code-conversion method and apparatus for analyzing and synthesizing human speech

Human speech is analyzed and a feature parameter is extracted from the human speech. The feature parameter is quantized and a quantization parameter is generated. Part of the bits of the quantization parameter is a predetermined bit pattern when the feature parameter is in a predetermined range. The part of the bits of the quantization parameter is tranformed into an additive parameter of at least one bit. The human speech is synthesized on the basis of the quantization parameter and the additive parameter. It is possible to nonlinearly quantize human speech into a quantization parameter having a small number of bits and to generate the synthesis parameter from the quantization parameter using a small size circuit.

Inter-Block Redundancy Reduction in Vector-Quantized Images by a Neural Predictor

A prediction scheme is proposed, which allows one to exploit residual interblock correlation after vector quantization (VQ), without loss in the quality of reconstructed images. Such a scheme is based on a Perceptron neural network, whose goal is to produce an estimation of a block, once the neighboring blocks have already been coded. The network is trained on a set of images by means of the classical back-propagation algorithm. The error to be minimized is the Euclidean distance (i.e., the mean square error, MSE) between the block generated by the network (using the four previously-coded neighboring blocks as input) and the current block. During the encoding phase, for each block vector-quantized, the block produced by the network is used to generate a reduced codebook that, in the case of a correct prediction, allows one to use a smaller number of bits for the transmission of the block. Experimental results show that the probability of a correct prediction ranges from 60 to 75 %, thus producing an increase in compression by a factor of 1.3 to 1.6. The main advantages of the presented approach lie in the efficiency of the neural predictor (even for images not included in the training set), and in reduced memory requirements in comparison with other methods previously proposed.

Partitioning circuits for improved testability

Exhaustive self-testing of combinational circuitry within the framework of the level-sensitive scan design (LSSD) discipline requires that every output node depend on a small number of input nodes. We present here efficient algorithms that take an arbitrary block of combinational logic and add to it the smallest number of bits of new LSSD registers necessary to: (1) partition the logic so that no output depends on more thank inputs, and (2) maintain timing within the block (so that all input-to-output paths encounter the same number of bits of register). Our partitioning algorithms conform to two different design constraints. We also show that the unconstrained partitioning problem is NP-complete.

Design of N-D digital filters with finite wordlength coefficients

A design technique for N-dimensional (N-D) IIR (infinite impulse response) digital filters is developed. Instead of using optimal step size in jth iteration of Fletcher-Powell algorithm, a suboptimal step size is used to increase the overall computation efficiency. Criteria are developed for the suboptimal step size to keep the mapping matrix D/sub j/ positive definite and to ensure the stability of the transfer function. The approach does not require checking the positive definiteness of the mapping matrix D/sub j/ or the stability of the filter. A optimization algorithm with finite-wordlength coefficients is developed. The algorithm minimizes the performance error caused by truncating the filter coefficients to a specified number of bits and provides a means for finding the smallest number of bits that results in an acceptable error. Two examples of 2-D digital filters are designed by the proposed technique. >

Digital filter realization based on simulation of the node equation of lc filters

AbstractOne method of synthesizing a digital filter involves the simulation of an LC analog filter. This method retains the low element sensitivity that is characteristic of an LC filter and is generally attractive in that it requires only a small number of bits in the multiplier that is used to produce the filter coefficients. The web digital filter is a typical application of this design method and is known to possess low‐element sensitivity, but it involves a somewhat complex configuration and the transfer functions which can be realized are therefore limited. This paper discusses a method of designing digital filters by simulation of the node equation of LC filters. Using this method, the transfer function of an analog filter which can be implemented by means of LCR elements, including transformers, can always be realized. With this method, we first transform the LC filter into an equivalent circuit and determine the applicable node equation. Then bilinear transformation is used to transform the s‐variable into the z‐variable and, to prevent the occurrence of a delay‐free loop within the circuit, after rearranging the circuit equation it is realized as a digital filter. In addition, a study is made of methods reducing the number of multipliers without degrading the coefficient sensitivity. Finally, the effectiveness of this design method is verified by means of computer simulation.

Efficient speech coding techniques by simplified orthogonal transformation

Abstract Digitally represented speech has to be quantized into as small number of bits as possible to save the storage memory or transmission bandwidth. Such lower bit rates can be achieved by encoding the parameters characterizing the special features of speech singals instead of trying to encode the speech waveform directly. Linear predictive coding (LPC) is one such technique. Selective coding is to allocate different numbers of bits to different parameters with different statistical variance. Orthogonal transformation can be applied to LPC parameters here to more efficiently allocate the bits. Because orthogonal transformation requires high complexity of computation, a “simplified orthogonal transformation” is proposed in this paper, which can be obtained by a series of training data, to tremendously reduce the computation but essentially preserve the advantages. Many efficient coding techniques in such case are proposed in this paper, the results indicate that very satisfactory speech quality can be ...

Nulling performance of null-steering arrays with digital phase-only weights

After a brief description of various application capabilities offered by phased array techniques, a simulated analysis is presented of null-steering arrays using microwave digital phase-only weights, i.e. in the presence of phase quantisation errors. It is shown that sufficiently good nulling performances requires only a small number of bits.

Optical computing for image bandwidth compression: analysis and simulation

Image bandwidth compression is dominated by digital methods for carrying out the required computations. This paper discusses the general problem of using optics to realize the computations in bandwidth compression. A common method of digital bandwidth compression, feedback differential pulse code modulation (DPCM), is reviewed, and the obstacles to making a direct optical analogy to feedback DPCM are discussed. Instead of a direct optical analogy to DPCM, an optical system which captures the essential features of DPCM without optical feedback is introduced. The essential features of this incoherent optical system are encoding of low-frequency information and generation of difference samples which can be coded with a small number of bits. A simulation of this optical system by means of digital image processing is presented, and performance data are also included.

Developing Large Binary to BCD Conversion Structures

Fast binary to BCD conversions consist of static implementation of a sequential algorithm and are limited to relatively small number of bits as the hardware increases rapidly with each added bit.

Image transmission by two-dimensional contour coding

The results of a computer simulation of an image transmission system are reported. A reduction in the total number of bits required to describe a picture by a factor of 4 to 23 is possible as compared with 6-bit PCM. In this system an image is treated as a two-dimensional signal of the spatial coordinates x and y. The large changes in brightness in a picture occur at the edges of objects and are accentuated by the visual system. The edge points can be isolated by the gradient or Laplacian operator. The fact that these edge points lie along connected contours in two dimensions is used to code the location and characteristics of each point efficiently. Two-dimensional reconstruction filters are derived to synthesize the high frequency picture from the decoded edge information. A two-dimensional low-pass or out-of-focus picture is also formed which can be transmitted with a relatively small number of bits. After a possible accentuation to make the picture appear "sharper," the "synthetic highs" signal is added to the low-pass picture to yield the final output.