Showing papers on "Encoder published in 1998"

Bandwidth-efficient turbo trellis-coded modulation using punctured component codes

Abstract: We present a bandwidth-efficient channel coding scheme that has an overall structure similar to binary turbo codes, but employs trellis-coded modulation (TCM) codes (including multidimensional codes) as component codes. The combination of turbo codes with powerful bandwidth-efficient component codes leads to a straightforward encoder structure, and allows iterative decoding in analogy to the binary turbo decoder. However, certain special conditions may need to be met at the encoder, and the iterative decoder needs to be adapted to the decoding of the component TCM codes. The scheme has been investigated for 8-PSK, 16-QAM, and 64-QAM modulation schemes with varying overall bandwidth efficiencies. A simple code choice based on the minimal distance of the punctured component code has also been performed. The interset distances of the partitioning tree can be used to fix the number of coded and uncoded bits. We derive the symbol-by-symbol MAP component decoder operating in the log domain, and apply methods of reducing decoder complexity. Simulation results are presented and compare the scheme with traditional TCM as well as turbo codes with Gray mapping. The results show that the novel scheme is very powerful, yet of modest complexity since simple component codes are used.

Source coding enhancement using spectral-band replication

Abstract: The present invention proposes a new method and apparatus for the enhancement of source coding systems. The invention employs bandwidth reduction (101) prior to or in the encoder (103), followed by spectral-band replication (105) at the decoder (107). This is accomplished by the use of new transposition methods, in combination with spectral envelope adjustments. Reduced bitrate at a given perceptual quality or an improved perceptual quality at a given bitrate is offered. The invention is preferably integrated in a hardware or software codec, but can also be implemented as a separate processor in combination with a codec. The invention offers substantial improvements practically independent of codec type and technological progress.

A space-time coding modem for high-data-rate wireless communications

Abstract: This paper presents the theory and practice of a new advanced modem technology suitable for high-data-rate wireless communications and presents its performance over a frequency-flat Rayleigh fading channel. The new technology is based on space-time coded modulation (STCM) with multiple transmit and/or multiple receive antennas and orthogonal pilot sequence insertion (O-PSI). In this approach, data is encoded by a space-time (ST) channel encoder and the output of the encoder is split into N streams to be simultaneously transmitted using N transmit antennas. The transmitter inserts periodic orthogonal pilot sequences in each of the simultaneously transmitted bursts. The receiver uses those pilot sequences to estimate the fading channel. When combined with an appropriately designed interpolation filter, accurate channel state information (CSI) can be estimated for the decoding process. Simulation results of the proposed modem, as applied to the IS-136 cellular standard, are presented. We present the frame error rate (FER) performance results as a function of the signal-to-noise ratio (SNR) and the maximum Doppler frequency, in the presence of timing and frequency offset errors. Simulation results show that for a 10% FER, a 32-state eight-phase-shift keyed (8-PSK) ST code with two transmit and two receive antennas can support data rates up to 55.8 kb/s on a 30-kHz channel, at an SNR of 11.7 dB and a maximum Doppler frequency of 180 Hz. Simulation results for other codes and other channel conditions are also provided. We also compare the performance of the proposed STCM scheme with delay diversity schemes and conclude that STCM can provide significant SNR improvement over simple delay diversity.

Servo controller in scanner of image formation device

Abstract: PROBLEM TO BE SOLVED: To provide a scanner controller of an image formation device which automatically changes the direction of the motor current and provides a rapid and proper scanner travelling performance, when a difference between the motor current and a control target current value at the time of speed reduction in the return operation of the scanner is large. SOLUTION: In an image formation device, a scanner for reading the image data from an original image is reciprocated by a motor M31. A servo controller is constituted of an H-bridge circuit, which rotates the motor in the forward and the reverse direction from a PWM(pulse width modulation) signal which determines the motor speed and a signal determining the direction of rotation, an encoder EC31 installed on a motor shaft, a microcomputer 30 which detects the speed of the rotation and calculates the speed, and controls the speed of the motor from a detection signal of the encoder, a detecting section 40 which detects the value and direction of current in the motor, and a feedback system which controls the speed from the deviation of the motor current value from a target command current value, obtained form the calculation of the motor speed and determines the direction of current to be allowed to flow in the motor from the comparison result between the motor current value and the target command current value and then controls the direction of rotation of the motor.

A frequency-domain video transcoder for dynamic bit-rate reduction of MPEG-2 bit streams

Abstract: Many of the forthcoming video services and multimedia applications are expected to use preencoded video for storage and transmission. Video transcoding is intended to provide transmission flexibility to preencoded bit streams by dynamically adjusting the bit rate of these bit streams according to new bandwidth constraints that were unknown at the time of encoding. In this paper, we propose a drift-free MPEG-2 video transcoder, working entirely in the frequency domain. The various modes of motion compensation (MC) defined in MPEG-2 are implemented in the discrete cosine transform (DCT) domain at reduced computational complexity. By using approximate matrices to compute the MC-DCT blocks, we show that computational complexity can be reduced by 81% compared with the pixel domain approach. Moreover, by using a Lagrangian rate-distortion optimization for bit reallocation, we show that optimal transcoding of high-quality bit streams can produce better picture quality than that obtained by directly encoding the uncompressed video at the same bit rates using a nonoptimized Test Model 5 (TM5) encoder.

Bit-rate control using piecewise approximated rate-distortion characteristics

Abstract: Digital video's increased popularity has been driven to a large extent by a flurry of international standards (MPEG-1, MPEG-2, H.263, etc). In most standards, the rate control scheme, which plays an important role in improving and stabilizing the decoding and playback quality, is not defined, and thus different strategies can be implemented in each encoder design. Several rate-distortion (R-D)-based techniques have been proposed aimed at the best possible quality for a given channel rate and buffer size. These approaches are complex because they require the R-D characteristics of the input data to be measured before making quantization assignment decisions. We show how the complexity of computing the R-D data can be reduced without significantly reducing the performance of the optimization procedure. We propose two methods which provide successive reductions in complexity by: (1) using models to interpolate the rate and distortion characteristics, and (2) using past frames instead of current ones to determine the models. Our first method is applicable to situations (e.g., broadcast video) where a long encoding delay is possible, while our second approach is more useful for computation-constrained interactive video applications. The first method can also be used to benchmark other approaches. Both methods can achieve over 1 dB peak signal-to-noise rate (PSNR) gain over simple methods like the MPEG Test Model 5 (TM5) rate control, with even greater gains during scene change transitions. In addition, both methods make few a priori assumptions and provide robustness in their performance over a range of video sources and encoding rates. In terms of complexity, our first algorithm roughly doubles the encoding time as compared to simpler techniques (such as TM5). However, the complexity is greatly reduced as compared to methods which exactly measure the R-D data. Our second algorithm has a complexity marginally higher than TM5 and a PSNR performance slightly lower than that of the first approach.

A forward error correction system for packet based real-time media

Abstract: A computationally simple yet powerful forward error correction code scheme for transmission of real-time media signals, such as digitized voice, video or audio, in a packet switched network, such as the Internet. According to one aspect, an encoder at the sending end derives p redundancy blocks from each group of a k payload blocks and concatenates the redundancy blocks, respectively, with payload blocks in the next group of k payload blocks. At the receiving end, a decoder may recover up to p missing packets in a group of k packets, provided with the p redundancy blocks carried by the next group of k packets. The invention may, for instance, append to each of a series of payload packets a single forward error correction code that is defined by taking the XOR sum of a preceding specified number of payload packets. The invention thereby enables correction from the loss of multiple packets in a row, without significantly increasing the data rate or otherwise delaying transmission.

Estimation of Angular Velocity and Acceleration from Shaft-Encoder Measurements

Abstract: This paper suggests a Kalman-filter approach to the estimation of angular velocity and acceleration from (quantized) shaft-encoder measurements Finite-difference estimates deteriorate as sampling rates are increased For small sampling periods, we show that the filtering problem is the dual of the cheap control problem, and we jus tify the use of all-integrator models We investigate Kalman filtering with constant sampling rate, and also with measurements triggered by encoder pulses Simulation and experimental results are given

Scalable audio coder and decoder

Abstract: The coder/decoder (codec) system of the present invention includes a coder and a decoder. The coder includes a multi-resolution transform processor, such as a modulated lapped transform (MLT) transform processor, a weighting processor, a uniform quantizer, a masking threshold spectrum processor, an entropy encoder, and a communication device, such as a multiplexor (MUX) for multiplexing (combining) signals received from the above components for transmission over a single medium. The decoder comprises inverse components of the encoder, such as an inverse multi-resolution transform processor, an inverse weighting processor, an inverse uniform quantizer, an inverse masking threshold spectrum processor, an inverse entropy encoder, and an inverse MUX. With these components, the present invention is capable of performing resolution switching, spectral weighting, digital encoding, and parametric modeling.

Adaptive senso-motor encoder for visual or acoustic prosthesis

Abstract: The invention concerns an adaptive senso-motor encoder for a visual or acoustic prosthesis, said encoder having a central checking unit for signal-processing functions, monitoring functions, control functions and external action functions. The encoder further comprises a group of adaptive spatio-temporal filters for converting sensor signals into stimulation pulse sequences, a bi-directional interface being provided for coupling the encoder to an implantable microstructure (2) for stimulating nerve or glia tissue and monitoring brain functions.

Method and system for indexing, sorting, and displaying a video database

Abstract: A method and system is provided for indexing, sorting, and displaying a video database Hardware and software components, and a novel encoding process are used to provide a searchable video and informational database Each encoder and the User use specially configured graphical user interfaces to access the system In the preferred embodiment, component clips of different videotape views of a sporting event are batch encoded and synchronized Data which remains constant for at least a part of a game is automatically reused A MasterPlayerId assigned to each player is used to index all video clips and information relating to that player A rating service provides ratings of individual plays or players The present invention includes powerful search features that permit a User to search the informational and video database according to numerous predefined and customized criteria A novel encoding scheme permit the viewing of MPEG format video clips in a form of slow motion A dedicated console permits the User to view selected information using a first display screen while simultaneously displaying some or all of this information to at least one other person using a second display screen The present invention also permits the User to select and save custom view sets of video clips, and to control the video display The User can switch this camera view during play or can replay a clip from a different camera view

System and method for fixed-rate block-based image compression with inferred pixel values

Abstract: An image processing system (205) includes an image encoder system (220) and an image decoder system (230) that are coupled together. The image encoder system (220) includes an image decomposer (315) and a block encoder (318) that are coupled together. The block encoder (318) includes a color quantizer (335) and a bitmap construction module (340). The image decomposer (315) breaks an original image into blocks. Each block (260) is then processed by the block encoder (318a-nth). Specifically, the color quantizer (335) selects some number of base points, or codewords, that serve as reference pixel values, such as colors, from which quantized pixel values are derived. The bitmap construction module (340) then maps each pixel colors to one of the derived quantized colors. The codewords and bitmap are output as encoded image blocks (320). The decoder system (230) includes a block decoder (505a-mth). The block decoder (505a-mth) includes a block type detector (520), one or more decoder units, and an output selector (523). Using the codewords of the encoded data blocks, the comparator and the decoder units determine the quantized colors for the encoded image block and map each pixel to one of the quantized colors. The output selector (523) outputs the appropriate color, which is ordered in an image composer with the other decoded blocks to output an image representative of the original image. A method for encoding an original image and for decoding the encoded image to generate a representation of the original image is also disclosed.

Seamless splicing of MPEG-2 multimedia data streams

Abstract: Respective encoders provide a first and second encoded MPEG-2 data streams for a first and second program respectively. Each stream includes at least video and audio components. The encoder provides seamless video splice-in and splice-out points. A play-to-air splicer is commanded to switch the broadcast output from the first input stream to the second input streams. The splicer identifies approximately aligned seamless video splice-in and seamless video splice-out points in the respective first and second video streams. The splicer splices the second video stream to the first video stream, but continues to broadcast the first audio stream. The splicer identifies corresponding audio splice-in and splice-out points. The splicer splices the second audio component to the first audio component. The splicer adjusts the decode and presentation times in the second stream after the respective slice-in to be consistent with such times in the first program. A decoder converts the compressed video and audio components output from the splicer into uncompressed form.

Multi-channel audio encoder

Abstract: A subband audio coder employs perfect/non-perfect reconstruction filters, predictive/non-predictive subband encoding, transient analysis, and psycho-acoustic/minimum mean-square-error (mmse) bit allocation over time, frequency and the multiple audio channels to encode/decode a data stream to generate high fidelity reconstructed audio. The audio coder windows the multi-channel audio signal such that the frame size, i.e. number of bytes, is constrained to lie in a desired range, and formats the encoded data so that the individual subframes can be played back as they are received thereby reducing latency. Furthermore, the audio coder processes the baseband portion (0-24 kHz) of the audio band-width for sampling frequencies of 48 kHz and higher with the same encoding/decoding algorithm so that audio coder architecture is future compatible.

Animation of synthetic faces in MPEG-4

Abstract: MPEG-4 is the first international standard that standardizes true multimedia communication-including natural and synthetic audio, natural and synthetic video, as well as 3D graphics. Integrated into this standard is the capability to define and animate virtual humans consisting of synthetic heads and bodies. For the head, more than 70 model-independent animation parameters defining low-level actions like "move left mouth corner" up to high-level parameters like facial expressions and visemes are standardized In a communication application. The encoder can define the face model using MPEG-4 BIFS (BInary Format for Scenes) and transmit it to the decoder. Alternatively, the encoder can rely on a face model that is available at the decoder. The animation parameters are quantized, predictively encoded using an arithmetic encoder or a DCT. The decoder receives the model and the animation parameters in order to animate the model. Since MPEG-4 defines the minimum MPEG-4 terminal capabilities in profiles and levels, the encoder knows the quality of the animation at the decoder.

Analysis, design, and iterative decoding of double serially concatenated codes with interleavers

Abstract: A double serially concatenated code with two interleavers consists of the cascade of an outer encoder, an interleaver permuting the outer codeword bits, a middle encoder, another interleaver permuting the middle codeword bits, and an inner encoder whose input words are the permuted middle codewords. The construction can be generalized to h cascaded encoders separated by h-1 interleavers, where h>3. We obtain upper bounds to the average maximum likelihood bit-error probability of double serially concatenated block and convolutional coding schemes. Then, we derive design guidelines for the outer, middle, and inner codes that maximize the interleaver gain and the asymptotic slope of the error probability curves. Finally, we propose a low-complexity iterative decoding algorithm. Comparisons with parallel concatenated convolutional codes, known as "turbo codes", and with the proposed serially concatenated convolutional codes are also presented, showing that in some cases, the new schemes offer better performance.

Modular conveyor system having multiple moving elements under independent control

Abstract: The modular conveyor system (20) comprises n interconnected track sections (26), forming a continuous track (24), wherein each track section (26) features a plurality of individually controlled coils (35) stretching along the length thereof. Plural pallets (22), each having thrust producing magnets, travel independently along the track (24). The track (24) also comprises multiple linear encoder readers (50) spaced at fixed positions therealong, and each pallet (22) includes a linear encoder strip (45) having a length R greater than the spacing E between the readers (50). Track section controllers (90) associate the encoder strips (45) with only one reader (50) at any time in order to resolve the position of the pallets based on the fixed position of the readers and the relative positions of the strips in relation thereto. The section controllers (90) also regulate and commutate the coils (35) of the corresponding track sections (26) in order to independently control each pallet (22). Communication links (92) interface adjacent section controllers (90) and enable them to transfer the pallet position-detecting responsibility between one another when a given pallet straddles adjacent track sections (26). The section controllers (90) also provide coil regulating signals to each other in the event a given moving element spans coils (35) situated in adjacent track sections (24). The electromagnetic structure and distributed control architecture of the conveyor system (20) enable it to independently control multiple practical pallets yet be constructed out of modular track sections, with little practical restriction on the length of the conveyor system or the number of pallets controlled thereby.

High-quality audio compression using an adaptive wavelet packet decomposition and psychoacoustic modeling

Abstract: This paper presents a technique to incorporate psychoacoustic models into an adaptive wavelet packet scheme to achieve perceptually transparent compression of high-quality (34.1 kHz) audio signals at about 45 kb/s. The filter bank structure adapts according to psychoacoustic criteria and according to the computational complexity that is available at the decoder. This permits software implementations that can perform according to the computational power available in order to achieve real time coding/decoding. The bit allocation scheme is an adapted zero-tree algorithm that also takes input from the psychoacoustic model. The measure of performance is a quantity called subband perceptual rate, which the filter bank structure adapts to approach the perceptual entropy (PE) as closely as possible. In addition, this method is also amenable to progressive transmission, that is, it can achieve the best quality of reconstruction possible considering the size of the bit stream available at the encoder. The result is a variable-rate compression scheme for high-quality audio that takes into account the allowed computational complexity, the available bit-budget, and the psychoacoustic criteria for transparent coding. This paper thus provides a novel scheme to marry the results in wavelet packets and perceptual coding to construct an algorithm that is well suited to high-quality audio transfer for Internet and storage applications.

Multiple-description wavelet based image coding

Abstract: We consider the problem of image coding for communication systems that use diversity to overcome channel impairments. We focus on the special case in which there are two channels of equal capacity between a transmitter and a receiver. Our designs are based on a combination of techniques successfully applied to the construction of some of the most efficient wavelet based image coding algorithms, with multiple description scalar quantizers (MDSQs). For a given image, we produce two bitstreams, to be transmitted over each channel. Should one of the channels fail, each individual description guarantees a minimum image quality specified by the user. However, if both descriptions arrive at destination, they are combined to produce a higher quality image than that achievable based on individual descriptions. We formulate a discrete optimization problem, whose solution gives parameters of the proposed encoder yielding optimal performance in an operational sense. Simulation results are presented.

Method for seamless splicing in a video encoder

Abstract: A method of achieving seamless switching of digitally compressed signals. The method includes the steps of identifying the point in a video signal where splicing to a second video signal is desired, and thereafter, maintaining adherence to certain parameters in the encoder buffer to ensure that the input signal is not being compressed at a rate that causes either underflow or overflow in the encoder buffer. The method also includes the steps of constraining the upper bound of the encoder buffer to ensure that data is not being outputted from the encoder buffer to the decoder buffer too slowly so as to cause an underflow of data in the decoder buffer. The method may also include the steps of constraining the lower bound of the encoder buffer to ensure that data is not being outputted from the encoder buffer to the decoder buffer too quickly so as to cause an overflow of data in the decoder buffer.

Code-spread CDMA using maximum free distance low-rate convolutional codes

Abstract: In code division multiple-access (CDMA) systems, maximum total throughput assuming a matched filter receiver can be obtained by spreading with low-rate error control codes. Previously, orthogonal, bi-orthogonal and super-orthogonal codes have been proposed for this purpose. We present in this paper a family of rate-compatible low-rate convolutional codes with maximum free distance. The performance of these codes for spectrum spreading in a CDMA system is evaluated and shown to outperform that of orthogonal and super-orthogonal codes as well as conventionally coded and spread systems. We also show that the proposed low rate codes will give simple encoder and decoder implementations. With these codes, any rate 1/n, n/spl les/512, are obtained for constraint lengths up to 11, resulting in a more flexible and powerful scheme than those previously proposed.

Adaptive error correction for a communications link

Abstract: A method of data transfer between a transmitter and a receiver over a communications link achieves maximum throughput by dynamically adapting a coding rate, and specifically an error correction encoder, as a function of a measured reverse channel signal parameter. The method comprises the steps of transmitting a signal from the transmitter to the receiver, the receiver receiving and measuring the signal to noise ratio of the transmitted signal. The receiver determines an appropriate code rate and encoding technique as a function of the measured signal to noise ratio and transmits an encoding identifier of the determined encoder to the transmitter. The transmitter encodes its data according to the encoding identifier and transmits the encoded message to the receiver. The receiver receives the encoded message and decodes the message according to the determined code rate and encoding technique.

Method and apparatus for non-sequential access to an in-progress video feed

Abstract: A method and system for delivering a live feed to a client is provided. According to one aspect of the invention, content data is generated by an encoder. Tag data that indicates locations of video frame data within the content data is generated while the content data is being generated. According to one embodiment, the tag data is generated by the encoder. According to an alternative embodiment, the tag data is generated by parsing the content data. The content data is at a location from which the content data is delivered to the client. The tag data is stored at a location from which the tag data may be used to provide the client non-sequential access to the content data. Before the encoder finishes generating the content data, a request is received for non-sequential access to the content data by the client, second content data is constructed based on the content data, the tag data and the request for non-sequential access, and the second content data is sent to the client.

Highly scalable wavelet-based video codec for very low bit-rate environment

Abstract: We introduce a highly scalable video compression system for very low bit-rate videoconferencing and telephony applications around 10-30 kbits/s. The video codec first performs a motion-compensated three-dimensional (3-D) wavelet (packet) decomposition of a group of video frames, and then encodes the important wavelet coefficients using a new data structure called tri-zerotrees (TRI-ZTR). Together, the proposed video coding framework forms an extension of the original zero tree idea of Shapiro (1992) for still image compression. In addition, we also incorporate a high degree of video scalability into the codec by combining the layered/progressive coding strategy with the concept of embedded resolution block coding. With scalable algorithms, only one original compressed video bit stream is generated. Different subsets of the bit stream can then be selected at the decoder to support a multitude of display specifications such as bit rate, quality level, spatial resolution, frame rate, decoding hardware complexity, and end-to-end coding delay. The proposed video codec also allows precise bit rate control at both the encoder and decoder, and this can be achieved independently of the other video scaling parameters. Such a scheme is very useful for both constant and variable bit rate transmission over mobile communication channels, as well as video distribution over heterogeneous multicast networks. Finally, our simulations demonstrated comparable objective and subjective performance when compared to the ITU-T H.263 video coding standard, while providing both multirate and multiresolution video scalability.

Method and apparatus for producing demand real-time television

Abstract: A demand television system comprising a broadcast encoder and a storage encoder The broadcast encoder encodes a real-time video frame sequence to form a broadcast bitstream and broadcasts the broadcast bitstream to a plurality of subscriber equipment, while simultaneously the storage encoder encodes the real-time video frame sequence to form a storage bitstream that is stored in an information server The subscriber equipment decodes the broadcast bitstream to display the broadcast program At any time, the subscriber equipment may request to review the information previously displayed in the broadcast bitstream As such, the storage bitstream is transmitted to the subscriber equipment The storage bitstream facilitates standard play of the previously broadcast information as well as trick play such as fast forward and fast reverse functions

Method and apparatus for compressing and decompressing images

Abstract: A method and apparatus for encoding (622) digital image data wherein a region of interest (606) can be specified either before the encoding process has begun or during the encoding process, such that the priority (616) of the encoder outputs are modified so as to place more emphasis on the region of interest, therefore increasing the speed and/or increasing the fidelity of the reconstructed region of interest. The system, therefore, enables more effective reconstruction of digital images over communication lines.

Apparatus and method for embedding/unembedding supplemental information

Abstract: A supplemental information embedding apparatus having a device for capturing image data, a first memory for storing image data indicating the presence of copyright protection, a first encoder for encoding the image data obtained by the image capturing means to a first format digital image data by n stages of processing (n is a positive integer greater than or equal to 2), a second memory for storing the first format digital image data produced in the digital image data producing means, a second encoder for encoding the image data indicating the copyright data supplied from the image capturing device or the first memory to a second format digital watermark by i stages of processing (i is a positive integer less than n), a third memory for storing the second format digital watermark encoded by the second encoder, a decoder for decoding the first format digital image data stored in the second memory up to a desirable stage of decoding, a divider for dividing the second format digital watermark stored in the third memory into predetermined format blocks, an embedder for dispersively embedding the blocks of the digital watermark into the image data encoded up to i stage by the first encoder or the irate data decoded up to i stage by the decoder, a third encoder for encoding the image data obtained by the embedder by exerting operations on and after j stage (j=n−1) of encoding so as to produce the first format digital image data, and a fourth memory for storing the first format digital image data encoded by the third encoder.

Method and apparatus for encoding and decoding a turbo code in an integrated modem system

Abstract: The present invention is a method and apparatus for encoding and decoding a turbo code. In the encoder, an interleaver interleaves and delays a block of input bits to generate interleaved input bits and delayed input bits. A first encoder generates a first, second, and third encoded bits. A second encoder generates a fourth encoded bit. A symbol generator generates a plurality of symbols which correspond to the input bits. In a decoder, a sync search engine detects a synchronizing pattern and extracts symbols from the encoded bits. An input buffer is coupled to the sync search engine to store the extracted symbols. A first soft-in-soft-out (SISO 1 ) is coupled to the input buffer to generate a first soft decision set based on the extracted symbols. An interleaver is coupled to the SISO 1 to interleave the first soft decision set. A second soft-in-soft-out (SISO 2 ) is coupled to the input buffer and the interleaver to generate a second soft decision set. A de-interleaver is coupled to the SISO 2 to de-interleave the second soft decision set. An adder is coupled to the SISO 1 and the de-interleaver to generate a hard decision set.

Image encoder and image decoder

Abstract: An image encoder for encoding the image data so as to make the image quality of a selected area better than that of the other areas without increasing the amount of data is disclosed. The encoder comprises an area selecting section for selecting a specific area in an image, an area position and shape encoding section for encoding the position and shape of the selected area, a coding parameter adjusting section for adjusting various parameters used to control the image quality and the amount of data in encoding a dynamic image and performing control so that the image quality of the area is encoded more preferably than the image quality of the other areas, a parameter encoding section for encoding the above parameters, a dynamic image encoding section for encoding input dynamic-image data by using the above various parameters, and an encoded data integrating section for combining encoded data by the area position and shape encoding section, the parameter encoding section and the dynamic image encoding section and transmit or store the data. An image decoder applied to the image encoder is also disclosed.

Data encoder/decoder for a high speed serial link

Abstract: An encoder/decoder is disclosed which is operative to convert an 8 bit value to a ten bit serial run length limited code for transmission over a serial data link. The encoding technique maintains DC balance within 2 bits over a single ten bit word and compensates for DC imbalance by inverting selected words in the transmission sequence to correct for a DC imbalance resulting from the transmission of a prior unbalanced word. One or more encoding lookup tables are employed at the encoder to map each byte into a ten bit run length limited code for serialization and transmission over the serial data link. A second decoding lookup table is employed at the decoder to map the received 10 bit run length limited code into the original 8 bit value.