Abstract: An expansion processor has a buffer defining unit for defining one of two buffers as a present inverse quantization buffer and defining one of two buffers as a present restoration buffer, an inverse quantization processor for inversely quantizing a quantized value read for each sample from a DCT data buffer, an IDCT processor for effecting an IDCT process on the inversely quantized data to restore time-domain audio data from frequency-domain data, a low-pass filter processor for removing a high-frequency component from the restored audio data, and an audio data output unit for outputting successive restored samples of audio data to a DAC to output sound from a speaker.

Abstract: Two related voiceband compression techniques are employed in order to enable an RF telecommunications system to accommodate data signals of high speed voiceband modems and FAX machines. A High Speed Codec enables the telecommunications system to pass voiceband modem and FAX transmissions at up to 9.6 kb/s. An Ultra-High Speed Codec supports voiceband modem and FAX transmissions up to 14.4 kb/s. The High Speed Codec operates using three 16-phase RF slots or four 8-phase RF slots, and the Ultra-High Speed Codec operates using four 16-phase RF slots. Because these codecs transmit information over several RF slots which can be contiguous, the slots within RF communication channels are dynamically allocated. The Dynamic Timeslot/Bandwidth Allocation feature detects and monitors the data transmission and forms a data channel from the necessary number of slots.

Abstract: A method and a system is provided for embedding and detecting additional information, such as copyright information, in audio data, so that a modification in the sonic quality due to the embedding is imperceptible to human beings, and does not drastically deteriorate the sonic quality.

Abstract: A time-scale modification method or apparatus performs time-scale modification (i.e., compression or expansion with respect to time) on original audio signals having waveforms. Adjacent wave segments are divided and cut from the waves of the original audio signals by various lengths. A certain number of samples are thinned out from each of the adjacent waveform segments to provide a reduced amount of data. Calculations are performed on the reduced amount of data to sequentially produce similarities between the adjacent wave segments in response to the various lengths. The similarities are evaluated to determine a length that provides a best similarity within the various lengths as a basic period. The waves of the original audio signals are divided and cut into two waves by the basic period. Time-scale modification is effected on the two waves to produce a mixed wave.

Abstract: Transceiver circuitry 1 comprises a first portion 10,20,30,41,50,100, having a first modulation means 41 operating at a first order of modulation, for transmitting and receiving voice signals; a second portion 20,30,42,50,100, having a second modulation means 42 operating at a second order of modulation, for transmitting and receiving digital signals at a higher data rate than is achievable by the first portion; and a data conversion means 20,30,100 operable to convert from or into voice signals intended for processing by the first portion into or from digital signals for processing by the second portion.

Abstract: A sound signal processing apparatus including extracting means for extracting from a composite sound signal, representing multiple sound sequences, digital sound signals corresponding to a portion of the composite sound signal. Each of the digital sound signals is individually sampled. Also included is a signal converter for converting the digital sound signals which have been extracted into analog sound signals.

Abstract: An audio reproducing apparatus comprises: audio decoding means for decoding an input audio signal frame by frame; data expanding/compressing means for subjecting data in a decoded frame to time-scale modification process; a frame sequence table which contains a sequence determined according to a given speed rate in which respective frames are expanded/compressed; frame counting means for counting the number of frames of the input audio signal; and data expansion/compression control means for instructing the dalta expanding/compressing means to subject the frame to one of time-scale compression process, time-scale expansion process, and process without time-scale modification process, with reference to the frame sequence table based on a count value output from the frame counting means, the data expanding/compressing means subjecting the audio signal to time-scale modification process in accordance with an instruction signal from the data expansion/compression control means.

Abstract: A method for identifying and categorizing an audio signal into subclasses to determine a subframe block size of a transform coder. A number of block sizes available for the transform coder is determined. An input audio signal is then sampled at predetermined time intervals to produce a plurality of samples, which are grouped into frames, in which each frame has an equal number of samples. The frames are analyzed in a time domain to produce at least one comparison index, after which an appropriate block size is selected for the transform coder.

Abstract: Loss-less data compression/decompression especially useful in a limited resource environment such as a handheld portable video game system allows graphics and/or attribute data to be efficiently and quickly decompressed on an as-needed basis in real time response to interactive user inputs. A two-level run-length-encoding is used to encode redundant patterns and redundant symbols. A common sentinel field format encodes whether data following the field is non-redundant data, a symbol run, or a pattern run. Compression ratios of 60% for representative symbol-mapped video display graphics/attribute files can be achieved.

Abstract: An audio information bit stream including audio control bits and audio data bits is processed for transmission in a communication system. The audio data bits are first separated into n classes based on error sensitivity, that is, the impact of errors in particular audio data bits on perceived quality of an audio signal reconstructed from the transmission. Each of the n different classes of audio data bits is then provided with a corresponding one of n different levels of error protection, where n is greater than or equal to two. The invention thereby matches error protection for the audio data bits to source and channel error sensitivity. The audio control bits may be transmitted independently of the audio data bits, using an additional level of error protection higher than that used for any of the n classes of the audio data bits. Alternatively, the control bits may be combined with one of the n classes of audio data bits and provided with the highest of the n levels of error protection.

Abstract: A data compression apparatus for data compressing an information signal, which is in n-level form, n being larger than 2. The data compression apparatus includes an input terminal for receiving the n-level information signal, an entropy coder, such as an arithmetic coder having an input for receiving an input signal, which is adapted to carry out a lossless encoding step on the input signal, so as to obtain a data compressed output signal at an output. The apparatus further includes a prediction filter for carrying out a prediction step on the n-level information signal so as to obtain a prediction signal and a probability signal determining unit for generating the probability signal in response to the prediction signal. An output terminal is available for supplying the data compressed output signal. Further, a data expansion apparatus is disclosed.

Abstract: Entropy encoding and decoding of data with a code book containing variable length entropy-type codes that are assigned to variable length input symbol groupings. The variable length input sequences are identified by scanning an input channel, such as a live broadcast, non-volatile data storage, or network connection (e.g., LAN, WAN, Internet). Each time a symbol grouping is recognized, a corresponding entropy-type code is output as a replacement for the input stream. Decoding is the inverse process of encoding, where a code word is looked up in the code book and the corresponding original input is obtained.

Abstract: In a communications system, multi-rate coding in accordance with the invention is implemented to generate multiple representations of an audio signal at different rates. These representations contain equivalent and/or various amounts of audio information. In an illustrative embodiment, at least one of the representations is a core representation containing core audio information. The remaining representations are enhancement representations containing enhancement audio information. The core representation is necessary for recovering the audio signal with minimal acceptable quality. Such quality is enhanced when the core representation, together with one or more of the enhancement representations, is used to recover the audio signal.

Abstract: A voice coding apparatus and a voice decoding apparatus are provided in order to improve the coding and decoding efficiency by eliminating useless coding levels. The apparatuses comprises an input terminal 24A for inputting the sub-frame length and for delivering it to the sub-frame dividing circuit 10 and an unit length calculating circuit 32A. The unit length calculating circuit calculates the unit length which determine a pulse interval from the sub-frame length supplied from the input terminal 24A and from a fundamental vector length supplied from the input terminal 26A, and delivers the obtained unit length to the table conversion circuit. The table designing circuit 34A designs the pulse position table based on the number of pulses supplied by the input terminal 26A and the fundamental vector length supplied by the input terminal 26A and deliver the unit length to the table circuit 36A.

Abstract: A method of managing multiple channels of audio data in an audio system having multiple speakers. A first channel signal is selectively passed through a software high-pass filter to selectively drive a first one of the speakers. A plurality of channel signals are selectively summed in software to generate a composite signal and the composite signal passed through a software low-pass filter to selectively drive a second one of the speakers.

Abstract: A portable audio player stores a large amount of compressed audio data on an internal disk drive, and loads a portion of this into an internal random access memory (RAM) which requires less power and less time to access. The audio player plays the data stored in RAM and monitors the amount of unplayed data. When the amount of unplayed data falls below a threshold, additional data is copied from the disk drive into RAM. Because the time necessary to copy a block of data from the disk drive to RAM is much less than the amount of time it takes to play the same block of audio data from RAM, this approach minimizes the amount of time that the disk drive must be operated, and thus minimizes the amount of power consumed by the system.

Abstract: A method for training of auditory and graphical discrimination in humans is provided within an animated game environment. The method provides a number of stimulus sets, each stimulus set having a target phoneme and a plurality of associated foils (similar sounding phonemes). Upon initiation of a trial, a target phoneme is presented to a subject. Subsequently, the target phoneme is presented to the subject, along with one of the associated foils, in randomized order. As the target phoneme and associated foil is presented, a graphical animation associates the target and foil each with its own graphical image. The subject then designates identification of the target phoneme by selecting its associated image. Speech processing is used to provide multiple levels of emphasis for enhancing the subject's ability to discriminate between the target phoneme and the foils.

Abstract: Of I, P, and B pictures contained in an MPEG 2 data stream, only the I picture is subjected to encryption such as scramble processing. Scramble rule data used at that time is stored in the lead-in area of an optical disk. A software DVD decoder reads the scramble rule data stored in the lead-in area, and its certification control module descrambles only the I picture. With this processing, the CPU power required for descramble processing can be reduced, and motion picture data can be decoded by the software DVD decoder in real time.

Abstract: The present invention relates to a system and method for automatic synchronization in multimedia presentations. According to an embodiment of the present invention, when a data stream is compressed, delay which would normally be compressed out is replaced by a delay token which indicates a length of time of the delay. When a data stream is decompressed and presented, the delay tokens may either be used or ignored. In particular, when data streams are presented together in a multimedia presentation the delay tokens may be used to synchronize the various data streams of the multimedia presentation. Otherwise, when data streams are presented alone without the other data streams of a multimedia presentation or are not part of a multimedia presentation, the delay tokens may be ignored. In such cases when the delay token is ignored, any data stream delay is simply skipped since there is no need to synchronize with other data streams.

Abstract: A monolithic integrated circuit for providing enhanced audio performance in personal computers. The monolithic circuit includes a wavetable synthesizer; a full function stereo coding and decoding circuit including analog-to-digital and digital-to-analog conversion; data compression, and mixing and muxing of analog signals; a local memory control module for interfacing with external memory; a game-MIDI port module; a system bus interface; and a control module for compatibility and circuit control functions.

Abstract: A fast and economical method for speeding up an audio signal without changing pitch can be accomplished by eliminating unneeded information from an audio signal. First, the signal is divided into chunks (frames or subframes), on which a mathematical manipulation such as a Fourier transformation is performed to identify the amplitudes of the componenet sinusoids (sines and cosines). These absolute values of the sine and cosine amplitudes for each frequency are averaged together, and the highest value(s) represents the signature, or dominant frequency/frequencies. The dominant frequency/frequencies or signatures from one chunk are compared to those of the next, and when identical the latter unit is marked as redundant. The final step consists of discarding redundant chunks from the original data, thus providing a shortened signal for replay. The pitch will not change because the only modification to the original signal was the elimination of redundant data.

Abstract: A method of automatically selecting processing parameters for encoding digital content. Metadata containing the genre of the digital content, receiving the compression level selected for encoding the digital content is received. An algorithm selected for encoding the digital content is received. And a previously defined table to select the processing parameters for encoding the digital content based on the genre of the content, the compression level selected and the algorithm selected is indexed and the processing parameters are retrieved.

Abstract: Data obtained by dividing digital data into a low-frequency signal and a high-frequency signal and converting the low-frequency signal and a signal obtained by subjecting the high-frequency signal to a lossless compression coding process into a transmission format is transmitted. The lossless compression coding process is effected by dividing a digital data string input in the unit of sample constructed by a preset number of bits into block units constructed by a plurality of samples, removing a constant number of bits having a common value from each sample starting from the sign bit side thereof for all of the samples in each block to form a compressed block and attaching information indicating the number of removed bits to the compressed block.

Abstract: An apparatus and method for training of auditory and graphical discrimination in humans is provided. The method and apparatus provides a number of stimulus sets, each stimulus set having a target phoneme, and associated grapheme, and a number of distractor phonemes, and associated graphemes. Upon initiation of a trial, a target phoneme is presented to a subject. A stimulus stream is then prepared that consists of a random sequence of distractor phonemes. Located within the sequence of distractor phonemes is the target phoneme. The stimulus sequence is presented to the subject for identification of the target phoneme within the sequence. Speech processing is used to provide multiple levels of emphasis for enhancing a subject's ability to discriminate between similarly sounding phonemes. The processing is applied to the presentation of the target phoneme and the stimulus stream.

Abstract: A method (100) of compressing a digital signal that is parametrically modeled and encoded includes the steps of storing (102) the digital signal in a memory in a plurality of frames having a plurality of parameters in each frame of the plurality of frames, wherein the digital signal was encoded at a higher rate and converting the digital signal to a lower rate by selecting (106) from each frame of the plurality of frames a subset of the plurality of parameters and discarding (108) the subset of the plurality of parameters within each frame of the plurality of frames.

Abstract: A data signal compression technique for real-time voice and data processing where the digitized signal is first compressed to obtain a first compressed signal, and the first compressed signal is then compressed again to obtain a second compressed signal. Within a digital signal processor, digital signals first undergo time scale compression after which the compressed signals undergo audio compression to achieve multiple-compressed signals. Upon reception in a second digital signal processor, the multiple-compressed signals are correspondingly decompressed to achieve a high-quality estimation of the original digital signals.

Abstract: An apparatus and method for training the cognitive and memory systems in a subject is provided. The apparatus and method incorporates a number of different programs to be played by the subject. The programs artificially process selected portions of language elements, called phonemes, so they will be more easily distinguished by the subject, and gradually improves the subject's neurological processing and memory of the elements through repetitive stimulation. The programs continually monitor a subject's ability to distinguish the processed language elements, and adaptively configures the programs to challenge and reward the subject by altering the degree of processing. Through adaptive control and repetition of processed speech elements, and presentation of the speech elements in a creative fashion, a subject's cognitive processing of acoustic events common to speech, and memory of language constructs associated with speech elements are significantly improved.

Abstract: An adaptive linear predictor is used to predict samples, and residuals from such predictions are encoded using Golomb-Rice encoding. Linear prediction of samples of a signal which represents digitized sound tends to produce relatively low residuals and those residuals tend to be distributed exponentially. Accordingly, linear prediction combined with Golomb-Rice encoding produces particularly good compression rates with very efficient and simple implementation. The accuracy of the linear predictor is improved by including, in the prediction of a current sample of a first channel of the digitized signal, look-ahead sample data from a corresponding second channel of the digitized signal. For example, prediction of a right channel sample of a digitized, stereo, audio signal is improved by inclusion of look-ahead left channel sample data in the right channel sample predictor.

Abstract: An audio decoder is described which supports simple sound-effect generation. The audio decoder includes a direct access pulse code modulation (PCM) first-in-first-out buffer (FIFO) to support simple sound effect generation. In one embodiment, the audio decoder additionally includes an input buffer, a decoding module, and an output interface. The input buffer buffers incoming data frames for the decoding module to retrieve and convert to a sequence of decoded audio samples. The FIFO is configured to receive and buffer audio sound effect samples from a control component external to the audio decoder. The output interface is configurable to retrieve decoded audio samples from the decoding module and audio sound effect samples from the FIFO. Any retrieved audio sound effect samples are included in a digital audio output signal provided by the output interface. The digital audio output signal may be provided directly to a digital-to-analog converter for sound reproduction.

Abstract: A multimedia decoder is provided with an audio decoder bypass module for forwarding undecoded audio bitstreams directly to external system components. In one embodiment, the multimedia decoder includes an audio decoder, and a bypass module. The audio decoder operates on the data in an audio bitstream buffer to convert at least a portion of the audio bitstream into a set of digital audio signals. The bypass module is configured to provide the full information content of the audio bitstream to an external system component which may be able to convert a greater portion of the audio bitstream into a second set of digital audio signals. As the audio decoder and bypass module each retrieve data from the audio bitstream buffer, they each use a pointer to track which location of the buffer to access next.

Abstract: In coding of an audio signal, coded signals with low quality and bit rate on the one hand and coded signals with high quality and bit rate on the other hand are transmitted to a decoder. At first, the audio signal is coded with low bit rate and is transmitted to the decoder before an additional coded signal is transmitted to the decoder, which either alone or together with the first coded signal upon decoding thereof provides a decoded signal with high quality within the decoder. In this manner, a low-quality decoded signal is generated first in the decoder before decoding of the high-quality signal is possible.

Abstract: An audio decoder makes use of various component sharing techniques and operates to efficiently prevent deadlock without introducing decoding errors or adding significant complexity to the audio decoder. In one embodiment, the audio decoder comprises a bitstreamer, a synchronization controller, a decode controller, a memory module, a data path, and an output buffer. The bitstreamer retrieves compressed data and provides token-aligned data to the synchronization controller and decode controller. The synchronization controller initially controls the bitstreamer to locate and parse audio frame headers. After each frame header is parsed, the decode controller controls the bitstreamer to parse the variable length code compressed transform coefficients. The coefficients are passed to the memory module and data path which operate under the control of the decode controller to inverse transform the coefficients and produce digital output audio data.

Abstract: An audio signal processor forms gaps or guard bands in sequences of blocks conveying encoded audio information and time aligns the guard bands with video information. The guard bands are formed to allow for variations in processing or circuit delays so that the routing or switching of different streams of video information with embedded audio information does not result in a loss of any encoded audio blocks.

Abstract: An information encoding method for encrypting and encoding information signals, such as PCM audio signals, in which the information signals can be reproduced with low quality even in the absence of the key information for encryption. For carrying out the information encoding method, the input PCM signals are converted by a transform unit into frequency signal components which are encoded by a signal component encoding unit. High frequency range side signal components are sent to an Ex-OR gate to take an Ex-OR of the high frequency range side signal components with a pseudo random bitstring from a pseudo random bitstring generating unit. A codestring generating unit 1606 generates a codestring having the low frequency range side components from a signal component encoding unit and the encrypted high frequency range side components from the Ex-OR gate.

Abstract: A video and voice signal processing apparatus is provided. The apparatus includes a signal receiving circuit for receiving an input signal containing a plurality of frames, each frame having an encoded voice signal block and an encoded video signal block. The signal receiving circuit separates the encoded voice signal block from the encoded video signal block in each frame. A voice signal processor converts the encoded voice signal block into a voice signal. Also included is a video extracting circuit which decimates a plurality of encoded video signal blocks and extracts one of the encoded video signal blocks as a representative video signal. A video signal processor converts the representative video signal into a video signal.

Abstract: The spectral range of a stochastic time series of information, including unvoiced speech is reduced to allow transmission over a substantially narrowed frequency band. Sets of autoregressive (AR) parameters are identified for successive time windows of the original time series and of subsequent stages of subsampled reduced-spectrum models of each window of the original time series are used. The AR parameters are transmitted together with subsampled windows of the original data. These AR parameters are used to reconstruct a least square stochastic estimate of the transmitted subsampled time series in a backwards manner from the most subsampled spectrum back to the original spectrum using a sequence of predictive feedback algorithms. Past prediction outputs are feedback for prediction whenever samples are missing. This process yields a high quality reconstructed signal that preserves not only speech parameters and intelligibility, but also near-natural speaker identifiability.

Abstract: An apparatus and method for training the sensory perceptual system in a language learning impaired (LLI) subject is provided. The apparatus and method incorporates a number of different programs to be played by the subject. The programs artificially process selected portions of language elements, called phonemes, so they will be more easily distinguished by an LLI subject, and gradually improves the subject's neurological processing of the elements through repetitive stimulation. The programs continually monitor a subject's ability to distinguish the processed language elements, and adaptively configures the programs to challenge and reward the subject by altering the degree of processing. Through adaptive control and repetition of processed speech elements, and presentation of the speech elements in a creative fashion, a subject's temporal processing of acoustic events common to speech are significantly improved.

Abstract: A sample rate conversion system and method uses a digital signal processor (DSP) and a separate sample rate conversion circuit (SRC) to perform multiple stream conversion and mixing of different rate input audio streams. The sample rate conversion system converts data, such as multiple streams of digital audio data sampled at different rates, and performs interpolation, decimation, FIR filtering, and mixing of multiple streams of data using the separate SRC. The SRC uses two bidirectional I/O memories for alternately storing input and output data as part of a sample rate converter. When the sample rate converter writes output to one of the bidirectional memories, it has the option of summing the data with the data already stored in the same I/O memory. Therefore a separate digital signal processor can use the sample rate converter circuit to perform some of the mixing for the multiple streams.

Abstract: A sample rate conversion system and method uses a digital signal processor (DSP) and a separate sample rate conversion circuit (SRC) to perform multiple stream conversion and mixing of different rate input audio streams. The sample rate conversion system converts data, such as multiple streams of digital audio data sampled at different rates, and performs interpolation, decimation, FIR filtering, and mixing of multiple streams of data using the separate SRC. The SRC uses two bidirectional I/O memories for alternately storing input and output data as part of a sample rate converter. When the sample rate converter writes output to one of the bidirectional memories, it has the option of summing the data with the data already stored in the same I/O memory. Therefore a separate digital signal processor can use the sample rate converter circuit to perform some of the mixing for the multiple streams.

Abstract: A method of processing exponent data in an audio decoder. A first block of audio data is received including encoded exponent data. The encoded exponent data is packed into packed encoded words and stored in memory. Exponents are generated from the packed encoded words in memory for processing the first block of audio data. A second block of audio data is received. A determination is made as to whether a reuse flag has been set for the second block, and if the reuse flag has been set, exponents are generated from the packed encoded words memory for processing the second block of data.

Abstract: An apparatus and method for training the sensory perceptual system in a language learning impaired (LLI) subject is provided. The apparatus and method incorporates a number of different programs to be played by the subject. The programs artificially process selected portions of language elements, called phonemes, so they will be more easily distinguished by an LLI subject, and gradually improves the subject's neurological processing of the elements through repetitive stimulation. The programs continually monitor a subject's ability to distinguish the processed language elements, and adaptively configures the programs to challenge and reward the subject by altering the degree of processing. Through adaptive control and repetition of processed speech elements, and presentation of the speech elements in a creative fashion, a subject's temporal processing of acoustic events common to speech are significantly improved.

Abstract: An audio signal compression and decompression method and apparatus that provide lossless, realtime performance. The compression/decompression method and apparatus are based on an entropy encoding technique using multiple Huffman code tables. Uncompressed audio data samples are first processed by a prediction filter which generates prediction error samples. An optimum coding table is then selected from a number of different preselected tables which have been tailored to different probability density functions of the prediction error. For each frame of prediction error samples, an entropy encoder selects the one Huffman code table which will yield the shortest encoded representation of the frame of prediction error samples. The frame of prediction error samples is then encoded using the selected Huffman code table. A block structure for the compressed data and a decoder for reconstructing the original audio signal from the compressed data are also disclosed.

Abstract: An audio decoder which can reduce a memory circuit capacity necessary for performing a series of decoding processes and can perform a down mixing. The audio decoder decodes audio data of a plurality of channels encoded in a frequency domain by using a time base to frequency base conversion. After a down mixing process was performed to the audio data of the frequency domain by frequency domain down mixing circuit, it is converted into audio data of a time domain by frequency base to time base converting circuit, thereby reducing memories by the number corresponding to the reduced number of channels. Further, by executing an inverse quantizing process of each channel and a frequency base to time base converting process of each channel by pipeline processes, a work buffer can be shared in both of the processes.

Abstract: An encoding system and an encoding method for encoding a digital signal having at least a first and a second digital signal component. The encoding system includes a splitter unit for dividing the bandwidth of the digital signal components into M successive frequency bands, and generating in response to the digital signal components M sub signals (SB.sub.m1,SB.sub.

Abstract: To modify the temporal scale of recorded speech, relative stress and relative speaking rate terms are computed for individual sections, or frames, of the speech. These terms are then combined into a single value denoted as audio tension. For a nominal time-scale modification rate, the audio tension is employed to adjust the modification rate of the individual frames of speech in a non-uniform manner, relative to one another. With this approach, compressed speech can be reproduced at a relatively fast rate, while remaining intelligible to the listener.

Abstract: A method and apparatus for providing high-speed data compressing without sacrificing the quality of data reconstruction. Each input vector or block of original data is expressed as a combination of a codebook index and an error differential, or as a compressed version of the original block of data, depending on whether the total number of bits needed to express the input vector as a combination of a codebook index and an error differential is less than the total bits needed to send the compressed version of the original block of data. In one embodiment, the flexibility to express video data in a compressed format or as a combination of a codebook index plus an error differential is provided through a video system employing a codebook, a scalar quantizer, and an entropy coder.

Abstract: Disclosed is a partitioning procedure for designing MPEG decoders, AC-3 decoders, and decoders for other audio/video standards. The procedure provides that some specified decoding functionality be implemented exclusively in the form of hardware and certain other specified decoding functionality be provided exclusively as firmware or software. A video decoder designed according to this procedure includes the following elements: (a) firmware or software for implementing, in conjunction with a CPU, video header processing functions; and (b) hardware for implementing preparsing assist, macroblock reconstruction, and video display control functions. An audio decoder designed according to this procedure includes the following elements: (a) firmware or software for implementing, in conjunction with a CPU, decoding fields containing parameters for processing the audio data; and (b) hardware for implementing matrixing and windowing functions on the audio data.

Abstract: This invention is for a single monolithic audio processing integrated circuit which includes a synthesizer module, a CODEC module and an external serial data port in the CODEC module for bi-directional serial data communication between the CODEC module and an external serial data device, such as a digital signal processor. A serial data path between the synthesizer module and the CODEC module is also included.

Abstract: An apparatus for generating the interrupt information includes an addressing device for generating the specified address information for specifying the desired information stored in a memory device, a readout address generating device for generating the readout address information of the desired information stored in the memory device, and a comparator device for comparing the specified address information from the addressing device and the readout address information from the readout address generating device and for generating the interrupt information in case of coincidence of the specified address information and the readout address information and supplying the interrupt information to a central processing unit.

Abstract: A method and apparatus for encoding an input signal in which the input signal is transformed into frequency components which are encoded by quantization from one encoding unit to another. The information specifying zero encoding units among the encoding units in which encoding is done on the assumption that all frequency components contained in them are deemed to be zero are encoded, is encoded, while the quantization step information of the zero encoding units is outputted without encoding. With the encoding method and apparatus, the number of the encoding bits may be decreased while deterioration in the input signal is prohibited for improving the encoding efficiency.