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: In a method for recording video images and the associated sound with a reduced resolution, in such a manner that the recorded information can be reproduced by means of standard equipment, video frames are filtered so as to derive browse sets having a reduced number of pixels. A number of 16 browse sets are stored in a frame memory in order to form a mosaic frame of 4×4 browse sets, after which the mosaic frame is recorded. Audio samples are processed to form audio browse samples: the most significant half (MSH) of the bits of a first audio sample is used as the MSH of an audio browse sample and the MSH of a ninth audio sample is used as the least significant half (LSH) of the bits of this audio browse sample, which is subsequently recorded. In this way, the information required to edit a recording is reduced by a factor of 16.

Abstract: A method and apparatus for communicating both voice and control data between a communication device (such as a cellular phone) and an external accessory (such as a hands-free kit) over a data bus. The method includes formatting a sequence of bits into a repeating sequence of first time slots and second time slots, transmitting the voice data in the first time slot, and transmitting the control data in the second time slot. Notably, a first bit of each of the second time slots comprises a clock bit that alternates between a high value and a low value (e.g. a ‘1’ or a ‘0’) as between consecutive second time slots.

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: A system (20) transmits a digital data stream (22) using a lossy compression service (24) having a communication path (28) through which compressed signals propagate. The system includes a data conditioner (36) configured to condition the received digital data stream (22) to produce data subsets (66). A frame generator (36) expands each of the data subsets (66) to generate distinct signal sets (68). The distinct signal sets (68) are expressed in a data size greater than the data subsets (66) to compensate for lossy compression cause by the lossy compression service (24). A transmitter is in communication with the frame generator (36) and transmits digitized ones of the distinct signal sets (68) in frames of a digital bit stream (126) over the communication path (28).

Abstract: A method for embedding a non-audio file into a compressed audio file is disclosed. The disclosed method is capable of decoding and constructing previously encoded audio files with embedded non-audio information. The method of the present invention also includes steps for embedding the non-audio information at the proper time within the compressed audio file, wherein the decoded non-audio information can link to an Internet document for viewing during playback of the decoded audio source.

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: A data processing device uses a portion of a random access memory as an output buffer for holding a frame of PCM sample data which is being output after being processed by a processing unit within the processing device. Fine grained synchronization between a reference clock and a stream of PCM data frames is provided by transferring only a portion of selected frame of PCM sample data PCM(n+1), in response to a time difference 971. A breakpoint address is determined to delineate the portion of the selected frame that is to be transferred. A sorted list of the addresses of the discontinuities is maintained in breakpoint queue. Since the buffer is managed in a FIFO manner, a single breakpoint register is sufficient to monitor addresses as they are provided by an address register for accessing the random access memory. When a breakpoint is detected, the breakpoint queue and the breakpoint register is updated by an update task 802.

Abstract: A proxy server has a connection to a client computer and to a remote server over the Internet. The proxy server receives a request for an audio file from the client computer and, in response, transmits a requests for the audio file to the remote server. Upon receiving the audio file, the proxy server determines whether transcoding of the audio file is appropriate. If appropriate, the proxy server transcodes the audio file received from the remote server and then transmits the transcoded audio file to the client. Transcoding may include changing the audio file type, compressing the audio file, reducing the number of audio channels, or reducing the sampling rate of the data. The proxy server determines the extent and type of transcoding to be performed on the audio file as the audio file is downloaded from the remote server.

Abstract: A frequency-domain audio coder selects among different entropy coding modes according to characteristics of an input stream. In particular, the input stream is partitioned into frequency ranges according to some statistical criteria derived from a statistical analysis of typical or actual input to be encoded. Each range is assigned an entropy encoder optimized to encode that range's type of data. During encoding and decoding, a mode selector applies the correct entropy method to the different frequency ranges. Partition boundaries can be decided in advance, allowing the decoder to implicitly know which decoding method to apply to encoded data. Or, adaptive arrangements may be used, in which boundaries are flagged in the output stream by indicating a change in encoding mode for subsequent data. For example, one can create a partition boundary which separates out primarily zero quantized frequency coefficients, from primarily non-zero quantized coefficients, and then apply a coder optimized for such data.

Abstract: There is provided a waveform data time expanding and compressing device, which includes a waveform memory for storing data of a PCM waveform; a block address memory for storing addresses of respective blocks, the respective blocks having a length equal to a wavelength of a pitch as a trend of the PCM waveform or an integral multiple thereof; a parameter determining unit for determining an expansion and compression parameter; and a waveform reproducer for carrying out waveform reproduction by determining a reading number for waveform data in a certain block in response to the expansion and compression parameter, sequentially reading out the block addresses according to the determined reading number, and reading out the PCM waveform data based on the block

Abstract: A sound decompressing apparatus which achieves special reproducing, also known as trick play, two examples of which are forward search reproducing and reverse search reproducing, by selecting and decompressing frames containing sound data at fixed or predetermined intervals. The apparatus may adjust the output level of reproduced sounds during special reproducing.

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: An audio encoding system includes quantization, phase matching, compressed domain processing and other improvements enabling high fidelity, low bit rate encoding systems to be formed. In a preferred embodiment a novel transient detector is used to divide an audio source into transient, low frequency non-transient and high frequency non-transient regions. Low frequency non-transients are sinusoid modeled and the residual is low frequency noise modeled. Transients are transform coded and high frequency non-transients and transform coded residual is high frequency noise modeled. The preferred embodiment also includes novel sinusoid, transform coded and noise quantization, among other methods for providing high fidelity data reduction and for interfacing sinusoid modeling and transform coding. Compressed domain time compression and expansion are further achieved with no detrimental effect on transients.

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.

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.

Abstract: A down-sampling system for digital waveforms performs real-time, “on the fly”, conversions and results in data of acceptable quality for many applications including applications dealing primarily with speech data. The down-sampler comprises a weight matrix calculator and a loop in which the system takes the input data from the producer's data stream, and at one chunk at a time, the system generates the output data. The loop comprises an input receiver, a chunk receiver, an output chunk generator, a chunk decider for deciding whether there is another chunk in the input, and an input decider for deciding whether there is more input.

Abstract: A time-scale modification method or apparatus is basically designed to effect a time-scale modification process (i.e., expansion or compression with respect to time) on rhythm source signals containing waves such that rhythm sounds are not substantially changed in pitches. Herein, attack positions are detected from the rhythm source signals by using thresholds which are determined in advance. Hence, the time-scale modification process is performed on intermediate signal portions of the rhythm source signals between the attacks in accordance with a desired time-scale modification factor. Then, the intermediate signal portions subjected to the time-scale modification process are smoothly connected with other signal portions such as the attacks and their proximal portions, which are not subjected to the time-scale modification process.

Abstract: An audio decoding device for decoding coded audio information with multiple channels includes a coded information memory section for storing the coded audio information; an information transmission section for reading the coded audio information stored at an arbitrary position in the coded information memory section; and an audio decoding section for decoding the coded audio information read by the information transmission section and outputting the resultant audio information in accordance with a time axis.

Abstract: Dolby AC-3 and MPEG-2 audio permit the transmission of audio signals with more than two independent audio channels. If a reproduction device has only a two-channel audio decoder (DEC1) then an external multi-channel audio decoder (DEC2) can be used for multi-channel sound reproduction. If the audio reproduction at the same time accompanies video reproduction, then a synchronization method is required in order to achieve lip synchronism between picture and sound. According to the invention, for the synchronization of a first decoder (DEC1), which merely has two-channel compatibility, with a second decoder, which has multi-channel compatibility, a counting variable is allocated a value (F) which is produced from system parameters such as the data coding method used, the transmission speed and/or the data rate. The data are received by the first decoder and output by the latter to the second decoder, the counting variable being decremented or incremented respectively for a specific volume of data.

Abstract: A message duration indicator apparatus and method for use with a portable voice messaging transceiver including a message recording memory, a microphone, a remaining message memory indicator and an indicator controller provide an indication to the user of the approaching end of the message period to stimulate the user to record a message in as short a time as possible.

Abstract: An audio encoder applies an adaptive block-encoding process to segments of audio information to generate frames of encoded information that are aligned with a reference signal conveying the alignment of a sequence of video information frames. The audio information is analyzed to determine various characteristics of the audio signal such as the occurrence and location of a transient, and a control signal is generated that causes the adaptive block-encoding process to encode segments of varying length. A complementary decoder applies an adaptive block-decoding process to recover the segments of audio information from the frames of encoded information. In embodiments that apply time-domain aliasing cancellation (TDAC) transforms, window functions and transforms are applied according to one of a plurality of segment patterns that define window functions and transform parameters for each segment in a sequence of segments.

Abstract: A technique for entropy coding information relating to frequency domain audio coefficients. For portions of a frequency spectrum having a predominate value of zero, a multi-level run length encoder statistically correlates sequences of zero values with non-zero values and assigns variable length code words. An encoder uses a specialized code book generated with respect to the probability of receiving an input sequence of zero-valued spectral coefficients followed by a non-zero coefficient. A corresponding decoder associates a variable length code word with a sequence of zero value coefficients adjacent a non-zero value coefficient.

Abstract: A method for adaptively training a human subject to process, and to distinguish between, similar acoustic events that are common in spoken language is provided. The method utilizes sequences of up/down frequency sweeps, of varying frequency and duration, and having varying inter stimulus intervals (ISI) between the frequency sweeps. A sequence is presented to the subject for order identification. The subject must listen to the up/down order of a sequence, and signal identification of the up/down order according to what s/he heard. Signal identification is provided utilizing a computer display, a mouse, and graphical buttons corresponding to the up/down frequency sweeps. Correct order identification causes the process to adaptively reduce the ISI separating the frequency sweeps, to reduce the duration of the frequency sweeps, to alter the frequency of the frequency sweeps, and to increase the number of frequency sweeps within a sequence.

Abstract: A method and apparatus for decoding a multi-channel audio bitstream in which adaptive frequency domain downmixer (3) is used to downmix, according to long and shorter transform block length information (17), the decoded frequency coefficients of the multi-channel audio (12,13,14,15) such that the long and shorter transform block information is maintained separately within the mixed down left and right channels. In this way, the long and shorter transform block coefficients of the mixed down let and right channels can be inverse transformed adaptively (4,5,6,7) according to the long and shorter transform block information, and the results of the inverse transform of the long and short block of each the left and right channel added together (8,9) to form the total mixed down output of the left and right channel.

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 and system for diagnosing data-processing system performance. Initially, unique audible sounds are associated with particular performance indicators within the data-processing system. Thereafter, performance indicators are identified, one or more of which indicate data-processing system performance. A diagnostic is then periodically run to detect performance indicators within the data-processing system. A unique audible sound is then generated associated with a particular performance indicator, in response to detecting the status of particular performance indicator via the diagnostic, such that potential data-processing system failures may be recognized by identifying the unique audible sound. The unique audible sound may be continuously generated at varying durations and volumes to indicate the presence of system failures. A trained user, accustomed to particular audible sounds, can identify and diagnose system failures by analyzing unique audible sounds generated by the data-processing system.

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.

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: A pitch of a tone to be generated is designated, and simultaneously control information to be used for time-axis stretch/compression control is generated. Discrete locations of waveform data to be read out from memory are designated with the time axis of the waveform data controlled to be stretched or compressed in accordance with the control information, and part of the waveform data at the designated locations are read out at a rate corresponding to the designated pitch. For example, virtual read addresses corresponding to the control information and actual read addresses corresponding to the designated pitch are generated, and the actual read addresses are controlled, at the individual discrete locations, to follow the virtual addresses.

Abstract: An encoding method and apparatus and a decodings method and apparatus in which the encoded information is decreased in volume and in which the encoding and decoding operations are performed with a smaller processing volume and a smaller buffer memory capacity. The apparatus includes a low range signal splitting circuit for separating low-range side signal components from L and R channel signals converted by a transform circuit into spectral signal components, and a channel synthesis circuit for synthesizing (L+R) channel signal components from the L and R channel spectral signal components.

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 audio playback apparatus for an input data stream (d0) delivered by an audio data source (1) in a standardized data format (F1) is disclosed comprising an input decoder (2) for forming a first data stream (d1), which contains the audio information and is separated by means of an output decoder (5) into its audio signal components (L, R). In addition to the first data stream (d1), the input decoder (2) generates a second data stream (d2) which corresponds to the data contained in the data field (D1) of the standardized data format (F1), and which is fed to an additional decoder (3) that detects a second data format (F2) possibly contained in the second data stream (d2) and passes the audio information of the second data format (F2) as a third data stream (d3) to the output decoder (5).

Abstract: Improved down mixing of audio channels of compressed digital audio signals by down mixing in the frequency domain. Fast virtual transform is applied to transform short DCT coefficients into long DCT coefficients, and down mixing is performed on the long DCT coefficients. Inverse discrete cosine transform is performed on the down mixed set of long DCT coefficients, generating signals in the windowing domain. The windowing domain signals are then overlapped and added to generate time domain signals suitable for further amplification. Down mixing in the frequency domain reduces the number of computations required.

Abstract: A digital sound processor for processing multiple standard sound signals and including an audio source which is connected to a digital control input of the sound processor and generates, via externally or internally applied control signals, an audible signal or an audible signal sequence which is fed via the output devices of the sound processor to reproducers.

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: 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.

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 which includes a coefficient memory and an arithmetic logic unit (ALU) can implement an efficient method for calculating a gain value specified by a range control field. In one embodiment, the audio decoder comprises coefficient memory, an ALU, frame control logic, and ALU control logic. The frame control logic extracts a range control field value from an audio packet header and provides it to the ALU control logic. The ALU control logic takes the binary representation of the range control field value and uses it to provide a sequence of addresses to the coefficient memory. In response to the sequence of addresses, the coefficient memory provides a sequence of pre-calculated factors to the ALU. The ALU control logic further directs the ALU to determine the product of the pre-calculated factors in the sequence. As a final step in finding the gain value, the ALU control logic may provide a shift instruction to the ALU.

Abstract: An audio decoder converts a linear PCM audio data packet into two concurrently provided digital audio sample sequences: a high-quality sequence and a decimated sequence. In one embodiment, the audio decoder is part of an audio system that further includes two audio devices. The first audio device is configured to produce an audio signal from a 96 kHz sequence, and the second audio device expects a 48 kHz sequence. The audio decoder includes an input interface, an arithmetic logic unit (ALU), and two output buffers. The input interface is configured to receive a linear PCM audio data packet and to reconfigure bytes as necessary to reconstruct a sequence of unscaled audio samples. The ALU multiplies each of the unscaled audio samples by a gain factor and buffers the resulting scaled audio sample sequence in a first output buffer.

Abstract: In a method for the cascaded coding and decoding of audio data the spectral components of the short-time spectrum associated with a data block are formed for each data block with a certain number of time input data, the coded signal is formed, by quantization and coding, on the basis of the spectral components for this data block and using a psycho-acoustic model to determine the bit distribution for the spectral components, whereupon time output data are obtained by decoding at the end of each codec stage.To prevent a deterioration in the sound quality in codec cascades with a plurality of stages, an identification code is added to the coded signal at an initial stage to mark the start of the data block; furthermore, the subsequent codec stages divide the data blocks to be coded on the basis of this identification code.

Abstract: An apparatus and method is disclosed for converting an input signal having frequency related information sustained over a first duration of time into an output signal sustained over a second duration of time at substantially the same first frequency by adding or subtracting to the effective wave length of the output signal. Preferably, the signals are converted in digital form with samples added or subtracted to frequency convert the signal.

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: A data compressing/decompressing apparatus is suitable for compressing data containing plural-byte characters, for instance, a Japanese language text. The data compressing/decompressing apparatus owns a homonym dictionary in which KANJI-character idioms, KANJI-character-reading, and homonym discrimination information are stored in correspondence with each other. This data compressing/decompressing apparatus converts a KANJI-character idiom contained in declarative sentence data into phonetic data, and further compresses this phonetic data to output the compressed phonetic data. The phonetic data is such data that this KANJI-character idiom is replaced by information constituted by character number discrimination information indicative of the character number about KANJI-character idiom reading, the reading thereof, and the homonym discrimination information thereof.

Abstract: An audio signal processing apparatus performs digital-to-analog conversion to convert at least a first audio data signal having a first sampling frequency and a second audio data signal having a second sampling frequency which is different from the first sampling frequency into corresponding analog audio signals, respectively. The audio signal processing apparatus is provided with: a frequency converting device for converting the second sampling frequency of the second audio data signal into the same sampling frequency as the first sampling frequency; a digital-to-analog converting device for performing digital-to-analog conversion to convert the first audio data signal having the first sampling frequency and the second audio data signal having the converted second sampling frequency which is the same as the first sampling frequency into the analog audio signals, respectively.

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: 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.

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.