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: In a coded speech decoding system, an n-channel time domain speech signal is converted to a frequency domain speech signal. A predetermined weighting adding process is executed on the frequency domain speech signal for each of a plurality of different transfer functions. The frequency domain speech signal obtained through the weighting adding process is converted to an m-channel (m<n) time domain speech signal. A predetermined windowing processing is executed on the time domain speech signal.

Abstract: In an embodiment of the invention, a method is presented to minimize the effect of pitch jitter in voicing determination of sinusoidal speech coders during voiced speech. In the method, the pitch of the input signal is normalized to a fixed value prior voicing determination in the analysis frame. After that, conventional voicing determination approaches can be used for the normalized signal. Based on experiments done, the method has been shown to improve the performance of sinusoidal speech coders during jittery voiced speech by increasing the accuracy of voicing classification decisions of speech signals.

Abstract: The invention provides an information recording medium, such as an optical disk, having a large capacity and being capable of performing read/write operations at high speeds. The recording medium includes an audio stream prepared for after-recording data, and a audio attribute information having a bit rate information to the recorded audio stream as a management information. A recorder according to the invention has a check unit for checking, in advance, the possibility of after-recording operation of the recorder to the audio stream to be after-recorded with reference to the bit rate information of the audio attribute information.

Abstract: When a copy of music data recorded in an HDD to another electronic equipment is instructed, an accounting process is performed or a sound quality deteriorating process is performed based on a selection. When the accounting process is performed, after a predetermined accounting procedure is performed, a data copying process is performed and the data is outputted to a copy destination. When the accounting process is performed, a quality of the data is held to be almost identical to that of the original data. When the sound quality deteriorating process is selected, a data conversion is performed by a predetermined sound quality deteriorating process, the quality of the data is deteriorated, and the deteriorated data is outputted to the copy destination. In this case, the accounting is not performed. Where the data is moved, the accounting process and the sound quality deteriorating process are not performed.

Abstract: Frequency information is selectively removed from a video signal in order to decrease the number of color values required for video compression. Removal of the frequency information includes both periodic raking out of narrow frequency bands, and rounding of frequency values. The frequency information removal is carried out selectively in those portions of the visible light spectrum in which the human eye's color response is strongest, thus allowing increases in video compression ratios without visible degradation of image quality.

Abstract: A band splitting section a01 splits an encoder input signal s00, entered into a subband encoding apparatus, into k band components. The band splitting section a01 outputs a subband signal s01 for each of n (<k) split bands, where n is determined by a processible upper-limit frequency on each application. A scale factor producing section a02 detects a maximum amplitude level of the subband signal s01 for each of n split bands. Then, the scale factor producing section a02 produces scale factor information s02 representing a normalized scale factor. A bit allocation producing section a04 outputs bit allocation information s04 for each of n split bands. A bit allocation value 0 is assigned to each of (n+1) to k split bands. A requantizng section a06 requalizes the subband signal s01 for each split band. A frame constructing section a07 constructs a coded frame.

Abstract: A wide band audio signal encoding apparatus, a wide band audio signal decoding apparatus, a wide band audio signal encoding and decoding apparatus and a wide band audio signal recording medium, each having a low bit rate, a wide band, a low distortion factor and a wide dynamic range are provided. Wide band audio data is divided into signal data of a predetermined natural number N of sub-bands, and the number of bits for quantization for sub-sampling is determined based on noise floor information of the above wide band audio data. The signal data of the N sub-bands are sub-sampled by the respective numbers of bits for quantization, and encoded data obtained by multiplexing the signal data of the sub-sampled N sub-bands are recorded on a wide band audio signal recording medium. Therefore, both he characteristics of an extremely wide dynamic range and a wide band can be concurrently achieved at a relatively low bit rate per channel.

Abstract: Scalable coding of audio into a core layer in response to a desired noise spectrum established according to psychoacoustic principles supports coding augmentation data into augmentation layers in response to various criteria including offset of such desired noise spectrum. Compatible decoding provides a plurality of decoded resolutions from a single signal. Coding is preferably performed on subband signals generated according to spectral transform, quadrature mirror filtering, or other conventional processing of audio input. A scalable data structure for audio transmission includes core and augmentation layers, the former for carrying a first coding of an audio signal that places post decode noise beneath a desired noise spectrum, the later for carrying offset data regarding the desired noise spectrum and data about coding of the audio signal that places post decode noise beneath the desired noise spectrum shifted by the offset data.

Abstract: A speech encoding system for use with a digital cellular communication device and a receiving station, includes a mechanism for determining whether a voice communications packet needs to be treated as a data communications packet; a voice recognition mechanism for receiving instructions by voice command; and a control mechanism for responding to said voice command and controlling a controlled entity. A method for encoding a voice command generated on a digital cellular communication device and transmitted over a wireless communication network to a receiving station for controlling a controllable entity includes recognizing a voice command; determining whether the voice command needs to be treated as a data communications packet; encoding the voice command; connecting the voice command to a voice recognition mechanism; and controlling a controlled entity with the voice command.

Abstract: A digital audio decoder is described. The digital audio decoder includes: (i) an audio core which defines a hardware for sub-band synthesis and windowing during decoding of MPEG and AC-3 digital audio signals; (ii) an input RAM coupled to the audio core and configured to store discrete samples in preparation for the sub-band synthesis and the windowing and configured to store intermediate values that are calculated by the audio core during the sub-band synthesis and written back to the input RAM. A process of decoding MPEG and AC-3 digital audio signals is also described.

Abstract: A communication device is provided having a speech encoder and a speech decoder and being able to retrieve and store voice messages in a memory. The messages are stored in the memory according to a message format. This format is more compressed than the speech encoding format which is provided by the speech encoder. The device includes a frame interpolation block for decompressing a stored message and thereby creating a signal according to the speech encoding format. A frame decimation block is also provided for compressing a speech encoded signal thereby allowing a corresponding voice message to be stored in the memory according to the message format.

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 method of compressing an audio signal comprising a series of values is disclosed, the method comprising: utilising a predetermined maximum number of bits B for representing values; dividing the audio signal into a series of blocks of a predetermined length N; and for each of the blocks: computing the maximum number of bits b required to represent the values within a block; packing an indicator of b bits in an output stream; and for each value within the block: where the value exceeds 2B writing only the b most significant bits to the output stream; and where the value does not exceed 2B writing the value to the output stream. The aforementioned methods are well suited to compressing impulse response functions. There is also disclosed a method of decoding an audio signal encoded by the aforementioned methods.

Abstract: An input separator separates the whole digital audio signal into a first audio signal belonging to the basic region and a second audio signal belonging to the extended region. A lossless encoder losslessly encodes the first audio signal and outputs a first bitstream and a first bit rate possessed by the first bitstream resulting from the lossless encoding. A psycho-acoustical encoder psycho-acoustically encodes the second audio signal and outputs a second bitstream and a second bit rate possessed by a second bitstream. The encoding apparatus encodes the input digital audio signal so that the sum of the first and second bit rates matches a predetermined bit rate. A decoding apparatus corresponding to the encoding apparatus losslessly decodes and psycho-acoustically decodes the bitstreams.

Abstract: A solid-state audio recording unit, capable of checking whether normal audio recording is performed or not on a real-time basis, includes one input buffer for receiving incoming audio data operating at a standard speed, another input buffer for writing audio data into a memory 9 operating at a high speed, one output buffer for receiving audio data from the memory 9 operating at a high speed, and another output buffer for delivering audio data as output operating at a standard speed. As such, it is possible to write/read data into/from the memory at a high speed; and thus operation is ensured enabling, on appearance, parallel processing of input and output, even though, in reality, a single memory is shared by the input and output ports; and thus it becomes possible to deliver audio data stored in a memory as an output on a real-time basis.

Abstract: A method and system for reduction of quantization-induced block-discontinuities arising from lossy compression and decompression of continuous signals, especially audio signals. One embodiment encompasses a general purpose, ultra-low latency, efficient audio codec algorithm. More particularly, the invention includes a method and apparatus for compression and decompression of audio signals using a novel boundary analysis and synthesis framework to substantially reduce quantization-induced frame or block-discontinuity; a novel adaptive cosine packet transform (ACPT) as the transform of choice to effectively capture the input audio characteristics; a signal-residue classifier to separate the strong signal clusters from the noise and weak signal components (collectively called residue); an adaptive sparse vector quantization (ASVQ) algorithm for signal components; a stochastic noise model for the residue; and an associated rate control algorithm.

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 redundance reduction method used in coding multichannel signals is proposed, where signals available in digitized form are predicted. A prediction error is computed, which is subsequently quantized and loaded for transmission over a transmission path. In the method, prediction is performed linearly in a backward-adaptive fashion for at least two channels simultaneously, and the statistical relationships of the signals within a channel and between at least two channels is taken into account. A device for decoding redundance-reduced multichannel signals is proposed. It comprises a linear backward-adaptive predictor for at least two channels.

Abstract: An MPEG-1 audio layer III decoding device, which can perform fast decoding of MP3 by performing fast inverse quantization of Huffman code data, includes a bit stream decomposing portion for decomposing an input bit stream of MP3 into side information including bit allocation information and Huffman table information, a scale factor and Huffman code data; a scale factor decoder for decoding the scale factor decomposed from the bit-stream based on the side information; a Huffman decoder for decoding the Huffman code data decomposed from the bit-stream based on the Huffman table information included in the side information; a zero detecting portion for detecting a band of the Huffman code data all providing values of zero; an inverse quantizer for performing inverse quantizing processing on the Huffman code data based on the output of the zero detecting portion, the side information, the scale factor and the Huffman code data; and a hybrid filter bank portion for inversely mapping and decoding the output of the

Abstract: A method extracts features from a mixture of signals. The method filters the mixture of signals by a filterbank to produce a plurality of band-pass signals. Each band-pass signal is windowed to produce a plurality of multi-dimensional observation matrices. The multi-dimensional observation matrices are reduced in their dimensionality. Features are extracted from the reduced dimensionality matrices using independent component analysis. The features can include temporal and spectral characteristics.

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: Provided is a dynamic bit allocation apparatus and method for audio coding which can be used widely for almost all digital audio compression systems and besides implemented simply with low cost. The bit allocation apparatus and method perform a very efficient bit allocation process, paying attention to a psychoacoustics behavior of the human audio characteristics with a simplified simultaneous masking model. In this process, peak energies of units in frequency divisional bands are computed, and a masking effect that is a minimum audio limit with the use of a simplified simultaneous masking effect model is computed and set as an absolute threshold for each unit. Then, a signal-to-mask ratio of each unit is computed, and then, based on this, an efficient dynamic bit allocation is performed.

Abstract: A digital transmission system is disclosed having a transmitter (11) and a receiver (12) for transmitting and receiving a digital audio signal. The digital audio signal is in the form of samples of a specific wordlength (WL) and occurring at a specific sampling rate.

Abstract: Techniques like Huffman coding can be used to represent digital audio signal components more efficiently using non-uniform length symbols than can be represented by other coding techniques using uniform length symbols Unfortunately, the coding efficiency that can be achieved by Huffman coding depends on the probability density function of the information to be coded and the Huffman coding process itself requires considerable processing and memory resources. A coding process that uses gain-adaptive quantization according to the present invention can realize the advantage of using non-uniform length symbols while overcoming the shortcomings of Huffman coding. In gain-adaptive quantization, the magnitudes of signal components to be encoded are compared to one or more thresholds and placed into classes according to the results of the comparison. The magnitudes of the components placed into one of the classes are modified according to a gain factor that is related to the threshold used to classify the components.

Abstract: A multi-channel audio compression technology is presented that extends the range of sampling frequencies compared to existing technologies and/or lowers the noise floor while remaining compatible with those earlier generation technologies. The high-sampling frequency multi-channel audio is decomposed into core audio up to the existing sampling frequencies and a difference signal up to the sampling frequencies of the next generation technologies. The core audio is encoded using a first generation technology such as DTS, Dolby AC-3 or MPEG I or II such that the encoded core bit stream is fully compatible with a comparable decoder in the market. The difference signal is encoded using technologies that extend the sampling frequency and/or improve the quality of the core audio. The compressed difference signal is attached as an extension to the core bit stream. The extension data will be ignored by the first generation decoders but can be decoded by the second generation decoders.

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 system comprises a refined psycho-acoustic modeler for efficient perceptive encoding compression of digital audio. Perceptive encoding uses experimentally derived knowledge of human hearing to compress audio by deleting data corresponding to sounds which will not be perceived by the human ear. A psycho-acoustic modeler produces masking information that is used in the perceptive encoding system to specify which amplitudes and frequencies may be safely ignored without compromising sound fidelity. The present invention includes a system and method for efficiently implementing a masking function in a psycho-acoustic modeler in digital audio perceptive encoding. In the preferred embodiment, the present invention comprises a non-logarithmically based representation of individual masking functions utilizing minimally-sized look-up tables.

Abstract: Blocks of encoded audio information are arranged in frames separated by gaps or guard bands that are aligned with frames of video information. The gaps are provided to protect the audio information against corruption caused by uncertainties or jitter in editing operations such as switching between two difference sources of video/audio information. The otherwise wasted space or bandwidth required to convey the gaps is utilized by conveying encoded segments of auxiliary information. When the encoded auxiliary information is subsequently decoded, an error recovery process provides substitute information for those segments that are corrupted by an editing operation. In one embodiment, the recovery process is adapted according to the choice of an auxiliary sync word conveyed in the segment.

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: A method for encoding a digital audio signal includes filtering a portion of the digital audio signal into a first number of frequency ranges to produce a respective first number of filtered signals and performing a discrete frequency analysis on each of the first number of filtered signals to produce a frequency representation of the digital audio signal. The method also includes generating a psychoacoustic representation of the portion of the digital audio signal based on the frequency representation of the digital audio signal and formatting the first number of filtered signals based on the psychoacoustic representation of the portion of the digital audio signal to produce a digitally-compressed encoded bit stream representing the portion of the digital audio signal.

Abstract: Coding bit rate converting method and apparatus for coded audio data are disclosed to convert input coded audio data to output coded data of a coding bit rate lower than a target coding bit rate. A control output is taken out when the frame size of the input coded audio data is larger than the frame size determined by the target coding bit rate, and the output coded data is provided as a converted output by controlling parameters defining the frame size of the input coded audio data by the use of the control output in a predetermined procedure until the frame size of the input coded audio data becomes smaller than the frame size determined by the target coding bit rate.

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: A scalable audio coding/decoding method and apparatus are provided. The coding method includes the steps of (a) signal-processing input audio signals and quantizing the same for each predetermined coding band, (b) coding the quantized data corresponding to the base layer within a predetermined layer size, (c) coding the quantized data corresponding to the next enhancement layer of the coded base layer and the remaining quantized data uncoded and belonging to the enhancement layer, within a predetermined layer size, and (d) sequentially performing the layer coding steps for all layers, wherein the steps (b), (c) and (d) each include the steps of (e) representing the quantized data corresponding to a layer to be coded by digits of a predetermined same number, and (f) coding the most significant digit sequences composed of most significant digits of the magnitude data composing the represented digital data.

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: Disclosed is an audio encoding apparatus for splitting an audio signal into a plurality of bands, allocating a number of quantization bits to each band and transmitting the audio signal of each band upon quantizing the audio signal by the number of allocated bits. The apparatus includes a bit allocation unit for (1) calculating an MNR for each band, where MNR is the ratio of an audio masking level M to a quantization noise level N, (2) comparing a set MNR with the smallest MNR from among the MNRs of the respective bands, (3) incrementing the number of quantization bits of the band that corresponds to the smallest MNR if the smallest MNR is smaller than the set MNR, and (4) performing allocation control for allocating quantization bits to each band until the smallest MNR becomes equal to or greater than the set MNR.

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: When errors occur in a received code sequence, an error concealment routine part 130 is actuated and a correct code sequence is generated based on a code sequence received before the errors are detected. A decoding routine part 21' executes a decoding process based on this code sequence and updates the internal state based on the decoded result. Thereafter, when the code errors are recovered, an error recovery routine part 140 is actuated. The error recovery routine part 140 re-estimates a correct code sequence during a time period when the errors are detected, based on a code sequence received before the errors are detected and a code sequence received after no more errors are detected, and generates a second estimated code sequence. The decode process part 21' executes the decoding process based on an internal state information for retaining the second estimated code sequence in an internal state storage part 23', and updates the internal state information based on the decoded result.

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: An audio recorder obtains a PCM audio signal from input analog or digital audio data, encodes the PCM audio signal, and stores the encoded signal in a semiconductor memory. The encoding method is selectable from at least two different methods, providing different levels of audio quality. The semiconductor memory may be internal to the audio recorder, or may be an external memory device such as a flash memory card. An audio player decodes the signal stored in the semiconductor memory, and converts the decoded signal to a selected one of at least two output formats.

Abstract: An adaptive audio signal compression computer system corrects multiple input audio streams, such as digital multichannel audio streams from various audio sources, for equalization deficiencies caused primarily by attenuation factors experienced by a listener's human ear. A compression stage (filter stage) adaptively compresses (filters) an input audio stream by selecting the ear response data that includes a plurality of selectable sets of filter coefficients wherein each set of filter coefficients corresponds to an inverse audible listening response curve for a predetermined audio volume level. When available from an audio source, the system provides audio source type data to an adaptive compression stage and also generates a compression control signal in response to the source type data to control the compression.

Abstract: An apparatus carries out an adaptive pulse code modulation process on voice data and records them in a solid memory which is detachably attached to the apparatus. The code bit number is switched, depending on the selected recording mode. A filter for suppressing a high-voice region of the frequency band of the voice data is used, and there is provided a selector for applying this filter at the time of voice reproduction selectively either always or according to the code bit number.

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: A device for determining the quality of an output signal to be generated by a signal processing circuit with respect to a reference signal. The device is provided with a first series circuit for receiving the output signal and a second series circuit for receiving the reference signal. The device generates an objective quality signal through a combining circuit which is coupled to the two series circuits. Poor correlation between the objective quality and subjective quality signals, the latter which will be assessed by human observers, can be considerably improved by a differential arrangement present in the combining circuit. This arrangement determines a difference between the two series circuit signals and reduces this difference by a certain value, preferably one that is a function of a series circuit signal. Poor correlation can be improved further by disposing a scaling circuit, between the two series circuits, for scaling at least one series circuit signal.

Abstract: A method is provided for compressing relatively time invariant binary data, such as speech data in a telephone answering device, using statistical analysis of changes in the data. An original record organized into multiple frames of multiple bits each is used to construct an XORed record of the same number of frames and bits. The XORed record has a base frame with the same bit value pattern as a corresponding base frame of the original record, and remaining frames with bit values given by the outputs of an exclusive-OR operation applied to the bit values of corresponding and prior frames of the original record. The bit positions of the XORed record frame set are analyzed and reordered, according to their bit value change activity and used to construct an output record. The output record may have a base frame with the same bit value pattern as the corresponding reordered XORed record base frame.

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: A telecommunication network establishes a telephone call from a first location that generates compressed voice packets to a second location. The telecommunication network includes a plurality of switches, a first interworking function ("IWF") coupled to the first location and coupled to a first switch of the plurality of switches, a second IWF coupled to the second location and coupled to a second switch of the plurality of switches, and a network database coupled to said first switch. When a telephone call is initiated from the first location, the first switch receives signaling information from the first location. The first switch then queries the network database based on the signaling information and determines whether the second location uses compressed voice packets based on the query. Finally, the first switch establishes a connection between the first IWF and the second IWF if it is determined that the second location uses compressed voice packets.

Abstract: The present invention provides a system for handling and transmitting a file over a communication channel wherein the file may be adaptably compressed to improve throughput. The invention also provides for a method of recovery in the event of a communication failure. The file may be encrypted while it is being transmitted. The compression and transmission may occur while the file is being written, so that the receiving location receives the data in near real time.

Abstract: A spatial audio coding system, including an encoder and a decoder, operates at very low bit-rates and is useful for audio via the Internet. The listener or listeners preferably are located within a predictable listening area, for example, users of a personal computer or television viewers. An encoder produces a composite audio-information signal representing the soundfield to be reproduced and a directional vector or "steering control signal." The composite audio-information signal has its frequency spectrum broken into a number of subbands, preferably commensurate with the critical bands of the human ear. The steering control signal has a component relating to the dominant direction of the soundfield in each of the subbands. Because the system is based on the premise that only sound from a single direction is heard at any instant, the decoder need not apply a signal to more than two sound transducers at any instant.