Abstract: Method and apparatus for time-scaling and/or pitch shifting by discarding and/or repeating segments of a signal. The signal is stored as a series of samples in a memory where it is readable by one or more read pointers. Periodicity of segments of the signal is determined by evaluating normalized cross-correlation over a range of possible periods. Transients are detected by monitoring changes in rms signal value. To achieve time compression or time stretching, a segment is skipped/discarded whenever a maximum time-discrepancy between the current output and an ideal output is reached or a high periodicity is detected, a jump of the optimal length would not make this time discrepancy too high, and no transient is present in the segment to be skipped/discarded.

Abstract: A data compression apparatus for data compressing a digital information signal obtained from a digital audio signal. The digital information signal includes p-bit samples, where p is an integer larger than 1. The apparatus has an input (16) for receiving the digital information signal, and a lossless compression unit (18) for carrying out a substantially lossless compression step on the digital information signal so as to obtain a data compressed digital information signal. The lossless compression unit includes a Rice encoder, which is distinguishable by a code parameter m. Further, an output terminal (22) is available for supplying the data compressed digital information signal. The Rice encoder has a generator unit (30) for generating the code parameter m from N samples of the digital information signal, in accordance with a formula which optimizes the value of m for each frame of N samples.

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 filter for a system for processing audio samples, which dynamically vaires its length responsive to a moving average of variations in an audio input rate. The filter lengthens at substantially constant input rate variations to reduce input noise, and shortens at rapid input rate variations to enhance responsiveness.

Abstract: A data processing apparatus for encoding data in which first information data from a first source of information data is supplied together with a reference timing value and subsequently a plurality of successive sources of information data of a predetermined processing unit are input when said first information data is finished being supplied. The data processing apparatus produces an encoding start point for the successive sources of data, as a function of a phase difference value between a predetermined reference timing value obtained before the successive sources of information data are input and a start point of the successive processing unit.

Abstract: A local server has a connection to a client and to a remote server over the Internet. The local server receives a request for an audio file from the client and, in response, transmits a requests for the audio file to the remote server. Upon receiving the audio file, the local 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 local 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. The extent and type of transcoding are based on the file formats which the client is capable of handling, the size of the requested audio file, the memory capacity of the client, the bandwidth of the connection between the local server and the client, and the desired level of audio quality.

Abstract: A method of transmitting digitized block coded audio signals includes forming scale factors of the digitized audio signals. The n(k-1) differences are formed from k successively in-time scale factors for each frequency sub-band or for a group of spectral values of the audio signal. The n(k-1) differences are grouped into at least two value classes. New scale factors are selected for each of the n sub-bands or spectral value groups based on a sequence of n(k-1) value classes. Identifying information, including the control information indicating at which locations in the sequence of n(k-1) value classes the selected new scale factors are disposed, is associated with each sequence of n(k-1) value classes. The associated selected new scale factors are assigned to each sequence of the sampled signal values and to the identifying information associated with each sequence of sampled signal values.

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

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

Abstract: A signal transfer acceleration system accelerates transfer of an audio signal through a communications channel, such as a telephone line, RF link, or other suitable audio signal communications channel. The connection along the communications channel takes much less time than would be required if the audio signal were played at its normal speed. The signal transfer acceleration system includes a specially designed transmitter system and a specially designed receiver system that are interfaced via the communications channel. The transmitter system includes: (1) a transmitter storage device configured to store the audio signal at a first sample rate; (2) a filter configured to reduce a frequency range of the signal by filtering frequencies from the signal; (3) a transmitter configured to transmit the signal along the communications channel at a second sample rate that is faster than the first sample rate so that the frequency range is expanded while a time period of the transfer is reduced.

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

Abstract: A shape data compression method for image generation by three-dimensional computer graphics, including a first stage inputting a code number and a quantization precision for encoding a three-dimensional coordinate point sequence and a two-dimensional coordinate point sequence, and reading the coordinate point sequence data. A second stage includes mapping transformed point sequences in a one-dimensional space. A third stage calculates an initial apace division width based on quantization precision, divides the normal space into partial spaces and analyzes distribution of the transformed point sequences therein, decides a division width and calculates distribution of transformed point sequences in each partial space at the division width. A fourth stage obtains a mean value of coordinate values of transformed point sequences distributed in the partial space, and encodes them to produce a code book. A fifth stage generates code sequences for encoding the transformed point sequences according to the code book.

Abstract: A method (400, 900, 1000, 1100), device (1205, 1306) and system (1207, 1308) provide memory-efficient encoding of the random-access lexicon for a text-to-speech synthesis system. The method generates word pronunciations based on efficient retrieval of stored pronunciation information and disambiguation information for an input word, by dividing the input word into a stem and a maximal suffix having a maximal suffix code and then generating at least one word pronunciation based on the stem and the maximal suffix.

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 method and apparatus for time-scale modification of speech using a modified version of the Waveform Similarity based Overlap-Add technique (WSOLA) comprises the steps of storing a portion of an input speech signal in a memory, analyzing the portion of the input speech signal to determined at least one filtered pitch value, calculating an estimated pitch value (12) from the at least one filtered pitch value, determining a segment size (14) in response to the estimated pitch value (12), the segment size (14) having a value greater than the estimated pitch value (12), and time-scale compressing (18) the input speech signal in response to the segment size determined.

Abstract: The present invention provides a system and a method for tracking parameters of a synthesized ?an! audio signal that reduces the amount of processing time without causing any discernible degradation in the sound quality of the audio signal. An audio signal is intelligently divided into multiple time slices and the parameters of the audio signal are tracked over the duration of the time slice. The time slices are selected so that the actual characteristic of the parameters over the duration of the time slice can be easily approximated by performing simple, non-processor intensive steps. The characteristics of various components of an audio signal such as a volume envelope, pitch envelope, low frequency oscillator, MIDI commands controlling the audio signal, and various other inputs are used to identify control points. Adjacent control points are then selected as the start point and end point of a time slice.

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: In an In-Flight Entertainment System (IFES), an audio distribution system transmits and synchronizes an audio data stream from multiple audio channels using the Adaptive Differential Pulse Code Modulation (ADPCM) technique for efficient transmission and preventing loss of synchronization. An encoder digitizes the analog audio signals, compresses the digital data, generates the synchronization parameters, including synchronization data for a selected channel, and creates a data frame to be transmitted to a number of decoders. Each decoder detects the synchronization header, extracts the compressed data patterns from the passenger selections, updates the ADPCM synchronization parameters, decompresses the compressed data patterns, and converts the digital audio data to analog audio signals to be delivered to the passenger seats.

Abstract: The invention enables the apparent display rate of an audiovisual display to be varied. The invention can modify an original set of audio data in accordance with a target display rate, then modify a related original set of video data to conform to the modifications made to the audio data set, such that the modified audio and video data sets are synchronized. When the modified audio and video data sets so produced are used to generate an audiovisual display, the audiovisual display has an apparent display rate that approximates the target display rate. The target display rate can be faster or slower than a normal display rate at which an audiovisual display system generates an audiovisual display from the original sets of audio and video data. The target display rate can be established solely by a user instruction, by analysis of the audiovisual data, or by modification of a user-specified nominal target display rate based upon analysis of the audiovisual data.

Abstract: An apparatus for compressing audio data is provided. An audio signal is sampled and divided into divided audio signals in a plurality of frequency bands. A predetermined process is applied to ones respective of the divided audio signals and a characteristic value for each of the divided audio signals is calculated after the predetermined process. An adaptive bit allocation circuit repeatedly allocates a number of bits to each of the divided audio signals based on the characteristic value and a bit rate of the input audio signal. The adaptive bit allocation circuit detects the frequency band containing one of the divided audio signals having a maximum characteristic value within a selected frequency range. A unit number of bits is repeatedly allocated to the one of the divided audio signals and the characteristics value is modified based on the unit number of bits. A counting member counts the number of allocated bits for the one of the divided audio signals.

Abstract: Apparatus for simultaneously decompressing and interpolating compressed audio data. The compressed audio data is stored in differential log format, meaning that the difference between each two consecutive data points is taken and the log of the difference calculated to form each compressed data point. To efficiently decompress and interpolate the compressed data, advantage is taken of the fact that addition of logs is equivalent to multiplication of linear values. Thus the log of an interpolation factor is added to each compressed data point prior to taking the inverse log of the sum. An integrator block completes the interpolation and decompression of the data.

Abstract: A conduct-along system that can give expressions to sounds and/or images, in which expressions are added to sounds and/or images following the playback of sounds and/or images in real-time based on any one or any combination of parameters, such as tempo, intensity, beat timing and accent, detected from the movement of an input device. The conduct-along system detects any one or any combination of parameters, such as tempo, intensity, beat timing and accent, from the movements of the input device, and plays back voices and/or images in real-time following any one or any combination of detected parameters, such as tempo, intensity, beat timing and accent.

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 MPEG audio/video decoder has memories, a signal processing unit (SPU) including a multiplier and a butterfly unit, a main CPU, and a memory controller which are time division multiplexed between decoding video and audio data. For audio decoding, the butterfly unit determines combinations of components of a frequency-domain vector to reduce the number of multiplies required to transform to the time domain (matrixing). Matrixing is interwoven with MPEG filtering to increase throughput of the decoder by increasing parallel use of the multiplier, the butterfly unit, and a memory controller.

Abstract: A data processing device uses a portion of random access memory 121 as an output buffer 124 for holding a portion of a stream of PCM data which is to be output to a digital to analog converter 530. D/A 530 forms a left analog channel and a right analog channel for speaker subsystems 814 and 815. The PCM data stream is stored in the output buffer so that PCM data samples which pertain to the left channel are stored at even address and PCM data samples which pertain to the right channel are stored at odd address. Control circuitry 145 monitors direct memory access (DMA) transfers which transfer PCM data samples to PCM serializer 142. By comparing the address of each DMA transfer to a left/right channel signal from the D/A, the control circuitry can verify that channel synchronization is correct. If a synchronization error is detected, an channel synchronization error correction procedure is invoked.

Abstract: A memory has storage segments at different addresses respectively. A write address signal represents an address which is periodically updated at a first frequency. Samples of an audio signal are sequentially written into storage segments of the memory at addresses represented by the write address signal respectively. A read address signal represents an address which is periodically updated at a second frequency lower than the first frequency. Samples of the audio signal are sequentially read out from storage segments of the memory at addresses represented by the read address signal respectively.

Abstract: A system for selection by a user and delivery to the user of selected audio data files at a delivery rate of 2-100 times the delivery rate for normal, audibly perceptible playback of an audio data file. The user registers the user's selection of audio material with a central library of data files and a digitized and compressed omnibus file containing the user's selections is prepared and transmitted to the user at a high data transfer rate. The user receives downloads the omnibus file to a removable, high density diskette or PCMCIA card that may hold ten ?Mbytes! Megabytes to one ?Gbyte! Gigabyte of digitized text or audio data, using a removable hard drive or its equivalent and a universal data interface that recognized and compensates for omnibus files received in any of a plurality of input signal formats. The user carries this diskette or PCMCIA card until the user has an opportunity to decompress and play back the text or audio data files in audibly perceptible form.

Abstract: A method an apparatus for modifying the play time of digital audio tracks. For each data block of samples taken from a plurality of such blocks comprising an audio stream or track, the original block data plus a time shifted copy of the block data are superimposed, in a manner dependent on either desired expansion or desired contraction of play time, to create an overlap region. The overlap region is selected at a position of best match as resulting from a normalized correlation over the overlap, the super position being weighted by a linear cross-fading function. All signal crossovers (signal transitions of opposite signal sign) in a block are located. A position at which to superimpose a copy of the data block with the data block, and the length of the data block, are determined based upon the location of said crossovers within a given data block.

Abstract: A radio paging system with voice transfer function for transmitting a voice message input from an ordinary push-button telephone set to a small-sized receive-only unsophisticated radio pager. A paging station is provided to transmit by radio the message from a telephone network to the radio pager as follows: voice information constituting the message is first converted from analog to digital format, compressed, stored in memory, and scrambled by a privacy function part for transmission. The radio pager in turn demodulates the received information, stores it in memory, retrieves a necessary message therefrom as designated, descrambles the designated message from scrambled state, expands the message from compressed state, and outputs the message as an audible output. In this manner, the user carrying the radio pager is able to get the message from the caller without the risk of being tapped by a third party.

Abstract: A method of reconstructing a stream of digital frequency domain audio signal samples into audio signals comprising parsing the stream of samples and reconstructing subband data in the frequency domain, processing the subband data to obtain a processed frequency domain digital audio signal, and constructing a time domain audio output signal from the processed frequency domain digital audio signal.

Abstract: An audio processing system is used to process digital audio signals that represent sound emanating from a source that is moving through three dimensional space. The audio processing system has a filter unit that employs infinite impulse response (IIR) filters to filter the audio signals. The IIR filters have filter coefficients that change when the sound source is stationary or moves from one location to the next in the 3D space. To minimize the transient response following a coefficient change, the filter unit initializes elements in the tap delay lines of the IIR filters to non-zero values. In one implementation, the tap delay line elements are changed to a set of predetermined non-zero values. In another implementation, the tap delay line elements are initialized to the final values produced by the filter for the previous sound source location.

Abstract: A dynamic range compression technique incorporates four novel concepts. The first is the use of a critical band multichannel structure for improved perceptual transparency. The second is the use of attack and release rates, instead of attack and release times, to affect gain control and adaptation of the compressor to changes in the input level. The third concept involves a level estimate control mode which permits increased adaptability using variable weightings of the contribution of both RMS and peak level estimates to gain control. Finally, the fourth concept involves the normalization of the level estimates to reduce or eliminate spectral distortion. These concepts provide a dynamic range compressor with improved perceptual transparency, especially with respect to music.

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: 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 audio encoder/decoder adopting a high-speed analysis filtering algorithm is provided. The audio encoder has a mapping unit for classifying a received audio signal according to a frequency band by using the high-speed band analysis filtering algorithm, a psychoacoustic model for assigning bits to each frequency band using psychoacoustic characteristics, a quantizing and encoding unit for quantizing and encoding the mapped signal according to the number of bits assigned to each frequency band, and a frame packing unit for generating a bit stream from a signal output from said quantizing and encoding unit. The audio decoder has a frame unpacking unit for unpacking a signal from a coded and received bit stream, a decoding and inverse-quantizing unit for decoding and inverse-quantizing said quantized signal, and an inverse-mapping unit for time/frequency-inverse-mapping the inverse-quantized signal by using the high-speed band synthesis filtering algorithm.

Abstract: A audio reproducing apparatus includes an audio decoder and a voice speed converting unit. The audio decoder decodes an audio data stream to produce an audio signal. The voice speed converting unit converts the audio signal in such a manner that when a bit rate is higher than a normal bit rate, a pitch of a reproduced sound interval is the same as the pitch of the sound interval in a normal playback mode and a voice speed in the reproduced sound interval approaches a voice speed in a sound interval in the normal playback mode. The voice speed converting unit further performs voice speed conversion on the audio signal in such a manner that when the bit rate is lower than the normal bit rate, the sound interval is not noticeably interrupted.

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 device for recording and/or reproducing or transmitting and/or receiving compressed data includes decoding circuits 31 to 33 for performing expansion during the compression process is disclosed. An error produced during the compression process is calculated by an input/output error calculating circuit 41 from the input data and data compressed by adaptive bit allocation encoding circuits 22 to 24 and expanded by the decoding circuits 31 to 33, and bit allocation is again calculated on the basis of the error produced during the compression process, with the input data remaining as it is. The bit allocation re-calculated in this manner is quantized by the encoding circuits 22 to 24. Besides, data annulling the error produced during the compression process is formulated by the input/output error calculating circuit 41 and summed to the input signals for subsequent quantization.

Abstract: Voice compression is performed in multiple stages to increase the overall compression between the incoming analog voice signal and the resulting digitized voice signal over that which would be obtained if only a single stage of compression were to be used. A first type of compression is performed on a voice signal to produce an intermediate signal that is compressed with respect to the voice signal, and a second, different type of compression is performed on the intermediate signal to produce an output signal that is compressed still further. As a result, compression better than 1920 bits per second (and approaching 960 bits per second) are obtained without sacrificing the intelligibility of the subsequently reconstructed analog voice signal. Voice compression is also performed by recognizing redundant portions of said voice signal, such as silence, and replacing such redundant portions with a special code in said compressed signal.

Abstract: A low bit rate encoder for compression-encoding digital audio signals of a plurality of channels makes use of both the property of the audio signal and the hearing sense of the human being. The encoder includes: an energy detecting section for detecting energies of the digital audio signals every digital audio signals of the respective channel, a bit allocation amount determining section for determining bit allocation amounts to the respective channels on the basis of the detected result, a compression-encoding section for compression-encoding the digital audio signals on the basis of the bit allocation amounts allocated for every respective channel in accordance with the determined bit allocation amounts, and a multiplexing section for multiplexing the compression-encoded signals every respective channels.

Abstract: A modulation scheme (600) useful in a voice paging system in which both of two orthogonal modulation components (500 and 510) are used to carry two halves of a single voice message destined for a receiver, or two separate voice messages for a receiver. The single voice message is transmitted in half the time.