Showing papers on "Audio signal processing published in 1991"

Computing and multimedia entertainment system

Abstract: A remotely controllable computing and multimedia entertainment system includes a personal computer (24) having an entertainment circuit (12) made up of a radio frequency circuit (48), a television circuit (46), and an audio multimedia circuit (18). A remote control circuit (50) provides programmable control of the entertainment circuit (12) to select among computer function operation, television and radio operation, and audio operation. An analog mixing circuit (70) within the audio multimedia circuit (18) provides mixing for a plurality of analog audio signals. A telephone circuit (44) integrates data, fax, and voice telephone signals in the entertainment circuit (12). A volume control circuit (318) within the audio multimedia circuit (18) provides varying volume, bass, and tone levels for each audio signal received by the analog mixing circuit (70). The analog audio signals received by analog mixing circuit (70) may include monaural and stereo audio signals.

Method and apparatus for digital audio broadcasting and reception

Abstract: A radio broadcasting system is provided for transmitting and receiving through free space a composite signal consisting of a frequency modulated (fm) analog signal and a multicarrier modulated digital signal which is especially adapted to be resistive to multipath degradation. The fm signal and the digital multicarrier modulated signal are fully coherent. Further according to the invention, the digital signal comprises a plurality of carriers having a maximum amplitude at least 20 dB below the unmodulated fm signal and preferably, 30 dB below the analog signal. The multicarrier modulated signal is phase locked according to the invention to the recovered analog fm pilot tone at 19 kHz in the composite baseband spectrum of the fm signal which is at least 20 dB above the multicarrier modulated signal, which enables rapid and reliable acquisition of signal for coherent detection. In a specific embodiment, the multicarrier modulated signal is a synthesized vector-modulated signal which is a quadrature phase shift keyed (QPSK) modulated set of synthesized carriers each occupying 9.5 kHz of spectrum replicated twenty-one times within a 199.5 kHz bandwidth with no more than two bits per vector. The broadcast system is designed to be used in support of compressed digital audio programming material. In a demodulation process according to the invention, a demodulator is operative to phase lock to the recovered high-amplitude analog pilot tone, to coherently demodulate the digital signal, and to format the recovered data stream for source decoding.

Enhanced wide area audio response network

Abstract: Apparatus and method to provide enhanced wide area audio response services through an enhanced wide area audio response network which includes a central controller and a plurality of audio peripherals distributed over a wide area, each audio peripheral being connected to telephone lines for receiving and originating telephone calls, converting received analog audio signals into digital representations, recording and storing digital representations, converting stored digital representations into analog audio signals, playing audio signals over connected telephone lines, and communicating with, including receiving commands from, the central controller over a Packet Switched Public Data Network (PSPDN), which controller is a highly reliable general purpose controller which offers utility grade service to each audio peripheral and utilizes Dialed Number Identification Service (DNIS) tables for various applications, including voice messaging, audio text, remote information provider accessing, and testing to provide error notification.

Multidimensional audio window management

Abstract: This paper proposes and organization of presentation and control that implements a flexible audio management system we call “audio windows”. The result is a new user interface integrating and enhanced spatial sound presentation system, an audio emphasis system, and a gestural input recognition system. We have implemented these ideas in a modest prototype, also described, designed as an audio server appropriate for a teleconferencing system. Our system combines a gestural front end (currently based on a DataGlove, but whose concepts are appropriate for other devices as well) with an enhanced spatial sound system, a digital signal processing separation of multiple sound sources, augmented with “filtears”, audio feedback cues that convey added information without distraction or loss of intelligibility. Our prototype employs a manual front end (requiring no keyboard or mouse) driving an auditory back end (requiring no CRT or visual display).

Signal Processing, Image Processing and Pattern Recognition

Abstract: A detailed exposition of the main areas of signal processing, this book is divided into three sections: one-dimensional signal processing and digital filters; two-dimensional signal processing and image processing; and pattern recognition. Among the more specific topics covered are: analog filters; discrete systems and signals; non-recursive filters; FFT; IIR design; quantization effects; and hardware and software design. There is also material on system stability, picture enhancement and restoration, and parallel processing methods, as well as a comprehensive treatment of syntactic methods, parsing and neural networks.

Sound imaging method and apparatus

Abstract: The illusion of distinct sound sources distributed throughout the three-dimensional space containing the listener is possible using only conventional stereo playback equipment by processing monaural sound signals prior to playback on two spaced-apart transducers. A plurality of such processed signals corresponding to different sound source positions may be mixed using conventional techniques without disturbing the positions of the individual images. Although two loudspeakers are required the sound produced is not conventional stereo, however, each channel of a left/right stereo signal can be separately processed according to the invention and then combined for playback. The sound processing involves dividing each monaural or single channel signal into two signals and then adjusting the differential phase and amplitude of the two channel signals on a frequency dependent basis in accordance with an empirically derived transfer function that has a specific phase and amplitude adjustment for each predetermined frequency interval over the audio spectrum. Each transfer function is empirically derived to relate to a different sound source location and by providing a number of different transfer functions and selecting them accordingly the sound source can be made to appear to move.

CCD correlators for auditory models

Abstract: Surface-channel charge-coupled devices (CCDs) provide a mechanism for analog signal delay that can be built using an ordinary double-poly CMOS digital process, such as offered by Orbit through MOSIS. This technique has been applied to implement the correlation processing needed in auditory models of the sort proposed by J.C.R. Licklider (1951) for monaural pitch perception and sound separation. The resulting chips take analog audio in and produce analog video out (moving pictures of the correlogram representation of the sound) in real time. These chips present a variety of interesting analog and digital design challenges, which are addressed here. The first working experimental chips show real-time correlograms with 84 cochlea taps and 70 CCD delay stages per tap. There remains at least one unanticipated and as yet not understood problem with these circuits, resulting in a serious skew in the displayed correlation levels. Several circuit improvements are planned for the next generation of experimental chips. >

Feature-map methods for extracting sound frequency modulation

Abstract: The role of frequency modulation (FM) in the auditory system is investigated. Psychoacoustic evidence shows that FM is a cue for noncontextual auditory scene analysis, and neurophysiological evidence shows that it is detected by neurons in some animal auditory systems. Some filters for detecting FM in sound are presented. One method uses correlation of a kernel with a time-frequency representation of cochlear neural impulses of a sound to reveal its FM components. Another correlation method is based on detecting FM in channel-by-channel autocorrelation images, with some data reduction and sharpening applied. The filters are tuned for various spatiotemporal frequencies to extract information about sound with various FM characteristics, producing maps showing time-frequency points at which the features are present. These maps can allow another process to decide what sources are present or to use FM cues for pattern recognition. >

Interrelational audio kinetic entertainment system

Abstract: Movable and audible toys and other animated devices (14) spaced apart from a television screen are provided with program synchronized audio and control data to interreact with the program viewer in relationship to its programming. A kinetic device actuating signal, sampling keyboard based encoder (12) is coupled via an audio mixer to an audiovisual programming source and television transmitter (16) carrying an audio based kinetic and audio signal complex. At a remote location (200), coded audio and kinetic device signals along with audiovisual programming are received and the audiovisual programming content is displayed for viewing and listening. Stereo sound band based program audio signals are decoded and separated from the stereo sound band based device audio and kinetic signals. The device audio and kinetic signals are retransmitted to a spaced-apart toy causing the device to be audible and to move in synchronization with the spaced-apart audiovisual programming. A single band low powered FM transmitter (28) carries the audio kinetic signal complex in proximity yet spaced apart from a receiver coupled to an audio cassette configured magnetic head transducer disposed in the kinetic device to communicate audio and kinetic information to a device speaker and motors to cause the device to move and be audible.

Subband coding of stereophonic digital audio signals

Abstract: The exploitation of left-right correlation in a subband code for stereophonic audio signals is investigated. A transform of left and right signals into decorrelated intensity and error signals is presented. Although this can be seen as the optimal exploitation of redundancy, it yields only marginal gain in bit rate. If the reduced phase-sensitivity of the human observer can be exploited by encoding only the intensity signal, a substantial gain can be obtained. Preliminary results of a stereo codec are promising: at 192 kb/s good coding results have been obtained. >

Sound mixing device

Abstract: A system for interactively controlling multiple parameters affecting an audio output, the system employing a controller with a visual display (e.g., a cathode ray tube) for displaying an icon that is a visual representation of an input sound signal in a multidimensional space and an input device (e.g., a mouse) used to control the location of the icon on the display. The controller generates a multiple parameter control signal that is based upon the location of the icon and is used by a sound signal processing circuit to control multiple parameters affecting an audio output. The icons are images of the sources of sound input signals. The sound signal processing circuit has M times N controllable amplifiers arranged in an M by N matrix in which each of the M inputs is distributed to N controllable amplifiers and the outputs of M controllable amplifiers are combined to provide each of the N outputs. Each controllable amplifier has a unique combination of an input signal and an output signal and receives a unique, continuously variable gain control signal.

Audio and video reproducing apparatus having independent control of selected signals

Abstract: An audio reproducing apparatus including first and second audio signal transmission paths each applied with a plurality of audio signals. Each of the first and second audio signal transmission paths has a selector for selecting one of the plurality of audio signals supplied thereto and a level controller for controlling the level of the selected audio signal. The first and second audio signal transmission paths are connected to first and second speaker units respectively each for converting the audio signal selected by the corresponding selector and whose level is controlled by the corresponding level controller into acoustic power.

System and method for receiving and decoding global positioning satellite signals

Abstract: A system receives and demodulates spread spectrum positional signals such as those generated by a GPS satellite by frequency shifting the signals substantially to baseband and utilizing digital signal processing techniques. The digital signal processing techniques utilized can be implemented by a standard audio digital signal processor due to the circuit design. The spread spectrum signal is frequency shifted substantially to baseband, forming in-phase and quadrature components which are processed substantially in parallel. Pseudo-range, carrier phase and doppler frequency, and the underlying data are thereby derived.

Apparatus and method for providing digital audio in the cable television band

Abstract: A method and apparatus are provided for transmitting, receiving, and reproducing digital audio signals as discrete carriers similar to standard FM broadcast signals. An audio signal is digitized using, for example, adaptive delta modulation techniques. Several channels of audio information, such as left and right stereo channels and a second audio program ("SAP") channel can all be digitized and incorporated onto the digital broadcast signal carrier. The digitized audio signal may be modulated using multiphase modulation of the carrier of an FM broadcast band or cable television band signal. A plurality of audio channels may be digitized and transmitted over the airwaves, or over a cable transmission network. Channels of nondigitized audio channels may be interspersed with the digitized audio channels in the Fm broadcast band. Source material for the digitized audio channels may be provided to a cable headend over the cable transmission network in the 5-30 MHz CATV upstream path, and rebroadcast over the cable transmission network in the forward or "downstream" band.

Method and apparatus for simultaneous output of digital audio and midi synthesized music

Abstract: A method and apparatus are disclosed for simultaneously outputting digital audio and MIDI synthesized music utilizing a single digital signal processor. The Musical Instrument Digital Interface (MIDI) permits music to be recorded and/or synthesized utilizing a data file containing multiple serially listed program status messages and matching note on and note off messages. In contrast, digital audio is generally merely compressed, utilizing a suitable data compression technique, and recorded. The audio content of such a digital recording may then be restored by decompressing the recorded data and converting that data utilizing a digital-to-analog convertor. The method and apparatus of the present invention selectively and alternatively couples portions of a compressed digital audio file and a MIDI file to a single digital signal processor which alternately decompresses the digital audio file and implements a MIDI synthesizer. Decompressed audio and MIDI synthesized music are then alternately coupled to two separate buffers. The contents of these buffers are then additively mixed and coupled through a digital-to-analog convertor to an audio output device to create an output having concurrent digital audio and MIDI synthesized music.

System for communicating television signals or a plurality of digital audio signals in a standard television line allocation

Abstract: A system is provided for selectively transmitting horizontal television lines including video and digital audio components or equivalent digital data lines containing a plurality of audio program signals. A conventional television signal transmission places a digital audio signal in the horizontal blanking interval, followed by analog video information. In accordance with the present invention, the "window" containing the analog video information is replaced with a plurality of digital audio signals that are time division multiplexed within the window. An additional audio channel is placed in the horizontal blanking interval, at the same location the audio is placed when video information is transmitted. Selector switches are provided in the encoder and decoder for processing a composite waveform as either a video signal with an associated audio channel, or as a multiple channel digital audio signal. Independent encryption and decryption of each of the multiple audio channels is provided.

Audio synchronization apparatus

Abstract: An audio synchronizer apparatus having a delay detector for accurately measuring the delay of a video apparatus, coupled with a controllable, variable delay audio delay circuit for accurately delaying an audio frequency signal by substantially the same amount as the video signal delay. The apparatus disclosed can operate successfully even with video delays exceeding one frame, without requiring an auxiliary correlation circuit. The variable audio delay may also contain precise filtering to compensate for adverse effects introduced by the delay.

Audio output circuit in electronic apparatus with composite display function

Abstract: An audio output circuit in an electronic apparatus equipped with a composite display function where a plurality of pictures can be displayed on composite screen regions of a single display screen. The circuit includes an audio signal switching circuit for selectively switching audio signals relative to the pictures displayed on the individual screen regions and outputting the selected audio signal to a speaker. The circuit also includes a simultaneous audio output switching circuit for selecting a plurality of audio signals out of those relative to the pictures displayed on the individual screen regions and then outputting the selected audio signals simultaneously through a plurality of speakers. Thus any audio signal(s) relative to the picture(s) displayed on the individual screen region(s) can be heard either selectively or simultaneously through the speaker(s).

Audio intrusion detection system

Abstract: A supervisory circuit for use with an audio intrusion detection system is disclosed. The supervisory circuit periodically generates an audio test signal which is supplied to a sounder which emits an audio test sound. The audio test sound is directed into a volume of space, in the same volume of space as which the audio intrusion detection system is directed to detect. The audio intrusion detection system detects the test sound and generates an audio test signal in response thereto. During the generation of the audio test sound, the comparing apparatus of the audio intrusion detection system is disabled. The audio test signal generated by the audio intrusion detection system is then compared to a test threshold signal. A test result signal is generated in response to the comparison with the test result signal indicative of the operability of the audio intrusion detection system.

Audio signal reproducing apparatus

Abstract: In the audio signal reproducing apparatus, information on the transfer characteristics representative of the transfer characteristics from virtual sound sources to both ears of a listener in at least a first quadrant of the rotational angular position of the head M of the listener is stored in storing unit 22. The transfer characteristics in rotational angular position of the head M represented by detection outputs from detecting unit 5L, 5R and 13 for detecting the rotational angle position depending upon the movement of the head of the listener are formed based upon the transfer characteristics information of at least the first quadrant stored in the storing unit and left and right channel audio signals are processed by audio signal processing unit 23 for achieving a proper binaural reproduction relative to the virtual sound sources. Also in the audio signal processing apparatus of the present invention, information on the transfer characteristics from virtual sound sources to both ears of a listener for each given rotational angle depending upon the movement of the head M of the listener is stored in storing unit 62. The rotational angular position of the head of the listener is detected by detecting unit 45L, 45R and 53 at a solution higher than that of the information of the transfer characteristics stored in said storing unit. The information on at least two transfer characteristics in the vicinity of the rotational angular position represented by the detection outputs from detecting unit is read from the storing unit. The information on the transfer characteristics in the rotational angular position of the head represented by the detection output is interpolation operated by interpolation operating unit 61. Based upon the information on the transfer characteristics determined by the interpolation operation unit, left and right channel audio signals are processed by audio signal processing unit 63 for achieving a proper binaural reproduction relative to the virtual sound sources.

System for generating correction signals for use in forming low distortion analog signals

Abstract: A method and apparatus for generating correction signals for use in forming low distortion analog signals. A digital representation of a desired analog waveform is encoded into a digital data signal which is outputted to memory. A digital correction signal is encoded, having an opposite phase and increased amplitude from the signal distortion which it is determined will occur when the digital data signal is repetitively read out of memory and decoded. This digital correction signal is outputted to memory. The digital data and correction signals are repetitively and synchronously read out of memory into a decoder. The decoder converts both digital signals into analog signals, so that the analog correction signal may superpose on the distortion in the analog data signal, resulting in a low distortion analog signal.

Sound processing device

Abstract: A sound processing device for processing sound is arranged to control an attenuation characteristic for attenuating the low-frequency signal of an input audio signal according to the level thereof, whereby it is possible to suppress a noise component contained in sound without substantial impairment in auditory sound quality.

Audio circuit for a television receiver with central speaker producing only human voice sound

Abstract: An audio circuit for a television receiver wherein a left-hand speaker is disposed on a left side of a screen of a television receiver, a right-hand speaker is disposed on a right side of the screen, and a central speaker is disposed above and below the screen, or wherein an audio signal accompanies a picture to be displayed on the screen. The audio circuit supplies the audio signal to the left-hand, right-hand and central speakers to produce sound including a frequency component extraction circuit for extracting predetermined frequency components of human voice from the audio signal and for supplying an extracted signal of the human voice frequency components to the central speaker so that the central speaker produces only human voice sound.

A MUSICAM source codec for digital audio broadcasting and storage

Abstract: Theoretical and practical aspects of the MUSICAM (masking pattern adapted universal subband integrated coding and multiplexing) system are presented. The system is briefly described. It is one of the few codecs able to achieve high audio quality at bit-rates in the range of 64 to 192 kb/s per monophonic channel. It is shown how it has been designed so as to meet the technical requirements of most applications (low delay, low complexity, error robustness, short access units, etc.). Two examples of applications in the field of digital audio broadcasting and multimedia are given. >

Digital communication device

Abstract: A digital communication device of the present invention is provided with a modulation circuit for modulating a digital transmit signal, a first interpolater for converting the modulated signal in frequency, a coding circuit for coding the signal converted in frequency into an audio PCM transmission code with reference to a voice companding code table, a decoding circuit for decoding a coded audio PCM receive code into a digital signal with reference to the voice companding code table, a second interpolater for converting the decoded digital signal in frequency, and a demodulation circuit for demodulating the converted signal, and digitally performs modem modulation/demodulation and voice codec processing.

Method and apparatus for creating de-correlated audio output signals and audio recordings made thereby

Abstract: An apparatus and method for generating audio output signals having a specified cross-correlation relationships is disclosed. The apparatus operates by phase-shifting (14) different frequency bands (12) of an input signal (x(t)) by differing amounts which depend on the desired cross-correlation. The amplitude spectrum of the input signal is not altered.

Real-time signal analysis apparatus and method for digital signal processor emulation

Abstract: An emulation system used to debug software for a digital signal process (DSP) (6) includes a built-in digital signal analyzer (18) which operates upon the same digital signals as those presented directly to and outputted by the DSP (6), bypassing the signal converters (4, 8) used to convert an input analog signal to digital format and the output digital signal to analog format. A host computer (12) communicates with the digital signal analyzer (18) via firmware in a control processor (30) and personality board (36), or is alternately connected directly with the analyzer (18). Communications between the digital signal analyzer and the DSP (6) are through the same contact probe (14) as that used for the emulation software. The analyzer may be used to trigger a software function within the emulator (20) based upon the real-time signal from the DSP (6), and is also capable of interpolating between successive digital values of an analyzed signal for display purposes.

Apparatus for discriminating an audio signal as an ordinary vocal sound or musical sound

Abstract: An apparatus for discriminating a received audio signal as vocal sound or musical sound includes a pre-processing circuit 100 for separating the audio signal into a vocal frequency band signal and a musical frequency band signal, an intermediate decision circuit having a plurality of decision units for producing a plurality of vocal and musical decision signals, each decision unit distinguishing whether vocal or musical frequency band signal includes properties of voice or music, and a final decision circuit 600 for systematically analyzing the vocal and musical decision signals to produce a final decision signal for discriminating the audio signal as the vocal or musical sound.

Method and structure for synchronizing multiple, independently generated digital audio signals

Abstract: A method for synchronizing multiple digital signals is disclosed herein which synchronizes samples of audio information generated by independent digital audio sources to a single master sampling clock reference. An interpolation filter associated with each independent digital audio source determines an interpolated audio sample based on audio samples from the associated digital audio source and based on a ratio between the master sampling rate and the sampling rate of the associated digital audio source. This ratio is derived from averaging a code indicating the number of samples received from an associated one of the independent digital audio sources during each master sampling clock period. The interpolation filter for each of the independent digital audio sources uses this ratio in combination with the audio samples to output an interpolated sample at each master sampling clock reference signal. Using this method, any number of independent digital audio sources may have their outputs interpolated with the above-described interpolation filter, and the interpolation filters will output interpolated samples which are synchronous with one another.

Motion picture digital sound system and method

Abstract: A digital time code is printed in an area of a motion picture film, between the normal analog optical sound track and the picture frames, that is exposed along with the sound track when a print is made. This area is partially redeveloped, and normally reserved to isolate the analog sound track from the picture frames. Digital audio for the motion picture is stored in a large capacity high integrity archival digital storage system. The time codes corresponding to known locations on the film are read as the film is played, and in an anticipatory pass the digital audio signals for these frames are transferred to a fast access data storage buffer which temporarily stores the data before it is converted to analog format for theater play. The time code is read with light that is absorbed by the film dyes produced when the film is developed. Temporary storage of the digital audio signal in the buffer memory accommodates breaks in the film, projector changeover and various time code validation schemes, and allows the digital data source in which the digital audio data is stored to be a relatively slow access high data reliability device such as a digital tape.