Showing papers on "Audio signal processing published in 1982"

Apparatus and method for receiving audio signals transmitted as part of a television video signal

Abstract: The apparatus and method for receiving one or more audio frequency signals transmitted on a television video signal includes a decoder comprised of a pulse separator to recover modulated pulses from said video signal and a demodulator to demodulate said modulated pulses from said separator to recover said audio signal. Also included is an encoding apparatus and method for encoding modulated pulses responsive to audio signals on a video signal.

Programmable signal processing device

Abstract: Programmable signal processing device mainly intended for heanng aids and of the kind which includes an electronically controlled signal processor, the device being able to select a number of different signal processes to suit different sound situations automatically or by the user himself. This is accomplished by a memory (6) arranged to store information data for at least two unique signal processes adjusted to different sound environment/listening situations and a control unit (5), manual or automatic, arranged to transmit information data for one of the unique signal processes from the memory (6) to the signal processor (4) to bring about one signal process adjusted to a particular sound environment/listening situation.

Audio/video quality monitoring system

Abstract: Method and means for evaluating the quality of audio and/or video transfer characteristics of a device upon which, or through which, audio and/or video information is contained, or passes, respectively. Both method and apparatus concern the evaluation of the quality of information transfer in the recording and playing back of a recording medium or in the transferring of audio and/or video information through an information handling device referred to as a throughput device. Unit evaluation is accomplished by establishing an input signal of known content, measuring selected parameters of selected parts of the input signal, feeding the input signal to the unit under test, measuring the parameters of parts of the output signal from the unit under test corresponding to the same selected parts of the input signal, and comparing the selected parameters of the input signal with the corresponding parameters of the output signal. Whether monitoring the quality of the signal transfer characteristics of a throughput device, a magnetic tape containing program material, or a video disc, master disc or replica, a "signature" is created for the unit under test, and subsequent analysis of the unit as it progresses along a production line or of a copy made on the same or alternate recording medium results in a second "signature" which is compared against the first signature to make a determination as to the quality of the signal handling or transfer characteristics of the unit. In this manner, out-of-tolerance conditions can be automatically detected, thereby eliminating subjectivity and providing consistency in the quality level of device testing.

Community antenna television arrangement for the reception and distribution of TV - and digital audio signals

Abstract: Community antenna television arrangement for the reception and distribution of TV signals and digital audio signals, in particular signals which are transmitted per satellite, including a head-end connected to a receiving antenna and a signal distribution network, a time-division multiplex signal which comprises the digital audio signals in a time-multiplex distribution, being applied to the head-end, which time-division multiplex signal is modulated on a sound carrier, the bit rate of the digital audio signals to be distributed being reduced in the head-end of the community antenna television arrangement by a TDM/FDM conversion in order to reduce signal echoes.

A microcomputer with digital signal processing capability

Abstract: A single microcomputer for realtime digital signal processing and high-speed controller applications, with a 200ns instruction cycle, 16 × 16 parallel multiplier, 32b arithmetic unit, 144 by 16 data memory, a 1536 by 16 program and coefficient memory, will be discussed.

Method and apparatus for recording and reproducing an information signal comprised of a video signal and an audio signal

Abstract: OF THE DISCLOSURE A method and apparatus for recording and reproducing an information signal comprised of a video signal and an audio signal in a plurality of successive parallel tracks on a magnetic tape, each track including a first audio section followed by a video section, by converting the audio signal into a PCM digital audio signal; compressing the PCM audio signal; frequency modulating the compressed PCM audio signal to produce an output PCM audio signal; and recording the output PCM audio signal and the video signal in the plurality of successive tracks such that one field interval of the video signal is recorded in the video section of each track and the output PCM audio signal corresponding to one field interval of the video signal is recorded in the audio section of each track and corresponds to the video signal in the leading portion of the previous successive track and the video signal in the trailing portion of the previous alternate track, wherein the audio signal recorded in each track is delayed by a maximum of .5 field intervals from the video signal to which it corresponds.

Automatic background noise suppressor

Abstract: A circuit for suppressing background noise of a continuous nature while enhancing speech signals, or signals having the transient temporal qualities of speech, includes a signal multiplier which, in the preferred embodiment, receives the composite audio signal along with a control signal present only when the speech component of the audio signal is present. The control signal may be derived from an AGC circuit having a slow attack, fast decay characteristic to establish a quiescent output level from the AGC amplifier in the absence of speech. An envelope detector is biased to provide a zero output amplitude in response to the quiescent amplifier output level. Speech components appearing in the amplifier output signal are then envelope-detected and filtered to provide the control signal. Alternatively, the control signal can be derived by envelope-detecting the audio signal, filtering the detected signal to remove its d.c. component representing the continuous noise, and then detecting and filtering again. In still another embodiment, the control signal acts upon a constant amplitude instead of the audio input signal in order to provide a speech-responsive tactile vibration for the deaf.

Direct digital loudspeaker

Abstract: A method of and apparatus for converting a digital signal to sound, the digital signal being encoded in a sequence of code words at a signal encoding frequency, the code words representing the analog sound pressure of an original audio signal, with decoding of the digital signal occurring after electro-acoustic transduction through mechanical rectification and characteristics of a listener's ear, includes utilizing a plurality of substantially identical sound pressure generating elements each having an individual driver associated therewith, and selectively energizing the drivers in a pulsed manner at the signal encoding frequency in combination in response to a respective order of the bits of each code word of a digital signal from a most significant bit to a least significant bit. The sum of the air pressures produced by the sound pressure generating elements in response to each of the successive code words of the digital signal has a magnitude corresponding to the analog value of the respective code word, and the auditory system of the listener has the characteristics of a low pass filter whereby the listener receives the sum of the air pressures as the analog sound pressure of the original audio signal.

Real time cochlear implant processor

Abstract: A real time cochlear implant processor (10) receives an audio input signal and extracts spectrum segments by operation of bandpass filters (18, 20). In each channel temporal circuitry detects the zero slope points of the filtered audio input signal and generates a control signal which operates a gate (66). Amplitude circuitry includes a reference voltage source for providing positive and negative polarity signals for a first mode of operation. The reference signals are routed through the gate (66) by the control signals. In a second mode of operation the audio input signal, after logarithmic amplification, is passed through the gate (66) by operation of the control signals. In a third mode of operation the audio input signal is squared through infinite clipping and the resulting signal is gated by the control signals through the gate (66). The outputs of the channels are combined in a summing circuit (78) and passed through a buffer (82) to produce a driver signal for an auditory implant electrode. The pulse outputs from the plurality of channels have different amplitudes where the higher frequency channels have lower amplitude pulses.

Toward an Intelligent Editor of Digital Audio: Signal Processing Methods

Abstract: In this article, we will describe signal processing methods that have been developed for use in an automatic music analysis system. A companion article by Chafe, Mont-Reynaud, and Rush, also in this issue of the Journal, deals with higher-level issues, namely the recognition of musical constructs. Unless one is willing to settle for a direct interface between musician and computer, such as the hardwired keyboard in the Xerox PARC system (Ornstein and Maxwell 1981), techniques must be developed to extract musical features from the sound itself. The approach of combining signal processing with knowledge engineering seems quite promising for music analysis. In contrast with many of the signals to which signal processing methods are applied, musical signals usually contain a great deal of order, chiefly in the form of quasi-periodicity (pitch and rhythm), and are not usually severely corrupted with random noise. By taking advantage of these features, one can construct mechanisms that provide musically significant descriptions of real data, such as tempo tracking ("foot-tapping"), meter analysis, attack characterization, pitch characterization (including vibrato), and timbre analysis. In this article and its companion, sample results of some promising strategies for accomplishing these goals are presented. In particular, we will concentrate on the problems of primary segmentation, that is, the first few passes through the data using little or no a priori knowledge. If we can mark the begin time for each new event in the music, the task of classifying and parameterizing each event is made easier. We have tried three approaches to this segmentation problem: (1) an amplitude thresholding method, (2) a linear predictive coding (LPC) method, and (3) a pitch detection method. While we will also discuss more advanced strategies, these are generally awaiting implementation and thus are not included in he examples.

Analyzing the signal transfer characteristics of a signal processing unit

Abstract: Method and means for evaluating the quality of audio and/or video transfer characteristics of a device upon which, or through which, audio and/or video information is contained, or passes, respectively. Both method and apparatus concern the evaluation of the quality of information transfer in the recording and playing back of a recording medium or in the transferring of audio and/or video information through an information handling device referred to as a throughput device. Unit evaluation is accomplished by establishing an input signal of known content, measuring selected parameters of selected parts of the input signal, feeding the input signal to the unit under test, measuring the parameters of parts of the output signal from the unit under test corresponding to the same selected parts of the input signal, and comparing the selected parameters of the input signal with the corresponding parameters of the output signal. Whether monitoring the quality of the signal transfer characteristics of a throughput device, a magnetic tape containing program material, or a video disc, master disc or replica, a "signature" is created for the unit under test, and subsequent analysis of the unit as it progresses along a production line or of a copy made on the same or alternate recording medium results in a second "signature" which is compared against the first signature to make a determination as to the quality of the signal handling or transfer characteristics of the unit. In this manner, out-of-tolerance conditions can be automatically detected, thereby eliminating subjectivity and providing consistency in the quality level of device testing.

Noise reduction circuitry for audio signals

Abstract: A signal substitution system is employed in audio processing circuitry to compensate for impulse noise and signal dropouts. Stored or delayed audio signal is substituted in the IF portion of the receiver circuitry ahead of the IF filters thereby precluding the noise impulses from exciting the filters into a ringing mode and significantly broadening what are otherwise very narrow signal disturbances. To insure that the substitution signal is in relative phase coherence with the substituted signal, the substitution signal is delayed an integral number, N, of cycles of the audio carrier. The number N is chosen to be small, resulting in the change in the modulating signal corresponding to the stored portion of the carrier being small. The affect on the baseband signal is that during instances where defects occurred the signal appears to have been sampled and held, but the duration is so short that they are not audibly distinguishable.

Method and apparatus for identifying sound recordings

Abstract: In an installation comprising a master assembly for reading the sound information signals and the end-of-program sequence signals recorded on a master tape, and at least one slave assembly to record a second time the read out signals, the slave assembly comprises circuits for detecting and regenerating end-of-sequence signals, which circuits are connected to the output of the said recording amplifier, digital coding circuits by derivation from a pilot frequency for producing a coded auxiliary digital signal, means to modify the digital coding at each new detection of end-of-sequence signals and an adder circuit to add the coded auxiliary digital signal with the sound information signal before re-recording the latter.

Modulator having improved bandwidth and power requirements

Abstract: A modulator is disclosed for use with digitized multiplexed voice channels in which service bits are transmitted using QPSK modulation while the audio information is transmitted using amplitude and phase modulation. The audio information in each channel is digitized and compressed using a mapping technique in which zero crossings of the audio signal are represented by binary O. The digitized channels are then multiplexed and coupled to the address inputs of a read-only memory. In the memory are stored digital numbers which represent in quadrature the amplitude and phase of the pulse to be transmitted corresponding to the digital data sample or service word then being presented at the address inputs of the memory. The digital numbers for the control words represent a constant amplitude while those corresponding to the data words represent an amplitude dependent upon the level of the original audio signal. No output carrier is produced for zero-value audio signals. A greater number of audio channels may be multiplexed within a given bandwidth than with previous systems while greatly reducing the required transmitter power.

A 2-channel, 16-bit digital sampling frequency converter for professional digital audio

Abstract: A digital sampling frequency converter for arbitrary ratios of sampling frequencies is presented. It is based on a multistage interpolating filter, and on a novel time-domain control of the filter stages by signals derived from the sampling frequency clocks. Time-domain resolution of ±300 picoseconds is obtained, compatible with digital audio of 16-bit resolution. In addition to the filter design and implementation, measurement results are presented. They indicate that 16-bit accuracy is indeed achieved, even with asynchronous, drifting and time-varying sampling frequencies. A number of applications (digital mastering, program transfer between conflicting digital audio formats, pitch control with constant sampling frequency in digital recorders, error concealment, interfaces in digital transmission) are presented.

Method of recording and reproducing visual information in audio recording medium and audio recording medium recorded with visual information

Abstract: A method of recording visual information such as words of a song in an audio recording medium and reproducing the same is provided. Picture information obtained through a TV camera is quantized into pulse signals. Using these pulse signals, a sine wave signal having an audio frequency band is modulated. Further, synchronizing signals are added to the modulated signal at predetermined intervals to form a character signal. Said character signal is recorded in one of the two channels of an audio recording medium where as a musical accompaniment is recorded in another channel. When played, such character signal is reproduced on a display screen such that a user can read the words to sing the song to the simultaneously reproduced accompaniment.

Apparatus for recording a digital audio signal in a track with a video signal

Abstract: Apparatus for recording and reproducing an information signal comprised of a video signal and an audio signal in a plurality of tracks extending obliquely on a magnetic tape includes a sample and hold circuit, an analog-to-digital converter and an encoder for converting the audio signal to a pulse code modulation (PCM) audio signal; a time base compression circuit for compressing the PCM audio signal; and two rotary magnetic heads for recording the compressed PCM audio signal in the plurality of tracks, each of the tracks being divided into a first leading audio track section, a contral video track section and a trailing audio track section, with the two heads recording one field interval of the video signal in the central track section of each track and recording the audio signal corresponding to one field interval of the video signal recorded in an adjacent track, in the first and second audio track sections of each track such that odd samples of the audio signal are recorded in each first leading audio track section and even samples of the audio signal are recorded in each second trailing audio track section

Switching arrangement for a stereophonic sound synthesizer

Abstract: A system for synthesizing stereophonic signals from an audio information signal source is provided. The system includes a transfer function circuit for producing a modulated signal which varies in amplitude as a function of frequency in response to the audio signal, first and second amplifiers for providing amplified output signals, and a switch circuit for coupling selected ones of the audio signal and the modulated signal to the inputs of the first and second amplifiers. In a first position, the switch circuit causes the first and second amplifiers to operate in a matrix mode for matrixing the input signals to provide first and second synthesized stereophonic output signals, and in a second position, the switch circuit causes the amplifiers to operate in a non-matrix mode with respect to the input signals supplied thereto for providing an amplified audio output signal.

Audio system with orderly shutdown sequence

Abstract: An audio system including audio processing circuit means interposed between a source of audio signals and a power amplifier. Power control circuitry including time delay means which respond to an on/off power switch for applying and removing power to the components of the audio system in a sequence which prevents audible pops associated with system start-up and shutdown.

Carding machine or roller card

Abstract: Apparatus for forming fiber material into a length of sliver. The apparatus includes: a machine for receiving such fiber material and forming it into the length of sliver; sensors for monitoring the operation of the machine and providing signals representative of that operation; an analog/digital converter connected to the sensors for receiving the signals provided by the sensors and converting those signals into digital representations of the operation of the machine; a digital electronic control unit connected to receive the digital representations formed by the analog/digital converter and including a microprocessor having memories, a device for generating digital representations of desired values of selected operating parameters of the machine, and a device performing operational, regulating, control and display functions, the control unit being arranged to provide digital signals for regulating the operation of the machine; a digital/analog converter connected to the control unit for deriving analog signals corresponding to the digital signals provided by the control unit; and a controllable regulator controlled by the analog signals derived by the digital/analog converter for regulating the operation of the machine.

The Lucasfilm Audio Signal Processor

Abstract: The requirements of audio processing for motion pictures present several special problems that both make digital processing of audio very desirable and relatively difficult. The difficulties can be summarized as follows: (1) Large amounts of numerical computation are required, on the order of 2 million integer multiply-adds per second per channel of audio, for some number of channels. (2) The exact processing involved changes in real time but must not interrupt the flow of audio data. (3) Large amounts of input/output capacity is necessary, simultaneous with numerical calculation and changes to the running program, on the order of 1.6 million bits per second per channel of audio. To this end, the digital audio group at Lucasfilm is building a number of audio signal processors the architecture of which reflects the special problems of audio.

Digital television receivers

Abstract: An analog carrier signal and its sidebands contain modulated information which is to be digitally encoded and processed. Demodulation and digital encoding is accomplished in a single process by an analog-to-digital converter, which samples the analog signals in response to a sampling signal of a frequency which is at least twice the bandwidth of the band of frequencies containing the carrier signal and its information-bearing sidebands. The analog signal samples are digitally encoded, producing a band of digital information signals corresponding to baseband signal components. When used in a television receiver to produce digital signal samples, the band of analog signals may include both sound and picture carriers and their audio and video information. By controlling the phase and frequency of the analog carrier signal in relation to the color subcarrier signal, and deriving the sampling signal from the analog carrier, an ease in video signal demodulation is provided.

System for recording and reproducing time-division multiplexed digital audio and video signals

Abstract: Analog primary color video signals of a still-picture converted to luminance and color difference components and converted to digital samples. The digital samples of each video component are written into a respective memory at the sampling rate and retrieved by a reading circuit at a rate lower than the sampling rate. Analog multi-channel audio signals are converted to digital samples which are time-division multiplexed with the retrieved digital samples of the video components. The multiplexed digital samples are recorded on a recording disc medium in a spiral track pattern.

Television signal bandwidth reduction using variable rate transmission

Abstract: A television signal transmission system incorporating circuits for processing and encoding a repetition reduced signal. The system contains a component separator for generating sampled digital values of the color and intensity components of a television signal, storage buffer for storing the separated components of the video signal, a data processor for comparing successive samples of the component video data and for generating variable-length blocks of data to represent either slowly varying signals or rapidly varying signals, circuitry for encoding and multiplexing audio and synchronization data into the signal stream, circuitry for encoding signal and control data for transmission to a receiver, circuit for processing the received signal to establish synchronization, circuit for demultiplexing the received signal to extract the audio information, and means for demultiplexing the component video information and reconstructing the composite video signal from the repetition-reduced representation.

Video signal recording and reproducing apparatus, including means for discriminating the mode of multiplexing of an audio signal

Abstract: A video signal recording/reproducing apparatus includes a recording medium having a video track and an audio track separately formed, wherein a video signal is recorded in the video track and an audio signal to be multiplexed and a discrimination signal of a low frequency and a low level representing the kind of multiplexing thereof are recorded in the audio track. Decision is made of the kind of multiplexing of the audio signal, such as a bilingual mode, a stereo mode or a monaural mode, as recorded in the audio track, in response to the descrimination signal reproduced from the audio track and a multiplexed audio signal is produced in response to the decision signal representing the kind of the audio signal and the audio signal reproduced from the audio track.

Personal acoustic alarm system

Abstract: The personal acoustic alarm unit produces an audible alarm signal in the audio range to alert persons in the area of the signal transmitter. This audio alarm signal is formed by a plurality of simultaneously generated audio frequency sonic signals, each of which has a different audio frequency. Receiver units are turned to receive the audio frequency sonic signals from a specific transmitter or group of transmitters.

Video and sound recording device with automatic focusing system

Abstract: There are video and sound recording devices such as movie cameras or video cameras with an automatic focusing system of the type for setting a photographic lens to a position at which an object can be sharply focused in response to the signal which is representative of a distance from the camera to the object and obtained by measuring the time interval from the time when the ultrasonic waves are transmitted at a predetermined repetition frequency to the object to the time when the echo from the object is received. Such devices have a common defect that a microphone picks up impulse-like noise generated from an ultrasonic transducer especially at the start point of ultrasonic waves. To overcome this problem, the present invention provides an audible noise elimination or suppression means or a control circuit which operates in synchronism with the ultrasonic wave transmission so that an audio signal processing system attenuates the audio output signal.

Digital-to-analog converter for bipolar signals

Abstract: In a digital-to-analog converter for bipolar signals all the bits change when the signals pass through the zero level. This result is a poor signal-to-noise ratio owing the small signal and the larger noise contribution of the switching transients. The invention is to add a digital number to or subtract it from the digital input signal as an offset. As a result of this, the point at which all the bits change is shifted towards a higher amplitude, which improves the signal-to-noise ratio and the distortion behaviour in the case of digital audio signals.