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Showing papers on "Audio signal processing published in 1978"


Book
05 Sep 1978
TL;DR: This paper presents a meta-modelling framework for digital Speech Processing for Man-Machine Communication by Voice that automates the very labor-intensive and therefore time-heavy and expensive process of encoding and decoding speech.
Abstract: 1. Introduction. 2. Fundamentals of Digital Speech Processing. 3. Digital Models for the Speech Signal. 4. Time-Domain Models for Speech Processing. 5. Digital Representation of the Speech Waveform. 6. Short-Time Fourier Analysis. 7. Homomorphic Speech Processing. 8. Linear Predictive Coding of Speech. 9. Digital Speech Processing for Man-Machine Communication by Voice.

3,103 citations


Book
01 Jan 1978
TL;DR: Applications of digital signal processing, Applications ofdigital signal processing , مرکز فناوری اطلاعات و £1,000,000; اوشاوρزی; کسراع رسانی ;
Abstract: Applications of digital signal processing , Applications of digital signal processing , مرکز فناوری اطلاعات و اطلاع رسانی کشاورزی

174 citations


Patent
03 Nov 1978
TL;DR: The block/sub-block format as mentioned in this paper provides an inherent simplification of error correction techniques, and consists of dividing the digital audio data into groups of digital words, and recording alternate words on separate tracks in the medium Cyclic redundancy check characters, synchronization, and parity information are selectively interspersed with the data to define sub-blocks which in turn are combined into separate blocks of the block arrangement.
Abstract: An audio signal is sampled, quantized and the values are stored in a specific digital data block/sub-block format on alternate tracks of a recording medium The block/sub-block format provides an inherent simplification of error correction techniques, and consists of dividing the digital audio data into groups of digital words, and recording alternate words on separate tracks in the medium Cyclic redundancy check characters, synchronization, and parity information are selectively interspersed with the data to define sub-blocks which in turn are combined into separate blocks of the block/sub-block arrangement On reproduce, any errors, dropouts, etc, are detected and corrected or concealed to reconstitute the original audio signal Editing is facilitated by the separated block arrangement which allows entering and exiting the record mode without destroying any data

118 citations


Patent
14 Sep 1978
TL;DR: In this article, a digital paging communication system including a transmitter and a plurality of receivers is disclosed, where the transmitter generates a preamble digital signal, a calling digital signal and an end mark digital signal in a predetermined sequence.
Abstract: A digital paging communication system including a transmitter and a plurality of receivers is disclosed. The transmitter generates a preamble digital signal, a calling digital signal and an end mark digital signal in a predetermined sequence. Each receiver demodulates a received signal into the preamble digital signal, calling digital signal and end mark digital signal. These signals are separately detected in synchronism with a recovered clock signal. Power is supplied to the radio frequency, intermediate frequency and demodulator sections of the receiver intermittently until such time as the calling signal of the receiver is detected, after which power is supplied continuously until an end mark signal is detected. The intermittent operation is a battery saving feature.

78 citations


Journal ArticleDOI
TL;DR: Real-time digital signal processing requires very fast multiplication, which is now becoming possible using mathematical techniques to take advantage of single-chip multipliers.
Abstract: Real-time digital signal processing requires very fast multiplication, which is now becoming possible using mathematical techniques to take advantage of single-chip multipliers.

71 citations




PatentDOI
TL;DR: In this paper, an improved audio signal processing system synthesizes from an audio signal, an enhanced audio signal by sensing signal energy of the audio signal within a preselected frequency portion of the signal, dividing the sensed signal energy into a plurality of discrete bands according to the frequency thereof and generating, responsively to the signal energy in each of the bands, a like plurality of second signals each of which includes frequency components which are subharmonics of the frequencies of the corresponding frequency band.
Abstract: An improved audio signal processing system synthesizes from an audio signal, an enhanced audio signal by sensing signal energy of the audio signal within a preselected frequency portion of the audio signal, dividing the sensed signal energy into a plurality of discrete bands according to the frequency thereof and generating, responsively to the signal energy in each of the bands, a like plurality of second signals each of which includes frequency components which are subharmonics of the frequencies of the corresponding frequency band. The second signals are combined so as to provide a combined signal and the latter is added to the audio signal to provide the enhanced audio signal.

32 citations


Patent
28 Aug 1978
TL;DR: In this paper, a random access memory is used to read out the previously entered values backwards, thus approximating the roughly symmetrical waveform envelope of normal high frequency musical signals so as to avoid discernable discontinuity in the audio output.
Abstract: The instantaneous amplitude of an analog audio signal is periodically sampled to be converted into multibit binary digital words representing the sampled amplitude. The bits of successive binary words are reproduced serially with appropriate word and frame synchronizing pulses in quaternary coding to pulse code modulate the FM carrier signal used in standard video recording. The recorded signal is demodulated on playback to reproduce successive digital words that are converted into discrete analog values which are low pass filtered to reproduce the original audio signal. The reproduced digital values are also stored in successive address locations of a random access memory. Upon detecting data drop-out, such as by monitoring the FM carrier in the reproduced signal, the memory address sequence is reversed to read out the previously entered values backwards, thus approximating the roughly symmetrical waveform envelope of normal high frequency musical signals so as to avoid discernable discontinuity in the audio output. At the same time, timing pulses are delivered to decrease gradually the bandwidth of the voltage controlled low pass filter to smooth the sample outputs from the memory. When actual data is reacquired, timing pulses used to increment the memory address gradually restore the maximum bandwidth of the voltage controlled low pass filter.

28 citations


Patent
Masao Inaba1, Sugimoto Atsumi1
01 Sep 1978
TL;DR: In this article, the authors proposed a delay compensation mechanism for a video signal transmitted through a plurality of frame synchronizers, which delay the video signal, and a normally undelayed audio signal, making up a complete television signal.
Abstract: Delay compensation apparatus for minimizing the delay differences between a video signal transmitted through a plurality of frame synchronizers, which delay the video signal, and a normally undelayed audio signal, the audio and video signal making up a complete television signal. Combinations of discreet delay intervals are imposed on the audio signal such that delay differences between the audio and video signal are minimized to the extent that the negligible remaining delay difference does not degrade the reproduced television picture.

27 citations


Patent
16 Nov 1978
TL;DR: In this paper, a system for recording stationary images on a convolution track of a video disk was proposed, in which a stationary image video signal of at least one frame was optically recorded on the track every revolution of the disk.
Abstract: A system for recording stationary images on a convolution track of a video disk in which a stationary image video signal of at least one frame is optically recorded on the track every revolution of the disk. An audio signal is recorded on portions of the track corresponding to every predetermined number of lines of the video signal. The audio signals are time compressed and are recorded on respective portions of the track corresponding to a predetermined number of lines of the video signal. During the read-out, the audio signals are time expanded. The video signal is reproduced from the video signals recorded in lines immediately preceding the audio signals. In another embodiment the video signal is reproduced with the mean of the video signals immediately preceding and subsequent to lines where the audio signals are recorded.

Patent
15 Dec 1978
TL;DR: In this article, a level-setter sets the level of the digital signal which has been compressed at a value which is lower than the white peak level of a composite video signal, and a mixer mixes the level set digital signal and the generated synchronizing signal to obtain a composite digital signal.
Abstract: A composite video signal recording and reproducing system also records and reproduces an audio signal after their conversion into a digital signal. The recording system converts an input analog audio signal into a digital signal and generates a synchronizing signal corresponding to a synchronizing signal of the composite video signal. There is a time-axis compression of the converted digital signal so that they do not exist in the period corresponding to the synchronizing signal. A level-setter sets the level of the digital signal which has been compressed at a value which is lower than the white peak level of the composite video signal. A mixer mixes the level set digital signal and the generated synchronizing signal, to obtain a composite digital signal. The composite digital signal is supplied to the recording and/or reproducing apparatus and recorded on a recording medium. The level-setter sets the amplitude of the digital signal at a magnitude such that the overshoots and undershoots occurring as a result of the passage of the digital signal through a pre-emphasis circuit of the recording and/or reproducing apparatus will not be clipped by a clipping circuit of the recording and/or reproducing apparatus.

Patent
Richard F. Abt1
18 Dec 1978
TL;DR: An audio processor for an FM stereo system including a peak clipping circuit to prevent overmodulation further includes means responsive to the audio signal before and after the clipping means for detecting the amount of energy lost due to peak clipping and gain control means for reducing the amplitude level of the clipped audio signal in proportion to the peak energy lost.
Abstract: An audio processor for an FM stereo system including a peak clipping circuit to prevent overmodulation further includes means responsive to the audio signal before and after the clipping means for detecting the amount of energy lost due to peak clipping and gain control means for reducing the amplitude level of the clipped audio signal in proportion to the peak energy lost to prevent overmodulation due to the shifting of the entire signal level which is associated with any asymmetry of the clipped signal.

Patent
12 Oct 1978
TL;DR: In this paper, a wireless transmission system for substantially noise-free transmission including a portable, battery powered, self contained transmitter for transmitting signals representative of an audio input to a remote receiver is presented.
Abstract: A wireless transmission system for substantially noise-free transmission including a portable, battery powered, self contained transmitter for transmitting signals representative of an audio input to a remote receiver The transmitter includes noise reduction encoding circuitry for processing a transduced audio input signal and providing a processed output signal to a transmission unit for transmission to the remote receiver The remote receiver receives the transmitted signals and supplies the received signals to a reception unit including noise reduction decoding circuitry for processing the received signal and providing an audio signal output with improved signal to noise ratio to a utilization device

Journal ArticleDOI
TL;DR: Applications of charge-coupled devices in signal processing systems, both analog and digital CCD concepts are considered, and projections for future uses of CCD's in signalprocessing systems are presented.
Abstract: The purpose of this paper is to discuss applications of charge-coupled devices in signal processing systems. Both analog and digital CCD concepts are considered. Recent developments in high-speed (~100 MHz) CCD's are discussed, and the uses of high-speed CCD and surface-acoustic wave (SAW) devices together are considered. Examples of the applications of CCD's in electro-optical systems, secure voice communication systems, sonar systems and radar systems are given. Finally, projections for future uses of CCD's in signal processing systems are presented.

Proceedings Article
01 Jan 1978
TL;DR: The DMX-1000 is an ultra-fast 16-bit minicomputer designed especially for audio signal processing applications, and gives this master computer, which is probably a slower, more generalpurpose machine, the ability to digitally synthesize highquality audio signals.
Abstract: The DMX-1000 is an ultra-fast 16-bit minicomputer designed especially for audio signal processing applications. The DMX-1000 is mostly digital. It synthesizes sound by calculating sampled waveform values and converting the stream of samples to analog form with one or more digital-toanalog converters (DACs). The DMX-1000 is designed to be added as a peripheral unit to another computer, called the master. It gives this master computer, which is probably a slower, more generalpurpose machine, the ability to digitally synthesize highquality audio signals. The DMX-1000 does the high-speed repetitive number-crunching required for the synthesis, and the master controls it in real-time. It is by microprogramming that the master computer controls the DMX-1000. The master provides a program and a set of parameters for it; the DMX 1000 executes the program over and over. Each execution of the program produces an output sample. Since the DMX-1000 has no branch instructions, any program will take the same amount of time to run each time it is executed; the samples will be evenly spaced. There are two high-speed memories in the DMX-1000the program memory and the data memory. The master computer may write into either of these at any time. The master provides the program for the DMX-1000 by writing it into the program memory. The data memory is used to hold constants, parameters, state variables, waveform lookup tables, and so forth.

Patent
13 Jul 1978
TL;DR: In this paper, an analog to digital converter for converting an analog signal into digital signals in the form of words each made up of a plurality of binary digits, and an encoding device for transforming the digital signals into digital words each having a predetermined number of identical binary digits.
Abstract: A digital processing system is disclosed, including an analog to digital converter for converting an analog signal into digital signals in the form of words each made up of a plurality of binary digits, and an encoding device for transforming the digital signals into digital words each having a predetermined number of identical binary digits.

PatentDOI
TL;DR: In this paper, a system for remote warning sounds reflected from a roadway surface supporting an automobile includes a sound transducer mounted in the underside of a fender well of the vehicle for producing an audio frequency electrical signal in response to the received reflected warning sounds.
Abstract: A system for receiving remote warning sounds reflected from a roadway surface supporting an automobile includes a sound transducer mounted in the underside of a fender well of the vehicle for producing an audio frequency electrical signal in response to the received reflected warning sounds. The system includes an audio amplifier connected to the sound transducer for amplifying the audio frequency signal. The output of the audio amplifier is connected to inputs of a plurality of tone decoder circuits. Each tone decoder circuit is responsive to an audio frequency signal in a predetermined frequency band. The frequency bands of the plurality of tone decoder circuits are mutually exclusive. Each tone decoder circuit produces an output signal having a first logic level if an electrical signal having a frequency within its frequency range is received from the audio amplifier and a second logic level if no such electrical signal is received. If any of the tone decoders receive an input signal within their respective ranges, a warning light visible to the automobile driver is turned on and a second sound transducer is activated to produce warning sound audible to the automobile driver.

Patent
13 Mar 1978
TL;DR: In this article, a method and system for permitting high quality analog signal processing to multiple channel digital information signals, which requires a minimal number of analog signals processing elements and which is transparent to a host communication system is presented.
Abstract: A method and system for permitting high quality analog signal processing to multiple channel digital information signals, which requires a minimal number of analog signal processing elements and which is transparent to a host communication system Multiple channel digital information signals are time compressed by a factor sufficient to permit subsequent digital-to-analog conversion and analog signal processing, such as DTMF or FSK filtering, and conversion from one digital code to another, to be effected by circuitry which is common to all of a given number of information channels, with the processed analog signals then converted to resulting digital signals which are time expanded in such a manner as to be completely compatible with the digital signal flow format of the host communication system Time compression is effected by storing the individual input digital signals in an input memory in their order of appearance at substantially the data input rate and reading the stored characters from the input memory at a substantially greater rate in information channel groups, each group comprising a predetermined number of samples from a particular channel Each information channel group is processed in the analog domain and the resulting signals are converted to digital form and time expanded by storing the converted digital signals in an output memory and reading the stored signals from the output memory at rates preselected to provide the complementary time expansion Both contiguous time processing and packet processing, as well as a special time encoding/time decoding technique can be employed in the invention

Patent
13 Nov 1978
TL;DR: In this article, a plurality of digital voice signals are summed to produce a new digital nonlinear signal representative of a combination of original voice signals for transmittal to each of the conferees in a conference call.
Abstract: Apparatus is illustrated for converting nonlinear or compressed digital information representative of an analog signal from a plurality of sources into a single composite digital signal. As specifically used and illustrated, a plurality of digital voice signals are summed to produce a new digital nonlinear signal representative of a combination of original voice signals for transmittal to each of the conferees in a conference call. This is accomplished without reverting to analog signals during the combination process.

Patent
16 Jun 1978
TL;DR: In this paper, a system for providing digital data representative of a selected analog signal on the data lines of a programmable controller using a central processing unit employs a conversion circuit which converts the input analog signal to digital data on output data terminals upon receiving of a conversion signal simultaneously with an analog signal and creates a completion signal when the conversion is completed.
Abstract: A system for providing digital data representative of a selected analog signal on the data lines of a programmable controller using a central processing unit. The system employs a conversion circuit which converts the input analog signal to digital data on output data terminals upon receipt of a conversion signal simultaneously with an analog signal and which creates a completion signal when the conversion is completed. The system can use at least two analog inputs that receive at least two analog conditions and convert a selected one of the analog conditions to an analog signal. Upon actuation of one of the input modules, the selected analog signal of the selected module is directed to the conversion circuit. After conversion, the conversion completion signal of a conversion circuit then deactivates the actuated input module for the next conversion cycle.

Patent
22 Sep 1978
TL;DR: In this paper, an attenuation control signal is generated from the difference between the radio AGC signal and a predetermined constant reference, which is used to attenuate the audio frequency signal.
Abstract: A radio receiver includes apparatus for attenuating the audio signal derived from weak and noisy radio frequency signals to render the resulting noisy output audio signals less objectionable. Apparatus is provided to attenuate the AGC audio frequency signal in a predetermined constant attenuation, linearly mix the AGC regulated and attenuated audio frequency signals in a proportion controlled by an attenuation control signal and generate the attenuation control signal from the difference between the radio AGC signal and a predetermined constant reference.

Patent
03 Oct 1978
TL;DR: In this article, the authors propose to obtain excellent audio effect with natural feeling of echo by reproducing the input signal with the main speakers after amplification and reproducing it with auxiliary speakers by delaying and amplifying it.
Abstract: PURPOSE:To obtain excellent audio effect with natural feeling of echo, by reproducing the input signal with the main speakers after amplification and reproducing it with auxiliary speakers by delaying and amplifying it. CONSTITUTION:The signal from the input terminal 11 having the signal R of right channel and the signal L of left channel, is reproduced from the main speakers 15 via the first power amplifier 12 as it is on one hand, and reproduced from the auxiliary spealers 16 from the second power amplifier 14 after being delayed by time tau, at the dalay element 13 on the other hand. The listener can listen to the stereo sound having feeling of echo by left and right ears for the mixing tone of the reproduced sound from the main and auxiliary speakers 15 and 16. In other case, as the input to the delay element 13, the mixing signal of left and right both channels can be used.

Patent
21 Sep 1978
TL;DR: In this paper, the first and second variable attenuators are mechanically interlocked and arranged in such a way that the sum of their attenuating factors remains constant irrespective of any attenuating operation.
Abstract: An audio signal transmission circuit comprising a first variable attenuator (i.e. volume controller) from which a level-adjusted audio signal is derived, a tone control circuit connected to the output of the first variable attenuator and supplying a tone-controlled audio signal to a loudspeaker, and a second variable attenuator connected between the tone control circuit and a record terminal and feeding the tone-controlled audio signal to the record terminal. The first and second variable attenuators are mechanically interlocked and arranged in such a way that the sum of their attenuating factors remains constant irrespective of any attenuating operation. At the record terminal there can be obtained the tone-controlled audio output signal with a substantially constant level (amplitude level), irrespective of the level adjusting operation of the signal to the loudspeaker.

Patent
30 Jan 1978
TL;DR: In this article, a method and apparatus for storing and recovering a digital signal indicative of digital data wherein an audio cassette tape recorder is used to record, in analog form, digital data.
Abstract: Method and apparatus are disclosed for storing and recovering a digital signal indicative of digital data wherein an audio cassette tape recorder is used to record, in analog form, digital data. The playback signal is processed through circuitry to produce one signal indicative of positive peaks in the playback signal, and a second signal indicative of negative peaks. The set and reset terminals of a flip-flop are responsive respectively to the latter two signals so as to reconstruct the original data signal at the output of the flip-flop.

Patent
30 Jun 1978
TL;DR: In this article, an audio frequency input connection is disclosed for connection to the antenna input of a television receiver, which includes means to convert an audio signal into a pleasantly varying form for display as a varying pattern on the picture tube of a TV receiver.
Abstract: An apparatus is disclosed for connection to the antenna input of a television receiver. The apparatus has an audio frequency input connection and includes means to convert an audio signal into a pleasantly varying form for display as a varying pattern on the picture tube of a television receiver. The apparatus constantly samples and converts the audio signal and stores a value into an image memory which represents an instantaneous time sampled pixel (picture element) and constantly displays the contents of the image memory on the picture tube of a television receiver. The apparatus samples and converts the audio signal and stores a value into the image memory which represents a Lissajous pattern which is varied in size and shape in accordance with the audio signal.

Patent
25 Sep 1978
TL;DR: The free space transmission system as mentioned in this paper transfers audio signals from a microphone to an amplifier and/or sound recorder, and converts the audio signal into a signal of higher frequency by suppressed-carrier single-sideband modulation and radiates it as ultrasound via an electroacoustic transducer.
Abstract: The free-space transmission system transfers audio signals from a microphone to an amplifier and/or sound recorder. It converts the audio signal into a signal of higher frequency by suppressed-carrier single-sideband modulation and radiates it as ultrasound via an electroacoustic transducer. The ultrasonic wave is received on the normal way by a receiver with relatively high directivity. The directivity may be achieved by using an interference tube or by using a reflector. The receiver may be fitted into a film camera.

Patent
10 Oct 1978
TL;DR: In this article, a circuit for suppressing pulse-shaped interferences in an audio signal caused by scratches on a phonograph disc, comprising a signal processing section connected between an audio frequency input and an audio-frequency output, including a noise suppressor having a control input, was proposed.
Abstract: A circuit for suppressing pulse-shaped interferences in an audio signal caused by scratches on a phonograph disc, comprising a signal processing section connected between an audio frequency input and an audio frequency output, comprising a noise suppressor having a control input; as well as a control signal section connected between the audio frequency input and the control input, said control signal section comprising a first threshold selection circuit connected to the audio frequency input for selecting pulse-shaped signals from the audio signal, followed by a second threshold selection circuit for distinguishing the pulse-shaped signals originating from noise signals from those originating from musical signals.

Journal ArticleDOI
TL;DR: The conceptual ideas and design features of a low cost digital signal processor intended for data acquisition/filtering and decision/control in telecommunications are discussed.
Abstract: The conceptual ideas and design features of a low cost digital signal processor intended for data acquisition/filtering and decision/control in telecommunications are discussed. The speed, lowcost and flexibility of the processor appear as a result of a mixed firmware/software architecture used in conjunction with a multi-level distribution of processing. The system is easily reprogrammable by replacing the appropriate ROM IC's.

Patent
05 Jan 1978
TL;DR: In this paper, a digital voltage accumulator is used to store a digital signal over an extended period of time, the digital signal representing the accumulated sum of values of an analog input signal as sampled during a system reset command.
Abstract: A digital voltage accumulator functioning to store a digital signal over an extended period of time, the digital signal representing the accumulated sum of values of an analog input signal as sampled during the period of time following system reset command. The accumulator is capable of selectively incrementing or decrementing the stored signal, following an update command, in response to amplitude and polarity changes in the analog input signal. Signal storage is effected using a digital counter and a bipolar digital to analog converter. The analog equivalent to the stored digital signal is arithmetically summed with the analog input signal. Update is accomplished by comparing this summed signal with the analog equivalent of the stored digital signal upon command. The digital counter increments when the sum signal is positive relative to the stored signal and decrements when the sum signal is negative relative to the stored signal. The update cycle stops when the comparator senses that the stored digital signal equals the arithmetic sum of the analog input signal plus the old value of the stored digital signal.