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Showing papers on "Audio signal processing published in 1984"



Patent
15 Aug 1984
TL;DR: In this article, a portable cordless telephone transceiver-radio receiver includes a headset which is driven by the audio output of a radio receiver until an incoming radio frequency (RF) telephone ring signal is detected by a telephone receiver.
Abstract: A portable cordless telephone transceiver-radio receiver includes a headset which is driven by the audio output of a radio receiver until an incoming radio frequency (RF) telephone ring signal is detected by a telephone receiver. A telephone audio ring signal is generated based on the detected RF ring signal and the audio ring signal is converted to sound at the headset. The audio ring signal operates a latch which blocks or squelches the audio output of the radio receiver from the headset until the latch is reset. A "talk/off" switch is operated by the user to initiate and terminate conversation and to automatically reset the latch. In another embodiment, the radio receiver is squelched or turned off directly by the "talk/off" switch. In a further embodiment, the audio ring signal is blocked from the head set until a prescribed number of audio ring signals have been detected.

97 citations


Patent
26 Sep 1984
TL;DR: In this article, the authors present a method and a system for rapidly attaining a large number of different sounds in an audio signal processing system linkable to, or incorporable in, any audio device.
Abstract: A method and a system for rapidly attaining a large number of different sounds in an audio signal processing system linkable to, or incorporable in, any audio device. A preferred embodiment comprises an audio signal processing system which includes a main housing (1), containing a main circuit apparatus, cartridge-receiving recesses (2), and cartridges (3) fabricated to be removable received in the recesses, each cartridge containing an audio effects circuit which affects the signal in a unique way to provide a different sound output. A plurality of electrical contacts (16, 20) of the recess and cartridge, respectively, provide disengageable connection of the audio effects circuit to the system. In order to avoid the generation of spurious signals into the main signal path during the changing of a cartridge, and to maintain the uninterrupted flow of the main signal path through the system, switching devices (4, 15) are connected such that the audio signal is noiselessly routed, flowing either to and from a recess via and through a cartridge therein, or passing by the recess to the next recess. Switches (15) are further provided for maintaining the continuity of the audio path while the recess is empty. The cartridges are inserted and removed producing, respectively, in the audio path, the quiet appearance and disappearance of the effect of its audio effects circuit. Pannel-mounted switches (8) and foot switches (72, 73) are provided to command bypass of any or all recesses whether said recess is devoid of a cartridge or contains a cartridge fully or otherwise installed therein.

50 citations


Patent
16 May 1984
TL;DR: In this article, an analog signal is converted into a digital signal to provide a sequence of digital signal samples corresponding to the analog audio signal, and each digital signal sample is compressed to provide compressed signal samples having a sign bit, three exponent bits and seven mantissa bits.
Abstract: In the scrambling system, an analog audio signal is converted into a digital signal to provide a sequence of digital signal samples corresponding to the analog audio signal. Each digital signal sample is compressed to provide compressed signal samples having a sign bit, three exponent bits and seven mantissa bits. Each bit of each compressed signal sample is exclusive-OR'd with a unique keystream to thereby scramble the audio signal. A Hamming code generator generates code bits for correcting singular errors in a combination of the sign bit, the exponent bits and the code bits; and a parity bit generator generates a parity bit for detecting double errors in a combination of the sign bit, the exponent bits and the code bits and for further detecting an error in the most significant mantissa bit and/or the parity bit. The bits from a plurality of successive compressed, error-encoded signal samples are interleaved and serialized in order to separate the bits from any single sample by at least a predetermined duration associated with an FM discriminator click. The serialized, interleaved, error-encoded, compressed signal samples are combined to provide two-bit digital words. The digital words are converted to digital PAM data signals which when converted to an analog signal by digital-to-analog conversion, provide a pulse-amplitude-modulated signal having a level related to the binary value of the digital words. The digital PAM data signals are converted to an analog signal to provide the pulse-amplitude-modulated signal. The descrambler system descrambles the scrambled audio signal by a process that is the converse of the scrambling process. Singular errors in a scrambled signal sample are detected and corrected by a Hamming error corrector. Double errors in a scrambled signal sample are detected by a parity bit check and compensated for by repeating the last received error free signal sample.

50 citations


Patent
06 Sep 1984
TL;DR: In this paper, a plurality of audio messages along with associated video data are combined along with corresponding audio message start address signals when recorded on a recording medium, and the address signals are utilized to access selectable messages for decoding and playback of the selected audio message.
Abstract: Method and apparatus for playing back a plurality of digitally encoded audio messages along with associated video data. The audio messages are combined along with a plurality of corresponding audio message start address signals when recorded on a recording medium. The digitally encoded information can be recorded in the audio domain, i.e. as digitally audio recorded on an audio channel of the medium, or in the video domain, i.e. as digitized audio recorded on the video channel of the medium. In playback, the address and audio data signals are retrieved from the recording medium and stored. The address signals are utilized to access selectable messages for decoding and playback of the selected audio message.

45 citations


Patent
19 Apr 1984
TL;DR: In this article, a microcomputer system for converting an analog input signal, such as an audio signal representative of sound into a digital form for storing in digital form in a highly condensed code and for reconstructing the analog signal from the coded digital form is disclosed.
Abstract: A microcomputer system for converting an analog input signal, such as an audio signal representative of sound into a digital form for storing in digital form in a highly condensed code and for reconstructing the analog signal from the coded digital form is disclosed. The system includes reductive analytic means where the original digital data stream is converted to a sequential series of frequency spectrograms, signal amplitude histograms and waveform code tables. Approximately 100 times less storage space than previously required for the storage of digitized audio signals of high fidelity quality is thereby obtained. Additive synthesis logic interprets the stored codes and recreates an output digital data stream for digital to analog conversion that is nearly identical to the original source material.

45 citations


Patent
02 Feb 1984
TL;DR: In this paper, a system and method for detecting and marking television commercials is presented, where a first comparator is utilized for comparing the television video signal to an adjustable reference voltage and for generating a triggering signal when a video signal is not present at the comparator input.
Abstract: A system and method for detecting and marking television commercials is provided. A first comparator is utilized for comparing the television video signal to an adjustable reference voltage and for generating a triggering signal when a video signal is not present at the comparator input. A second comparator compares the television audio signal to a second adjustable reference voltage to provide a triggering signal when an audio signal is not present at the comparator input. The outputs from the two comparators are supplied to a one-shot timer mechanism which is adapted to generate an output signal of a predetermined time period only upon the simultaneous absence of an audio and video signal, thus signifying a fade break in the television broadcast signal. The timed output signal from the timer mechanism is connected to a low frequency oscillator which generates a low frequency output signal upon being actuated by the output from the timer mechanism.

44 citations


Patent
26 Sep 1984
TL;DR: In this paper, a method for recording and reproducing a video format signal including audio data and a video data signal, the audio data is divided into groups of audio data, and identification codes are inserted into the video format signals as required.
Abstract: In a method for recording and reproducing a video format signal including audio data and a video data signal, the audio data is divided into groups of audio data, and identification codes for the groups of audio data are inserted into the video format signal as required. For reproducing the video format signal, a sound specifying instruction is compared with the identification codes to detect the group of audio data which has been specified so that the group of audio data thus detected is reproduced while the video data signal is reproduced, for instance, as a still picture.

41 citations


Patent
Tomimitsu Yasuharu1
05 Oct 1984
TL;DR: In this article, a digital signal processing apparatus with a rate-conversion function has at least two digital filters and a memory, and the memory temporarily stores a time-division-multiplexed signal which is sequentially read out of the memory and selectively transferred to the digital filters.
Abstract: A digital signal processing apparatus with a rate-conversion function has at least two digital filters and a memory. A digital signal stored in the memory is selectively transferred to the at least two digital filters. These digital filters perform a filtering operation in parallel, and the results from each of the digital filters are alternately derived by a multiplexer. Thus, high-speed filtering can be executed. The memory temporarily stores a time-division-multiplexed signal which is sequentially read out of the memory and selectively transferred to the digital filters. Thus, a plurality of digital signals can be filtered by the same digital filters without an increase in hardware elements. Therefore, the digital filter section can be integrated in a single semiconductor chip. A shift register may be used as the memory, whereby a circuit arrangement of the digital filter section can be extremely simplified.

38 citations


Proceedings ArticleDOI
19 Mar 1984
TL;DR: The GRASP system (for Grouping Research on Auditory Sound Processing) is a part of that overall framework for sound separation and uses the physical cues present in the acoustic signal to decide how many sounds are present and of what each sound consists.
Abstract: Any speech recognition system designed to function in environments with other sounds present must deal with extraneous sounds. A hierarchical separation and recognition system is proposed as a computational model of how the auditory system processes sounds. The GRASP system (for Grouping Research on Auditory Sound Processing) is a part of that overall framework for sound separation. GRASP is a computational model of the data-driven aspects of auditory separation. It uses the physical cues present in the acoustic signal (such as pitch and onsets) to decide how many sounds are present and of what each sound consists. Its initial use is to separate simultaneously spoken digits from different speakers.

36 citations


Patent
07 Nov 1984
TL;DR: In this paper, a low-pass filter is used to limit the frequency band of the input signal to be supplied to the dynamic range compressor (20) to at most half the sampling frequency of the MODEM (9) and thus permits the noise reduction circuit (6) to compress and expand signals supplied in the same frequency band.
Abstract: An audio signal transmission system having a noise reduction circuit (6), which includes a transmission medium (101) such as a magnetic tape to be used to transmit a digital signal converted from an input audio signal and thereafter to output the audio signal reproduced from the digital signal, a digital signal transmitter having a pulse-code modulator and demodulator (MODEM) (9). The noise reduction circuit (6) includes a compressor (20) for compressing the dynamic range of the input audio signal according to the detection level obtained by detecting at least the high-frequency components thereof in order to supply the output from the compression to the digital signal transmitter and an expan- ,der (21) for expanding the dynamic range of the demod-I ulated signal from the transmitter according to the detection level of at least the high-frequency components thereof. The transmission system has a low-pass filter which limits the frequency band of the input signal to be supplied to the dynamic-range compressor (20) to at most half the sampling frequency of the MODEM (9) and thus permits the noise reduction circuit (6) to compress and expand signals supplied thereto in the same frequency band without causing any erroneous operation.

Patent
10 Sep 1984
TL;DR: In this article, a reproducing device (10) and a pitch changing device (20) are used to reproduce audio signals at a variable speed, at variable sampling frequency, at a recording medium, on which the audio signal data has been recorded at a predetermined sampling frequency FS.
Abstract: An audio signal reproducing apparatus includes a reproducing device (10) and a pitch changing device (20). The reproducing device (10) reproduces, at variable speed, audio signal data from a recording medium (11) on which the audio signal data has been recorded at a predetermined sampling frequency FS. The pitch changing device (20) writes into a memory the reproduced audio signal data obtained by the reproducting device (10) by means of a write clock which corresponds to a reproduction sampling frequency fs and reads out the reproduced audio signal data from the memory by means of a read clock which corresponds to an oscillation frequency fp from an oscillator which oscillates at a predetermined sampling frequency Fs, thereby effecting pitch change processing with respect to the reproduced audio signal. The pitch changing device (20) is composed of a pitch change processing section (21) which effects pitch change processing with respect to the reproduced audio signal at any desired pitch change ratio, and a pitch ratio control section (22) which automatically sets the pitch change ratio in relation to the pitch change processing section to a predetermined value by means of a signal representing the start or end point of the reproduced audio signal.

Patent
02 Mar 1984
TL;DR: In this paper, a code signal representing either of the two-channel digital audio signals and the digital data is recorded and control data indicative of the linear velocity of the digital disc upon reproduction are inserted in every frame of the recording signal.
Abstract: Digital data consisting of different kinds of digital audio signals having sampling frequencies, numbers of quantization bits and the like that are different from those of the two-channel digital audio signals, display data, program data, and the like are recorded in the same signal format as that of the two-channel digital audio signals. A code signal representing either of the two-channel digital audio signals and the digital data is recorded and control data indicative of the linear velocity of the digital disc upon reproduction are inserted in every frame of the recording signal. One block of the digital data is formed by a predetermined amount of data and the same control data used for controlling the processing of the digital data is recorded at least twice in one block. The digital disc is rotated at a constant linear velocity on the basis of the control data upon reproduction. The reproduced audio data from the reproducing circuit is D/A converted, and the pass band of the low-pass filter to which the reproduced analog audio signals are supplied is adapted to the band of the reproduced analog audio signals.

Patent
09 Nov 1984
TL;DR: In this paper, a digital signal transmitting system includes a signa' generator (22 to 24) for producing a time division multiplexed ditigal audio signals and service bit signals, modulator (26) for modulating a carrier in accordance witt the time division-multiplexed signals and a transmitter (20) to transmit the modulated carrier through a CATV (cable television) transmission line.
Abstract: A digital signal transmitting system includes a signa' generator (22 to 24) for producing a time division multiplexed ditigal audio signals and service bit signals, modulator (26) for modulating a carrier in accordance witt the time division-multiplexed signals and a transmitter (20) to transmit the modulated carrier through a CATV (cable television) transmission line. At a receiver side, the service bit signals are decoded so as to control the channel change-over of a television receiver (30) when an emergency broadcast signal is transmitted. A muting control circuit is also provided so as to mute an audio signal from a D/A (digital-to-analog) converter (52).

PatentDOI
TL;DR: In a hearing aid of the type having an electrical amplifier, wherein the improvement provides an audio signal processing circuit incorporated into the amplifier and including integral computing circuitry for controlling the operation thereof.
Abstract: In a hearing aid of the type having an electrical amplifier, wherein the improvement provides an audio signal processing circuit incorporated into the amplifier and including integral computing circuitry for controlling the operation thereof; the processing circuit providing circuitry for receiving an audio frequency electrical signal, circuitry for separating the audio frequency signal into a plurality of frequency bandwidths, circuitry for separately amplifying any audio frequency signal present in each of the bandwidths, circuitry for summing the amplified audio frequency signals from each of the bandwidths to produce a reconstituted signal, and circuitry for controlling amplification including circuitry for sampling any audio frequency signal present in each of the bandwidths and circuitry for determining, in response to the audio frequency signals sampled from all of the bandwidths the amplification level for each bandwidth for the separate amplification.

Patent
08 Mar 1984
TL;DR: In this paper, a data signal reproduction apparatus reproduces a digital signal from a recording medium on which a digital data signal, which is processed according to run length limited code modulation, is recorded with a synchronizing signal portion.
Abstract: A data signal reproduction apparatus reproduces a data signal from a recording medium on which a digital data signal, which is processed according to run length limited code modulation, is recorded with a synchronizing signal portion. The apparatus is improved so that erroneous detection of the synchronizing signal is prevented when there is a defective part on the recording surface of the recording medium. The apparatus is arranged so that the supply of a rectangular-wave reproduction signal from a waveform shaping unit (9) to a reproduction signal processing unit (13) is inhibited when the amplitude of a read signal detected by an amplitude detection unit (14 to 17) is lower than a predetermined value, thereby preventing erroneous detection of the synchronizing signal. The apparatus is suitable for the reproduction of a data signal from a digital audio disc.

Proceedings ArticleDOI
01 Mar 1984
TL;DR: The Integrated Signal Processing System is a Lisp machine-based workstation which provides a unified environment for signal data processing and the development of signal processing algorithms.
Abstract: The Integrated Signal Processing System (ISP) is a Lisp machine-based workstation which provides a unified environment for signal data processing and the development of signal processing algorithms. ISP is based on a model of signal processing computation in which the fundamental activities are creating and manipulating abstract signal objects. ISP consists of three main subsystems. The signal representation language (SRL) formalizes the semantic foundation of ISP and provides a set of facilities for defining signal classes and creating instances of signal objects. The ISP environment provides a signal stack, signal pictures and signal display layouts which are used to create and view selected signals from the universe defined by SRL. Finally, the user interface consists of a number of interactive mechanisms for manipulating components of the environment.

Patent
02 Mar 1984
TL;DR: In this paper, a code signal representing either of the two-channel digital audio signals and the digital data is recorded and control data indicative of the linear velocity of the digital disc upon reproduction are inserted in every frame of the recording signal.
Abstract: PCT No. PCT/JP84/00081 Sec. 371 Date Oct. 23, 1984 Sec. 102(e) Date Oct. 23, 1984 PCT Filed Mar. 2, 1984 PCT Pub. No. WO84/03580 PCT Pub. Date Sep. 13, 1984.Digital data consisting of different kinds of digital audio signals having sampling frequencies, numbers of quantization bits and the like that are different from those of the two-channel digital audio signals, display data, program data, and the like are recorded in the same signal format as that of the two-channel digital audio signals. A code signal representing either of the two-channel digital audio signals and the digital data is recorded and control data indicative of the linear velocity of the digital disc upon reproduction are inserted in every frame of the recording signal. One block of the digital data is formed by a predetermined amount of data and the same control data used for controlling the processing of the digital data is recorded at least twice in one block. The digital disc is rotated at a constant linear velocity on the basis of the control data upon reproduction. The reproduced audio data from the reproducing circuit is D/A converted, and the pass band of the low-pass filter to which the reproduced analog audio signals are supplied is adapted to the band of the reproduced analog audio signals.

Patent
Clyde Robbins1
04 Dec 1984
TL;DR: In this article, a cable television converter is retrofitted for use in reproducing stereophonic broadcast television sound, and a composite signal is produced comprising the scrambled output signal from the converter tuner, the reception enable control signal from a digital processor, and the output signal of the converter's remote control detection circuit.
Abstract: A cable television converter is retrofitted for use in reproducing stereophonic broadcast television sound. A composite signal is produced comprising the scrambled output signal from the converter tuner, the reception enable control signal from the converter's digital processor, and the output signal from the converter's remote control detection circuit. The composite signal is applied to a stereo adapter which separates the audio signal portion of the composite signal into left and right stereo signals, extracts the enable control signal from the composite signal, and outputs the left and right stereo signals for audio reproduction upon authorization indicated by the extracted enable control signal. Remote volume control and selection of a second audio program (SAP) is also provided.

Patent
30 Mar 1984
TL;DR: In this paper, a portable battery powered tone source with a built-in audio cable tester is presented, which can be used to test any direct point-to-point connection such as common ground or a PC board run with a simple test lead adaptor.
Abstract: A portable battery powered tone source with a built-in audio cable tester. Plural outputs consist of -40 dB, mic level; -10 dB line level and 1W speaker level. Continuity in the system under test from input to speakers causes an audio frequency signal to be present at speakers. This systems own speakers can be used as monitors in the testing process. Moving the device (source) enables the user to audibly locate the point at which the continuity starts/stops. At such time the user can test the audio castle at that point. The various plugs of the device are compatible with all common audio jacks enabling the user to test all types of audio equipment as well as all types of audio cables with only one device. The device can also be used to test any direct point-to-point connection such as common ground or a PC board run with a simple test lead adaptor.

Journal ArticleDOI
TL;DR: A discussion of the principles and equipment used in a compact disk digital audio system can be found in this article, where the authors present a detailed analysis of the equipment and software used.
Abstract: A discussion is presented of the principles and equipment used in a compact-disk digital audio system.

Patent
03 Dec 1984
TL;DR: In this paper, a preamplifier adapted to receive audio frequency electrical signals generated by a musical instrument for providing controlled output signals therefrom having selected distortion and harmonic signal content is shown.
Abstract: A preamplifier adapted to receive audio frequency electrical signals generated by a musical instrument for providing controlled output signals therefrom having selected distortion and harmonic signal content is shown wherein the preamplifier includes an amplifier for amplifying audio frequency electrical signals generated by the musical instrument, a first electrical signal processing channel operatively coupled to the amplifier for receiving the amplified audio frequency electrical signals and for producing in response thereto a first controlled output signal wherein the first electrical signal processing channel includes a plurality of selectively interconnectable signal processing circuits at least one of which is capable of at least one of selectively amplifying, distorting and adding preselected harmonic content to the first controlled output signal, a second electrical processing channel operatively coupled to the amplifier for receiving the amplified audio electrical signals and for producing response thereto a second controlled output signal and wherein the second electrical processing channel includes a plurality of selectively interconnectable signal processing means at least one of which is capable of at least one of selectively amplifying, distorting and adding preselected harmonic content to the second controlled amplitude output signal and an output circuit operatively coupled to the first and second electrical signal processing channels for selectively applying one of the first and second controlled output signals to an output terminal is shown. A method for providing controlled output signals having selected distortion and harmonic signal content is also shown.

Patent
30 Nov 1984
TL;DR: In this article, an analog video signal digitization and correction system scans a medium to produce video data signals representing images printed on the medium, and two reference quantities representing analog reference signals produced by scanning a densely printed image and an area of the medium having no printed image are used to establish the maximum and minimum values of a range of possible values for each video data signal.
Abstract: An analog video data signal digitization and correction system scans a medium to produce video data signals representing images printed on the medium. Two reference quantities representing analog reference signals produced by scanning a densely printed image and an area of the medium having no printed image are used to establish the maximum and minimum values of a range of possible values for each video data signal. The analog video data signals are compared with this range and a corrected digital value obtained for each data signal depending on its position in the range. In generating the reference quantities each analog reference signal is converted into a digital reference signal dependent on the position of the analog signal in a range of possible values. For each analog reference signal only a portion of the range of possible values is selected in order to increase the accuracy of the corresponding digital reference signal. The number of bits in the digital reference signal is kept low but the accuracy of each digital reference signal remains high. In converting each digital reference signal into an analog reference quantity additional high order bits are added to the digital reference signal so that each analog reference quantity has the same accuracy as its corresponding digital reference signal.

Patent
21 Mar 1984
TL;DR: In this article, a combined audio and visual signal unit is coupled in parallel on a single cable pair in such a way that only one of the audio or visual components is drawing power from the power supply at any given instant.
Abstract: A combined audio and visual signal unit with the audio and visual components actuated alternately and powered over a single cable pair in such a manner that only one of the audio and visual components is drawing power from the power supply at any given instant. Thus, the power supply is never called upon to provide more energy than that drawn by the one of the components having the greater power requirement. This is particularly advantageous when several combined audio and visual signal units are coupled in parallel on one cable pair. Typically, the signal unit may comprise a horn and a strobe light for a fire alarm signalling system.

Patent
19 Apr 1984
TL;DR: In this article, the authors proposed a method for processing a speech signal with variable gain, where the gain is set to an initial value before processing, and then the optimum gain setting is determined from the histogram of frequencies.
Abstract: For processing a signal applied to an input of audio signal processing devices, for example a speech signal received via a microphone (2) in a speaker-identification or speech recognition system, the signal must have a specific level. For this purpose the signal is applied via an amplifier (4) with variable gain. Before processing begins, the gain is set to an initial value. During processing of the speech signal, it is checked, at regular time intervals, in which range of a plurality of predetermined amplitude ranges the amplified signal is situated, and whether overloading takes place and the frequency which the amplitude of the signal occurs in each of the amplitude ranges is counted. When the signal ends, the optimum gain setting is determined from the histogram of frequencies. In the case of a signal which is specific of an individual speaker and which must be processed in speech recognition or speaker identification systems, this optimum gain setting may be stored in a memory (28) at a location assigned to the speaker, and when the system is subsequently used by the same speaker this value is used as the initial setting of the amplifier gain. In this way the gain is set to an optimum value during the first time that the system is used by a specific speaker and it is corrected continuously when the system is used again by the same speaker.

Patent
15 Oct 1984
TL;DR: In this paper, a multiplier arrangement was proposed to ameliorate a perception problem associated with expectations as to the direction (left or right) from which the audio signal comes when viewing a receiver showing a narrow-field-of-view picture.
Abstract: A stereophonic television signal processing arrangement includes a television camera having a zoom lens capable of wide and narrow viewing angles. A sensor coupled to the zoom lens generates a control signal indicative of the current field of view. An audio signal generator such as a pair of microphones and a matrix generates L+R and L-R audio signals. In order to ameliorate a perception problem associated with expectations as to the direction (left or right) from which the audio signal comes when viewing a receiver showing a narrow-field-of-view picture, a multiplier arrangement multiplies the L+R and L-R signals by factors K 1 and K 2 , respectively, which are directly and inversely related, respectively, to the control signal indicative of current field of view.

Patent
30 Mar 1984
TL;DR: In this paper, a digital color camera for converting an input optical image to a standard color television signal by using digital signal processing techniques includes an image sensor for converting optical signals to a series of pixel signals, an analog-to-digital converting circuit for producing digital pixel signals; a digital signal detecting circuit for separating the digital pixel signal signals into a digital luminance signal and a digital chrominance difference signal.
Abstract: A digital color camera for converting an input optical image to a standard color television signal by using digital signal processing techniques includes: an image sensor for converting optical signals to a series of pixel signals; an analog to digital converting circuit for producing digital pixel signals; a digital signal detecting circuit for separating the digital pixel signals into a digital luminance signal and a digital chrominance difference signal; a digital chrominance signal processing circuit for producing two independent digital chrominance difference signals; a digital luminance signal correction circuit for making a gamma correction on the digital luminance signal; a color television signal encoding circuit for producing an analog standard color television signal from the two independent digital chrominance difference signals and the gamma corrected luminance signal These circuits are controlled by timing pulses produced by a control timing generating circuit driven by a driving circuit The digital signal detecting circuit includes: a pixel signal separation circuit for separating the digital pixel signals into two digital color pixel signals; a first digital addition circuit for adding the two digital color pixel signals; a pixel signal inverting circuit for inverting one of the two digital color pixel signals, and a second digital addition circuit for adding the other of the two digital color pixel signals with an output signal of the pixel signal inverting circuit

PatentDOI
TL;DR: In this article, a triphonic sound system is proposed for multi-channel television broadcasting, in which three independent stereophonically related audio frequency source signals L, R and C are combined to derive three other audio frequency signals M, S, and T which respectively comprise (L+1.4C+R, (L-R) and (-1. 4C).
Abstract: A triphonic sound system in which three independent stereophonically related audio frequency source signals L, R and C are combined to derive three other audio frequency signals M, S, and T which respectively comprise (L+1.4C+R, (L-R) and (-1.4C). In a preferred transmitter embodiment, useful in television broadcasting, the audio signals S and T modulate two quadrature-related sub-carriers of the same frequency to develop two double-sideband, suppressed-carrier signals, the frequency of the subcarriers being sufficiently high as to assure a frequency gap between the lower sidebands of the modulated subcarrier signals and the audio frequency signal (L+1.4C+R). The aforementioned signals, and a pilot signal having a frequency which lies within the frequency gap, are combined and frequency-modulated onto a high frequency carrier for the purpose of transmitting the same to one or more remote receivers. The receiver is operative in response to reception of the high frequency carrier to reproduce each of the audio frequency source signals L, R and C, and includes means for reproducing conventional monophonic and two-channel stereophonic broadcasts. The described matrix equations are amenable to and useful with multi-channel television sound systems currently under consideration for future broadcast service in the United States.

Patent
02 Jul 1984
TL;DR: In this article, a digital data transmission system comprises a first circuit for transmitting digital data related to an information signal data in terms of sectors each having k words with respect to each channel, where k is a natural number.
Abstract: A digital data transmitting system comprises a first circuit for transmitting digital data related to an information signal data in terms of sectors each having k words with respect to each channel, where k is a natural number. The transmission of the digital data is performed within a predetermined transmission time period once or a plurality of times according to a tolerance of data error rate for an information content of the digital data. The information signal data are among digital data related to a plurality of channels of information signals which are subjected to a digital modulation, and the plurality of channels of information signals are made up of a combination at least including the information signal data among three kinds of information signals. The three kinds of information signals are a non-compressed audio signal, a video signal, and the information signal data other than the non-compressed audio signal and the video signal. The system also comprises a second circuit for adding a code and a synchronizing signal to the digital data which are obtained from the first circuit, and for transmitting the digital data having the first code and the first synchronizing signal added thereto. The first code is at least indicative of a number of times the digital data are repeatedly transmitted within the predetermined transmission time period.

Patent
28 Sep 1984
TL;DR: In this article, a method and apparatus for digitizing audio signals being generated from a plurality of parties (A, B) in order to provide audio communication between the parties with a minimum of interference is presented.
Abstract: A method and apparatus for digitizing audio signals being generated from a plurality of parties (A, B) in order to provide audio communication between the parties with a minimum of interference. Each of the parties has an audio transmitter and receiver (38, 58) provided for transmitting and receiving audio signals. The audio signals are converted (40) to a PCM companded eight-bit parallel signal followed by conversion (42) to a serial signal for transmitting to a remote location and then reconverting (46) each of the companded serial signals to a first eight-bit parallel signal. A pre-programmed ROM (48) is located at the remote location and has a plurality of inputs and outputs. Each of the eight-bit parallel signals from the individual parties are fed to the ROM causing the ROM to produce an eight-bit signal on the output representing the instantaneous sum of the eight-bit parallel signals being supplied to the inputs thereof. The eight-bit parallel signal appearing on the output of the ROM is converted to a serial-digital signal by a parallel-to-serial converter (50) for transmitting over a single line (52) to another serial-to-parallel converter (54) prior to being fed through a digital-to-analog converter to the receiver (58) for producing the audio signal at the receiver.