Showing papers on "Audio signal processing published in 1990"

Analog & digital signal processing

Abstract: The Fourier series in spectral analysis and function approximation the Fourier transformation and generalized signals the Laplace transformation and some of its applications analogue signal processing systems digitization of analogue signals discrete signals and systems design of digital filters the fast fourier transform and its applications stochastic signals and power spectra finite word-length effects in digital signal processors linear estimation and adaptive filtering.

System transmitting and receiving digital and analog information for use in locating concealed conductors

Abstract: An improved locator system is provided which transmits digital and analog information over an electromagnetic field that is radiating from an underground source. The locator system includes a transmitter that modulates its output signal by turning its output signal on and off in response to a sequence of digital words. The transmitter also modulates an analog signal onto the transmitted signal. The modulated analog signal can include an audio frequency or voice signal. The improved locator system also includes a receiver that senses and detects the transmitted digital or analog signal impressed on the radiated electromagnetic field. The receiver also demodulates the transmitted audio frequency signal and provides an audio output signal to an operator.

Multiple clock rate teleconferencing network

Abstract: A hub-resident data switching apparatus for a multinode teleconferencing network enables plural node sites, operating at differing clock rates, to communicate with each other asynchronously and simultaneously. The switching apparatus includes an internal TDM bus to which plural node interface units, an audio combiner unit and a timing and control unit are coupled. Each node interface unit is associated with a respective node and is operative to receive and forward communication signals with respect to its node at the clock rate of the service used by that node. The hub's TDM bus includes a video/PC file bus and an audio/command bus and operates at a clock rate that is a multiple of the number of nodes in the network times the highest clock frequency of any node in the network divided by the bit width of the TDM bus. Each unit interfaces with the TDM bus by way of a pipeline bus interface unit. The audio combiner unit outputs digitized `combined audio` signal packets onto the audio bus for transmission to its associated node. The output section of the audio combiner unit includes an audio packet input buffer whose contents are processed in dependence upon the occupancy status of each audio packet input buffer in the combiner. As long as there is sufficient audio data stored in one or more buffers to ensure effectively continuous audio processing, audio data is processed and forwarded for transmission. If the audio packet input buffers have very little or no audio data, then processing is delayed.

Multiplier policies for digital signal processing

Abstract: The successful design of digital signal processing (DSP) systems and subsystems is often predicated on realizing fast multiplication in digital hardware. This tutorial provides the reader with a broad perspective of this important field and the pedagogy needed to understand the basic principles of digital multiplication. Both conventional and nonconventional methods of implementing multiplication, representing a mix of speed/complexity tradeoffs, are presented. Some are based on traditional shift-add structures, whereas others strive for greater mathematical sophistication. Topics include stand-alone fixed-point multipliers, cellular arrays, memory intensive policies, homomorphic systems, and modular arithmetic. >

Analog and Digital Signal Processing

Abstract: From the Publisher: Provides well balanced, detailed coverage of the techniques of signal processing in both the analog and digital domains and the ways in which they are linked in practical applications. Topics include spectral analysis of continuous and discrete signals, analysis of continuous and discrete systems and networks using transform methods, design of analog and digital filters, digitization of analog signals, power spectrum estimation of stochastic signals, the fast Fourier transform algorithms, finite word-length effects in digital signal processors and linear estimation and adaptive filtering.

Programmable hybrid hearing aid with digital signal processing

Abstract: Programmable hybrid hearing aid with digital signal processing comprising a main section (1) which can be inserted in the meatus (6). The main section (1) comprises an open connection between the ear opening and an inner portion of the meatus (6), providing an acoustic transmission channel with low-pass characteristic and resonant amplification. The main section further comprises an electroacoustic transmission channel based on digital signal processing and a signal processor (DSP) and with possibility for suppressing a possible acoustic signal feedback through the acoustic transmission channel. A variant of the hearing aid is provided with a microphone (M1) and the feedback signal is suppressed by digital filtering. Another variant of the hearing aid employs two microphones (M1.M2). and the feedback signal may then be suppressed by phasing out before the digital signal processing, while the digital signal processing also comprises cancellation of the feedback signal in case of high gain. A number of response functions are stored in a memory (RAM2) in a control unit and is freely chosen by the user in regard of adaption to hearing function and acoustic environment. All the electronics of the electroacoustic channel in the hearing aid is implemented as a monolithic integrated circuit (3) in CMOS technology.

Method and apparatus for digitally processing a high definition television augmentation signal

Abstract: A method and apparatus for digital encoding are described for compressing the augmentation channel signals (chrominance and luminance signals for panel information and high frequency luminance and line difference signal) so that this information can be transmitted in a 3 MHz wide RF channel using a digital transmission scheme such as QPSK. Analog signal components are sampled and converted to digital signals. Each of the signals is fed into a separate coder which reduces the number of bits/pixel required to reconstruct the original signal. Compression is achieved by quantization and removal of redundancy. The compression scheme is based on the use of DCT together with VLC. Each augmentation signal has its own coder, which is adapted to the unique statistics of this signal.

Broadcast digital sound processing system

Abstract: A broadcast digital sound processing system includes an ISA (Industry Standard Architecture) bus compatible personal computer with a hard disk drove and a sound processor board installed in an expansion slot of the computer. The board includes a stereo input, analog to digital converter (ADC) and a stereo set of digital to analog converters (DAC's) interfaced to a digital signal processor (DSP) chip. A stereophonic audio signal is converted to digital data by the ADC and communicated to the computer by the DSP chip through a two port record first-in/first-out (FIFO) buffer for storage on the disk. A program is played back by communicating a program data file through a two port playback FIFO buffer to the DSP and from there to the DAC's for reconstruction to a stereo set of analog signals. The reconstructed audio signals may then be used as a modulating signal for radio broadcasting.

Extending the notion of a window system to audio

Abstract: With audio's increasing importance in computer applications, users will soon need presentation, management and organizational capabilities similar to visual window systems to avoid a confusing cacophony of multiple audio sources sounding at once. The ways in which an audio window system could be used are described. These include multimedia documents, spatial data management systems, and teleconferencing. The signal processing methods used to create hierarchical and spatial distribution among nearly arbitrary (not pure sine wave) audio sources are discussed. A prototype system, combining hierarchical and spatial processing functions with a computer-controlled switch, software and human input devices, is presented. Two envisioned implementations, a terminal-based system and a network-based server, are described. Preliminary work suggests that an effective audio window system needs much less complexity and fewer levels of digital signal processing precision than the current prototype. >

Audio signal processing system having independent and distinct data buses for concurrently transferring audio signal data to provide acoustic control

Abstract: An audio signal data processing system comprises input device for sequentially supplying an audio signal data, data memory control device for writing the audio signal data into a data memory and reading-out the data from the data memory, delay memory control device for sequentially reading-out the audio signal data from the data memory and storing the data into a location of a delay memory indicated by a writing address and for reading-out the audio signal data from a location of the delay memory indicated by a reading address and writing the data into the data memory, address designating devices for designating the writing and reading addresses, arithmetic device for multiplying a predetermined coefficient data to the audio signal data having been read-out by the delay memory control device and written into the data memory, and output device for providing the audio signal data in accordance with a result of operation by the arithmetic device. The data memory control device and the delay memory control device write and read the audio signal data into and from the data memory through data buses independent from each other.

Signal processing device and method

Abstract: A signal processing device for generating an output signal corresponding to a multi-dimensional input signal such as a two-dimensional image and a method of processing such a signal. The device includes an array of processing elements which are congruent and shaped so that they can be arranged on a processing element so that adjacent pairs of elements considered as a unit are geometrically similar to each processing element. Output signals of individual processing units are linearly ordered in such a manner as to maintain adjacency of signals from adjacent processing elements in the array. The ordering facilitates one-dimensional Haar transform processing of the signals in such as a manner as to localize signal energy.

Wireless audio and video signal transmitter and receiver system apparatus

Abstract: A wireless audio and video signal transmitter and receiver system apparatus capable of accepting both external baseband audio and video signal inputs and a modulated external audio/video signal modulated at a particular television channel. A transmitter transmits an audio/video signal to one or more remote receivers which regenerate the original audio/video signal providing both baseband audio and video signal outputs as well as a modulated audio/video signal output at the particular television channel. The transmitter includes a modulated audio/video signal input section and baseband audio and video signal input sections which provide audio sub-carrier and video baseband signals to a video/RF switch which permits the user to select between modulated and baseband inputs to the transmitter. The audio/video signal is AM modulated and up converted before being transmitted to the receiver. The receiver down converts the received audio/video signal and through which a modulated audio/video signal output section connected to baseband audio and video signal output sections serves to provide both modulated and baseband audio/video signal outputs.

Integrated digital signal processor/general purpose CPU with shared internal memory

Abstract: An integrated data processing platform for processing a digital signal that includes a general purpose processor and a digital signal processor (DSP) module. The DSP module recovers digital data from a digital signal utilizing a sequence of DSP operations selected by the general purpose processor. The general purpose processor processes the digital data recovered by the DSP module, but is also available to perform general purpose tasks. A shared internal memory array selectively provides information to the DSP module and to the general purpose processor. The information stored in the internal memory array includes operands utilized in the execution of the DSP algorithm and selected instructions and data utilized by the general purpose CPU either for controlling the execution of the DSP algorithm or for executing its own general purpose tasks. While in many applications the data processing system will include an analog front end that converts a modulated input signal received on an analog transmission channel to a corresponding digital signal for processing by the data processing system, the data processing system may also receive the digital signal directly from a digital source.

Data communication apparatus which can use either an analog or a digital line

Abstract: A data communication apparatus includes an analog communication unit for performing analog communication, an A/D converter for converting the analog signal from the analog communication unit into a digital voice signal, and an ISDN communication control unit for sending the digital voice signal from the A/D converter onto a digital communication line. Data used in an ISDN are voice data, image data, facsimile communication data, etc.

Analog-to-digital converting unit with broad dynamic range

Abstract: An analog-to-digital converting unit comprises an analog level varying circuit having a plurality of gains and varying a magnitude of an analog input signal for producing a plurality of analog output signals different in magnitude from one another with the respective gains, a plurality of analog-to-digital converting circuits respectively supplied with the analog output signals and producing a plurality of digital code signals, respectively, a controlling circuit supplied with two of the digital code signals and calculating a difference therebetween for producing a first control signal indicative of varying one of the digital code signals in value so as to be equal in value to the other digital code signal, a calculating circuit responsive to the first control signal and causing one of the digital code signals to be varied in value for producing a candidate of a digital output signal, and an output circuit coupled to one of the analog-to-digital converting circuits and to the calculating circuit and supplying the digital output signal to the output node. Since the value of the input analog signal is retrieved from the digital code signal on the basis of the difference, the analog-to-digital unit is free from any secular changes and any variation of operational ambient.

Apparatus for recording an audio signal with protection against re-recording thereof, and recording apparatus which indicates the presence of such protection

Abstract: Low-frequency components of the left and right channels of the audio signal to be recorded are derived by means of filters (3, 4). First and second auxiliary signals are added to such low-frequency components in such a way that during reproduction of the audio signal so recorded, including the first and the second auxiliary signals added thereto, by means of two loudspeaker units arranged in a stereo configuration, the two auxiliary signals are not audible. However, they will inhibit re-recording by apparatus for recording audio signals which are not copy-protected in the form of the aforesaid two auxiliary signals, or will cause such apparatus to indicate that the audio signal to be re-recorded is protected.

Holter ECG report generating system

Abstract: A system used by a technician for evaluating a Holter ECG tape having a signal representing a series of waveforms thereon and for generating a report reflecting the evaluation. A tape reader generates from the ECG tape an analog signal representative of the ECG waveforms recorded on the tape. An A/D converter converts the analog signal into a digital signal. A computer including a data bus, a memory, a processor controls the operation of the system. A display displays waveforms representing the ECG. A storage device is connected to the data bus. A direct memory access device moves the digital signals from the converter to the memory for storing the digital signal in the storage device. The converter provides at least a 12 bit a digital signal corresponding to the analog signal and includes means for converting the 12 bit digital signal to an 8 bit digital signal.

High-quality coding of telephone speech and wideband audio

Abstract: Digital speech technology is reviewed, with the emphasis on applications demanding high-quality reproduction of the speech signal. Examples of such applications are network telephony, ISDN terminals for audio teleconferencing, and systems for the storage of audio signals, which include the important subclass of wideband speech. Depending on the application, the bandwidth of input speech can vary from about 3 kHz to nearly 20 kHz. Coding for digital telephony at 4 and 8 kb/s, network quality coding at 16 kb/s, and coding for audio at 7 and 20 kHz are examined. Future directions in the field are discussed with respect to anticipated technology applications and the algorithms needed to support these technologies. >

Transform coding of audio signals at 64 kbit/s

Abstract: The coding of high-quality sound at 64 kb/s is of interest for applications such as ISDN. The algorithm described allows the reduction to such a bit rate while maintaining the original quality. It is based on transform coding, and uses a time-domain aliasing cancellation (TDAC) transformation. Perceptual properties and the interblock redundancy of the spectrum are involved when coding the transform coefficients. The complexity of the algorithm allows its real-time implementation on a one floating-point digital signal processor, such as the ATT DSP 32C. The performance and subjective results of the coding system are discussed. >

System for coding and decoding an orthogonally transformed audio signal

Abstract: A system for coding and decoding an audio signal by using an orthogonal and inverse orthogonal transformation of a block units comprises a coding unit having a circuit (10) for obtaining a power level of the audio signal of a segment unit having a predetermined time interval shorter than the block unit, a circuit (11) for generating a gain control signal from the power level, a circuit (12) for performing a predetermined adaptive gain control responsive to the gain control signal to generate and output the adaptive gain control signal to a decoding unit, thereby performing a pre-treatment, and a coding portion (20) for coding the adaptive gain control signal by using the orthogonal transformation to generate and output a coded signal; and the decoding unit having a decoding portion (30) for decoding the coded signal, dequantizing and inversely and orthogonally transforming a decoded audio signal, and a circuit (13) for performing an inverse gain control for the decoded audio signal responsive to the adaptive gain control signal from the adaptive gain control circuit to reproduce and output an audio signal, thereby performing post-treatment.

Digital dialog editor

Abstract: A system is provided for editing dialog in the digital domain. The system generally consists of three sections, a front end, a plurality of audio processor modules and an inputloutput system. The front end section interfaces with the user and provides overall system control. In one embodiment, the user interfaces with the system via AT compatible hardware using proprietary software with a mouse (7) and keyboard (8). A graphic representation of the sound is presented on a high resolution graphics monitor (2). An intelligent machine control processor controls the overall operation of the system. Each audio processor module includes a processor for preforming operations on the associated track of data. A shared memory architecture is preferably used, whereby the audio processor modules are linked by a VME bus. In operation, the analog master recordings are converted to digital by the input system. Each track is stored separately for a audio processor module and disk drive. The user may call up and display segments of a graphic representation of sound on the monitor. The sound may be modified by action of the mouse. The system operates to assemble the edited master from the edit decision list.

Timing audio and visual signals with coincidental markers

Abstract: A method for timing audio and video component signals of a television signal with coincidental markers generates (22-24) a field pulse every Nth field of the television signal at the transmitter. The duration of the field pulse is approximately one video field, and is used to enable an audio tone generator (30) for the one video field and to switch (10) a distinctive video signal, such as a flat video signal, into the video component signal for the one video field. At the receiver the two component signals are input through a synchronizer (16) to a waveform display device (18), such as an oscilloscope, to observe the time differential between the two markers in the audio and video component signals. The audio component signal is delayed by the synchronizer until the markers are again in time coincidence as at the transmitter.

Audio scrambling system using in-band carrier

Abstract: Audio signals are scrambled by single side band modulating with a modulation signal carrier having a frequency lying within the audio frequency band so that the signals are frequency translated upward. The scrambled audio signals are descrambled after broadcasting or recording/reproducing using a substantially identical modulation carrier signal to restore the original audio signals. Security is enhanced by varying the frequency of the modulation carrier signal in a pseudo random manner in response to start (ACLK) and rate (A0, A1) control signals. The control signals accompany the scrambled audio signals and are used during the descrambling process to aid in the generation of the descrambling carrier modulation signal. The frequency translation technique reduces the adverse effect of wow and flutter frequency fluctuations introduced during recording and reproduction of the audio signals (as compared to systems using frequency spectrum inversion), and also reduces the adverse affect of high frequency headroom crashing experienced in pre-emphasis broadcasting and equalizer recording applications.

Audio signal data processing system

Abstract: An audio signal data processing system comprises input device (1) for sequentially supplying an audio signal data, a data memory control device (31-33) for writing the audio signal data into a data memory (5,6) and reading-out the data from the data memory, a delay memory control device (35) for sequentially reading-out the audio signal data from the data memory and storing the data into a location of a delay memory (15) indicated by a writing address and for reading-out the audio signal data from a location of the delay memory indicated by a reading address and writing the data into the data memory, address designating devices (35) for designating the writing and reading addresses, an arithmetic device (8,11) for multiplying a predetermined coefficient data to the audio signal data having been read-out by the delay memory control device and written into the data memory, and an output device (37) for providing the audio signal data in accordance with a result of operation by the arithmetic device. The data memory control device and the delay memory control device write and read the audio signal data into and from the data memory through data buses (4,14) independent from each other.

Repeater of digital audio interface signal

Abstract: A digital audio interface signal repeater capable of displaying how many generations a further copying of a digital audio interface signal is allowed includes: a receiving device for receiving a digital audio interface signal, a demodulator for demodulating predetermined information included in the received digital audio interface signal, and an information display device for displaying the demodulated information The predetermined information included in the digital audio interface signal received by the receiving device is demodulated by the demodulator The demodulated information is displayed by the information display device Since the digital audio interface signal includes information regarding the allowed number of subsequent copying, necessary information such as the allowed number of copying of the digital audio interface signal being processed can be known by confirming a display on the information display device

Apparatus for recording on a disk an audio signal that is recorded after the recording of a video signal thereon

Abstract: Apparatus for recording a still video picture signal and an audio signal on a disk, comprising an after-recording switch manually operable to initiate the recording of an audio signal on the disk, and a controller which responds to the operation of the after-recording switch to control the recording of the audio signal.

Apparatus for improved recording of signals having video and audio components

Abstract: A signal recording apparatus for recording a TV signal in which an audio signal such as a MUSE signal and a video signal are multiplexed in a time sharing manner in units of a given period of time, is arranged to code these video and audio signals alike, to dispersively allocate a discretely separated audio signal within the video signal, to provide symbols other than the symbols of the audio and video signals within each synchronizing block for the purpose of permitting the use of other video signal recording apparatuses. The arrangement permits simplification and high-quality of the apparatus for recording the signals of this kind and thus contributes to the popularization of the apparatus.

High quality coding of telephone speech and wideband audio

Abstract: Advances in signal processing which have created several new technologies for high-quality digital audio are discussed. For traditional telephony, characterized by a signal bandwidth of about 3.2 kHz, the transmission rate for network quality speech is now down to 16 kb/s, with the prospect of a new standard from the CCITT. Robust communications quality appropriate for cellular radio has been realized at 8 kb/s. The current focus is on 4 kb/s, with the aim of improving naturalness and speaker identification. In the coding of wideband audio, an important point of reference is the CCITT standard for 7 kHz speech at a rate of 64 kb/s. Results of recent research are pointing to better capabilities-higher signal bandwidth at 64 kb/s, and 7 kHz bandwidth at lower bit rates such as 32 kb/s. The coding of audio with a signal bandwidth of 20 kHz is receiving significant attention due to recent activity in the ISO (International Standards Organization), with a goal of storing a CD-grade monophonic audio channel at a bit rate not exceeding 128 kb/s. Prospects for accomplishing this are found to be very good. As a side result, emerging algorithms will offer very attractive options at lower rates such as 96 and 64 kb/s. >

Process for generating an entertainment audio signal interrupted by advertising audio signals and device for carrying out the process

Abstract: A process for generating an entertainment audio signal interrupted by advertising audio signals in which pilot signals associated with the advertising audio signals cause switching between the entertainment audio signals and the advertising audio signals makes it possible to broadcast different radio programs of the desired acoustic quality in a shop by means of a carrier frequency. To this end, a high-frequency control signal is generated from advertising audio signals and pilot signals which precede or follow the advertising audio signals. The control signal and preferably the entertainment audio signal are modulated to a high-frequency carrier signal, possibly with several channels, for the separate recovery of the signals. The modulated carrier signal is transmitted, the transmitted carrier signal is demodulated and, if necessary, the signals are separated from each other. The pilot signals of the transmitted control signals are monitored for the presence of fading or flashback conditions. If fading conditions are present, an advertising audio signal forming part of the control signal which follows the pilot signal is faded over the supplied entertainment audio signal and the resultant signal is reproduced. If flashback conditions are present or if the advertising audio signal is interrupted for a certain time interval, the entertainment audio signal is faded in and the resultant signal is reproduced.