Showing papers on "Audio signal processing published in 1988"

Introduction to Digital Signal Processing

Abstract: An Introduction to Digital Signal Processing is written for those who need to understand and use digital signal processing and yet do not wish to wade through a multi-semester course sequence. Using only calculus-level mathematics, this book progresses rapidly through the fundamentals to advanced topics such as iterative least squares design of IIR filters, inverse filters, power spectral estimation, and multidimensional applications--all in one concise volume.

Digital signal encoding and decoding apparatus

Abstract: Digital data is conveyed along with the analog signal by selectively quantizing the analog signal in response to the level of each of the digital bits to be sent. By determining which quantization function was used, a decoder may recover the embedded digital data.

Adaptive, programmable signal processing and filtering for hearing aids

Abstract: A hearing aid system utilizes digital signal processing to correct for the hearing deficit of a particular user and to maximize the intelligibility of the desired audio signal relative to noise. An analog signal from a microphone is converted to digital data which is operated on by a digital signal processor, with the output of the digital signal processor being converted back to an analog signal which is amplified and provided to the user. The digital signal processor includes a time varying spectral filter having filter coefficients which can be varied on a quasi-real time basis to spectrally shape the signal to match the hearing deficit of the user and to accommodate ambient signal and noise levels. The coefficients of the spectral filter are determined by estimating the energy in several frequency bands within the frequency range of the input signal, and using those energy estimates to calculate desired gains for the frequency bands and corresponding spectral filter coefficients. The spectral energy analysis may be carried out using pairs of high pass and low pass digital filters in cascade relation, with the output of each low pass filter being provided to the next pair of high pass and low pass filters. The rate at which output data is provided from the filters in each pair may be reduced from the sample rate of input data by one half for succeeding pairs of filters in the cascade to thereby reduce the computation time required.

Adaptive, programmable signal processing hearing aid

Abstract: A hearing aid system utilizing digital signal processing is programmable to fit the hearing deficit of a particular user and adaptive to the sound environment to maximize the intelligibility of the desired audio signal relative to noise. A signal picked up from microphone (30) is amplified and filtered by preemphasis circuit (32) and converted to digital data through an 8-bit linear A-to-D converter (47). The processor (50) preferably performs spectral shaping on the data to match the user's preference and performs a non-linear adaptive amplification function on the digital data. The amplification gain function may include several piecewise linear sections, including a first section providing expansion up to a first knee (K1), a second section providing linear amplification from the first to a second knee (K2), and a third section providing compression for signals above the second knee to reduce the effect of over range signals and minimize loudness discomfort to the user.

A method for indicating the presence of speech in an audio signal

Abstract: A voice operated switch employs digital signal processing techniques to examine audio signal frames having harmonic content to identify voiced phonemes and to determined whether the signal frame contains primarily speech or noise. The method and apparatus employ a multiple-stage, delayed-decision adaptive digital signal processing algorithm implemented through the use of commonly available electronic circuit components. Specifically the method and apparatus comprise a plurality of stages, including (1) a low-pass filter to limit examination of input signals to below about one kHz, (2) a digital center-clipped autocorrelation processor whih recognizes that the presence of periodic components of the input signal below and above a peak-related threshold identifies a frame as containing speech or noise, and (3) a nonlinear filtering processor which includes nonlinear smoothing of the frame-level decisions and incorporates a delay, and further incorporates a forward and backward decision extension at the speech-segment level of several tenths of milliseconds to determine whether adjacent frames are primarily speech or primarily noise.

Wireless remote speaker system

Abstract: A device comprised of separate transmitter and receiver sections, which converts an audio signal into an F.M. signal, transmits the F.M. signal over A.C. power lines, and reconverts the F.M. signal into an audio signal, which can then be outputted to loudspeakers or other devices. By transmitting the signal over A.C. lines, no long wires are needed to connect remote loudspeakers or other devices to the source of the audio signal.

Audio frequency based market survey method

Abstract: A method for obtaining audience preference market survey data, such as a radio and/or television listening audience survey and/or supplemental data, such as bar coded data (156), from a plurality of diverse locations for accumulative processing by a remote data processor, involves recording (22, 30, 40, 42, 44, 56, 54, 52) a plurality of audio signals (46, 48, 50) at each of the diverse locations which corresponds to the ambient radio and/or television audio sound at predetermined synchronized discrete sampling times (42, 60, 64, 66, 62) or windows which are synchronized to a master recording (110) of the programs being surveyed. The sampling windows are of short duration with respect to the measurement interval. The master recording (110) audio signals frequency intervals are matched against the frequency of the diverse location audio samples to provide an indication of audience preference and tested for a correct match in a configurable filter array (120, 122, 124). Respondents at the diverse locations may be provided with portable tape recorders (30) which are automatically activated at synchronized clock times to obtain the audio samples. Bar code scanning information (150, 24) may also be provided in the form of audio signals by using the scanning signal (152) to drive a voltage controlled audio oscillator (160).

Simultaneous audio and video transmission with restricted bandwidth

Abstract: Method for simultaneously sending audio and video signals over standard telephone lines or other channels having a restricted bandwidth which comprises obtaining a video image, digitizing the image, modulating a signal with the digitized image, obtaining audio signals and filtering the audio signals to a frequency range of the band outside that of the modulated video signal, combining the filtered audio signals and the video signal, and transmitting such signals through the restricted bandwidth channel, together with a method for receiving such signals and apparatus for sending and receiving such signals.

Method for transmitting an audio signal and an auxiliary signal

Abstract: With an audio signal (T), an auxiliary signal (Z) is intended to be transmitted which cannot be separated from the audio signal (T), is inaudible on reproduction and does not impair sound reproduction quality. The auxiliary signal (Z) is transmitted using psycho-acoustic masking effects with a level which is lower than the respective level of the audio signal (T). In particular for the transmission of a code signal for copy protection in recordings with CD players or DAT recorders.

Telecommunication system with video and audio frames

Abstract: An interactive system transfers video and audio information from a central facility to terminals by means of addressed television frames. Each audio frame includes information representing a single channel of time compressed audio. The terminals store correctly addressed frames. Video frames are repeatedly replayed to provide series of still-frame images. The audio frame is used to generate a normal speed audio output.

Feedback suppression in digital signal processing hearing aids

Abstract: Acoustic feedback in digital signal processing hearing aids is suppressed by using signal processing techniques in the digital processor. A first processing technique causes the data to the main signal processing path in the digital signal processor to be delayed by varying amounts over time, preferably in a periodic manner, to disrupt the buildup of feedback resonances. In a second technique, a digital filter receives the input data and has its coefficients adjusted so that the output of the filter is substantially an optimal estimate of the current input sample based on past input samples. The output of the filter is then subtracted from the input signal data to provide difference signal data which substantially cancels out the resonant frequencies. In a third technique, the acoustic feedback path from the output to the input of the hearing aid is modeled in the digital signal processor as a delay and a linear filter. The output of the main signal processing path in the digital signal processor is delayed and the delayed data passed through the linear filter, with the output of the filter then being substracted from the input signal data to provide difference signal data which is provided to the main signal processing path. The coefficients of the digital filter in the feedback path are adjusted so that the signal passed through the feedback filter substantially corresponds to of the acoustic feedback signal to thereby cancel the same.

Signal processing the modern approach

Abstract: Signal processing: the modern approach , Signal processing: the modern approach , مرکز فناوری اطلاعات و اطلاع رسانی کشاورزی

Apparatus and method for providing digital audio on the sound carrier of a standard television signal

Abstract: A television transmission system replaces the standard FM audio portion of a television signal with digital audio. Three digital audio channels are time division multiplexed on the sound carrier, using combined multi-phase and AM modulation. The audio signals are digitized using adaptive delta modulation techniques. Video vertical and horizontal framing, as well as the audio carrier phase reference, audio data bit time and frame reference, and various control data is carried using AM modulation. The digital audio information is carried using multi-phase modulation. The composite data stream may be serially encrypted to provide security and prevent unauthorized reproduction of the video and/or audio portions of the television signal.

Digital sound data storing device

Abstract: This invention relates to a digital sound data storing device wherein analog sound data is synthesized from digital sound data. The device includes a case and an IC card detachably attached to the case. A plurality of digitized sound data groups are stored at a built-in memory of the IC card. The digitized sound data groups are obtained by recording sentences for foreign language conversation practice through a microphone to thereby obtain analog sound signal which is converted to digital sound signal. When the IC card is attached to the case, a signal processing circuit operates to read out the digital sound data group, which is converted to the analog sound signal. The analog sound signal is supplied to an amplifying circuit and reproduced as sound.

Telecommunication system with burst and continuous audio signals

Abstract: A telecommunication system (10) for transmitting video and plural audio information on a transmission medium (13) includes a facility (11) and an addressable terminal (12). The facility (11) includes arrangements for providing addressed video frames containing information corresponding to still images and for providing addressed audio frames containing information corresponding to audio bursts. There is also means for providing non-addressed signals containing information corresponding to a least one channel of continuous audio. The addressed video frames, addressed audio frames, and non-addressed signals are coupled to the transmission medium (13) and sent to the terminal (12). The addressable terminal (12) includes means for detecting said addressed video frames, addressed audio frames, and non-addressed signals on said transmission medium (13). Circuits select video frames and audio frames in response to their corresponding addresses. Arrangements are made for storing still images contained by selected video frames and for storing and playing audio bursts contained by selected audio frames. Circuits provide continuous audio signals contained by said non-addressed signals. A circuit attenuates the continuous audio signals in response to playing of an audio burst.

Method and apparatus for after-recording sound on a medium having pre-recorded video thereon

Abstract: A recording apparatus for recording audio signals and video signals on a recording medium in a mixed form is disclosed. The recording apparatus has a device for reproducing video signals which correspond to audio signals to be recorded prior to the execution of recording of the audio signals. Therefore, an operator can perform sound recording while confirming a correspondence between an image and the sound, since the image associated with the sound is reproduced as a visual image.

Method and apparatus for producing an audio magnetic tape recording at high speed from a preselected music library

Abstract: The present invention relates to a novel method and apparatus for taking an analog audio signal, converting the signal into a digital audio signal, and thereafter converting the digital audio signal into a standard bandwidth video signal and thereby substantially compressing it In this format, it can be placed onto a master video tape from which it can be transferred to a video disk The present invention further comprises a method and apparatus from which a selected number of recordings on the video disk can be retrieved by playing the video disc in still frame mode to create an analog video signal, converting the analog video signal into a string of digital values, converting the digital audio stream into an analog audio signal and thereafter recording the audio signal onto an audio tape at high speed This method and apparatus is used as the central technology in a consumer electronic music center wherein a consumer can randomly select a given number of song selected from a music library and thereafter create his or her own customized audio tape in a matter of minutes

Tone visualizing apparatus

Abstract: A tone visualizing apparatus for visualizing an inputted audio signal to thereby display an image corresponding to this audio signal includes at least a detector, image display (such as a CRT display unit) and display controller. The detector detects characteristics of the audio signal such as envelope, chord, spectrum signal components, number of zero-cross points and energy of the audio signal. The image display displays an image based on given image information which can be generated from a video tape recorder (VTR), a video disk unit or an image memory constituted by a semiconductor memory. The display controller controls the image display so that a display parameter of the image will be controlled based on the detected characteristics of the audio signal. For example, the display parameter can be set as size, brightness or colors of the image. Thus, the image is controlled so that the impression of the image will be matched with that of the audio tone.

Digital signal processing system

Abstract: A digital signal processing system for use as a digital service unit within a communication switching system, comprised of a plurality of digital signal processing (DSP) modules for connection via a dedicated programmable digital switch forming part of a circuit switch matrix, to one or more input/output ports such as line circuits, trunk circuits, etc., under control of a main system controller, such as a microprocessor. Applications programs for implementing predetermined service features, are downloaded from the main controller via the circuit switch matrix and digital switch, to one or more of the digital signal processing modules for storage within internal memories thereof. The main controller dynamically allocates circuit switch and message channels of the programmable digital switch in accordance with the signal bandwidth and computation power required to implement the predetermined service features. Thus, an extremely high signal bandwidth efficiency is obtained for performing various service features such as tone generation and detection. DTMF tone detection, digital conferencing, speech synthesis, etc., utilizing simple, inexpensive, time-shared modules.

Video recording camera

Abstract: A hand-held video recording camera includes a lens and an image pickup operatively arranged to receive image information from the lens for converting the image information into an electric image data output signal. A microphonic arrangement in or on the camera is operatively arranged to receive sound from a subject or subjects within view of the lens for producing audio signals corresponding to the sound received. Means are provided within the camera for recording signals representing the image data and audio signals. A source of further audio signals is provided within the hand-held camera. Audio signal mixing means within the hand-held camera mix the audio signals corresponding to the sound received by the microphone arrangement and to the further audio signals from the internal source. The audio mixing means has its output coupled to the recording means so that signals representing the audio signals can be recorded, as well as signals representing image data. Reproducing means carried by or within the hand-held recording camera and responsive to the further audio signals from the source of further audio signals within the hand-held camera projects sound corresponding to the further audio signals toward a subject or subjects being recorded.

Finite impulse response (fir) filter using a plurality of cascaded digital signal processors (dsps)

Abstract: An FIR digital filter device of the symmetrical coefficient type or the antisymmetrical coefficient type or which processes digital data signals such as digital audio signals. In the FIR digital filter device, a digital input data is transferred in the forward direction through a plurality of digital signal processors (F, B) connected in cascade and is then transferred in the reverse direction to form M pieces of delayed digital data. The digital signal processor in each stage effects the algebraic addition of said delayed digital data and the digital input data through a first operation unit (46), effects the multiplication result through a multiplier (47), adds up the multiplied result through a second operation unit (37), and transfers the obtained data as carry-over data to the next stage.

Method and apparatus for programmed audio annotation

Abstract: An apparatus for programmed audio annotation comprises a user's computer system having a display and a modem associated therewith, and a host computer system having an audio synthesizer unit and a memory unit associated therewith. The modem of the user's system and the audio synthesizer unit of the host system are coupled telephonically. An application program is executed on the user's computer system. The application program generates a sequence of screens for display to the user. Associated with each such screen is a digital word that is provided as part of a command to the modem. In response to such command, the modem transmits a dual tone multiple frequency (DTMF) signal which is received by the audio synthesizer unit of the host computer system. The memory unit of the host computer system contains digitized audio data corresponding to audio messages associated with each of the display screens of the application program. The audio synthesizer unit receives the DTMF signal and converts it to a digital address specifying a location in the memory unit. The specified digitized audio data is retrieved and is provided to the audio synthesizer unit which then synthesizes an audio signal. This audio signal is transmitted telephonically to the modem in the user's computer system where the signal is converted to an audible sound by the modem's internal speaker. Accordingly, the user hears the audible sound associated with a visual screen substantially concurrently with the display thereof.

Method and apparatus for maintaining audio/ video synchronism in a television signal read-out from a digital buffer memory by a reference signal

Abstract: Both the analog and the video component of a television signal are converted to digital form, the audio component being converted at a much lower sample rate with a word length which is an integral multiple of the video word length. The digital audio component is then compressed by use of a temporary memory and read-out therefrom at a rate that is half of the sample rate of the digtal video component, after which the word length is halved and the word rate doubled in a shift register circuit. In that form the audio component is inserted in horizontal blanking intervals of the video component in a compatible form by a multiplexer, the output of which is read into a single picture field or full picture memory under control of an address generator clocked in synchronism with the incoming video component. The buffer memory is read-out at a rate controlled by a reference signal such as is used for synchronism in a television studio. On the output side the digital video and audio components of the television signal are separated by a demultiplexer, the audio word length is doubled and the word rate halved and the still compressed audio signal is expanded, to make available a substantially continuous digital component. The audio component can then be converted to analog form and will be correctly timed for accompanying the digital video component converted to analog form at the output of the demultiplexer. Separate audio signal delay circuits are thus avoided by the use of relatively simple multiplexing circuits and audio processing circuits.

Endoscope signal processing apparatus using sequential and synchronous imaging devices

Abstract: The endoscope signal processing apparatus is provided with a first signal processing device processing the signal for a field sequential type imaging device and a second signal processing device processing the signal for a synchronous type imaging device. One signal processing device has a circuit partly used in common with the other signal processing device. Both signal processing device have a common video signal output end from which video signals from both signal processing devices are output. This apparatus is further provided with a discriminating device discriminating the imaging system of the endoscope to be connected and an output switching device whereby the video signal transmitted through the video signal processing device corresponding to the imaging system discriminated by this discriminating device is led selectively to the common video signal output.

Circuit for processing a digital signal having a blanking interval

Abstract: A digital signal processing circuit for carrying out a series of processings for a digital signal having a signal effective interval and blanking interval such as a video signal. Operation control data added to the blanking interval of the input digital signal controls the signal processings of a plurality of signal processing blocks, so that the circuit construction can be simplified and complicated signal processings can be achieved.

Digital signal mixing apparatus

Abstract: A digital signal mixing device comprises a plurality of input channels (29, 30, 31, 32, 33) each having a respective signal input port (34, 35, 36, 37, 38) for analogue signals, an interface circuit (49) incorporating an analogue-to-digital converter, digital signal processing means (51, 66, 52, 50) for conditioning the digital signal to effect for example volume control, tone control and introduce other musical effects, and summing means (55) interconnecting the individual channel with next adjacent channels upstream and downstream thereof in a sequence such that mixing of the signals is effected by successive addition of the conditioned digital signals produced by an input channel to a signal representing the addition of the output signals from all input channels earlier in the sequence.

Digital signal processor for providing for timbral change in arbitrary audio signals

Abstract: An audio signal processor in which the harmonic content of the output signal (Figure 3g) varies with the amplitude of the input signal (Figure 3a). The preferred embodiment includes an analog to digital converter (102), a sample and hold circuit (101), timing circuits (106), a RAM look-up table (103) for performing non-linear transformation, a digital to analog converter (104) and a post filter (105) from which processed analog audio is output.

Acoustic processing apparatus

Abstract: An apparatus and method of recording and processing acoustics, such as body sounds, for assessment of the sound and possible diagnosis of any abnormalities associated with the sound. The apparatus includes an electronic stethoscope for acoustic pick-up and a convertor section for converting the acoustic analog signal into a digital signal. The operator can track the acoustic signal on an acoustic headset. The digital signal is continuously stored in computer memory and the operator can selectably retain a portion of the digital signal in volatile memory. The apparatus further includes a signal editing function for selectably altering the retained digital signal to isolate the waveform of interest. To this end, the retained signal is fed through the conversion section for conversion to an output analog signal for display on the monitor and play on the acoustic headset. The retained signal can be further edited as desired and stored in nonvolatile memory. The edited waveform is not only useful in diagnosing abnormalities but is also easily preserved for historical interest.

Analog to digital conversion system utilizing dither

Abstract: A system suitable for digitizing an analog audio signal, with the addition of analog dither to the audio signal and, after the digitization of the resulting analog audio/dither signal, the subtraction of digital dither from the digital audio/dither signal. For the production of the analog and digital dither signals, a digital dither generator is employed which generates digital dither of, e.g., 16 bits. The analog dither signal is derived from this 16 bits dither by first translating the same into analog dither of equivalent magnitude and then by reducing its magnitude to a value equivalent to e.g., 10 bits. The digital dither signal, on the other hand, is provided by reducing the magnitude of the 16 bits digital dither to that of substantially 10 bits. The analog dither signal is added to the audio signal, and the resulting analog audio/dither signal is translated by an analog to digital converter of, e.g., 16 bits into a digital audio/dither signal. Then the substantially 10 bits digital dither signal is subtracted from the digital audio/dither signal. A limiter may be connected to the output of the subtracter for limiting the output level when the analog to digital converter is overloaded.