Showing papers on "Audio signal processing published in 1996"

Digital wireless speaker system

Abstract: This invention discloses a digital wireless speaker system for use in consumer audio applications. A digital radio frequency transmitter is connected to an analog or digital audio source and a digital radio frequency receiver provides for reception of the transmitted audio information in remote locations. In addition, the digital receiver will be able to receive control information to implement such things as volume, tone controls, or other auxiliary information. This allows the user to listen to high quality audio in a variety of locations without the need of independent stereos or external wires. The system is based on digital circuitry to improve the performance of the system and provide for compact disc quality sound. The digital circuitry incorporates forward error correction techniques and interleaving to enable the system to account for errors in transmission and thus improve the overall performance of the system.

System for transmitting desired digital video or audio signals

Abstract: A method for transferring desired digital video or digital audio signals. The method comprises the steps of forming a connection through telecommunications lines between a first memory of a first party and a second memory of a second party. The first memory has the desired digital video or digital audio signals. Then, there is the step of selling electronically by the first party to the second party through telecommunications lines, the desired digital video or digital audio signals in the first memory. Then, there is the step of transferring the desired digital video or digital audio signals from the first memory of the first party to the second memory of the second party through the telecommunications lines while the second memory is in possession and control of the second party. Additionally, there is a system for transferring digital video or digital audio signals.

Method and apparatus for transporting auxiliary data in audio signals

Abstract: Auxiliary data (x(m)) is transported in a conventional audio signal (s(t)) by hiding the data in the form of colored noise (y(t)). The colored noise has a spread spectrum that simulates the spectrum of the primary audio signal. The data to be transported is first converted to a spread spectrum signal (p(n)). The primary audio signal is analyzed to determine its spectral shape (88). The same spectral shape is imparted to the spread spectrum signal (94), which is then combined with the primary audio signal for transmission (100). The spectral shaping can be performed using time domain modeling and synthesis such as linear predictive coding (88) or by using subband coding techniques such as fast Fourier transforms. A plurality of different auxiliary information streams can be transported on the audio signal. By adjusting the gain of individual spread spectrum signal carrier(s) and the power of the colored noise (98), the auxiliary information stream(s) can be rendered inaudible in the primary audio signal, or at any desired level below or above an audible threshold.

System for assigning multichannel audio signals to independent wireless audio output devices

Abstract: A home entertainment and information system is provided which assigns and transmits audio programming to audio output devices. Digital and analog signals from a variety of program sources are received by the home entertainment and information system. The system assigns and transmits to an audio output device a program that is distinct from programs assigned and transmitted to other audio output devices within the same system, and thus where two users are viewing different programs visually displayed on the same or different monitors, they hear the audio portion of the respective program they are viewing through individual audio output devices. An audio output device is also assignable to a remote control device such that when a program is selected using the remote control device, the audio portion of the program is transmitted to the assigned audio output device.

Common receive module for a programmable digital radio

Abstract: A digital submodule is included in a software programmable common receive module for receiving intermediate frequency signals and producing a serial bit stream. The digital submodule is programmable based on a selected application of a plurality of radio applications and, if present, a selected function of a plurality of functions of the selected radio application. The digital submodule may include an analog to digital converter for converting intermediate frequency signals received from an analog submodule into digital signals. The digital signals are supplied to a programmable signal processing unit which is configured, according to the selected radio application and, if present, the selected function, to perform control functions, processing and analysis of the digital signals and generate output signals. The output signals are then formatted by a formatting unit producing formatted digital signals. The formatted digital signals are then supplied to a system bus. The programmable signal processing unit may include a digital downconverter for selective use depending on the selected application of radio communication, for generating a baseband signal. Additionally, a central processing unit is included to perform further signal processing for selected radio applications.

Directional microphone utilizing spaced-apart omni-directional microphones

Abstract: A sound processing apparatus includes a plurality of microphones spaced apart from each other, each microphone producing electrical signals representative of sound signals incident thereon. The sound processing apparatus also includes a signal processing unit that produces a specific direction sound signal by processing the electrical signals according to a specific sound direction. The plurality of microphones can be positioned about a periphery of a computer display, and adaptive beam forming techniques can be employed to provide a directional input sound signal for use in sound processing.

Multi-scene recording medium and apparatus for reproducing data therefrom

Abstract: A multi-scene recording disk and a data reproducing apparatus which enables a user to select and enjoy one of simultaneously proceeding scenes, and also enables a program editor to edit programs using a new concept and novel fashion. Related program movements, each consisting of program bars, are recorded on the disk 100. The data reproducing apparatus includes a data string processing section 203, a system control section 204, an operator panel 205, a video processing section 206, and audio processing sections 211 and 213. The sections 203 and 204 and the panel 205 select any one of the program movements and switch one movement to another, and select any one of the program bars of the movement selected and switch one bar to another. The information thus selected is supplied to the video processing section 206 and the audio processing sections 211 and 213.

User independent, real-time speech recognition system and method

Abstract: A system and method for identifying the phoneme sound types that are contained within an audio speech signal is disclosed. The system includes a microphone (12) and associated conditioning circuitry (14), for receiving an audio speech signal and converting it to a representative electrical signal. The electrical signal is then sampled and converted to a digital audio signal with a digital-to-analog converter (34). The digital audio signal is input to a programmable digital sound processor (18), which digitally processes the sound so as to extract various time domain and frequency domain sound characteristics. These characteristics are input to a programmable host sound processor (20) which compares the sound characteristics to standard sound data. Based on this comparison, the host sound processor (20) identifies the specific phoneme sounds that are contained within the audio speech signal. The programmable host sound processor (20) further includes linguistic processing program methods to convert the phoneme sounds into English words or other natural language words. These words are input to a host processor (22), which then utilizes the words as either data or commands.

Automatic volume control to compensate for ambient noise variations

Abstract: A system and method for automatically adjusting the volume of an audio system to compensate for variations in ambient noise. The system includes a microphone for monitoring the ambient audio environment which includes output of the audio system plus environmental noise. The system also includes processing circuitry connected to the microphone. The processing circuitry varies the volume of the output of the audio system in proportion to changes in the environmental noise. The processing circuitry comprises the microphone, located to detect the ambient sound in the listening environment, an analog-to-digital converter connected to the output of the microphone, and a digital signal processor connected to the output of the analog-to-digital converter. The output signal of the DSP is an input to the volume control of the audio system.

Analog spread spectrum wireless speaker system

Abstract: An analog spread spectrum wireless speaker system (20) for use in consumer audio applications for providing reliable and high fidelity stereo sound. The system (20) includes a transmitter (22) that accommodates any analog input from a variety of audio devices (26) such as compact disk players, cassette players, AM/FM tuners and transmits this information in the 2.4-2.4835 GHz band to a receiver (24) at a remote location. The receiver (24) is capable of reproducing the audio signal (26) with good frequency and signal-to-noise performance.

Low bit rate high quality audio coding with combined harmonic and wavelet representations

Abstract: We describe a novel high quality audio coding method using adaptive signal representation, based on sinusoidal and wavelet analysis of signals. First, we perform a harmonic analysis of the signal to remove strong periodic structures or tones from the signal. Then we carry out wavelet analysis that are useful in tracking the transients of the signal. These transients are then removed from the wavelet coefficients. The remaining coefficients have broadband noise-like structure. Since this method separates out tones (sinusoids), transients, and broadband noise, we may use tonal, noise, and temporal masking information to individually encode the tones and the wavelet coefficients. Our experiments suggest that this method yields a nominal bit rate of 1 bit/sample for high quality audio compression.

Acquisition and error recovery of audio data carried in a packetized data stream

Abstract: Audio data is processed from a packetized data stream carrying digital television information in a succession of fixed length transport packets. Some of the packets contain a presentation time stamp (PTS) indicative of a time for commencing the output of associated audio data. After the audio data stream has been acquired, the detected audio packets are monitored to locate subsequent PTS's for adjusting the timing at which audio data is output, thereby providing proper lip synchronization with associated video. Errors in the audio data are processed in a manner which attempts to maintain synchronization of the audio data stream while masking the errors. In the event that the synchronization condition cannot be maintained, for example in the presence of errors over more than one audio frame, the audio data stream is reacquired while the audio output is concealed. An error condition is signaled to the audio decoder by altering the audio synchronization word associated with the audio frame in which the error has occurred.

Sound generator synchronized with image display

Abstract: Digital sound source data is stored in a sound source data memory 22. When a first display object (an enemy character, a waterfall, or the like) so defined as to generate a sound is displayed in a three-dimensional manner on a display screen 41 of a television 40, a audio processing unit 12 reads out the corresponding sound source data from the sound source data memory 22, to produce first and second sound source data. The first and second sound source data are converted into analog audio signals by digital-to-analog converters 16a and 16b, and are then fed to left and right speakers 42L and 42R. At this time, the audio processing unit 12 calculates delay time on the basis of a direction to the first display object as viewed from a virtual camera (or a hero character), and changes delay time of the second sound source data from the first sound source data. Further, the audio processing unit 12 individually controls the sound volume levels of the first and second sound source data depending on the distance between the first display object and the virtual camera (or the hero character). Consequently, sounds having a spatial extent corresponding to the change of a three-dimensional image can be respectively generated from the left and right speakers 42L and 42R.

Method and apparatus for echo data hiding in audio signals

Abstract: A method of hiding information in a host audio signal introduces one or more echoes into the signal. The separation in time between the host signal and an echo is associated with the value of a datum embedded in the signal. The identity of the embedded datum is determined by observing the delay between the host signal and the echo.

Dual-mode radiotelephone apparatus for digital or analog modulation

Abstract: A dual-mode radiotelephone capable of operation in analog or digital modes. According to exemplary embodiments, a digital signal processor receives a speech signal to be transmitted in the digital or analog mode and generates In-phase (I) and Quadrature (Q) modulating signals. The I and Q signals are supplied to a quadrature modulator for generating a digitally modulated signal and are supplied to an analog modulator for generating an analog modulated signal.

Multiple audio and video signal providing apparatus

Abstract: A video signal provider multiplexes digital video signals for four channels and provides a signal the bandwidth of which is limited to 6 MHz by RF-modulation. An analog video signal provider provides another signal the bandwidth of which is limited to 6 MHz by RF-modulation. A digital audio signal provider multiplexes digital audio signals for 32 channels and provides another signal the bandwidth of which is limited to 6 MHz by RF-modulation. Such signals are combined and transmitted through a coaxial cable 14 to a controller. In this example, such signal providing apparatus can provide a maximum of 84 channels (=126 MHz (the bandwidth for video signal of the coaxial cable)/6 MHz×4 ch), which number is significantly larger than the 20 channels provided in the prior art.

Multimedia communications with system-dependent adaptive delays

Abstract: A method and system in which lip synchronization is restricted to a single system in a multiple system communication of video data and audio data. In the preferred embodiment, the receiving system synchronizes video and audio signals for presentation. The originating system forms and processes video and audio signals separately. For many systems, the processing involves dissimilar delays. Consequently, when the information is transmitted through a network to the receiving system, the video and audio data is nonsynchronous. The originating system provides delay information that is indicative of the dissimilarity of video and audio processing time at the originating system. The delay information is utilized at the receiving system to determine an adaptive compensation delay for introduction at the appropriate video or audio processing path. By providing a single compensation delay for multi-system communications, the total delay is potentially reduced.

Electronic program guide device

Abstract: An electronic program guide device includes a demultiplexer for filtering a multiplexed bit stream transmitted through a transmission channel with respect to a selected channel and separating the filtered multiplexed bit stream into a video stream and an audio stream; a controlling part coupled to the demultiplexer for extracting electronic program guide information from the multiplexed bit stream and storing the extracted electronic program guide information in a memory; a video processing part coupled to the demultiplexer and the controlling part for decompressing the video stream from the demultiplexer to restore an original image, synthesizing the original image and the electronic program guide information, and displaying the synthesized original image and the electronic program guide information on a screen; an audio processing part coupled to the demultiplexer for decompressing the audio stream from the demultiplexer; a voice source processing part coupled to the controlling part for vocally synthesizing the electronic program guide information received from the controlling part; and a selecting part coupled to the audio processing part and the voice source processing part for selecting an output from the audio processing part and the voice source processing part and sending the output to a speaker.

Am compatible digital waveform frame timing recovery and frame synchronous power measurement

Abstract: In a broadcast system having a complex waveform of digital and analog segments, frame timing is recovered (290) and the frame synchronous power (284) of the digital portion of the composite AM compatible Digital Audio Broadcast (DAB) waveform is measured by examining a portion of the signal (284) The frame timing necessary for proper demodulation is recovered from the received signal (200), and a signal power measurement for scaling (242) is provided (294) The composite AM compatible DAB waveform is produced by a modulation method in which a digital representation of an audio signal is encoded together with an analog amplitude modulated (AM) signal and transmitted simultaneously in the same frequency channel

A Digital Signal Processing Primer: With Applications to Digital Audio and Computer Music

Abstract: a digital signal processing primer with applications to digital signal processing primer with applications to digital signal processing primer with applications to a digital signal processing primer with applications to a digital signal processing primer with applications to digital signal processing the computer laboratory digital signal processing the computer laboratory digital signal processing cems home signal processing ece, rutgers ece 431 digital signal processing lecture notes signals and systems, 1997, 957 pages, alan v. oppenheim toyota repair manual browserfame balboa lite digital system manual towies pharmacotherapy in primary care browserfame tiger products co ltd user manual foserv digital signal processing for audio applications vbou q a revision guide evidence 2013 and 2014 law questions digital audio signal processing ku leuven electrical,electronics,communications,power,precision and advances in silicon carbide processing and applications document about student solutions manual for stewarts repair manual cp99 saosey applications of minicomputers to library and related problems multiple choice questions on obstetric nursing ebook | www elvis presley a biography auzww close to home a book of postcards zaraa computer controlled urban transportation gurka munch und deutschland boscos a sounding brass elect trilogy 3 zaraa a biographical history of the french revolution trupin gurps y2k the countdown to armageddon gurps ser generic new treasury of poetry oilys nv53 service manual saosey hp 7310xi manual subiuk early readers level 2 xcelr birding the front range awandc the heart of parenting raising an emotionally intelligent 1987 crusader engine manual fakof letters from rising pharmacy starsadvice on creating and 2010 nissan pathfinder service repair manual beelo essays on race and empire mdmtv solomon goldbard v empire state mutual life insurance co melancholie der ankunft blwood nissan bluebird manual mdmtv simulation and analysis of audio signal processor ijca yamaha outboard f115 lf115 factory service repair workshop

Dual audio program system

Abstract: Dual program audio apparatus is provided having two or more sources of input audio program signals (34, 35, 36, 37, 38), switching circuitry (41) for selecting two of the input audio signals for amplification, one or more amplifiers (42, 43) for amplifying the selected input audio program signals, and one or more audio speakers (39, 40) for enabling a listener to hear both audio programs simultaneously Volume control circuitry (44, 45) is provided for selecting a higher volume level for a foreground program and a relatively lower volume for a background program and circuitry is provided for selectively interchanging the foreground and background programs either instantaneously or gradually and continuously

Digital alias-free signal processing in the GHz frequency range

Abstract: The advanced Information Technology we call Digital Alias-free Signal Processing (DASP) is discussed in this paper to draw attention to the facts that, first, this technique has already reached a considerable degree of maturity so that it can now be used as a widely applicable Digital Signal Processing (DSP) tool and, second, that it is especially competitive in the area of Microwave and Radio Frequency (RF) signal processing. Its utility arises from its applicability to digital processing of signals at frequencies considerably exceeding half of the mean sampling rate, which traditionally limit classical DSP applications.

Analog/digital system for television services

Abstract: A combined IRD scrambler/modulator is provided in a headend. A service provider transmits a digital program signal in a digital data stream to a headend which distributes analog program signal to at least one subscriber. The headend receives an encoded digital program signal in the digital data stream from the service provider and decrypts the digital program signal therefrom. The digital program signal includes an audio and a video portion. The video portion of the signal is scrambled by a scrambler. The digital audio portion of the digital program signal and scrambled video portion of the digital program signal are converted to an analog program signal and transmitted to one or more subscribers.

Apparatus and method for synchronizing audio and video frames in an MPEG presentation system

Abstract: Synchronization of MPEG audio and video presentations is attained by detecting a lack of synchrony between the data presentations and modifying the audio data stream based on the detected lack of synchrony. Synchrony of the MPEG audio and video presentations is monitored by tracking the amount of data transferred to audio and video interfaces over time. Synchronization of MPEG audio and video presentations is achieved in a system in which audio and video interfaces are controlled by separate and independent timing signal generators by tracking the amount of audio and video data output by the respective audio and video interfaces. The amount of audio and video data output by the interfaces is compared and, as a result of the comparison, the audio data stream is modified to restore synchrony. Alternatively, presentation of the video data stream is modified to achieve synchrony.

Digital spread spectrum GPS navigation receiver

Abstract: Disclosed are digital processing techniques for efficiently receiving, sampling and recovering multiple direct-sequence spread-spectrum signals by multiplexing key signal processing elements across many logical channels. Global Positioning System (GPS) receivers serve to illustrate the technique. They receive many signals, each with independent timing, phase, amplitude, and data modulation, which may arrive on multiple carrier frequencies and at multiple spreading rates. To employ digital processing techniques, the carrier or carriers are reduced to one or more digital sample streams. Digital processing (e.g., products, sums, decisions, etc.) is then used to estimate, or recover, the received signal and its characteristics. When these functions can be partitioned by processing rate, the usual case, the lower-rate functions can be efficiently shared with negligible efficiency loss.

Perceptual audio compression based on loudness uncertainty

Abstract: A new technique for the determination of the masking effect of an audio signal is employed to provide transparent compression of an audio signal at greatly reduced bit rates. The new technique employs the results of recent research into the psycho-physics of noise masking in the human auditory system. This research suggests that noise masking is a function of the uncertainty in loudness as perceived by the brain. Measures of loudness uncertainty are employed to determine the degree to which audio signals are "tone-like" (or "noise-like"). The degree of tone-likeness, referred to as "tonality," is used to determine masking thresholds for use in the compression of audio signals. Tonality, computed in accordance with the present invention, is used in conventional and new arrangements to achieve compression of audio signals.

Audio activity detection circuit to increase battery life in portable computers

Abstract: A power management system for a computer system having an audio circuit which generates an audio output signal. An activity detection circuit senses audio signal activity on the analog audio signal. The power management system maintains power to at least the audio circuit when activity is detected on the analog audio circuit by the activity detection circuit. The power management system causes the audio circuit to enter a power conservation mode when no activity is detected by the audio detection circuit on the audio analog signal for a predetermined amount of time.

MicroUnity's MediaProcessor architecture

Abstract: This media processor extends general-purpose computer systems for communicating and processing digital video, audio, data, and radio frequency signals at broadband rates.

A digital audio broadcasting (DAB) receiver

Abstract: This paper describes the features and functions of the newly developed Eureka-147 DAB (digital audio broadcasting) receiver. In this paper, we also describe an AFC (automatic frequency control) and a synchronization technique using a DSP (digital signal processor).

AM-PSK system for broadcasting a composite analog and digital signal using adaptive M-ary PSK modulation

Abstract: A radio-frequency broadcasting system and method for simultaneously broadcasting analog and digital signals in a standard AM broadcasting channel wherein the system includes: a transmitter for providing, and broadcasting, a composite signal containing an adaptively-modulated phase-shift-keyed digital signal with analog programming material modulated thereon; and a receiver for detecting the composite signal, extracting the analog audio signal and digital audio, or data, therefrom, and playing back the received audio programming. The adaptively-modulated phase-shift-keyed digital signal is responsive to the transmitted signal, dynamically increasing the digital data transmission rate as the transmitted signal power increases. Symbol states are generally arranged around concentric arcs, and adjacent symbol states are generally equidistant.