| 序号 | 专利名 | 申请号 | 申请日 | 公开（公告）号 | 公开（公告）日 | 发明人 |
|---|---|---|---|---|---|---|
| 181 | LOW-COST HEARING TESTING SYSTEM AND METHOD OF COLLECTING USER INFORMATION | PCT/US2005/020437 | 2005-06-10 | WO2005125002A2 | 2005-12-29 | BURROWS, Mark; CRONIN, John; EDWARDS, Nancy; GABEL, Steven, D.; NARSANA, Tushar; SHAYA, Steven, A.; SINGARAYAR, John, Anthony |
System and method for diagnosing hearing loss in an individual using a self-executable, interactive electronic hearing loss diagnosis apparatus including a data storage media and a media player for accessing data on the media. The diagnostic apparatus provides hearing loss diagnostic data to the individual in the form of coded data. A hearing loss professional can use the coded data to further diagnose the hearing loss of the individual. |
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| 182 | MULTI-CHANNEL DIGITAL FEEDBACK REDUCER SYSTEM | PCT/US2004006124 | 2004-03-01 | WO2004082321A3 | 2005-03-31 | ABRAHAM MATTHEW T |
| The present invention provides method and apparatus that improve processing of acoustic signals by reducing acoustic feedback in an acoustic system. An aspect of the invention is a multi-channel digital feedback reducer (DFR) system that comprises a plurality of channel elements. Each channel element comprises a notch filter configuration having an adaptive notch filter and an operative notch filter. The operative notch filter processes a signal received from an acoustic input device and provides the processed signal to an acoustic output device, in which acoustic feedback between the acoustic input device and the acoustic output device is ameliorated. If acoustic feedback is detected by a channel element, the channel element informs other channel elements of the multi-channel DFR system about the detected feedback to ensure that all channel elements may incorporate the same notch filters. During the notification, the other channel elements may continue searching for feedback on the associated channels. | ||||||
| 183 | A FULLY PARAMETRIC EQUALIZER | PCT/DK2002/000831 | 2002-12-09 | WO2004054099A1 | 2004-06-24 | CHRISTENSEN, Knud, Bank; PEDERSEN, Kim, Rishøj |
The invention relates to a parametric equalizer comprising filtering means (FM), user interface means (UIM), audio signal input means and audio signal output means, said filtering means comprising at least one filter block (FIB) said user interface means (UIM) facilitating adjustment of corner frequency (fc), Shape (Q) and gain (G), said user interface means (UIM) comprising a further adjustment parameter (SYM), said user interface means being mapped by means of coefficient adjustment algorithms (CAD) into filter coefficient settings (FCP) of the at least one filter block (FIB), which when established reflects the adjustment of the user interface means (UIM) said further adjustment parameter (SYM) providing a filter coefficient setting (FCS) comprising a combined adjustment of at least one zero frequency, pole frequency, zero Q and pole Q of at least one filter block. |
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| 184 | CIRCUIT AND METHOD FOR ELIMINATING POP NOISE IN DIGITAL AUDIO AMPLIFIER USING DUAL POWER SUPPLY | PCT/KR2003/001875 | 2003-09-09 | WO2004025830A1 | 2004-03-25 | RYOO, Tae-Ha; JANG, Byung-Tak |
Provided are a circuit for and a method of eliminating pop noise in a digital audio amplifier using dual power supply, which are simple, drop in price, and can be readily implemented in a semiconductor chip. According to an existing technique, pop noise is eliminated using a relay. However, in the present circuit and method, pop noise is eliminated using the small number of discrete electronic devices. The circuit for eliminating pop noise controls a voltage at a gate of a power switch, i.e., a power MOS transistor, when power supply voltages are applied and the application of power supply voltages is stopped. |
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| 185 | CIRCUIT ARRANGEMENT FOR REDUCING THE DYNAMIC RANGE OF AUDIO SIGNALS | PCT/EP0307442 | 2003-07-09 | WO2004010412A2 | 2004-01-29 | SPIELBAUER GEORG; CHRISTOPH MARKUS |
| Circuit arrangement having a first transformation device (3) to which an incoming audio signal (2) is supplied and which transforms this audio signal (2) from the time domain to the frequency domain resulting in an input spectrum (4), a spectral processing device (5), which is connected downstream from the first transformation device (3), to receive the input spectrum (4) and use it to produce an output spectrum (6) such that the output spectrum (6) has a narrower dynamic range than the input spectrum (4), and a transformation device (16), which is connected downstream from the spectral processing device (5), is supplied with the output spectrum (6) and transforms this output spectrum (6) from the frequency domain to the time domain, resulting in an output audio signal (17). | ||||||
| 186 | DYNAMIC ALLOCATION OF POWER SUPPLIED BY A POWER SUPPLY AND FREQUENCY AGILE SPECTRAL FILTERING OF SIGNALS | PCT/US0146702 | 2001-12-05 | WO0249211A3 | 2003-09-18 | CALDWELL ROCKY |
| The power voltage supply for a power amplifier is switched between an operational full power mode and a reduced power mode by the switching of a dual state impedance device connected in series between the power amplifier and the power supply. A control circuit detects the level of a signal to be amplified by that amplifier and switches the impedance states thus reducing the power output capability of the power amplifier and placing the power amplifier into a reduced power mode. In one embodiment, the signal is produced by spectrally filtering with a low pass filter, with the cross-over frequency being adjustable. Additionally, the gain of the amplifier can be reduced for high level signals. | ||||||
| 187 | FILTERSCHALTUNG UND VERFAHREN ZUR VERARBEITUNG EINES AUDIOSIGNALS | PCT/EP2002/007703 | 2002-07-10 | WO2003009469A3 | 2003-01-30 | NEUMANN, Wolfgang; GIER, Hermann |
Die Erfindung betrifft eine Filterschaltung zur Verarbeitung eines Audiosignals, mit einem ersten Zweig, in dem das Audiosignal im Wesentlichen unverändert zu einer Ausgangs-Summierstufe geführt wird, und einem zweiten Zweig, der mehrere in Reihe geschaltete Filterstufen aufweist, wobei das Audiosignal in eine erste Filterstufe der in Reihe geschalteten Filterstufen eingegeben und über die in Reihe geschalteten Filterstufen zu der Ausgangs-Summierstufe geführt wird, und wobei der zweite Zweig mehrere Nebenzweige aufweist, über die das Audiosignal weiteren Filterstufen der in Reihe geschalteten Filterstufen direkt zugeführt wird, und wobei wenigstens eine der Filterstufen einstellbar ist und die Einstellung der einen Filterstufe die Operation wenigstens einer anderen Filterstufe beeinflußt. |
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| 188 | MICROPHONE-TAILORED EQUALIZING SYSTEM | PCT/US2002/013554 | 2002-04-30 | WO2002089521A3 | 2002-11-07 | SCHWARTZ, Stephen, R. |
A method and system is described to improve the reproduction of sound of an acoustic musical instrument (10). According to one embodiment, a first microphone (11) is placed at a proximate location to the musical instrument (10) to pick up the sound of the musical instrument (10). The sound as picked up by the first microphone (11) is compared to a reference sound of the instrument (e.g., the sound of the instrument as perceived at a normal listening position). Based on this comparison, a tailor-made equalizer (13) is designed to compensate for the differences between the sounds as picked up by the first microphone (11) and the reference sounds of the musical instrument (10). |
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| 189 | DIGITAL AUDIO PROCESSOR | PCT/EP2002/001104 | 2002-02-04 | WO2002080358A3 | 2002-10-10 | WEI, Bin; LIU, Shuang, Ming |
The present invention is related to a method for adjusting the frequency characteristic of a digital audio processor having adjustable parameters including at least low frequency gain, high frequency gain and volume gain. The invention is also related to an audio device executing the inventive method every time the characteristic frequency response curve is adjusted. Today's audio devices frequently offer the feature that the user can select among a group of different sound characteristics. Typically the user may want to adapt the sound characteristic of the device to the kind of music he is listening, e.g. rock, jazz, or classic music. In technical terms the sound characteristic is widely determined by the characteristic frequency response curve of the audio signal processing. Those characteristic curves are essentially defined by a gain value in the low, middle and high frequency range. Normally, a 3-band audio processor, which allows adjusting these three parameters independently is used for this purpose. However, 3-band audio processors are relatively more expensive than 2-band audio processors, which allows only adjusting the gain value at the low frequency range and high frequency range. The invention suggests a method how to adjust the volume gain, the low and high frequency gain of a 2-band audio processor to achieve the same features a 3-band audio processor offers. |
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| 190 | ACTIVE FILTER CIRCUIT WITH DYNAMICALLY MODIFIABLE INTERNAL GAIN | PCT/US0204668 | 2002-01-10 | WO02056558A2 | 2002-07-18 | PALASKAS GEORGE; TSIVIDIS YANNIS; TOTH LASZLO |
| Techniques are provided for the implementation of a signal processing circuit (100) which expands the dynamic range of the signal processing circuit (100) without interrupting the output of the circuit. The techniques can receive an input signal (104), proce ss the signal (104) through one of a plurality of dynamically modifiable signal processing circuits, and switch (130) to processing the signal through another of the plurality of signal processing circuits without disturbing the output of the system. | ||||||
| 191 | METHOD AND APPARATUS FOR DYNAMIC SOUND OPTIMIZATION | PCT/US0112250 | 2001-04-13 | WO0180423A3 | 2002-05-10 | CHRISTOPH MARKUS |
| A device and method is presented in which an adjustment to the noise conditions is made for the purpose of controlling the volume and other variables of a desired signal offered in a monitored space, in the course of which, for the purpose of adjustment, a monitoring signal occurring at the monitoring point is picked up and split into a desired-signal component and a noise-signal component. These two components then become the basis for the adjustment. Also in accordance with an embodiment of the invention, an adaptive warped filter is provided. The adaptive warped filter is used to extract a noise related signal which is then used as part of the basis for the adjustment. In an embodiment, the extracted noise signal is corrected and used to control a dynamic controller which adjusts the volume. | ||||||
| 192 | PORTABLE EQUALIZER WITH FREQUENCIES ADJUSTABLE ACCORDING TO THE USER'S HEARING CAPACITY | PCT/IT2001/000074 | 2001-02-16 | WO01063757A3 | 2002-03-21 | |
| The invention is a new portable equalizer with several audio inputs and one audio output, which separates the incoming audio signals into various predetermined frequency bands in the range of audible frequencies, emphasizes or attenuates each frequency band as set by the user according to his/her hearing deficiencies, mixes the emphasized or attenuated frequency bands and conveys them to the audio output. It can be provided with a specific input for the telephone connection instead of the handset. Thanks to this new equipment, even the persons with partial hearing deficiencies will be able to hear completely, it is possible to define the gain or the attenuation for the various frequency bands, it is possible to adjust the general volume of the equalizer, anyone who has hearing deficiencies can carry out a personalized adjustment of the operation of the new equalizer according to his/her own specific needs. The device may feature analogic or digital technology, even with the use of DSP (digital signal processor) technology. | ||||||
| 193 | VEHICLE ACCESSORY MICROPHONE | PCT/US0031708 | 2000-11-17 | WO0137519A3 | 2001-11-29 | WATSON ALAN R; KNAPP ROBERT C; TURNBULL ROBERT R; SPENCE WILLIAM R; POE G BRUCE; BRYSON MICHAEL A |
| A microphone assembly (1700) includes one or more transducers (1702) positioned in a housing (1732). Circuitry (Fig. 8) is coupled to the transducer for outputting an electrical signal such that the microphone has a main lobe directed forwardly and attenuates signals originating from the sides and/or rear. The transducers can advantageously include multiple transducers, which, with the circuit, produce a desired sensitivity pattern. The microphone assembly can be employed in a vehicle accessory. | ||||||
| 194 | COMPLEMENTARY TRANSFER FUNCTION DESIGN OF CROSSOVER FILTERS IN LOUDSPEAKER SYSTEMS | PCT/US2000/013667 | 2000-05-18 | WO01089083A1 | 2001-11-22 | |
| The invention provides a system and method of designing complementary transfer functions (CTF) [109] for crossover filters that adjust for various crossover considerations. According to the invention, a computer-implemented system and procedure, in a specific audio environment, processes crossover factors [110], designs a CTF filter in a computer simulation [111], correlates measurements of crossover parameters with the computer simulations [112], and adjusts the CTF design [14] until there is a high or predetermined degree of correlation between the computer simulation and the crossover measurements. In particular, the inventive system and method achieves proper blending between chosen sets of subwoofers [105] and satellite speakers [101] in loudspeaker systems. | ||||||
| 195 | AUDIO DEVICE DRIVER | PCT/US2001/000224 | 2001-01-04 | WO01050598A2 | 2001-07-12 | |
| A device driver for a computer operating system. The operating system is adapted to receive digitized audio data from a media source. The device driver comprises an intercepting component driver which is capable of initiating at least one of sound modification processing of the digitized data and storage of the digitized data. | ||||||
| 196 | NOISE REDUCING/RESOLUTION ENHANCING SIGNAL PROCESSING METHOD AND SYSTEM | PCT/US2000/033139 | 2000-12-06 | WO01041427A1 | 2001-06-07 | |
| A noise reduction/resolution enhancement signal processing system (Figs. 21-24) and a corresponding method (Fig. 20) wherein the influence of noise spikes and gaps is substantially reduced. The noise reduction data may be amplitudes (Fig. 3, "bx") measured at corresponding values (x) over a given domain D, wherein the data defines a composite wave form, which is decomposed into instances of a discrete wave form, each having reduced noise amplitudes. A candidate point c in D (Fig. 20) for, e.g., an amplitude extreme is determined for each discrete wave from instance (having unknown amplitude), wherein a minimization technique (Fig. 20) determines a first set of discrete wave from instances (having known amplitudes), by collapsing on the amplitudes (bx) from above, wherein a maximization technique determines a second set of discrete wave form instances (having known amplitudes) by rising up to the amplitudes (bx) from below. For each point c, a determination is made as which of the corresponding instances from the first and second sets is a better reduction of noise, which is used to provide a resulting discrete wave form instance at c. Embodiments are disclosed for mass spectrometry (Fig. 21), digital imaging (Fig. 23) and audio applications (Fig. 24). | ||||||
| 197 | RECORDING AND PLAYBACK CONTROL SYSTEM | PCT/US1999/023279 | 1999-10-06 | WO01026220A1 | 2001-04-12 | |
| The invention is a system for recording and reading both program data and acoustical control data and playing back the data to optimize performance of audio reproduction and recreate the effect of an original acoustic environment. The system has a recording apparatus (10), a playback apparatus and a recording media. The recording apparatus (10) produces recording media having both acoustic control information and audio data. The playback apparatus gives the user some ability to override otherwise automatic parameter adjustments. Optionally, a metadata display system (90) takes information about the physical arrangement of instruments and other characteristics of the recording session and the recording studio and makes that visually available to the listener. A player type register (82) identifies the characteristics of the playback device to cause an adjustment of the characteristics of the playback system. | ||||||
| 198 | DIRECTIONAL MICROPHONE ASSEMBLY FOR MOUNTING BEHIND A SURFACE | PCT/US9902336 | 1999-02-03 | WO9939545A3 | 1999-10-21 | JULSTROM STEPHEN D; SCHULEIN ROBERT B |
| A directional microphone suitable for subsurface mounting. A directional pickup pattern is developed from the outputs of a plurality microphone elements (145, 147, 149) mounted in an assembly behind a surface such that their acoustic excitation comes from the opposite side of the surface through small openings in the surface (61, 63). The openings may have varying dimensions and may be covered with acoustically semitransparent material without significantly degrading the assembly frequency response or polar pattern. Precautions are taken to ensure design robustness considering practical microphone element characteristics and potential high levels of low-frequency excitation. | ||||||
| 199 | DIRECTIONAL MICROPHONE ASSEMBLY FOR MOUNTING BEHIND A SURFACE | PCT/US1999/002336 | 1999-02-03 | WO99039545A2 | 1999-08-05 | |
| A directional microphone assembly suitable for subsurface mounting. A directional pickup pattern is developed from the outputs of a plurality of omnidirectional microphone elements mounted in an assembly behind a surface such that their acoustic excitation comes from the opposite side of the surface through small openings in the surface. The openings may have varying dimensions and may be covered with acoustically semitransparent material without significantly degrading the assembly frequency response or polar pattern. Precautions are taken to ensure design robustness considering practical microphone element characteristics and potential high levels of low-frequency excitation. | ||||||
| 200 | SERIAL DIGITAL DATA COMMUNICATIONS RECEIVER WITH IMPROVED AUTOMATIC CABLE EQUALIZER, AGC SYSTEM, AND DC RESTORER | PCT/CA1998/000296 | 1998-04-01 | WO98045955A2 | 1998-10-15 | |
| A serial digital data communications receiver with an improved automatic cable equalizer that is less susceptible to jitter and has greater multi-standards capability, and an improved automatic gain control system with a DC restorer that provides optimal edge jitter performance while avoiding the possibility of a latch-up condition at the start of data transmission. The automatic cable equalizer for equalizing signals received over cables of different lengths has multiple stages each having a transfer function of 1 + Ki[fi(j omega )] wherein each of the Ki varies in accordance with a sequential gain control methodology. The AGC system uses the difference between band-pass filtered versions of the amplitudes of the input and output of a DC restorer based on quantized feedback, to regulate the AGC circuit. The DC restorer comprises a comparator for generating a quantized output and further clamps the input with a clamping circuit so that a version of the quantized output is fed back to the input while avoiding the possibility of operational failure of the comparator at the onset of data transmission. | ||||||
