With Amplitude Compression/expansion Patents (Class 381/106)
  • Patent number: 7272235
    Abstract: A request is received to play an audio file. A determination is made regarding whether volume normalization parameters associated with the audio file are stored in a media library. If the volume normalization parameters associated with the audio file are stored in the media library, the volume normalization parameters are retrieved from the media library. If the volume normalization parameters associated with the audio file are not stored in the media library, retrieving the volume normalization parameters from the audio file. The volume normalization parameters are applied while playing the audio file. The volume normalization process can be applied across multiple audio files during playback.
    Type: Grant
    Filed: June 26, 2003
    Date of Patent: September 18, 2007
    Assignee: Microsoft Corporation
    Inventors: Phillip Lu, Adil Sherwani, Kipley J. Olson
  • Patent number: 7269264
    Abstract: A method for reproducing audio signals of at least two different sources, a first volume for reproducing audio signals of a first source being able to be preselected, in which, during simultaneous reproduction of the audio signals of the first and the at least second source, the reproduction of the audio signals of the at least second source occurs at a volume that is raised compared to the first volume at least by a differential volume. This allows for a comprehensible reproduction of prioritized audio signals of the second source even against the background of a continuous reproduction of audio signals of a first source.
    Type: Grant
    Filed: August 24, 2001
    Date of Patent: September 11, 2007
    Assignee: Robert Bosch GmbH
    Inventor: Berthold Gierse
  • Patent number: 7254243
    Abstract: In accordance with the invention, audio signals are specially processed for sound presentation in a high noise environment. The electrical signal representative of the sound is first subjected to equalization to preferentially reduce the magnitude of bass signals while increasing the magnitude of treble signals. The equalized signal is then compressed, and the compressed signal is subjected to “mirror image” equalization which increases the magnitude of bass signals while reducing the magnitude of treble signals. The resulting signal fed to the speakers provides a sound presentation of compressed volume range and a bass-rich sound spectrum. It is particularly useful for providing quality sound presentation in a high noise environment.
    Type: Grant
    Filed: August 10, 2004
    Date of Patent: August 7, 2007
    Inventor: Anthony Bongiovi
  • Patent number: 7233833
    Abstract: A method of modifying low frequency components of a digital audio signal having left and right channel signals, including the steps of: a) filtering the left and right channels signals using respective left and right high-pass filters to form left and right high-pass filtered signals; b) filtering the left and right channel signals using respective left and right band-pass filters to form left and right low frequency signals; c) modifying the amplitude of the left and right low frequency signals to give modified left and right low frequency signals whereby signals with amplitude a where 0<a<a1 are amplified by a first constant value C1, signals with amplitude a1?a<a2 are amplified proportional to 1/a, signals with amplitude a=2a are unchanged, signals with amplitude a2<a<a3 are attenuated proportional to 1/a, and signals with amplitude a=a3 are attenuated by a second constant value C2; and d) combining the modified band-pass filtered left and right signals with the respective left and right hig
    Type: Grant
    Filed: September 10, 2003
    Date of Patent: June 19, 2007
    Assignee: Creative Technology Ltd
    Inventor: Max Andrew Little
  • Patent number: 7224808
    Abstract: A system configured to dynamically adjust the ultrasonic carrier level in a parametric array system in response to changing source signal input levels, and which employs a look-ahead delay strategy to enable optimal modulation of the carrier wave to eliminate constant ultrasonic carrier emission and reduce the ultrasonic carrier emission to what is actually needed to accommodate the db range of the source material, and at the same time, to also minimize noticeable distortion and sound artifacts of a high-power ultrasonic carrier, and/or distortion/artifacts arising from modulation of an ultrasonic carrier to reduce average power output; and thus it realizes advantages of carrier modulation based on source-signal level, while minimizing inherent drawbacks of carrier modulation.
    Type: Grant
    Filed: September 3, 2002
    Date of Patent: May 29, 2007
    Assignee: American Technology Corporation
    Inventors: Michael E. Spencer, James J. Croft, III
  • Patent number: 7216221
    Abstract: A system and method for improved audio controls on a personal computer is provided. The system and method provide a unified architecture for audio controls across hardware and software interfaces of the personal computer. An intelligent facility may automatically change audio controls for users to simply interact with various communications and media applications. To this end, a configurable audio controller intelligently handles various aspects of the system's audio devices by following various rules that may be based at least in part on user-configurable settings and a current operating state. The present invention also provides audio controls so that a user may easily change audio settings such as the volume of an audio output device. There are many applications that may use the present invention for automatic control of audio devices based upon the user's context.
    Type: Grant
    Filed: September 30, 2003
    Date of Patent: May 8, 2007
    Assignee: Microsoft Corporation
    Inventors: Eric Gould Bear, Chad Magendanz, Aditha May Adams, Carl Ledbetter, Steve Kaneko, Dale C. Crosier
  • Patent number: 7212640
    Abstract: Variable attack and release system and method are disclosed for dynamically modifying the various elements of the system, including modifying compander response time to dynamically adjust for changing parameters such as environmental or input signal changes, to maintain output signals within predetermined limits. The system and method include permitting accurate and quick response to noise sources having a duration which exceeds a predetermined threshold while at the same time ignoring transient or short duration environmental noise.
    Type: Grant
    Filed: November 29, 2000
    Date of Patent: May 1, 2007
    Inventor: Karl M. Bizjak
  • Patent number: 7206420
    Abstract: A method of matching input amplitudes in a system wherein one or more of a plurality of inputs may be selected, with each input capable of having different characteristics, involving selecting an input signal and mapping the input signal to a predetermined signal amplitude through the use of level matching logic. The level matching logic may include a gain cell for increasing or decreasing the amplitude of the input signal.
    Type: Grant
    Filed: November 29, 2000
    Date of Patent: April 17, 2007
    Assignee: SYFX Tekworks
    Inventor: Karl M. Bizjak
  • Patent number: 7181028
    Abstract: An audio converting device including a digital high-pass filter, an expander, a digital low-pass filter, a delta-sigma modulator, a digital-to-analog converter, an analog low-pass filter and a gain control unit is provided. The digital high-pass filter in this invention can filter out a direct-current component of digital audio data such that the production of noise is avoided when the volume is adjusted by users.
    Type: Grant
    Filed: March 21, 2002
    Date of Patent: February 20, 2007
    Assignee: Realtek Semiconductor Corp.
    Inventors: Shih-Yu Ku, Wen-Chi Wang, Yi-Shu Chang, Chao-Cheng Lee
  • Patent number: 7110559
    Abstract: The invention concerns a method (300) and system (100) for controlling audio output. The method includes the steps of inputting (312) an audio signal and a voltage level signal, measuring (314) the audio signal and the voltage level signal, mapping (316) the audio signal against at least one table (134) of predetermined corresponding gain targets (138) and selecting (318) at least one gain target for the audio signal. The mapping step and the selecting step are based at least in part on the measurement of the voltage level signal and the measurement of the audio signal. The method also includes the step of applying (320) the gain target to the audio signal.
    Type: Grant
    Filed: November 6, 2003
    Date of Patent: September 19, 2006
    Assignee: Motorola, Inc.
    Inventors: Ali Behboodian, Audley F. Patterson
  • Patent number: 7110555
    Abstract: A sound signal reproducing apparatus for supplying sound signals of a plurality of channels to speakers via sound amplifier circuits includes: a power supply voltage controller for controlling a power supply voltage to the sound amplifier circuits; a plurality of filter circuits for passing predetermined frequencies of the output signal of each of the sound amplifier circuits in the plurality of channels; a plurality of level detection circuits for detecting levels of output signals of the plurality of filter circuits; an adder circuit for adding output signals of the plurality of level detection circuits; and a comparator circuit for comparing an output signal of the adder circuit with a reference level, so that an output signal of the comparator circuit controls the power supply voltage controller.
    Type: Grant
    Filed: July 30, 2002
    Date of Patent: September 19, 2006
    Assignee: Sony Corporation
    Inventor: Hideaki Shiobara
  • Patent number: 7107109
    Abstract: The present invention relates to a process for adjusting the sound volume of a digital sound recording characterized in that it comprises: a step consisting of determining, in absolute values, for a recording, the maximum amplitude values for sound frequencies audible for the human ear, a step consisting of calculating the possible gain for a specified sound level setting, between the maximum amplitude value determined above and the maximum amplitude value for all frequencies combined, a step consisting of reproducing the recording with a sound card by automatically adjusting the amplification gain level making it possible to obtain a sound level for the recording of a specified value so that it corresponds to the gain calculated for this recording.
    Type: Grant
    Filed: June 1, 2000
    Date of Patent: September 12, 2006
    Assignee: Touchtunes Music Corporation
    Inventors: Guy Nathan, Dominique Dion
  • Patent number: 7031905
    Abstract: An audio signal encoding apparatus includes a device for compressing multiple-channel digital audio signals into compression-resultant multiple-channel signals respectively. The multiple-channel digital audio signals relate to a sampling frequency and a quantization bit number. The compression-resultant multiple-channel signals, a signal representative of the sampling frequency, and a signal representative of the quantization bit number are formatted into a formatting-resultant signal. The formatting-resultant signal contains a sub packet and a sync information portion. The sub packet contains at least portions of the compression-resultant multiple-channel signals. The sync information portion contains the signal representative of the sampling frequency and the signal representative of the quantization bit number.
    Type: Grant
    Filed: May 27, 2004
    Date of Patent: April 18, 2006
    Assignee: Victor Company of Japan, Ltd.
    Inventors: Yoshiaki Tanaka, Shoji Ueno, Norihiko Fuchigami
  • Patent number: 7027981
    Abstract: Output control system and method includes calculating gain for an input signal, detecting a predetermined condition, and modifying the gain calculation in accordance with one or more kneepoints to provide a varying output signal. A companding ratio is then established in accordance with the kneepoints. System gain is then adjusted, and the companding ratio may be further adjusted by adjusting one of the kneepoints relative to another.
    Type: Grant
    Filed: November 29, 2000
    Date of Patent: April 11, 2006
    Inventor: Karl M. Bizjak
  • Patent number: 7024260
    Abstract: Maximal and minimal values represented by samples of a digital audio signal are detected. A number of samples from a sample representing a minimal value to a maximal-value-corresponding sample is detected. A number of samples from a sample representing a maximal value to a minimal-value-corresponding sample is detected. Calculation is given of a first difference between the maximal-value-corresponding sample and the immediately-preceding sample. Calculation is given of a second difference between the minimal-value-corresponding sample and the immediately-preceding sample. First and second coefficients are calculated from the detected sample numbers. The first coefficient and the first difference are multiplied to generate a first multiplication result. The second coefficient and the second difference are multiplied to generate a second multiplication result. The maximal value represented by the maximal-value-corresponding sample is incremented by the first multiplication result.
    Type: Grant
    Filed: November 8, 2001
    Date of Patent: April 4, 2006
    Assignee: Victor Company of Japan, Ltd.
    Inventor: Toshiharu Kuwaoka
  • Patent number: 7010491
    Abstract: With the goal of presenting a waveform compression and expansion apparatus with which the sound quality of such things as musical tones that are expressed by waveforms is satisfactory following the compression and expansion of the waveforms of the musical tones etc., a method and system for waveform compression and expansion is disclosed in which all of the multiple number of band divided waveforms that comprise the original waveform which has been band divided are apportioned to at least two kinds of compression and expansion formats and form a multiple number of compressed and expanded waveforms by compression or expansion an identical amount only in the direction of the temporal axis.
    Type: Grant
    Filed: December 9, 1999
    Date of Patent: March 7, 2006
    Assignee: Roland Corporation
    Inventor: Tadao Kikumoto
  • Patent number: 6970570
    Abstract: A hearing aid device providing instantaneous gain compression for sound signals and adaptive control of nonlinear waveform distortion, the device comprising: (a) at least one bandpass nonlinearity (BPNL) amplifier comprising a first bandpass filter, a second bandpass filter, and a memoryless nonlinear (MNL) compressive audio amplifier configured to receive a sound signal from the first bandpass filter and provide an MNL compressive audio amplifier output to the second bandpass filter, wherein the MNL compressive audio amplifier is configured to produce the MNL compressive audio amplifier output by providing memoryless gain compression directly on a sound signal that is (1) received from the first bandpass filter and (2) exhibits instantaneous amplitudes greater than a compression threshold, the BPNL amplifier thereby producing a desired gain compression on the received sound signal at an output of the second bandpass filter, and (b) a controller in communication with the BPNL amplifier, the controller being c
    Type: Grant
    Filed: August 23, 2001
    Date of Patent: November 29, 2005
    Assignee: Hearing Emulations, LLC
    Inventor: Julius L. Goldstein
  • Patent number: 6970571
    Abstract: An earmuff comprising a headpiece and a circuit. The headpiece supports the earmuff on the head of an individual. The circuit has an input device for receiving external sound energy and converting the external sound energy to electrical sound signals.
    Type: Grant
    Filed: February 3, 2003
    Date of Patent: November 29, 2005
    Assignee: Jackson Products, Inc.
    Inventors: Jon P. Knorr, Benjamin P. Knapp
  • Patent number: 6937912
    Abstract: A method for computing the “clippings” of an audio signal in the digital domain to prevent aliasing is disclosed. An offset signal is found by subtracting a threshold from samples of the audio signal. These are multiplied by a pulse after the pulse is lined up in time with the crossing of the audio signal at the threshold. The pulse is the sum of two step functions, both of which are bandwidth-limited.
    Type: Grant
    Filed: April 5, 2000
    Date of Patent: August 30, 2005
    Assignee: Orban, Inc.
    Inventor: Robert A. Orban
  • Patent number: 6934593
    Abstract: A device for decoding audio signals subjected to a noise-reduction encoding such as a Dolby-B encoding comprises a plurality of processing blocks (612, 616, 620, 624, 628, 632, 636, 640) for generating, starting from an input signal (610) containing the audio signal subjected to encoding superimposed on a noise component, an output signal (626) consisting of a replication of the audio signal with the noise component reduced. The aforesaid processing blocks are implemented in a digital form and comprise a sliding-band filtering structure (612) fed with the input signal (610) and designed to generate a filtered signal (614). The filtered signal (614) is fed, according to a general feedforward scheme, to an overshoot-suppression stage (616) in view of the generation, in an adder node (620), of a difference signal (622) starting from which the output signal is obtained via filtering (624).
    Type: Grant
    Filed: June 12, 2002
    Date of Patent: August 23, 2005
    Assignee: STMicroelectronics S.r.l.
    Inventors: Federico Fontana, Mario Bricchi
  • Patent number: 6920188
    Abstract: An apparatus is provided for processing a wide dynamic range analog signal which comprises multiple components such as a signal with an in-phase component and a quadrature-phase component in for example separate I and Q data channels, wherein each channel has a dynamic range compressor stage and an operator stage which processes the compressed signals. Optionally the apparatus has a dynamic range expander stage following the operator stage. A method according to the invention involves processing I and Q information after first independently compressing the dynamic range of the signal according to a logarithmic transfer characteristic over a frequency range of interest. A mathematical operation through a F(i,q) function (corresponding to the operator stage) is performed on the compressed components, thereby producing normalized components. The operating transfer function F(i,q) cross links the data channels to effect normalization based on amplitude of information in each of the channels.
    Type: Grant
    Filed: November 16, 2000
    Date of Patent: July 19, 2005
    Assignee: Piradian, Inc.
    Inventors: James Terrell Walker, Kamran Khorram Abadi, Robert Gustav Lorenz
  • Patent number: 6907129
    Abstract: A circuit and related method for digital volume control are provided, where the circuit includes a digital filter configured to process samples of an input stream in a manner that processes a previous input sample during a time interval before a subsequent input sample, and outputs a series of exponentially decaying waveforms. The result is an exponential response to a volume change made by a user, where the change feels more pleasant and natural than a conventional linear response.
    Type: Grant
    Filed: December 9, 2003
    Date of Patent: June 14, 2005
    Assignee: ESS Technology, Inc.
    Inventor: Andrew Martin Mallinson
  • Patent number: 6885752
    Abstract: A hearing compensation system for the hearing impaired comprises a plurality of bandpass filters having an input connected to an input transducer and each bandpass filter having an output connected to the input of one of a plurality of multiplicative AGC circuits whose outputs are summed together and connected to the input of an output transducer. The multiplicative AGC circuits attenuate acoustic signals having a constant background level without the loss of speech intelligibility. The identification of the background noise portion of the acoustic signal is made by the constancy of the envelope of the input signal in each of the several frequency bands. The background noise that will be suppressed includes multi-talker speech babble, fan noise, feedback whistle, florescent light hum, and white noise.
    Type: Grant
    Filed: November 22, 1999
    Date of Patent: April 26, 2005
    Assignee: Brigham Young University
    Inventors: Douglas Melvin Chabries, Richard Wesley Christiansen, Aaron Michael Hammond, William Charles Borough
  • Patent number: 6882735
    Abstract: Computer processor method and apparatus for creating an audio multiplier control signal for controlling the dynamic range of a recorded audio work. The technique includes determining an envelope of the amplitude of amplitude versus time values the audio signal, and then determining, for values of the envelope, respective minimum and maximum multiplication factors (MinMF and MaxMF) that can be multiplied times the values such that the products are above a predetermined minimum amplitude and below a predetermined maximum amplitude of the dynamic range. Then a control signal function of amplitude versus time is created such that all values of the control signal function at particular times are between respective MinMF and MaxMF values for the times, and such that segments of the control signal function have reduced slopes. Also disclosed is a method of creating a reduced-slope series of line segments passing through a pair of Max and Min limiting functions specifying y values with respect to a variable x.
    Type: Grant
    Filed: January 11, 2001
    Date of Patent: April 19, 2005
    Assignee: Autodesk, Inc.
    Inventor: Clinton A. Staley
  • Patent number: 6873709
    Abstract: Improved approaches are disclosed to filter and compress sound signals so as to achieve not only speech audibility and intelligibility at low levels but also preserves spectrum contrast at high levels. According to one aspect of the invention, gain amounts for different frequency bands are individually constrained based on signal levels for the frequency bands. Hence, the gain amounts for each of the frequency bands may or may not be constrained depending on the corresponding signal levels. As a result, the most critical information for speech intelligibility, speech clarity, and speech quality can be made available to hearing impaired people over wide range of signal level. The invention is particularly useful for hearing aids or other sound systems for the hearing impaired.
    Type: Grant
    Filed: August 7, 2001
    Date of Patent: March 29, 2005
    Assignee: Apherma Corporation
    Inventor: Zezhang Hou
  • Publication number: 20040258246
    Abstract: The invention relates to an apparatus (1) for amplifying bass frequencies of an audio signal (AL, AR). The bass components are applied to a controllable amplifier (17).
    Type: Application
    Filed: April 14, 2004
    Publication date: December 23, 2004
    Inventor: Loic Bernard Tanghe
  • Publication number: 20040249489
    Abstract: The present invention involves a digital audio player and a method for processing encoded digital audio data, wherein the digital audio data is encoded using one of a plurality of encoding formats. The audio data player has a hard disk or other data storage medium for storing data files, a microcontroller buffer memory for anti-skip protection, and an audio decoder. The encoded audio data files and associated decoder files are downloaded from a personal computer or similar device to the audio data player hard drive. The player provides a menu-driven user interface for selection, sorting, and playback of stored audio data files. During playback, elapsed playback time is computed and displayed. For variable bit-rate audio data files, the audio data player generates an elapsed playback timekeeping map concurrently with playback and fast forward scan of audio data files.
    Type: Application
    Filed: March 5, 2004
    Publication date: December 9, 2004
    Inventor: Robert James Dick
  • Publication number: 20040225388
    Abstract: The invention relates to a fully digitized audio system comprising a power supply and a decode and sound field effect process unit, characterized in that a digital audio signal output from the decode and sound field effect process unit is transmitted to a control and encode unit where audio and control signals are encoded, and coupled to a digital sound box through a digital transmission terminal in the control and encode unit. The invention adopts totally digitized audio signal processing according to the concept of mechatronics, solving the problem of distortions rising throughout the procedure from input, process, distribution, transmission and amplification to sounding, to ensure a controllable hi-fi output of audio signal. The audio system according to the present invention provides high fidelity, good controllability, and easiness of assembling, and is suitable for mass production of advance audio systems.
    Type: Application
    Filed: July 17, 2002
    Publication date: November 11, 2004
    Inventors: Guohua Zhang, Limin Lai, Enyi Zhan, Aihua Zhou
  • Publication number: 20040213420
    Abstract: A single control determines both volume and the degree of compression in the reproduction of motion picture soundtracks. For some settings of the control, compression or limiting reduces the highest levels on a movie soundtrack, leaving the dialogue level substantially unchanged, thus removing the reason for complaints from the audience without the danger of making the dialogue too quiet for intelligibility.
    Type: Application
    Filed: April 24, 2003
    Publication date: October 28, 2004
    Inventors: Kenneth James Gundry, John Iles, Roger Wallace Dressler
  • Patent number: 6807280
    Abstract: An audio signal processing circuit reduces noise in an incoming audio signal and is particularly useful in telephone communication systems utilizing one or more hands-free microphones. A preferred embodiment of the audio signal processing circuit includes a pre-emphasis circuit receiving the audio signal from a microphone or other transducer, an amplifier circuit receiving the pre-emphasized audio signal from the pre-emphasis circuit and a de-emphasis circuit receiving the amplified signal from the amplifier circuit. An output of the de-emphasis circuit provides the processed audio signal having an improved signal to noise ratio with minimum audible distortion. A preferred embodiment of the amplification circuit includes an amplifier defining a non-linear transfer function therethrough which provides low gain to the lower amplitude noise signals and higher gain to the higher amplitude audio signals.
    Type: Grant
    Filed: January 26, 1998
    Date of Patent: October 19, 2004
    Assignee: Delphi Technologies, Inc.
    Inventors: Richard Sidney Stroud, Gary Michael McQuilling
  • Publication number: 20040190734
    Abstract: A multi-channel signal processing system adapted to provide binaural compressing of tonal inputs is provided. Such a system can be used, for example, in a binaural hearing aid system to provide the dynamic-range binaural compression of the tonal inputs. The multi-channel signal processing system is essentially a system with two signal channels connected by a control link between the two signal channels, thereby allowing the binaural hearing aid system to model behaviors, such as crossed olivocochlear bundle (COCB) effects, of the human auditory system that includes a neural link between the left and right ears. The multi-channel signal processing system comprises first and second channel compressing units respectively located in first and second signal channels of the multi-channel signal processing system. The first and second channel compressing units receive first and second channel input signals, respectively, to generate first and second channel compressed outputs.
    Type: Application
    Filed: January 27, 2003
    Publication date: September 30, 2004
    Applicant: GN ReSound A/S
    Inventor: James M. Kates
  • Publication number: 20040184621
    Abstract: Systems and methods for detecting clipping conditions in an audio signal and processing the signal to reduce the clipping conditions. In one embodiment, a system comprises a noise shaper, a modulator, an output stage and other components. A detector detects clipping in the noise shaper and a signal processor processes the audio signal input to the noise shaper based on feedback received from the detector. The signal processor may function to modify the input audio signal in different ways in response to different conditions that are detected by the detector. A filter may be included to filter the output of the detector before being provided to the signal processor. A flag circuit may be coupled between the filter and the signal processor to assert an output signal until the signal processor resets the flag circuit.
    Type: Application
    Filed: March 19, 2004
    Publication date: September 23, 2004
    Inventors: Jack B. Andersen, Larry E. Hand, Wilson E. Taylor
  • Patent number: 6788792
    Abstract: An amplitude adjustment device such as an amplitude compression device and amplitude expansion device is basically configured by a PWM modulator, a demodulator and an amplitude detector. Herein, the PWM modulator effects pulse-width modulation on an input signal to produce a pulse-width modulated signal, which is demodulated by the demodulator to produce an output signal. In addition, the amplitude detector detects an amplitude of a demodulated signal or an amplitude of the input signal to produce a control signal. A modulation factor of the pulse-width modulation is adjusted based on the control signal. In the case of the amplitude compression device, an input/output gain is changed inversely proportional to the amplitude of the input signal or amplitude of the output signal. Thus, it is possible to compress a dynamic range with respect to input/output characteristics.
    Type: Grant
    Filed: June 24, 1999
    Date of Patent: September 7, 2004
    Assignee: Yamaha Corporation
    Inventors: Toshio Maejima, Masao Noro
  • Publication number: 20040165737
    Abstract: An audio codec and a method of compressing audio data makes use of a filterbank which automatically adapts itself to changes in the sampling frequency/bit rate to mimic the characteristics of the human auditory system. The algorithm used compares the bandwidth of each sub-band at a given depth with the critical bandwidth. If the critical bandwidth is less than the bandwidth of the sub-band, then the sub-band is split into two at the next level, and the process is repeated until the bandwidth of every sub-band is less than the critical bandwidth at the corresponding frequency. The codec thus automatically adapts itself to changes in sampling frequency/bit rate, which is particularly advantageous when very low bandwidths are in use.
    Type: Application
    Filed: April 12, 2004
    Publication date: August 26, 2004
    Inventor: Donald Martin Monro
  • Patent number: 6763253
    Abstract: Proposed by the invention is a device for the bidirectional transfer of audio and/or video signals, in particular in the context of sound and/or image reports, with: at least one means (6) for providing an audio input signal; a first mixing device (10), which is connected to the means (6) for providing the audio input signal and which is designed to output a mixed audio transmission signal; a transmission and/or reception device coupled to the first mixing device (10) for transmitting the mixed audio transmission signal and/or receiving an audio reception signal; a control device (22) coupled to the first mixing device (10) for controlling the first mixing device (10); a compression and/or decompression device (12, 32) for compression of the mixed audio transmission signal or, as the case may be, for decompression of the audio reception signal, which compression and/or decompression device is connected to the first mixing device (10) for taking up the mixed audio transmission signal or, as the case may
    Type: Grant
    Filed: October 26, 2000
    Date of Patent: July 13, 2004
    Assignee: Sennheiser Electronics GmbH & Co. KG
    Inventors: Wolfgang Niehoff, Axel Haupt
  • Publication number: 20040131204
    Abstract: A perceptual encoder divides an audio signal into successive time blocks, each time block is divided into frequency bands, and a scale factor is assigned to each of ones of the frequency bands. Bits per block increase with scale factor values and band-to-band variations in scale factor values. A preliminary scale factor for each of ones of the frequency bands is determined, and the scale factors for the each of ones of the frequency bands is optimized, the optimizing including increasing the scale factor to a value greater than the preliminary scale factor value for one or more of the frequency bands such that the increase in bit cost of the increasing is the same or less than the reduction in bit cost resulting from the decrease in band-to-band variations in scale factor values resulting from increasing the scale factor for one or more of the frequency bands.
    Type: Application
    Filed: January 2, 2003
    Publication date: July 8, 2004
    Inventor: Mark Stuart Vinton
  • Patent number: 6760452
    Abstract: An audio system limits gain in individual audio channels having a shared clip detect signal from a multi-channel power amplifier such that only channels likely to be exceeding the distortion threshold are gain limited. In addition, any arbitrary channels may be grouped together for common gain limiting. The invention monitors the power level of each audio channel, compares the power level to a power threshold that indicates whether a high power condition with the potential to cause excess distortion exists or not, and makes a decision whether to activate each channel's gain limiter based on whether the corresponding clip detect signal is active and the high power condition exists simultaneously for that channel.
    Type: Grant
    Filed: October 24, 2002
    Date of Patent: July 6, 2004
    Assignee: Visteon Global Technologies, Inc.
    Inventors: Kai Kwang Lau, Robert Kelly Cadena, John Elliott Whitecar
  • Patent number: 6757396
    Abstract: A digital audio dynamic range compressor includes a root mean square estimator receiving first and second audio input samples and generating root mean square values of the samples. A gain calculator receives the root mean square values and computes a gain for each input sample in the linear domain, not in the logarithmic or dB domain. A minimum selector receives the computed gain of each input sample and determines a minimum. An attack and release filter receives the minimum gain value and filters the minimum gain value according to attack and release coefficients and generate a gain output. A multiplier receives the gain output and multiplies the first and second audio input samples with the gain output.
    Type: Grant
    Filed: September 27, 1999
    Date of Patent: June 29, 2004
    Assignee: Texas Instruments Incorporated
    Inventor: Rustin W. Allred
  • Patent number: 6757659
    Abstract: An audio signal encoding apparatus includes a device for compressing multiple-channel digital audio signals into compression-resultant multiple-channel signals respectively. The multiple-channel digital audio signals relate to a sampling frequency and a quantization bit number. The compression-resultant multiple-channel signals, a signal representative of the sampling frequency, and a signal representative of the quantization bit number are formatted into a formatting-resultant signal. The formatting-resultant signal contains a sub packet and a sync information portion. The sub packet contains at least portions of the compression-resultant multiple-channel signals. The sync information portion contains the signal representative of the sampling frequency and the signal representative of the quantization bit number.
    Type: Grant
    Filed: November 2, 1999
    Date of Patent: June 29, 2004
    Assignee: Victor Company of Japan, Ltd.
    Inventors: Yoshiaki Tanaka, Shoji Ueno, Norihiko Fuchigami
  • Patent number: 6747678
    Abstract: As a desired digital signal processor (DSP) mode is selected from DSP setting buttons displayed in a DSP setting window, only those slide bars for changing parameters of the selected DSP mode are displayed. As a slider bar corresponding to a parameter to be changed is selected from the slide bars displayed in the DSP setting window, an area of an impulse response diagram to be influenced by a change in the parameter value is indicated in a discriminatory manner by an arrow or the like. A user can visually confirm the acoustic effect to be changed by the value of each DSP parameter and can change the parameter value with ease.
    Type: Grant
    Filed: June 15, 2000
    Date of Patent: June 8, 2004
    Assignee: Yamaha Corporation
    Inventors: Masaki Katayama, Yasuhiro Fujimura, Tetsuya Matsuyama
  • Publication number: 20040081324
    Abstract: An audio system limits gain in individual audio channels having a shared clip detect signal from a multi-channel power amplifier such that only channels likely to be exceeding the distortion threshold are gain limited. In addition, any arbitrary channels may be grouped together for common gain limiting. The invention monitors the power level of each audio channel, compares the power level to a power threshold that indicates whether a high power condition with the potential to cause excess distortion exists or not, and makes a decision whether to activate each channel's gain limiter based on whether the corresponding clip detect signal is active and the high power condition exists simultaneously for that channel.
    Type: Application
    Filed: October 24, 2002
    Publication date: April 29, 2004
    Inventors: Kai Kwong Lau, Robert Kelly Cadena, John Elliott Whitecar
  • Patent number: 6725110
    Abstract: A digital audio decoder decodes or expands compressed data such as bit stream data, which are compressed based on the MPEG/Audio standard. Inverse quantization circuits perform inverse quantization on plural bit stream data, which are supplied thereto in connection with multiple channels respectively, thus producing inversely quantized data with respect to a prescribed number (e.g., thirty two) of sub-band samples respectively. The inversely quantized data are combined together among the multiple channels with respect to the prescribed number of the sub-band samples respectively. Then, a filter bank synthesizes together combined data corresponding to all of the sub-band samples, thus reproducing original digital audio signals. Multipliers are provided for use in gain control on the inversely quantized data with respect to the sub-band samples respectively.
    Type: Grant
    Filed: May 23, 2001
    Date of Patent: April 20, 2004
    Assignee: Yamaha Corporation
    Inventor: Toshihiko Suzuki
  • Publication number: 20040071300
    Abstract: A system for sharing wavelet domain components among encoded signals receives a set of signals decomposed and encoded according to a wavelet transform. The decomposed and encoded signals each include a set of wavelet coefficients at each level of the decomposition of the encoded signal. Using a vector quantization technique, the system identifies one or more sets of wavelet coefficients that are sharable among two or more of the decomposed and encoded signals at a particular level of decomposition. The system then stores the sets of wavelet coefficients of the decomposed and encoded signals. Each identified sharable set of wavelet coefficients at a particular level of decomposition is stored only once and shared by two or more of the decomposed and encoded signals.
    Type: Application
    Filed: October 10, 2002
    Publication date: April 15, 2004
    Applicant: Texas Instruments Incorporated
    Inventors: Daniel L. Zelazo, Steven D. Trautmann
  • Patent number: 6714093
    Abstract: In a dynamics compressor, an analog signal to be compressed is supplied to a first amplifier stage as an input signal, and the output signal thereof is supplied to the next amplifier stage as an input signal, and so on for a number of successive amplifier stages. Each amplifier stage amplifies its input signal with a stage amplification until it reaches a stage limit level. The analog signal and the respective output signals of the amplifier stages are summed in a summation stage to form a sum signal. Above a minimal level, the magnitude of the curve of the sum signalroughly corresponds to an exponent characteristic of the magnitude of the analog signal.
    Type: Grant
    Filed: October 1, 2002
    Date of Patent: March 30, 2004
    Assignee: Siemens Aktiengesellschaft
    Inventors: Ralph Oppelt, Markus Vester
  • Publication number: 20040049305
    Abstract: A conventional audio/video transmission system including an audio/video transmitter apparatus and an audio/video receiver apparatus suffers from delay of video relative to audio resulting from reproducing audio from the audio signal output from the audio/video transmitter apparatus and reproducing video from the video signal output from the audio/video receiver apparatus. To solve this inconvenience, according to the invention, in an audio/video transmission system including an audio/video transmitter apparatus and an audio/video receiver apparatus, the audio/video transmitter apparatus additionally has an audio delay circuit that outputs the input analog audio signal with a predetermined delay time.
    Type: Application
    Filed: September 5, 2003
    Publication date: March 11, 2004
    Applicant: Sharp Kabushiki Kaisha
    Inventors: Hiroshi Takahashi, Takeshi Morimoto
  • Publication number: 20040022400
    Abstract: This invention generally relates to audio signal processing apparatus and methods for altering, and particularly increasing, the perceived level of bass frequencies in an audio signal. The apparatus comprises an audio input (202) to receive an audio input signal; a compressor (204) coupled to the audio input and having an output, to compress said audio input signal; a high-cut filter coupled to the output of said compressor to provide a filtered compressor output; and a combiner (206) to combine a signal from said compressor output with a signal from said audio input to provide a combined audio output; and wherein said compressor is configured to distort said audio input signal such that said distortion is perceivable as an increase in the level of bass in said combined audio output.
    Type: Application
    Filed: August 9, 2002
    Publication date: February 5, 2004
    Inventor: Anthony J. Magrath
  • Publication number: 20030231775
    Abstract: A method (200) and apparatus (100) for classifying a homogeneous audio segment are disclosed. The homogeneous audio comprises a sequence of audio samples (x(n)). The method (200) starts by forming a sequence of frames (701-704) along the sequence of audio samples (x(n)), each frame (701-704) comprising a plurality of the audio samples (x(n)). The homogeneous audio segment is next divided (206) into a plurality of audio clips (711-714), with each audio clip being associated with a plurality of the frames (701-704). The method (200) then extracts (208) at least one frame feature for each clip (711-714). A clip feature vector (f) is next extracted from frame features of frames associated with the audio clip (711-714). Finally the segment is classified based on a continuous function defining the distribution of the clip feature vectors (f).
    Type: Application
    Filed: May 28, 2003
    Publication date: December 18, 2003
    Applicant: CANON KABUSHIKI KAISHA
    Inventor: Timothy John Wark
  • Publication number: 20030182000
    Abstract: The present invention includes methods of preprocessing soundtracks adapted to hearing impairment, environmental noise or other factors and making the preprocessed soundtracks available as alternate soundtracks. Particular aspects of the present invention are described in the claims, specification and drawings.
    Type: Application
    Filed: March 22, 2002
    Publication date: September 25, 2003
    Applicant: Sound ID
    Inventors: Hannes Muesch, Brent W. Edwards, Sunil Puria
  • Patent number: 6618486
    Abstract: A feedback controller, embedded in an FM audio processor, that controls the “integrated multiplex power” to the requirements of ITU-R 412 (2.51) is disclosed. This regulation specifies the maximum power, which is the same as the power produced by a sinewave modulating the carrier ±19 kHz (where ±75 kHz is 100% peak modulation). The controller applies the square of the multiplex signal to an integrator. A constant is removed from the integrator; representing the maximum power threshold. The output of the integrator is sampled periodically with a sample-and-hold circuit. A second integrator receives the output of the sample-and-hold. The output of the second integrator circuit is added to the threshold setting of a compressor, which determines the average power output of the compressor. The output of the compressor is applied through a peak controller to the input of the multiplex coder, closing a feedback loop, which controls the integrated multiplex power to a preset threshold.
    Type: Grant
    Filed: May 2, 2001
    Date of Patent: September 9, 2003
    Inventor: Robert A. Orban
  • Publication number: 20030142840
    Abstract: A feedback controller, embedded in an FM audio processor, that controls the “integrated multiplex power” to the requirements of ITU-R 412 (2.51) is disclosed. This regulation specifies the maximum power, which is the same as the power produced by a sinewave modulating the carrier ±19 kHz (where ±75 kHz is 100% peak modulation).
    Type: Application
    Filed: May 2, 2001
    Publication date: July 31, 2003
    Inventor: Robert A. Orban