279results about "Digital signal tone/bandwidth control" patented technology

A prefilter (5) for an audio system comprising a loudspeaker (1) in a room (2), which corrects both amplitude and phase errors due to the loudspeaker (1) by a linear phase correction filter response and corrects the amplitude response of the room (2) whilst introducing the minimum possible amount of extra phase distortion by employing a minimum phase correction filter stage. A test signal generator (8) generates a signal comprising a periodic frequency sweep with a greater phase repetition period than the frequency repetition period. A microphone (7) positioned at various points in the room (2) measures the audio signal processed by the room (2) and loudspeaker (1), and a coefficient calculator (6) (e.g. a digital signal processor device) derives the signal response of the room and thereby a requisite minimum phase correction to be cascaded with the linear phase correction already calculated for the loudspeaker (1). Filter (5) may comprise the same digital signal processor as the coefficient calculator (6). Applications in high fidelity audio reproduction, and in car stereo reproduction.

A technique for enhancing audio signals generated from compressed digital audio files is described. The technique uses a Bass Maximizer module, a Harmonic Exciter module and a Quasi Stereo module. The Bass Exciter module enhances the intensity, depth and punch of the bass audio content by creating harmonic sequences from low frequency components contained in the original input signal. The Harmonic Exciter module adds to the treble audio content of the original input signal by generating harmonic series from the high frequency components contained in the input signal. The Quasi Stereo Module creates a stereo image of the enhanced input signal by adding and subtracting delayed and filtered versions of the enhanced input signal with itself to create left and right channeled stereo-like outputs. The technique provides a useful tool to regenerate from an audio signal more pleasant and joyful sounds.

A computer readable memory to direct a computer to improve the perceived audio quality of a speaker included in that computer. The computer readable memory stores a first, second and third set of instructions. The first set of instructions causes the computer to determine the speaker type. The second set of instructions causes the computer to select a set of default filter coefficients for a digital filter based upon the speaker type. Finally, the third set of instructions causes the computer to realize a digital parametric equalizer using a digital filter and the set of default filter coefficients. Thus, the digital filter alters the audio signal that is input to the speaker, thereby improving the perceived quality of the speaker.

Embodiments disclosed herein enable detection and improvement of the quality of the audio signal using a mobile device by determining the loss in the audio signal and enhancing audio by streaming the remainder portion of audio. Embodiments disclosed herein enable an improvement in the sound quality rendered by rendering devices by emitting an test audio signal from the source device, measuring the test audio signal using microphones, detecting variation in the frequency response, loudness and timing characteristics using impulse responses and correcting for them. Embodiments disclosed herein also compensate for the noise in the acoustic space by determining the reverberation and ambient noise levels and their frequency characteristics and changing the digital filters and volumes of the source signal to compensate for the varying noise levels.

An equalizer bank with interference reduction is disclosed. The equalizer bank with interference reduction comprises a front-shelving filter, a plurality of peak filters, a rear-shelving filter and a plurality of compensators. The center frequency and the quality factor of each compensator are designed intentionally to be identical to those of the corresponding peak filter. When a user selects a specific gain level, the corresponding parameters that determine the peak filter and the compensator are retrieved directly and a complex calculation is skipped. Therefore, an audio signal can be equalized more efficiently with lower hardware cost.

A sound compensation system alters an electrical audio signal for input to a sonic reproduction device having associated behavioral characteristics. The behavior characteristics of the device are defined by individual or groups of individual components of the sonic reproduction device and include mechanical, acoustic and electromagnetic behaviors. The model includes a plurality of filters that simulate at least one of the behavior characteristics of the sonic reproduction device. The filters are defined by digital signal processes or by analog circuits and are characterized by one or more of an associated frequency, time, phase and transient response. These responses combine to define an overall response for the model. The filters include adjustable parameters which are used to alter filter responses to produce responses that are conjugates to the responses of the unaltered filters and thus the sonic reproduction device. A controller modifies the parameters.

Audio transducers (headphones, speakers, microphones) inherently do not accurately reproduce the signal presented to them at the input. This can be compensated for by taking into account the transducer characteristics and transforming the input signal using a digital signal processor (DSP) to counteract the inaccuracies. However, for the compensation to take place, the DSP needs to know the characteristics of the transducer. For systems with built-in transducers (like laptops with internal speakers) the characteristics of the internal speakers can be stored on the hard-drive of the laptop and the DSP can read this data and make the appropriate compensations. Because a transducer (headphone, speaker, microphone) has its own characteristics that need to be compensated for separately, a profile is supplied to the DSP either by a database lookup based on an identification made by the user or transducer itself or by profile information stored on the transducer. Once the characteristics of a transducer are known, many additional DSP algorithms can be applied in order to improve the audio performance and even safety of the system.

An equalizer control system is provided. The equalizer control system includes a first sound mode identifier to identify a first sound mode for controlling the equalizer from a plurality of sound modes, by classifying an audio feature value, extracted from a sequential audio stream, into a category selected from a plurality of categories, the category corresponding to the first sound mode, a second sound mode identifier to identify a second sound mode for controlling the equalizer, from the plurality of sound modes, by segmenting a sound source into a music genre using a highlight extracted from stored music, and an equalizer controller to analyze the first sound mode and the second sound mode, select one of the first sound mode and the second sound mode, and control the equalizer according to the selected sound mode.

A method of generating a desired shelving filter for audio data, the desired shelving filter having a low-corner frequency U and a high-corner frequency f₂, wherein the difference between the gain of the frequency response of the desired shelving filter at f₂ and the gain of the frequency response of the desired shelving filter at U is substantially equal to a non-zero desired amount, the method comprising: determining an order for a first shelving filter of a first predetermined type, wherein the low-corner frequency of the first shelving filter is U and the high-corner frequency of the first shelving filter is fΣ, determining, for a second shelving filter of a second predetermined type and of a predetermined order m, a frequency fz for the low-corner frequency of the second shelving filter and a frequency Z₄ for the high-corner frequency of the second shelving filter, where h is at least fi, f₄ is greater than f₃, and f₄ is at most fi, and forming the desired shelving filter as a filter combination of the first shelving filter and the second shelving filter; wherein fz, f₄ and the order of the first shelving filter are determined such that difference between the gain of the frequency response of the filter combination at fe and the gain of the frequency response of the filter combination at fi is substantially equal to the desired amount.

A method for use with equipment having filters, the method characterized by multiplying a matrix having a specified cut-boost setting of a filter of a graphic equalizer by a correction matrix to create a matrix having a corrected cut-boost setting of the filter of the graphic equalizer; adjusting an actual cut-boost setting of the filter of the graphic equalizer to be substantially equal to the corrected cut-boost setting of the filter of the graphic equalizer; and configuring the filter to have a Q value substantially equal to a linearizing Q value. In addition to the foregoing, other method embodiments are described in the claims, drawings, and text forming a part of the present application. In one or more various embodiments, related systems include but are not limited to circuitry and/or programming for effecting the foregoing-referenced method embodiments; the circuitry and/or programming can be virtually any combination of hardware, software, and/or firmware configured to effect the foregoing-referenced method embodiments depending upon the design choices of the system designer.

An audio compensation system (20) for producing a sound compensated output signal (S_o). The system comprises a volume control circuit (24) for producing a volume-adjusted signal (S_ic) by applying a volume adjustment to an audio signal in response to a volume setting, wherein the sound compensated output signal is responsive to the volume-adjusted signal. The system also comprises circuitry (26, 30, 32) for producing an amplified signal (S_ifa) by amplifying a selected bandwith of signals in response to the volume setting. The sound compensated output signal is also responsive to the amplified signal.

An integrated audio codec includes a high-pass filter to prevent damage to personal computer speakers and other components. The audio codec may be compliant with HD audio standards and can operate with generic software drivers. Tuning of the high-pass filter is provided through an external pin-out where either an external capacitor or external resistors provide an ability to tune the high-pass filter. In one implementation, a tuning voltage is digitized into a tuning code used by a digital high-pass filter. In addition, multiplexers can be used to insure only the audio path leading to the speakers is filtered.

The invention relates to a method for automated tuning of a sound system, the sound system comprising delay lines, equalizing filters, and at least two loudspeakers, the method comprising the steps of reproducing a useful sound signal through the loudspeakers, measuring sound pressure values at least one location, providing a target transfer function for tuning the delay lines and the equalizing filters of the sound system, the target transfer function representing a desired transfer characteristics of the sound system, adjusting the delay of the delay lines, and adjusting amplitude responses of the equalizing filters such, that the actual transfer characteristics of the sound system approximates the target function.

A device may include logic configured to detect information about a condition of a speaker in a communications device. The logic may process the information, load equalization logic based on the processed information, and may apply the equalization logic to an input signal of the speaker when the information indicates that a frequency response for the speaker should be changed.

An apparatus for processing an audio signal for reproduction by a sound transducer includes an equalization parameter determinator for determining a set of equalization parameters and an equalizer configured to equalize an input audio signal, to obtain an equalized audio signal. Different concepts for the determination of the set of equalization parameters include an image recognition, an evaluation of an identification signal which is provided by the sound transducer via an audio connection, and a measurement of the impedance of the sound transducer over frequency. Also, an upload functionality and a download functionality are provided.

The present disclosure provides a system for the synthesis of external sound of a vehicle, the system comprising a hazard analysis unit configured to detect a collision hazard between the vehicle and at least one further road user, in particular an outside traffic participant; at least one electroacoustic transducer; and a sound processing unit configured to generate an audio signal representing an, in particular vehicle dependent, synthetic noise signal and to control the at least one electroacoustic transducer to output a synthetic external sound based on the audio signal, wherein the sound processing unit is configured to modify the audio signal to enhance the perceptibility of the synthetic external sound by the further road user upon detection of the collision hazard.

A portable assistive listening system for enhancing sound for hearing impaired individuals includes a fully functional hearing aid and a separate handheld digital signal processing (DSP) device. The focus of the present invention is directed to the handheld DSP device and a unique method of processing incoming audio signals. The DSP device includes a programmable digital signal processor, a UWB transceiver for communicating with the hearing aid and/or other wireless audio sources, an LCD display, and a user input device (keypad). The handheld device is user programmable to apply different sound enhancement algorithms for enhancing sound signals received from the hearing aid and/or other audio source. The handheld device is capable of receiving audio signals from multiple sources, and gives the user control over selection of incoming sound sources and selective enhancement of sound. Specifically, the invention focuses on a proprietary method of adjusting the master volume and equalization of each of the multiple incoming audio signals prior to mixing all of the enhanced signals for output. Applying a Master Volume and Equalization setting prior to mixing provides for a more evenly enhanced sound and better overall sound intelligibility, as well as reducing processing power drained from the system during use.

According to one embodiment, during mixing of an N-channel input signal to generate an M-channel output signal, in at least one frequency sub-band, magnitude equalization is applied to the mixed channel signals such that an amplitude sum magnitude for the N input channels (e.g., the magnitude of a sum of estimated amplitudes of the N input channels) is approximately equal to an amplitude sum magnitude for the M output channels (e.g., the magnitude of a sum of estimated amplitudes of the M output channels). In one implementation, magnitude equalization is applied to one or more sub-bands (e.g., those below 1 kHz), and power equalization is applied to one or more other sub-bands (e.g., those above 1 kHz) to reduce coloration effects in the output signal.

An audio entertainment system and method for providing controlled equalization of sound from an external source is provided. The audio entertainment system includes a first audio device having an equalizer with first equalization settings and an output for providing an audio output signal. The system also includes an input for receiving audio signals from an external second audio device having second equalization settings. The second audio device inputs to the first audio device second equalization settings. The system further includes a controller for processing the equalization settings and controlling the audio output signal based on the equalization settings.