160results about "Manually operated tone/bandwidth control" patented technology

Systems and methods for optimizing speaker performance. The system includes a self-contained speaker unit that includes a speaker, an amplifier coupled to the speaker, and a processor coupled to the amplifier. The processor receives a first sound signal from a receiver and a second sound signal from a microphone, processes the first sound signal based on a plurality of parameters, outputs the processed sound signal to the speaker, and generates a video signal based on the second sound signal. A wireless remote control allows a user to manipulate the parameters. The processor generates a test sound signal and outputs it to the receiver. The receiver processes the test sound signal and returns it to the processor for output through the speaker. The video signal includes a graphical user interface having a frequency response graph of the second sound signal and an eight band equalizer.

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.

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.

The present technique relates to a sound processing apparatus, a method, and a program capable of alleviating degradation of the quality of sound in a case where the gain of a sound signal is amplified.

When equalizer processing for adjusting the gain of each frequency band of an input signal on the basis of a gain setting value is performed, an input signal is attenuated by an input attenuation amount derived from the gain setting value, and the equalizer processing is performed on the input signal attenuated. The amount of amplification of the gain of the input signal in the equalizer processing is estimated on the basis of the gain setting value and a weight coefficient of each frequency band derived from a generally-available music signal prepared in advance, and a difference of the estimation value and the input attenuation amount is calculated as a gain correction amount. Further, nonlinear amplification processing is performed on the input signal so as to actually amplify the input signal, which has been subjected to the equalizer processing, by a gain correction amount, and an output signal is obtained. The present technique can be applied to a sound processing apparatus.

An audio output apparatus has a measuring circuit which measures the levels of at least two sound signals, a sound level adjusting module (a sound level adjusting circuit and a gain control circuit) which adjusts a sound level so as to equal the levels of the sound signals based on the levels measured at the measuring circuit, and an array speaker unit (a delay circuit, a multiplier, an adder, an amplifier and a speaker unit) which emits sounds in accordance with the sound signals outputted from the sound level adjusting module in different directivities respectively.

A low noise amplifier including a variable gain amplifier stage configured to accept an input signal and to provide a load driving signal; a tunable bandpass filter connected as a load to the variable gain amplifier stage, wherein the bandpass filter includes a cross-coupled transistor pair, and at least one cross-coupled compensation transistor pair biased in a subthreshold region configured to add a transconductance component when the load driving signal is of a magnitude large enough to decreases a transconductance of the cross-coupled transistor pair; and, a controller circuit configured to tune the bandpass filter. The filter can be tuned in respect to the frequency and the quality factor Q.

The present invention provides a long time constant integrator circuit as part of an integrated circuit. The integrator circuit is fully integrated on chip with no external capacitive or resistive components for enhancing the circuit's time constant. It achieves a -3 dB cut-off frequency of 1.6 Hz. The circuit is realisable on a very small area of silicon being formed by a bipolar process using npn transistors, resistive and capacitive elements. The integrator circuit comprises a transconductance stage as an input to an operational amplifier. The circuit design is fully differential and employs realisable resistors and capacitors.

An audio processing apparatus (20) connected between an audio source (10) and an audio output (30) is provided. The audio processing apparatus is suitable for singing practice of users and includes a vocal removing unit (210) and a pitch change unit (220). The vocal removing unit is for removing vocal signals from audio signals of the audio source, and the pitch change unit is for changing pitch of the audio signals and outputting the audio signals to the audio output.

The invention relates to the technical field of audio processing, and particularly relates to a method for adjusting audio equalizing and an audio system for providing equalizing adjustment. The method comprises the steps that a gain adjusting unit receives a current frame audio signal amplified by a power amplifier EQ unit; the gain adjusting unit calculates the energy value of each band of the current frame audio signal according to the current frame audio signal and obtains gain adjusting parameters according to the energy value of each band of the current frame audio signal and the gain parameters corresponding to each band; and the gain adjusting unit transmits the gain adjusting parameters to a host EQ unit so that the host EQ unit is enabled to perform gain processing on the next frame audio signal according to the gain adjusting parameters. According to the method for adjusting audio equalizing and the audio system for providing equalizing adjustment, the problem in the prior art that the auditory requirements of the user cannot be met due to distortion of the audio signal caused by the fact that gain processing is performed on the audio signal through the host EQ unit and then the audio signal is amplified by the power amplifier EQ unit can be solved.

The present invention relates to an audio control device using a multi-screen. The audio control device has a touch screen on which a bass axis and a treble axis for the sound of audio equipment are displayed in a multidimensional fashion, and on which a point selected by a user for audio control is inputted. An audio control section collectively calculates a bass value and a treble value corresponding to the point selected for audio control. A control section controls a speaker based on a result calculated by the audio control section.

Apparatuses, methods and systems are provided for controlling volume of a media system. A remote control is configured to control volume of a media unit and an external device simultaneously or consecutively.

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.

The present invention relates to a bass enhancement system comprising a bass substitution filter with variable gain and variable band width, wherein said variable gain and variable bandwidth at least partly depends on characteristics of an input signal.

The present invention further relates to a bass enhancement system according to the above, further comprising a hearing compensation filter and a loudspeaker bass response compensation filter, wherein said input signal is coupled to said hearing compensation filter, the output of which is coupled to said loudspeaker bass response filter, the output of which is coupled to said bass substitution filter.

The present invention further relates to a method of enhancing bass perception comprising boosting a reproducible bass frequency band in accordance with characteristics of signal content in a substantially non-reproducible bass frequency band.

Methods and apparatus for adjusting a tonal quality of an audio signal include functional elements and/or steps, comprising producing a first intermediate signal containing a band of frequencies at a lower mid-range of the audio signal; producing a second intermediate signal containing a band of frequencies at an upper mid-range of the audio signal; combining the first intermediate signal, the second intermediate signal and the audio signal to produce a tonally adjusted version of the audio signal; and adjusting respective gains of the first and second intermediate signals to affect the tonal quality of the audio signal without substantially altering the tonal quality of the audio signal at frequencies between the lower mid-range and the upper mid-range.