Abstract: A headphone for use with a wireless audio device. The headphone determines whether the origin of an incoming transmission is a cell phone or a device such as a land line device or a computer. The headphone applies a different equalization pattern depending on whether the origin is a land line device or a computer or whether the origin is a cell phone. The headphone may measure the amplitude of the incoming transmission above a first threshold frequency, or below a second threshold frequency, or both to determine if the origin is a land line device or a computer or if the origin is a cell phone.

Abstract: Signal of a channel selected in accordance with a cue instruction is supplied to a cue bus. Cue signal processing section provided in the cue bus can perform one or more signal processing operations, selected from among a predetermined plurality of different signal processing operations, such as “Delay”, “Insert” and “Equalizer”. For each of a plurality of channel types, a set of setting information is stored which contains information for setting ON or OFF of individual ones of the plurality of different signal processing operations. The set of setting information corresponding to the type of the channel selected via the cue instruction is referenced, and one or more signal processing operations is determined in such a manner that each signal processing operation set in an ON state in the setting file in correspondence with the type of the selected channel is performed on the signal supplied to the cue bus.

Abstract: A mobile communication device and a method of setting tone color, which allow a user to set the tone color of received sound. Provided are a normal mode, which sets the equalizer using GCF standards stored in an internal memory or equalizer setting values selected by a provider, a country-specific mode, which uses country-specific setting, and a user mode, in which a user can set frequency-specific gains of the received sound, and one mode is selected from the provided mode, so that the tone color of the received sound can be adjusted according to the selection. Telephone speech quality can be optimized for user preference, network environments and language characteristics.

Abstract: A bass enhancement system can provide an enhanced bass effect for speakers, including relatively small speakers. The bass enhancement system can apply one or more bass enhancements to an input audio signal. For example, the bass enhancement system can exploit how the human ear processes overtones and harmonics of low-frequency sounds to create the perception that non-existent (or attenuated) low-frequency sounds are being emitted from a loudspeaker. The bass enhancement system can generate harmonics of at least some low-frequency fundamental frequencies in one embodiment. Playback of at least some harmonics of a low-frequency fundamental frequency can cause a listener to perceive the playback of the low-frequency fundamental frequency. Advantageously, in certain embodiments, the bass enhancement system can generate these harmonics without performing processing-intensive pitch-detection techniques or the like to identify the fundamental frequencies.

Abstract: A method and apparatus for generating vibration based on sound characteristics in a mobile terminal are provided. The method includes converting audio data into an audio signal upon generation of a sound play request; determining whether to generate vibration based on a sound volume of the audio signal; setting an actuator to be driven for the audio signal among at least two actuators based on frequency distribution characteristics of the audio signal if it is determined upon determining to generate the vibration; and driving the actuator being set for the audio signal when outputting the audio signal.

Abstract: Systems, methods, and apparatus to filter audio are disclosed. An example device includes first and second audio speakers having first audio characteristics, a third audio speaker having second audio characteristics, wherein the third speaker is positioned between the first and second audio speakers, a first audio filter to process an audio input signal to have a first frequency response including a first cutoff frequency, the first audio filter to output a first audio output signal to the first audio speaker, and a second audio filter to process the audio input signal to have a second frequency response to compensate for interference between the first and second frequency responses caused by a position of the first audio speaker relative to the second audio speaker.

Abstract: In some embodiments, techniques for audio processing may include receiving a first characterization of a first audio device, including a first audio parameterization that includes at least one parameter relating to characteristics of the first audio device; receiving a second characterization of a second audio device, including a second audio parameterization that includes at least one parameter relating to characteristics of the second audio device; determining a processing parameterization that includes a combination of a plurality of audio processing factors, wherein the audio processing factors include the first characterization and the second characterization; receiving a first electronic audio signal, as the input signal; generating a second electronic audio signal, as the output signal, wherein generating the second electronic audio signal includes processing the input signal using the processing parameterization; and providing the output signal via an electronic audio output interface.

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.

Abstract: A user is allowed to select an acoustic equalizer setting from among a plurality of predetermined acoustic equalizer settings. Upon receiving an indication of a selection of a given acoustic equalizer setting, a handheld telephony device processes voice call downlink audio, based on the given acoustic equalizer setting and drives an audio output device with the processed audio signal.

Abstract: A case for a handheld computing device is disclosed. The case includes a housing sized and dimensioned to receive a handheld computing device securely therein. A data port is configured to electrically connect to the data connector on the handheld computing device. A microprocessor is electrically connected to the data port. The microprocessor is configured and arranged to authorize audio stored on the handheld computing device to play through the data connector and connected data port. Audio filters are electrically connected to the data port. An audio output electrically connector to the audio filters.

Abstract: The present invention provides methods and systems for digitally processing audio signals in broadcasting and/or transmission applications. In particular, the present invention includes a pre-transmission processing module which is structured and configured to generate a partially processed signal. A transmitter is then structured to transmit or broadcast the partially processed signal to a receiver, where the signal is then fed to a post-transmission processing module. The post-transmission processing module is structured and configured to further processes the signal based upon, for example, the listening environment, profile(s), etc. and generate a final output signal.

Abstract: The present invention provides for methods and systems for digitally processing an audio signal. Specifically, the present invention provides for a speaker system that is configured to digitally process an audio signal in a manner such that studio-quality sound that can be reproduced.

Abstract: Disclosed herein is a recording apparatus including: an audio signal correction block configured to execute correction for flattening the frequency characteristic of an audio signal supplied from a microphone and/or correction of the level of the audio signal; a correction control block configured to make the audio signal correction block adjust the level of the audio signal attenuate the reference value with time, and, if an audio signal with a level thereof exceeding the reference value is entered, use the absolute value of the level of the audio signal exceeding the reference value as a new reference value; and a recording block configured to record the audio signal to a recording media.

Abstract: A noise suppression system reduces low-frequency noise in a speech signal using linear predictive coefficients in an adaptive filter. A digital filter may update or adapt a limited set of linear predictive coefficients on a sample-by-sample basis. The linear predictive coefficients may be used to provide an error signal based on a difference between the speech signal and a delayed speech signal. The error signal represents an enhanced speech signal having attenuated and normalized low-frequency noise components.

Abstract: A method and device of channel equalization and beam controlling for a digital speaker array system includes (1) converting digital format; (2) performing channel equalization; (3) controlling beam-forming; (4) performing multi-bit ?-? modulation; (5) performing thermometer code conversion; (6) performing dynamic mismatch-shaping processing; and (7) extracting the channel information to send to the digital power amplifier and drive the array sound. A device includes a sound source, a digital converter, a channel equalizer, a beam-former, a ?-? modulator, a thermometer coder, a dynamic mismatch shaper, an extraction selector, a multi-channel digital power amplifier and a speaker array. Each unit connects to each other serially.

Abstract: The present invention relates to a method for controlling one or more loudspeakers provided in an enclosure, such as a listening room or an automobile cabin, the method comprising the steps of: (i) providing said one or more loudspeakers (2, 3, 4) with an audio input signal (5) whereby a sound field (10) is generated in the enclosure (1), and determining the corresponding acoustic power output APO(f) emitted from the one or more loudspeakers (2, 3, 4) into said enclosure (1); (ii) determining an acoustic contribution or room gain RG(f) of the enclosure (1) to the generated sound field (10); (iii) optionally determining a listening position interface LPI(f) that characterises a listener's ability to receive sound energy from a sound field at the specific place in the sound Field, in which he is located; and (iv) determining a filter characteristic as a function of the acoustic power output, the acoustic contribution or room gain RG(f) of the enclosure to the sound field in the enclosure and optionally the list

Abstract: In certain embodiments, an improved audio equalization filter can be generated by frequency warping one or more digital filters having a plurality of frequency bands. Frequency warping can include, for example, transforming at least some of the frequency bands of the one or more digital filters into lower frequency bands. As a result, in various implementations the audio equalization filter may be more accurate than certain currently-available IIR equalization filters. The audio equalization filter may also be more computing-resource efficient than certain currently-available FIR equalization filters.

Abstract: Disclosed herein is an audio outputting device for switching a plurality of processes to perform a process on an audio signal, and acoustically reproducing and outputting the audio signal, the audio outputting device including, a control section for, when changing a process performed on an audio signal from one process to another process, stopping the one process on the audio signal, outputting sound based on the audio signal unprocessed by either of the one process and the other process, and performing the other process on the audio signal after passage of a predetermined period of time.

Abstract: A signal processing apparatus is configured to change volume level or frequency characteristics of an input signal with a limited bandwidth in a first frequency range. The apparatus includes: an information extracting unit configured to extract second frequency characteristic information from a collection signal with a limited bandwidth in a second frequency range different from the first frequency range; a frequency characteristic information extending unit configured to estimate first frequency characteristic information from the second frequency characteristic information extracted by the information extracting unit, the first frequency characteristic information including the first frequency range; and a signal correcting unit configured to change volume level or frequency characteristics of the input signal according to the first frequency characteristic information obtained by the frequency characteristic information extending unit.

Abstract: In an embodiment, an oversampled data converter includes a lowpass filter having a filter stage comprising a dynamic limiter, where the dynamic limiter having a limit set by an signal level at an input to the oversampled data converter. The oversampled data converter also includes a quantizing block comprising an input coupled to an output of the lowpass filter and an output coupled to an input of the lowpass filter.

Abstract: A sound processing apparatus according to the present invention acquires a test signal for measuring a standing wave state emitted in a listening room, and determines a peak position or a dip position due to a standing wave based on frequency characteristics of the test signal. Next, the sound processing apparatus emits a burst signal corresponding to the frequency of the peak position or the dip position, and acquires this signal. The sound processing apparatus calculates an increment ?P of the acquired signal, which indicates an amount of increase of a peak in the trailing edge portion corresponding to the end position of the burst signal relative to a peak in the portion corresponding to the stationary portion of the burst signal, and attenuates the frequency of the above peak position or dip position of a sound signal to be output by an attenuation depending on ?P.

Abstract: The present invention is an acoustic traveling wave tube system for propagating a directional acoustic wave comprising an acoustic traveling wave tube having a cylindrical shape with a load on one end of the tube, a plurality of excitation rings positioned around a circumference of the tube and spaced at predetermined intervals along a length of the tube and a microprocessor having a database containing a plurality of waveforms representative of acoustic signals. The microprocessor energizes one of the plurality of excitation rings to form an acoustic wave, sequentially energizes one or more of the remaining excitation rings along the length of the tube to amplify the acoustic wave as the acoustic wave travels along the length of the tube, and propagates the acoustic wave from an end of the tube opposite the load as a shaped directional acoustic wave.

Abstract: A method and accompanying system are disclosed for tuning each channel of a high-speed SerDes link interface arranged in a configuration linking a local side to a remote side. The method includes transmitting a flow control packets from the local side to the remote side to change remote side transmission characteristics in a link channel; monitoring the bit error rate (BER) in the link channel; transferring additional flow control packets to adjust the remote side transmission characteristics; and processing the BER data at the local side to generate the remote side transmission characteristics for the link channel.

Abstract: A system and method for controlling a sound level of a mobile audio device are disclosed herein. In accordance with at least some embodiments, a system includes a transducer, a phase estimator, and a sound level control. The transducer converts an electrical signal applied to the transducer into audible sound. The phase estimator estimates a phase difference between a voltage and a current of the electrical signal applied to the transducer. The sound level control controls the loudness of sound produced by the transducer based, at least in part on the estimated phase difference.

Abstract: A system and method provide at least a single stage optimization process which maximizes the flatness of the net subwoofer and satellite speaker response in and around a cross-over region. A first stage determines an optimal cross-over frequency by minimizing an objective function in a region around the cross-over frequency. Such objective function measures the variation of the magnitude response in the cross-over region. An optional second stage applies all-pass filtering to reduce incoherent addition of signals from different speakers in the cross-over region. The all-pass filters are preferably included in signal processing for the satellite speakers, and provide a frequency dependent phase adjustment to reduce incoherency between the center and left and right speakers and the subwoofer. The all-pass filters are derived using a recursive adaptive algorithm.

Abstract: A method for determining coefficients of a family of cascaded second order Infinite Impulse Response (IIR) parametric filters used for equalizing a room response. The method includes determining parameters of each IIR parametric filter from poles or roots of a reasonably high-order Linear Predictive Coding (LPC) model. The LPC model is able to accurately model the low-frequency room response modes providing better equalization of loudspeaker and room acoustics, particularly at the low frequencies. Advantages of the method include fast and efficient computation of the LPC model using a Levinson-Durbin recursion to solve the normal equations that arise from the least squares formulation. Due to possible band interactions between the cascaded IIR parametric filters, the method further includes optimizing the Q value of each filter to better equalize the room response.

Abstract: A method of optimizing the low frequency audio response emanating from a pair of low frequency transducers housed within a cabinet. The low frequency transducers are electrically connected to a power amplifier and source of audio content. The resonant frequency (Fs) and amplitude (Q) are characterized as to the high-pass pole of the low frequency transducers as they are mounted within the cabinet. An equalizer is placed between the amplifier and source of audio content for canceling the complex pole of the low frequency transducers and for establishing a new complex pole at a cut off frequency below which the sound generated by the low frequency transducers will diminish.

Abstract: A frequency equalization system is provided that substantially equalizes the room frequency responses generated by at least one loudspeaker within a listening area so that the frequency responses in the listening area are substantially constant and flat within a desired frequency range with minimum signal latency. The frequency equalization system may use multiple microphones to measure the audio signals of one or more subwoofers to achieve an improved bass response that is flat across the relevant frequency range. The system employs an algorithm that is a closed-form, non-iterative, mathematical solution and features short computation time with reduced delays.

Abstract: Disclosed herein is a sound outputting apparatus, including: an electro-acoustic conversion section disposed in a housing and configured to acoustically reproduce a first sound signal; a sound collection section configured to collect sound outside said housing and output a second sound signal; a surrounding noise evaluation section configured to evaluate surrounding noise outside said housing based on the second electric signal; and a control section configured to perform predetermined control based on a result of the evaluation of said surrounding noise evaluation section.

Abstract: Methods and apparatus are provided for adaptive link partner transmitter equalization. According to one aspect of the invention, a local transceiver adapts one or more equalization parameters of a link partner by receiving a training frame over a channel between the link partner and the local transceiver, wherein the training frame is comprised of a predefined training pattern; adjusting one or more of the equalization parameters of the link partner; and determining whether the equalization of the channel satisfies one or more predefined criteria based on whether the predefined training pattern is properly received by the local transceiver. The predefined training pattern can be a pseudo random pattern, such as a PN11 pattern Noise margins and jitters margins for the channel can optionally be improved.

Abstract: A method of adjusting a loudness of an audio signal may include receiving an electronic audio signal and using one or more processors to process at least one channel of the audio signal to determine a loudness of a portion of the audio signal. This processing may include processing the channel with a plurality of approximation filters that can approximate a plurality of auditory filters that further approximate a human hearing system. In addition, the method may include computing at least one gain based at least in part on the determined loudness to cause a loudness of the audio signal to remain substantially constant for a period of time. Moreover, the method may include applying the gain to the electronic audio signal.

Abstract: A general model is provided for predicting a loudspeaker preference rating, where the model's predicted loudspeaker preference rating is calculated based upon the sum of a plurality of weighted independent variables that statistically quantify amplitude deviations in a loudspeaker frequency response. The independent variables selected may be independent variables determined as maximizing the ability of a loudspeaker preference variable to predict a loudspeaker preference rating. A multiple regression analysis is performed to determine respective weights for the selected independent variables. The weighted independent variables are arranged into a linear relationship on which the loudspeaker preference variable depends.

Abstract: A screen displayed in response to the selection of a tag button is called a tag screen and groups of the tag screens corresponding to the tag buttons are called screen groups. One of the screen groups is assigned to a home button as a home screen group, and clicking this home button when an arbitrary screen is displayed causes a large display to display a setting screen belonging to the home screen group. Here, in each of the screen groups, the selected tag button, the position of a mouse cursor, and so on are saved, and when an original screen is switched to another screen during an adjustment work or the like and the original screen is thereafter displayed again, the adjustment work or the like that was executed in the original screen can be continued at once.

Abstract: A method of operating a loudspeaker includes providing a digital audio signal and identifying a target transfer function to be applied to the signal. At least one coefficient of an FIR filter is generated. The generating includes performing Heyser spiral curve fitting, and fitting a three-dimensional curve based on a magnitude and phase of a target transfer function. The digital audio signal is filtered through the FIR filter. The filtered signal is inputted into the loudspeaker.

Abstract: In an equalizer system with a plurality of equalizers, the variation and setting of various equalizer parameters (gains, center frequencies, Q factors) may be automated by an adjustment circuit that may utilize a procedure, for example, a nonlinear curve fitting. Upon each variation, a figure of merit (e.g., cost function) may be formed and the parameter variation may continue in the direction in which the cost function decreases until it reaches a predetermined level. After each new setting of the equalizers, the procedure can be performed and the desired parameter set can be passed on to an equalizer implementing routine. External control signals may be provided to the adjustment circuit which then generates corresponding internal control signals for setting the gains of the equalizers at the respective center frequencies, the internal control signals being modified relative to the associated external control signals in such a way that the interferences occurring between the equalizers are at least reduced.

Abstract: In a user terminal, a reproducing unit reads out from a music file memory unit a music file selected and input by a user and reproduces the music file. In a normal reproduction, an output unit outputs the reproduced sound as an audio. When a notification information reception unit receives information that needs to be noticed such as an e-mail or a phone call, a notification information analysis unit analyzes the setting of the information or notification sound and determines the degree of importance. A frequency band allocation unit selects from allocation patterns of frequency bands stored in an allocation information memory unit a pattern for the music and for the notification sound according to the degree of importance and the frequency band of the notification sound. An audio processing unit extracts and synthesizes the components of the frequency bands allocated to music and the notification sound, and the output unit outputs accordingly.

Abstract: The present invention relates to the field of communication devices, e.g. wireless communication devices. More particularly, the present invention relates to the field of signal equalization, especially minimum mean square error equalization. The present invention especially relates to an equalizer for a communication device, a method of equalizing one or more received signals and a software program product for carrying out the method. The present invention reduces the size of a look-up table needed for a division operation and, generally, provides for a reduced complexity of the equalizer and receiver.

Abstract: Disclosed herein are apparatus, method, and computer program product whereby a device receives an acoustic signal, where the acoustic signal has a variable sound pressure level. A device outputs an electrical signal from an input audio transducer, where the input audio transducer is connected to a supply voltage. The device determines a distortion level of the electrical signal; and increases or decreases the supply voltage based on the distortion level.

Abstract: The present invention provides methods and systems for digitally processing audio signals. Some embodiments receive an audio signal and converting it to a digital signal. The gain of the digital signal may be adjusted a first time, using a digital processing device located between a receiver and a driver circuit. The adjusted signal can be filtered with a first low shelf filter. The systems and methods may compress the filtered signal with a first compressor, process the signal with a graphic equalizer, and compress the processed signal with a second compressor. The gain of the compressed signal can be adjusted a second time. These may be done using the digital processing device. The signal may then be output through an amplifier and driver circuit to drive a personal audio listening device. In some embodiments, the systems and methods described herein may be part of the personal audio listening device.

Abstract: A processing apparatus which is suitable for signal communication with a system having at least one of an input module and an output module. The processing apparatus can be configured to receive input signals from the input module of the system. The processing apparatus includes a first channel processing portion to which the input signals are communicable. The first channel processing portion can be configurable to receive and process the input signals in a manner such that bass frequency audio signals are extracted from the input signals. The bass frequency audio signals can be further processed via at least one of linear dynamic range processing, manipulation of dynamic range via compression and manipulation of dynamic range via expansion.

Abstract: A signal processing apparatus includes a signal output unit for outputting a measurement signal, the measurement signal being produced by synthesizing a signal composed of a concatenation of 2d period signals with a sinusoidal signal, each period signal having a time-domain waveform period being 2n samples, the sinusoidal wave having a wave count within the concatenation period of 2d period signals being other than an integer multiple of 2d, and n and d being respectively natural numbers, and an analyzing unit for frequency analyzing a response signal obtained as a result of picking up the measurement signal output from the signal output unit.

Abstract: A system is provided for configuring an audio system for a given space. The system may statistically analyze potential configurations of the audio system to configure the audio system. The potential configurations may include positions of the loudspeakers, numbers of loudspeakers, types of loudspeakers, listening positions, correction factors, filters, or any combination thereof. The statistical analysis may indicate at least one metric of the potential configuration including indicating consistency of predicted transfer functions, flatness of the predicted transfer functions, differences in overall sound pressure level from seat to seat for the predicted transfer functions, efficiency of the predicted transfer functions, or the output of predicted transfer functions.

Abstract: Sound quality is enhanced in a sound system including handsets and headsets. Handset sound enhancing algorithms are implemented in a handset. The handset automatically determines which, if any, of a plurality of headset sound enhancing algorithms are active in a headset in communication with the handset. The handset determines how to use the handset sound enhancing algorithms in a sound processing channel based on which of the headset sound enhancing algorithms are active in the headset.

Abstract: A system quantifies listener envelopment in a loudspeakers-room environment. The system includes a binaural detector that receives frequency modulated audible noise signals from multiple loudspeakers. The binaural detector generates detected signals that are analyzed to determine an objective listener envelopment. The envelopment is based on binaural activity of one or more sub-bands of the detected signal.

Abstract: The present invention relates to a receiver apparatus and method of channel estimation in a telecommunication system which provides at least two pilot sequences, and to a computer program product. Channel estimation is achieved by estimating channel taps separately for each of the at least two pilot sequences in every transmission block, and for applying estimated channel taps obtained from the estimation to at least one of a temporal and spatial filtering or combining operation to refine the channel estimate. Accordingly, temporal correlations and cross-correlations of the at least two pilot sequences are exploited without requiring knowledge of path delays and beamforming parameters.

Abstract: A dynamic bass equalization circuit has an amplitude dependent gain that is dependent upon the audio electrical signal amplitude and a dynamically adjusted frequency response that varies with the amplitude dependent gain. In one implementation, the dynamic bass equalization circuit includes a Sallen-Key high pass filter that includes an amplifier with a negative feedback path. The dynamically adjusted frequency response is provided by a parallel pair of reversed diodes connected in the negative feedback path.

Abstract: The invention provides a method and device for enhancing the listening qualities of an audio file by providing the listener with a plurality of modified equalized audio files. Each modified equalized audio file having a consistent loudness level but different audio characteristics. Hence, for an input audio file the current invention allows the listener to individually select the best audio characteristics for them to listen to the content of the input audio file according to their particular requirements without them needing to adjust the loudness level in playback. The invention further enables the listener to switch between the multiple equalized audio files during playback. The invention further includes a SN detector and reducer to eliminate the adverse effects of the presence of sudden, strong noise in the input audio file in the process of generating the plurality of modified equalized audio files.

Abstract: An apparatus for providing real-time calibration for two or more microphones. A calibrator for receiving a left microphone signal and a right microphone signal and generating phase difference data. A phase and amplitude correction system for receiving one of the left microphone signal or the right microphone signal the phase difference data and generating calibration data for a beamformer. The beamformer receiving the calibration data, the left microphone signal and the right microphone signal and generating a monaural beamformed signal.

Abstract: A method for clipping and post-clipping processing an audio signal, includes clipping an audio signal to provide a clipped audio signal; filtering, by a first filter, the audio signal to provide a filtered unclipped audio signal; and filtering, by a second filter, the clipped audio signal to provide a filtered clipped audio signal. The method further includes differentially combining the filtered clipped audio signal and the clipped audio signal to provide a differentially combined audio signal; and combining the filtered unclipped audio signal and the differentially combined audio signal to provide an output signal.

Abstract: The present invention relates to a method and corresponding system for predicting the perceived spatial quality of sound processing and reproducing equipment. According to the invention a device to be tested, a so-called device under test (DUT), is subjected to one or more test signals and the response of the device under test is provided to one or more means for deriving metrics, i.e. a higher-level representation of the raw data obtained from the device under test. The derived one or more metrics is/are provided to suitable predictor means that “translates” the objective measure provided by the one or more metrics to a predicted perceived spatial quality. To this end said predictor means is calibrated using listening tests carried out on real listeners. By means of the invention there is thus provided an “instrument” that can replace expensive and time consuming listening tests for instance during development of various audio processing or reproduction systems or methods.