Abstract: A system and method are provided for capturing hearing characteristics from self-administered hearing tests, including loudness tolerance levels at different sound frequencies, as an individualized shaped auditory profile for automatically enhancing audio to complement and address as closely as possible an individual's hearing deficits experienced via a particular signal pathway, the signal pathway including the audio device used to administer the test, the sound environment, and the user's hearing capabilities. The user may self-administer a hearing test on a convenient personal apparatus, such as a smartphone. The shaped auditory profile may then be used to produce enhanced sound subsequently transmitted to the individual via the same signal pathway. In certain of its aspects, the invention is useful for any individual seeking an enhanced hearing experience, whether having hearing within normal range or hearing that is impaired.

Abstract: Apparatus for reproducing sound including at least three loudspeakers (5, 6, 7) mounted in a substantially sealed enclosure. The three loudspeakers may be mounted to a wall of the enclosure so that they are all directed away from, and evenly spaced around, a common point. Two speakers may be driven with the respective out of phase signals comprising the difference between two stereo channels and another speaker with the sum of those channels.

Abstract: Various embodiments relate to a system and method for decorrelating an audio signal with a hybrid filter. The hybrid filter is generated by first generating a decorrelation filter. A frequency-dependent warping is applied to the decorrelation filter. The warped decorrelation filter is then mixed with a carrier filter to generate the hybrid filter. The carrier filter may include filters for spatial processing of an audio signal, filters for upmixing an audio signal, and/or filters for downmixing an audio signal.

Abstract: A method of distinguishing sound sources includes the step of transforming data, collected by at least two transducers which each react to a characteristic of an acoustic wave, into signals for each transducer location. The transducers are separated by a distance of less than about 70 mm or greater than about 90 mm. The signals are separated into a plurality of frequency bands for each transducer location. For each band a comparison is made of the relationship of the magnitudes of the signals for the transducer locations with a threshold value. A relative gain change is caused between those frequency bands whose magnitude relationship falls on one side of the threshold value and those frequency bands whose magnitude relationship falls on the other side of the threshold value. As such, sound sources are discriminated from each other based, on their distance from the transducers.

Abstract: According to one embodiment, an acoustic characteristic correction coefficient calculation apparatus includes a frequency converter, a smoother, a frequency inverter, a cutter, and a calculator. The frequency converter is configured to convert a first impulse response corresponding to an input acoustic signal to a frequency domain. The smoother is configured to smooth amplitude and phase corresponding to the frequency domain converted by the frequency converter. The frequency inverter is configured to convert a frequency characteristic smoothed by the smoother to a time domain. The cutter is configured to cut out a second impulse response configured by the time domain obtained by converting the frequency characteristic by use of the frequency inverter by a preset tap number. The calculator is configured to calculate a correction coefficient used to correct an acoustic characteristic based on a result cut out by the cutter.

Abstract: A system and method are provided for processing sound signals. The processing may include identifying individual harmonic sounds represented in sound signals, determining sound parameters of harmonic sounds, classifying harmonic sounds according to source, and/or other processing. The processing may include transforming the sound signals (or portions thereof) into a space which expresses a transform coefficient as a function of frequency and chirp rate. This may facilitate leveraging of the fact that the individual harmonics of a single harmonic sound may have a common pitch velocity (which is related to the chirp rate) across all of its harmonics in order to distinguish an the harmonic sound from other sounds (harmonic and/or non-harmonic) and/or noise.

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: A signal processing apparatus includes: one or more detection means for detecting movement of a diaphragm of a speaker in correspondence with feedback methods that are different feedback methods; analog-to-digital conversion means for converting one or more detection signals acquired by the detection means into a digital form; feedback signal generating means for generating feedback signals corresponding to the feedback methods using the digital detection signals; synthesis means for combining an audio signal to be output as a driving signal of the speaker with the feedback signals; correction equalizer means for setting an equalizing characteristic to allow a sound reproduced by the speaker to have a target frequency characteristic by changing the digital audio signal; feedback operation setting means for setting feedback methods in which a feedback operation up to combining the audio signal with the feedback signal is performed and the feedback operation is not performed equalizing characteristic changing a

Abstract: The invention relates to modifying the loudness of an audio signal by measuring the weighted broadband level of the audio signal and modifying that weighted broadband level as a function of a spectral localization estimate of the audio signal.

Abstract: An audio signal processing device includes: a microphone configured to collect noise; an analyzing unit configured to analyze an audio signal collected by the microphone to detect the level and frequency property of the collected audio signal; and a signal processing unit configured to subject an audio signal to be reproduced to signal processing based on the analysis results of the analyzing unit.

Abstract: An audio circuit and associated method for enhanced intelligibility of audio content includes a first means for receiving reproduced audio content, a microphone for providing a microphone output signal in accordance with ambient noise, a second means for enabling the microphone output signal when the reproduced audio content is off, and disabling the microphone output signal when the reproduced audio content is on, and a signal processor, in communication with the first and second means. The signal processor applies a transfer function to the reproduced audio content for increasing gain to the reproduced audio content as a function of increasing amplitude of the microphone output signal, and decreasing gain to the reproduced audio content signal as a function of decreasing amplitude of the microphone output signal, and applies an equalization curve to the audio content to boost frequencies in a range that enhances consonant perception thus increasing speech intelligibility.

Abstract: Disclosed is a tuning sound (TS) feedback apparatus. The tuning sound (TS) feedback apparatus includes: a TS pick-up casing which is disposed and fixed inside or near the hearing organ of a sound engineer who manages tuning operations for sound output from a sound output terminal of a sound device having an equalizer; a sound/electric conversion element which is included in the TS pick-up casing, picks up the TS output from the sound output terminal of the sound device, and converts the picked-up TS into an electrical signal; and a TS output module which is included in the TS pick-up casing, converts the TS converted into the electrical signal by the sound/electric conversion element into wired or wireless communication data to be used for external communication, and transmits the data to the sound device.

Abstract: A parametric speaker comprises a generally planate radiating element, suitable for radiating ultrasonic vibrations into a fluid medium, and an emitter, having an ultrasonic output and/or resonant frequency, the emitter being intimately coupled to the radiating element. The radiating element is physically configured to have a mechanical resonance that substantially matches the output and/or resonant frequency of the emitter.

Abstract: Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include creating at least two iterative filters based on at least two separate frequency ranges of the original frequency response, calculating an error difference between the frequency response modified by the at least two iterative filters and the original frequency response, and applying the error difference to the audio signal.

Abstract: Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include identifying a target sub-region of the frequency response which has a predetermined area percentage of a total area under a curve generated by the frequency response, determining whether the target sub-region is a narrow energy region, creating a filter to adjust the frequency response, and applying the filter to the audio signal.

Abstract: A system is provided to protect a loudspeaker (144) by controlling a level of an applied audio signal. A control signal is generated by applying an input audio signal (115) to the collective operations of a gain control system (100). The gain control system (100) uses the input audio signal (115) in conjunction with at least one parameter to derive an estimated stress associated with the loudspeaker (144). The estimated stress is compared with a protection threshold stress (127). If the protection threshold stress is exceeded, a gain applied by a gain component (134) is selectively adjusted to modify the input audio signal (115). The resulting gain-controlled audio signal (116) is employed to drive the loudspeaker (144).

Abstract: Electronic devices and accessories for electronic devices such as headsets are provided. The electronic devices may produce audio output. The headsets may include earbuds with speakers that play the audio output for a user while the earbuds are located in the user's ears. Circuitry in an electronic device and a headset may be used in evaluating how well the earbuds are sealed to the user's ears. In response to seal quality measurements, informative messages can be generated for the user, overall earbud volume may be increased, balance adjustments may be made to correct for mismatched balance between left and right earbuds, equalization settings may be adjusted, and noise cancellation circuitry settings can be changed. Electrical impedance measurements and acoustic measurements can be used in evaluating seal quality.

Abstract: A reproducing apparatus includes: a digital signal processing block configured to execute first boost processing for setting a volume level for an entered digital audio signal and boosting an amplitude level of the signal; a digital-to-analog conversion block configured to convert the digital audio signal into an analog one; an analog signal processing block configured to execute second boost processing for boosting an amplitude level of the analog audio signal; an analog volume adjusting block configured to set a volume level for the analog audio signal from the analog signal processing block; a loudspeaker configured to output the analog audio signal from the analog volume adjusting block; an operating block configured to indicate a volume level of an audio signal from the loudspeaker and turn on/off the boost processing for the audio signals; and a control block configured to control components in accordance with an operation by the operating block.

Abstract: An audio signal processing apparatus and method for extracting components from an input signal, generating additional components, combining components, and level-controlling components. An audio signal processing apparatus may include a harmonic overtone adder and an equalizer. A harmonic overtone adder may include a high-pass filter, a low-pass filter, an harmonic overtone generator, and a combining unit. An equalizer may include a level detector and a gain controller.

Abstract: A system and method for enhancing audio, the method including receiving audio input tracks, with at least one audio input track including a restriction parameter, determining a restricted audio input track, where the restricted audio input track is the audio input track including the restriction parameter, manipulating another audio input track based on musical properties of the restricted audio input track, and combining the restricted audio input track and the manipulated audio input track into a single output audio track.

Abstract: A multiband limiter with selective sideband linking includes first and second frequency band splitters, a first and second plurality of limiters, first and second summers, and a plurality of selectable links coupling the first plurality of limiters to the second plurality of limiters. The first plurality of limiters each have a band input coupled to one of the first plurality of band outputs, a link port and a limiter output, and the first summer is receptive to the limiter outputs of the first plurality of limiters and has a first channel output. The second plurality of limiters each have a band input coupled to one of the second plurality of band outputs, a link port and a limiter output, and the second summer is receptive to the limiter outputs of the second plurality of limiters and has a second channel output.

Abstract: In order to generate a multi-channel signal having a number of output channels greater than a number of input channels, a mixer is used for upmixing the input signal to form at least a direct channel signal and at least an ambience channel signal. A speech detector is provided for detecting a section of the input signal, the direct channel signal or the ambience channel signal in which speech portions occur. Based on this detection, a signal modifier modifies the input signal or the ambience channel signal in order to attenuate speech portions in the ambience channel signal, whereas such speech portions in the direct channel signal are attenuated to a lesser extent or not at all. A loudspeaker signal outputter then maps the direct channel signals and the ambience channel signals to loudspeaker signals which are associated to a defined reproduction scheme, such as, for example, a 5.1 scheme.

Abstract: Scaling, by a desired amount sm, the overall perceived loudness Lm of a multichannel audio signal, wherein perceived loudness is a nonlinear function of signal power P, by scaling the perceived loudness of each individual channel Lc by an amount substantially equal to the desired amount of scaling of the overall perceived loudness of all channels sm subject to accuracy in calculations and the desired accuracy of the overall perceived loudness scaling sm. The perceived loudness of each individual channel may be scaled by changing the gain of each individual channel, wherein gain is a scaling of a channel's power. Optionally, in addition, the loudness scaling applied to each channel may be modified so as to reduce the difference between the actual overall loudness scaling and the desired amount of overall loudness scaling.

Abstract: A communication system has a controller for transmitting data to be played back by a plurality of playback devices corresponding to a plurality of channels, and a plurality of adapters required for executing playback by the playback devices. The controller has a setting unit which sets data to be played back by the playback devices and control information required to control playback of the data in time slots of a sync transmission frame, and a transmission unit which transmits the sync transmission frame the by the setting unit to the adapters. Each adapter has a reception unit which receives the transmitted sync transmission frame, and a playback control unit which extracts data corresponding to the channel to be played back by the adapter from the sync transmission frame, and controls the playback timing of the data based on control information corresponding to the data.

Abstract: A system and method for using an audiogram and audio filters to provide improved audio characteristics for hearing impaired listeners by equalizing the received sound level of the signal across the frequency spectrum. The system and method further provide for modifying an audio signal based on the personal hearing characteristics of a listener to compensate for hearing loss.

Abstract: A method estimates noise power spectral density (PSD) in an input sound signal to generate an output for noise reduction of the input sound signal. The method includes storing frames of a digitized version of the input signal, each frame having a predefined number N2 of samples corresponding to a frame length in time of L2=N2/sampling frequency. It further includes performing a time to frequency transformation, deriving a periodogram comprising an energy content |Y|2 from the corresponding spectrum Y, applying a gain function G(k,m)=f(?s2(km),?w2l (k,m?1), |Y(k,m)|2), to estimate a noise energy level |?|2 in each frequency sample, where ?s2 is the speech PSD and ?w2 the noise PSD. It further includes dividing spectra into a number of sub-bands, and providing a first estimate |{circumflex over (N)}|2 of the noise PSD level in a sub-band and a second, improved estimate |{circumflex over (N)}|2 of the noise PSD level in a subband by applying a bias compensation factor B to the first estimate.

Abstract: An automatic gain control derives the gain from the distance between the sound source and the microphone. The distance-based automatic gain control normalizes signal level changes caused by a speaker not maintaining a constant distance with respect to the microphone. Also, a proximity-effect compensation that derives the adaptive filter from the distance between the sound source and the microphone. The proximity-effect compensation corrects frequency response changes due to undesired proximity-effect variations. Determination of the distance between a sound source and a microphone permits accurate compensation for both frequency response changes and distance-related signal level changes.

Abstract: The present invention provides for methods and systems for digitally processing an audio signal to reproduce high quality sounds on various materials. In various embodiments, a method comprises filtering the signal with a low shelf filter and/or high shelf filter, passing the signal through a first compressor that, filtering the signal again with a low shelf filter and/or high shelf filter, processing the signal with a graphic equalizer based on a selected material profile, passing the signal through a second compressor, and outputting the signal to a transducer.

Abstract: A synthesis filter 106 synthesizes a plurality of wide-band speech signals by combining wide-band phoneme signals and sound source signals from a speech signal code book 105, and a distortion evaluation unit 107 selects one of the wide-band speech signals with a minimum waveform distortion with respect to an up-sampled narrow-band speech signal output from a sampling conversion unit 101. A first bandpass filter 103 extracts a frequency component outside a narrow-band of the wide-band speech signal and a band synthesis unit 104 combines it with the up-sampled narrow-band speech signal.

Abstract: An audio processor for converting a multi-channel audio input signal, such as a B-format sound field signal, into a set of audio output signals, such as a set of two or more audio output signals arranged for headphone reproduction or for playback over an array of loudspeakers. A filter bank splits each of the input channels into frequency bands. The input signal is decomposed into plane waves to determine one or two dominant sound source directions. The(se) are used to determine a set of virtual loudspeaker positions selected such that the dominant direction(s) coincide(s) with virtual loudspeaker positions. The input signal is decoded into virtual loudspeaker signals corresponding to each of the virtual loudspeaker positions, and the virtual loudspeaker signals are processed with transfer functions suitable to create the illusion of sound emanating from the directions of the virtual loudspeakers.

Abstract: The present invention(s) provides for methods and systems for digitally processing a ringtone. In various embodiments, a method comprises generating a profile comprising a plurality of filter equalizing coefficients, configuring a plurality of filters of a graphic equalizer using the plurality of filter equalizing coefficients from the profile, receiving a ringtone for processing, adjusting the plurality of filters using a first gain, equalizing the ringtone using the plurality of filters of the graphic equalizer, and outputting the equalized ringtone.

Abstract: A multiband dynamics compressor implements a solution for minimizing unwanted changes to the long-term frequency response. The solution essentially proposes undoing the multiband compression in a controlled manner using much slower smoothing times. In this regard, the compensation provided acts more like an equalizer than a compressor. What is applied is a very slowly time-varying, frequency-dependent post-gain (make-up gain) that attempts to restore the smoothed long-term level of each compressor band.

Abstract: For generating a low-frequency channel for a low-frequency loudspeaker arranged at a predetermined low-frequency loudspeaker position, a plurality of audio objects are initially provided, each audio object having an object position and an object description associated with it. Hereafter, a calculation of an audio object scaling value is performed for each audio object on the basis of the object description, so that an actual amplitude state at least comes close to a target amplitude state at a reference playback position. Thereafter, each object signal is scaled with an associated audio object scaling value so as to then sum the scaled object signals. From the composite signal obtained there, a low-frequency channel is subsequently derived for the low-frequency loudspeaker, and is provided to the respective low-frequency loudspeaker.

Abstract: Computer implemented methods for the control of properties in computer simulations (such as video games) based on bitmaps are disclosed. In these methods, a bitmap is used as a reference file for an Audio Zone, and the bitmap values indicate properties within the virtual scene. To compute the value of a property (such as audio loudness), the bitmap is sampled at coordinates that correspond to the virtual location of the observer. Properties of the bitmap, such as the brightness associated with a pixel in the bitmap, may be designated to correspond the loudness of an associated sound. Bitmap-based Audio Zones are intuitive to use for the scene designer. Programming code for accessing bitmaps is well known and robust, and can have a small computational footprint and reduced memory requirements. The method is therefore suitable for real-time simulations such as video games, and resource-limited devices such as hand-held computers or smartphones.

Abstract: A pseudo bass generating apparatus includes a first 4th-order LPF, an absolute value circuit, a clip circuit, a multiplier, a first adder for subtracting an output signal of the multiplier from an output signal of the clip circuit, a second adder for adding an output signal of the first adder and an output signal of the absolute value circuit, a second 4th-order LPF, and a third adder for adding the input signal and an output signal of the second 4th-order LPF.

Abstract: Using frequency characteristics determined for individual ones of a plurality of analyzed bands of a predetermined audio frequency range with frequency resolution that becomes finer in order of lowering frequencies of the analyzed bands, a synthesized band is set for each one or for each plurality of the analyzed bands, and then a time-axial response waveform is determined for each of the synthesized bands. The response waveforms of the synthesized bands are then added together to thereby provide a response waveform for the whole of the audio frequency range.

Abstract: Exemplary embodiments include a frequency modulation (FM) transmitter and a non-FM receiver, which may be implemented on the same IC chip. The FM transmitter may include a digital FM modulator, a lowpass filter, an amplifier, and an LC tank circuit. The digital FM modulator may receive a digital input signal, perform FM modulation with the digital input signal, and provide a digital FM signal. The lowpass filter may filter the digital FM signal and provide a filtered FM signal. The amplifier may amplify the filtered FM signal and provide an output FM signal. The LC tank circuit may filter the output FM signal. The digital FM modulator may perform FM modulation by changing a variable divider ratio of a multi-modulus divider within a PLL. A delta-sigma modulator may receive the digital input signal and generate a modulator output signal used to obtain the variable divider ratio.

Abstract: An improved filtering system for loudspeaker equalization is comprised of two filtering sections, a single pole filtering section having single pole filters for shaping the frequency response of the audio signal over the operative bandwidth of the equalized loudspeaker system, and a parametric filtering section that provides for localized parametric equalization for compensating for room or other environmental resonances. The filtering system of the invention can be used to create a wide variety of useful filter shapes, while minimizing phase distortion that can harm the transient response of phase-aligned loudspeakers of a loudspeaker system.

Abstract: Methods and apparatus automatically cancel or attenuate an unwanted signal (such as low frequencies from wind buffets) from, and/or control frequency response of, a condenser microphone, or control the effective condenser microphone sensitivity before the signal reaches an ASIC or other processing circuit. As a result, the maximum amplitude signal seen by the processing circuit is limited, thereby preventing overloading the input of the processing circuit. Remaining (wanted) frequencies can be appropriately amplified to reduce the noise burden on further processing circuits. A corrective signal is applied to a bias terminal of the condenser microphone to cancel the unwanted signal. Optionally or alternatively, a controllable impedance is connected to a line that carries the signal generated by the MEMS microphone, so as to attenuate unwanted portions of the signal.

Abstract: A method and apparatus for customizing audio signal processing for a user by a mobile device is provided. The method includes identifying hearing characteristics of the user by testing hearing abilities of the user, by the mobile device, at a plurality of frequencies; adjusting a dynamic range of each of the plurality of frequencies based on the hearing characteristics; processing a decoded audio signal based on the adjusted dynamic range of each of the plurality of frequencies; and outputting the processed audio signal.

Abstract: 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.

Abstract: A method, circuit and article of manufacture for restoring high-frequency content in a first signal includes down-sampling the first signal by a factor to give a second signal; up-sampling the second signal by the same factor to give a third signal; low-pass filtering the third signal to give a fourth signal; high-pass filtering the fourth signal to give a fifth signal; and, adding the fifth signal to the first signal.

Abstract: A spectral management system may be used in an audio system to receive and process audio signals having multiple distributed audio channels, such as a right, left, center, right side, left side, right rear and left rear channels. The spectral management system may separate and route a frequency range of audio content included in one or more of the distributed audio channels to other distributed audio channels. The separated and routed frequency range of audio content may be combined with audio content present on the other distributed audio channels to which the separated frequency range of audio content is routed. Separation, routing and combination may include bass audio content routing, mid-bass audio content routing, subwoofer audio content routing and treble audio content routing.

Abstract: A multimedia processing system is provided. The system comprises: a depth analyzing unit configured to receive an input image and retrieve a depth image according to the input image; and a audio processing unit configured to receive an input audio signal and the depth image, detect an audio object and position information corresponding to the audio object from the depth image, and retrieve an acoustic frequency range corresponding to the audio object from the input audio signal; wherein when the position information exceeds a predetermined range, the audio processing unit adjusts the acoustic frequency range of the input audio signal according to the position information to generate an output audio signal.

Abstract: A haptic conversion system is provided that intercepts audio data, such as a digital audio signal, analyzes the audio data in frequency, and divides the analyzed audio data into one or more audio frequency regions, where each audio frequency region includes one or more audio sub-signals. The haptic conversion system further maps the one or more audio frequency regions to one or more haptic frequency regions, where each haptic frequency region includes one or more haptic signals. The haptic conversion system further maps the one or more haptic effects to one or more actuators. The haptic conversion system further sends the one or more haptic signals to one or more actuators, in order to generate one or more haptic effects.

Abstract: An audio input signal is processed to produce a processed audio output signal. An audio input signal is received as an original signal. The audio input signal is dynamically filtered to produce a first stage signal consisting of a selected frequency band of the input signal. Gain applied to the first stage signal is dynamically controlled in response to a control signal to produce a second stage signal. The control signal is derived from the first stage signal. Processing the original signal in combination with the second stage signal to produce a processed audio output signal. Processing the original signal in combination with the second stage signal and the first stage signal to produce a processed audio output signal.

Abstract: A method or apparatus for delivering audio programming such as music to listeners may include identifying, capturing and applying a listener's audio perception characteristics (sometimes referred to as the listener's Personal Waveform Profile) to transform audio content so that the listener perceives the content similarly to how the content would be perceived by a different listener. An audio testing tool may be implemented as software application to identify and capture respective listeners' Personal Waveform Profiles. A signal processor may operate an algorithm processing source audio content using the respective listeners' Personal Waveform Profiles to provide audio output based on a difference between different profiles.

Abstract: Improved approaches to flexibly implementing graphic equalizers on media players are disclosed. These approaches provide dynamic order reduction of a multi-band graphic equalizer so that equalizer effects can be timely performed with only limited computational resources. In one embodiment, a media player receives a media item and associated equalizer settings for a multi-band graphic equalizer. The media player can then automatically (i.e., without user action) approximate the multi-band graphic equalizer with the equalizer settings for the media item using a fewer number of filters. Fewer filters means order reduction, and thus reduction in computational requirements. After the multi-band graphic equalizer is approximated, the media player can present the media item to its user in accordance with the reduced complexity, approximated equalizer.

Abstract: In many audio playback systems, frequencies below a given cut off frequency are suppressed either due to speaker constraints or safety constraints. For example, some speakers are only capable of generating signals above a certain frequency. Prolonged low frequency sound can cause damage to speakers or other components. An audio driver can be equipped with a phantom bass module which by doubling, tripling and/or quadrupling frequencies below a cutoff frequency can simulate the bass experience. The doubling, tripling and quadrupling methods disclosed provide a low complexity formulation of a frequency doubling, tripling and quadrupling. In addition, the frequency doubling, tripling and quadrupling formulations are easily adapted to multi-rate processing, where computational savings can be very high.

Abstract: A signal processing device includes: a sound adjustment amount calculation unit which calculates a sound adjustment amount for adjusting sound characteristics of each channel to a predetermined sound characteristic for each channel, using a sound signal that is obtained by collecting the outputs of each channel; an evaluation value calculation unit which calculates a coefficient allocation evaluation value for allocating a size of a filter coefficient necessary for the sound adjustment of the respective channels for each channel, based on the sound adjustment amount that is calculated by the sound adjustment amount calculation unit; and a filter coefficient calculation unit which calculates the filter coefficient for each channel using the coefficient allocation evaluation value that is calculated by the evaluation value calculation unit.