Abstract: The present invention relates to a device for controlling a loudspeaker (14) in an enclosure, comprising: an input for an audio signal (Saudio_ref) to be reproduced; an output for supplying an excitation signal from the loudspeaker. It comprises a control unit comprising: means (24, 25) for calculating a desired dynamic value (Aref) of the loudspeaker diaphragm based on the audio signal (Saudio_ref) to be reproduced and the structure of the enclosure; means (26) for calculating a plurality of desired dynamic values (Aref, dAref/dt, Vref, Xref) of the loudspeaker diaphragm at each moment based on only the desired dynamic value (Aref); a mechanical model (36) of the loudspeaker; and means (70, 80, 90) for calculating the excitation signal of the loudspeaker at each moment, without feedback loop, from the mechanical model (36) of the loudspeaker and desired dynamic values (Aref, dAref/dt, Vref, Xref).

Abstract: An estimate of a fractional chirp rate of a signal may be computed by using multiple frequency representations of the signal. A first frequency representation may be computed using a first fractional chirp rate and a first score may be computed using the first frequency representation that indicates a match between the first fractional chirp rate and a fractional chirp rate of the signal. A second frequency representation may be computed using a second fractional chirp rate and a second score may be computed using the second frequency representation that indicates a match between the second fractional chirp rate and the fractional chirp rate of the signal. The fractional chirp rate of the signal may be estimated using the first score and the second score, for example, by selecting a fractional chirp rate corresponding to a highest score.

Abstract: Systems and methods for handling silence in audio streams are disclosed. In one aspect, a transmitter detects a halt in an audio stream. After detection of the halt in the audio stream, the transmitter embeds a silence signal into the audio stream and transmits the silence signal to associated receivers. The associated receivers may respond to the embedded silence signal by “playing” silence or by using the silence signal to activate a silence protocol. In either event, the associated receivers do not receive the original audio halt and do not produce an unwanted audio artifact.

Abstract: A multi-band limiter, a sound recording apparatus, and a program are provided having a multi-band limit function with a flat frequency characteristic while removing a dead zone of a multi-band limiter. A cross-over filter unit divides an input sound signal into a low-band component, a middle-band component, and a high-band component so that the adjacent bands cross at ?6 dB.

Abstract: The present invention provides a vibration audio system for transmitting an audio signal outputted from a sound source to a listener in the form of vibration while reducing output level of the signal and power consumption. The system includes an envelope detection unit (204) for detecting an envelope signal of the audio signal outputted from a sound source, a vibration transmission member for allowing the listener to perceive vibration of a low-frequency sound outputted from a low-frequency output speaker that outputs audio signals, and a frequency conversion unit (205) for generating an audio signal frequency-converted on the basis of resonant frequencies by multiplying the envelope signal by sine waves having the same frequencies as resonance frequencies obtained from an impulse response of the low-frequency output speaker disposed in the vibration transmission member. The audio signal frequency-converted by the frequency conversion unit (205) is outputted from the low-frequency output speaker.

Abstract: Embodiments of systems and methods are described for reducing undesired leakage energy produced by a non-front-facing speaker in a multi-speaker system. For example, the multi-speaker system can include an array of forward-facing speakers, one or more upward-facing speakers, and/or one or more side-facing speakers. Filters coupled to any two of the speakers in the multi-speaker system can generate audio signals output by the coupled speakers to reduce, attenuate, or cancel a portion of an audio signal output by one or more non-front-facing speakers that acoustically propagates along a direct path from the respective non-front-facing speaker to a listening position in a listening area in front of the multi-speaker system.

Abstract: The invention relates to amplifiers of audio power which operate with loudspeaker systems. The main technical result is significant extension of the reproducible frequency range without reduction of the loudspeaker system fundamental resonance frequency. This result can be obtained by means of adding only one low power resistor and one low capacitance capacitor to a typical power amplifier. In the described embodiment this is achieved by means of a rise of the amplifier FR at low frequencies using positive feedback and the quasi-resonance effect.

Abstract: An unknown content audio signal that is input to an audio power amplifier and speaker that have a known gain and speaker sensitivity, is buffered. A portion of the buffered signal is provided to an automatic content recognition (ACR) system, and then a content identifier, of the unknown content signal, as provided by the ACR system is received. Previously determined metadata is also received, for a reference audio content signal that has been assigned to the content identifier. The metadata includes a previously determined reference measure of an audio signal characteristic for the reference content. An input measure of the audio signal characteristic is computed, for the unknown content, and compared with the reference measure. On that basis, and also based on the known gain and speaker sensitivity, at least one of a scalar gain and spectral shaping filtering that is being applied to the unknown content is adjusted. Other embodiments are also described and claimed.

Abstract: Techniques are disclosed for systems and methods to provide wind sensor motion compensation for wind sensors mounted to moving platforms. A wind sensor motion compensation system may include a wind sensor, a wind sensor accelerometer, one or more additional sensors, actuators, controllers, user interfaces, and/or other modules mounted to or in proximity to a vehicle. The wind sensor motion compensation system may be implemented with one or more logic devices adapted to receive sensor signals and determine a sensor-motion compensated wind velocity. The logic devices may be adapted to receive a wind sensor acceleration and a relative wind velocity from a wind sensor, determine a wind sensor velocity from the wind sensor acceleration, and determine a sensor-motion compensated relative wind velocity from the wind sensor velocity and the relative wind velocity.

Abstract: Provided are, among other things, systems, methods and techniques for processing an audio signal to add virtual bass. In one representative embodiment, an apparatus includes: (a) an input line that inputs an original audio signal in the time domain; (b) a bass extraction filter that extracts a bass portion of the original audio signal, which also is in the time domain; (c) an estimator that estimates a fundamental frequency of a bass sound within the bass portion; (d) a frequency translator that shifts the bass portion by a positive frequency increment that is an integer multiple of the fundamental frequency estimated by the estimator, thereby providing a virtual bass signal; (e) an adder having (i) inputs coupled to the original audio signal and to the virtual bass signal and (ii) an output; and (f) an audio output device coupled to the output of the adder.

Abstract: Provided are, among other things, systems, methods and techniques for processing an audio signal to add virtual bass. In one representative embodiment, an apparatus includes: (a) an input line that inputs an original audio signal; (b) a transform module that transforms the original audio signal into a set of frequency components; (c) a bass extraction filter that extracts a bass portion of such frequency components; (d) an estimator that estimates a fundamental frequency of a bass sound within such bass portion; (e) a frequency translator that shifts the bass portion by a frequency that is an integer multiple of the fundamental frequency estimated by the estimator, thereby providing a virtual bass signal; (f) an adder having (i) inputs coupled to the original audio signal and to the virtual bass signal and (ii) an output; and (g) an audio output device coupled to the output of the adder.

Abstract: Disclosed are a method and electronic device for voice recognition. The voice recognition method includes recognizing, in a first processor using low power mode, a voice signal inputted through a microphone, entering an active state and performing voice recording in a second processor if the recognized voice signal is a previously set keyword, and performing voice recognition in the second processor if the end of a voice input is determined during the voice recording.

Abstract: A voice signal processing apparatus and a voice signal processing method are provided. A first sampling point of an mth original frequency-lowered signal frame phase-matched to the sampling point corresponding to a phase reference sampling point number is determined according to the phase reference sampling point number of an (m?1)th original frequency-lowered signal frame corresponding to a middle sampling point of an (m?1)th renovating frequency-lowered signal frame. The q consecutive sampling points starting from the first sampling point are used as the sampling points of an mth renovating frequency-lowered signal frame.

Abstract: Example techniques may involve calibration with multiple recording devices. An implementation may include detecting, via a microphone, one or more calibration sounds as emitted by one or more playback devices of one or more zones during a calibration sequence. The implementation may further include determining a first response, the first response representing a response of a given environment to the one or more calibration sounds as detected by the first recording device and receiving data indicating a second response, the second response representing a response of the given environment to the one or more calibration sounds as detected by a second recording device. The implementation may also include determining a calibration for the one or more playback devices based on the first response and the second response and sending, to the one or more zones, an instruction that applies the calibration to playback by the one or more playback devices.

Abstract: Separation of speech and background from an audio mixture by using a speech example, generated from a source associated with a speech component in the audio mixture, to guide the separation process.

Abstract: An apparatus for processing a medical image includes: a receiver configured to receive a signal having a plurality of frequency bands; an image reconstructor configured to segment the signal into a first signal of a first frequency band and a second signal of a second frequency band based on a signal strength, and configured to generate a first reconstructed image of the first frequency band and a second reconstructed image of the second frequency band; and an image synthesizer configured to synthesize the first reconstructed image and the second reconstructed image.

Abstract: Technologies pertaining to calibration of filters of an audio system are described herein. A mobile computing device is configured to compute values for respective filters, such as equalizer filters, and transmit the values to a receiver device in the audio system. The receiver device causes audio to be emitted from a speaker based upon the values for the filters.

Abstract: A voice signal processing apparatus and a voice signal processing method are provided. A last sampling point of an mth original frequency-lowered signal frame is determined according to a phase reference sampling point number of the mth original frequency-lowered signal frame. Here, the phase reference sampling point number corresponds to a middle sampling point of an mth renovating frequency-lowered signal frame, and the last sampling point is phase-matched with a sampling point corresponding to the phase reference sampling point number in the mth original frequency-lowered signal frame. P consecutive sampling points starting from the last sampling point are applied as sampling points of an (m+1)th renovating frequency-lowered signal frame.

Abstract: One or more techniques and/or systems are provided for communicating between two or more drivers respectively controlling and/or managing different channels through which data is transferred between a wireless communication device and a computer system and/or between a controller of the computer system and an application of the computer system. Typically, at least one of the channels is configured to transmit time sensitive data (e.g., such as audio data) while another channel is configured to transmit time insensitive data (e.g., such as call control data). A device driver interface may be configured to provide a medium through which the two or more drivers can communicate. The techniques and/or systems find particular application with respect to Bluetooth headsets used in combination with a computer system comprising a system on chip architecture, but other applications are also contemplated.

Abstract: Devices related to digital MEMS loudspeakers are discussed. Such devices may include an air pressure source, MEMS valves coupled to the air pressure source, and an audio modulator coupled to the MEMS valve to receive an audio signal and to control the MEMS valves via a modulation signal to provide an acoustic output.

Abstract: A computing system automates the configuration and management of a live sound system that includes a processor and memory for building in a GUI of a display a representation of the live sound system for a venue. The system loads a venue template that includes loudspeaker arrays and related properties including a setup configuration of the loudspeaker arrays and tuning data for constituent loudspeakers that are operable to provide an audio coverage pattern for the venue. The system overlays on top of the representation of the loudspeaker arrays a wiring circuit representation indicating interconnections of the loudspeakers that define bandpass inputs for each array. The system generates a plurality of amplifiers in the representation to drive the arrays, and associates amplifier channels of the amplifiers with the bandpass inputs. The amplifier channels include representations of output channels of DSPs coupled with respective amplifier channels.

Abstract: A speech signal processing system is described for use with automatic speech recognition and hands free speech communication. A signal pre-processor module transforms an input microphone signal into corresponding speech component signals. A noise suppression module applies noise reduction to the speech component signals to generate noise reduced speech component signals. A speech reconstruction module produces corresponding synthesized speech component signals for distorted speech component signals. A signal combination block adaptively combines the noise reduced speech component signals and the synthesized speech component signals based on signal to noise conditions to generate enhanced speech component signals for automatic speech recognition and hands free speech communication.

Abstract: A mobile multi-function device that includes a speaker, two or more microphones, and a beamformer processor is described. The beamformer processor uses the microphones to perform beamforming operations. One of the microphones shares a receiver acoustic opening with the speaker while the other microphone uses a separate acoustic opening. The receiver acoustic opening may be an earpiece opening that is held to the ear of a user while conducting a phone call with the device and provides acoustic input and output paths for the microphone and the speaker, respectively.

Abstract: An apparatus for driving loudspeakers of a sound system is provided. The sound system has at least two loudspeakers of a basic system and at least three loudspeakers of a focus system, each of the loudspeakers having a position in an environment. The apparatus has a basic channel provider for providing basic system audio channels to drive the loudspeakers of the basic system, and a focused source renderer for providing focus system audio channels to drive the loudspeakers of the focus system. The focused source renderer is configured to calculate a plurality of delay values for the loudspeakers of the focus system based on the positions of the loudspeakers of the focus system and based on a position of a focus point, and to generate at least three focus group audio channels for at least some of the loudspeakers of the focus system based on the plurality of delay values and based on a focus audio base signal to provide the focus system audio channels.

Abstract: An audio processing apparatus and method are provided. The audio processing apparatus includes an envelope detector to detect an envelope of an input signal with respect to a low frequency band, and a signal restorer to restore the input signal including a high frequency band by performing frequency folding of frequency sub-band according the envelope.

Abstract: Method of improving voice quality using a wireless headset with untethered earbuds starts by receiving first acoustic signal from first microphone included in first untethered earbud and receiving second acoustic signal from second microphone included in second untethered earbud. First inertial sensor output is received from first inertial sensor included in first earbud and second inertial sensor output is received from second inertial sensor included in second earbud. First earbud processes first noise/wind level captured by first microphone, first acoustic signal and first inertial sensor output and second earbud processes second noise/wind level captured by second microphone, second acoustic signal, and second inertial sensor output. First and second noise/wind levels and first and second inertial sensor outputs are communicated between the earbuds. First earbud transmits first acoustic signal and first inertial sensor output when first noise and wind level is lower than second noise/wind level.

Abstract: An auditory prosthesis device for neural stimulation of a patient's hearing having an audio signal input device; a sound processor for generating a neural stimulation signal; and an implantable stimulation assembly having plural stimulation channels for stimulation of the patient's hearing, the sound processor having a filter bank for dividing the input audio signal into plural analysis channels, each containing a frequency domain signal representative of a portion of the audio signal, a signal level determiner for each analysis channel for analyzing the respective frequency domain signal, a neural stimulation signal generator for each analysis channel, a mapping unit for allocating the analysis channels to the stimulation channels according to an adjustable mapping scheme, and a control unit for controlling the mapping unit such that a standard mapping scheme is used in a standard operation mode and a low bandwidth mapping scheme is used in a low bandwidth operation mode.

Abstract: Analysis of the audio component in multimedia data is disclosed. Rooms can be described through room impulse responses (RIR), the “fingerprint” of a specific room. The method uses machine learning techniques to identify rooms from ordinary audio recordings.

Abstract: A high-pass filter is configured to remove a low-pass filter component of a first channel audio signal and a low-pass filter component of a second channel audio signal. A control unit is configured to detect a wind noise magnitude based on at least one from among the first channel audio signal and the second channel audio signal, and to increase a cutoff frequency of the high-pass filter according to an increase in the wind noise magnitude thus detected. The control unit is configured to set the cutoff frequency of the high-pass filter to a predetermined minimum value fMIN when the wind noise magnitude thus detected is smaller than a predetermined minimum value, and to gradually increase the cutoff frequency when the wind noise magnitude becomes greater than the minimum value.

Abstract: A method and an apparatus that can display a page of an e-book according to user input information about reading of the e-book are provided. The includes displaying an opened page of the e-book; detecting a pointing position indicated by a pointer and an amount of position variation of the pointer with respect to the opened page; modifying the opened page based on the pointing position and the detected amount of position variation; and displaying a modified page.

Abstract: The present invention provides methods and systems for digital processing of an input audio signal. Specifically, the present invention includes a high pass filter configured to filter the input audio signal to create a high pass signal. A first filter module then filters the high pass signal to create a first filtered signal. A first compressor modulates the first filtered signal to create a modulated signal. A second filter module then filters the modulated signal to create a second filtered signal. The second filtered signal is processed by a first processing module. A band splitter splits the processed signal into low band, mid band, and high band signals. The low band and high band signals are modulated by respective compressors. A second processing module further processes the modulated low band, mid band, and modulated high band signals to create an output signal.

Abstract: A method and system for providing a moving-screen sound system for a portable communication device such as a cell phone employs a fixed, pivoted, or hinged display screen driven directly or indirectly by an audio actuator. While the frequency response of the screen may have defects that would negatively impact overall sound quality from the device, in embodiments a supplemental audio transducer is placed adjacent the screen or elsewhere on the device so as to supplement or correct the frequency response of the screen. In an embodiment, the supplemental audio transducer is driven to correct a notch or other defect in the screen response. In another embodiment, the supplemental audio transducer is driven to extend the bass response of the screen.

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 method of panning includes panning an input signal or a low frequency portion of the input signal to split the input signal or the low frequency portion of the input signal into first and second channels, panning the input signal or a high frequency portion of the input signal to split the input signal or the high frequency portion of the input signal into third and fourth channels such that localization of the high frequency portion of the input signal coincides with localization of the low frequency portion of the input signal.

Abstract: A system and method for selectively applying Intensity Stereo coding to an audio signal is described. The system and method make decisions on whether to apply Intensity Stereo coding to each scale factor band of the audio signal based on (1) the number of bits necessary to encode each scale factor band using Intensity Stereo coding, (2) spatial distortions generated by using Intensity Stereo coding with each scale factor band, and (3) switching distortions for each scale factor band resulting from switching Intensity Stereo coding on or off in relation to a previous scale factor band.

Abstract: Chart animation control may be provided. Upon receiving a change to a displayed visualization comprising a plurality of data elements, a determination may be made as to whether to animate an update of the visualization according to the change to the displayed visualization. In response to determining to animate the update of the visualization according to the change to the data value, an animation of the update may be constructed and rendered and the updated visualization may be displayed.

Abstract: An audio processing device is provided. The audio processing device includes a shaper. The shaper receives a digital input audio signal and generates an output audio signal at an output terminal according to the digital input audio signal. The shaper includes a gain adjustment unit and a shaping unit. The gain adjustment unit receives the digital input audio signal and adjusts the digital input audio signal to generate a gained audio signal. The shaping unit receives the gained audio signal. When the shaper operates in a first mode, the shaping unit shapes an envelope of the gained audio signal to generate a shaped audio signal, and the shaped audio signal serves as the output audio signal.

Abstract: A noise suppression device includes a phase difference derived suppression coefficient computation section that over a phase difference utilization range computes for each frequency a phase difference derived suppression coefficient based on a phase difference, an amplitude ratio derived suppression coefficient computation section that computes for each frequency an amplitude ratio derived suppression coefficient based on an amplitude ratio or an amplitude difference, and based on the amplitude conditions, and a suppression section that suppresses noise contained in the input sound signals based on a suppression coefficient determined by using the phase difference derived suppression coefficient and the amplitude ratio derived suppression coefficient.

Abstract: Systems and methods for providing bass compensation to correct for uneven bass response are disclosed. An example bass compensation system includes a low pass filter configured to receive an audio signal from an audio source and provide a filtered audio signal, the low pass filter having a roll off of at least 18 dB per octave. The bass compensation system further includes a summing amplifier coupled to the low pass filter and configured to sum the audio signal from said audio source and the filtered audio signal to provide a summed audio signal, wherein the summed audio signal provided by the summing amplifier provides a bass boost at a first frequency and mid bass cut at a second frequency greater than the first frequency.

Abstract: There is provided an acoustic processing device capable of applying acoustic processing matching listener's sense to reproduce an audio signal with satisfactory sound quality in terms of auditory sense irrespective of the characteristics of a sound source. The acoustic processing device (1) includes a gain calculation section (5) that calculates a gain correction amount corresponding to predetermined auditory sense characteristics, an offset gain calculation section (6) that calculates a gain offset based on a frequency characteristics of an audio signal analyzed by a frequency analysis section (3), an acoustic signal generation section (7) that generates an acoustic signal based on the gain correction amount and the gain offset, and an acoustic addition section (8) that adds the generated acoustic signal to the audio signal.

Abstract: An apparatus for processing an audio signal to focus an acoustic signal by an arrangement of a plurality of loudspeakers comprises a frequency analyzer, a signal processor and a signal output interface. The acoustic signal is based on the audio signal. The frequency analyzer is configured to determine a fundamental frequency in a frequency spectrum of the audio signal depending on a geometry parameter of the arrangement of the plurality of loudspeakers. The signal processor is configured to adapt an overtone of the fundamental frequency to obtain the processed audio signal and the signal output interface is configured to output the processed audio signal to the plurality of loudspeakers.

Abstract: The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency Q?+r?0 is generated on the basis of existing components at ? and ?+?0. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and/or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank (101), a subband processing unit (102) configurable by control data (104) and a synthesis filter bank (103).

Abstract: An audio system includes a level controller receiving an input audio signal from an audio medium. The level controller includes a gain adjust module producing an output audio signal based on the input audio signal. A filter arrangement is communicatively coupled to the level controller. The filter arrangement receives the output audio signal and divides the output signal into a plurality of filtered signals. Each of the filtered signals corresponds to a respective frequency band. A multiband dynamics controller is communicatively coupled to the filter arrangement. The multiband dynamics controller receives the filtered signals and applies a respective gain characteristic to each of the filtered signals to thereby produce a plurality of gain adjust signals. The gain adjust signals are combined into a multiband dynamics controller output signal.

Abstract: The present disclosure relates to an audio player and speaker system, and particularly, to a wireless audio player and speaker system. More particularly, the present disclosure relates to a portable, rechargeable, wireless boombox-style audio player, for playing audio from an external audio source, and including a wireless charging element for wirelessly recharging an external media device.

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: In a wind noise suppressor, a divider divides the frequency band an input sound into a first frequency band having a possibility that wind noise is included and a second frequency band having a frequency higher than a frequency of the first frequency band, a calculator calculates a probability that the input sound includes wind noise from feature parameters of a sound in the first frequency band, a suppressor suppresses wind noise included in the first frequency band in accordance with an intensity calculated from the probability, and an adder mixes and outputs the sound in the second frequency band divided by the divider and the sound in the first frequency band by which wind noise is suppressed by the suppressor.

Abstract: A standing wave attenuation device is installed in a cabin of a vehicle so as to reduce a standing wave caused by external noise such as road noise. The standing wave attenuation device provides a closed loop including a feedback comb filter with a feedback loop, a microphone, a speaker, and a delay element. The delay element adjusts the phase of the output signal of the feedback comb filter such that the time needed for one-time circulation of a signal through the feedback loop matches a half period of the standing wave. An original sound including the standing wave is picked up by the microphone and subjected to processing so that the speaker produces a sound wave with the inverse phase against the phase of a sound wave constituting the standing wave, so that the standing wave is canceled out by the sound wave emitted from the speaker.

Abstract: A parametric loudspeaker system comprises a pre-compensator (305) for generating a pre-compensated envelope signal by applying a pre-compensation to the input audio signal where the pre-compensation compensates for distortion of in-air demodulation of the modulated ultrasound signal. A pre-modulator (307) generates a complex base band signal generating a phase signal from the pre-compensated envelope signal using a predetermined function for determining a phase signal from an amplitude signal such that the corresponding complex signal has either suppressed negative or positive frequencies. The complex base band signal is then generated to have an amplitude corresponding to the pre-compensated envelope signal and a phase corresponding to the phase signal. A modulator (309) quadrature modulates the complex base band signal on an ultrasonic quadrature carrier and an output circuit (311) drives the ultrasound transducer (301) from the modulated signal.

Abstract: Disclosed is an audio device that plays back compressed audio signals that can be sufficiently comfortable to be audible even in a play back environment with improved acoustic quality. In order to correct a weak signal component near a frequency with a high output level of a compressed audio, weak-signal component adding unit (121) extracts a specific frequency region based on input audio signals, generates a weak-signal component composed of harmonics signals composed of a harmonic component and the like, and a noise signal and adds the generated weak-signal component to the input audio signal. In order to correct a high-frequency band output level from which compressed audio signals are omitted, high-frequency band audio adding unit (122) generates audio of a high-frequency band composed of a harmonic signal and a noise signal and adds the generated high-frequency band audio to the input audio signal.

Abstract: The method comprises the steps of: a) converting the digital audio signal (PCM) into a voltage signal (VE); b) first lowshelf filtering of fixed gain (G2); c) calculating a value of excursion (x) of the loudspeaker; d) comparing the excursion with a maximum value and calculating a first gain of possible attenuation (G3); e) second lowshelf filtering of gain (G4+G3) taking into account the first gain of possible attenuation; g) comparing with the maximum saturation or clipping voltage (vMAX) and calculating a second gain of possible attenuation (G5); h) third lowshelf filtering gain (G6+G3+G5) taking into account the first and/or second gains of possible attenuation; i) comparing with the maximum saturation or clipping voltage (vMAX) and applying a gain of possible overall attenuation (G7); j) optionally compensating for the nonlinearities of the loudspeaker response; and k) reversely converting the signal (Vs) into a digital audio signal (S) without dimension, for later amplification.