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: 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 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: 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: 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.

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: Electronic circuitry is described. The electronic circuitry includes a first microelectromechanical system (MEMS) structure that exhibits a first frequency response in a voice frequency range and that captures a first signal. The electronic circuitry also includes a second MEMS structure coupled to the first MEMS structure. The second MEMS structure exhibits a second frequency response in an ultrasound frequency range and captures a second signal.

Abstract: The present invention provides for a cable which enhances an audio signal between two devices, such as an mp3 player and a speaker system, or a set-top box and home theater system. The cable includes integrated hardware and software elements for processing the signal. In various embodiments the cable may also include a user interface and selectable profiles which adjust the processing scheme for various purposes, such as a profile for environments with high ambient noise, or profiles for certain styles of music.

Abstract: Systems and methods can support constructing an infinite impulse response (IIR) filter. A desired frequency response may be received. An initial filter model may be constructed comprising complimentary pairs of component IIR filters based upon the desired frequency response. The filter model may be converged according to stepwise refinement of individual terms within the filter model. A global error for the converged filter model may be computed. An additional complimentary pair of component IIR filters may be incorporated into the filter model in response to the global error exceeding a maximum acceptable error. In response to incorporation of additional component IIR filters, convergence and evaluation of the filter model may be iterated. Upon final convergence of the filter model, an aggregate IIR filter may be generated by combing the component IIR filters.

Abstract: A speaker system includes a first speaker (203) and a second speaker (205). A driving circuit receives an audio signal and has a first drive circuit (209) generating a first drive signal for the first speaker (203) in response to a first filtering of the audio signal with a first passband. A second drive circuit (211) generates a second drive signal for the second speaker (205) in response to a second filtering having a second passband which includes a frequency band below the first passband. A delay (213) delays the second drive signal relative to the first drive signal. The sound from the second speaker is directionally radiated with a directional radiation pattern having a notch towards the listening position (111). The system uses the precedence effect and non-direct low frequency audio radiation to ensure that directional cues are predominantly provided by the first speaker (203) which may be small and positioned remote from the second speaker (205).

Abstract: A band division apparatus obtains an analog low-band signal, an analog intermediate-band signal and an analog high-band signal from a digital sound signal. In the band division apparatus, a digital filter separates the digital sound signal into a digital intermediate-band signal having the intermediate frequency band while the high frequency band and the low frequency band are attenuated, and a digital low-high-band signal having a frequency band combining the low frequency band and the high frequency band while the intermediate frequency band are attenuated. A DA converter converts the digital low-high-band signal into an analog low-high-band signal. Another DA converter converts the digital intermediate-band signal into the analog intermediate-band signal. An analog filter separates the analog low-high-band signal into the analog low-band signal and the analog high-band signal.

Abstract: An array loudspeaker includes a plurality of drivers arranged in an array configuration. A digital signal processor-based control system processes direct audio signal and indirect audio signal inputs for the loudspeaker to simultaneously produce direct sound in the form of a directed beam or wavefront, and indirect sound as a perceptually diffuse soundfield.

Abstract: A method and device for frequency compression of audio signals to reduce the occurrence of artifacts. A component of the audio signal having a plurality of frequency channels is shifted from a first frequency channel into a second frequency channel. A dominant instantaneous frequency is determined in the first frequency channel. During shifting or mapping, first the entire first frequency channel, including the dominant instantaneous frequency, is shifted or mapped into the second frequency channel, wherein the dominant instantaneous frequency obtains an intermediate frequency position. A final frequency position for the dominant instantaneous frequency is determined using a predefined compression characteristic in the second frequency channel, starting from the frequency position of the dominant instantaneous frequency in the first frequency channel. Finally, the dominant instantaneous frequency is shifted or mapped from the intermediate frequency position to the final frequency position.

Abstract: A distributed self-scaling network audio processing system includes end nodes interconnected by packet-switched network and operating as peers on the network. Each of the end nodes supports local input processing, mixing, and output processing. The input processing includes the option of dual input channels for supporting separate front-of-house and monitor workflows. End nodes are added to the system to support specific audio processing applications, based on the number of audio sources, the number of output mixes required, and the number of locations from which users choose to interact with the system.

Abstract: The subject matter described herein includes methods, systems, and computer readable media for synthesizing sounds using estimated material parameters. According to one aspect, a method for sound synthesis using estimated material parameters is provided. The method includes at a computing platform including a memory and a processor, estimating, by the processor and based on a recorded audio sample of an impact on a physical object, one or more material parameters of a physical object for use in synthesizing a sound associated with a virtual object having material properties similar to the physical object. The method further includes storing the one or more material parameters in memory.

Abstract: A computer-implemented method of determining a configuration for an audio processing operation, wherein the audio processing operation comprises a predetermined set of one or more audio processing sub-operations, each audio processing sub-operation being configurable with one or more respective configuration parameters, the method comprising: specifying the predetermined set of one or more audio processing sub-operations; specifying a target frequency response; and performing a convergent optimization process to determine a configuration for the audio processing operation that reduces a difference between the frequency response of the audio processing operation and the target frequency response, wherein the configuration comprises a respective value for each configuration parameter of each audio processing sub-operation.

Abstract: An electrical system including multiple bodies having an underlying structure resembling a double helix may be used to produce useful electromagnetic fields for various applications.

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 differential high impedance circuit for use in an acoustic apparatus includes a first set of transistor devices and a second set of transistor devices. The first set of transistor devices includes a first transistor (302) and a second transistor (306), and the first transistor (302) coupled to Vdd and the second transistor (306) coupled ground. The second set of two transistor devices includes a third transistor (304) and a fourth transistor (308). The third transistor (304) is coupled to the first transistor (302) and provides a first output (Out+), and the fourth transistor (308) is coupled to the second transistor (306) and provides a second output (Out?). The first and second outputs configured to provide a resistance.

Abstract: A device for producing a frequency-shifted audio signal based on an audio input signal is provided. The device has an interface and a frequency-shifting unit. The interface is configured for receiving the audio input signal. The frequency-shifting unit is configured for producing the frequency-shifted audio signal. The frequency-shifting unit is additionally configured to produce one of the second subband values based on one of the first subband values such that the second phase angle of this second subband value differs from the first phase angle of this first subband value by a phase angle difference, the phase angle difference being dependent on frequency information indicating by which frequency difference the audio input signal is to be shifted in order to obtain the frequency-shifted audio signal, and the phase angle difference being dependent on a frequency bandwidth of one of the first subbands.

Abstract: An application specific integrated circuit (ASIC) is used with an acoustic device. An input clock signal is received. The frequency of the input clock signal is determined, and the frequency is indicative of one of a plurality of operational modes of the ASIC. Based upon the determined frequency, an amount current provided to one or more operational blocks of the ASIC is changed.

Abstract: According to one embodiment, sound processor includes: communication module; outputting module; recording module; display; input module; controller; and calculating module. The controller (i) displays, on display, message prompting user to move a sound input device to position proximate to speaker, (ii) causes the outputting module to output the test sound and causes the recording module to record first sound, (iii) displays, after the first sound is recorded, on the display, message prompting the user to move the sound input device to listening position, and (iv) causes the outputting module to output the test sound and causes the recording module to record second sound. The calculating module finds a first frequency characteristic of the first sound and a second frequency characteristic of the second sound, and calculates, based on a difference between the first and second frequency characteristics, a value for correcting the second frequency characteristic to a target frequency characteristic.

Abstract: A high frequency energy converter which has application as an acoustic actuator for converting incoming high frequency energy into outgoing harmonized high frequency mechanical (e.g., sound) and electromagnetic waves. The energy converter is adapted to improve the quality of sound heard by a listener by reducing random and spurious harmonics that are introduced by the environment in which the listener is located. The energy converter includes an outer body and a reactive crystalline material (e.g., quartz) lying at the bottom of the outer body that is responsive to the incoming high frequency energy. A dispersion horn is located at the top of the outer body to be seated upon the crystalline material. The dispersion horn has a throat extending therethrough so that both incoming high frequency energy and outgoing high frequency mechanical and electromagnetic waves are transmitted through the throat of the horn in opposite directions.

Abstract: An audio signal processing device receives a plurality of audio signals via a left channel (L) and a right channel (R) so as to produce a composite signal L+R and a difference signal L?R. The composite signal L+R is changed in phase with an all-pass filter, while the difference signal L?R is changed in phase and frequency characteristic with a band-pass filter (e.g. a center frequency of 1 kHz). The band-pass filter has a gently curved frequency characteristic achieving a broad passing band. Additionally, a phase difference of 90 degrees is maintained between the all-pass filter and the band-pass filter over the entire audio frequency range. The composite signal and the difference signal are adjusted in their levels and then mixed together to produce a monaural signal achieving an audio surround effect for widely propagating sound into the surrounding space without degrading sound quality.

Abstract: Frequency equalization system 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. The frequency equalization system uses multiple microphones to measure the impulse responses of the room and uses the impulse responses to design filters to process 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 very short computation time.

Abstract: A device, system and method of playing back a digital audio stream wherein large amounts of pre-emphasis of the high frequencies is applied before the digital to analog conversion and before an interpolation or digital filter, followed by de-emphasis in the analog domain in order to yield better audio fidelity.

Abstract: An audio power management system manages operation of audio devices in an audio system. The audio power management system includes a parameter computer, a threshold comparator and a limiter. Audio signals generated with the audio system may be provided to the audio power management system. Based on a measured actual parameter of the audio signal, such as a real-time actual voltage and/or a real-time actual current, the parameter computer can derive estimated operational characteristics of audio devices, such as a loudspeaker included in the audio system. The threshold comparator may use the estimated operational characteristics to develop a threshold and manage operation of one of more devices in the audio system by monitoring the measured actual parameter, and selectively directing the limiter to adjust the audio signal, or another device in the audio system to protect or optimize performance.

Abstract: Embodiments are directed toward user profile based audio adjustment techniques. The techniques are used to render various audio and/or audio/video content having different audio output parameter values in accordance with a user profile that characterizes a user's desired value and/or range of one or more of the output parameter levels.

Abstract: Two audio samples and/or sets of audio samples are identified. The pitch distributions of the audio samples and/or sets of audio samples are identified, the pitch distribution of an audio sample or set of audio samples referring to how much of each of multiple pitches of notes is present in the audio sample or set of audio samples. Based on the pitch distributions of the audio samples and/or sets of audio samples, at least one pitch of one of the audio sample and/or set of audio samples can be automatically adjusted (but need not be, depending on the pitch distributions) to increase harmonic coherence of the audio samples and/or sets of audio samples.

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: Methods and systems for amplitude modulation in a parametric speaker system are provided that perform truncated double sideband (TDSB) frequency modulation of audio signal in which most of the processing is performed in the frequency domain, thus permitting use of fast processing techniques for amplitude modulation (AM) and filtering and reducing computation cost over time domain processing. A maximum envelope value of the time domain audio signal may be to the carrier signal in the frequency domain that avoids emitting the carrier signal when the input signal level is low or mute. The application of the envelope value may be smoothed to reduce discontinuity at input block boundaries.

Abstract: Apparatuses for and methods of carrying out dynamic equalization processing of an audio signal, and apparatuses for and methods of controlling such equalization processing of the audio signal to dynamically adjust the time-varying spectrum of an audio signal to more closely match a user specified target time-invariant perceived audio signal spectrum while preserving the original dynamic range of the audio signal. The dynamic equalization is according to a user-defined spectral profile specified by a control interface that allows a user to define, create, modify and/or apply the user-defined spectral profile.