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: 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: 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: 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: 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: 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: 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 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: 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.

Abstract: A driver, includes a driver block, a controller block, and a comparison block. The driver block includes an adjustable current source configured to produce a digital output stream. The controller block is coupled to the driver block. The comparison block is coupled to the driver block and the controller block. The comparison block is configured to compare the digital output stream to a reference value at a time delayed with respect to a master clock and based upon the comparison cause the controller block to adjust a strength of the driver block.

Abstract: In some embodiments, a virtual bass generation method including steps of: performing harmonic transposition on low frequency components of an input audio signal (typically, bass frequency components expected to be inaudible during playback of the input audio signal using an expected speaker or speaker set) to generate transposed data indicative of harmonics (which are expected to be audible during playback, using the expected speaker(s), of an enhanced version of the input audio which includes the harmonics); generating an enhancement signal in response to the transposed data; and generating an enhanced audio signal by combining (e.g., mixing) the enhancement signal with the input audio signal. Other aspects are systems (e.g., programmed processors) and devices (e.g., devices having physically-limited bass reproduction capabilities, such as, for example, a notebook, tablet, mobile phone, or other device with small speakers) configured to perform any embodiment of the method.

Abstract: A band broadening apparatus includes a processor configured to analyze a fundamental frequency based on an input signal bandlimited to a first band, generate a signal that includes a second band different from the first band based on the input signal, control a frequency response of the second band based on the fundamental frequency, reflect the frequency response of the second band on the signal that includes the second band and generate a frequency-response-adjusted signal that includes the second band, and synthesize the input signal and the frequency-response-adjusted signal.

Abstract: A method of attenuating an input signal to obtain an output signal is described. The method comprises receiving the input signal, attenuating the input signal with a gain factor to obtain the output signal, applying a filter having a frequency response with a frequency-dependent filter gain to at least one of a copy of the input signal and a copy of the output signal to obtain a filtered signal, the frequency-dependent filter gain being arranged to emphasize frequencies within a number N of predetermined frequency ranges, N>1; wherein the filter comprises a sequence of N sub-filters, each one of the N sub-filters having a frequency response adapted to emphasize frequencies within a corresponding one of the N predetermined frequency ranges; determining a signal strength of the filtered signal, and determining the gain factor from at least the signal strength.

Abstract: An arrangement of overlapping filter banks comprises a synthesis stage and an analysis stage. The synthesis stage receives a first signal segmented into time blocks and outputs, based thereon, an intermediate signal to be received by the analysis stage forming the basis for the computation of a second signal segmented into time frames. In an embodiment, the synthesis stage is operable to release an approximate value of the intermediate signal in a time block located L?1 time blocks ahead of its output block, which approximate value is computed on the basis of any available time blocks of the first signal, so that the approximate value contributes, in the analysis stage, to the second signal. The delay is typically reduced by L?1 blocks. Applications include audio signal processing in general and real-to-complex conversion in particular.

Abstract: A method of audio processing lowers the frequency of a high frequency audio area of an input audio in order to generate a lowered frequency audio area. The lowered frequency audio area is combined with the input audio to generate an output audio such that the output audio comprises the high frequency audio area, a low frequency audio area, and a lowered frequency audio area.

Abstract: An audio signal processing device includes: an input section to which an audio signal is input; a bias processing section configured to add a bias signal to the audio signal; a calculation section configured to perform a power calculation on the audio signal to which the bias signal has been added by the bias processing section; and an adding unit configured to add the audio signal on which the power calculation is performed by the calculation section to the audio signal having been input to the input section.

Abstract: A microphone system (10) includes a microphone sub-system including a microphone (14); and an acoustic filter (12). The acoustic filter (12) can include a resistive element (20) and a closed volume configured to act as an acoustic compliance. The acoustic filter (12) is configured to have a frequency response substantially equal and opposite to a difference between a frequency response of the microphone sub-system and a desired frequency response of the system (10).

Abstract: Provided is a method and apparatus for focusing sound using an array speaker system. The method includes generating a plurality of delayed signals to be focused to a predetermined position from an input signal, filtering a low-frequency signal having a frequency that is lower than a reference frequency from the delayed signals, generating low-frequency focusing signals divided into 2 groups by adjusting a gain of the filtered low-frequency signal, and applying the low-frequency focusing signals divided into the 2 groups to speaker units of the array speaker system at both sides with respect to a center portion of the array speaker system and outputting the low-frequency focusing signals through the speaker units. In this way, the performance of sound focusing for the low-frequency signal can be improved and thus a listener located a predetermined distance from and in a predetermined direction relative to the array speaker system can clearly listen to the low-frequency focusing signals.

Abstract: According to one embodiment, a characteristic correcting device includes: a correction filter configured to correct sound quality characteristics of a plurality of bands in a frequency range of an input signal based on a frequency characteristic which is set in advance to generate an output signal; an input module configured to input a surrounding sound signal of sound around an output device outputting the output signal; and an adjusting module configured to reduce the number of the bands of which the sound quality characteristics are to be corrected of the bands in the frequency range of the input signal in accordance with an increase in amplitude of the surrounding sound signal which is input.

Abstract: A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to measure the output of the transducer in order to control the adaptation of the anti-noise signal and to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit that performs the adaptive noise canceling (ANC) function also either adjusts the frequency response of the anti-noise signal with respect to the reference microphone signal, and/or by adjusting the response of the adaptive filter independent of the adaptation provided by the reference microphone signal.

Abstract: A headphone comprises a plurality of actuatable equalization selectors. Each of the selectors corresponds to an equalization setting that includes a preset distribution of relative amplitudes of sounds in predetermined frequency ranges. In one embodiment, each of the plurality of actuatable equalization selectors is a button-type switch. A knob-type switch or a voice recognition mechanism could also actuate an equalization setting. In a preferred embodiment, an equalizer identification indicator produces a communication perceivable to a headphone wearer and which corresponds to an equalization setting. The communication can be audible, preferably a human voice, or tactile, preferably vibration patterns corresponding to equalization settings.

Abstract: A modulation device includes: a spread code generation unit which generates a spread code having a predetermined cycle; an audio signal input unit to which an audio signal is input; a first modulation unit which phase-modulates the spread code in each cycle on the basis of a data code; and a combining unit which combines the audio signal with a modulation signal which has been generated on the basis of the phase-modulated spread code and distributed in a frequency range higher than a predetermined frequency to output a combined signal.

Abstract: A device for mixing multiple (N) pulse density modulated (PDM) bit streams of a bit rate, the device comprises an input logic, an error accumulation circuit, an error correction circuit and an adder of more than N bits; wherein the device is arranged to output an output PDM bit stream that represents a mixture of the multiple input PDM bit streams; wherein the output PDM bit stream comprises a plurality of output PDM bits, wherein a certain output PDM bit of a plurality of output PDM bits that form the output PDM bit stream is generated during a certain clock cycle; wherein the input logic is arranged to select, during each fraction of the certain clock cycle, a current bit of a selected PDM bit stream, wherein different PDM bit streams are selected during different fragments of the certain clock cycle; wherein the error accumulation circuit is arranged to store intermediate values during a first fraction till a penultimate fraction of the certain clock signal and to store a last value during a last fraction

Abstract: A method of selecting a compensation method to compensate a sound source signal output from a sound source playing apparatus includes; displaying a plurality of areas indicating each of a plurality of compensation methods with respect to a signal output from the sound source playing apparatus, outputting a test signal compensated by a compensation method of the plurality of compensation methods, the compensation method corresponding to at least one area of the plurality of areas, selecting any one of the plurality of areas based on a response of a user to the output test signal, and storing the compensation method corresponding to the selected area.

Abstract: A system for remapping an audio range to a human perceivable range includes an audio transducer configured to output audio and a processing circuit. The processing circuit is configured to receive the audio from an audio input, analyze the audio to determine a first audio range, a second audio range, and a third audio range. The processing circuit is further configured to use frequency compression on the first audio range based on the second audio range and third audio range to create a first open frequency range, move the second audio range into the first open frequency range to create a second open frequency range, move the third audio range into the second open frequency range, and provide audio output including the compressed first audio range, the moved second audio range, and the moved third audio range.

Abstract: Systems and methods for reducing the length of music data files having decaying sound patterns are provided. A system and method can include analyzing a music data file including an attack portion and a tail portion. A fadeout range can be associated with the music data file, and a cut threshold of the tail portion can be identified. A modified version of the music data file can be played back. The playback can include reducing sound levels of the music data file in accordance with the fadeout range and ending the playback when the cut threshold of the tail portion is reached.

Abstract: An apparatus for generating a drive signal for a sound transducer (109) comprises a sound generator (101) which provides an input audio signal. A divider (101) divides the input audio signal into at least a low frequency signal and a high frequency signal and an expander (105) generates an expanded signal by applying a dynamic range expansion to the low frequency signal. A combiner (107) then generates the drive signal by combining the expanded signal and the higher frequency signal. The threshold for applying the dynamic range extension may be adjusted depending on the amplitude of the low frequency signal. The low frequency signal may furthermore be compressed into a narrow frequency band around a resonance frequency. The approach may allow improved audio quality especially from high Q low frequency sound transducers by attenuating decay parts of bass signals thereby reducing sustain or ringing for bass notes.

Abstract: The present invention relates to a speaker housing having a rhomboid geometric shape. The rhomboid shape is a polyhedron with six faces, each of which is a parallelogram. The opposite sides of each parallelogram are substantially equal in length and their adjacent angles are substantially supplementary. The rhomboid shape of the speaker housing allows the listener to position the speakers in order to optimize the trajectory of the driver in a variety of listening situations, providing an accurate and optimized response for the listener. The rhomboid shape of the speaker housing and the utilization of point source, coaxial, dual concentric, triaxial, tri-concentric or other multi-concentric driver configurations and/or single driver loudspeaker driver technologies in the speaker can create a unique performance and functionality to this design.

Abstract: A channel divider that can suitably set a crossover frequency when a biwiring-connectable multiway speaker system including network circuits are used is provided. The channel divider includes a low-pass filter LPF and a high-pass filter HPF for dividing a band of a sound signal into a first output signal and a second output signal so as to output the signals, and a control circuit for controlling the filters. The channel divider has a sound output mode in which the control circuit sets a cutoff frequency fcL of LPF to a value higher than fc0 by about ? to 1 Oct., sets a cutoff frequency fcH of HPF to a value lower than fc0 by about ? to 1 Oct., and outputs the first output signal and the second output signal including frequency bands fcL to fcH when any crossover frequency fc0 for defining the band division is specified.

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 carried out 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.

Abstract: The invention relates to a method of processing an audio signal in a portable listening device, the audio signal comprising a low frequency part having an LF-bandwidth ?fLF and a high-frequency part having a HF-bandwidth ?fHF. The invention further relates to a listening device and to a listening system. The object of the present invention is to improve performance or save power in a portable listening device. The problem is solved in that the method comprises a) providing an audio input signal consisting of said low frequency part having an LF-bandwidth ?fLF; b) performing at least one signal processing step on the low frequency part of the audio signal; and c) performing a bandwidth extension process on said low frequency part of the audio signal to generate said high-frequency part of the audio signal, thereby generating or regenerating said audio output signal with a full bandwidth ?ffull comprising said LF-bandwidth ?fLF and said HF-bandwidth ?fHF.

Abstract: Estimation of a high band extension of a low band audio signal includes the following steps: extracting (S1) a set of features of the low band audio signal; mapping (S2) extracted features to at least one high band parameter with generalized additive modeling; frequency shifting (S3) a copy of the low band audio signal into the high band; controlling (S4) the envelope of the frequency shifted copy of the low band audio signal by said at least one high band parameter.

Abstract: The present document relates to the technical field of audio coding, decoding and processing. It specifically relates to methods of recovering high frequency content of an audio signal from low frequency content of the same audio signal in an efficient manner. A method for determining a first banded tonality value (311, 312) for a first frequency subband (205) of an audio signal is described. The first banded tonality value (311, 312) is used for approximating a high frequency component of the audio signal based on a low frequency component of the audio signal.