Abstract: A howling canceller is adapted to an acoustic system having a speaker and first and second microphones. The speaker and the first microphone form a first acoustic feedback loop; the speaker and the second microphone form a second acoustic feedback loop. The howling canceller includes a howling suppressing unit for performing suppression processing in such a way that: frequency components at which howling is possibly occurring are detected in each of the sound signals picked up by the first and second microphones; the detected frequency components of the sound signals picked up by the first and second microphones are compared with each other on a per-frequency basis and a frequency component having larger power is detected; and based on the comparison results, the larger power frequency component of at least one of the sound signals picked up by the first and second microphones is suppressed.

Abstract: A hearing protector comprises two protective muffs (1) with passive noise damping, a microphone (3) disposed in at least the one protective muff (1) and connected to an analog and a digital signal processing device (5 and 7, respectively), the signal processing devices (5, 7) being connected to loudspeakers (4) disposed interiorly in the protective muffs (1) for extinguishing interior noise in the protective muffs (1). The analog signal processing device is rehearsed for processing non-repeatable noise, while the digital signal processing device is rehearsed for processing repeatable noise. According to the invention, a further microphone (8) is disposed outside the passive noise damping and is connected to the digital signal processing device for tracing and locking onto repeatable noise.

Abstract: A microphone unit which can suppress collection of wind noise and minimize or eliminate digital signal processing has at least a microphone, a first acoustic transmissive material, and a second acoustic transmissive material, the first acoustic transmissive material is a fiber material in which fibers are intertwined with each other, the second acoustic transmissive material is a mesh-like member or a porous member having a plurality of holes, and the microphone is configured to be protected by the first acoustic transmissive material and the second acoustic transmissive material in this order.

Abstract: A noise cancellation system, comprising: an input for a digital signal, the digital signal having a first sample rate; a digital filter, connected to the input to receive the digital signal; a decimator, connected to the input to receive the digital signal and to generate a decimated signal at a second sample rate lower than the first sample rate; and a processor. The processor comprises: an emulation of the digital filter, connected to receive the decimated signal and to generate an emulated filter output; and a control circuit, for generating a control signal on the basis of the emulated filter output. The control signal is applied to the digital filter to control a filter characteristic thereof.

Abstract: An input signal is processed through noise suppression (NS) and echo control (EC) via a multipath model that reduces noise pumping effects while maintaining EC performance. A copy of a “noisy” input signal is sent to an EC component before the noisy signal is sent to a NS component, which processes the signal first, when there is a consistent noise level for estimation. The copy of the pre-processing noisy signal is sent to the EC component along with a “clean” or “noise-suppressed” signal output from the NS component. The EC component analyzes the noisy signal as if the EC was the first component in the signal chain to determine what actions to take. The EC component then applies these actions to the clean signal received from the NS component.

Abstract: An anti-noise signal is produced in accordance with an active noise cancellation process (ANC), at an input of a speaker so as to control how much background noise a user can hear. Strength of the anti-noise signal is adjusted gradually, rather than abruptly, in proportion to decreasing or increasing sound pressure level (SPL) of the background noise, during inactivation or activation of the ANC process. Other embodiments are also described and claimed.

Abstract: Described herein are multi-channel noise suppression systems and methods that are configured to detect and suppress wind and background noise using at least two spatially separated microphones: at least one primary speech microphone and at least one noise reference microphone. The multi-channel noise suppression systems and methods are configured, in at least one example, to first detect and suppress wind noise in the input speech signal picked up by the primary speech microphone and, potentially, the input speech signal picked up by the noise reference microphone. Following wind noise detection and suppression, the multi-channel noise suppression systems and methods are configured to perform further noise suppression in two stages: a first linear processing stage that includes a blocking matrix and an adaptive noise canceler, followed by a second non-linear processing stage.

Abstract: There is provided a sound reproduction device including a left channel housing unit that accommodates at least a left channel driver unit outputting a left channel sound, a left channel microphone recording an external sound, and a left channel micro-computer controlling a setting for a noise cancelling processing based on a recording signal of the left channel microphone; and a right channel housing unit that accommodates at least a right channel driver unit outputting a right channel sound, a right channel microphone recording an external sound, and a right channel micro-computer controlling a setting for a noise cancelling processing based on a recording signal of the right channel microphone. The left channel micro-computer and the right channel micro-computer are configured to communicate data with each other.

Abstract: A method of reducing noise in an environment where the noise source is in a fixed location relative to a pair of microphones, such as in a camera with a zoom motor, involves receiving signals x1(t), x2(t) from the respective microphones, and filtering each of the signals x1(t), x2(t) with respective first and second linear filters having filter coefficients obtained by computing eigenfilters corresponding to data samples from the respective microphones for noise only and signal only conditions.

Abstract: A method for adjusting the performance of an electroacoustic transducer includes receiving, by gain adjustment circuit, a displacement signal corresponding to a relative motion between a magnetic structure of the electroacoustic transducer and a voice coil of the electroacoustic transducer. The method includes detecting, by the gain adjustment circuit, a displacement signal value of the displacement signal as one of meeting or exceeding a displacement signal threshold. The method includes modifying, by the gain adjustment circuit, a loop gain of an active noise reduction loop associated with the electroacoustic transducer when the displacement signal value of the displacement signal one of meets or exceeds the displacement signal threshold.

Abstract: A method includes detecting an accessory device at a master device. The method also includes receiving, at the master device, active noise cancellation (ANC) coefficients associated with the accessory device in response to detecting the accessory device. The method also includes modifying audio content, at the master device, based on the ANC coefficients.

Abstract: A method for operating an active noise reduction system that is designed to reduce the harmonic or sinusoidal noise emanating from a rotating device, where there is an active noise reduction system input signal that is related to the frequency of the noise to be reduced, and where the active noise reduction system comprises one or more adaptive filters that output a generally sinusoidal noise reduction signal that is used to drive one or more transducers with their outputs directed to reduce the noise. Distortions of the noise reduction signal are detected. A distortion is based at least in part on differences between the frequency of the noise reduction signal and the frequency of the harmonic noise. The noise reduction signal is altered based on the detected distortion.

Abstract: A method of processing digitized microphone signal data in order to detect wind noise. First and second sets of signal samples are obtained simultaneously from two microphones. A first number of samples in the first set which are greater than a first predefined comparison threshold is determined. A second number of samples in the first set which are less than the first predefined comparison threshold is determined. A third number of samples in the second set which are greater than a second predefined comparison threshold is determined. A fourth number of samples in the second set which are less than the second predefined comparison threshold is determined. If the first number and second number differ from the third number and fourth number to an extent which exceeds a predefined detection threshold, e.g. as determined by a Chi-squared test, then an indication that wind noise is present is output.

Abstract: A microphone array apparatus includes: an acquisition unit configured to acquire samples from a sound signal inputted from each of a plurality of microphones, at predetermined time intervals; an operation unit configured to calculate a value based on volumes of the sound signal possessed by a plurality of the samples for each of the sound signals inputted from the plurality of microphones; a correlation coefficient calculator configured to calculate a coefficient of correlation between the sound signals, on the basis of the values calculated for the respective sound signals; and a gain calculator configured to calculate reduction gain for the sound signals inputted from the plurality of microphones, on the basis of the coefficient of correlation.

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 estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit uses the reference and/or error microphone, optionally along with a microphone provided for capturing near-end speech, to determine whether one of the reference or error microphones is obstructed by comparing their received signal content and takes action to avoid generation of erroneous anti-noise.

Abstract: On an active noise controller, an adjustment reference signal generator generates an adjustment reference signal Sra by reading the waveform data sequentially at a read location shifted by a given amount from a read location for a reference signal Sr with respect to the waveform data at a reference signal generator. A one-tap adaptive filter generates a control signal Scp by multiplying the adjustment reference signal Sra by a filter factor W. A gain controller outputs a compensation control signal Sca generated by multiplying the control signal Scp by gain G. A speaker outputs the compensation control signal Sca as a cancellation sound.

Abstract: Improvements in voice signals transmitted within communication systems are obtained by use of adaptive filters, front and rear microphones, noise cancelling systems and other means and methods. Disclosed embodiments include the use of directional microphones, primary inputs, secondary inputs, adaptive weight generators, canceller outputs to improve signal to noise ratios and other communication attributes.

Abstract: The headset comprises two earpieces each having a transducer for playing back the sound of an audio signal and received in an acoustic cavity defined by a shell having an ear-surrounding cushion. The active noise control comprises, in parallel, a feedforward bandpass filter receiving the signal from an external microphone, a feedback bandpass filter receiving as input an error signal delivered by an internal microphone, and a stabilizer bandpass filter locally increasing the phase of the transfer function of the feedback filter in an instability zone, in particular a waterbed effect zone around 1 kHz. A summing circuit delivers a weighting linear combination of the signal delivered by these filters together with the audio signal to be played back. Control is non-adaptive, with the parameters of the filters being static.

Abstract: Disclosed, in general, are devices that provide an air-tight chamber over a sound source while absorbing all fields of sounds from the sound source. In general, the devices feature: an anechoic chamber that is configured to receive a sound source in an air-tight manner; and an anechoic channel that is in fluid communication with the ambient atmosphere. Suitably, the anechoic chamber is adapted to capture air containing sound energy generated by the sound source, and distribute the air about an internal surface area on the inside of the chamber, wherein the internal surface area is sufficiently large to dampen or otherwise absorb the sounds energy. Preferably, the air is directed from the anechoic chamber through an anechoic channel extending therefrom to the ambient to further dampen or absorb the sound energy. In one configuration, the outer wall of the apparatus is configured to reflect ambient sounds.

Abstract: Systems and methods for controlling adaptivity of noise cancellation are presented. One or more audio signals are received by one or more corresponding microphones. The one or more signals may be decomposed into frequency sub-bands. Noise cancellation consistent with identified adaptation constraints is performed on the one or more audio signals. The one or more audio signals may then be reconstructed from the frequency sub-bands and outputted via an output device.

Abstract: Near-end equipment for a communication channel with far-end equipment. The near-end equipment includes at least one loudspeaker, at least two microphones, a beamformer, and an echo canceller. The communication channel may be in one of a number of communication states including Near-End Only state, Far-End Only state, and Double-Talk state. In one embodiment, when the echo canceller determines that the communication channel is in either the Far-End Only state or the Double-Talk state, the beamformer is configured to generate a nearfield beampattern signal that directs a null towards a loudspeaker. When the echo canceller detects the Near-End Only state, the beamformer is configured to generate a farfield beampattern signal that optimizes reception of acoustic signals from the near-end audio source. Using different beamformer processing for different communication states allows echo cancellation processing to be more successful at reducing echo in the signal transmitted to the far-end equipment.

Abstract: The present invention provides a noise reduction control method using a microphone array and a noise reduction control device using a microphone array wherein the method comprises the steps of: S1: collecting, by the microphone array, acoustic signals; S2: estimating incidence angles of all acoustic signals of the microphone array; S3: conducting a statistics on signal components according to incidence angles; S4: determining a parameter ? from a ratio of noise components according to the statistical result and using the parameter ? as a control parameter for controlling an adaptive filter. With the present invention, space position information of the sound is obtained directly with the microphone array to control update of the adaptive filter more accurately, so as to eliminate noise, enhance SNR and protect speech quality well at the same time.

Abstract: A Personal Noise Reduction for enclosed cabins is disclosed. According to one embodiment, an input audio source corresponding to sound received from multiple microphones situated equidistantly in both directions in a two dimensional plane, is converted to a digital signal via an analog to digital (A/D) convertor. The AID converted audio is analyzed for content to identify ambient noise. The frequency, amplitude and phase of the identified ambient noise is subsequently determined. A Noise correction sound wave is generated with negative phase of that corresponding to the identified ambient noise. The noise correction sound wave is added to the identified noise to create a noise corrected sound.

Abstract: A method and system for reducing ambient noise distraction with an electronic personal display is disclosed. One example determines when the electronic personal display is in reader mode. In addition, ambient noise around the electronic personal display is also detected. Noise cancelling sound waves are generated at the electronic personal display for reducing ambient noise distraction. The noise cancelling sound waves are then output from at least one speaker coupled with the electronic personal display.

Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

Abstract: A method of controlling a vehicle includes detecting vibration in a drivetrain of the vehicle with a fore/aft accelerometer during a launch event. An operating regime of the vehicle is identified concurrently with the detected vibration when the detected vibration is outside a predefined allowable range. A control strategy of the drivetrain is adjusted to define a revised control strategy to avoid the identified operating regime. The revised control strategy is applied to control the drivetrain to mitigate vibration in the drivetrain.

Abstract: One embodiment provides an audio processing device including: a first audio input receiver configured to receive mainly environment noise; a second audio input receiver configured to receive mainly a voice; a first frequency band difference calculator configured to calculate a first power difference, obtained through accumulation over plural frequency bands, of signals output from the first audio input receiver and the second audio input receiver; a first noise suppression amount calculator configured to calculate a first noise suppression amount based on the first power difference so as to increase the first noise suppression amount if the first power difference is relatively small; and a noise suppressor configured to perform a noise suppression for the signal that is output from the second audio input receiver based on the first noise suppression amount.

Abstract: A computationally implemented system and method that is designed to, but is not limited to: electronically modulating two or more acoustic ultrasonic signals according to output information to be transmitted as two or more acoustic ultrasonic signals from two or more portable electronic device emitters of a portable electronic device; and electronically projecting said two or more acoustic ultrasonic signals from said portable electronic device emitters to produce a first set of one or more acoustic audio signals from a first set of said two or more acoustic ultrasonic signals at a first location and to produce a second set of one or more second acoustic audio signals from of a second set of said two or more acoustic ultrasonic signals at a second location. In addition to the foregoing, other method aspects are described in the claims, drawings, and text forming a part of the present disclosure.

Abstract: A Noise Control Process for targeted environments is disclosed. According to one embodiment, an input audio source corresponding to sound received from a group of microphones placed at certain locations in the targeted environment, is converted to a digital signal via an analog to digital (A/D) convertor. The ND converted audio is analyzed for content to identify ambient noise. The frequency, amplitude and phase of the identified ambient noise is subsequently determined. A Noise correction sound wave is generated with negative phase of that corresponding to the identified ambient noise. The noise correction sound wave is added to the identified noise to create a noise corrected sound.

Abstract: A Noise Cancellation Process for a motorcycle comprising an input audio source. Input audio source is combined in a mixer. Noise analyzer analyzes output from the mixer for amplitude and harmonic content. A replicate wave that is identical to the input audio source is created. The phase of the replicate wave is changed by ?180 degrees. Amplitude from the Noise Analyzer is monitored and amplitude of the replicated wave is matched to the original wave. The phase of the replicate wave and the input audio wave are combined to create Audio Out. Audio Out is played on the system.

Abstract: A voice activity detector (100) includes a frame divider (201) for dividing frames of an input signal into consecutive sub-frames, an energy level estimator (202) for estimating an energy level of the input signal in each of the consecutive sub-frames, a noise eliminator (203) for analyzing the estimated energy levels of sets of the sub-frames to detect and eliminate from enhancement noise sub-frames and to indicate remaining sub-frames as speech sub-frames, and an energy level enhancer (205) for enhancing the estimated energy level for each of the indicated speech sub-frames by an amount which relates to a detected change of the estimated energy level for a current speech sub-frame relative to that for neighboring speech sub-frames.

Abstract: A personal audio device, such as a wireless telephone, includes an adaptive noise canceling (ANC) 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 is also provided proximate the speaker to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit determines a degree of coupling between the user's ear and the transducer and adjusts the adaptive cancellation of the ambient sounds to prevent erroneous and possibly disruptive generation of the anti-noise signal if the degree of coupling lies either below or above a range of normal operating ear contact pressure.

Abstract: There is provided a noise cancellation system, comprising: an input for a digital signal, the digital signal having a first sample rate; a digital filter, connected to the input to receive the digital signal; a decimator, connected to the input to receive the digital signal and to generate a decimated signal at a second sample rate lower than the first sample rate; and a processor. The processor comprises: an emulation of the digital filter, connected to receive the decimated signal and to generate an emulated filter output; and a control circuit, for generating a control signal on the basis of the emulated filter output. The control signal is applied to the digital filter to control a filter characteristic thereof.

Abstract: An acoustic noise canceling microphone arrangement and processor that uses a principal microphone and other microphones that may be incidentally or deliberately located in the vicinity of the principal microphone in order to derive an audio signal of enhanced signal-to-background noise ratio. In one implementation, the principal and incidental microphones comprise the microphone built into a mobile phone and the microphone built into a Bluetooth headset.

Abstract: Provided are a method for estimating a spectrum density of diffused noises. Also provided is a processor for implementing the method. The processor includes at least two sound receiving units and a spectrum density estimating unit for estimating spectrum density.

Abstract: A system, method, and computer program product are provided for cleaning an audio segment. For a given audio segment, an offset amount is calculated where the audio segment is maximally correlated to the audio segment as offset by the offset amount. The audio segment and the audio segment as offset by the offset amount are averaged to produce a cleaned audio segment, which has had noise features reduced while having signal features (such as voiced audio) enhanced.

Abstract: The present invention relates to a sound emitting and collecting apparatus including a sound collecting portion that captures surrounding sound using two microphones, and a sound emitting portion that emits sound from at least one speaker. The apparatus includes a sound source separating portion that extracts a target sound from a sound source in a predetermined direction, based on an input sound signal obtained by capturing surrounding sound using the two microphones, and an emission non-target sound removing portion that removes a non-target sound that is emitted from the speaker and captured by each of the microphones, based on sound source data for the sound emitting portion. The emission non-target sound removing portion is provided on a path that reaches the sound source separating portion. The emission non-target sound removing portion has a structure similar to an acoustic echo canceller, for example.

Abstract: Psychoacoustic Bass Enhancement (PBE) is integrated with one or more other audio processing techniques, such as active noise cancellation (ANC), and/or receive voice enhancement (RVE), leveraging each technique to achieve improved audio output. This approach can be advantageous for improving the performance of headset speakers, which often lack adequate low-frequency response to effectively support ANC.

Abstract: An active vibration noise control device having a pair of speakers, including: a basic signal generating unit generating a basic signal based on a vibration noise frequency; an adaptive notch filter generating a first control signal provided to one speaker using a first filter coefficient and generating a second control signal provided to the other speaker using a second filter coefficient to cancel the generated vibration noise; a microphone detecting cancellation error between the vibration noise and the control sounds and outputting an error signal; a reference signal generating unit generating a reference signal based on a transfer function from the speakers to the microphone; a filter coefficient updating unit updating first and second filter coefficients, minimize the error signal; and a phase difference limiting unit limiting a phase difference between control sounds generated by different speakers. Therefore, it becomes possible to appropriately ensure a uniform and wide noise-cancelled area.

Abstract: A microphone array device includes a first sound reception unit configured to obtain a first sound signal that is input from a first microphone, a second sound reception unit configured to obtain a second sound signal that is input from a second microphone, a noise state evaluation unit configured to compare the first sound signal and the second sound signal and to obtain an evaluation parameter to evaluate an influence of a non-target sound included in the second sound signal on a target sound included in the first sound signal according to a result of the comparison, a subtraction adjustment unit configured to set a suppression amount for the second sound signal based on the evaluation parameter and to generate a third sound signal; and a subtraction unit configured to generate a signal to be output based on the third sound signal and the first sound signal.

Abstract: A sound focusing technique is provided to transfer sound to a particular direction. In a sound focusing apparatus, first and second speakers may be arranged to emit sound in opposite directions to form a sound zone. An amplitude and/or a phase of a received signal may be adjusted by a signal processing unit to assign the received signal and the adjusted signal to the first and second speakers, respectively.

Abstract: A method of reducing noise in an acoustic system, the method comprising at a first user terminal: receiving an audio signal from at least one further user terminal over a network; executing a communication client on a processing unit, the client configured so as when executed to: supply the audio signal to an audio signal processing module of the first user terminal, wherein the processing module processes the audio signal, whereby a level of gain is applied to the audio signal, and outputs a processed audio signal to a speaker; estimate a noise level of the audio signal and the processed audio signal and estimate the gain applied by the processing module taking into account both the noise level estimates; selectively apply a system gain reduction step to at least one of the audio signal and an audio signal received via a microphone, based on at least the estimated gain.

Abstract: Systems and methods for suppressing pop-up noise in an audio signal are disclosed. The system includes a driver circuit shared by a pin interface and a complementary pin interface. A control unit is coupled to the pin interface and the complementary pin interface. To activate the pin interface, the control unit is configured to first activate the driver output at the complementary pin interface. Once the complementary pin interface achieves a preset voltage, the driver output is switched to the pin interface by the control unit. In addition, the driver circuit can be calibrated for a DC offset on the complementary pin interface by re-using calibration data calculated at the pin interface. Further, DC correction signals can be provided from a pre-biasing circuit based on the calibration data of the driver circuit.

Abstract: Quality of industrial products is evaluated by evaluating non-stationary operation sound, which is a kind of operation sound, from an aspect of tone, using closely simulated evaluation levels of evaluation of non-stationary sound by used of a human sense of hearing. An operation sound of a conforming product sample is converted into sound waveform data by a sound collecting unit, and the sound waveform data is input into a computer via an A-D converter, and then converted into psychoacoustic parameters. Data of pseudo conforming products is additionally obtained from the psychoacoustic parameters of a plurality of conforming product samples by making use of deviation in the data of the conforming product samples. Threshold data is obtained using thresholds and masking data for evaluation calculated from psychoacoustic parameters of data of the conforming product samples and the pseudo conforming products by a statistical technique.

Abstract: A purpose is to suppress recognition process delay generated due to load in signal processing. Included is a speech input means 10 that inputs a speech signal, an output evaluation means 20 that evaluates whether or not the speech signal input by the speech input means 10 is the speech signal in a sound section, which is a speech section assuming that a speaker is speaking, and outputs the speech signal as a speech signal to be processed only when evaluated as the speech signal in the sound section, a signal processing means 30 that performs signal processing to the speech signal, which is output by the output evaluation means 20 as the speech signal to be processed, and a speech recognition processing means 40 that performs a speech recognition process to the speech signal which is signal-processed by the signal processing means 30.

Abstract: A method and a device for reducing snore annoyances include determining a snore sound pattern of a snoring person to predict an upcoming snore sound level. A faked snore sound is played to flatten the resulting snore sound level.

Abstract: A wireless audio system is provided which incorporates a main device and a holder, for use with a listening device and an audio output device. The main device is capable of connecting wirelessly to an audio output device and generally includes a chip for sound transmission, a series of controls to command audio output device and a port. The holder connects to and receives an audio transmission from the main device at port. The holder transmits the audio received to an external listening device via a second connector.

Abstract: To provide a noise reduction transmitter which can secure clarity of sounds collected in very noisy environments and maintain a quality of sounds without devising a noise insulation cover particularly. A transmission microphone 7 is arranged inside a noise insulation cover 2 worn on and covering at least a user's 1 mouth. A noise detection microphone 9 which detects external noises is arranged outside the noise insulation cover, and a noise component cancellation circuit 11 is provided which generates a noise component cancellation signal based on an output signal from the noise detection microphone. An electroacoustic transducer 8 is arranged in the noise insulation cover to reproduce a noise component cancellation sound based on an output signal from the noise component cancellation circuit 11.

Abstract: A frequency domain active noise cancellation system and method are provided that achieves superior active reduction of offending noise sources. The frequency-domain active noise cancellation identifies and cancels one or more of the strongest relatively narrow frequency bands in the power spectra of the offending sound sources.

Abstract: A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.