Abstract: A system and method for digital processing including a gain element to process an input audio signal, a high pass filter to then filter the signal and create a high pass signal, a first filter module to filter the high pass signal and create a first filtered signal and a splitter to split the high pass signal into two high pass signals. The first filter module filters one high pass signals before a first compressor modulates the signal or a high pass signal to create a modulated signal. A second filter module filters the modulated signal to create a second filtered signal that is processed by a first processing module including a band splitter that splits the signal into low and high band signals that are then modulated by compressors. A second processing module processes the modulated low and high band signals to create an output signal.

Abstract: An audio mixer is provided comprising: an audio receiver configured to receive an audio signal comprising a plurality of audio channels and a controller comprising one or more modules configured to: receive an audio content setting from a user interface device, adjust the relative volumes of the audio channels according to the audio content setting to provide a plurality of adjusted audio channels and combine the adjusted audio channels to generate a custom audio content.

Abstract: A content presentation system including a display screen and an array of speakers behind the display screen is described herein. In some examples, speaker-associated screen areas, the volume of a sound, and a screen location of a sound source may be used to determine one or more speakers that are associated with the sound. For example, the volume of a sound and the screen location of a sound source may be used to determine a sound range for the sound that may be used to associate one or more speakers with the sound. In some examples, upon determination of the sound range, one or more speaker-associated screen areas that are wholly or partially included within the sound range may then be identified. Each speaker that is represented by an identified speaker-associated screen area may then be associated with the sound.

Abstract: A method of augmenting an ambiently experienced audio content is provided, the method comprises recording an acoustic audio signal from the ambiently experienced audio content; receiving a wireless signal comprising a first audio signal and a second audio signal, the wireless signal received using a first wireless communication system, wherein the first audio signal comprises sound that is also audible within the ambiently experienced audio content and the second audio signal comprises sound that is substantially inaudible within the ambiently experienced audio content; comparing the first audio signal with the acoustic audio signal to determine a delay between the acoustic audio signal and the first audio signal; delaying the second audio signal by the determined delay; and reproducing the delayed second audio signal at an audible volume, such that it is substantially synchronized with the ambiently experienced audio content.

Abstract: An ASC apparatus, which actively generates a sound effect inside a cabin of a vehicle provided with an internal combustion engine and an electric motor as driving sources, includes: a reference signal generating unit which generates a harmonic reference signal based on a vehicle-speed-corresponding frequency, being a frequency based on a vehicle speed, by referring to waveform data; a driver seat speaker which outputs a sound including the sound effect; and a signal processing unit which generates a control signal that forms the sound effect by multiplying the reference signal by a sound effect gain related to the reference signal, and outputs the control signal to the driver seat speaker. The signal processing unit sets the sound effect gain based on a state of charge of a storage battery.

Abstract: A system for controlling a sound level of a prime mover permitted to emanate from or within a vehicle includes a global positioning satellite (GPS) receiver operable for receiving a GPS signal and a sound control device. The sound control device is operable for increasing or decreasing a sound level of the prime mover in response to a control signal. A controller in communication with the GPS receiver includes a calendar and clock, and is programmed to execute a method, including receiving the GPS signal via the GPS receiver, and determining an actual location of the vehicle using the GPS signal. The controller also determines an actual date and time of day using the calendar and the clock, respectively, and transmits the control signal to the sound control device to increase or decrease the sound level using at least one of the actual location, date, and time of day.

Abstract: A software application characterizes a nonlinear system by applying a series of test signals at full power and performing an algorithm on the resulting outputs to determine the true linear response and the order-separated outputs. The application generates a baseline test signal, multiplied by a gain factor. The application inputs the test signal to the system to produce a response. The application then generates another test signal by multiplying the baseline signal by a different gain factor. The application iterates generating test signals by multiplying by differing gain factors. The application then constructs a Vandermonde matrix of the gain factors, computes the inverse Vandermonde matrix, and convolves the inverse Vandermonde matrix with a matrix of the system responses to each of the test signals. The elements of the resulting convolution represent the order-separated outputs including the linear response output of the nonlinear system at full power.

Abstract: A device which produces the necessary directional audio signals for a 3-dimensional audio playback and which in that case uses as input signals the available channels of an audio recording intended for 2-dimensional audio playback. By taking psychoacoustic effects into account the desired spatial 3D audio effect is produced by a targeted use of signal delays, frequency-dependent amplitude matchings and a limited use of reverberation effects in conjunction with a targetedly asymmetric processing.

Abstract: An audio system of a vehicle includes a mode module that sets a mode signal to a first mode when a vehicle speed is greater than a predetermined speed and a longitudinal acceleration of the vehicle is less than a predetermined acceleration. The predetermined acceleration is less than zero and the predetermined speed is greater than zero. A sound control module, when the mode signal is in the first mode, selectively sets audio characteristics for a deceleration event of the vehicle based on randomization parameters. An audio driver module, based on the audio characteristics, applies power to speakers to output sound within a passenger cabin of the vehicle.

Abstract: An audio system of a vehicle includes a total frequencies module that selectively varies a total number of output frequencies based on one or more present operating parameters of the vehicle. An allocation module, based on one or more present operating parameters of the vehicle, determines an allocation indicative of: (i) a first portion of the total number of output frequencies allocated to outputting first predetermined sounds for internal combustion engine operation; and (ii) a second portion of the total number of output frequencies allocated to outputting second predetermined sounds for electric motor operation. Based on the total number and the first and second portions, an audio driver module applies power to speakers of a passenger cabin of the vehicle to: (i) output first predetermined sounds at harmonic frequencies of the internal combustion engine; and (ii) output second predetermined sounds at harmonic frequencies of the electric motor.

Abstract: A system and method for addressing acoustic signal reverberation is provided. Embodiments may include receiving, at one or more microphones, a first audio signal and a reverberation audio signal. Embodiments may further include processing at least one of the first audio signal and the reverberation audio signal. Embodiments may also include limiting a model based reverberation equalizer using a temporal constraint for direct sound distortions, the model based reverberation equalizer configured to generate one or more outputs, based upon, at least in part, at signal least one of the first audio signal and the reverberation audio signal.

Abstract: A method and apparatus processes multi-channel audio by encoding, transmitting or recording “dry” audio tracks or “stems” in synchronous relationship with time-variable metadata controlled by a content producer and representing a desired degree and quality of diffusion. Audio tracks are compressed and transmitted in connection with synchronized metadata representing diffusion and preferably also mix and delay parameters. The separation of audio stems from diffusion metadata facilitates the customization of playback at the receiver, taking into account the characteristics of local playback environment.

Abstract: A system for controlling distortion comprising a total harmonic distortion (THD) modeling system configured to apply a chirp signal to a system and to identify one or more frequency bands at which distortion is present, and to apply a ramping signal to identify for each of the one or more frequency bands an input signal level at which distortion is initiated. A signal processing system configured to receive an input signal, to determine whether frequency components are present in the input signal that are associated with the one or more frequency bands at which distortion is present, and to limit the amplitude of the input signal at the one or more frequency bands, such as by applying dynamic range compensation.

Abstract: A system and a method of indicating the reception of voice in a hearing assistance system that is constructed and arranged to assist a user to better hear the voice of another person. The hearing assistance system includes a detector that is capable of determining whether or not speech has been received by the hearing assistance system. In response to detecting the reception of the voice of another person by the hearing assistance system, the reception of the voice of another person by the hearing assistance system is visually indicated.

Abstract: A motor vehicle audio system, and a method of operating such a system. There is at least one rear far-field speaker that is located behind a front seat of the motor vehicle, and at least one near-field speaker located in front of the far-field speaker and close to an intended position of a listener's head. There is an audio signal processor configured to detect when the rear far-field speaker is shut off while the near-field speaker remains on, and in response automatically increase the output energy of the near-field speaker.

Abstract: The present invention relates to a signal processing apparatus and method, a program, and a data recording medium configured such that the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis. An analyzer 21 generates mapping control information in the form of the root mean square of samples in a given segment of a supplied audio signal. A mapping processor 22 takes a nonlinear function determined by the mapping control information taken as a mapping function, and conducts amplitude conversion on a supplied audio signal using the mapping function. In this way, by conducting amplitude conversion of an audio signal using a nonlinear function that changes according to the characteristics in respective segments of an audio signal, the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis. The present invention may be applied to portable playback apparatus.

Abstract: There is provided a microphone unit for recording audio signals comprising an audio processing unit for processing the recorded audio signals, a detection unit for detecting metadata belonging to the audio signal, and a first transmitting and receiving unit for transmitting the processed audio signal and the metadata.

Abstract: Apparatus and associated methods relate to training FIR filter coefficients by deconvolving a first input signal and a second input signal in the frequency domain, both of the signals being generated in response to an undetermined broadband excitation applied to an acoustic body instrument, until a fidelity of the second signal convolved with the trained coefficients meets predetermined fidelity criteria relative to the first signal. In an illustrative example, a musical instrument pickup signal and a microphone signal from the musical instrument may be sampled, segmented, and transformed to the frequency domain. FIR filter coefficients may be, for example, trained by block deconvolution in the frequency domain of the microphone signal and the pickup signal. In various examples, the trained FIR filter coefficients may adapt the pickup signal to mimic microphone performance, including full-body acoustic content.

Abstract: An apparatus for controlling a virtual engine sound in response to auto-cruise speed settings may include: a controller area network (CAN) communication unit configured to receive information on one or more of auto-cruise speed settings and whether an auto-cruise function is activated, and generate a CAN signal; a control unit configured to control a virtual engine sound function in response to the CAN signal; a signal processing unit configured to process and output a virtual engine sound signal according to control of the control unit; and a digital amp unit configured to amplify the virtual engine sound signal outputted from the signal processing unit and produce a virtual engine sound.

Abstract: Features are disclosed for estimating reverberation an audio signal without prior knowledge of the sound captured in the audio signal and without prior knowledge of the environment in which the sound was captured. This reverberation estimation may be referred to as “blind estimation.” The reverberation may be blindly estimated using autocorrelation of the sound intensity of the signal with respect to time. The blind estimation can be used to decide whether or not to mitigate the effects of reverberation in the signal prior to performing subsequent processes, such as automatic speech recognition.

Abstract: A jitter-associated data generator is provided for generating data associated with jitter. The jitter-associated data generator comprises a first circuit, a second circuit, and a third circuit. The first circuit generates an output including frequency components in a first frequency band common to first and second jitter, from a received digital input, as digital data associated with first jitter. The second circuit generates an output including frequency components in a second frequency band corresponding to the difference between the first and second jitter, from the received input, as differential digital data. The third circuit generates an output including frequency components in a third frequency band including the first and second frequency bands, from the received first jitter-associated digital data and differential digital data, as digital data associated with the second jitter.

Abstract: A method for providing information about a failure in a virtual engine sound system (VESS) unit for a vehicle includes sending a VESS unit failure message to a cluster, and displaying the failure message on the cluster. In the unit failure information providing method, VESS unit failure information may be provided by the use of a navigation unit or electric/electronic subassemblies.

Abstract: Apparatuses, systems, media and methods may provide for environment actuation by one or more augmented reality elements. A location module may determine a location of one or more networked devices in a real space and/or establish a location of the one or more augmented reality elements in a virtual space, which may be mapped to the real space. A coordinator module may coordinate a virtual action in the virtual space of the one or more augmented reality elements with an actuation event by the one or more networked devices in the real space. The actuation event may correspond to the virtual action in the virtual space and be discernible in the real space.

Abstract: In general, techniques are described for compensating for loudspeaker positions using hierarchical three-dimensional (3D) audio coding. An apparatus comprising or more processors may perform the techniques. The processors may be configured to perform a first transform that is based on a spherical wave model on a first set of audio channel information for a first geometry of speakers to generate a first hierarchical set of elements that describes a sound field. The processors may further be configured to perform a second transform in a frequency domain on the first hierarchical set of elements to generate a second set of audio channel information for a second geometry of speakers.

Abstract: A power and control unit for a low or medium voltage apparatus, said apparatus comprising at least a couple of electrical contacts that can be coupled/separated by means of an electro-magnetic actuator, first power storage means for supplying electric power for the operations of said apparatus and power supply means for charging first said power storage means, characterized in that it comprises: a primary control device for managing the operations of said apparatus when said power supply means are available; a secondary control device for managing the operations of said apparatus when said power supply means are not available, said secondary control device being able to receive electric power directly from said first power storage means.

Abstract: An adjustable microphone. The microphone includes a MEMS microphone, a charge pump, a preamplifier, a first analog-to-digital converter, a root mean square (RMS) power detector, and a logic circuit. The MEMS microphone is configured to provide a signal indicative of sound detected by the MEMS microphone. The charge pump provides a bias voltage to the MEMS microphone. The preamplifier receives the signal from the MEMS microphone, and outputs an amplified signal indicative of sound detected by the MEMS microphone. The first analog-to-digital converter receives the amplified signal and converts the amplified signal to a digital signal. The root mean square power detector is configured to detect a power level of the amplified signal and output an indication of the power of the amplified signal. The logic circuit receives the RMS power detector output and a control input, and adjusts the operation of the microphone based on the control input.

Abstract: A method of emulating a sound of an audio apparatus by using a sound emulation apparatus includes obtaining k past samples (where k is a natural number that is equal to or greater than 1) by delaying a current sample of an audio signal; applying a plurality of characteristic functions indicating an input/output relationship of the audio apparatus to the current sample and the k past samples, respectively; and adding the current sample and the k past samples, to which the plurality of characteristic functions have been applied, respectively, to generate an emulation sound of the audio apparatus.

Abstract: A signal processing apparatus includes a low-pass-filter processing unit configured to perform processing for limiting a band of an input audio signal on the basis of a cutoff frequency; a high-pass-filter processing unit configured to perform processing for limiting a band of the input audio signal on the basis of the cutoff frequency; a delay processing unit configured to perform processing for delaying the audio signal band-limited by the high-pass-filter processing unit; and a combination processing unit configured to combine the audio signal band-limited by the low-pass-filter processing unit and the audio signal subjected to the delay processing performed by the delay processing unit.

Abstract: A method and apparatus for processing a signal, and more particularly, to an apparatus for processing a mix signal and method thereof are disclosed, by which a mix signal such as an audio signal and a video signal can be encoded/decoded. The present invention includes receiving at least one of a mix signal and source signals and generating a unified side information corresponding to a unified source signal using the mix signal and the at least one of the source signals, wherein the unified source signal is generated by grouping at least one source signal.

Abstract: In a method of reproducing audio signals using an electronic device, the electronic device decodes an audio file to generate original digital audio signals, and attenuates the original digital audio signals having frequencies that are lower than a preset cutoff frequency. The electronic device further amplifies each of attenuated digital audio signals and the original digital audio signals having frequencies that are higher than the preset cutoff frequency with a preset gain. Once the electronic device converts the amplified digital audio signals into analog audio signals, and outputs the analog audio signals to an audio amplifier, the audio amplifier amplifies the analog audio signals to drive a speaker to output the analog audio signals.

Abstract: A device comprises one or more processors configured to apply a binaural room impulse response filter to spherical harmonic coefficients representative of a sound field in three dimensions so as to render the sound field.

Abstract: Psychoacoustic bass audio signal enhancement can be accomplished using a monotonic, asymmetric polynomial distortion. A non-linear process applies a monotonic, asymmetric polynomial distortion function that has continuous first and second derivatives to generate even and odd harmonics of missing fundamental frequencies. This polynomial distortion produces the desired psychoacoustic effect with a fairly rapid rolloff so as to avoid unpleasant aliasing. Moreover, the lack of first-order discontinuities prevents clicks or glitches.

Abstract: A broadcast transmitting apparatus and method, and a broadcast playback apparatus and method for providing an object-based audio by encoding and decoding a multichannel audio signal are provided. The broadcast transmitting apparatus may generate audio identification information used to determine whether the multichannel audio signal is an object-based audio signal. When the multichannel audio signal is determined to be the object-based audio signal, based on the audio identification information, the broadcast playback apparatus may control and output the multichannel audio signal for each channel.

Abstract: A speaker system comprised of more than one speaker, which utilizes the known characteristics and location of each speaker to enhance low frequency (low (bass) frequency) performance of the system.

Abstract: A sound effect generating apparatus includes a delay unit for giving a delay to a sound effect output from a rear speaker depending on either an accelerator opening change which represents a change per unit time in an accelerator opening or the accelerator opening itself.

Abstract: A device comprising one or more processors is configured to apply adaptively determined weights to a plurality of channels of the audio signal to generate a plurality of adaptively weighted channels of the audio signal. The processors are further configured to combine at least two of the plurality of adaptively weighted channels of the audio signal to generate a combined signal. The processors are further configured to apply a binaural room impulse response filter to the combined signal to generate a binaural audio signal.

Abstract: A control system is provided for a vehicle having an engine which transitions between an activated mode and a deactivated mode. The control system includes a vehicle bus transmitting a signal indicating a vehicle selected mode and if the engine is operating in one of the activated mode and the deactivated mode. The control system also includes an engine sound enhancement (“ESE”) module configured to receive the signal. The ESE module is configured to select at least one ESE tone and a set of ancillary tones associated with one or more of the deactivated mode, the activated mode, and an activation transition. The ESE module selects a specific type of ancillary tones based on the vehicle selected mode.

Abstract: A computer-implemented method for providing a dialog with a user, the method comprising the steps of: (a) creating (501) dialog's plot scenario as a narrative graph structure containing the dialogs; (b) assigning (502) coefficients of dialog transitions maps to narrative graph nodes; (c) providing (503) speech input for recognition in a given context and dialog phase represented by current narrative graph node; (d) applying (504) algorithm for updating dialog coefficients based on user's speech or user's other behavior; (e) applying (505) at least one fuzzy logic algorithm which using user speech, and other coefficients on the transition map, determines transition to another narrative graph node (phase of a dialog or a plot), or updates a position on the transition map continuing the dialog in the same narrative graph node (f) determining (507) a response based on coefficients of the narrative graph; (g) repeating steps (c) to (f) for a particular narrative graph node until a transition is decided in step (e)

Abstract: A vehicle comprises electric motor, generator of simulated engine sounds, decider of whether or not to generate simulated engine sounds and controller of the level of simulated engine sounds. The controller smoothly increases level of simulated engine sounds upon decision of sound generation on detection of pedestrian, crosswalk, narrow road, or road with no sidewalk. The decider is responsive to vehicle navigation system, or ETC, or camera of EDR. EDR records the decision as circumstantial evidence. The decision may be optionally possible, but is forcibly made upon necessity. Simulated engine sounds can be greater than, or equal to, or less than real engine sounds. The controller makes a soft peak of simulated engine sound upon brake or accelerator operated. Balance of simulated engine sounds among front, rear, right and left of vehicle is changeable in response to shift lever or blinker lever operation. The decision is visually indicated inside vehicle.

Abstract: A movement actuated tone effects system is provided for manipulating the electrical signal of an electric instrument by moving the electrical instrument in relation to a set of reference axes. A movement actuated tone effects unit is positioned within the electric instrument or within an effects cartridge attached to the electric instrument via a cartridge receiver. The movement actuated tone effects unit includes a motion sensor unit, a processor, a tone effects circuit, and a power source, wherein the motion sensor unit measures the motion of the electric instrument and generates electrical signals directed to the processor. The processor analyzes the electrical signals, determines the appropriate tone effect to apply, and then directs the tone effects circuit to produce the desired tone effect. The movement actuated tone effects unit may further include an integrated speaker, a signal converter, a transmitter, and a device terminal.

Abstract: Provided is a sound generating device (20) for a vehicle, including: a plurality of pressure sensors (21a to 21d) for detecting air pressures of an air intake sound of an engine (12) and outputting the air pressures as sound pressure signals; a signal processing unit (24) for performing processing of changing the sound pressure signals in accordance with a driving condition of a vehicle (10); and loudspeakers (28a and 28b) for outputting the sound pressure signals as the air intake sound of the engine (12). Then, the plurality of pressure sensors (21a to 21d) are provided at an interval in a circumferential direction of an outer periphery of an air intake duct (15) at positions in a vicinity of an air flow meter (18) on an air cleaner (16) side with respect to a center of the air intake duct (15), the air intake duct (15) connecting the air cleaner (16) to a throttle body (17).

Abstract: The present invention relates to a bass enhancement system comprising a bass substitution filter with variable gain and variable band width, wherein said variable gain and variable bandwidth at least partly depends on characteristics of an input signal. The present invention further relates to a bass enhancement system according to the above, further comprising a hearing compensation filter and a loudspeaker bass response compensation filter, wherein said input signal is coupled to said hearing compensation filter, the output of which is coupled to said loudspeaker bass response filter, the output of which is coupled to said bass substitution filter. The present invention further relates to a method of enhancing bass perception comprising boosting a reproducible bass frequency band in accordance with characteristics of signal content in a substantially non-reproducible bass frequency band.

Abstract: A digital version of both amplitude and phase of a received optical is developed by employing direct differential detection in conjunction with digital signal processing. The signal is split into three copies. An intensity profile is conventionally obtained using one of the copies. Phase information is obtained by supplying each remaining copy to a respective one of a pair of optical delay interferometers that have orthogonal phase offsets, followed by respective balanced intensity detectors. The output of each of the balanced intensity detectors, and the intensity profile, are each converted to respective digital representations. Signal processing is used to develop the phase information from the digital representations of the output of the balanced intensity detector outputs.

Abstract: A tone effects system for use with electric instruments that allows for reduced or eliminated signal loss before sound effects are applied. A cartridge receiver and a changeover switch are integrated into the body of a desired electric instrument, while an effects cartridge is attached to the electric instrument via the cartridge receiver. The cartridge receiver provides electronic connections between the electric instrument and the effects cartridge, while the changeover switch is used to direct the electronic signal of the electric instrument through the effects cartridge. The effects cartridge provides an effects circuit for manipulating the received electronic signal, as well as an at least one effects control for adjusting the extent to which the effect is applied to the electronic signal. As the effects cartridge is attached to the electric instrument, the at least one effects control is readily accessible while the electric instrument is being played.

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: An audio effects chain sequencing apparatus alters the sequence of effects units in an effects chain so that different sequences of effects units can be connected. In one example the audio effects chain sequencing apparatus can have an external input port and an external output port, at least two component ports each having an internal input port and an internal output port, where the at least two component ports receive an input from the external input port, transfer that input in a component sequence to the component ports, and transfer the input to the external output port at the end of the component sequence and having a sequence selection module which selects the component sequence in which the at least two component ports transfer the input.

Abstract: The present invention relates to a control method of a D/A converter, a D/A converter, a control method of an A/D converter, and an A/D converter that can suppress an existing n-th harmonic without using a large-scale circuit, such as a bootstrap. A D/A converter (10) of the present invention is a D/A converter (10) that can suppress the generation of an existing n-th harmonic (n is an integer of 2 or more) of an analog output signal. The D/A converter (10) includes a D/A conversion unit (11) that converts an input digital signal into an analog signal and a control unit (12) that arbitrarily controls the timing of the sampling phase and the integral phase of the D/A conversion unit (11). The D/A conversion unit (11) is configured to generate an arbitrary n-th harmonic and superimpose the arbitrary n-th harmonic on an analog output signal including the existing n-th harmonic.

Abstract: For reproducing sound in a rectangular room in association with a video playback screen (10), from audio signals of the kind comprising at least three main channels (2, 3, 4) and a bass effects channel (1), at least five loudspeakers (11, 12, 13, 14, 15) are used that are specialized for reproducing audio frequencies lower than 120 Hz, four of which (11, 13, 14, 15) are arranged in four respective corners of the room and at least one of them (12) is arranged in the proximity of the screen. The two loudspeakers (14, 15) arranged in the corners furthest away from the screen receive either a signal constituted by the sum of the signals for the other loudspeakers with polarity inversion, with individual delays, and with lowpass filtering, or else a signal coming from a microphone (23, 25) arranged in the immediate proximity of each of the two loudspeakers (14, 15) concerned, the microphone signal being inverted in polarity and lowpass filtered.

Abstract: A system includes a controller for receiving first and second signals representative of a load on an engine, and generating based on the signals. Harmonic scaling elements generate a scaled version of a received engine harmonic signal. Each harmonic scaling element includes a harmonic specific mapping element for mapping the control parameter to a harmonic specific scaling factor. Some of the harmonic specific scaling factor values of the harmonic specific mapping element are mapped to control parameter values associated with a negative load on the engine, and some are mapped to control parameter values associated with a positive load on the engine. The control parameter values associated with the negative load are derived from the first signal, and control parameter values associated with the positive load are derived from the second signal. An adjustable gain element applies the harmonic specific scaling factor to the received engine harmonic signal.

Abstract: A system and method may be configured to reconstruct an audio signal from transformed audio information. The audio signal may be resynthesized based on individual harmonics and corresponding pitches determined from the transformed audio information. Noise may be subtracted from the transformed audio information by interpolating across peak points and across trough points of harmonic pitch paths through the transformed audio information, and subtracting values associated with the trough point interpolations from values associated with the peak point interpolations. Noise between harmonics of the sound may be suppressed in the transformed audio information by centering functions at individual harmonics in the transformed audio information, the functions serving to suppress noise between the harmonics.