Abstract: A noise suppression apparatus using spectral subtraction is provided. A noise estimation unit estimates noise components included in a mixed signal. A fundamental frequency of the mixed signal is detected. A subtraction factor in the spectral subtraction is set based on the detected fundamental frequency. The spectral subtraction for the mixed signal is executed using the set subtraction factor and the estimated noise components. A boundary frequency at the fundamental frequency or a frequency lower than the fundamental frequency is set, and a subtraction factor for a frequency lower than the boundary frequency is set to assume a value larger than a subtraction factor for a frequency not less than the boundary frequency.

Abstract: Comfort noise, such as can be used in voice communications can be generated using methods in the frequency domain and/or in the time domain. In various embodiments, a comfort noise spectrum can be generated in the frequency domain as the product of a background noise sample and a random noise sample. In other embodiments, the comfort noise can be generated directly in the time domain as the convolution of a background noise sample and a random noise sample.

Abstract: A signal processing apparatus includes: one or more detection means for detecting movement of a diaphragm of a speaker in correspondence with feedback methods that are different feedback methods; analog-to-digital conversion means for converting one or more detection signals acquired by the detection means into a digital form; feedback signal generating means for generating feedback signals corresponding to the feedback methods using the digital detection signals; synthesis means for combining an audio signal to be output as a driving signal of the speaker with the feedback signals; correction equalizer means for setting an equalizing characteristic to allow a sound reproduced by the speaker to have a target frequency characteristic by changing the digital audio signal; feedback operation setting means for setting feedback methods in which a feedback operation up to combining the audio signal with the feedback signal is performed and the feedback operation is not performed equalizing characteristic changing a

Abstract: One aspect of the invention provides a method for enhancing speech output by an electro-acoustical transducer in a noisy listening environment. In some embodiments, this method includes: filtering an input audio signal x(t) using a filter H(z) to produce a filtered audio signal x(t) formula (I), wherein x(t) formula (I)—H(z)x(t); providing to an electro-acoustical transducer a signal corresponding to the filtered audio signal x(t) formula (I) to produce a sound wave corresponding to the filtered audio signal; and prior to filtering the audio signal using the filter, configuring the filter such that, with respect to one or more frequencies, the filtered audio signal has a higher signal level than the input audio signal, and such that the overall signal level of the filtered audio signal (slƒ) is substantially related to the overall signal level of the input signal (slr) such that si/=sl/×c.

Abstract: A sound reproduction device includes a modulator having an output terminal for outputting a modulated carrier wave signal obtained by modulating a carrier wave signal in a ultrasonic band with an audible sound signal, a super-directivity loudspeaker connected to the output terminal, a capacitor connected between a ultrasonic wave source and a ground, first and second current detectors for detecting currents flowing through the super-directivity loudspeaker and the capacitor, a high-pass filter for outputting a filtered signal obtained by eliminating a low-frequency band component of the current detected by the first current detector, and a differential amplifier unit for outputting a signal corresponding to a difference between the filtered signal and the current detected by the second current detector. The ultrasonic wave source is configured to output the carrier wave signal such that the signal output from the differential amplifier unit is constant.

Abstract: Zoom motor noise in a camera audio recording is reduced by detecting activity of the zoom motor, transforming a audio signal into the frequency domain during zoom motor activity, and scaling the frequency domain signal during zoom motor activity in each of a series of frequency bins by a scaling factor derived from a pre-stored zoom motor noise spectrum to produce a processed audio signal in the frequency domain. The processed audio signal is then transformed back to the time domain.

Abstract: Provided are systems and methods for dynamic detection of changes in signal bandwidth associated with audio samples received by a communication device during handovers, where the communication device is moved through geographical areas served by different wireless network technologies. Based on the detection, operation parameters of an audio processor of the communication device may be adjusted or switched to achieve optimal performance of audio enhancing procedures (e.g., noise suppression) performed by the audio processor and to preserve minimal power consumption.

Abstract: An audio processing device includes a setting section that sets a reproduction sampling frequency Fplay and a recording sampling frequency Frec higher than Fplay, a digital-to-analogue converter that based on Fplay converts a sound source signal that is a digital signal into a reproduction signal that is an analogue signal, an analogue-to-digital converter that based on Frec converts a recording signal that is an analogue signal converter into an input signal that is a digital signal, a signal separator that separates the input signal into a low region signal contained in a band of less than Fplay and a high region signal contained in a band of the Fplay and higher, and a breakup detector that detects whether or not breakup is occurring in the reproduced sound based on power of the high region signal.

Abstract: A speech signal processing device is equipped with a power acquisition unit, a probability distribution acquisition unit, and a correspondence degree determination unit. The power acquisition unit accepts an inputted speech signal and, based on the accepted speech signal, acquires power representing the intensity of a speech sound represented by the speech signal. The probability distribution acquisition unit acquires a probability distribution using the intensity of the power acquired by the power acquisition unit as a random variable. The correspondence degree determination unit determines whether a correspondence degree representing a degree that power acquired by the power acquisition unit in a case that a predetermined reference speech signal is inputted into the power acquisition unit corresponds with predetermined reference power is higher than a predetermined reference correspondence degree, based on the probability distribution acquired by the probability distribution acquisition unit.

Abstract: A correction spectrum calculation unit 6 obtains a correction spectrum by smoothing an estimated noise spectrum in accordance with the degree of its variations, and a suppression quantity limiting coefficient calculation unit 7 decides a suppression quantity limiting coefficient from the correction spectrum. A suppression quantity calculation unit 9 obtains a suppression coefficient based on the suppression quantity limiting coefficient, and the spectrum suppression unit 10 carries out amplitude suppression of spectral components of an input signal.

Abstract: The present invention provides a spatially pre-processed target-to-jammer ratio weighted filter and a method thereof, which uses two microphones to receive audio signals. The audio signals are divided into a plurality of sinusoidal waves by a fast Fourier transform (FFT) module, and a beamformer uses the sinusoidal waves to generate beamformed signals. A reference generator generates at least one reference signal. The beamformed signals and reference signals are used to work out power spectral densities (PSD), and a target-to-jammer ratio (TJR) is worked out with the power spectral densities. TJR is used to determine whether a sound source exists. According to the determination result, a noise estimator is switched to eliminate noise from the beamformed signals and generate output signals. An inverse fast Fourier transform (IFFT) module recombines the output signals and then outputs the recombined signals.

Abstract: A method and system to reduce the noise floor of a communications system is disclosed. The system may be incorporated into any device that provides binary samples from a datastream, such as a cordless telephone system. The system is configured to determine a number of bits of the binary samples that are affected by noise. The system is then able to remove the noise by setting those bits to a fixed value. The fixed value may depend on whether the sample is positive or negative. The value to set may be chosen so that the least significant bits of each sample come as close as possible to 0 for that particular numerical representation system. The system can be integrated with other known signal processing methods.

Abstract: A method of combining at least two audio signals for generating an enhanced system output signal is described.

Abstract: A novel system prevents surrounding sound to enter through a hearing apparatus, for instance through a ventilation opening, and reach an eardrum of the wearer in the form of interference sound. Contrary to auditory accessories designed especially to protect against noise, it is not possible for many hearing apparatus to compensate for such an interference sound by means of active noise cancellation. The hearing apparatuses do not have the special components needed. No compensation sound signal can therefore form with a correct phase. In accordance with the invention, a compensation sound is only generated for a relatively narrow spectral band. This spectral band is determined as a function of a hearing ability of the wearer of the hearing apparatus and/or as a function of a spectral distribution of the energy of the interference sound or a sound producing the interference sound. The improvement is particularly suited to compensating for an interference sound in a hearing device.

Abstract: A standalone or integrated electronic unit for selectively modifying audio or video data for use with an output device. The electronic unit has one or more audio or video inputs and one or more audio or video outputs for outputting the modified audio or video source material. The unit may include sensors, selectors, and/or logic for selecting from among a plurality of modification profiles and applies the selected profile to the audio or video source material before conveying it to the output device. The modification profile may be used to adapt an audio or video source for a specific type of audio or video playback or output device. For audio, the modification profile may be used to modify audio quality by altering the transfer function, and for video, the modification profile may be used to modify video quality by altering the visual characteristics of the video source signal.

Abstract: A method and apparatus are provided for estimating the noise spectrum of an audio signal occupying a certain frequency range. The amplitude of the audio signal is measured at only a subset of the frequency range having a low expected signal to noise ratio at the time of measuring such that the measured amplitude is indicative of noise amplitude at that subset of the frequency range. A predefined noise spectral shape is then adapted by fitting to the noise amplitude so as to form the estimated noise spectrum. The noise spectrum so estimated is used to remove interference noise from the audio signal.

Abstract: Processing a signal includes: receiving data that includes an input signal; filtering the input signal to generate a filtered signal, such that if the input signal includes at least one instance of a nonlinear distortion of a desired signal then the filtered signal includes a signature signal corresponding to the nonlinear distortion, the nonlinear distortion characterized by a time duration that is within a predetermined range; and detecting whether or not the filtered signal includes the signature signal.

Abstract: A system and method for use in filtering of an acoustic signal are provided for producing an output signal of attenuated amount of diffuse sound in accordance with predetermined parameters of desired output directional response and required attenuation of diffuse sound. The system includes a filtration module and a filter generation module including a directional analysis module and filter construction module.

Abstract: A method and system are provided in which a device, such as an acoustic echo canceller, may reduce the residual echo that may be heard at the far end of a conversation when an external speaker volume is changed. The device may compute a gain based on an echo estimate produced by a filter and on a near-end signal comprising audio information. The gain may be based on a correlation of the echo estimate and the near-end signal that tracks the changes in volume. Once computed, the gain may be validated to ensure that it is being applied when appropriate. The echo estimate may be adjusted by first applying the gain to an output of the filter and subsequently scaling a value of each of the coefficients of the filter based on the gain. The gain may be smoothed out over consecutive frames based on several adaptation schemes.

Abstract: There is provided a signal processing apparatus, for suppressing a noise, which includes a first calculator to obtain a phase difference between two spectrum signals in a frequency domain transformed from sound signals received by at least two microphones to estimate a sound source by the phase difference, a second calculator to obtain a value representing a target signal likelihood and to determine a sound suppressing phase difference range at each frequency, in which a sound signal is suppressed, on the basis of the target signal likelihood, and a filter. The filter generate a synchronized spectrum signal by synchronizing each frequency component of one of the two spectrum signals to each frequency component of the other of the two spectrum signals for each frequency when the phase difference is within the sound suppressing phase difference range and to generate a filtered spectrum signal.

Abstract: An audio decoder (100) comprising: effect means, decoding means, and rendering means. The effect means (500) generate modified down-mix audio signals from received down-mix audio signals. Said received down-mix audio signals comprise a down-mix of a plurality of audio objects. Said modified down-mix audio signals are obtained by applying effects to estimated audio signals corresponding to audio objects comprised in said received down-mix audio signals. Said estimated audio signals are derived from the received down-mix audio signals based on received parametric data. Said received parametric data comprise a plurality of object parameters for each of the plurality of audio objects. Said modified down-mix audio signals based on a type of the applied effect are decoded by decoding means or rendered by rendering means or combined with the output of rendering means.

Abstract: A multiband dynamics compressor implements a solution for minimizing unwanted changes to the long-term frequency response. The solution essentially proposes undoing the multiband compression in a controlled manner using much slower smoothing times. In this regard, the compensation provided acts more like an equalizer than a compressor. What is applied is a very slowly time-varying, frequency-dependent post-gain (make-up gain) that attempts to restore the smoothed long-term level of each compressor band.

Abstract: The present invention relates to a loudspeaker device, comprising first and second closely located and individually acoustically isolated loudspeaker elements. The first and second elements are arranged to receive a first signal and a second signal, respectively, at least part of said first signal being in anti-phase relative to said second signal. The device further includes third and fourth loudspeaker elements, being located in close proximity to said first and second loudspeaker elements, respectively.

Abstract: A sound processing device includes: a plurality of sound input units; a detecting unit for detecting a frequency component of each sound input to the plurality of sound signal unit, the each sound arriving from a direction approximately perpendicular to a line determined by arrangement positions of two sound input units among the plurality of sound input units; a correction coefficient unit for obtaining a correction coefficient for correcting a level of at least one of the sound signals generated from the input sounds by the two sound input units so as to match the levels of the sound signals with each other based on the sound of the detected frequency component; a correcting unit for correcting the level of at least one of the sound signals using the obtained correction coefficient; and a processing unit for performing a sound process based on the sound signal with the corrected level.

Abstract: The specification and drawings present a new method, apparatus and software product for calibrating multiple microphones (e.g., a microphone array) to match their sensitivity using an ambient noise by creating and updating one or more calibration signal level difference histograms.

Abstract: Improved systems and methods for psychoacoustic adaptive notch filtering are provided. By accounting for psychoacoustic properties of an audio signal as well as finer characteristics of noise which may be present in the audio signal (e.g., the shape of the spectral density of the noise), more effective strategies for dealing with undesirable components of the audio signal may be realized.

Abstract: Provided is an audio processing apparatus including a frequency domain conversion unit configured to convert an audio signal input from a microphone to a frequency domain for each of frames, and a gain adjustment unit configured to perform gain adjustment for each of bands on the audio signal converted to the frequency domain. The gain adjustment unit acquires an autocorrelation value of power of the audio signal between the frames for each of the bands, and sets an adjustment amount of the gain in accordance with the acquired autocorrelation value.

Abstract: A method of suppressing wind noise in a voice signal determines an upper frequency limit that lies within the frequency spectrum of the voice signal, and for each of a plurality of frequency bands below the upper frequency limit, compares the average power of signal components in a first portion of the signal to the average power of signal components in a second portion of the signal, where the second portion is successive to the first portion. Signal components are identified in at least one of the plurality of frequency bands as containing impulsive wind noise in dependence on the comparison, and the identified signal components are attenuated.

Abstract: An audio data processing apparatus and method to reduce wind noise. The apparatus includes a wind noise detecting unit to detect a wind noise section from an input audio signal, and a signal processing unit to reduce a low-frequency band signal of the input audio signal in the detected wind noise section. The apparatus determines whether wind is present and automatically reduces wind noise based on the determined result. Accordingly, the apparatus can effectively reduce wind noise.

Abstract: Enhancements to hardware architectures (e.g., a RISC processor or a DSP processor) to accelerate spectral band replication (SBR) processing are described. In some embodiments, instruction extensions configure a reconfigurable processor to accelerate SBR and other audio processing. In addition to the instruction extensions, execution units (e.g., multiplication and accumulation units (MACs)) may operate in parallel to reduce the number of audio processing cycles. Performance may be further enhanced through the use of source and destination units which are configured to work with the execution units and quickly fetch and store source and destination operands.

Abstract: It is described a method for controlling an adaptation of a behavior of an audio device (100) to a current acoustic environmental condition. The method comprises (a) monitoring an audio output signal (x(t), x?(t)) being provided to an acoustic output device (110) of the audio device (100) for outputting an acoustic output signal, (b) measuring an audio input signal (z(t)) being provided by an acoustic input device (120) of the audio device (100), wherein the audio input signal (z(t)) is indicative for a feedback portion of the acoustic output signal and for the current acoustic environmental condition, (c) determining a relation between the audio output signal (x?(t)) and the audio input signal (z(t)) and (d) adapting the behavior of the audio device (100) based on the determined relation.

Abstract: In one aspect thereof the invention provides a method for noise suppression of a speech signal that includes, for a speech signal having a frequency domain representation dividable into a plurality of frequency bins, determining a value of a scaling gain for at least some of said frequency bins and calculating smoothed scaling gain values. Calculating smoothed scaling gain values includes, for the at least some of the frequency bins, combining a currently determined value of the scaling gain and a previously determined value of the smoothed scaling gain. In another aspect a method partitions the plurality of frequency bins into a first set of contiguous frequency bins and a second set of contiguous frequency bins having a boundary frequency there between, where the boundary frequency differentiates between noise suppression techniques, and changes a value of the boundary frequency as a function of the spectral content of the speech signal.

Abstract: A system and method for enhancing the sound signal produced by an audio system in a listening environment by compensating for ambient sound in the listening environment, comprises producing an audio sound in the time domain from an electrical sound signal in the time domain. The electrical sound signal in the time domain is transformed into an electrical sound signal in the frequency domain and the electrical sound signal in the frequency domain is retransformed into an audio sound in the time domain. The total sound level in the environment is measured and a signal representative thereof is generated. The audio sound signal and the total sound signal are processed to extract a signal representing the ambient sound level within the environment, and equalization is performed in the frequency domain to adjust the output from the audio sound signal to compensate for the ambient noise level.

Abstract: An electronic device for suppressing noise in an audio signal is described. The electronic device includes a processor and instructions stored in memory. The electronic device receives an input audio signal and computes an overall noise estimate based on a stationary noise estimate, a non-stationary noise estimate and an excess noise estimate. The electronic device also computes an adaptive factor based on an input Signal-to-Noise Ratio (SNR) and one or more SNR limits. A set of gains is also computed using a spectral expansion gain function. The spectral expansion gain function is based on the overall noise estimate and the adaptive factor. The electronic device also applies the set of gains to the input audio signal to produce a noise-suppressed audio signal and provides the noise-suppressed audio signal.

Abstract: A device and method processing microphone signals from at least two microphones is presented. A first beamformer processes the signals from the microphones and provides a first beamformed signal. A power estimator processes the signals from the microphones and the first beamformed signal from the first beamformer in order to generate, in frequency bands, a first statistical estimate of the energy of a first part of an incident sound field. A gain controller processes said first statistical estimate in order to generate in frequency bands a first gain signal, and an audio processor for processing an input to the signal processing device in dependence of said generated first gain signal. The invention provides a new and improved noise reduction device and noise reduction method for use in the signal processing in devices processing acoustic signals, e.g. microphone devices.

Abstract: A method and an apparatus for removing white noise in a portable terminal are provided. The method for removing the white noise in the portable terminal includes measuring a volume variation of a voice signal output from a power amplifier; detecting a frequency band including white noise using the measured volume variation; and removing signals of the detected frequency band in the voice signal before output to speaker.

Abstract: A robust noise reduction system may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. The system may receive acoustic signals from two or more microphones in a close-talk, hand-held or other configuration. The received acoustic signals are transformed to frequency domain sub-band signals and echo and noise components may be subtracted from the sub-band signals. Features in the acoustic sub-band signals are identified and used to generate a multiplicative mask. The multiplicative mask is applied to the noise subtracted sub-band signals and the sub-band signals are reconstructed in the time domain.

Abstract: A system and method for dynamic residual noise shaping configured to reduce hiss noise in an audio signal. The system and method may detect an amount and type of hiss noise. The system and method may limit calculated noise suppression gains responsive to the detected amount and type of hiss noise. The limited noise suppression gains may be applied to the audio signal and may reduce the hiss noise.

Abstract: A device for processing parameters representing Head-Related Transfer Functions includes an input stage configured to receive audio signals of sound sources, a determinor configured to receive reference parameters representing Head-Related Transfer Functions and configured to determine, from the audio signals, position information representing positions and/or directions of the sound sources. A processor is configured to process the audio signals; and an influencer is configured to influence the processing of the audio signals based on the position information yielding an influenced output audio signal.

Abstract: A signal processing apparatus is configured to change volume level or frequency characteristics of an input signal with a limited bandwidth in a first frequency range. The apparatus includes: an information extracting unit configured to extract second frequency characteristic information from a collection signal with a limited bandwidth in a second frequency range different from the first frequency range; a frequency characteristic information extending unit configured to estimate first frequency characteristic information from the second frequency characteristic information extracted by the information extracting unit, the first frequency characteristic information including the first frequency range; and a signal correcting unit configured to change volume level or frequency characteristics of the input signal according to the first frequency characteristic information obtained by the frequency characteristic information extending unit.

Abstract: A technique for suppressing non-stationary noise, such as wind noise, in an audio signal is described. In accordance with the technique, a series of frames of the audio signal is analyzed to detect whether the audio signal comprises non-stationary noise. If it is detected that the audio signal comprises non-stationary noise, a number of steps are performed. In accordance with these steps, a determination is made as to whether a frame of the audio signal comprises non-stationary noise or speech and non-stationary noise. If it is determined that the frame comprises non-stationary noise, a first filter is applied to the frame and if it is determined that the frame comprises speech and non-stationary noise, a second filter is applied to the frame.

Abstract: A method of enhancing an audio signal includes the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-

Abstract: According to one embodiment, an acoustic correction apparatus includes an input module, a calculator, a divider, a converter, an extractor, a synthesizer, and a generator. The input module receives an audio signal propagated through a sound field. The calculator calculates an impulse response from the audio signal. The divider divides the impulse response into first and second impulse responses. The converter converts the first and second impulse responses into first and second frequency spectrums. The extractor specifies an amplitude component of the first frequency spectrum with a peak relatively higher than that of the amplitude component of the first frequency spectrum, and extracts the peak as a resonance component. The synthesizer synthesizes a first property and a second property for attenuating the resonance component. The generator generates a correction filter for performing correction to obtain the synthesized property.

Abstract: Systems and methods provided herein decrease an idle channel noise floor and reduce power during an idle channel input for low power audio devices that include a digital pulse width modulation (PWM) amplifier having a noise shaper. An audio data signal is monitored for an idle channel condition. The noise shaper performs quantization of the audio data signal and uses noise shaper filter coefficients to shape noise resulting from the quantization. Predetermined values for the noise shaper filter coefficients are used to shape the noise resulting from quantization while the idle channel condition is not being detected. The values of the noise shaper filter coefficients are reduced so that the values move toward zeros, and the reduced values of the noise shaper filter coefficients are used to attenuate noise resulting from quantization, while the idle channel condition is being detected.

Abstract: The invention relates to a method for sequencing spectral components of elements to be encoded (A1, . . . , AQ) originating from an audio scene comprising N signals (Sii=1 to N), in which N>1, an element to be encoded comprising spectral components associated with respective spectral bands, characterised in that it comprises the following steps: calculation of the respective influence of at least some spectral components which can be calculated as a function of the spectral parameters originating from at least some of the N signals on the mask-to-noise ratios determined over the spectral bands as a function of the encoding of said spectral components; and allocation of an order of priority to at least one spectral component as a function of the influence calculated for said spectral component compared to the other influences calculated.

Abstract: An audio signal compensation device includes: a signal processor configured to perform filtering on an input audio signal; a filter coefficients storage module configured to store a plurality of filter coefficients; a user interface configured to provide options for a determination of filter coefficients to a user and to obtain a selection result from the user; and a filter coefficients determining module configured to determine a set of filter coefficients among the plurality of filter coefficients based on the selection result.

Abstract: A system for detecting noise in a signal received by a microphone array and a method for detecting noise in a signal received by a microphone array is disclosed. The system also provides for the reduction of noise in a signal received by a microphone array and a method for reducing noise in a signal received by a microphone array. The signal to noise ratio in handsfree systems may be improved, particularly in handsfree systems present in a vehicular environment.

Abstract: Various embodiments reduce noise within a particular environment, while isolating and capturing speech in a manner that allows operation within an otherwise noisy environment. In one embodiment, an array of one or more microphones is used to selectively eliminate noise emanating from known, generally fixed locations, and pass signals from a pre-specified region or regions with reduced distortion.

Abstract: An apparatus and a method for canceling noise in a voice signal in an electronic apparatus are provided. The apparatus includes a Generalized Sidelobe Canceller (GSC) and a decision unit. The GSC cancels noise components from signals with different phases input via a plurality of microphones. The decision unit estimates a Signal-to-Noise Ratio (SNR) of an input signal to determine a step-size of a filter included in the GSC.

Abstract: The invention relates to a method for ordering spectral parameters of ambisonic components to be encoded (A1, . . . , AQ) originating from an audio scene comprising N signals (Sii=1 to N), in which N>1, comprising the following steps: calculation of the respective influence of at least some spectral parameters, taken from a set of spectral parameters to be ordered, on an angle vector defined as a function of energy and velocity vectors associated with Gerzon's criteria and calculated as a function of a reverse ambisonic transformation in relation to said quantified ambisonic components; and allocation of a precedence order to at least one spectral parameter as a function of the influence calculated for said spectral parameter compared to the other calculated influences.