Abstract: A voice recognizing apparatus includes a microphone 12 which inputs an input voice including speech voice uttered by a user speaker and interference voice uttered by an interference speaker other than the user speaker, superimposition amount determining unit 14 which determines a noise superimposition amount for the input voice on the basis of a speech voice and an interference voice separately input as the input voice, a noise superimposing unit 16 which superimposes noise according to the noise superimposition amount onto the input voice and outputs the resultant voice as noise-superimposed voice; and a voice recognizing unit 18 which recognizes the noise-superimposed voice.

Abstract: An enhancement system extracts pitch from a processed speech signal. The system estimates the pitch of voiced speech by deriving filter coefficients of an adaptive filter and using the obtained filter coefficients to derive pitch. The pitch estimation may be enhanced by using various techniques to condition the input speech signal, such as spectral modification of the background noise and the speech signal, and/or reduction of the tonal noise from the speech signal.

Abstract: An enhancement system improves the estimate of noise from a received signal. The system includes a spectrum monitor that divides a portion of the signal at more than one frequency resolution. Adaptation logic derives a noise adaptation factor of the received signal. A plurality of devices tracks the characteristics of an estimated noise in the received signal and modifies multiple noise adaptation rates. Weighting logic applies the modified noise adaptation rates derived from the signal divided at a first frequency resolution to the signal divided at a second frequency resolution.

Abstract: In one embodiment, a signal processing method is disclosed. The method can perform filter processing of convoluting a tap coefficient in a first signal sequence to generate a second signal sequence. The method can subtract the second signal sequence from a third signal sequence to generate a fourth signal sequence. The third signal sequence includes an echo signal of the first signal sequence. The method can correct the tap coefficient in accordance with an amount of correction determined using a function. The function includes at least one of a first region and a second region, and has values limited. The first region is included in a negative value region of the fourth signal sequence. The second region is included in a positive value region of the fourth signal sequence.

Abstract: An audio privacy system reduces the intelligibility of speech in an audio signal while preserving prosodic information, such as pitch, relative energy and intonation so that a listener has the ability to recognize environmental sounds but not the speech itself. An audio signal is processed to separate non-vocalic information, such as pitch and relative energy of speech, from vocalic regions, after which syllables are identified within the vocalic regions. Representations of the vocalic regions are computed to produce a vocal tract transfer function and an excitation. The vocal tract transfer function for each syllable is then replaced with the vocal tract transfer function from another prerecorded vocalic sound. In one aspect, the identity of the replacement vocalic sound is independent of the identity of the syllable being replaced.

Abstract: The present invention provides a method and system for defining the mean opinion score (MOS) as a function of frame error rate (FER) and pilot signal strength. In an embodiment of the invention, an entity receives MOS scores that have been obtained using subjective tests for certain calls made within the network. Next, the entity receives FER and pilot signal strength samples that have been obtained for the calls for which MOS scores have been subjectively obtained. Finally, the entity calculates an equation for the MOS as a function of FER and pilot signal strength using a non-linear regression analysis.

Abstract: Systems and methods for intelligent control of microphones in speech processing applications, which allows the capturing, recording and preprocessing of speech data in the captured audio in a way that optimizes speech decoding accuracy.

Abstract: The systems and methods described herein may filter and eliminate noise from natural language utterances to improve accuracy associated with speech recognition and parsing capabilities. In particular, the systems and methods described herein may use a microphone array to provide directional signal capture, noise elimination, and cross-talk reduction associated with an input speech signal. Furthermore, a filter arranged between the microphone array and a speech coder may use band shaping, notch filtering, and adaptive echo cancellation to optimize a signal-to-noise ratio associated with the speech signal. The speech signal may then be sent to the speech coder, which may use adaptive lossy audio compression to optimize bandwidth requirements associated with transmitting the speech signal to a main unit that provides the speech recognition, parsing, and other natural language processing capabilities.

Abstract: Various embodiments for components and associated methods that can be used in a binaural speech enhancement system are described. The components can be used, for example, as a pre-processor for a hearing instrument and provide binaural output signals based on binaural sets of spatially distinct input signals that include one or more input signals. The binaural signal processing can be performed by at least one of a binaural spatial noise reduction unit and a perceptual binaural speech enhancement unit. The binaural spatial noise reduction unit performs noise reduction while preferably preserving the binaural cues of the sound sources. The perceptual binaural speech enhancement unit is based on auditory scene analysis and uses acoustic cues to segregate speech components from noise components in the input signals and to enhance the speech components in the binaural output signals.

Abstract: Enhanced estimation of the noise component of a signal is accomplished by using a plurality of filters. Each filter provides an estimate of a minimum sample in a sample set that includes a plurality of signal samples. A comparator, coupled to the plurality of filters, successively compares the estimates among the plurality of filters, and selects the signal estimate having the lowest magnitude. The selected signal estimate represents an enhanced estimate of the noise component of the signal.

Abstract: A howling canceller which suppresses occurrence of howling even when an open loop gain exceeds “1” in the whole reproduction band. In the howling canceller, an adaptive filter (107) operates a digital received voice signal with a tap coefficient to generate a pseudo echo; a subtractor (108) subtracts the pseudo echo from a digital transmitted voice signal to generate a residual signal; and an amplitude limiting circuit (110) limits the absolute value of the amplitude of the digital received voice signal to be equal to or smaller than a predetermined threshold which ensures that all of a D/A converter (101), a power amplifier (102), a speaker (103), a microphone (104), a microphone amplifier (105), and an A/D converter (106) operate in a linear operation area, and outputs the amplitude-limited digital received voice signal to the D/A converter (101) and the adaptive filter (107).

Abstract: A method and device are provided for the objective evaluation of voice quality of a speech signal. The device includes: a module for extracting a background noise signal, referred to as a noise signal, from the speech signal; a module for calculating the audio parameters of the noise signal; a module for classifying the background noise contained in the noise signal on the basis of the calculated audio parameters, according to a predefined set of background noise classes; and a module for evaluating the voice quality of the speech signal on the basis of at least the resulting classification relative to the background noise in the speech signal.

Abstract: Acoustic noise suppression is provided in multiple-microphone systems using Voice Activity Detectors (VAD). A host system receives acoustic signals via multiple microphones. The system also receives information on the vibration of human tissue associated with human voicing activity via the VAD. In response, the system generates a transfer function representative of the received acoustic signals upon determining that voicing information is absent from the received acoustic signals during at least one specified period of time. The system removes noise from the received acoustic signals using the transfer function, thereby producing a denoised acoustic data stream.

Abstract: A two microphone noise reduction system is described. In an embodiment, input signals from each of the microphones are divided into subbands and each subband is then filtered independently to separate noise and desired signals and to suppress non-stationary and stationary noise. Filtering methods used include adaptive decorrelation filtering. A post-processing module using adaptive noise cancellation like filtering algorithms may be used to further suppress stationary and non-stationary noise in the output signals from the adaptive decorrelation filtering and a single microphone noise reduction algorithm may be used to further provide optimal stationary noise reduction performance of the system.

Abstract: A system distinguishes a primary audio source and background noise to improve the quality of an audio signal. A speech signal from a microphone may be improved by identifying and dampening background noise to enhance speech. Stochastic models may be used to model speech and to model background noise. The models may determine which portions of the signal are speech and which portions are noise. The distinction may be used to improve the signal's quality, and for speaker identification or verification.

Abstract: The subject matter of this specification can be embodied in, among other things, a method that includes receiving an audio signal, determining an energy-independent component of a portion of the audio signal associated with a spectral shape of the portion, and determining an energy-dependent component of the portion associated with a gain level of the portion. The method also comprises comparing the energy-independent and energy-dependent components to a speech model, comparing the energy-independent and energy-dependent components to a noise model, and outputting an indication whether the portion of the audio signal more closely corresponds to the speech model or to the noise model based on the comparisons.

Abstract: An apparatus and a method for voice communication of a mobile terminal are provided. More particularly, an apparatus and a method for clearly receiving a counterpart user's voice signal in a mobile terminal positioned at a place where a noise occurs are provided. The apparatus includes an input unit, an extension signal generator, and an adder. The input unit receives a voice signal. The extension signal generator generates, based on a voice signal received via the input unit, a harmonics signal corresponding to a frequency band that represents a reaction sensitive to a sense of hearing. The adder merges the generated harmonics signal with the received voice signal.

Abstract: Disclosed herein are system, method and apparatus with environmental noise cancellation. The instant disclosure is particularly adapted to a receiver module having at least two inputs. The two inputs respectively receive a main audio portion and the audio with majority of environmental noise. The system firstly calibrates the audio signals to reduce the error caused by the difference between the two inputs. An adaptive beamforming technology and a speech extractor are respectively used to extract the environmental noise portion with less main audio and the main audio portion with less noise. After a process of time-to-frequency domain transformation, a non-linear noise suppression technology is introduced into estimating the environmental noise and acquiring a gain. After noise suppression processed with the gain, a sequence of audio signals is output after a frequency-to-time domain transformation.

Abstract: Automatic level control of speech portions of an audio signal is provided. An audio signal is received in the form of a sequence of samples and may contain speech portion and non-speech portions. The sequence of samples is divided into a sequence of sub-frames. Multiple sub-frames adjacent to a present sub-frame are examined to determine a peak value of samples in the sub-frames. A gain factor is computed for the present sub-frame based on the peak value and a desired maximum value for said speech portion, and each sample in the present sub-frame is amplified by the gain factor. In an embodiment, variations in filtered energy values of multiple sub-frames enable determination of whether a sub-frame corresponds to a speech or non-speech/noise portion.

Abstract: Disclosed herein are systems, methods, and computer-readable storage media for processing a message received from a user to determine whether an estimate of intelligibility is below an intelligibility threshold. The method includes recognizing a portion of a user's message that contains the one or more expected utterances from a critical information list, calculating an estimate of intelligibility for the recognized portion of the user's message that contains the one or more expected utterances, and prompting the user to repeat at least the recognized portion of the user's message if the calculated estimate of intelligibility for the recognized portion of the user's message is below an intelligibility threshold. In one aspect, the method further includes prompting the user to repeat at least a portion of the message if any of a measured speech level and a measured signal-to-noise ratio of the user's message are determined to be below their respective thresholds.

Abstract: A method of estimating the reverberations in an acoustic signal (y) comprises determining the frequency spectrum (Y) of the signal (y), providing a first parameter (?) indicative of the decay of the reverberations part (r) of the signal over time, and providing a second parameter (?) indicative of the amplitude of the direct part (d) of the signal relative to the reverberations part (r). An estimated frequency spectrum ({circumflex over (R)}) of the reverberations signal (r) is produced using the frequency spectrum (Y) of a previous frame, the first parameter (?), and the second parameter (?). The second parameter (?) is preferably inversely proportional to the early-to-late ratio of the signal (y).

Abstract: A speech enhancement system improves the speech quality and intelligibility of a speech signal. The system includes a time-to-frequency converter that converts segments of a speech signal into frequency bands. A signal detector measures the signal power of the frequency bands of each speech segment. A background noise estimator measures a background noise detected in the speech signal. A dynamic noise reduction controller dynamically models the background noise in the speech signal. The speech enhancement renders a speech signal perceptually pleasing to a listener by dynamically attenuating a portion of the noise that occurs in a portion of the spectrum of the speech signal.

Abstract: A noise estimation apparatus includes a correlation calculator configured to calculate a correlation value of a spectrum between a plurality of frames in sound information obtained using one or more microphones, a power calculator configured to calculate a power value indicating a sound level of one target frame among the plurality of frames, an update determiner configured to determine an update degree indicating a degree to which the sound information of the target frame is to be reflected in a noise model stored in a storage, or determine whether or not the noise model is to be updated to another noise model, based on the power value of the target frame and the correlation value, and an updater configured to generate the other noise model based on a determined result, the sound information of the target frame, and the noise model.

Abstract: The present invention is a system and method that improves upon voice activity detection by packetizing actual noise signals, typically background noise. In accordance with the present invention an access network receives an input voice signal (including noise) and converts the input voice signal into a packetized voice signal. The packetized voice signal is transmitted via a network to an egress network. The egress network receives the packetized voice signal, converts the packetized voice signal into an output voice signal, and outputs the output voice signal. The egress network also extracts and stores noise packets from the received packetized voice signal and converts the packetized noise signal into an output noise signal. When the access network ceases to receive the input voice signal while the call is still ongoing, the access network instructs the egress network to continually output the output noise signal.

Abstract: An audio apparatus has a function of correcting an audio signal in response to a noise level. The audio apparatus includes a correction unit that corrects an input audio signal on the basis of a weighting factor, an output unit that produces a played-back audio sound on the basis of the corrected audio signal, a microphone for receiving an external sound that includes the played-back audio sound and noise, a noise-extracting unit that extracts a noise signal from an external sound signal, the noise-extracting unit including a speech-removing unit that removes a speech signal from the noise signal on the basis of noise spectrum data, and a weighting factor calculation unit that calculates the weighting factor on the basis of the extracted noise signal and supplies the calculated weighting factor to the correction unit.

Abstract: A sound source signal from a target sound source is allowed to be separated from a mixed sound which consists of sound source signals emitted from a plurality of sound sources without being affected by uneven sensitivity of microphone elements. A beamformer section 3 of a source separation device 1 performs beamforming processing for attenuating sound source signals arriving from directions symmetrical with respect to a perpendicular line to a straight line connecting two microphones 10 and 11 respectively by multiplying output signals from the microphones 10 and 11 after spectrum analysis by weighted coefficients which are complex conjugate to each other. Power computation sections 40 and 41 compute power spectrum information, and target sound spectrum extraction sections 50 and 51 extract spectrum information of a target sound source based on a difference between the power spectrum information.

Abstract: A method and an apparatus for transmitting a speech signal are provided. A speech signal transmitter includes a quadrature mirror filter, a base sub-band encoder, an enhancement sub-band encoder, and a network connector. The quadrature mirror filter receives a speech signal, divides the speech signal into an enhancement band speech signal and a base band speech signal, and outputs the enhancement band speech signal and the base band speech signal. The base sub-band encoder receives and encodes the base band speech signal. The enhancement sub-band encoder receives and encodes the enhancement band speech signal. The network connector multiplexes the encoded enhancement band speech signal and the encoded base band speech signal based on the kinds of networks over which speech signals are transmitted, and transmits the multiplexed signals to the networks. A speech signal is multiplexed and transmitted by various methods based on the kinds of networks. Thus, the speech signal can be efficiently transmitted.

Abstract: According to one embodiment, there is provided a sound quality control apparatus, including: a characteristic parameter extractor; a speech score calculator; a music score calculator; a power value acquisition module; a first storage configured to store speech scores and music scores; a second storage configured to store power values; a power-based score corrector configured to correct a current music score or a current speech score based on a first comparison result between a current power value and past power values, a second comparison result between the current music score and past music scores and a third comparison result between the current speech score and past speech scores; and a sound quality controller configured to perform a sound quality control by using at least one of the speech score and the music score corrected by the power-based score corrector.

Abstract: A method for processing multichannel acoustic signals which processes input signals of a plurality of channels including the voices of a plurality of speaking persons. The method is characterized by detecting the voice section of each speaking person or each channel, detecting overlapped sections wherein the detected voice sections are common between channels, determining a channel to be subjected to crosstalk removal and the section thereof by use of at least voice sections not including the detected overlapped sections, and removing crosstalk in the sections of the channel to be subjected to the crosstalk removal.

Abstract: An invention for eliminating the noise generated by a user speaking into a microphonic instrument is disclosed herein. In a first embodiment, the invention comprises a soundproofed housing arranged to cover a user's mouth region and a loudspeaker that outputs a processed sound wave having a phase that is opposite of the user's voice thereby canceling out the user's voice that was confined inside the housing. In a second embodiment, the invention comprises a soundproofed housing arranged to enclose a user's mouth area, a microphone to capture the user's speech, and a loudspeaker that outputs a processed sound wave having a phase that is opposite of the user's voice thereby canceling out the user's voice that was confined inside the housing.

Abstract: A fast accurate multi-channel frequency dependent scheme for analyzing noise in a signal processing system is described herein. Noise is decomposed within each channel into frequency bands and sub-band noise is propagated. To avoid the computational complexity of a convolution, traditional methods either assume the noise to be white, at any point in the signal processing pipeline, or they just ignore spatial operations. By assuming the noise to be white within each frequency band, it is possible to propagate any type of noise (white, colored, Gaussian, non-Gaussian and others) across a spatial transformation in a very fast and accurate manner. To demonstrate the efficacy of this technique, noise propagation is considered across various spatial operations in an image processing pipeline. Furthermore, the computational complexity is a very small fraction of the computational cost of propagating an image through a signal processing system.

Abstract: A noise reduction system and a method of noise reduction includes a microphone array comprising a first microphone, a second microphone, and a third microphone. Each microphone has a known position and a known directivity pattern. An instantaneous direction-of-arrival (IDOA) module determines a first phase difference quantity and a second phase difference quantity. The first phase difference quantity is based on phase differences between non-repetitive pairs of input signals received by the first microphone and the second microphone, while the second phase difference quantity is based on phase differences between non-repetitive pairs of input signals received by the first microphone and the third microphone. A spatial noise reduction module computes an estimate of a desired signal based on a priori spatial signal-to-noise ratio and an a posteriori spatial signal-to-noise ratio based on the first and second phase difference quantities.

Abstract: A noise adaptive mobile communication device including a noise collecting microphone which collects noise from a peripheral environment; a noise sensing unit which senses the collected noise; a frequency-component detecting unit which detects a frequency component of the sensed noise; a sound generating unit which generates a noise-adaptive sound from the detected frequency component; a call-sound synthesizing unit which synthesizes received call sound with the noise-adaptive sound; and an operation control unit which controls the call-sound synthesizing unit to operate each predetermined time.

Abstract: A system and method are disclosed for enhancing audio signals by nonlinear spectral operations. Successive portions of the audio signal are processed using a subband filter bank. A nonlinear modification is applied to the output of the subband filter bank for each successive portion of the audio signal to generate a modified subband filter bank output for each successive portion. The modified subband filter bank output for each successive portion is processed using an appropriate synthesis subband filter bank to construct a modified time-domain audio signal. High modulation frequency portions of the audio signal may be emphasized or de-emphasized, as desired. The modification may be applied within one or more frequency bands.

Abstract: According to one embodiment of the invention, a method for managing time-sensitive packetized data streams at a receiver includes receiving a time-sensitive packet of a data stream, analyzing an energy level of a payload signal of the packet, and determining whether to drop the packet based on the energy level of the payload signal.

Abstract: According to one embodiment, a sound quality control device includes: a time domain analysis module configured to perform a time-domain analysis on an audio-input signal; a frequency domain analysis module configured to perform a frequency-domain analysis on a frequency-domain signal; a first calculation module configured to calculate first speech/music scores based on the analysis results; a compensation filtering processing module configured to generate a filtered signal; a second calculation module configured to calculate second speech/music scores based on the filtered signal; a score correction module configured to generate one of corrected speech/music scores based on a difference between the first speech/music score and the second speech/music score; and a sound quality control module configured to control a sound quality of the audio-input signal based on the one of the corrected speech/music scores.

Abstract: A novel enhanced beamforming technique that improves beamforming operations by incorporating a model for the directional gains of the sensors, such as microphones, and provides means of estimating these gains. The technique forms estimates of the relative magnitude responses of the sensors (e.g., microphones) based on the data received at the array and includes those in the beamforming computations.

Abstract: The present technology provides a robust noise suppression system which may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. An acoustic signal may be received and transformed to cochlear domain sub-band signals. Features such as pitch may be identified and tracked within the sub-band signals. Initial speech and noise models may be then be estimated at least in part from a probability analysis based on the tracked pitch sources. Speech and noise models may be resolved from the initial speech and noise models and noise reduction may be performed on the sub-band signals and an acoustic signal may be reconstructed from the noise-reduced sub-band signals.

Abstract: A technique is described herein for reducing audible artifacts in an audio output signal generated by decoding a received frame in a series of frames representing an encoded audio signal in a predictive coding system. In accordance with the technique, it is determined if the received frame is one of a predefined number of received frames that follow a lost frame in the series of the frames. Responsive to determining that the received frame is one of the predefined number of received frames, at least one parameter or signal associated with the decoding of the received frame is altered from a state associated with normal decoding. The received frame is then decoded in accordance with the at least one parameter or signal to generate a decoded audio signal. The audio output signal is then generated based on the decoded audio signal.

Abstract: A method and a device for tracking background noise in a communication system, where the method includes: calculating a SNR of a current frame according to input audio signals; increasing a frame counter, and calculating tone features and signal steadiness features of the current frame if the SNR of the current frame is not smaller than a first threshold; judging the possibility of a time window including a noise interval according to the calculated tone feature values and signal steadiness feature values of each frame of the time window when the frame counter is increased to the length of the time window; and extracting noise features in the time window. Existence of background noise is analyzed continuously in a time window, so that background noise that changes frequently and dramatically can be detected or tracked rapidly.

Abstract: Disclosed herein is a signal processing apparatus including: a first decimation processing section for generating, based on a digital signal in a first form, a digital signal in a second form; a second decimation processing section for generating, based on the digital signal in the second form, a digital signal in a third form; a first signal processing section for processing the digital signal in the third form; an interpolation processing section for converting a digital signal in the third form outputted from the first signal processing section into a digital signal in the second form; a second signal processing section for processing the digital signal in the second form outputted from the first decimation processing section; and a combining section for combining the digital signals in the second form outputted from the interpolation processing section and the second signal processing section.

Abstract: A triangular microphone assembly (101) for use in a vehicle accessory includes a mirror housing (106) adapted for attachment to the interior of the vehicle. A mirror is disposed in an opening of the mirror housing (106) and a plurality of virtual digital microphones (108a, 108b, 108c) are arranged in a substantially triangular configuration in the mirror housing (106). A digital signal processor (DSP) (537) is used for receiving signals from the plurality of digital microphones (108a, 108b, 108c) such that the digital microphones exhibit directional characteristics for reducing undesirable noise in at least one direction by normalizing the phase of the received signals as a function of signal frequency.

Abstract: A method for a multi microphone noise reduction in a complex noisy environment is proposed. A left and a right noise power spectral density for a left and a right noise input frame is estimated for computing a diffuse noise gain. A target speech power spectral density is extracted from the noise input frame. A directional noise gain is calculated from the target speech power spectral density and the noise power spectral density. The noisy input frame is filtered by Kalman filtering method. A Kalman based gain is generated from the Kalman filtered noisy frame and the noise power spectral density. A spectral enhancement gain is computed by combining the diffuse noise gain, the directional noise gain, and the Kalman based gain. The method reduces different combinations of diverse background noise and increases speech intelligibility, while guaranteeing to preserve the interaural cues of the target speech and directional background noises.

Abstract: A method determines a bias reduced noise and interference estimation in a binaural microphone configuration with a right and a left microphone signal at a time-frame with a target speaker active. The method includes a determination of the auto power spectral density estimate of the common noise formed of noise and interference components of the right and left microphone signals and a modification of the auto power spectral density estimate of the common noise by using an estimate of the magnitude squared coherence of the noise and interference components contained in the right and left microphone signals determined at a time frame without a target speaker active. An acoustic signal processing system and a hearing aid implement the method for determining the bias reduced noise and interference estimation. The noise reduction performance of speech enhancement algorithms is improved by the invention. Further, distortions of the target speech signal and residual noise and interference components are reduced.

Abstract: An enhancement system improves the estimate of noise from a received signal. The system includes a spectrum monitor that divides a portion of the signal at more than one frequency resolution. Adaptation logic derives a noise adaptation factor of the received signal. A plurality of devices tracks the characteristics of an estimated noise in the received signal and modifies multiple noise adaptation rates. Weighting logic applies the modified noise adaptation rates derived from the signal divided at a first frequency resolution to the signal divided at a second frequency resolution.

Abstract: In a noise suppression apparatus for suppressing noise contained in a speech signal, the speech signal is converted to a first vector of spectral speech components and a second vector of spectral speech components identical to the first vector. A vector of noise suppression coefficients is determined based on the first vector spectral speech components. A vector of estimated noise components is determined based on the first vector spectral speech components, and a speech section correction factor and a nonspeech section correction factor are calculated from the estimated noise components and the first-vector spectral speech components to produce a combined correction factor. The noise suppression coefficients are weighted by the combined correction factor to produce a vector of post-suppression coefficients. The second vector spectral speech components are weighted by the post-suppression coefficients to produce a vector of enhanced speech components.

Abstract: Disclosed is an apparatus and method for processing signals such as sound signals. The sound processing apparatus includes a sound signal input unit for receiving sound signals, a harmonic noise separator for separating a harmonic region and a noise region from the received sound signals, a noise restraint index determination unit for determining an optimal noise restraint index k according to a system and circumstance, and a noise restrainer for restraining the separated noise region depending on the noise restraint index k so as to output noise attenuated signals.

Abstract: A system improves the perceptual quality of a speech signal by dampening undesired repetitive transient noises. The system includes a repetitive transient noise detector adapted to detect repetitive transient noise in a received signal. The received signal may include a harmonic and a noise spectrum. The system further includes a repetitive transient noise attenuator that substantially removes or dampens repetitive transient noises from the received signal. The method of dampening the repetitive transient noises includes modeling characteristics of repetitive transient noises; detecting characteristics in the received signal that correspond to the modeled characteristics of the repetitive transient noises; and substantially removing components of the repetitive transient noises from the received signal that correspond to some or all of the modeled characteristics of the repetitive transient noises.

Abstract: An signal processing apparatus, system and software product for audio modification/substitution of a background noise generated during an event including, but not be limited to, substituting or partially substituting a noise signal from one or more microphones by a pre-recorded noise, and/or selecting one or more noise signals from a plurality of microphones for further processing in real-time or near real-time broadcasting.

Abstract: An acoustic noise suppression filter including attenuation filtering with a noise-free estimate based on a codebook of line spectral frequencies.