Abstract: A band division noise suppressor suppressing noise sufficiently with a small amount of processing and a little voice distortion. In the band division noise suppressor, a band dividing section (101) divides an input voice signal into a low band voice signal and a high band voice signal. The low band voice signal is subjected to decimate at a decimation section (102), subjected to noise suppression at a low band noise suppressing section (103), and then interpolated at an interpolation section (104). On the other hand, the high band voice signal is subjected to noise suppression at a high band noise suppressing section (105). A band combination section (106) composes the bands of low-band and high-band voice signals subjected to noise suppression and outputs a voice signal subjected to noise suppression over the entire band.

Abstract: Digital audio data with error detection bits added thereto is inputted to an error detecting and correcting device (4). The correcting device (4) corrects an error when the error is detected in the digital audio data. The digital audio data outputted from the error detecting and correcting device (4) is inputted to an impulse noise suppressing circuit (6). The suppressing circuit (6) operates for a predetermined time period when the correcting device (4) detects an error.

Abstract: A method and system is provided for enhancing an audio signal based on spectral subtraction. The noise power spectrum for each frame of an audio signal is dynamically estimated based on a plurality of signal power spectrum values computed from a corresponding plurality of adjacent frames. An over-subtraction factor is then dynamically computed for each frame based on the noise power spectrum estimated for the frame. The signal power spectrum of the audio signal at each frame is then reduced in accordance with the over-subtraction factor computed for the corresponding frame.

Abstract: Methods and apparatus for post-processing of speech signals are disclosed herein.

Abstract: In a speech signal decoding method, information containing at least a sound source signal, gain, and filter coefficients is decoded from a received bit stream. Voiced speech and unvoiced speech of a speech signal are identified using the decoded information. Smoothing processing based on the decoded information is performed for at least either one of the decoded gain and decoded filter coefficients in the unvoiced speech. The speech signal is decoded by driving a filter having the decoded filter coefficients by an excitation signal obtained by multiplying the decoded sound source signal by the decoded gain using the result of the smoothing processing. A speech signal decoding apparatus is also disclosed.

Abstract: In order to enhance the quality of a communication signal derived from speech and noise, a filter divides the communication signal into a plurality of frequency band signals. A calculator generates a plurality of power band signals each having a power band value and corresponding to one of the frequency band signals. The power band values are based on estimating, over a time period, the power of one of the frequency band signals. The time period is different for different ones of the frequency band signals. The power band values are used to calculate weighting factors which are used to alter the frequency band signals that are combined to generate an improved communication signal.

Abstract: The present invention provides a diagnostic device that presents, to a patient, a Noise-Vocoded Speech Sound signal that is obtained by dividing at least one portion of a sound signal into a single or a plurality of frequency band signals and subjecting the frequency band signals to noise, and analyzing the content of a response recognized by the patient and the presented stimulus to diagnose a disease of the patient based on the analysis results, so that diagnosis including determining the disease of the patient and estimating a damaged site can be performed.

Abstract: Methods, apparatus, and systems for source separation include a converged plurality of coefficient values that is based on each of a plurality of M-channel signals. Each of the plurality of M-channel signals is based on signals produced by M transducers in response to at least one information source and at least one interference source. In some examples, the converged plurality of coefficient values is used to filter an M-channel signal to produce an information output signal and an interference output signal.

Abstract: A multimedia decoder unit having error concealment and fast muting capabilities. The audio decoder provides error concealment using a dynamic recovery delay that is based on the error rate of an input digital bitstream and also uses frame repeating. The decoder allows fast audio muting whereby audio can be muted within two audio frames of a mute signal that immediately freezes the video frame, e.g., a channel change. With respect to the dynamic recovery delay, a template of fixed length is used to inspect the last frames within the template. If error is found, then the error sum is used as an index into a table length which provides a dynamic template length. Error within the dynamic template length is computed and if larger than a tolerance, the current frame is muted. This allows the recovery delay to be adaptive and based on the error rate while still allowing mute merging.

Abstract: A method of estimating noise in data containing voice information and noise includes receiving the data as a sequence of input values; transforming the data by applying a first non linear mapping to the input values wherein the derivative function of the mapping decreases in magnitude as the input values increase in magnitude smoothing the transformed data; and transforming the smoothed transformed data by applying a second non linear mapping that is opposite to the first non linear mapping, to determine an estimate of the noise in the inputted data.

Abstract: The ability to combine multiple audio signals captured from the microphones in a microphone array is frequently used in beamforming systems. Typically, beamforming involves processing the output audio signals of the microphone array in such a way as to make the microphone array act as a highly directional microphone. In other words, beamforming provides a “listening beam” which points to a particular sound source while often filtering out other sounds. A “generic beamformer,” as described herein automatically designs a set of beams (i.e., beamforming) that cover a desired angular space range within a prescribed search area. Beam design is a function of microphone geometry and operational characteristics, and also of noise models of the environment around the microphone array. One advantage of the generic beamformer is that it is applicable to any microphone array geometry and microphone type.

Abstract: In accordance with an embodiment, the invention provides a spectral enhancement system that includes a plurality of distributed filters, a plurality of energy distribution units, and a weighted-averaging unit. At least one of the distributed filters receives a multi-frequency input signal. Each of the plurality of energy-detection units is coupled to an output of at least one filter and provides an energy-detection output signal. The weighted-averaging unit is coupled to each of the energy-detection units and provides a weighted-averaging signal to each of the filters responsive to the energy-detection output signals from each of the energy-detection units to implement distributed gain control. In an embodiment, the energy detection units are coupled to the outputs of the filters via a plurality of differentiator units.

Abstract: Modifying an audio signal comprising a plurality of channel signals is disclosed. At least selected ones of the channel signals are transformed into a time-frequency domain. The at least selected ones of the channel signals are compared in the time-frequency domain to identify corresponding portions of the channel signals that are not correlated or are only weakly correlated across channels. The identified corresponding portions of said channel signals are modified.

Abstract: An apparatus for adaptively suppressing noise in an input signal frequency spectrum derived from overlapping input frames is provided. The system includes a psychoacoustic power computation module configured to compute a noisy signal power in psychoacoustic bands, a voice activity scoring module configured to compute a probabilistic score for a presence of a speech, and a noise estimation module configured to estimate a noise power in the psychoacoustic bands based on information of past frames, the probabilistic score, and the computed noisy signal power. The system also includes a gain computation module configured to compute a gain for each frequency, based on a probabilistic heuristic, the probabilistic score and the information on the past frames, and a gain post-processing module configured to perform a gain time smoothing, a gain frequency smoothing, and a gain regulation for the computed gain.

Abstract: Provided is a signal distortion elimination apparatus comprising: an inverse filter application means that outputs the signal obtained by applying an inverse filter to an observed signal as a restored signal when a predetermined iteration termination condition is met and outputs the signal obtained by applying the inverse filter to the observed signal as an ad-hoc signal when the predetermined iteration termination condition is not met; a prediction error filter calculation means that segments the ad-hoc signal into frames and outputs a prediction error filter of each frame obtained by performing linear prediction analysis of the ad-hoc signal of each frame; an inverse filter calculation means that calculates an inverse filter such that a concatenation of innovation estimates of the respective frames becomes mutually independent among their samples, where the innovation estimate of a single frame (an innovation estimate) is the signal obtained by applying the prediction error filter of the corresponding frame

Abstract: A method and system for improving speech quality may include estimating at least one component of a distorted portion of a speech signal from at least one component of an undistorted portion of the speech signal and reinforcing the component of the distorted portion based on the estimating. The components may include the pitch, spectral envelope and spectral energy of the speech signal. The undistorted portion of the speech signal may be delayed and the components of the distorted portion may be interpolated from the components of a delayed undistorted portion and a current undistorted portion of the speech signal. The components of the distorted portion of the speech signal may be extrapolated from a current undistorted portion of the speech signal. Components of the distorted portion of the speech signal may be estimated from frequency bands other than the frequency band affected by the distortion.

Abstract: A softbit speech decoder includes a speech loss concealment circuit receiving bit information, bit error probability data, and a speech information flag from an equivalent channel based on a received signal provided to the equivalent channel. The speech loss concealment circuit also contains a speech data judging circuit for judging whether the speech information flag indicates that the received signal contains speech data, a parameter generating circuit for generating output information, including a speech-parameter error probability vector when the speech information flag indicates that the received signal does not contain speech data. When the received signal does not contain speech data, the speech-parameter error probability vector generated by the parameter generating circuit allows the softbit speech decoder to continue operating.

Abstract: A transient noise removal system removes or dampens undesired transients from speech. When the transient noise removal system receives a speech frame, the system performs a wavelet transform analysis. The speech frame may be represented by one or more wavelet coefficients across one or more wavelet levels. For a given wavelet level, the transient noise-removal system may determine a wavelet threshold. The transient noise removal system may compare the threshold corresponding to a wavelet level to the wavelet coefficients within that level. The transient noise removal system may attenuate each wavelet coefficient based on a comparison to a threshold.

Abstract: A system and method of concealing bit errors in a signal are provided. An exemplary method detects bit errors in an input signal having at least a current signal segment and a previous signal segment. The previous signal segment has a log-gain value qlg(m?1) and immediately precedes the current signal segment. The method comprises estimating a level lvl(m?1) of the input signal and determining a log-gain value qlg(m) of the current signal segment within the input signal. The method also comprises determining a difference between the gain value of the current signal segment and the previous signal segment and determining whether the difference exceeds a threshold. wherein the threshold is adaptive to the input signal level.

Abstract: The present invention relates to an apparatus and method of improving intelligibility of a voice signal. A method of improving intelligibility of a voice signal according to an embodiment of the present invention includes analyzing a background noise signal on a call receiving side, classifying a received voice signal into a silence signal, an unvoiced sound signal, and a voiced sound signal, and intensifying the classified unvoiced sound signal and voiced sound signal on the basis of the analyzed background noise signal on the call receiving side.

Abstract: There is provided a communication device for effectively encoding an audio/music signal while maintaining a predetermined quality by controlling the transmission bit rate of the transmission side considering the use environment of the reception side. In this device, a transmission mode decision unit (101) detects an environment noise contained in the background of the audio/music signal in the input signal and decides the transmission mode controlling the transmission bit rate of the signal transmitted from a communication terminal device (150), which is a communication terminal of the partner side, according to the environment noise level. A signal decoding unit (103) decodes encoded information transmitted from the communication terminal device (150) via a transmission path (110) and outputs the obtained signal as an output signal.

Abstract: A method for multifunctional processing of signals in frequency subbands performs subband decomposition and signal processing in two stages. A fullband signal is first splitted, with downsampling, into wide frequency subband (WFS) signals. Processing algorithms not requiring a high frequency resolution but benefiting from downsampling (such as subband acoustic echo cancellation), are applied to the WFS signals by wide subband processing blocks. Processed WFS signals are splitted, preferably without downsampling, into groups of narrow frequency subband (NFS) signals. The NFS signals are processed using processing algorithms (noise suppression, etc.) requiring a higher resolution. Processed NFS signals are synthesized into processed WFS signals, which are recombined into an output signal. Two-stage processing makes it possible to optimize signal processing, while keeping computational costs at low level and avoiding undesirable time delays.

Abstract: A method is provided whereby, before being subjected to a low rate voice coding, an incoming digital voice signal is chronologically segmented into blocks, the blocks are broken down respectively, in chronological order, into frequency components by a transformation in the frequency range and the frequency components are multiplied by weight factors depending on the frequency and modifiable in time, a frequency component being multiplied by the last weight factor calculated for the frequency component if the factor is less than the current weight factor.

Abstract: A system and method for nonlinear signal enhancement is provided. The method comprises: performing a linear transformation on a measured signal comprising a source component and a noise component; determining a modulus of the linear transformed signal; estimating a noise-free part of the linear transformed signal; and reconstructing the source component of the measured signal using the noise-free part of the linear transformed signal.

Abstract: A system and method of processing sound signals are disclosed. In one embodiment, a speech coder applies a first sound signal enhancement process to a first part of a sound signal and applies a second sound signal enhancement process to a second part of the sound signal. The sound signal is then coded using the enhanced first part of the sound signal and the enhanced first part of the sound signal and the enhanced sound part of the sound signal. Examples of the portions of the sound signal that are separately processed include an excitation signal component and a spectral component of the sound signal.

Abstract: Methods and systems are described for changing a communication quality of a communication session based on a meaning of speech data. Speech data exchanged between clients participating in a communication session is parsed. A meaning of the parsed speech data is determined for identifying a service quality indicator for the communication session. An action is performed to change a communication quality of the communication session based on the identified service quality indicator.

Abstract: An integrated circuit (IC) includes at least one baseband processing module, an RF section, and an interface. The baseband processing module converts outbound voice signal into an outbound voice symbol stream; an inbound voice symbol stream into an inbound voice signal; outbound data into an outbound data symbol stream; an inbound data symbol stream into inbound data; outbound wireless network data into an outbound wireless network data symbol stream; and an inbound wireless network data symbol stream into inbound wireless network data.

Abstract: A method and apparatus to reduce background noise in speech signals in order to improve the quality and intelligibility of processed speech. In mobile communications environment, speech signals are degraded by additive random noise. A randomness of the noise, which is often described in terms of its first and second order statistics, make it difficult to remove much of the noise without introducing background artifacts. This is particularly true for lower signal to background noise ratios. The method and apparatus provides noise reduction without any knowledge of the signal to background noise ratio.

Abstract: An apparatus and method for controlling a volume of comfort noise in a mobile communication terminal. The mobile communication terminal includes a vocoder for detecting a decoding rate of voice data received after establishing a voice call to output a detection signal and an amplifier for amplifying an output signal of the vocoder. The method comprises the steps of determining whether the decoding rate of the voice data from the vocoder is ?, recognizing the voice data as silence period data containing only the comfort noise and providing the vocoder with a control signal to lower a volume of the comfort noise, if the decoding rate of the voice data is ?, and controlling a volume of the voice data such that it becomes a volume preset by a user, if the decoding rate of the voice data is not ?.

Abstract: A method of reducing noise by cascading a plurality of noise reduction algorithms is provided. A sequence of noise reduction algorithms are applied to the noisy signal. The noise reduction algorithms are cascaded together, with the final noise reduction algorithm in the sequence providing the system output signal. The sequence of noise reduction algorithms includes a plurality of noise reduction algorithms that are sufficiently different from each other such that resulting distortions and artifacts are sufficiently different to result in reduced human perception of the artifact and distortion levels in the system output signal.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.

Abstract: A frame erasure concealment device and method that is based on reestimating gain parameters for a code excited linear prediction (CELP) coder is disclosed. During operation, when a frame in a stream of received data is detected as being erased, the coding parameters, especially an adaptive codebook gain gp and a fixed codebook gain gc, of the erased and subsequent frames can be reestimated by a gain matching procedure. By using this technique with the IS-641 speech coder, it has been found that the present invention improves frame erasure concealment device and method improve the speech quality under various channel conditions, compared with a conventional extrapolation-based concealment algorithm.

Abstract: A method, medium, and system scalably encoding/decoding audio/speech. The method includes splitting an input signal into a low frequency band signal that is lower than a predetermined frequency and a high frequency band signal that is higher than the predetermined frequency, scalably encoding the split low frequency band signal into a core layer and one or more extension layers and then decoding the encoded core layer and the encoded extension layers, generating an error signal by using the split low frequency band signal and a decoded signal of the encoded core layer and the encoded extension layers, and encoding the error signal and the high frequency band signal into a signal-to-noise ratio (SNR) enhancement layer and a bandwidth extension layer.

Abstract: Disclosed herein is a noise reduction method for automatic speech recognitionl.

Abstract: A noise suppressor for altering a speech signal is trained based on a speech recognition system. An objective function can be utilized to adjust parameters of the noise suppressor. The noise suppressor can be used to alter speech signals for the speech recognition system.

Abstract: A method of classifying a spectro-temporal interval of an input audio signal (x(t)) is disclosed. A spectro-temporal interval of the input audio signal is first modelled (62 . . . 71) according to a perceptual model to provide a first representation (Rep 1). The spectro-temporal interval is then modelled (62 . . . 71) using a modified noise substituted input signal according to the same perceptual model to provide a second representation (Rep 2). The spectro-temporal interval is then classified as being noise or not based on a comparison of the first and second representations.

Abstract: A method and apparatus for shaping quantization noise generated when compressing audio data at a low bit rate is disclosed. A predetermined quantization noise threshold allowed during quantization of sampled audio data and quantization noise energy information of a quantized MDCT coefficient are received in all frequency bands of an audio frequency. The quantization noise energy of the quantized MDCT coefficient is attenuated in a predetermined number of frequency bands in which a difference between the predetermined quantization noise threshold and the quantization noise energy of the quantized MDCT coefficient is large.

Abstract: Acoustic noise for wireless or landline telephony is reduced through optimal filtering in which each frequency band of every time frame is filtered as a function of the estimated signal-to-noise ratio and the estimated total noise energy for the frame. Non-speech bands, non-speech frames and other special frames are further attenuated by one or more predetermined multiplier values. Noise in a transmitted signal formed of frames each formed of frequency bands is reduced. A respective total signal energy and a respective current estimate of the noise energy for at least one of the frequency bands is determined. A respective local signal-to-noise ratio for at least one of the frequency bands is determined as a function of the respective signal energy and the respective current estimate of the noise energy. A respective smoothed signal-to-noise ratio is determined from the respective local signal-to-noise ratio and another respective signal-to-noise ratio estimated for a previous frame.

Abstract: The invention is directed to an audience response detection interactive presentation tool. An interactive method for controlling a presentation in accordance with an embodiment of the present invention includes: presenting a slide to an audience during a presentation; activating a noise level measuring system; measuring audience noise as a pointer is successively positioned over each of a plurality of selection mechanisms; deactivating the noise level measuring system; automatically selecting the selection mechanism associated with the loudest measured audience noise; and automatically performing a predetermined action based on the selected selection mechanism.

Abstract: An enhanced noise pre-processor in a speech codec smoothes channel energy estimate moving toward a first smoothing constant if a prior signal to noise ratio estimate for more than five channels are above a threshold and toward a second smaller smoothing constant otherwise. Forming a signal to noise ratio estimate for each channel includes conditionally boosting if a signal energy estimate is more than a predetermined factor of a noise energy estimate and signal to noise ratio estimates are above a threshold for more than five channels. The estimated signal to noise ratio is conditionally modified if two long term prediction coefficients are above a predetermined factor. The estimated signal to noise ratio is not modified and a voice metric is set greater than a voice metric threshold upon matching templates corresponding to the fricative and nasal speech sounds. An adaptive minimum channel gain is chosen based on a current signal to noise ratio estimate.

Abstract: A system and method are provided that accurately estimate noise and that reduce noise in pattern recognition signals. The method and system define a mapping random variable as a function of at least a clean signal random variable and a noise random variable. A model parameter that describes at least one aspect of a distribution of values for the mapping random variable is then determined. Based on the model parameter, an estimate for the clean signal random variable is determined. Under many aspects of the present invention, the mapping random variable is a signal-to-noise ratio variable and the method and system estimate a value for the signal-to-noise ratio variable from the model parameter.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal.

Abstract: A method of detecting speech in an audio signal comprises a step of obtaining information on the energy of the audio signal, the energy information then being used to detect speech in the audio signal. The method further comprises a step of obtaining information on the voicing of the audio signal, the voicing information then being used in conjunction with the energy information to detect speech in the audio signal.

Abstract: In one aspect the invention is a method for decoding. The method includes receiving encoded data and decoding the encoded data using a noise-adaptive decoder. The data may include first-order Reed-Mueller (FORM) based codes. The data may be based on Complementary Code Keying. Using a noise-adaptive decoder may include determining values of a hard decision based on a first decoding process and discarding the values of the hard decision if a noise sensitivity parameter is above a threshold value. The method may further include using a second decoder process if the noise sensitivity parameter is above the threshold value.

Abstract: A noise suppression device calculates a subband SN ratio calculation based on a noise likeness signal, an input signal spectrum and a subband-based estimated noise spectrum. The device calculates a subband-based input signal average spectrum, calculates a subband-based mixture ratio of the subband-based estimated noise spectrum to the subband-based input signal average spectrum on the basis of the noise likeness signal, and calculates the subband-based SN ratio on the basis of the subband-based estimated noise spectrum, the subband-based input signal average spectrum and the mixture ratio.

Abstract: A method and apparatus determine a channel response for an alternative sensor using an alternative sensor signal, an air conduction microphone signal. The channel response and a prior probability distribution for clean speech values are then used to estimate a clean speech value.

Abstract: An unfiltered frame portion (2) from a second frame (503) is blended together with a filtered frame portion (1) from a first frame (501) to produce a combined frame portion (507). The combined frame portion (507) is then buffered (110) along with the filtered frame (501) for LPC analysis.

Abstract: A method for discriminating noise from signal in a noise-contaminated signal involves decomposing a frame of samples of the signal into decorrelated components, and using a difference between probability distributions of the noise contributions and the signal contributions to identify signal and noise. A Gaussian distribution is used to determine whether the components are only noise whereas a Laplacian distribution is used to determine whether the components contain the signal. Such discrimination may be used in speech enhancement or voice activity detection apparatus.

Abstract: In a signal processing method and apparatus, a predetermined correcting signal having a same frame length as a second frame signal in which predetermined processing is performed to a frequency spectrum of a first frame signal of a frame length to which a predetermined window function is performed and is converted into a time domain is adjusted so that amplitudes of both ends of the correcting signal become equal to amplitudes of both or one of frame ends of the second frame signal, and a corrected frame signal is obtained by subtracting an adjusted correcting signal from the second frame signal.

Abstract: This document describes tools capable of enabling and/or adaptively generating comfort noise. The tools may do so by receiving some background noise, analyzing that noise, and generating comfort noise based on the received background noise. In some embodiments, for example, the tools build and continuously adapt a history based on segments of background noise as they are received from the sender. The tools may use this history to generate comfort noise that is pleasing, relatively accurate, and/or dynamically changing responsive to changes in a speaker's background noise.