Abstract: A noise estimation method for a noisy speech signal according to an embodiment of the present invention includes the steps of approximating a transformation spectrum by transforming an input noisy speech signal to a frequency domain, calculating a smoothed magnitude spectrum having a decreased difference in a magnitude of the transformation spectrum between neighboring frames, calculating a search spectrum to represent an estimated noise component of the smoothed magnitude spectrum, and estimating a noise spectrum by using a recursive average method using an adaptive forgetting factor defined by using the search spectrum. According to an embodiment of the present invention, the amount of calculation for noise estimation is small, and large-capacity memory is not required. Accordingly, the present invention can be easily implemented in hardware or software. Further, the accuracy of noise estimation can be increase because an adaptive procedure can be performed on each frequency sub-band.

Abstract: A sound quality improvement method for a noisy speech signal according to an embodiment of the present invention comprises the steps of estimating a noise signal of an input noisy speech signal by performing a predetermined noise estimation procedure for the noisy speech signal; measuring a relative magnitude difference to represent a relative difference between the noisy speech signal and the estimated noise signal; calculating a modified overweighting gain function with a non-linear structure in which a relatively high gain is allocated to a low-frequency band than a high-frequency band by using the relative magnitude difference; and obtaining an enhanced speech signal by multiplying the noisy speech signal and a time-varying gain function obtained by using the overweighting gain function. Accordingly, the amount of calculation for noise estimation is small, and large-capacity memory is not required.

Abstract: An audio input signal is filtered using an adaptive filter to generate a prediction output signal with reduced noise, wherein the filter is implemented using a plurality of coefficients to generate a plurality of prediction errors and to generate an error from the plurality of prediction errors, wherein the absolute values of the coefficients are continuously reduced by a plurality of reduction parameters.

Abstract: The invention relates to a method and means for encoding background noise information during voice signal encoding methods. A basic idea of the invention is to provide the scalability known for transmitting voice information in a similar manner when forming an SID frame. The invention provides encoding of a narrowband first component and of a broadband second component of a piece of background noise information and formation of an SID frame which describes the background noise with separate areas for the first and second components.

Abstract: A method for reconstructing an erased speech frame is described. A second speech frame is received from a buffer. The index position of the second speech frame is greater than the index position of the erased speech frame. The type of packet loss concealment (PLC) method to use is determined based on one or both of the second speech frame and a third speech frame. The index position of the third speech frame is less than the index position of the erased speech frame. The erased speech frame is reconstructed from one or both of the second speech frame and the third speech frame.

Abstract: A system and method for improving voice recognition processing at a server system that receives voice input from a remotely located user system. The user system includes a microphone, a processor that performs front-end voice recognition processing of the received user voice input, and a communication component configured to send the front-end processed user voice input to a destination wirelessly over a network. The server system includes a communication component configured to receive the sent front-end processed user voice input, and a processor configured to complete voice recognition processing of the sent front-end processed user voice input.

Abstract: The present invention relates to a method and apparatus of a digital filter for noise suppression of a signal representing an acoustic recording. The method comprises determining a desired frequency response (H(?)) of the digital filter; and generating a noise suppression filter based on the desired frequency response. The desired frequency response is determined in a manner so that the desired frequency response does not exceed a maximum level, wherein the maximum level is determined in response to the signal to be filtered.

Abstract: A noise cancellation system is provided, for generating a noise cancellation signal to be added to a wanted signal to mitigate the effects of ambient noise. The system comprises: an input, for receiving an input signal representing ambient noise; a detector, for detecting a magnitude of said input signal; and a voice activity detector, for determining voiceless periods when said input signal does not contain a signal representing a voice. The detector is adapted to detect the magnitude of said input signal during said voiceless periods, and the system is adapted to operate in a first mode when said input signal is above a threshold value, and a second mode when said input signal is below the threshold value. The first mode comprises generating a noise cancellation signal with a first magnitude for at least partially cancelling the ambient noise. The second mode comprises generating a noise cancellation signal with a second magnitude that is less than the first magnitude.

Abstract: A method of measuring the degree of enhancement made to a voice signal by receiving the voice signal, identifying formant regions in the voice signal, computing stationarity for each identified formant region, enhancing the voice signal, identifying formant regions in the enhanced voice signal that correspond to those identified in the received voice signal, computing stationarity for each formant region identified in the enhanced voice signal, comparing corresponding stationarity results for the received and enhanced voice signals, and calculating at least one user-definable statistic of the comparison results as the degree of enhancement made to the received voice signal.

Abstract: A noise reducing apparatus includes: a voice signal inputting unit inputting an input voice signal; a noise occurrence period detecting unit detecting a noise occurrence period; a noise removing unit removing a noise for the noise occurrence period; a generation source signal acquiring unit acquiring a generation source signal with a time duration corresponding to a time duration corresponding to the noise occurrence period; a pitch calculating unit calculating a pitch of an input voice signal interval; an interval signal setting unit setting interval signals divided in each unit period interval; an interpolation signal generating unit generating an interpolation signal with the time duration corresponding to the noise occurrence period and alternately arranging the interval signal in a forward time direction and the interval signal in a backward time direction; and a combining unit combining the interpolation signal and the input voice signal, from which the noise is removed.

Abstract: The device and method of the present invention improves electronic communication which have behavioral consequences, including for example, flight communication, two-way closed circuit communication such as for fire, police, miners, scuba divers and other heath and safety workers, and even for mobile communication which happens during activities such as cellular or mobile conversations during driving. Dichotic listening techniques are altered to enhance dyadic (involving two people) interactions with a partner. The speech of at least the first member of the dyad is filtered to isolate the component below 0.5 Khz, which will be input with a gain to the left ear of the second person (provided that they are right-handed), and thus their right cerebral hemispheres, and the component with a frequency above 0.5 Khz. will be input to their right ears, and thus their left cerebral hemispheres.

Abstract: A method for speech signal processing is provided. Energy attenuation gain values are set for background noise signals corresponding to obtained background noise frames subsequent to an erasure concealment frame, so that differences between the energy attenuation gain values of the background noise signals corresponding to the background noise frames and the energy attenuation gain values of signals corresponding to their respective previous frames are within a threshold range. Energy attenuation of the background noise signals corresponding to the background noise frames is controlled by using the energy attenuation gain values. An apparatus for speech signal processing is also provided in embodiments of the present invention. By using the embodiments of the present invention, the energy transition between the area of erasure concealment signal and the area of background noise signal may be made natural and smooth, so as to improve the audio comfortable sensation of the listener.

Abstract: Techniques pertaining to noise reduction are disclosed. According to one aspect of the present invention, noise in an audio signal is effectively reduced and a high quality of a target voice is recovered at the same time. In one embodiment, an array of microphones is used to sample the audio signal embedded with noise. The samples are processed according to a beamforming technique to get a signal with an enhanced target voice. A target voice is located in the audio signal sampled by the microphone array. A credibility of the target voice is determined when the target voice is located. The voice presence probability is weighted by the credibility. The signal with the enhanced target voice is enhanced according to the weighed voice presence probability.

Abstract: The invention relates to the processing of a digital signal originating from a decoder and a noise reduction post-processing step, including, in particular, limitation of distortion introduced by the post-processing step in order to deliver a corrected output signal (SOUT), assigning said corrected output signal (SOUT) with: a current amplitude having an intermediary value between a current amplitude value of the post-processed signal (SPOST) and a corresponding current amplitude value of the decoded signal (S?MIC), or the current amplitude of the post-processed signal (SPOST), according to the respective values of the current amplitude of the post-processed signal (SPOST) and by the corresponding current amplitude of the decoded signal (S?MIC).

Abstract: A voice mixing device for mixing a plurality of voice signals, comprises: a speaker selection unit selecting at least one voice signal among said plurality of voice signals; a full signal adder unit adding all of at least one voice signal selected by said speaker selection unit; respective subtractor unit subtracting only one of said selected voice signals from an addition result of said full signal adder unit; a common noise suppression unit suppressing noise of a common voice signal, being an addition result of said full signal adder unit; individual noise suppression unit suppressing noise of respective individual voice signals, being subtraction results of said subtractor unit; and memory switching unit copying information of noise suppression obtained in said common noise suppression unit based on a selection result of said speaker selection unit, to information of noise suppression in said individual noise suppression unit.

Abstract: A method of regenerating wideband speech from narrowband speech, the method comprising: receiving samples of a narrowband speech signal in a first range of frequencies; modulating received samples of the narrowband speech signal with a modulation signal having a modulating frequency adapted to upshift each frequency in the first range of frequencies by an amount determined by the modulating frequency wherein the modulating frequency is selected to translate into a target band a selected frequency band within the first range of signals; filtering the modulated samples using a target band filter to form a regenerated speech signal in the target band; and combining the narrow band speech signal with the regenerated speech signal in the target band to regenerate a wideband speech signal, the method comprising the step of controlling the modulated samples to lie in a second range of frequencies identified by determining a signal characteristic of frequencies in the first range of frequencies.

Abstract: A signal processor includes: a first adaptive filter that takes a first signal as input and generates a first pseudo signal; a first subtractor that subtracts the first pseudo signal from a second signal to supply a first differential signal as output; a second adaptive filter that takes the first signal as input to generate a second pseudo signal; a second subtractor that subtracts the second pseudo signal from the second signal to supply a second differential signal as output; a first step size control circuit that generates a first step size used in updating the first adaptive filter in accordance with the relation between the second pseudo signal and the second differential signal; and a second step size control circuit that generates a second step size used in updating the second adaptive filter in accordance with the relation between the first signal and the second signal.

Abstract: A speech analysis device which accurately analyzes an aperiodic component included in speech in a practical environment where there is background noise includes: a frequency band division unit which divides, into bandpass signals each associated with a corresponding one of frequency bands, an input signal representing a mixed sound of background noise and speech; a noise interval identification unit which identifies a noise interval and a speech interval of the input signal; an SNR calculation unit which calculates an SN ratio; a correlation function calculation unit which calculates an autocorrelation function of each bandpass signal; a correction amount determination unit which determines a correction amount for an aperiodic component ratio, based on the calculated SN ratio; and an aperiodic component ratio calculation unit which calculates, for each frequency band, an aperiodic component ratio of the aperiodic component, based on the determined correction amount and the calculated autocorrelation function.

Abstract: Systems and methods for receiving natural language queries and/or commands and execute the queries and/or commands. The systems and methods overcome the deficiencies of prior art speech query and response systems through the application of a complete speech-based information query, retrieval, presentation and command environment. This environment makes significant use of context, prior information, domain knowledge, and user specific profile data to achieve a natural environment for one or more users making queries or commands in multiple domains. Through this integrated approach, a complete speech-based natural language query and response environment can be created. The systems and methods creates, stores and uses extensive personal profile information for each user, thereby improving the reliability of determining the context and presenting the expected results for a particular question or command.

Abstract: An echo suppressing system includes: a sound output device for outputting sound based on a sound signal, including a passing section for allowing passage of a component of a different frequency band, and a plurality of sound output sections, each of which outputs sound based on each of the plurality of sound signals passed through the passing section; a summer for summing the plurality of sound signals to generate a reference sound signal; a sound input device for converting input sound into a sound signal; and an echo suppressor for suppressing echo based on the sound output by the sound output device, including an input section to which a sound signal is input from the sound input device as an observation sound signal, and a correction section for correcting the observation sound signal so as to suppress echo included in the observation sound signal.

Abstract: An audio encoding device which can improve encoding performance while performing division search on an algebraic codebook in an audio encoding. In a distortion minimizing unit (112) of a CELP encoding device: a maximum correlation value calculation unit (221) calculates a correlation value by using each pulse and a target signal in each candidate position for four pulses constituting the fixed codebook so as to acquire a maximum value of the correlation value for each pulse and calculates a maximum correlation value by using the maximum value of the correlation value; a sorting unit (222) divides the four pulses into two subsets each having two pulses; and a search unit (224) performs a division search on the fixed codebook and acquires a code indicating the positions and polarities of the four pulses where the encoding distortion is minimum.

Abstract: A method for enhancing speech includes extracting a center channel of an audio signal, flattening the spectrum of the center channel, and mixing the flattened speech channel with the audio signal, thereby enhancing any speech in the audio signal. Also disclosed are a method for extracting a center channel of sound from an audio signal with multiple channels, a method for flattening the spectrum of an audio signal, and a method for detecting speech in an audio signal. Also disclosed is a speech enhancer that includes a center-channel extract, a spectral flattener, a speech-confidence generator, and a mixer for mixing the flattened speech channel with original audio signal proportionate to the confidence of having detected speech, thereby enhancing any speech in the audio signal.

Abstract: A method of filtering a speech signal for speech encoding in a communications network, includes determining a cut off frequency for a filter, wherein a component of the speech signal in a frequency range less than the cut off frequency is to be attenuated by the filter; receiving the speech signal at the filter; determining at least one parameter of the received speech signal, the at least one parameter providing an indication of the energy of the component of the received speech signal that is to be attenuated; and adjusting the cut off frequency in dependence on the at least one parameter, thereby adjusting the frequency range to be attenuated.

Abstract: Methods, media and apparatus for smoothing a time-varying level of a signal. A method includes estimating a time-varying probability density of a short-term level of the signal and smoothing a level of the signal by using the probability density. The signal may be an audio signal. The short-term level and the smoothed level may be time series, each having current and previous time indices. Here, before the smoothing, computing a probability of the smoothed level at the previous time index may occur. Before the smoothing, calculating smoothing parameters using the probability density may occur. Calculating the smoothing parameters may include calculating the smoothing parameters using the smoothed level at the previous time index, the short-term level at the current time index and the probability of the smoothed level at the previous time index. Calculating the smoothing parameters may include calculating the smoothing parameters using breadth of the estimated probability density.

Abstract: The intelligibility of speech signals is improved in the many situations where a voice signal is communicated or stored. Means and methods are disclosed for developing a scheme with high voice signal intelligibility without sacrificing the voice quality. The disclosed method comprises certain steps, including, but not limited to: Learning the noise on near-end side and enhancing the far-end voice as a function of the noise type and noise level on the near-end side. The disclosed method and apparatus are especially useful to increase the intelligibility of the communication device's loudspeaker output. The invention includes processing of an input speech signal to generate an enhanced intelligent signal. The FFT spectrum of the speech received from the far-end is modified in accordance with the LPC spectrum of the local background noise to generate an enhanced intelligent signal.

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: A noise detection apparatus includes a time-frequency transform unit configured to transform an input signal from a time domain to a frequency domain to produce a spectrum, a power spectrum calculating unit configured to obtain powers of frequencies from the spectrum, a peak stationarity detecting unit configured to use peaks of the powers of frequencies in each frame to detect frequencies at which a stationary peak of the powers exists, a power stationarity detecting unit configured to use magnitudes of the powers of frequencies in each frame to detect frequencies at which the magnitudes of the powers are stationary, and a check unit configured to use the frequencies detected by the peak stationarity detecting unit and the frequencies detected by the power stationarity detecting unit to check whether there is a noise that has at least one of peak stationarity and power stationarity in the frequency domain.

Abstract: A method of reducing noise in an audio signal, comprising the steps of: using a furrow filter to select spectral components that are narrow in frequency but relatively broad in time; using a bar filter to select spectral components that are broad in frequency but relatively narrow in time; analyzing the relative energy distribution between the output of the furrow and bar filters to determine the optimal proportion of spectral components for the output signal; and reconstructing the audio signal to generate the output signal. A second pair of time-frequency filters may be used to further improve intelligibility of the output signal. The temporal relationship between the furrow filter output and the bar filter output may be monitored so that the fricative components are allowed primarily at boundaries between intervals with no voiced signal present and intervals with voice components. A noise reduction system for an audio signal.

Abstract: A scalable coding apparatus is provided to suppress deterioration of a quality of a coded signal in a normal frame next to a frame compensated for the occurrence of a data loss. The scalable coding apparatus is provided with a core-layer coding section (11) to carry out core-layer coding for the n-th frame input audio signal, an ordinary coding section (121) to generate expanding-layer ordinary-coding layer L2(n) by carrying out ordinary-coding of an expanding layer for the input audio signal, a deterioration-compensation coding section (123) to generate an expanding-layer-deterioration coding data L2?(n) by carrying out compensation for quality deterioration of coded audio in a current frame due to a past frame loss, a judging section (125) to determine whether either the expanding-layer ordinary-coding data L2(n) or the expanding-layer deterioration-coding data L2?(n) should be output from the expanding-layer coding section (12) as expanding-layer coding data of the current frame.

Abstract: A computer-readable medium recording a program allowing a computer to execute: setting a plurality of frames on a common time axis between a first waveform of an input to the audio processing and a second waveform of an output from the audio processing, detecting a voice frame and a noise frame in the first and second waveform, calculating a first and second spectrum from the first and second waveform, adjusting the level of the first or second spectrum of the noise frame, and setting the adjusted first and second spectrum of the noise frame as a third and fourth spectrum, calculating a distortion amount of the noise frame from the third and fourth spectrum, estimating a noise model spectrum from the first or second spectrum, and calculating a distortion amount of the voice frame from the first and second spectrum of the voice frame at the selected frequency.

Abstract: In a noise estimation apparatus, a microphone converts sound into an electric signal and outputs the electric signal as a sound signal. A noise estimator performs estimation for estimating a magnitude of a noise component contained in the sound signal so as to generate an estimated noise signal. The noise estimator limits a minimum value of a noise level of the noise component contained in the sound signal during the estimation to a predetermined default value, and integrates the noise level having the limited minimum value to generate the estimated noise signal.

Abstract: Disclosed are an apparatus and a method for voice processing in a mobile communication terminal. A plurality of microphones are used to remove environmental noise at the time of voice communication, so that it is possible to perform high-quality voice communication and video telephony. Moreover, it is possible to perform voice recording even when a user does not open a mobile communication terminal. Furthermore, when voice is recorded or sound is recorded during moving image photographing, a plurality of microphones are effectively utilized to achieve good-quality recording and to perform recording conveniently even when the folder or the slider of the mobile communication terminal is closed. Therefore, it is possible to provide improved convenience in using the mobile communication terminal.

Abstract: Provided is an audio decoding device which can adjust the high-range emphasis degree in accordance with a background noise level.

Abstract: The present invention is a system and method for packetizing actual noise signals, typically background noise, received by an access gateway from a speaking party and transmitting these packetized noise signals via a network to an egress gateway. The egress gateway converts the packetized noise signal into noise signals suitable for output and transmits the output noise signals to a listening party. When the access gateway detects that no voice signal is being received and only a noise signal is being received for a predetermined period of time, the access gateway instructs the egress network to continually transmit output noise signals to the listening party and ceases to transmit packetized noise signals to the egress gateway.

Abstract: Bidirectional differential point to point simultaneous high speed signalling is provided between integrated circuits with highly effective echo canceling. Each integrated circuit comprises a transmitter for transmitting a first signal to another integrated circuit and a receiver for receiving a second signal from the other integrated circuit. The transmitter has an output buffer; a receiver has a receiver buffer and is co-located on the same integrated circuit; and a differential buffer is coupled between the input of the transmitter buffer and the output of the receiver buffer. To increase the quality of receiving the second signal, a third signal adjusted in phase and amplitude is coupled at the output of the receive buffer, so that the echoing of the first signal is canceled. Preferably, the rise time of the third signal is also adjusted.

Abstract: First encoded voice bits are transcoded into second encoded voice bits by dividing the first encoded voice bits into one or more received frames, with each received frame containing multiple ones of the first encoded voice bits. First parameter bits for at least one of the received frames are generated by applying error control decoding to one or more of the encoded voice bits contained in the received frame, speech parameters are computed from the first parameter bits, and the speech parameters are quantized to produce second parameter bits. Finally, a transmission frame is formed by applying error control encoding to one or more of the second parameter bits, and the transmission frame is included in the second encoded voice bits.

Abstract: Provided are an apparatus and method for estimating noise and a noise reduction apparatus employing the same. The noise estimation apparatus estimates noise by blocking audio signals from a direction of a target sound source from received audio signals, and compensating for distortions from directivity gains of a target sound blocker blocking the audio signals from the target sound source.

Abstract: In a method of smoothing stationary background noise in a telecommunication speech session, initially receiving and decoding S10 a signal representative of a speech session, where the signal comprises both a speech component and a background noise component. Subsequently, providing S20 a noisiness measure for the signal, and adaptively S30 smoothing the background noise component based on the provided noisiness measure.

Abstract: The system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, a transmitter analysis filter for receiving the digital speech signal and dividing it into a baseband signal and an enhancement residual band signal, a standard baseband encoder for accepting the baseband signal and coding it using an ITU-T encoder, an additional baseband encoder for reducing standard coding distortion in the baseband signal, an enhancement residual band encoder for coding a signal obtained by removing the coded baseband signal from the original digital speech signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.

Abstract: The present invention describes a speech enhancement method using microphone arrays and a new iterative technique for enhancing noisy speech signals under low signal-to-noise-ratio (SNR) environments. A first embodiment involves the processing of the observed noisy speech both in the spatial- and the temporal-domains to enhance the desired signal component speech and an iterative technique to compute the generalized eigenvectors of the multichannel data derived from the microphone array. The entire processing is done on the spatio-temporal correlation coefficient sequence of the observed data in order to avoid large matrix-vector multiplications. A further embodiment relates to a speech enhancement system that is composed of two stages. In the first stage, the noise component of the observed signal is whitened, and in the second stage a spatio-temporal power method is used to extract the most dominant speech component.

Abstract: A voice signal processing apparatus and method includes determining maximum amplitude values of a plurality of different voice frame signals obtained by giving different amounts of phase shift to frequency components of voice frame signals having a predetermined length which are divided from a digital voice signal, and selecting a voice frame signal whose maximum amplitude value is the minimum from among the amplitude values of the plurality of different voice frame signals.

Abstract: A system for a multimodal unification of articulation includes a voice signal modality to receive a voice signal, and a control signal modality which receives an input from a user and generates a control signal from the input which is selected from predetermined inputs directly corresponding to the phonetic information. The interactive voice based phonetic input system also includes a multimodal integration system to receive and integrates the voice signal and the control signal. The multimodal integration system delimits a context of a spoken utterance of the voice signal by using the control signal to preprocess and discretize into phonetic frames. A voice recognizer analyzing the voice signal integrated with the control signal to output a voice recognition result. This new paradigm helps overcome constraints found in interfacing mobile devices. Context information facilitates the handling of the commands in the application environment.

Abstract: An apparatus for a speech recognition based on source separation and identification includes: a sound source separator for separating mixed signals, which are input to two or more microphones, into sound source signals by using independent component analysis (ICA), and estimating direction information of the separated sound source signals; and a speech recognizer for calculating normalized log likelihood probabilities of the separated sound source signals. The apparatus further includes a speech signal identifier identifying a sound source corresponding to a user's speech signal by using both of the estimated direction information and the reliability information based on the normalized log likelihood probabilities.

Abstract: A transmitted data that includes audio data and a transmitted spectral sharpness parameter representing a spectral harmonic/noise sharpness of a plurality of subbands are received. A measured spectral sharpness parameter is estimated from received audio data. The transmitted spectral sharpness parameter is compared with the measured spectral sharpness parameter. A main sharpness control parameter is formed for each of the decoded subbands. The main sharpness control parameter for each of the decoded subbands is analyzed. Ones of the decoded subbands are sharpened if the corresponding main sharpness control indicates that a corresponding subband is not sharp enough, wherein sharpened subbands are formed. Likewise, ones of the decoded subbands are flattened if the corresponding main sharpness control indicates that a corresponding subband is not flat enough, wherein flattened subbands are formed.

Abstract: A system and methods of subtraction of a shaped component of a noise reduction spectrum from a combined signal are disclosed. In an embodiment, a method includes identifying a selected frequency component using a corresponding frequency component of a noise sample spectrum. A noise set is comprised of the noise sample spectrum. The method further includes forming a shaped component of a noise reduction spectrum using a processor and a memory based on a combined signal spectrum and the selected frequency component. The method also includes subtracting the shaped component of the noise reduction spectrum from the combined signal spectrum.

Abstract: MDCT or FFT-based audio coding algorithms often have the problem named here spectral pre-echoes when coding an energy attack signal. This invention presents several possibilities to avoid the spectral pre-echoes existing in decoded signal segment before the energy attack point. The spectral envelope before the attack point can be improved by performing spectrum smoothing, replacing the segment of having spectral pre-echoes or filtering the segment with a combined filter obtained by doing LPC analysis.

Abstract: A double talk detector for controlling the echo path estimation in a telecommunication system by indicating when a received coded speech signal is dominated by a non-echo signal; i.e., that so-called double talk exists. This is determined by extracting LSPs from a coded speech frame of the received coded speech signal when the signal power exceeds a first threshold value, converting each of said extracted LSPs into LSFs, and calculating the distance between each two adjacent LSFs. For each distance that is smaller than a second threshold, a spectral peak is located between the two LSFs, and it is determined whether said spectral peak is an echo or not. When a predetermined number of non-echo spectral peaks are located in the received speech signal, double talk will be indicated, and the echo path estimation may be disabled.

Abstract: An apparatus for noise suppression having a linear prediction analysis circuit having an LP error filter (LFF), which takes a first, noisy voice signal y(n)=x(n)+?(n) as a basis for producing an LP-error-filter output signal e(n), having a coefficient calculation unit (KBE), which updates the coefficients of the LP error filter on the basis of the internal signals (including the input and out signals y(n) and e(n)) in the LP error filter, and having a subtraction unit, which subtracts the LP error filter output signal e(n) from the first voice signal y(n) in a subtractor and, following the subtraction, outputs the remainder as a second voice signal x(n)=y(n)?e(n) in which the noise is suppressed.

Abstract: The present invention relates to a method for signal processing comprising the steps of providing a set of prototype spectral envelopes, providing a set of reference noise prototypes, wherein the reference noise prototypes are obtained from at least a sub-set of the provided set of prototype spectral envelopes, detecting a verbal utterance by at least one microphone to obtain a microphone signal, processing the microphone signal for noise reduction based on the provided reference noise prototypes to obtain an enhanced signal and encoding the enhanced signal based on the provided prototype spectral envelopes to obtain an encoded enhanced signal.

Abstract: A clipping detection device calculates an amplitude distribution of an input signal for each predetermined period, calculates a deflection degree of the distribution on the basis of the calculated amplitude distribution, and then detects clipping of a communication signal on the basis of the calculated deflection degree of the distribution.