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

Abstract: Configurations disclosed herein include systems, methods and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context. Example embodiments may decode two sets of encoded frames from an encoded audio signal. The two frame sets may be encoded using different encoding schemes. For example, the bit rate or coding mode may differ between the two encoded frame sets. Based on information from one of the decoded sets of frames, a context component included in a signal represented by the other frame set may be suppressed. Other embodiments may generate an audio context signal within the mobile user terminal, and mix the generated audio signal with another decoded audio signal.

Abstract: Configurations disclosed herein include systems, methods, and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context. Enhancing the context of a voice communication may first include suppressing an existing context component from the digital audio signal to obtain a context suppressed signal. This signal may then be mixed with a new context signal to create a context enhanced signal, which may then be encoded before transmission. When this new context enhanced signal includes a speech component, it may be encoded and transmitted at a particular bit rate. When the context enhanced signal does not include a speech component, it may also be encoded at a similar bit rate. However, depending on the state of a process control signal, portions of a digital audio signal that lack a speech component may also be transmitted at a lower bit rate.

Abstract: Speech frames of a first speech coding scheme are utilized as speech frames of a second speech coding scheme, where the speech coding schemes use similar core compression schemes for the speech frames, preferably bit stream compatible. An occurrence of a state mismatch in an energy parameter between the first speech coding scheme and the second speech coding scheme is identified, preferably either by determining an occurrence of a predetermined speech evolution, such as a speech type transition, e.g. an onset of speech following a period of speech inactivity, or by tentative decoding of the energy parameter in the two encoding schemes followed by a comparison. Subsequently, the energy parameter in at least one frame of the second speech coding scheme following the occurrence of the state mismatch is adjusted. The present invention also presents transcoders and communications systems providing such transcoding functionality.

Abstract: Automatic correcting of user's speech impairment in speech may include obtaining the audio signal of a given user's speech, and analyzing the obtained audio signal to identify artifacts caused by the user's impairment. The obtained audio signal may be modified by eliminating the identified artifacts from it. The modified audio signal may be provided, e.g., to be played or broadcast or transmitted.

Abstract: A noise reduction device of the present invention comprises a control filter unit for generating a control sound signal to cancel out a noise, a control speaker for outputting a control sound according to the control sound signal from the control filter unit, an error microphone for detecting a residual sound by superimposing the noise upon the control sound output from the control speaker, and an obstacle detector for detecting an obstacle around the error microphone, wherein the control filter unit generates the control sound signal according to data from the error microphone and the obstacle detector.

Abstract: A system is provided for transmitting information through a speech codec (in-band) such as found in a wireless communication network. A modulator transforms the data into a spectrally noise-like signal based on the mapping of a shaped pulse to predetermined positions within a modulation frame, and the signal is efficiently encoded by a speech codec. A synchronization sequence provides modulation frame timing at the receiver and is detected based on analysis of a correlation peak pattern. A request/response protocol provides reliable transfer of data using message redundancy, retransmission, and/or robust modulation modes dependent on the communication channel conditions.

Abstract: A noise reduction device is disclosed, in which noise reduction device, a controlling sound generator outputs a white noise generated by a white-noise generator, and this white noise is sensed by an error sensor for identifying an acoustic transmission function covering a path from the controlling sound generator to the error sensor. At this time, an identification controller prompts the white noise generator to generate a white noise for identifying the acoustic transmission function provided that an ambient noise level sensed by the error sensor is not greater than a given threshold.

Abstract: A sound analysis device comprises: a sound parameter calculation unit operable to acquire an audio signal and calculate a sound parameter for each of partial audio signals, the partial audio signals each being the acquired audio signal in a unit of time; a category determination unit operable to determine, from among a plurality of environmental sound categories, which environmental sound category each of the partial audio signals belongs to, based on a corresponding one of the calculated sound parameters; a section setting unit operable to sequentially set judgement target sections on a time axis as time elapses, each of the judgment target sections including two or more of the units of time, the two or more of the units of time being consecutive; and an environment judgment unit operable to judge, based on a number of partial audio signals in each environmental sound category determined in at least a most recent judgment target section, an environment that surrounds the sound analysis device in at least the

Abstract: A noise feedback coding (NFC) system and method that utilizes a simple and relatively inexpensive general structural configuration, but achieves improved flexibility with respect to controlling the shape of coding noise. The NFC system and method utilizes an all-zero noise feedback filter that is configured to approximate the response of a pole-zero noise feedback filter.

Abstract: A voice reproduction apparatus includes an ambient sound analysis unit to analyze a characteristic of an ambient sound, a characteristic analysis unit to analyze an acoustic characteristic of a signal for reproduction, a reproduction timing adjusting unit to record the signal for reproduction and to read the signal for reproduction at a reproduction timing of follow-up reproduction, a reproduction speed changing unit to change a reproduction speed of the read signal for reproduction, and a control unit to control the reproduction timing adjusting unit so that the signal for reproduction is reproduced at the reproduction timing corresponding to an analysis result of the ambient sound analysis unit and to control the reproduction speed changing unit so that the signal for reproduction is reproduced at the reproduction speed corresponding to the analysis result of the ambient sound analysis unit and the acoustic characteristic obtained by the characteristic analysis unit.

Abstract: Provided are a signal separating apparatus and a signal separating method capable of solving the permutation problem and separating user speech to be extracted. The signal separating apparatus separates a specific speech signal and a noise signal from a received sound signal. First, a joint probability density distribution estimation unit of a permutation solving unit calculates joint probability density distributions of the respective separated signals. Then, a classifying determination unit of the permutation solving unit determines classifying based on shapes of the calculated joint probability density distributions.

Abstract: A device comprising an audio information processor to receive at least one audio stream encoded according to a first protocol by a remote network processing device, the audio stream having associated comfort noise information to indicate a level of background noise available for presentation during silence periods associated with the audio stream, the audio information processor to decode the received audio stream according to the first protocol and to encode the decoded audio stream according to a second protocol, and a background noise translator to convert the comfort noise information received with the audio stream into a format compatible with the second protocol.

Abstract: According to various embodiments of the invention, a new and effective keyboard click noise reduction scheme is presented. The keyboard click noise reduction scheme may have various processing units including: Dynamic Signal Modeler, Smart Model Selector, Adaptive Filtering Module, Keyboard/Impulse Noise and Voice Activity Detectors, and a Post-Processing Unit. By adaptively changing the coefficients of the proposed adaptive filter through minimizing the output energy, the scheme can provide the target signal/voice with nearly zero keyboard click noise. The scheme could be used in real-time to minimize keyboard click noise or any kind of unwanted noise, especially noise having transient impulse characteristics.

Abstract: The present technology provides a robust noise suppression system that 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. An acoustic signal may be reconstructed from the noise-reduced sub-band signals.

Abstract: A method for performing a call between a near-end user and a far-end user, which includes the following operations performed during the call by the near-end user's communications device. Automatic gain control (AGC) is performed to update a gain applied to an uplink speech signal. A frame is detected in a downlink signal that contains speech; in response, the updating of the gain is frozen. Other embodiments are also described and claimed.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for enhancing speech recognition accuracy. In one aspect, a method includes receiving an audio signal that corresponds to an utterance recorded by a mobile device, determining a geographic location associated with the mobile device, identifying a set of geotagged audio signals that correspond to environmental audio associated with the geographic location, weighting each geotagged audio signal of the set of geotagged audio signals based on metadata associated with the respective geotagged audio signal, and using the set of weighted geotagged audio signals to perform noise compensation on the audio signal that corresponds to the utterance.

Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing frame.

Abstract: Encoding audio signals with selecting an encoding mode for encoding the signal categorizing the signal into active segments having voice activity and non-active segments having substantially no voice activity by using categorization parameters depending on the selected encoding mode and encoding at least the active segments using the selected encoding mode.

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 noise-environment storing unit stores therein a compensation vector for compensating a feature vector of a speech. A feature-vector extracting unit extracts the feature vector of the speech in each of a plurality of frames. A noise-environment-series estimating unit estimates a noise-environment series based on a feature-vector series and a degree of similarity. A calculating unit obtains a compensation vector corresponding to each noise environment in estimated noise-environment series based on the compensation vector present in the noise-environment storing unit. A compensating unit compensates the extracted feature vector of the speech based on obtained compensation vector.

Abstract: Apparatus and methods are provided for measuring perceptual quality of a signal transmitted over a communication network, such as a circuit-switching network, packet-switching network, or a combination thereof. In accordance with one embodiment, a distributed apparatus is provided for measuring perceptual quality of a signal transmitted over a communication network. The distributed apparatus includes communication ports located at various locations in the network. The distributed apparatus may also include a signal processor including a processor for providing non-intrusive measurement of the perceptual quality of the signal. The distributed apparatus may further include recorders operatively connected to the communication ports and to the signal processor, wherein at least one of the recorders processes the signal at one of the communication ports and the recorder sends the signal to the signal processor to measure the perceptual quality of the signal.

Abstract: A device and method are provided for automatically adjusting gain, including a conversion module for converting an audio time-domain signal to an audio frequency-domain signal, an analysis module for analyzing the audio frequency-domain signal in accordance with an equal-loudness level contour of human hearing so as to generate strength weightings and generating a signal strength in accordance with the weightings, a calculation module for calculating a gain by analysis of the audio frequency-domain signal when the signal strength falls outside a default range, and a control module for generating an audio output signal in accordance with the gain and the audio time-domain signal.

Abstract: A method of pre-processing an audio signal transmitted to a user terminal via a communication network and an apparatus using the method are provided. The method of pre-processing the audio signal may prevent deterioration of a sound quality of the audio signal transmitted to the user terminal by pre-processing the audio signal, and by enabling a codec module, encoding the audio signal, to determine the audio signal as a speech signal. The method of pre-processing may include separating the audio signal into channels, measuring the channel energy for each of the channels, selecting a specific channel energy, and amplifying the specific channel energy. The method may include encoding an audio signal using a speech codec and/or decoding an encoded audio signal using the speech codec.

Abstract: Methods and apparatus for audio watermarking a substantially silent media content presentation are disclosed. An example method to audio watermark a media content presentation disclosed herein comprises obtaining a watermarked noise signal comprising a watermark and a noise signal having energy substantially concentrated in an audible frequency band, the watermarked noise signal attenuated to be substantially inaudible without combining with a separate audio signal, associating the watermarked noise signal with a substantially silent content component of the media content presentation, the media content presentation comprising one or more media content components, and outputting the watermarked noise signal during presentation of the substantially silent content component.

Abstract: An encoding device improves the sound quality of a stereo signal while maintaining a low bit rate. The encoding device includes: an LP inverse filter which LP-inverse-filters a left signal L(n) by using an inverse quantization linear prediction coefficient AdM(z) of a monaural signal; a T/F conversion unit which converts the left sound source signal Le(n) from a temporal region to a frequency region; an inverse quantizer which inverse-quantizes encoded information Mqe; spectrum division units which divide a high-frequency component of the sound source signal Mde(f) and the left signal Le(f) into a plurality of bands; and scale factor calculation units which calculate scale factors ai and ssi by using a monaural sound source signal Mdeh,i(f), a left sound source signal Leh,i(f), Mdeh,i(f), and right sound source signal Reh,i(f) of each divided band.

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: A speech enhancement system enhances transitions between speech and non-speech segments. The system includes a background noise estimator that approximates the magnitude of a background noise of an input signal that includes a speech and a non-speech segment. A slave processor is programmed to perform the specialized task of modifying a spectral tilt of the input signal to match a plurality of expected spectral shapes selected by a Codec.

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: An audio signal quality enhancement apparatus and method. The apparatus includes a pitch calculating unit to extract a pitch period of an audio signal, a frequency domain transforming unit to transform the audio signal to a frequency domain, a frequency band dividing unit to classify the transformed audio signal into audio signals for each of the plurality of frequency bands based on the extracted pitch period, and a pitch enhancement unit to determine a gain based on a volume of the transformed audio signal, and to generate an output signal by multiplying each of the classified audio signals with respect to each of the plurality of frequency bands by the gain, thereby enhancing quality of the audio signal.

Abstract: Provided are methods and systems for enhancing the quality of voice communications. The method and corresponding system may involve classifying an audio signal into speech, and speech and noise and creating speech-noise classification data. The method may further involve sharing the speech-noise classification data with a speech encoder via a shared memory or by a Least Significant Bit (LSB) of a Pulse Code Modulation (PCM) stream. The method and corresponding system may also involve sharing acoustic cues with the speech encoder to improve the speech noise classification and, in certain embodiments, sharing scaling transition factors with the speech encoder to enable the speech encoder to gradually change data rate in the transitions between the encoding modes.

Abstract: A method of embedding data into an audio signal provides a data sequence for embedding in the audio signal and computes masking thresholds for the audio signal from a frequency domain transform of the audio signal. The masking thresholds correspond to subbands of the audio signal, which are obtained from a masking model used to compress the audio signal. The method applies the masking threshold to the data sequence to produce masked data sequence and inserts the masked data sequence in the audio signal to produce an embedded audio signal. A method of detecting data embedded in an audio signal analyzes the audio signal to estimate the masking threshold used in embedding the data and applies the estimated masking threshold to the audio signal to extract the embedded data.

Abstract: According to an aspect, a mobile communication device includes a housing, a speaker, a microphone, a detecting unit, and a processing unit. The speaker is provided in the housing, and outputs an incoming voice according to an incoming voice signal. The microphone is provided in the housing. The microphone receives an outgoing voice and outputs an outgoing voice signal in response to reception of the outgoing voice. The detecting unit detects vibration of the housing and outputs a housing-vibration signal indicating the vibration of the housing. The processing unit performs echo cancellation to the outgoing voice signal based on the incoming voice signal and the housing-vibration signal.

Abstract: Provided are a method and apparatus for sinusoidal audio coding, which employs a tracking method for further effective coding of sinusoids extracted in the process of a sinusoidal analysis of parametric coding. The sinusoidal audio coding method includes: extracting sinusoids of a current frame by performing a sinusoidal analysis on an input audio signal; with respect to each of the extracted sinusoids, setting a mode selected from a birth mode in which a sinusoid is newly generated irrespective of sinusoids of a previous frame, a continuation mode in which the sinusoid is only one sinusoid continued from one of the sinusoids of the previous frame, and a branch mode in which the sinusoid is one of a plurality of sinusoids continued from one of the sinusoids of the previous frame; and coding the extracted sinusoids according to the selected mode. Accordingly, a plurality of sinusoids that can be continued from one previous track component are set to the continuation mode or the branch mode.

Abstract: A speech enhancement method operative for devices having limited available memory is described. The method is appropriate for very noisy environments and is capable of estimating the relative strengths of speech and noise components during both the presence as well as the absence of speech.

Abstract: In a noise reduction apparatus for controlling noise up to a predetermined upper limited frequency, a distance from a noise source to control point X is made larger than a distance obtained by subtracting a one-half wavelength from a distance, obtained by adding up a distance from the noise source to a noise detecting microphone, a distance corresponding to time as a sum of respective delay time of the noise detecting microphone, a noise controller, and a control speaker, and a distance from the control speaker to control point X, where one wavelength is a period corresponding to the upper limited frequency.

Abstract: The invention relates to audio signal processing. More specifically, the invention relates to enhancing multichannel audio, such as television audio, by applying a gain to the audio that has been smoothed between segments of the audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for enhancing speech recognition accuracy. In one aspect, a method includes receiving geotagged audio signals that correspond to environmental audio recorded by multiple mobile devices in multiple geographic locations, receiving an audio signal that corresponds to an utterance recorded by a particular mobile device, determining a particular geographic location associated with the particular mobile device, generating a noise model for the particular geographic location using a subset of the geotagged audio signals, where noise compensation is performed on the audio signal that corresponds to the utterance using the noise model that has been generated for the particular geographic location.

Abstract: A device and a method for filtering out noise from speech of caller are disclosed. The method is applied to the device, includes: inputting a speech sound of a caller; converting the speech sound to digital signals by an analyzing-to-digital converting unit; analyzing the digital signals to identify a pure speech of the caller and filtering out an extraneous noise thus obtaining pure speech signals of the caller; encoding the pure speech signals by a coder and decoder unit, and submitting the encoded speech signals to the receiver.

Abstract: A method for processing and iteratively enhancing and estimating a source audio signal received at two audio receivers is provided. In one embodiment, the method involves the use of codebook constrained iterative binaural Wiener filter (CCIBWF). The provided CCIBWF embodiment can improve the quality of speech received at two audio receivers both in terms of noise reduction and speech intelligibility. In one embodiment, optimum speech enhancement performance was achieved within two iterations of the CCIBWF scheme. Further, the embodiment of the CCIBWF scheme introduces minimal distortion to the binaural cues, such as the interaural time delay cues, thereby preserving localization information of the audio source. The embodiment of the CCIBWF is also able to relatively accurately track the Time Delay of Arrival (TDOA) when the audio source is moving. This ensures that the performance of the CCIBWF scheme is not significantly degraded due to the selection of wrong codebooks.

Abstract: A method of pre-processing an audio signal transmitted to a user terminal via a communication network and an apparatus using the method are provided. The method of pre-processing the audio signal may prevent deterioration of a sound quality of the audio signal transmitted to the user terminal by pre-processing the audio signal, and by enabling a codec module, encoding the audio signal, to determine the audio signal as a speech signal. Also, the method of pre-processing the audio signal may improve a probability that the codec module may determine a corresponding audio signal as a speech when the audio signal is transmitted via the communication network by pre-processing the audio signal using a speech codec.

Abstract: Presented herein are systems and methods for processing sound signals for use with electronic speech systems. Sound signals are temporally parsed into frames, and the speech system includes a speech codebook having entries corresponding to frame sequences. The system identifies speech sounds in an audio signal using the speech codebook.

Abstract: A device for suppressing ambient sounds from speech received by a microphone array is provided. One embodiment of the device comprises a microphone array, a processor, an analog-to-digital converter, and memory comprising instructions stored therein that are executable by the processor.

Abstract: Systems and methods for adaptive processing of a close microphone array in a noise suppression system are provided. A primary acoustic signal and a secondary acoustic signal are received. In exemplary embodiments, a frequency analysis is performed on the acoustic signals to obtain frequency sub-band signals. An adaptive equalization coefficient may then be applied to a sub-band signal of the secondary acoustic signal. A forward-facing cardioid pattern and a backward-facing cardioid pattern are then generated based on the sub-band signals. Utilizing cardioid signals of the forward-facing cardioid pattern and backward-facing cardioid pattern, noise suppression may be performed. A resulting noise suppressed signal is output.

Abstract: An apparatus processes an acoustic input signal to provide an output signal with reduced noise. The apparatus weights the input signal based on a frequency-dependent weighting function. A frequency-dependent threshold function bounds the weighting function from below.

Abstract: A non-intrusive signal quality assessment apparatus includes a feature vector calculator that determines parameters representing frames of a signal and extracts a collection of per-frame feature vectors (?;(n)) representing structural information of the signal from the parameters. A frame selector preferably selects only frames (?\with a feature vector (?;(n)) lying within a predetermined multi-dimensional window (?). Means determine a global feature set (?) over the collection of feature vectors (?;(n)) from statistical moments of selected feature vector components ((1^,02, . . . O11). A quality predictor predicts a signal quality measure (Qj from the global feature set (?)).

Abstract: Systems and methods for utilizing inter-microphone level differences (ILD) to attenuate noise and enhance speech are provided. In exemplary embodiments, primary and secondary acoustic signals are received by omni-directional microphones, and converted into primary and secondary electric signals. A differential microphone array module processes the electric signals to determine a cardioid primary signal and a cardioid secondary signal. The cardioid signals are filtered through a frequency analysis module which takes the signals and mimics a cochlea implementation (i.e., cochlear domain). Energy levels of the signals are then computed, and the results are processed by an ILD module using a non-linear combination to obtain the ILD. In exemplary embodiments, the non-linear combination comprises dividing the energy level associated with the primary microphone by the energy level associated with the secondary microphone.

Abstract: A method for reducing ringing in a signal output from a filter comprising inputting a signal into a filter; filtering a first portion of the input signal to generate a filtered portion of the output signal; analyzing the filtered portion of the output signal; detecting if ringing is present in the filtered portion of the output signal based on said analysis; and adjusting the filter characteristics to reduce ringing in a subsequent filtered portion of the output signal if it is determined that ringing is present.

Abstract: A system and method for providing an augmented version of a Low-Complexity Sub-band Coder (LC-SBC) is described herein. In accordance with the method, a series of input audio samples representative of the frame are received. A series of sub-band samples is generated for each of a plurality of frequency sub-bands based on the input audio samples. A determination is made as to whether the frame is a voice frame or a noise frame. Responsive to a determination that the frame is a noise frame, an index representative of a previously-processed series of sub-band samples stored in a history buffer for at least one of the frequency sub-bands is encoded instead of encoding the series of sub-band samples generated for the frequency sub-band.

Abstract: A device for improving the intelligibility of a signal arising from a source subjected to a noisy environment, said source marking the signal with a specific signature, the device comprising a processing circuit receiving the signal; and means for analyzing the signal and parameterizing the processing circuit according to characteristics of the signature present in the signal. A first channel with low distortion conveys the signal from the source to the means for analyzing, and a second channel, susceptible to introduce a distortion, conveys the signal from the source to the processing circuit.