Abstract: There is provided an echo cancellation method that comprises computing a Discrete Cosine Transform (DCT) domain over a plurality of samples of a first signal to generate a plurality of first DCT samples, selecting one or more first coefficients from the plurality of first DCT samples, computing a DCT domain over a plurality of samples of a second signal to generate a plurality of second DCT samples, selecting one or more second coefficients from the plurality of second DCT samples, wherein the one or more second coefficients are same coefficients as the one or more first coefficients, applying normalized cross correlation to the one or more first coefficients and the one or more second coefficients to generate normalized cross correlation values, cancelling an echo of the second signal from the first signal by determining a bulk delay and a double talk condition based on the normalized cross correlation values.

Abstract: A system and method facilitating acoustic echo cancellation convergence detection is provided. The invention includes an acoustic echo cancellation convergence detector having a center clipping component, a convergence calculator, a convergence filter and a convergence statistic component. The invention provides for the acoustic echo cancellation convergence detector to provide an output regarding convergence of an adaptive filter based, at least in part, upon a filtered convergence statistic. Optionally, the acoustic echo cancellation convergence detector can provide an output associated with convergence history.

Abstract: A method and apparatus for detecting a singing frequency in a signal processing system using two neural-networks is disclosed. The first one (a hit neural network) monitors the maximum spectral peak FFT bin as it changes with time. The second one (change neural network) monitors the monotonic increasing behavior. The inputs to the neural-networks are the maximum spectral magnitude bin and its rate of change in time. The output is an indication whether howling is likely to occur and the corresponding singing frequency. Once the singing frequency is identified, it can be suppressed using any one of many available techniques such as notch filters. Several improvements of the base method or apparatus are also disclosed, where additional neural networks are used to detect more than one singing frequency.

Abstract: A system such as a speakerphone may include a processor, memory, a speaker and a microphone. The processor may be configured (via program instructions stored in the memory) to calibrate the speaker by: outputting a stimulus signal; receiving an input signal corresponding to the stimulus signal; computing a midrange sensitivity and a lowpass sensitivity for a transfer function derived from a spectrum of the input signal and a spectrum of the output signal; subtracting the midrange sensitivity from the lowpass sensitivity to obtain a speaker-related sensitivity; performing an iterative search for current parameters of a speaker model using the input signal spectrum, the stimulus signal spectrum and the speaker-related sensitivity; and updating averages of the speaker model parameters using the current parameter values. The stimulus signal may be transmitted during periods of silence in the external environment. The parameter averages may be used to perform echo cancellation.

Abstract: An echo canceller reduces an echo signal produced when the transmitted signal leaks back into the receiver via a hybrid. The echo canceller estimates the echo signal from the transmitted signal, and then subtracting the estimated echo signal from the received signal. In practice, the echo path channel in a DMT-modem is much longer than the cyclic prefix, and therefore, the received echo signal will be subjected to both ISI (inter-symbol-interference) and ICI (inter-carrier-interference). A traditional echo canceller, designed for a xDSL-modem, uses either a time domain adaptive FIR-filter or a combined echo canceller implemented in both time and frequency domain. A matrix-based adaptive echo canceller is implemented in the frequency domain. Various example embodiments are disclosed.

Abstract: An audio signal processing apparatus includes a splitting unit for splitting an audio signal of a first system and another audio signal of a second system into pluralities of frequency band components, a level comparing unit for calculating a level ratio or a level difference between each of the frequency bands of the first system and each of the frequency bands of the second systems, and an output control unit for removing frequency band components whose level ratio or level difference calculated by the level comparing unit is equal and substantially equal to a predetermined value from at least one of the first and second systems.

Abstract: A system for canceling acoustic feedback includes an input for receiving a digital audio signal and a processor configured to detect acoustic feedback signal in the digital audio signal and to determine the frequency of the feedback signal The system also includes a plurality of bandpass filters for attenuating the feedback signal. The processor is further configured to: select a bandpass filter from among the plurality of bandpass filters. The selected bandpass filter comprises a response characteristic that attenuates parts of the signal at the frequency of acoustic feedback signal.

Abstract: In one embodiment, an echo canceller configured to cancel echo in a wideband voice conference is provided. A double-talk condition may be when a plurality of users are speaking substantially simultaneously. When a double-talk condition is detected in the wideband conference, a high-frequency process is enabled and used to process signals in the high band to reduce echo. Accordingly, echo in the high band may not be produced by end devices being used by the users' speaking. Also, the users speaking have the echo cancelled in the low band and substantial echo does not result. This results in the users speaking experiencing the conference in the narrowband. The other users that are not speaking, however, continue to receive wideband signals. The users not speaking also continue to have echo cancellation performed for the high band and low band because these users are not speaking and thus attenuation of their voices is not a consideration.

Abstract: A method of removing reverberation from audio signals is disclosed. The method comprises spectro-temporally analyzing the first audio signal and the second audio signal to derive an energy function of time for a plurality of frequency bands. The method further comprises determining a delay stability between the energy function of time for the first audio signal and the second audio signal in each band, determining a gain function in each band based on the delay stability, adjusting the energy of the first audio signal and the second audio signal using the gain function within each band, and resynthesizing audio signals from the energy in each band of the first audio signal and the second audio signal.

Abstract: A method and apparatus for adapting dual filters is disclosed. In one aspect, a method may include transforming a signal, adapting a first adaptive filter based on the transformed signal, estimating a delay of an impulse response based on the adaptation of the first filter, delaying a signal based on the estimated delay, and adapting a second adaptive filter based on the delayed signal. In one aspect, an echo or other unwanted signal may be reduced or cancelled based on the adaptation of the second filter.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Abstract: The present invention is directed to implementing theoretical foundations of frequency domain methods aiming to locally decrease the impact of aggregate side lobes of a few discrete frequency tones, in the vicinity of other “victim” frequencies. The family of frequency methods referred to makes use of one or more “canceller tones.” The “canceller tones” loads are optimized to minimize the aggregate power in the neighborhood of the victim frequencies.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Abstract: In an echo processing method and device which can detect an accurate echo section without effects of a far end signal, an echo delay, and a reduction of an echo cancellation amount, a signal of a specified frequency band is generated in conformity with a near end signal, and the signal of the specified frequency band is added to the near end signal to form a transmitting signal. Receiving signals are separated into the signal of the specified frequency band and a signal of a band other than the specified frequency band. An echo section is detected based on the signal of the specified frequency band separated. An echo component in the signal of the band other than the specified frequency band is removed and a level of the echo component is detected based on the near end signal in the echo section.

Abstract: An echo cancellation circuit includes delay units at its input unit and output unit in order to adjust the delay caused by a FIFO. The delay times of the delay units are trained using a training signal, such the REVERB signal, that is a system initializing signal. In addition, an upstream pilot tone is used to tune the delay time. The performance of the echo cancellation circuit is optimized according to the device and the method described.

Abstract: The present invention relates to systems and methods that remove echo from a signal via a novel echo cancellation technique that supports arbitrary playback sampling rates. The novel echo cancellation technique transforms a playback signal to a frequency domain representation and converts its sampling rate to a sampling rate of a frequency domain transformed received signal for the appropriate number of frequency bins. This conversion is achieved via an exact or interpolated approached. The re-sampled playback signal transform is then utilized in connection with the received signal transform to remove echo associated with the playback signal from the received signal.

Abstract: A system and method facilitating acoustic echo cancellation convergence detection is provided. The invention includes an acoustic echo cancellation convergence detector having a center clipping component, a convergence calculator, a convergence filter and a convergence statistic component. The invention provides for the acoustic echo cancellation convergence detector to provide an output regarding convergence of an adaptive filter based, at least in part, upon a filtered convergence statistic. Optionally, the acoustic echo cancellation convergence detector can provide an output associated with convergence history.

Abstract: A circuit arrangement for two-wire/four-wire conversion in a DMT system, which is connected to a digital reception path, a digital transmission path and also an analog transmission/reception path and which has nonlinear echo cancellation in the time domain of a signal and also linear echo cancellation in the frequency domain of the signal, furthermore comprises a device for adaptation of the nonlinearities in the frequency domain.

Abstract: A method of controlling an echo canceller in a communications channel, is disclosed wherein input signals from the communications channel are first subbanded into a subband. Echo locations are then identified within the subband and are used to control the echo canceller. Typically, the echo canceller will be a fullband echo canceller with an adaptive filter, in which case the echo locations are used to control the filter coefficients.

Abstract: Disclosed is an apparatus for cancelling far endpoint echo signals in audio signals transmitted from a near endpoint to a far endpoint. In one embodiment, the apparatus includes a near endpoint analysis filter bank operable to divide a near endpoint signal into a plurality of near endpoint subband signals, a far endpoint analysis filter bank operable to divide a far endpoint signal into a plurality of far endpoint subband signals, and a background signal power estimator operable to determine background noise at the near end.

Abstract: A method of estimating an echo return loss of a communication link measures the peak power value of a signal received from a far end of a communication link and the corresponding peak power value of a signal received from the near end of the communication link. When the near end signal is free from near end voice and excessive noise, the near end corresponding peak will be the echo of the far end power. The far end power is monitored and when a peak is detected, a period, which can be equal to a filter length period, is initiated. When the period has run, the delay line attached to the near end is polled to determine the peak power in the delay line. This peak power will correspond to the far end peak. If a peak, exceeding the first peak is detected before the period has run, the period is reinitiated. The echo return loss estimate is calculated as a ratio of the far-end peak power value to the near-end peak power value.

Abstract: The invention provides an apparatus and method for bulk delay insertion prior to echo cancellation, with echo suppression. The exemplary apparatus embodiment (100) bandpass filters (120) and down samples (130), by a down sampling factor, a reference signal and a combined signal that includes an echo of the reference signal. An adaptive echo delay filter (140) determines a plurality of filter coefficients, using the down sampled reference signal, and using an error signal formed as a difference between the down sampled combined signal and the adaptive echo delay filter output. When the filter coefficients have converged, an echo delay time period is determined as a product of the tap location of the maximum filter coefficient and the down sampling factor. When the echo delay time period is greater than the capacity of an associated echo canceller, an adaptive delay inserter (110) inserts the echo delay time period prior to echo cancellation.

Abstract: A method of eliminating interference components in a microphone signal by generating a compensation signal and subtracting the compensation signal from the microphone signal. The compensation occurs completely in the frequency domain and the output signal is processed in the frequency domain as well. Measures for reducing the expenditure during the signal processing are specified. For example, advantageous modifications provide that a filter setting, obtained during a preceding speech pause, be used for eliminating interference in a voice signal and/or that the simulation filter be divided into several partial filters for long pulse responses. In particular, the invention is suitable for eliminating interference signal components, e.g., caused by a radio or the like, from a voice input signal in a motor vehicle, the source signal of which is available as reference signal.

Abstract: Real and imaginary components of the discrete Fourier transform (DFT) of the residual echo in an acoustic echo cancellation system are separately estimated using mapping functions derived without an assumption of statistical independence between separate components of a near-end microphone input signal. The residual echo estimator calculates the real and imaginary components using an approximation of the microphone input signal without the echo. A different approximation is used depending on whether the near-end talker and/or the far-end talker are talking.

Abstract: The present invention relates to a method for improving acoustic noise attenuation, in which the method includes an adaptive control for partial-band echo compensation as well as global-band post-filtering for suppressing residual echo in hand-free devices. This method also uses a level balance as well as controllable frequency-selective echo compensation with partial-band processing. After the frequency-selective echo compensation, the signal is submitted to a post-filtering in another frequency-selective filter using a Wiener-equation adjustment algorithm (Wiener filtering). A single control value (increment vector) is used for controlling both the frequency-selective echo compensation and the other filter. This method can thus be implemented with a very reduced amount of calculations so that it can also be used in simple consumer-directed processors.

Abstract: Echo-canceling apparatus of the invention includes transfer function estimation unit which estimates a transfer function corresponding to the reverberation of a room attached to a voice after it is output from a loudspeaker and before it is input to a microphone, a first filter unit which operates using the transfer function, a first subtraction unit which subtracts the output signal of the first filter unit from the signal from the microphone, a second filter unit which operates using the transfer function copied from the first filter unit in case the estimation accuracy of the transfer function estimation unit is high, a second subtraction unit which subtracts the output signal of the second filter unit from the signal from the microphone, a singing detection unit which detects singing, a notch filter unit which notches a specific frequency band component in the signal received from a far-end speaker, and a switch unit which selects between the signal from the far-end speaker processed by the notch filter u

Abstract: An echo canceller processing echo, noise and near end talk in a narrower, but still intelligible, frequency band for reducing required processing power and complexity. In a preferred embodiment of the present invention, an input audio signal of captured sound in an audio communication system is decimated and then divided into a number of sub bands by an analyze filter. Each sub band is processed as in background audio echo cancelling by subtracting the signal with an echo estimate from a model of the acoustic signal in the respective sub band, except from that the signal is also bypassed, adjusted by a filter and subtracted from the processed signal. The resulting signals are then recombined by a synthesize filter and interpolated to the original sampling rate and bandwidth. Finally, the output from the synthesize filter is added to the input audio signal, which has been delayed and adjusted by a filter.

Abstract: A hybrid circuit models a plurality of attributes of a communication system including a subscriber line coupled to a line interface circuit. The plurality of attributes includes subscriber line impedance, and impedance of a coupling transformer. The attributes are based on other factors, such as whether or not bridge taps exist at or near the line interface circuit on the subscriber line. The plurality of attributes are modeled by a specific arrangement of resistive and capacitive elements to substantially duplicate the collective transforming effects of the attributes on a transmission signal being sent out on the subscriber line. The transmission signal is transformed and provided to an output, where it is subtracted from a composite signal representing a combination of the actual transformed transmission signal and a receive signal. The subtraction yields an isolated receive signal, which is later processed to recover the full receive signal.

Abstract: A method for equalizing a signal in a transceiver includes receiving an analog signal and adaptively equalizing the analog signal in an adaptive equalization filter to produce an analog filtered signal. The method also includes converting the analog filtered signal to a digital signal, digitally adapting the digital signal in a digital finite impulse response (FIR) filter, and modifying at least one digital filter coefficient of the digital FIR filter according to a signal error associated with an output of the digital FIR filter. The method further includes providing the at least one modified digital filter coefficient of the digital FIR filter to an analog equalization controller. In addition, the method includes using the at least one modified digital filter coefficient of the digital FIR filter in the analog equalization controller to adaptively equalize the analog signal in the adaptive equalization filter.

Abstract: A method corrects an input DC offset signal generated in front-end analog signal processing circuitry in a transceiver capable of operating in a high frequency local area network. An output signal from the front-end analog signal processing circuitry of the transceiver is received in a DC offset correction controller. A DC offset signal component in the output signal from the front-end analog signal processing circuitry due to an accrued DC offset is detected. A DC offset correction signal is sent from the DC offset correction controller to an adjustable biasing circuit coupled to a differential amplifier of a receiver line driver of the transceiver to modify the DC offset signal component.

Abstract: A double-talk detector (109) for an acoustic echo canceler (104) of a VoIP terminal (101) converts (216,226) received far-end signals (212) and transmitted near-end signals (222) into the frequency domain and high-pass filters out (218,228) low-frequencies representing noise to obtain a complex reference signal and a complex error signal, respectively. It then correlates (230) the complex signals and computes (232) the instantaneous error energy (234) and smoothed (long-term average) error energy (235) of the complex error signal. If the convergence value is below 0.06 (300), the signals are converged and double-talk detection is enabled (306). If the convergence value is above 0.09 (312), the signals are diverged and double-talk detection is disabled (316). If double-talk detection is not enabled, an error threshold (236) is set (304,318) to the value of the smoothed error energy.

Abstract: A howling control apparatus and howling control method for controlling time up to the cancellation of howling suppression according to a howling occurrence situation, thereby eliminating the repetition of howling suppression and cancellation.

Abstract: An apparatus for enhancing intelligibility of a voice signal in a noisy environment includes a first noise estimator which estimates a far-end noise component from the far-end signal, and a second noise estimator which estimates a near-end noise component from the near-end signal. A noise reduction calculator determines a noise reduction gain from the estimated far-end noise, and an echo gain calculator determines an echo control gain. A master gain calculator combines both the echo control gain and the noise control gain into a master gain which is applied to the far-end signal. A comfort noise generator applies the pre-set minimum threshold of the master gain and constantly matches the spectrum of the far-end noise to synthesize a background noise for selectively mixing onto the far-end signal when echo is determined. Echo is determined based on a comparison of the near-end signal and the far-end signal spectra and by compensating the total gain applied.

Abstract: Processing complexity in an acoustic echo canceller can be reduced by partitioning the input near-end and far-end signals into respective pluralities of frequency subband signals. Also, due to the complex conjugate symmetry of conventional Discrete Fourier Transform matrices, not all of the subband signals of each plurality need be subjected to echo cancellation processing. The subband signals that would have otherwise been produced by echo cancellation processing are instead provided by producing complex conjugates of selected ones of the subband signals that have been subjected to echo cancellation processing.

Abstract: Method and apparatus for an improved echo canceller provide for faster convergence in echo canceling, which also contribute to better voice quality and provide for an increase in the capacity (number of channels) of a multi-channel voice processing products. Whitening filters are used to flatten the spectrum of the signals that pass through the echo canceller. Higher convergence rate is achieved when flattened signals pass through the echo canceller. A reconstruction filter is used to introduce the spectral envelop back into the outbound signal.

Abstract: A system and method of Acoustic Echo Cancellation includes a digital filter used to estimate the echo path is split into 2 filters: one fixed infinite impulse response (IIR) filter and one adaptive finite impulse response (FIR) filter. The fixed filter models the composite frequency response of the loudspeaker, its cabinet enclosure and the direct coupling path between the loudspeaker and the microphone(s). The adaptive filter models the time varying characteristics of the indirect coupling path between the loudspeaker and microphone.

Abstract: A speaker-phone that is operable to perform system change detection, double-talk detection and if desired, optimal step-size control. The invention differentiates between system change and double-talk and is operable to these two situations differently. The invention does not rely solely on the energy measurements that are employed by conventional speaker-phones. Highly reliable system change detection is offered when compared to conventional speaker-phones. The direction and the amplitude of the system change is measured with very good detection probability in the invention. For double-talk detection, the invention does not rely on assumptions that the far-end echoes are of lower amplitude that the near-end signal in performing double-talk detection. The invention performs double-talk detection within a large diversity of echo-paths and background noise levels. A designer may tune the speaker-phone system to ensure proper double-talk detection near low levels of the near-end signal.

Abstract: A method and noise reduction apparatus comprises a microphone array including a plurality of microphone elements for receiving a training signal including a plurality of training signal samples, and a working signal including a plurality of working signal samples, and at least one frequency domain convertor coupled to the plurality of microphone elements for converting the plurality of training signal samples and the plurality of working signal samples to the frequency domain. A signal spatial correlation matrix estimator is coupled to the at least one frequency domain convertor for estimating a signal spatial correlation matrix using the converted plurality of training signal samples. An inverse noise spatial correlation matrix estimator is coupled to the at least one frequency domain convertor for estimating an inverse noise spatial correlation matrix using the converted plurality of working signal samples.

Abstract: Switch 141 continuously switches between received signal 2 and a supplemental signal, which is obtained by processing the received signal 2 through filter 145 to use the output of switch 141 in place of received signal 2. Accordingly, adaptive filters 122 and 124 operate sometimes by using received signal 2 as the input signal and sometimes by using the supplemental signal as the input signal, so that it is possible to obtain adaptive filter coefficients by using twice the number of conditional equations as the case of using only received signal 2 as the input signal. Therefore, since the adaptive filter coefficients do not becomes indefinite, it is possible to converge the coefficients to the correct values. Further, since switching period between the original and the supplemental signals is controlled to be longer than the sampling period of the received signal, it is possible to suppress aliasing distortion of the received signal directly supplied to a speaker and to be maintain better sound quality.

Abstract: The invention is a multiple channel teleconferencing system employing a stereophonic acoustic echo canceler that exploits the coherence between multiple channels. In accordance with the invention a small non-linearity is incorporated into each channel path between the microphone and the speaker and an efficient frequency domain adaptive algorithm is implemented in the echo canceler circuit. The frequency domain algorithm converges to a solution much more quickly, than, for instance, a time domain, normalized least mean squares, solution.

Abstract: A communication device and method for detecting an infinite return loss (IRL) condition and temporarily halting or bypassing operations of an echo canceller in response to detecting that condition. The echo canceller includes an IRL detection module that analyzes the echo return loss and the echo return loss enhancement measured from the far-end and near-end signals. Based on the analysis, the IRL detection module determines whether to bypass the subtraction of the estimated echo in the outgoing near-end signal or continue normal operations.

Abstract: A signalling tone detector for use in a communications network carrying voice traffic incorporating single frequency and dual frequency signalling tones comprises a first output path, a second output path, and a switch arranged to selectively couple an input signal to one or other of the output paths. The first output path comprises an automatic frequency control circuit for determining a mean frequency of an input signal, comparison means for comparing the mean frequency with stored frequency values corresponding to single tone signalling frequencies and mean values of pairs of dual tone signalling frequencies, a first discriminator for determining the presence of either a single frequency or a pair of frequencies, and a second discriminator for providing a signal output indicative of the presence of a single frequency or pair of frequencies.

Abstract: In methods and apparatus for echo cancellation using sub-band adaptive filtering, the bandwidth of each of a number of sub-band analysis filters, as well as the bandwidth of each of a number of synthesis filters, is increased as compared to a corresponding bandwidth in a conventional system. More specifically, the −3 dB bandwidths of adjacent filters in an analysis or synthesis filter bank according to the invention are designed to overlap one another. For example, each of M (an integer) sub-band filters in an analysis or synthesis filter bank according to the invention can include a −3 dB bandwidth somewhat greater than 1/M times a total bandwidth of interest. The increase in bandwidth tends to cancel aliasing effects resulting from sub-band down-sampling, and an echo canceling device according to the invention can therefore provide superior output signal reconstruction as compared to conventional devices.

Abstract: Echo cancellation in data communication between modems utilizes analog echo cancellation to lessen reduction of usable dynamic range of the received signal at the input to the analog-to-digital converter (DAC) in the receiver. Two digital-to analog (D/A) conversions are provided in the modem's analog front end (AFE). One generates the analog signal for transmission. The other generates an analog representation of a cancellation signal that is used to electronically cancel the echo before analog-to-digital (A/D) conversion of the received signal. A preferred embodiment utilizes multiplexed DAC architecture to emulate two DACs by sharing DAC circuitry between data paths of the two D/A conversions.

Abstract: An echo canceller (110) for a full-duplex communication system (100) such as a speakerphone determines a gain factor to be used in sub-band adapter coefficient updates by using both time domain and sub-band signals to determine the state of the system (100). In one embodiment a gain factor controller (120) uses the sampled time domain signal to determine if the system (100) is in the silence state, and if not uses the sub-band signals to determine whether the system (100) is in the talk, double-talk, or listen state. The gain factor controller (120) selectively changes the gain factor and selectively performs coefficient updating based on the state. In another embodiment the gain factor controller (120) determines the state and selectively updates coefficients using state information developed on a band-by-band basis.

Abstract: The magnitude processing circuit 607 non-linearly processes the received signal 2 and then outputs a non-linear processed signal. The delay processing circuit 300 further delays the non-linear processed signal and then outputs a delayed processed signal. The delayed processed signal is time-multiplexed with the non-linear processed signal. The multiplexed signal is used instead of the received signal 2. The adaptive filters 122 and 124 operate alternately with the non-linear processed signal or with the delayed processed signal obtained by non-linearly processing and then delaying the received signal 2 as the input signal. Adaptive filter coefficients can be obtained by using twice the number of conditions compared with the number is twice that for the case where only the received signal 2 is used as the input signal.

Abstract: In an acoustic echocanceller (6), an estimate of an echo signal is determined by an adaptive filter (10) and is subtracted from the input signal by a subtracter (14). The spectrum estimator (12) determines the frequency spectrum of the estimate of the echo signal, and the filter (16) filters the output signal of the subtracter (14) with a filter having a transfer function dependent on the spectrum determined by the estimator (12). The use of this combination results in a substantial improvement of the suppression of the echo signal.

Abstract: A compander circuit (20) for echo cancellation in telecommunications apparatus with an estimator module (21) for estimating an echo coupling and a delay of a line echo and a characteristic module (23) with an associated multiplier (24) for controlling the short-time output level of zout(k) as a function of the short-time input level of zin(k) is characterized in that at all inputs and outputs of the compander circuit (20), a plurality of N communications channels are applied and outputted, respectively, at an N-fold clock rate as compared with a single communications channel; that all memory units in the compander circuit (20) are designed as multiplexer-buffer-demultiplexer (MBD) units (30); and that a modulo-N counter (26) is provided which controls all MBD units (30) so that in each of the MBD units (30), always the same memory cell (351, 352, 35N) assigned to a particular communications channel is addressed.

Abstract: A process for suppressing total echoes comprising a plurality of partial echoes in telecommunications (TK) equipment such as terminals, transmission systems, or switching equipment by means of adaptive FIR (=Finite Impulse Response) filters that simulate the total echo and subtract it from the echo-affected useful signal transmitted to the respective TK equipment, whereby n partial FIR filters are provided, which respectively simulate a partial echo, is characterized in that m parallel, mutually independent TK channels are provided with an echo suppression function and that for this purpose a variable sub-quantity of the partial FIR filters is assigned to a TK channel to be operated, respectively, as required depending on the number of partial echoes and the size of the occurring delay times between signal and echo.

Abstract: In an acoustic echocanceller (6), an estimate of an echo signal is determined by an adaptive filter (10) and is subtracted from the input signal by a subtracter (14). The spectrum estimator (12) determines the frequency spectrum of the estimate of the echo signal, and the filter (16) filters the output signal of the subtracter (14) with a filter having a transfer function dependent on the spectrum determined by the estimator (12). The use of this combination results in a substantial improvement of the suppression of the echo signal.