Abstract: A method for suppressing echo signals generated in a communication path such as acoustic coupling between a speaker and a microphone in a telephone device handset includes the steps of monitoring signals supplied to the communication path to determine an attribute thereof and masking signals received from the communication path as a function of the determined attribute of the monitored signals thereby to suppress echo.

Abstract: Self-interference cancellation in two-way relayed communications is improved by creating models of upconverter and downconverter imperfections and then compensating for those imperfections before self interference cancellation processing. The model includes compensation for phase offset, for amplitude imbalance and for leakage in the mixers.

Abstract: Device and method for noise suppression in telephone equipment, in particular with hands-free equipment, having a level discriminator, an additional filter with a group propagation time of less than 2 ms being inserted into the transmission and receiving paths of the telephone equipment.

Abstract: A telephone includes a first channel having a microphone input and a line output, a second channel having a line input and a speaker output, and each channel having a plurality of band pass filters with their outputs coupled to a multiplex circuit. The multiplex circuit selects subsets of the signals in each channel in accordance with the magnitudes of the signals in each band. Thus, full duplex operation is provided in accordance with the spectral content of each channel. In one embodiment of the invention, the multiplex circuits initially allocate alternate bands within each channel and complementary bands between the channels, thereby emulating a complementary comb filter. Thereafter, bands are allocated according to spectral content.

Abstract: A method includes receiving a plurality of packets from at least two voice packet streams. The method also includes generating at least two voice signals using the plurality of packets. Each voice signal is associated with one of the voice packet streams. In addition, the method includes identifying an echo of one of the voice signals in at least one other of the voice signals.

Abstract: An echo canceller includes two parallel filters and a controller. The controller chooses the best of the two filters for the final echo cancellation. The filter coefficients are exchanged between the two filters constantly so that both filters retain performance all the time even if double-talk and echo path variations occur very closely in time. One filter is strongly robust and will not diverge during double-talk, and another filter is weakly robust and will converge rapidly during echo path variation.

Abstract: System and method for processing communication signals in a communication system having a detector for detecting a parameter of a communication signal. A communication signal is provided to a non-linear processor (NLP) adapted to examine the signal and to decide whether or not to enter an active state based upon a parameter of the signal. If the NLP enters an active state, the NLP performs non-linear processing on the signal. The NLP communicates to the detector whether the NLP is active or inactive. If the NLP is active, a processing step of the detector is disabled.

Abstract: A method and apparatus that performs specific automatic gain control in an echo canceller (16). The disclosed method and apparatus utilize a low complexity algorithm that performs automatic gain control to achieve greater speed of convergence of the echo canceller (16) and an increased ability to distinguish between near-end speech and echo through the use of specific automatic gain control. A double talk threshold of the echo canceller is adjusted by using, in part, computed gains of inbound and outbound signals internally within the echo canceller in order to adjust the echo canceller double talk threshold.

Abstract: A base band echo cancellation method and apparatus utilizing Laguerre echo estimation. The method begins with a series of transmitted symbols that are stored in a buffer. The buffer is segmented into a head portion and a tail portion. The symbols in the tail portion are passed through a Laguerre or IIR filter and then reinserted into the tail portion of the buffer. An echo estimation signal is generated based on the symbols in the buffer. The apparatus includes a transmit symbol buffer that is coupled through a modulator to a Laguerre or IIR filter. A demodulator couples the filter back to the buffer. An echo estimator is also coupled to the buffer.

Abstract: An echo canceler has its otherwise conventional tapped delay line basis functions augmented with at least one pair of basis functions realizing two quadrature responses of a complex pole. The frequency of this complex pole is selected as substantially equal to the frequency of the dominant pole of a highpass filter inserted into the echo path of the echo canceler, where the dominant pole is the pole closest to the unit circle. This echo canceler arrangement results in a substantial improvement in attainable echo-return-loss enhancement (ERLE).

Abstract: Echoes caused by discontinuities in a telecommunications transmission lines cause double talk which cause divergence problems with echo cancellers. Divergence problems in echo cancellers cause considerable problems in interactive caller response systems, they cause less problems if all parties to a call are human. One of the biggest problems is the effect of the echoes on automatic speech recognition systems. The echo canceller method and system presented cancels echoes to reduce their interference with ASR and yet provides stability in the presence of double talk.

Abstract: A timer operated hardware switching device to control the connection between a computer and telephone or data circuit. The device improves the security of the computer by limiting external access from the Internet or other networks. Security software programs such as fire walls are enhanced by the invention, which prevents electronic attacks during periods when the switching device is in the disconnected state. The device is simple to understand and can be easily verified in operation. Additionally, such a device can be used with conventional telephone applications to prevent calls from being made during specific time periods. The invention can be embodied as a standalone device, a device for use with a conventional power timer, and a device built into a computer or other computer accessory.

Abstract: A system and method provide ASR talkoff suppression in a prompt-and collect voice transaction system in which a prompt signal is sent and an input signal is received. A comparison is made of a characteristic of the prompt signal and the input signal. Processing of the input signal to reduce the likelihood of an ASR response thereto is performed if the result of the comparison satisfies a predetermined criterion.

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 apparatus for reducing echo feedback in a wireless communication system (10) is accomplished when a receiving communication unit (24) senses a feedback signal, or echo, via an ancillary communication path. The receiving communication unit is a targeted recipient of an original audio signal generated by a transmitting communication unit (22), where the original audio signal (42) is conveyed to the receiving communication unit. Upon detecting the feedback signal and determining that it exceeds a feedback threshold, the receiving communication unit attenuates an audible output of the original signal to reduce echo to the transmitting communication unit. In addition, the receiving communication unit, and/or the transmitting communication unit include echo canceller to further minimize the echo within the digital communication system.

Abstract: An echo canceller circuit for use in an echo canceller system is set forth that provides sensitive double-talk detection. The echo canceller circuit comprises a second digital filter having adaptive tap coefficients to simulate an echo response occurring during the call. The adaptive tap coefficients of the second digital filter are updated over the duration of the call using a Least Mean Squares process having an adaptive gain a. A channel condition detector is used to detect channel conditions during the call. The channel condition detector is responsive to detected channel conditions for changing the adaptive gain a during the call. For example, the channel condition detector may detect the presence of a double-talk condition and set the adaptive gain a to zero. Similarly, the channel condition detector may detect the occurrence of a high background noise condition and set the adaptive gain a to a level less than 1 that is dependent on the detected level of the background noise.

Abstract: The present invention relates to an echo reduction method implemented with a multi-sensor sound pickup device (8) forming an antenna (10), and a sound playback device (13). The method consists in subjecting the output signals ym(t) from the sensors (11m) to complex weights w(f), said weights w(f) being calculated by maximizing the directivity factor Fd(f) under constraints at low frequencies and in the near field. The invention also provides an echo reduction apparatus (6). The apparatus comprises: a multi-sensor sound pickup device (8) forming an antenna (10); means for calculating a first constraint; and means for calculating a second constraint.

Abstract: Handsfree telephony continues to be an increasingly desirable feature of modern telecommunications, whether in a conference room or mobile setting. Fundamental to the user acceptability of these systems is the performance of algorithms for acoustic echo cancellation, the purpose of which are to prevent the far-end signal from being transmitted back to the far-end talker. Most speech coding algorithms are based on some variant of Linear Predictive Coding (LPC), which reduces the amount of bits sent across a channel. Instead of doing echo cancellation in the time domain, the invention involves operating an acoustic echo canceller on the LPC parameters at the receiver, before the decoding stage.

Abstract: An echo canceller is provided in which a down-sampling circuit converts a 16-kHz sampling frequency of a wide-band voice signal output to an 8-kHz sampling frequency of a narrow-band voice signal supplied at an input terminal, an adaptive filter estimates, from the wide-band voice signal whose sampling frequency has been down-sampled to 8 kHz in the down-sampling circuit, an echo signal coming from a speaker to a microphone and having an echo path characteristic imparted to the echo signal by an echo path filter, and a subtraction circuit subtracts from the microphone input signal the echo signal having been estimated by the adaptive filter.

Abstract: A comfort noise generating apparatus designed to reduce a feeling of fracture in a voice sound and a feeling of unsuitableness at timings before and after a process by an NLP or the like. A comfort noise according to a voice signal after an unnecessary component has been removed is formed and added to this voice signal. The apparatus comprises: a noise generator for generating a noise which is multiplexed to a voice signal after completion of a process by an NLP; a signal level analyzing part for measuring a signal power of a voice signal before the process; a multiplier for varying noise characteristics from the noise generator in accordance with an analysis result of the signal level analyzing part; a signal frequency characteristics analyzing part for selecting the optimum coefficient from a coefficient register; a band pass filter for forming a comfort noise by the selected coefficient and a white noise signal; and an adder for multiplexing the comfort noise to the voice signal.

Abstract: An echo suppressor uses a multi-state operating sequence to provide non-disruptive monitoring of a remotely broadcast program during a performer's live performance at a local site. The operating sequence is actuated by a voice activated switch that initiates signal processing for correlating a locally originating audio (e.g., the performer's voice) to its echo in the broadcast program. During correlation states in the operating sequence, the echo suppressor switches monitoring to a local microphone signal to allow the performer to hear his or her own voice while preventing echo from being heard before correlation is achieved and echo suppression can begin. The correlation states include an initial correlation at a reduced sampling rate to detect echo within a wide delay time window and determine the locally originating audio is on-the-air.

Abstract: An echo canceller circuit for use in an echo canceller system is set forth that provides sensitive double-talk detection. The echo canceller circuit comprises a second digital filter having adaptive tap coefficients to stimulate an echo response occurring during the call. The adaptive tap coefficients of the second digital filter are updated over the duration of the call using a Least Mean Squares process having an adaptive gain a. A channel condition detector is used to detect channel conditions during the call. The channel condition detector is responsive to detected channel conditions for changing the adaptive gain a during the call. For example, the channel condition detector may detect the presence of a double-talk condition and set the adaptive gain a to zero. Similarly, the channel condition detector may detect the occurrence of a high background noise condition and set the adaptive gain a to a level less than 1 that is dependent on the detected level of the background noise.

Abstract: An interactive apparatus allows the user to interrupt an outgoing prompt and remove component which is normally found in the users' responses (e.g., a frequency band) from the outgoing output prompt. An input signal analysis unit in the apparatus is able to detect the response of the user (and distinguish it from an echo of the outgoing prompt) by noting the presence of the component which is lacking from the outgoing prompt. As an alternative, the apparatus may force spaced timeslots in the outgoing signal to silence. In that case, the input signal analysis unit can detect the presence of the user's signal over a predetermined time interval. As well as being applicable to apparatuses which involve the user in prompt/response dialogues, the invention is also useful in relation to the interruption of messages being replayed by voice-controllable answerphones of the like.

Abstract: An echo cancellation device (ECD) comprises an echo canceller (EC) including a transfer function estimator (EST, H) and a subtractor (ADD) and a residual echo suppression device (G, ADD2). The residual echo suppression device (G) comprises a residual echo filter (G) having an adjustable filter function (g). This filter function (g) can be adapted to either remove from the subtractor output (TNE′) the spectral characteristics relating to the reception signal (RFE) and/or to emphasize in the subtractor output signal (TNE′) a background signal spectral content relating to the transmission signal (TNE). A noise generation means (NGM′) can be provided at the output of the adaptable filter (G) for injecting a noise process in to the filter output signal (TNE′) prior to a speech coding in a speech coder (COD). The noise process masks in the filter output signal a spectral content relating to the reception signal (RFE).

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 methods and apparatus for echo cancellation, a noise estimate is selectively added to a downlink communications signals to improve the convergence speed and stability of an echo-canceling adaptive filter. The added signal content enables the echo-canceling adaptive filter to more quickly track echo path changes during user communications and prevents divergence of the adaptive filter during periods in which the downlink signal does not contain information (e.g., speech) sufficient to develop a quality echo path estimate. To prevent performance degradation from a user perspective, the added noise is made to resemble existing system noise (e.g., either near-end or far-end background noise). In exemplary embodiments, filter convergence speed and stability are further improved by whitening the spectrum and increasing the power level of the added noise whenever a near-end user is active and masking the added noise.

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 videoconferencing system uses an audio input unit including: two directional microphones disposed in directions with directivity being opposite to each other, the microphones forming a pair and a plurality of the pairs being located with directivity shifted from each other, cancellation elements for canceling sound components of the same phase from sounds inputted to the microphones of the pair, extraction elements for extracting sound components of voice band from sounds inputted to the microphones of the pair, a calculation element for calculating a difference between levels of the sound components of voice band extracted by the extraction elements to decide the microphone which has the larger level in the pair having the largest level difference, and selection elements for selecting sounds as input sound, in which the sound components with the same phase are canceled by the cancellation elements from sound inputted to the microphone decided by the calculation element.

Abstract: Methods and apparatus for compensating for the background noise modulation caused by the operation of a non-linear processor on an audio signal include a controller that receives operating status information from the non-linear processor and generates a noise reduction control signal. The methods and apparatus further include a noise reduction circuit that receives the noise reduction control signal from the controller. The noise reduction circuit reduces the background noise in the audio signal when a noise reduction control signal is generated by the controller and injects synthesized background noise into the audio signal when a noise reduction control signal is not generated by the controller.

Abstract: Methods and apparatus for processing at least one voice signal in which a centralized voice processing unit controls operation of a plurality of voice processing block. In a first embodiment, the centralized voice processing unit comprises a centralized voice activity detector that provides at least one voice activity indication to the plurality of voice processing blocks. In a second embodiment, the centralized voice processing unit comprises a centralized noise estimator that provides at least one noise estimate to the plurality of voice processing blocks. In a third embodiment, the centralized voice processing unit comprises a centralized signal characteristic estimator that provides at least one signal characteristic estimate to the plurality of voice processing blocks.

Abstract: Methods and apparatus for processing a transmitted voice signal include a centralized frame controller providing at least one boundary control signal to voice processing blocks and controlling the operation of the voice processing blocks on the transmitted voice signal based upon the boundary control signal.

Abstract: An enhanced near-end voice signal may be generated in a hands-free environment by receiving an audio signal, generating an estimated acoustic echo signal, and generating a processed signal by removing the estimated acoustic echo signal from the audio signal. A near-end enhanced spectrum is then determined, that has one or more ranges of contiguous frequencies over which the detector spectrum takes on its largest values, wherein the range of contiguous frequencies are those associated with a relatively high echo return loss in the processed signal. The processed signal is filtered in accordance with the near-end enhanced spectrum, thereby generating an enhanced near-end voice signal. The enhanced near-end voice signal may then be applied to any of a number of components that are intended to process near-end speech. For example, when applied to a voice activity detector, the amount of energy contained in the enhanced near-end voice signal is then measured.

Abstract: A hybrid echo suppresser and associated methods of use provide echo suppression in a bi-directional communications system, while minimizing desired-voice signal disruption. The hybrid echo suppresser includes a linear echo canceller (LEC), a non-linear processor (NLP), and a control circuit. The LEC operates on an echo-causing signal to produce an estimated-echo signal, which is subtracted from an input echo-containing signal, thereby producing an echo-cancelled signal. The NLP operates on this echo-cancelled signal to eliminate residual echo arising from system non-linearities. The control circuit repeatedly compares the echo-containing signal and the estimated-echo signal to determine if the echo-containing signal includes a desired-voice component. If not, the control circuit operates the NLP in an echo-voice mode having substantial signal attenuation.

Abstract: A network for providing a telecommunications service with automatic speech recognition to a telecommunications user, including a switch in communication with a telecommunications device associated with the telecommunications user for storing identifying information regarding a prior calling party to the telecommunications user, and for detecting a trigger specific to the service in response to a communication from the telecommunications device, and an intelligent resource server in communication with the switch for receiving from the switch the communication from the telecommunications device and a message including the identifying information, for playing an audible message for the telecommunications user in response to receiving the communication, the audible message containing the information regarding the prior calling party and prompting the telecommunications user to place an outgoing communication to the prior calling party, and for automatically recognizing a predetermined keyword spoken by the teleco

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: Full acoustic coupling and echo cancellation is provided in implementations where there is acoustic coupling between acoustic zones.

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: To be able to successfully compensate the echos occurring in bidirectional telecommunications systems, comprising for example 2-wire/4-wire converters and/or AD/DA converters and/or satellite transmission links, the echo coupling factor and echo delay time for a system must be determined with a very high degree of accuracy and independently of the double talk situation. For this purpose, in accordance with the invention a process is provided wherein, by analyzing a dialogue, states and time segments are determined in which the probability of an incorrect evaluation of the coupling factor between the transmission paths and the echo delay time is very low. To improve the measurement accuracy, auxiliary quantities dependent upon the speech dynamics and the characteristic curve of the speech signal are introduced.

Abstract: A speech processor uses at least two speech processing features to enhance the quality of speech signals received by a user during a telephone call. The speech processing features are applied to the speech signals. However, the user only hears speech signals affected by one speech processing feature until both features have fully converged or ramped-up, and the two features are no longer interfering with each other. At that point, a “flash-cut” of the second speech processing feature is activated. The flash-cut instantaneously switches to speech signals affected by both features. This quick transition makes the speech processing features more noticeable to the user, and the user is not subjected to the period where the features interfere. Further, an optional audio indicator is generated before implementing the flash-cut, so the user is alerted to the flash-cut, and the speech processing features are even more noticeable.

Abstract: A method of controlling an echo canceller for providing an echo cancellation signal by subtracting an echo replica signal either from a transmit signal or a receive signal. The method includes the steps of a first step of performing a speech detection based on the echo cancellation signal provided by the echo canceller; a second step of detecting a double talk condition based on a result of the speech detection performed by the first step; and a third step of controlling a learning operation of the echo canceller based on a result of the double talk detection performed by the second step.

Abstract: An echo canceling apparatus suitable for multiplexed transmission lines is provided that achieves a short convergence time and a small number of arithmetic operations. Adaptive filters are located in respective transmission lines. Each adaptive filter adaptively controls tap positions such that tap coefficients are located in dispersive regions which are obtained by removing fixed delays from the impulse response in an echo path. A controller receives tap position information from the adaptive filters to mutually compare the convergence degrees among the adaptive filters. Thus, the controller respectively supplies coefficient update control signals to the adaptive filters so as to suppress coefficient adaptation of an adaptive filter whose convergence is leading compared with the others.

Abstract: An object of the present invention is to provide an echo canceller capable of generating, when the double-talk detection precision is degraded and the learning of a filter coefficient swings by noises, an echo replica signal using a filter coefficient having a low degree of error at the time of detecting a double-talk state. A backup filter coefficient is calculated by adding a predetermined weighting factor times a filter coefficient obtained by the learning and a predetermined weighting factor times the preceding backup filter coefficient stored in second storage means.

Abstract: In accordance with the invention, from the characteristic curve of the speech signal there are derived pulse sequences which are dependent upon the frequency but not the amplitude of the speech signal, so that in the case of a real echo, independently of the attenuation, identical pulse sequences occur in the case of the transmitted signal and, with a time shift, in the case of a received echo. The correspondence between the pulse sequences of transmitted signal and received signal is an indication that the received signal is an echo of the transmitted signal.

Abstract: A telephone system including a base station and a mobile unit in communication with each other is provided with an acoustic echo canceler. A sound signal is input. A reference signal that matches the sound signal is formed. A plurality of encoders and decoders receive the reference signal and form a modified reference signal. The plurality of encoders and decoders match a set of encoders and decoders that operate on the sound signal in the base station and the mobile unit. An adaptive filter receives the modified reference signal and forms a replica signal. A subtractor subtracts the replica signal from an echo signal received by the base station.

Abstract: A hands-free communications unit employs a downlink peak power memory element (150, FIG. 1), a downlink voice activity detector (140), and an uplink voice activity detector (180). The downlink voice activity detector (140) evaluates the incoming signal to determine if the signal includes voice. In a similar manner, the uplink voice activity detector (180) determines if an outgoing signal includes voice. In the event that the incoming signal includes voice and the energy of the outgoing signal exceeds a threshold calculated from the peak power of the incoming signal, the uplink voice activity detector (180) indicates that a double-talk condition exists (350, FIG. 3). In the event that the incoming signal does not include voice, the uplink voice activity detector (180) compares the uplink signal to a voice threshold above a noise floor to determine if the outgoing signal includes voice.

Abstract: An adaptive filter (and a corresponding adaptive filtering method) for use in an echo cancellation environment, comprising a finite impulse response filter having a vector of adaptive coefficients, and an adaptive coefficient vector update module which modifies the vector of adaptive coefficients in response to an error signal and based on a predetermined probability density function for said error signal. In accordance with several illustrative embodiments of the present invention, the probability density function is advantageously non-Gaussian, and has certain advantageous characteristics which provide for robustness in the adaptive algorithm based thereupon.

Abstract: A multi-device audio-video system contains speech recognizing facilities and echo canceling facilities. In particular, plural and functionally separate such speech recognizing facilities and echo canceling facilities are present. Now, the echo canceling facilities combine their forces for by one or more thereof canceling one or more mutually unique cancelable speech entities and combining such cancelled entities for overall non-recognition by the system.