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: 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: Apparatuses, systems, and methods for a multi-carrier communication system that detects and reduces an echo from a non-linear element present on the transmission medium. In an embodiment, a training period is established between a first transmitter-receiver device and a second transmitter-receiver device in the discrete multiple tone system that separates communication signals into two or more separate frequency bands. Noise caused by an echo generated by a non-linear element present on the transmission medium is detected during the training period. The significance of the non-linear echo contribution to the overall ambient noise level present in the system may be determined.

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: 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 method and apparatus for performing double-talk detection in an acoustic echo canceller in which a detection statistic is advantageously computed based on an estimate of a cross-correlation between the far-end signal and the return signal which has been normalized with use of an estimate of a covariance matrix of the far-end signal. The estimate of the cross-correlation between the far-end signal and the return signal may be further normalized with use of either an estimate of a variance of the return signal or an estimate of a covariance matrix of the return signal. In certain illustrative embodiments of the invention, one or more of these quantities may be estimated based on signal samples sampled over a predetermined time window. And in another illustrative embodiment of the present invention, the coefficients of the adaptive filter employed in the acoustic echo canceller itself are advantageously used to compute the detection statistic.

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: 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: The filtering coefficients of a frequency domain stereophonic echo canceller are adapted by a method which takes account of the cross-correlation between the input signals relating to the two channels. In particular, the adaptation process takes account of the coherence function, reducing the problems commonly encountered with stereophonic cancellation schemes when relatively correlated input signals occur.

Abstract: Apparatus and methods for adaptively, reliably, and accurately measuring the duration of an alerting or other signaling tone. A tone detector in accordance with the principles of the present invention accurately measures a duration of a tone burst (i.e., tone pulse) using a long frame length DFT and a short frame length DFT. Based on information obtained by the short frame length DFT, the long (i.e., standard) frame length DFT may be interrupted and reset to begin a new frame of data synchronized with the start of the next short frame corresponding to the processing by the short frame length DFT to minimize noise in the processing by the long frame length DFT and to greatly improve the resolution of the measurement of the duration of a particular tone to correspond to the lengths of the short frames used by the short frame length DFT.

Abstract: A zero-delay structure to be used in sub-band echo cancellation systems. In telecommunications, a delay is usually introduced in sub-band decomposition and/or sub-band reconstruction. This invention will help remove this. This structure not only removes the delay but does not sacrifice the performance of the echo canceller. It is to be used in combination with any sub-band-based echo canceller or wavelet echo canceller.

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: Echo is canceled in a communication device including at least one microphone and a loudspeaker. Signals directed to the loudspeaker are filtered through at least one first filter. The filtered signals are subtracted from signals received by the microphone. At least one second filter filters the subtracted results to produce a signal corresponding to speech signals received by the microphone. At least one third filter filters the subtracted results to produce a signal corresponding to echo from the loudspeaker. The output from the third filter is subtracted from the output from the second filter to produce an echo-canceled signal. The device may include a number of microphones, and the first, second, and third filters may each include a number of filters equal to the number of microphones. The outputs from the second filter may be added, the outputs from the third filter may be added, and the sum of the outputs from the third filter may be subtracted from the sum of the outputs of the second filter.

Abstract: A DFT is used to compute the magnitude and phase of the frequencies desired to be detected in a frame of sampled data by performing 2 dot products for each such frequency. The two dot products are calculated using the samples in the frame and an equal number of equally spaced cosine and sine coefficients for each frequency to be detected.

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 a splitterless Digital Subscriber Line (DSL) modem 30, a receiver architecture (100) that permits monitoring of harmonics within the upstream channel of a DSL connection. The receiver (100) can detect the harmonics by monitoring the harmonics in some empty tones or specified tones. A threshold can be set to differ the normal noise power and harmonics power in those empty or specified tones. When a POTS device (10) coupled to same wire line pair (20) as the DSL modem (30) goes off-hook, the DSL modem (30) can switch from the normal state to an off-hook state in which the harmonics are reduced to minimum by transmitting only the upper sub-band tones of the upstream transmission channel at reduced power levels.

Abstract: A technique for separating an acoustic signal into a voiced (V) component corresponding to an electrolaryngeal source and an unvoiced (U) component corresponding to a turbulence source. The technique can be used to improve the quality of electrolaryngeal speech, and may be adapted for use in a special purpose telephone. A method according to the invention extracts a segment of consecutive values from the original stream of numerical values, and performs a discrete Fourier transform on the this first group of values. Next, a second group of values is extracted from components of the discrete Fourier transform result which correspond to an electrolaryngeal fixed repetition rate, F0, and harmonics thereof. An inverse-Fourier transform is applied to the second group of values, to produce a representation of a segment of the V component. Multiple V component segments are then concatenated to form a V component sample stream.

Abstract: The problem of talk-off and talk-on errors occurring when a common detection method for predefined signaling signal like a calling alerting signal CAS is used.