Abstract: In a wave field synthesis apparatus for driving an array of loudspeakers with drive signals, the loudspeakers being arranged at different defined positions, a drive signal for a loudspeaker being based on an audio signal associated with a virtual source having a virtual position with reference to the loudspeaker array and on the defined position of the loudspeaker, at first relevant loudspeakers of the loudspeaker array are determined on the basis of the position of the virtual source, a predefined listener position, and the defined positions of the loudspeakers, so that artifacts due to loudspeaker signals moving opposite to a direction from the virtual source to the predefined listener position are reduced.

Abstract: An echo cancelling device includes a filter-coefficient polarity inverter configured to invert the polarity of a filter coefficient at intervals of a predetermined frame length, an adaptive filter configured to estimate a signal to be inputted from an unknown system by multiplying a signal to be outputted from a speaker by the filter coefficient, and generate the resultant estimated signal, a subtracter configured to calculate an error signal from an input signal from a microphone and the estimated signal, and an error-signal polarity inverter configured to invert the polarity of the error signal in synchronization with the inversion of the polarity of the filter coefficient at intervals of the predetermined frame length. The adaptive filter updates the filter coefficient on the basis of the error signal by using binary fixed-point arithmetic, and therein, performing a truncation operation.

Abstract: Provided is a dereverberation system or the like which copes with an arbitrary condition flexibly and is capable of recognizing a sound or a sound source signal. According to the dereverberation system, an inverse filter (h) is set by using a pseudo-inverse matrix (R+) of a non-square matrix (R) as a correlation matrix of input signals (x). On the basis of the inverse filter (h) and an estimated correlation matrix (R?) generated according to a window function (w), an error cost (J(h) between a correlation value of the input signals (x) and output signals (y) and a desired correlation value (d) is calculated. On the basis of the error cost (J(h)), the inverse filter (h) is adaptively updated according to a gradient method.

Abstract: The invention concerns a method and device for processing echo between at least two communication devices in order to attenuate, in a picked-up signal of a communication device comprising at least one microphone, the components of the signal broadcasted by at least one other communication device comprising at least one loudspeaker, characterised in that the echo processing device comprises: means for receiving, by means of a connection with at least one other device, information representing at least one broadcasted signal of at least one other communication device, means for modifying the picked-up signal of the communication device according to information representing the broadcasted signal and information representing the coupling separating a loudspeaker of the said at least one other communication device from the microphone of the communication device.

Abstract: A method and system is provided for eliminating acoustical feedback in a system. The method determines a parameter for at least one notch filter, adjusting the notch filter based on the parameter, processing the digital signals through the notch filter, testing at the effect of the notch filter in the system, and removing the notch filter if the notch filter is not effective. Also disclosed is a method and system of selecting candidate frequencies which might be feedback, as opposed to other wanted sound frequencies. The selection method sampling the digital signals, converting the time domain digital signal samples by a fast Fourier transform algorithm into the frequency domain, using a ballistics approach to find prominences in the frequency spectrum, and testing the sizes of the prominences.

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 multiple channel steered spatialized signal is generated from a signal input modified according to respective spatialization gain functions to generate a plurality of audio channels. An echo cancellation signal is applied to a return path using a combined spatialization and echo path estimate. The estimate is derived from the gain functions applied to the respective channels. When the gain functions applied in the respective channels are changed, for instance to represent a different apparent position of the sound source, a new estimate of the echo paths is generated, based on a previous estimate of the echo path and on the new gain functions.

Abstract: A multichannel acoustic echo reduction system is described herein. The system includes an acoustic echo canceller (AEC) component having a fixed filter for each respective combination of loudspeaker and microphone signals and having an adaptive filter for each microphone signal. For each microphone signal, the AEC component modifies the microphone signal to reduce contributions from the outputs of the loudspeakers based at least in part on the respective adaptive filter associated with the microphone signal and the set of fixed filters associated with the respective microphone signal.

Abstract: In one embodiment, a two-way telecommunication device may perform acoustic echo cancellation on incoming signals. An audio decoding module may produce an audio render signal. An audio capture interface may receive an audio capture signal. A short length adaptive filter may determine a time delay between the audio render signal and the audio capture signal by adaptively predicting a sub-band of the audio capture signal using a corresponding sub-band of the audio render signal.

Abstract: To provide an audio conference apparatus and an audio conference system which can smoothly proceed with the audio conference by removing a recursion sound of the conference voice is achieved. An audio conference apparatus 1 outputs ring tones from corresponding channels before a communication control unit 12 outputs audio signals from the unused channels (S1 to S3). Speakers SP1 to SP16 emits the ring tone from predetermined sound source positions corresponding to the respective channels. Microphones MIC1A to MIC16A and microphones MIC1B to MIC16B collect audio signals including a recursion sound of the ring tone. The echo cancel unit 20 generates a pseudo-recursion sound signal on the basis of an input signal, and subtracts the pseudo-recursion sound signal from the collected audio signals. An audio conference system is configured to connect a plurality of the audio conference apparatuses to each other.

Abstract: A distributed audio signal processing system having a plurality of linked audio signal processing units is disclosed. Each audio signal processing unit has physical channels for receiving and sending local audio signals and a high bandwidth interface for exchanging audio signals with other linked audio signal processing units. Each of the physical channels of each of the audio signal processing units is mapped to a corresponding global channel. Global channels can be combined to form virtual channels that can be processed as a signal channel.

Abstract: An audio signal decorrelator for deriving an output audio signal from an input audio signal has a frequency analyzer for extracting from the input audio signal a first partial signal descriptive of an audio content in a first audio frequency range and a second partial signal descriptive of an audio content in a second audio frequency range having higher frequencies compared to the second audio frequency range. A partial signal modifier for modifies the first and second partial signals, to obtain first and second processed partial signals, so that a modulation amplitude of a time variant phase shift or time variant delay applied to the first partial signal is higher than that applied to the second partial signal, or for modifying only the first partial signal. A signal combiner combines the first and second processed partial signals, or combines the first processed partial signal and the second partial signal, to obtain an output audio signal.

Abstract: A method is provided for discriminating between the case of a single talker with an acoustic reflection and the case of two talkers, regardless of their power levels. The method is implemented in real time by performing a cross-correlation between pairs of average power signals originating from pairs of beamformers. A detection decision is then made based on the value of the cross correlation and its lag.

Abstract: A speech enhancement system controls the gain of an excitation signal to prevent uncontrolled gain adjustments. The system includes a first device that converts sound waves into operational signals. An ambient noise estimator is linked to the first device and an echo canceller. The ambient noise estimator estimates how loud a background noise would be near the first device before or after an echo cancellation. The system then compares the ambient noise estimate to a current ambient noise estimate near the first device to control a gain of an excitation signal.

Abstract: Techniques to reduce distortion in acoustic signals in mobile computing devices are described. For example, a mobile computing device may comprise a speaker operative to receive a first signal and output a second signal. The mobile computing device may further comprise a first microphone operative to receive the second signal and a second microphone operative to receive a third signal. An echo canceller may be coupled to the first microphone and the second microphone and may be operative to compare the second signal and the third signal and reduce distortion in the third signal based on the comparison. Other embodiments are described and claimed.

Abstract: An acoustic device is provided with first and second one or more acoustic elements to generate a first signal that includes mostly undesired audio and substantially void of desired audio, and a second signal that includes desired as well undesired audio respectively. The first one or more acoustic elements are designed and arranged to generate a Cardioid beam with a null at an originating direction of the desired audio. The second one or more acoustic elements are designed and arranged to generate a complementary beam that includes the desired audio. A system is provided with an appropriate signal processing logic to recover the desired audio using the first and second signals. The signal processing logic may practice echo cancellation like techniques or blind signal separation techniques.

Abstract: Techniques for multi-channel acoustic echo cancellation include adaptive filtering. An adaptive filter can use a lattice predictor of order M coupled to an adaptive LMS/Newton filter of length N, wherein M<N. The lattice predictor can provide decorrelation of the input to the LMS/Newton filter and can provide faster convergence for the LMS/Newton filter. Efficient operation of the LMS/Newton filter can also be provided by using output from the lattice predictor to provide low complexity update of weights for the LMS/Newton filter.

Abstract: In an audio system having a microphone, a speaker coupled to a source of audio output, and an echo canceller coupled to the speaker and microphone, latency between the source of audio output and the speaker may be compensated in echo cancellation performed by the echo canceller. The echo canceller may use a reference signal derived from a signal from the source of audio output in echo cancellation. The latency may be compensated by measuring the latency between the signal from the source of audio output and the speaker, determining a delay amount from the latency, delaying the reference signal by the delay amount to produce a delayed reference signal, and using the delayed reference signal as the reference signal in the echo canceller.

Abstract: A method for reducing echo includes detecting whether an output sound volume is greater than a threshold value, and setting an input sensitivity from a first designated sensitivity value to a second designated sensitivity value when the output sound volume is detected to be greater than the threshold value. The method further includes detecting whether an interrupt signal is received, determining whether the interrupt signal is triggered by detecting that the output sound volume is greater than the threshold value when receiving the interrupt signal, detecting whether the input sensitivity is the second designated sensitivity value when determining that the interrupt signal is triggered by detecting that the output sound volume is greater than the threshold value, and setting the input sensitivity as the second designated sensitivity value when detecting that the input sensitivity is not the second designated sensitivity value.

Abstract: A linear full duplex system and method for acoustic echo cancellation is disclosed. In one embodiment, a method includes calculating a residual echo after subtraction of an echo estimate from a near end signal associated with a communication, refining a far end and a residual signal associated with the communication, updating, based on the far end signal, the echo estimate of an echo associated with the communication, subtracting the echo from the near end signal based on the echo estimate to cancel the echo associated with the communication, updating, based on the refined far end and refined residual signal, the adaptive filter module used for echo estimation, and detecting a steady state and, during the steady state, dynamically detecting internal substates and switching between the internal substates and detecting and managing double talk associated with the communication.

Abstract: An echo canceler is constituted of an adaptive filter for generating an echo replica simulating an echo occurring in an echo path lying between a speaker and a microphone, a subtracter for subtracting the echo replica from the sound received by the microphone, a double-talk detector which extracts a prescribed frequency band, in which the signal level is attenuated due to the echo path, from the output signal of the subtracter and which determines a double-talk event with respect to the extracted frequency band, and a controller for controlling the adaptive filter to update the echo replica in the double-talk event.

Abstract: Systems and methods for envelope-based acoustic echo cancellation in a communication device are provided. In exemplary embodiments, a primary acoustic signal is received via a microphone of the communication device, and a far-end signal is received via a receiver. Frequency analysis is performed on the primary acoustic signal and the far-end acoustic signal to obtain frequency sub-bands. An echo gain mask based on magnitude envelopes of the primary and far-end acoustic signals for each frequency sub-band is generated. A noise gain mask based on at least the primary acoustic signal for each frequency sub-band may also be generated. A combination of the echo gain mask and noise gain mask may then be applied to the primary acoustic signal to generate a masked signal. The masked signal is then output.

Abstract: An echo canceler uses an adaptive filter to remove an echo of an incoming far-end signal from an outgoing near-end signal in the presence of ambient noise, updating the filter coefficients while the far-end signal is active and the near-end signal is inactive. The near-end and far-end signals are also sampled to obtain data vectors, which are averaged to generate simulated near-end and far-end signals, from which substitute filter coefficients are calculated and updated while the far-end signal is silent, the substitute filter coefficients being used when the far-end signal becomes active again. The number of data vectors averaged is varied according to the ratio of echo power to ambient near-end noise power, or according to the echo attenuation ratio, thereby speeding up convergence of the filter coefficients. Data sampled while the near-end and far-end signals are both active are excluded from the updating process.

Abstract: An audio processing device for reducing the effect on a first signal of echo from a second signal, the device comprising: an echo reduction processor for processing the first signal to reduce echo in it, the echo reduction unit having: a first mode of operation for reducing echo of a first function from the first signal; and a second mode of operation for reducing echo of a second function from the first signal, the second function being more complex than the first function and the echo reduction processor being such as to consume more power in the second mode of operation than in the first mode of operation; and an echo reduction controller for controlling the echo reduction processor to operate in a selected one of the first mode of operation and the second mode of operation.

Abstract: Harmonic distortion residual echo suppression (HDRES) technique embodiments are presented which act to suppress the residual echo remaining after a near-end microphone signal has undergone AEC, including harmonic distortion in the signal that was caused by the speaker audio signal playback. In general, an AEC module is employed which suppresses some parts of the speaker audio signal found in a near-end microphone signal and generates an AEC output signal. A HDRES module then inputs the AEC output signal and the speaker audio signal, and suppresses at least a portion of a residual part of the speaker audio signal that was left unsuppressed by the AEC module. This includes at least a portion of the harmonic distortion exhibited in the AEC output signal.

Abstract: A method is designed for emphasizing a howling frequency component of a sound signal observed in an acoustic feedback system during an observation period having a predetermined length.

Abstract: Sound signal reception is improved by utilizing a plurality of microphones to capture sound signals which are then weighed to dynamically adjust signal quality. A first sound signal and a second sound signal are obtained from first and second microphones, respectively, where the first and second sound signals originate from one or more sound sources. A first signal characteristic (e.g., signal power, signal signal-to-noise ratio, etc.) is obtained for the first sound signal and a second signal characteristic is obtained for the second sound signal. The first and second sound signals are weighed or scaled based on their respective first and second signal characteristics. The weighed first and second sound signals are then combined to obtain an output sound signal.

Abstract: In an embodiment, a method for removing an echo component is a first signal received at an audio input device includes outputting a second signal from an audio output device, receiving at the audio input device the first signal wherein the echo component in the first signal is the result of the second signal traversing an echo path; detecting if the first signal only comprises the echo component; determining an estimate indicative of the echo path by comparing the first signal and the second signal when it is detected that the first signal only comprises echo; applying the estimate indicative of the echo path to the first signal to determine an equalised first signal; comparing the equalised first signal with the second signal to determine an estimate indicative of the echo component; and removing the echo component from the first signal in dependence on the estimate indicative of the echo component; wherein the step of detecting if the first signal only comprises the echo component comprises comparing a char

Abstract: Different sampling rates between a playout unit and a capture unit are compensated for via a system, method and computer program product. The playout unit receives samples from a computational unit, and the capture unit sends samples to the computational unit. A playout FIFO buffer operates in a playout time domain, and a capture FIFO buffer operates in a capture time domain. The computational unit is synchronized to a common clock. A first relationship is calculated between the common clock and a playout fifo buffer read pointer, and a second relationship is calculated between the common clock and a capture FIFO buffer write pointer. For each sample in the playout time domain a corresponding sample in the samples from said computational unit is found and sent to the playout FIFO buffer. For each sample in the common clock time domain the corresponding sample in the capture time domain is found and sent to the computational unit.

Abstract: It is an object of the present invention to provide an echo suppressor which is simple in construction in comparison with the conventional echo suppressor, and which can suppress echoes corresponding to the spatial transfer paths, and reduce the amount of calculations necessary to suppress echoes. The echo suppressor is operative under the condition that transfer functions corresponding to spatial transfer paths between two or more loudspeakers (105, 106) and one or more microphones (107, 108) are estimated on the basis of symmetrical arrangement of the loudspeakers (105, 106) and the microphones (107, 108).

Abstract: A system and method for providing microphonic isolation on a transmission line. The transmission line has a first part and a second part. The first part of transmission line carries a data signal and a microphonic signal. The microphonic signal has frequencies that include those in a range of substantially 20 Hz to substantially 20 kHz. The system includes an isolation apparatus. The isolation apparatus has an input in electrical communication with a first part of the transmission line, an output in electrical communication with the second part of the transmission line, and a filter in electrical communication with the input and the output. The filter is arranged to substantially remove the microphonic signal received at the input from first part of transmission line and pass the data signal to the output.

Abstract: A voice processing apparatus includes a band dividing portion dividing a first voice signal generated by a first microphone and a second voice signal generated by a second microphone into predetermined frequency bands, a sound source segregating portion segregating an echo component of a voice emitted by a first sound source included in a voice emitted by a second sound source in each of the predetermined frequency bands based on the power of the first and second microphones, and a band synthesis portion synthesizing the first and second voice signals from which the echo component of the first sound source has been segregated by the sound source segregating portion into a voice signal including the voice emitted by the first sound source and a voice signal including the echo component of the first sound source.

Abstract: Converter 100 uses either the output signal of a sound pickup device or the signal obtained by subtracting the output signal of an echo canceller from the output signal of the sound pickup device as a first signal, uses an estimated crosstalk value indicative of an estimated value of the amount of crosstalk of an echo leaking into the first signal to correct the first signal, and limits the corrected first signal not to be smaller than estimated near-end noise.

Abstract: A method and apparatus for removing an echo signal in a signal transmission/reception apparatus of a communication system are provided. A signal transmission/reception apparatus determines an echo channel impulse response using a reception signal, generates an echo signal removing coefficient using the echo channel impulse response, removes an echo signal from the reception signal using the echo signal removing coefficient, and transmits a signal in which the echo signal is removed.

Abstract: Far-end audio signal data generated by a far-end device during voice cross-talk between the far-end device and a near-end device is received via a network. Energy levels of frequency components of the received far-end audio signal data are determined. Near-end audio signal data produced by a microphone array is received, the near-end audio signal data including components contributed by a loudspeaker playing at least some of the far-end audio signal data. Which frequency components of the near-end audio signal data to use to compute likely near-end sound source directions is controlled dynamically, where the controlling is based on the determined energy levels of the frequency components of the received far-end audio signal data.

Abstract: Frames containing audio data may be received, the audio data having been derived from a microphone array, at least some of the frames containing residual acoustic echo after having acoustic echo partially removed therefrom. Probability distribution functions are determined from the frames of audio data. A probability distribution function comprises likelihoods that respective directions are directions of sources of sounds. An active speaker may be identified in frames of video data based on the video data and based on audio information derived from the audio data, where use of the audio information as a basis for identifying the active speaker is controlled by determining whether the probability distribution functions indicate that corresponding audio data includes residual acoustic echo.

Abstract: An input sound processor compares power at each frequency component of an input sound with a reference value, and sets multiplication points indicating frequency components at which the total power of the input sound is to be determined. A product-sum operation is performed at the multiplication points on the power at each frequency component and the square amplitude of each filter coefficient indicating the transfer characteristic from a loudspeaker to a microphone to estimate the total power of the input sound at the position of the microphone.

Abstract: Disclosed herein is an echo cancellation and echo channel estimation system that can relay transmit signals without interference with echo signals by canceling the undesired echo signals received by a relay. The echo cancellation and echo channel estimation system is designed to relay signals between a transmitter and a receiver through a relay using multiple antennas. The echo cancellation and echo channel estimation system includes a receive antenna, a preprocessing vector generation module, an echo cancellation module, and a transmit array antenna. The receive antenna receives a transmit signal from the transmitter and an echo signal via an echo channel. The preprocessing vector generation module generates a preprocessing vector and applies the preprocessing vector, the transmit signal and the echo signal, received from the receive antenna, to the echo cancellation module.

Abstract: A system and method for Acoustic Echo Cancellation. The system and method include a subband affine projection filter and a variable step size controller configured to cancel an estimated echo from a near-end signal. The system and method also include a divergence detector adapted to reset the subband affine projection filter in response to determining a divergence is occurring. Additionally, the system and method include a double talk detector adapted to transmit a signal to mask an output signal when double talk is detected.

Abstract: A method is provided for estimating a reverberation signal component of an acoustic signal detected by a microphone where the acoustic signal is comprised of a direct sound component and a reverberation signal component. A method for dereverberation of an acoustic signal is further provided.

Abstract: Speech dereverberation is achieved by accepting an observed signal for initialization (1000) and performing likelihood maximization (2000) which includes Fourier Transforms (4000).

Abstract: An echo canceller includes a residual signal generation unit, a double talk detection unit, a nonlinear processor, a speech detection unit, and an input/output characteristic change unit. The residual signal generation unit generates a pseudo echo signal, and generates a residual signal by using the pseudo echo signal. The double talk detection unit detects the state of the transmission signal. The nonlinear processor attenuates the residual signal that has been inputted thereto to a signal level which is based on a predetermined input/output characteristic, and that outputs the attenuated residual signal. The speech detection unit detects whether or not speech is included in the reception signal. The input/output characteristic change unit changes the input/output characteristic of the nonlinear processor to a predetermined input/output characteristic when a single talk state has been detected at the double talk detection unit and speech has been detected at the speech detection unit.

Abstract: A signal processing system enhances an audio signal. The audio signal is divided into audio sub-band signals. Some audio sub-band signals are excised. Other audio sub-band signals are processed to obtain enhanced audio sub-band signals. At least a portion of the excised audio sub-band signals are reconstructed. The reconstructed audio sub-band signals are synthesized with the enhanced audio sub-band signals to form an enhanced audio 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: The present invention provides an echo control retrofit apparatus and method that advantageously reduces echo when communicating over packet-switched networks. An echo control retrofit apparatus, including an echo control circuit, is operably coupled between a headset or handset device and an audio source. The echo control circuit receives a sound signal from the audio source and a transmit signal from the headset or handset device and provides an adjusted sound signal to the audio source. Advantageously, a variety of existing headsets and handsets may be used in accordance with the present invention to provide reduced caller echo without the need to purchase new headsets or handsets.

Abstract: A device for reducing echo by controlling microphone and/speaker gain in a telecommunications system is disclosed. An exemplary embodiment of the device may have an echo detector, echo level detector, and a gain controller. The echo detector detects an echo in a transmitted audio communication signal. The echo level detector determines the volume level of the detected echo. The gain control may adjust the gain of a microphone transmitting and/or speaker broadcasting the audio communication signal based on the echo level detected. The gain control and echo cancellation may cooperate with each other.

Abstract: Implementations related to echo cancellation are depicted and described herein.

Abstract: Microphones arranged in an array shape along a longitudinal direction are respectively formed in both the longitudinal side surfaces of a housing 2 with substantially an elongated rectangular parallelepiped shape, and speakers arranged in an array shape along the longitudinal direction are formed in a lower surface. The speaker array forms sound emission beams based on sound emission directivity set according to a conference environment. On the other hand, when the microphone array forms sound collection beams by sound collection signals collected, a talker direction is detected from these beams and an output sound signal corresponding to this direction is formed and also is reflected on setting of the sound emission directivity. Also, when there are plural input sound signals, the sound emission directivity is set according to a use situation of the plural input sound signals.

Abstract: An apparatus capable of performing acoustic echo cancellation and a method thereof are provided. The apparatus comprises a mapping matrix, first and second speakers, first and second microphones, a reference generator, and a multi-channel acoustic echo canceller. The mapping matrix generates an output signal according to the first and second far end signals. The first and second speakers, coupled to the mapping matrix, play the output signal. The first and second microphones receive the first and second echo signals that are acoustically coupled from the first and second speakers to the first and second microphones, wherein the first and second echo signals are correlated to the output signal. The reference generator generates a reference signal linearly correlated to the output signal according to the first and second far end signals.

Abstract: An echo canceller circuit is set forth. The echo canceller circuit includes a digital filter having adaptive tap coefficients to simulate an echo response occurring during a call. The adaptive tap coefficients are updated during the call using a Means Squares process. A tap energy detector is also employed. The tap energy detector identifies and divides groups of taps having high energy from groups of taps having low energy. The high energy tap groups are smaller in number than the low energy tap groups. The high energy tap groups are adapted separately from the low energy tap groups using the Least Squares process. Still further, the high energy tap groups may be adapted using an adaptive gain constant a while the low energy tap groups are adapted using an adaptive gain constant a?, wherein a>a?.