94results about How to "Cancel echo" patented technology

Systems and methods for receiving natural language queries and / or commands and execute the queries and / or commands. The systems and methods overcomes the deficiencies of prior art speech query and response systems through the application of a complete speech-based information query, retrieval, presentation and command environment. This environment makes significant use of context, prior information, domain knowledge, and user specific profile data to achieve a natural environment for one or more users making queries or commands in multiple domains. Through this integrated approach, a complete speech-based natural language query and response environment can be created. The systems and methods creates, stores and uses extensive personal profile information for each user, thereby improving the reliability of determining the context and presenting the expected results for a particular question or command.

Some embodiments provide an architecture for establishing multi-participant audio conferences over a computer network. This architecture has a central distributor that receives audio signals from one or more participants. The central distributor mixes the received signals and transmits them back to participants. In some embodiments, the central distributor eliminates echo by removing each participant's audio signal from the mixed signal that the central distributor sends to the particular participant.

This invention provides one double microwave sound strength method and device suitable for small mobile communication device to process the input signal x1 and x2 and to adopt wave beam forming technique and use aim sound signal source and noise signal source difference to isolate signals to get the sound signal S(k) and noise signal n(k); using two paths of signals relationship to remove noise part and sound point to get síõ(k) and níõ(k).

The invention provides an adaptive echo eliminator and the echo eliminating method thereof; wherein the device consists essentially of a voice state detector, an NLMS (energy normalized least mean square error) controller and a sliding window FIR filter. The method mainly includes the following steps: sampling both a far-end voice signal and a near-end voice signal, and confirming the speaking status information of the current network according to the estimated values of the short-time energy of the sampled far-end speech signal sample and the near-end speech signal sample. Then, the coefficient of a sliding window FIR filter is configured according to the speaking status information of the current network, and the sliding window FIR filter step-by-step filters the near-end speech signal and the far-end speech signal filled into the buffer area according to the mobile length set by the configured coefficient. Utilizing the invention, the echo of the digital hands-free speaking system can be effectively eliminated, thereby effectively eliminating the echo of the digital communication system based on the embedded system.

A wireless repeater includes an internal feedback path for adaptively cancelling an echo between an output antenna and an input antenna. The internal feedback path employs an adaptive algorithm implementing a list having a plurality of list elements. Each list element has one or more echo cancellation parameters and one or more repeater settings. The list may be pruned by employing a minimum distance between elements within the list. A method for stabilizing a wireless repeater includes obtaining an autocorrelation of a signal in a signal path of the repeater, detecting an echo in the signal path based on the autocorrelation, providing a list of elements indicating past successful echo cancellation coefficients and associated repeater settings, and adaptively adjusting an error of the echo cancellation coefficients to cancel the echo in the signal path.

The invention relates to a self-adapting volume control method, a device and a communication terminal belonging to the communication technology field. The invention discloses the self-adapting volume control method comprising: receiving an audio signal from a sound source; acquiring sound source corresponding to the audio signal and an input, output gain parameters corresponding to the sound information; controlling an input, output volumes according to the input, output gain parameters. The invention also discloses the self-adapting volume control device comprising a receiving unit, an acquisition unit and a control unit. The invention also discloses the communication terminal comprising said device. The advantage of the invention is to automatically control the input, output volume and enable the user to obtain satisfactory listening effect.

A communication apparatus capable of echo cancellation is provided. The apparatus is configured to be provided with an input signal receiving interference. The input signal has a frequency range. The interference includes an echo component and a noise component. The apparatus has a first interference eliminator configured to reduce an echo component included in the input signal so as to produce an intermediate signal. The apparatus has a selector configured to select one of the input signal and the intermediate signal. The apparatus has a second interference eliminator configured to reduce at least one of a noise component and an echo component included in one of the input signal and the intermediate signal selected by the selector.

Some embodiments provide an architecture for establishing multi-participant audio conferences over a computer network. This architecture has a central distributor that receives audio signals from one or more participants. The central distributor mixes the received signals and transmits them back to participants. In some embodiments, the central distributor eliminates echo by removing each participant's audio signal from the mixed signal that the central distributor sends to the particular participant.

The invention discloses a time unifying method and a time unifying device for audio data and a reference signal and belongs to the communication field. The method comprises the steps as follows: obtaining the audio data which is collected currently and obtaining the reference signal corresponding to the audio data from a reference queue; calculating a first time delay estimation expected value according to a correlation value between the audio data and the reference signal; detecting whether there is beating effect between the audio data and the reference signal in the reference queue according to a second time delay estimation expected value and the first time delay estimation expected value, wherein the second time delay estimation expected value is a time delay estimation expected value which is obtained by last calculation before current time; unifying a time relation between the audio data and the reference signal in the reference queue if the beating effect is existed. The device comprises an obtaining module, a calculating module, a detecting module and a unifying module. The time unifying method and time unifying device of the invention improve robustness of echo cancellation.

The invention discloses a frequency-convertible echo cancellation method based on near-end audio signal calibration and correction and relates to the technical field of audio and video processing. The frequency-convertible echo cancellation method based on the near-end audio signal calibration and correction is a near-end audio signal calibration and correction method and includes the first step of carrying out one-time calibration on the frequency conversion of a microphone of equipment and the frequency conversion of a loudspeaker of the equipment to generate a set of frequency correction parameters, and the second step of firstly carrying out one-time correction on collected audio signals through the utilization of the frequency correction parameters calibrated before, then allowing estimation echo audio signals generated from a self-adaptive filter module to be subtracted from the corrected audio signals and finally achieving the echo cancellation function. The frequency-convertible echo cancellation method based on the near-end audio signal calibration and correction is capable of well canceling echoes after the frequency conversion, simplifying the calculated amount of a self-adaptive filter, enabling the calculated amount of the self-adaptive filter to have better convergence, being suitable for processing the long echoes and reducing noises and constraint.

The invention discloses a method for eliminating echo. The method comprises the steps of acquiring a mixed signal mixed by a user sound given out by a user and echo given out by a loudspeaker, acquiring a sound signal given out by the loudspeaker according to the volume of the sound given out by the loudspeaker, and using the sound signal as a reference signal; comparing the mixed signal with the reference signal, so as to obtain a gain coefficient of the reference signal, and acquiring a gain signal of the reference signal according to the gain coefficient; acquiring a compensation signal of the reference signal according to the gain signal and a corresponding preset compensation coefficient; reversing phase of the compensation signal; and combining the compensation signal with the phase reversed with the mixed signal, so as to eliminate the echo given out by the loudspeaker. The invention also discloses a system for eliminating echo. Through adoption of the method and system for eliminating echo, the effect of eliminating echo in the mixed signal can be improved.

The invention provides an echo signal offset method and system as well as a television, belonging to the technical field of electroacoustics. The method comprises the following steps: acquiring variation information of time delay, phase shift and amplitude of an echo signal of a test signal; correcting coefficients of a filter based on the variation information; estimating an echo signal generated by a far-end voice signal based on the corrected coefficients of the filter; and eliminating the estimated echo signal from received local voice signals, wherein the local voice signals comprise the voice signal of a local user and the echo signal locally practically generated by the far-end voice signal. Based on full consideration of influence of environment factors on echo signals, echo can be eliminated effectively.

The invention discloses a network voice intercom method, device and system. The method comprises the following steps: data synchronization is performed on an audio stream recorded locally and an audio stream sent by the opposite end; the echo in the audio stream recorded locally is filtered by taking the audio stream sent by the opposite end as the reference audio stream; and the filtered audio stream is encoded and compressed and sent through a network. The method, device and system of the embodiments of the invention are used to perform data synchronization on the audio stream recorded locally and the audio stream sent by the opposite end and take the audio stream sent by the opposite end as the reference audio for filtering the echo, the encoding and decoding can be optimized, the echo can be eliminated, and the network voice intercom call quality can be improved.

This invention relates to a method for eliminating acoustic echo including the following steps: the minimum mean square algorithm of adaptive step wave band standardization, eliminating wave beams of noises to form non-linear filter to eliminate acoustic echoes. This invention applies an exclusive adaptive step non-linear filter technology to divide signals into multiple wave bands for process and alter steps adaptively according to the echo environment to eliminate echoes to increase convergence speed and reduce radiation speed to improve the ability for eliminating echo.

An acoustic echo control system and a double talk control method thereof that eliminate echo signals which are generated in a hand-free communication system. In order to discriminate a double talk generation and a change of an echo path, the acoustic echo control system accurately detects a double talk section and an echo path change section using a near end signal from a near end talker and a far end signal from a far end talker and in the double talk section an adaptive echo remover of the system suspends a filter coefficient adaption and estimates an acoustic echo path which is supplied to a microphone over a speaker by performing the filter coefficient adaption in the echo path change section, thereby improving communication quality and discriminating the double talk and the change of the echo path without time delay.

An embodiment of the invention discloses an echo eliminating method, a server, a terminal and a system, wherein the echo eliminating method is applied to a server cluster and comprises the steps of supplying a detecting signal to one or a plurality of terminals; receiving an acquiring signal which is transmitted by one or the plurality of terminals, wherein the acquiring signal is an audio signalwhich is acquired by one or the plurality of terminals in playing the detecting signal by one or the plurality of terminals; and determining an echo characteristic parameter which corresponds with oneor the plurality of terminals based on the acquiring signal and the detecting signal, wherein the echo characteristic parameter is used for echo elimination of the terminal. The echo eliminating method, the server, the terminal and the system can continuously correct and restore the instable filter state, thereby effectively eliminating the echo.

The invention discloses artificial intelligence simultaneous interpretation equipment for on-site meetings. A microphone array is capable of improving the pickup capacity in each direction, so that the voices in each direction of a meeting desk can be recognized by equipment; an echo elimination module is capable of eliminating echoes in the voices, and a noise eliminating module is capable of eliminating various noises in the received voices, so that the intelligibility of the received voices can be improved; a voiceprint recognition module is capable of recognizing and marking a speaker through voiceprints and indicating the feathers of gender and age of the speaker; an AS voice recognition module is capable of translating the voices of the user into words; an automatic language judgingmodule is capable of automatically judging the languages of the speaker, so that the voices of each people can be translated into words required by each language in a meeting room; a neural network translating module is capable of cooperating with a translation library, giving out words of translation results corresponding to the national languages, and the TTS broadcasting module is capable of generating the voices of people in the corresponding countries according to the words and playing the voices, and the display screen is capable of displaying finally translated word results.

A wireless repeater includes an internal feedback path for adaptively cancelling an echo between an output antenna and an input antenna. The internal feedback path employs an adaptive algorithm implementing a list having a plurality of list elements. Each list element has one or more echo cancellation parameters and one or more repeater settings. The list may be pruned by employing a minimum distance between elements within the list. A method for stabilizing a wireless repeater includes obtaining an autocorrelation of a signal in a signal path of the repeater, detecting an echo in the signal path based on the autocorrelation, providing a list of elements indicating past successful echo cancellation coefficients and associated repeater settings, and adaptively adjusting an error of the echo cancellation coefficients to cancel the echo in the signal path.

The invention discloses an echo cancellation method and device. The method comprises steps of a near-field voice signal and a far-field voice signal; acquiring a frequency domain echo time delay and time domain convolution factors in echo cancellation of the last near-field voice signal; according to the time domain convolution factors, the frequency domain echo time delay and a far field voice signal, carrying out echo cancellation processing on the near-field voice signal in the time domain so as to obtain echo cancellation signals of the time domain and based on the echo cancellation signal, updating time domain convolution factors; acquiring the frequency domain signal of the echo cancellation signal and the frequency domain signal of the far-field voice signal, and according to the frequency domain signal of the echo cancellation signal and the frequency domain signal of the far-field voice signal, acquiring the frequency domain echo time delay and inhibition factors; and according to the inhibition factors, carrying out echo cancellation processing on the echo cancellation signals in the frequency domain so as to obtain signals subjected to echo cancellation. According to theinvention, by use of the method of combining the time domain and the frequency domain, it is ensured that echo can be well cancelled under the condition of quite less voice distortion.

Embodiments of the invention provide an echo eliminating method, device and system. The method comprises steps of acquiring near-end input signals and far-end input signals; filtering the near-end input signals corresponding to all sub-bands according to the far-end input signals corresponding to all sub-bands, and generating residual signals corresponding to all sub-bands; combining the residualsignals corresponding to all sub-bands, performing frequency-to-time conversion on the combined residual signal, so as to acquire a near-end input signal with echo eliminated, wherein when the near-end input signals and the far-end input signals are processed, association of the near-end input signals and the far-end input signals corresponding to all sub-bands is detected according to a set period, when it is determined that sequence numbers of sub-bands with highest association of the near-end input signals and far-end input signals in all the sub-bands are less than K in N consecutive periods, near-end sub-band signals corresponding to sub-band K to sub-band M-1 are stopped filtering, wherein K is less than or equal to M-1.

System and method for blind echo cancellation in a received terahertz signal in a pulsed terahertz system for imaging or spectroscopy. Blind signal processing methods estimate the impulse response of the reflection mechanism and do not require a reference measurement to be taken. The reference signal may be recovered using a successive approach wherein the reference is first estimated using cross-correlation with the received signal and the received signal is represented as a function of the reference signal. For each successive echo, the calculated echo may be subtracted from the received signal and then the estimate of the reference signal is refined. Using an analytical approach, the parameters of a transfer function modeling the reflection mechanism may be estimated by optimizing a cost function.

The invention relates to a time delay estimation method and device and electronic equipment. The time delay estimation method comprises the steps that an acoustical signal collected by a microphone and a far-end voice signal output by a loudspeaker are acquired; the acoustical signal and the far-end voice signal are taken as fingerprint input signals respectively, and dynamic change feature extraction of audio energy is carried out on the fingerprint input signals to obtain an audio fingerprint of the acoustical signal and an audio fingerprint of the far-end voice signal; fingerprint comparison is carried out on the audio fingerprint of the acoustical signal and the audio fingerprint of the far-end voice signal to obtain time delay estimation results. The time delay estimation method and device and the electronic equipment have the advantages that the response speed of the time delay estimation can be increased, and the problem that echo leakage caused by the fact that the echo cancellation cannot be performed normally during time delay change is avoided.

An application sound suppression method includes acquiring application sound data; acquiring ambient sound data collected by the microphone; performing an echo cancellation process on the ambient sound data according to the echo cancellation algorithm and the applied sound data. The application of sound data and its echoes is highly correlated, As a result, echo cancellation processing is performed on ambient sound data using an echo cancellation algorithm and sound data, The method can eliminate the echo of the applied sound data in the ambient sound data, has remarkable effect on the suppression of the applied sound, obviously improves the speech signal-to-noise ratio, can receive the pure speech signal at the receiving end, and improves the user experience. The present application alsodiscloses an application sound suppression apparatus, an apparatus and a storage medium.

A method (200) of cancelling echo in a duplex communication device (100). The method can include detecting a level of noise present on an uplink signal path (104), generating a noise classifier (194) based on the detected level of noise, detecting whether uplink audio is present on the uplink signal path (104) and detecting whether downlink audio is present on a downlink signal path (102). The method further can include generating a double talk flag (136) based at least on the noise classifier, whether uplink audio is present on the uplink signal path, and whether downlink audio is present on the downlink signal path. In addition, the double talk flag, the noise classifier and an uplink signal can be processed to generate an output signal (120) having reduced echo.

The invention discloses an echo cancellation method and device, electronic equipment and a readable storage medium. The method comprises the following steps: acquiring a first audio signal received byat least one loudspeaker and a second audio signal received by a microphone; performing delay compensation processing on the first audio signal to obtain a third audio signal; acquiring an amplitudevalue of a transfer function of transferring the audio signal from the loudspeaker to the microphone; by utilizing the third audio signal and the amplitude value of the transfer function, determiningan echo signal generated by the first audio signal in the second audio signal; and cancelling the echo signal from the second audio signal.

A multi-receiving terminal echo cancellation method and a multi-receiving terminal echo cancellation system are disclosed. The present invention performs echo filtering on receiving terminal signals in M channels by means of echo filters to obtain filtered receiving terminal signals in M channels, and subtracts the filtered receiving terminal signals in M channels from a sending terminal signal to obtain a system output signal in which receiving terminal echoes have been cancelled; and at the same time, the present invention buffers the receiving terminal signals in M channels by means of buffers, calculates a decorrelation matrix according to each of the receiving terminal signals in M channels that are buffered within preset length, decomposes the buffered receiving terminal signals in M channels into decorrelated receiving terminal signals in M channels by means of the decorrelation matrix, and calculates update amounts of the echo filters according to the decorrelation matrix, the decorrelated receiving terminal signals in M channels and the system output signal that is fed back. The technical solutions of the present invention can support the situation of two or more receiving terminals, and are suitable for situations in which the correlation between a plurality of receiving terminal signals is variable.

A network communications device is capable of promoting connection quality, and includes: a plurality of transmitting units for outputting a plurality of analog transmit-signals to another network communications device based on a plurality of digital transmit-signals; a plurality of receiving units for outputting a plurality of digital receive-signals based on a plurality of analog receive-signals sent from the another network communications device; an echo canceller for providing a signal for canceling an echo in one of the digital receive-signals; a near end crosstalk canceller for providing a signal for canceling near end crosstalk in one of the digital receive-signals; a decoding circuit for generating a decoded signal based on one of the digital receive-signals subsequent to cancellation of the echo and the near end crosstalk therein; and a power-increasing control circuit for increasing operating power so as to promote connection quality.

The invention discloses a method and a system for reducing noise based on frequency characteristics. The method comprises the following steps: S10: acquiring a sound signal; S20: filtering the sound signal, so as to obtain a filtered signal, and decomposing the filtered signal into sound frequency domain signals; S30: acquiring a sound frequency domain which is required by a user; S40: implementing enhancement processing on a sound frequency domain signal which is required by the user, and implementing weakening processing on signals of other frequency domains, so that a playing frequency domain signal is obtained; and S50: reducing the playing frequency domain signal and playing sound. The method provided by the invention, which reduces the noise on the basis of the natural properties of the sound, is easy to implement and broad in application scope; and a voice scheme can be freely selected, so that voice delivery becomes more intuitive and valid, user perception is clearer and voice is easier to recognize.