874results about "Two-way loud-speaking telephone systems" patented technology

Techniques for canceling echo and suppressing noise using an array microphone and signal processing. In one system, at least two microphones form an array microphone and provide at least two microphone input signals. Each input signal may be processed by an echo canceller unit to provide a corresponding intermediate signal having some echo removed. An echo cancellation control unit receives the intermediate signals and derives a first gain used for echo cancellation. A noise suppression control unit provides at least one control signal used for noise suppression based on background noise detected in the intermediate signals. An echo cancellation and noise suppression unit derives a second gain based on the control signal(s), cancels echo in a designated intermediate signal based on the first gain, and suppresses noise in this intermediate signal based on the second gain. The signal processing may be performed in the frequency domain.

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.

A processor operates on samples of a digital output signal to determine samples of a digital correction signal. The output signal samples are directed to an output channel for transmission from a speaker. The digital correction signal samples are supplied to a first digital-to-analog converter for conversion into an analog correction signal. The subtraction circuit generates a difference between a first analog signal provided by a microphone and the analog correction signal. The analog correction signal is an estimate of a contribution to the first analog signal due to a direct path transmission between the speaker and the microphone. The processor also receives a digital input signal derived from the difference signal, and, performs acoustic echo cancellation on the digital input signal to obtain a resultant signal. The acoustic echo cancellation is configured to remove contributions to the digital input signal due to reflected path transmissions.

The present invention provides a solution to the needs described above through a method and apparatus for reducing echo and noise. The apparatus includes a microphone array for receiving and audio signal, the audio signal including a voice signal component and a noise signal component. The apparatus further includes a voice processing path having an input coupled to the microphone array and a noise processing path having an input coupled to the microphone array. The voice processing path is adapted to detect voice signals and the noise processing path is adapted to detect noise signals. A first echo controller is coupled to the voice processing path and a second echo controller is coupled to the noise processing path. A noise reducer is coupled to the output of the first echo controller and second echo controller.

An adaptive noise canceling microphone system for extracting a desired signal, in particular a desired speech signal, comprising two microphones being arranged at a predefined distance from each other; a signal forming system (SFS) being adapted to receive a first and second input signals resulting from sounds received by the two microphones wherein an acoustical signal component in the first input signal is determined, wherein an acoustical signal component in the second input signal is determined, wherein the acoustical signal component in the first input signal is enhanced to generate a speech enhanced signal, and wherein the acoustical signal component in the second input signal is suppressed to generate a speech nulled signal; an adaptive noise cancellation filtering circuit being adapted to receive the speech enhanced signal and the speech nulled signal, wherein the noise in the speech enhanced signal is cancelled using the speech nulled signal as reference, thereby generating an output filtered signal representing the desired signal.

A two-stage structure for performing non-linear echo cancellation is described in which a first echo canceller is used to attenuate linear echo components of a microphone signal and a second echo canceller is used to attenuate non-linear echo components of the output signal generated by the first echo canceller. One or both of the echo cancellers may be implemented using closed-form solutions, including a closed form solution for a hybrid method in the frequency domain.

A desired acoustic signal is extracted from a noisy environment by generating a signal representative of the desired signal with processor (30). Processor (30) receives aural signals from two sensors (22, 24) each at a different location. The two inputs to processor (30) are converted from analog to digital format and then submitted to a discrete Fourier transform process to generate discrete spectral signal representations. The spectral signals are delayed to provide a number of intermediate signals, each corresponding to a different spatial location relative to the two sensors. Locations of the noise source and the desired source, and the spectral content of the desired signal are determined from the intermediate signal corresponding to the noise source locations. Inverse transformation of the selected intermediate signal followed by digital to analog conversion provides an output signal representative of the desired signal with output device (90). Techniques to localize multiple acoustic sources are also disclosed. Further, a technique to enhance noise reduction from multiple sources based on two-sensor reception is described.

A protective acoustic cover assembly including a metal foil with perforations, and a treatment on one or more surfaces of said metal foil. The treatment is a hydrophobic or oleophobic treatment, or both. The protective acoustic cover assembly has an average specific acoustic resistance of less than about 11 Rayls MKS from 250-300 Hz, an average specific acoustic reactance magnitude of less than about 1 Rayls MKS from 250-300 Hz, and an instantaneous water entry pressure value of greater than about 11 cm. The perforations of the metal foil preferably have an average maximum pore size of less than about 150 micrometers. The protective acoustic cover assembly further includes an adhesive mounting system, and the preferred metal foil is nickel.

A speech processing apparatus which, in the process of performing echo canceling by using a pseudo acoustic echo signal, continuously uses an impulse response used for the previous frame as an impulse response to generate the pseudo acoustic echo signal when a voice is contained in the microphone input signal, and which uses a newly updated impulse response when a voice is not contained in the microphone input signal.

An apparatus for enhancing audibility of a far-end speech signal from a far-end user to a near-end user in a telephony system includes a near-end background noise signal level estimator (26) at the near-end user. A near-end speech signal level estimator (24) is also provided at the near-end user. A gain control logic (22) determines a gain (G) for amplification of the far-end speech signal based on both estimated speech and background noise signal levels.

A modem monitor for monitoring full duplex communications comprises one receiver for each data stream and an echo canceler for canceling "far" echo. Prior to each receiver decoding the data stream it is designed to monitor, it replicates an echo expected to be found in its received data stream from a sample of each of the other predecoded data streams. The echo canceler then subtracts the expected echo from the received data stream to remove all far echoes caused by the other data streams. The echo canceled data stream is then decoded. The monitor can also include a data communications network interface and a sample rate converter operating with the PSTN for converting from the network transmission rate to the data transmission rate.

A signal processing system detects reverberation. The system may suppress the reverberation and improve signal quality. The system analyzes frequency bands of an input signal to determine whether reverberation characteristics are present. When reverberation is detected, the system may attenuate the reverberant frequency band to reduce or eliminate the reverberation.

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.

Systems and methods are disclosed for packet voice conferencing. An encoding system accepts two sound field signals, representing the same sound field sampled at two spatially-separated points. The relative delay between the two sound field signals is detected over a given time interval. The sound field signals are combined and then encoded as a single audio signal, e.g., by a method suitable for monophonic VoIP. The encoded audio payload and the relative delay are placed in one or more packets and sent to a decoding device via the packet network. The decoding device uses the relative delay to drive a playout splitter—once the encoded audio payload has been decoded, the playout splitter creates multiple presentation channels by inserting the transmitted relative delay in the decoded signal for one (or more) of the presentation channels. The listener thus perceives a speaker's voice as originating from a location related to the speaker's physical position at the other end of the conference. An advantage of these embodiments is that a pseudo-stereo conference can be conducted with virtually the same bandwidth as a monophonic conference.

A telecommunications system includes a network; a plurality of user devices operably coupled to said network and including a microphone and a speaker; a collaboration server configured to cause a playing of one or more test tones at said plurality of user devices and compensate for feedback if said test tones are detected by microphones associated with said plurality of user devices.

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).

Hands-free phones with voice activity detection using a comparison of frame power estimate with an adaptive frame noise power estimate, automatic gain control with fast adaptation and minimal speech distortion, echo cancellation updated in the frequency domain with stepsize optimization and smoothed spectral whitening, and echo suppression with adaptive talking-state transitions.

Systems and methods that enable high quality audio teleconferencing are disclosed. In one embodiment of the present invention, a signal processor receives signals from a spatially dispersed set of directional microphones, processing the microphone signals and the far-end received audio into a signal for transmission to a far-end party. The processing may comprise the use of one or more algorithms that reduce conference room noise and may selectively increase participant audio levels by processing the microphone signals using beamforming techniques. An embodiment of the present invention may also comprise one or more omni-directional microphones that may be used in cooperation with the directional microphones to adjust for background noise, acoustic echo, and the existence of side conversations.

A mobile communications device contains at least two microphones. One microphone is designated by a selector to provide a voice dominant signal and another microphone is designated to provide a noise or echo dominant signal, for a call or a recording. The selector communicates the designations to a switch that routes the selected microphone signals to the inputs of a processor for voice signal enhancement. The selected voice dominant signal is then enhanced by suppressing ambient noise or canceling echo therein, based on the selected noise or echo dominant signal. The designation of microphones may change at any instant during the call or recording depending on various factors, e.g. based on the quality of the microphone signals. Other embodiments are also described.

A case for a portable electronic device such as cell phones, cameras, MP3 players and PDAs wherein said case includes material at the corners to provide protection to the electronic device. To reduce bulk at the corners and allow the material forming the case to conform with the contours of the device, the protection at the corners includes strips of material defining apertures there between. The apertures provide reduction in bulk and allow the case to conform to the corners of the device therein. Alternately, the protection at the corners can be provided by other structures co-molded into the corners of the cases. Such structures include material that is of a reduced thickness than other material used in the case, or structures that are formed to conform to the corners of the case. Such structures can be joined to the material forming the panels of the case by co-molding.

PCT No. PCT/GB95/02618 Sec. 371 Date Jul. 2, 1997 Sec. 102(e) Date Jul. 2, 1997 PCT Filed Nov. 7, 1995 PCT Pub. No. WO96/15597 PCT Pub. Date May 23, 1996An echo detecting system includes data stores for storing signals from up and down channels respectively. The signals undergo pre-processing to identify signals forms characteristic of speech and instruct a measurement unit to carry out comparison using cross-correlation techniques between the signals stored in the stores only when such characteristics are detected. This reduces the processing power required and raises the accuracy of the correlations. Parallel processing techniques allow echoes with longer delay periods to be detected. The results of the measurement may be used to generate an echo-cancellation signal.

The present invention is directed to a telephone equipped with multiple microphones that provides improved performance during operation of the telephone in a speaker-phone mode. For example, the multiple microphones can be used to improve voice activity detection, which in turn, can improve echo cancellation. In addition, the multiple microphones can be configured as an adaptive microphone array and used to reduce the effects of (i) room reverberation, when a near-end user is speaking, and/or (ii) acoustic echo, when a far-end user is speaking.

A conferencing system is described which includes a plurality of conferencing devices and a hub device. Each conferencing device is operable to process near-end speech energy signals to improve the intelligibility of the near-end speech, transmit the processed near-end signals to the hub device, receive far-end signals from the hub device, process the far-end signals, and present the processed far-end signals over its speaker. The hub device is operable to communicate with each of the conferencing devices and a voice communication system, selectively combine the processed near-end signals received from the conferencing devices, transmit the combined near-end signals to the voice communication system, receive the far-end signals from the voice communication system, and transmit the far-end signals to the conferencing devices.

A method and an apparatus for selecting an antenna from an antenna diversity array in a wireless communication system are provided. The method includes the steps of modulating an incoming signal with a spread-spectrum type modulation, measuring a signal quality value for each antenna in the antenna diversity array, and selecting an antenna from the antenna diversity array on the basis of the measured signal quality values. The step of measuring a signal quality value for each antenna may include computing peak-to-average ratio values by dividing a peak sample value by an average sample value, where the average sample value is determined by averaging all of the predetermined number of sample values for each of the plurality of incoming spreading codewords, and the peak sample value is determined by selecting the maximum of all of the predetermined number of sample values for each of the plurality of incoming spreading codewords.

A multi-channel acoustic cancellation system 40 with, for example, stereo speakers and a pair of microphones in the transmitting and receiving rooms (11 and 21) has time varying all-pass filters (45, 47) in the signal path between the microphones (13, 15) in the transmitting room (11) and the speakers (27, 29) in the receiving room (21) to provide decorrelation.

A signal enhancement system improves the understandability of speech or other audio signals. The system reinforces selected parts of the signal, may attenuate selected parts of the signal, and may increase SNR. The system includes delay logic, a partitioned adaptive filter, and signal reinforcement logic. The partitioned adaptive filter may track and enhance the fundamental frequency and harmonics in the input signal. The partitioned filter output signals may approximately reproduce the input signal, delayed by an integer multiple of the period of the fundamental frequency of the input signal. The reinforcement logic combines the input signal and the filtered signals to produce an enhanced output signal.

A downlink activity and double talk probability detector and method for an echo canceler circuit (10) improves the stability of an echo canceler adaptive filter (300) and improves the attenuation of post-echo canceler uplink data (388). The echo canceler circuit (10) includes a downlink activity and double talk probability data generator (30) and an echo canceler stage (20). The downlink activity and double talk probability data generator (30) receives pre-echo canceler uplink data (40) and downlink data (50) and in response produces double talk probability data (60) and downlink activity data (70). The echo canceler stage (20) receives the downlink data (50), the pre-echo canceler uplink data (40), the double talk probability data (60) and the downlink activity data (70), and in response produces uplink data (80).

A method including several steps is provided for selectively combining single-channel and multi-channel signals for loudspeaker output. A single-channel signal (54) is created (208) based on an inbound multi-channel signal (32, 36). A local voice activity level and a remote voice activity level are detected (210). If the remote voice activity level dominates the local voice activity level, α is set equal to a first percentage (212). Otherwise, α is set equal to a second percentage higher than the first percentage (214). At least one loudspeaker output signal (22, 24) is mixed comprising a proportion of the single-channel signal based on α and a proportion of the inbound multi-channel signal based on 1−α. A computer program product is also provided for the preceding method. An apparatus is also provided, including a receive combiner (52), a sound activity monitor (72), a mix and amplitude selector (56), and a monaural and stereo mixer (78). A system is also provided having a receive channels analysis filter (120).

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.

There is disclosed an echo canceller circuit for use in a full duplex transceiver of the type comprising a line driver capable of sending analog transmit signals through a cable and comprising a line receiver capable of receiving analog receive signals from the cable. An echo canceller impedance model circuit is coupled to an output of the line driver and is coupled to an input of the line receiver. The echo canceller impedance model circuit generates an echo canceller current that is equal in magnitude and opposite in phase to a current that represents signal echoes that are present in the analog receive signals. The echo canceller impedance model circuit has a variable impedance for generating the echo canceller current. The variable impedance has at least one variable resistor and at least one variable capacitor. The values of resistance and capacitance in the echo canceller impedance model circuit are varied in response to control signals from a echo canceller control circuit to compensate for and cancel signal echoes.