231results about "Echo suppressors/cancellers" patented technology

A signal processing method including the steps of: receiving a first audio signal; detecting the presence of one or more shrieks within an audible frequency range of said audio signal; creating one or more filters to selectively attenuate the respective one or more shrieks within the audible frequency range; filtering the audio signal using the one or more filters; and transmitting the filtered audio signal to an audio telephone device.

The invention relates to a method and device for controlling the sound effect. The method for controlling the sound effect comprises the step of detecting whether a hands-free call channel of a mobile terminal is sheltered or not when the mobile terminal executes hands-free calls, and the step of adjusting the configuration of the hands-free call channel of the mobile terminal and / or outputting a warning signal to prompt a user that the hands-free call channel is sheltered when the hands-free call channel of the mobile terminal sheltered. According to the method for controlling the sound effect, when the mobile terminal executes the hands-free calls, whether the hands-free call channel of the mobile terminal is sheltered or not is detected, and when the hands-free call channel is sheltered, the configuration of the hands-free call channel of the mobile terminal is adjusted and / or the warning signal is output, and therefore the hands-free call quality of the mobile terminal can be effectively improved.

An audio conferencing system in which each participant in the conference generates a monaural signal for transmission on an output channel to the other participants, and receives a plurality of input channels from the other participants. The input channels are individually converted to stereophonic signal pairs by respective filter banks having selected predetermined transfer functions. The stereo channels are combined linearly to provide a stereophonic audio output. The system has an echo cancellation system in which an adaptive filter process is applied to the input channels to generate a combined echo cancellation signal for applying to the output channel.

A videoconferencing system has a videoconferencing unit that use portable devices as peripherals for the system. The portable devices obtain near-end audio and send the audio to the videoconferencing unit via a wireless connection. In turn, the videoconferencing unit sends the near-end audio from the loudest portable device along with near-end video to the far-end. The portable devices can control the videoconferencing unit and can initially establish the videoconference by connecting with the far-end and then transferring operations to the videoconferencing unit. To deal with acoustic coupling between the unit's loudspeaker and the portable device's microphone, the unit uses an echo canceller that is compensated for differences in the clocks used in the A / D and D / A converters of the loudspeaker and microphone.

The invention relates to a method for the transmission of call control parameters for switching over between a voice transfer mode and a data transfer mode between two media gateway controllers (6, 7), which are used by way of an IP network between two telecommunication terminal devices separated from any medium or bearer connection, whereby the call control parameters are transferred with conversion into the SIP or SIP_T protocol or from this into a standard signaling protocol.

A new audio echo cancellation (AEC) approach is disclosed. To facilitate echo cancellation, the method adjusts for errors (called drift) in sampling rates for both capturing audio and playing audio. This ensures that the AEC module receives both the signals at precisely the same sampling frequency. Furthermore, the far-end signal and near-end mixed signal are time aligned to ensure that the alignment is suitable for application of AEC techniques. An additional enhancement to reduce errors utilizes a concept of native frequency. A by-product of drift compensation allows for excellent buffer control for capture / playback and buffer overflow / underflow errors from drift errors are eliminated.

A universal reconfigurable system and method are provided for reducing nonlinear echo and residual echo, and for echo leakage prevention in various time-varying and complex environments is proposed in this invention. According to one embodiment an echo cancellation system includes (1) an adaptive linear filter implemented in either time-domain or frequency-domain; (2) a nonlinear echo suppression; (3) echo leakage prevention; (4) direct current (DC), low frequency residual echo and noise reduction; (5) time-domain nonlinear processor (NLP) to reduce the residual echo; and (6) frequency-domain NLP to further reduce the residual echo. The echo cancellation system is universally applicable to acoustic echo cancellation (AEC) and / or electrical echo cancellation applications. In terms of AEC, this invention is reconfigurable for one or more microphones and / or one or more reference channels. The numbers of microphones and reference channels are user configurable.

Certain aspects of a method and system for a dual mode subband acoustic echo canceller with integrated noise suppression may include splitting an input signal into a lowband component and a highband component. The subbands of each of the lowband component and the highband component may be processed in order to reduce an echo associated with the input signal and to suppress the noise associated with the input signal.

Apparatuses and methods are disclosed for automatic volume control of an audio stream reproduced by a captioning communication service for use by a call assistant in generating a text transcription of a communication session between a hearing-impaired user and a far-end user. The automatic volume control automatically adjusts a volume of the audio stream reproduced by the captioning communication service responsive to a volume control command identifying which of the far-end voice signal and the near-end voice signal is active at a given time. The system further includes an echo modifier configured to add distortion to an echo portion of the far-end voice signal when generating the audio stream.

A method and apparatus for detecting whether a remote party has added a secondary telephone destination to a telephone call, for example through the activation of three-way calling service, conference calling or two lining bridging, by identifying an echo characteristic to the telephone connection between the local and the remote telephone and monitoring the echo characteristic to determine whether there is a significant change. The addition of a secondary telephone destination can be verified by continuing to monitor the echo characteristic to determine whether it has returned to its original value.

Methods and corresponding systems for reducing an echo signal include assigning a subsegment delay to each of a plurality of subsegment adaptive filters. A send signal and a delayed receive signal are coupled to each of the subsegment adaptive filters, wherein the delayed receive signal is delayed according to a respective subsegment delay. A set of filter coefficients in each of the subsegment adaptive filters are adapted, in parallel, to correspond to a respective subsegment impulse response of a connected system. Each set of filter coefficients is analyzed to produce a pure delay, and the pure delay is used to delay the receive signal for a main adaptive filter. An echo signal replica is produced using the main adaptive filter and an error signal. The echo signal replica is subtracted from the send signal to produce the error signal.

In an audio conferencing environment, including multiple users participating by means of a series of associated audio input devices for the provision of audio input, and a series of audio output devices for the output of audio output streams to the multiple users, with the audio input and output devices being interconnected to a mixing control server for the control and mixing of the audio inputs from each audio input devices to present a series of audio streams to the audio output devices, a method of reducing the effects of cross talk pickup of at least a first audio conversation by multiple audio input devices, the method including the steps of: (a) monitoring the series of audio input devices for the presence of a duplicate audio conversation input from at least two input audio sources in an audio output stream; and (b) where a duplicate audio conversation input is detected, suppressing the presence of the duplicate audio conversation input in the audio output stream.

A system and method for audio multicast includes a flexible number of active and passive conferencing endpoints in packet communication with a server. The server creates a mixed audio stream from the received audio packets from the active endpoints. The server multicasts the mixed audio to all the conferencing endpoints. The endpoints determine if the received mixed audio includes any self-generated audio by comparing the received packet to a sample packet of self-generated audio stored prior to transmission to the server. The endpoint encodes and decodes its own audio twice to match the transformations that occurred for the endpoint's contribution to the mixed audio. If a match is present, the endpoint removes the self-generated audio from the mixed audio and plays the conference audio.

A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

A teleconferencing system comprises a conference bridge having a multichannel connection to each customer equipment. The customer equipment has a unit to separately process each channel to provide one output representing each of the other participants. These outputs can be combined in a spatializer to provide a spatialized output in which each participant is represented by a virtual sound source. The conference bridge comprises a concentrator, having a unit to identify the currently active input channels and to transmit only those active channels over the multichannel connection, together with control information identifying the transmitted channels.

A signal processing system which discriminates between voice signal and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

The present invention proposes a method of providing information of the echo path of a speech connection in a Packet Data Network, said method comprising the steps of requesting delay information by means for triggering echo removal, said delay information being obtained by signaling a non-speech measurement data unit to said Packet Data Network by an echo estimation means, and measuring the delay of the echo of said measurement data unit by said echo estimation means; replying the echo delay of said measurement data unit from said echo estimation means; generating delay information including at least said echo delay of said measurement data unit; and adjusting a shift register being related to an echo removing device by the use of said delay information such that the performance of said echo removing device is optimized in view of its echo removing capability.

Monitoring voice quality passively using line echo cancellation data across a telecommunications network and reporting monitoring data to a central network management system. Network is monitored for potential voice quality issues for pro-active isolation of problems prior to customer complaints about the problems. Line echo cancellation related and non-related data for IP and other networks is gathered and correlated together to provide voice quality assessments of network performance.

A speech enhancement system improves speech conversion within an encoder and decoder. The system includes a first device that converts sound waves into operational signals. A second device selects a template that represents an expected signal model. The selected template models speech characteristics of the operational signals through a speech codebook that is further accessed in a communication channel.

The present invention relates to a desktop speakerphone constructed to make the desktop speakerphone less space-consuming while having improved audio qualities. The desktop speakerphone preferably has two microphone clusters 6, 7 mounted at the upper side of the housing 2 closer towards respective longitudinal ends 8 of the latter, so that each microphone cluster 6, 7 can receive voice sound Av from one or more of the users. Each microphone cluster 6, 7 preferably comprises three pressure microphones 10, 11, 12. Furthermore, within each microphone cluster 6, 7, the second and third sound inlets 14, 15 are preferably arranged symmetrically on opposite sides of the respective median plane 18. Within each microphone cluster 6, 7, the relative arrangement of the three sound inlets 13, 14, 15 defines a respective microphone axis 9, 19 for each of the microphone pairs 10, 11, 10, 12. The microphone axis 9 of the first microphone pair 10, 11 extends through the first and the second sound inlet 13, 14, while the microphone axis 19 of the second microphone pair 10, 12 extends through the first and the third sound inlet 13, 15. The microphones are connected to multiple array processors, each configured to provide an array signal in dependence on two or more of the multiple microphone signals. First and second array processor configured to provide array signals in dependence on microphone signals.

A communication system adaptively cancels noise and / or interference from signals communicated through a communication channel, such as signals communicated by a telecommunication network. The system, based on a common mode signal of a received signal, generates an estimate of noise or interference within a differential mode signal of the received signal. The system then subtracts the estimate from the differential mode signal in an effort to remove noise from the differential mode signal thereby providing a differential mode signal that is substantially free of the estimated noise or interference.

A novel and useful mechanism for the detection and estimation of far-end cross talk (FEXT) caused by transmissions over adjacent wire pairs or cables. The detection and estimation technique of the invention can be used to identify FET impairments in Ethernet and DSL systems. The detection and estimation scheme is performed utilizing an adaptive FEXT canceller filter for detection purposes during normal operation of the system. FEXT filter tap coefficients are determined one at a time using a mechanism shared amongst all taps. Once adapted, the filter coefficients can be used to estimate FEXT strength and to determine spectral measurements of the FEXT impairment noise.

Audio from a near-end that has been acoustically coupled at the far-end and returned to the near-end unit is detected and suppressed at the near-end of a conference. First and second energy outputs for separate bands are determined for the near-end audio being sent from the near-end unit and for the far-end audio being received at the near-end unit. The near-end unit compares the first and second energy outputs to one another for each of the bands over a time delay range and detects the return of the sent near-end audio in the received far-end audio based on the comparison. The comparison can use a cross-correlation to find an estimated time delay used for further analysis of the near and far-end energies. The near-end unit suppresses any detected return by muting or reducing what far-end audio is output at its loudspeaker.

A digital audio conferencing system has a fixed base station that is in communication with a far end (R.E.) system over a communication network. The base station is associated with a wireless loudspeaker and one or more wireless microphones. The base station operates to receive F.E. audio signals to be played by the wireless loudspeaker, and it operates to remove acoustic echo picked up by the wireless microphones. A first clock controlling F.E. audio signal sampling at the base station, and a second clock controlling audio signal sampling and at a wireless microphone are synchronized to one master, reference clock that controls the operation of the base station. Acoustic echo included in an audio signal picked up by a wireless microphone is removed by AEC functionality running in the base station.

In an example embodiment, a control connection is introduced between an adaptive jitter buffer (AJB) and an adaptive bulk delay (ABD) buffer of an echo canceller (ECAN) in an Internet Protocol (IP) conference bridge. The control connection allows the AJB to control the amount of delay inserted by the ABD in the ECAN convolution processor (CP) signal path. The adjustment in ABD delay restores the time alignment of the ECAN internal echo estimate and offsets variations in echo tail delay introduced by network induced AJB delay changes. Time-invariance is preserved in the echo tail path.