Abstract: A method for adjusting an operational characteristic of an audio device can include a series of steps. The method can include receiving a user spoken utterance from an audio speech source and detecting a position of the audio speech source relative to the audio device. The method further can include generating proximity data corresponding to the detected position and processing the received user spoken utterance with a selected signal processing technique based upon the proximity data. The signal processing technique can distinguish the user spoken utterance from background noise.
Type:
Grant
Filed:
April 25, 2001
Date of Patent:
October 4, 2005
Assignee:
International Business Machines Corporation
Abstract: A method and apparatus for reducing echo feedback in a wireless communication system (10) is accomplished when a receiving communication unit (24) senses a feedback signal, or echo, via an ancillary communication path. The receiving communication unit is a targeted recipient of an original audio signal generated by a transmitting communication unit (22), where the original audio signal (42) is conveyed to the receiving communication unit. Upon detecting the feedback signal and determining that it exceeds a feedback threshold, the receiving communication unit attenuates an audible output of the original signal to reduce echo to the transmitting communication unit. In addition, the receiving communication unit, and/or the transmitting communication unit include echo canceller to further minimize the echo within the digital communication system.
Type:
Grant
Filed:
April 5, 2004
Date of Patent:
September 20, 2005
Assignee:
Motorola, Inc.
Inventors:
Robert Novorita, Eric Ziolko, Gary Grube
Abstract: A telephone apparatus comprises a first audio input device, a second audio input device, a first audio output device, and a second audio output device. The telephone apparatus is operative to couple the first audio input device and the first audio output device to a first telephone line. In response to a first signal, the coupling between the first audio input device and the first telephone line is modified, and the second audio input device and the second audio output device are coupled to a second telephone line.
Type:
Grant
Filed:
August 26, 1999
Date of Patent:
September 6, 2005
Assignee:
SBC Properties, L.P.
Inventors:
Marc Ira Lipton, Robert Derrick Gourdine
Abstract: A data processing system comprises an audio processing means receiving data within said data processing system for converting the data into an analog signal with a first and a second analog channel components; an audio output means receiving the analog signal and providing a first and second output signal wherein the first output signal is provided for a loudspeaker and the output second signal is provided for a headset. The control of the audio processing means by a control signal providing an audio signal on the first or on the second channel or on both.
Type:
Grant
Filed:
March 29, 1999
Date of Patent:
August 30, 2005
Assignee:
Siemens Information & Communication Networks, Inc.
Abstract: To achieve an improved convergence behaviour for, e.g., echo cancellation there is provided a digital adaptive filter which includes a filter coefficient update means to successively update filter coefficients in accordance with an input signal, an estimated power of the input signal, and an error signal between the input signal filtered in the digital adaptive filter and the input signal propagated along an external path being modelled by the digital adaptive filter. Here, an input signal power estimation means is adapted to perform recursive smoothing for an increasing input power and a decreasing input power in an asymmetric fashion with different smoothing factors. In case the estimation is carried out in the frequency domain the step size for the update of filter coefficients may be calculated for each frequency band individually.
Abstract: A telephone having a hands-free mode of operation. The telephone includes a pair of microphones spaced apart from each other. Each microphone receives sound in hands-free mode of operation and provides audio signals representative of received sounds. The audio signals from each microphone may be converted to digital audio signals. The digital audio signals are presented to a fixed delay path and a variable delay path. Audio signals from both paths are combined and filtered in an adjustable filter to remove noise based upon a prior determination of the noise source location and the voice spectrum derived from the digital audio signals.
Type:
Grant
Filed:
November 27, 2001
Date of Patent:
August 31, 2004
Assignee:
Siemens Information and Communication Networks, Inc.
Abstract: In a speakerphone system, the signal that is directed to a loudspeaker is filtered by a dynamically controllable highpass filter. The filter characteristics are adjusted on the basis of the power of the signal and the degree of recent microphone activity. The filtered signal is a far-talker signal, while the microphone activity is represented as a near-talker signal that includes an echo component as a consequence of loudspeaker-to-microphone coupling. The degree of microphone activity is determined by a subband analyzer. A noise floor estimator may be used to determine the degree to which the near-talker signal is comprised of background noise, so that the noise can be eliminated in the process of controlling the highpass filter.
Type:
Grant
Filed:
January 8, 2002
Date of Patent:
August 31, 2004
Assignee:
Signalworks, Inc.
Inventors:
Hugh J. McLaughlin, Tamara L. Logan, Jeff L. Schenck
Abstract: Voice activity is detected by comparing a signal with two thresholds and producing data representing the energy of the signal. The data, in binary form, is compared with thresholds to determine voice activity. In accordance with another aspect of the invention, the thresholds are adjusted based upon statistical information. In accordance with another aspect of the invention, the data can be weighted to provide an indication of the quasi-RMS energy of an input signal. In accordance with another aspect of the invention, voice activity detectors, individually weighted, are provided at each input and each output of a telephone for reliably controlling echo cancelling circuitry within the telephone.
Abstract: A system and a method for the transmission and reception of digital data wirelessly using acoustic tones comprises a device having an encoder, a modulator system, a demodulator system, and a decoder. The modulator system uses a plurality of acoustical tones with a fundamental frequency and multiple harmonics that are amplitude modulated. The received signal is demodulated and detected by quadrature detectors comprising replicas of the acoustical tones. An extended Golay code based decoder is used for correcting errors occurring during transmission.
Type:
Application
Filed:
June 19, 2003
Publication date:
April 29, 2004
Applicant:
Disney Enterprises, Inc.
Inventors:
Thomas Richard McKnight, Eric C. Haseltine, Jesse N. Schell
Abstract: The present invention is directed to the provision of a personalized speaker phone and hands-free telephony. In particular, the present invention allows communications to be output along a narrowly defined path, rather than being broadcast. In this way, a private voice communication signal can be provided to a user, even though the user is not holding the output device to the user's ear. Furthermore, by providing audible signals along narrowly defined paths, different audible signals may be provided to users at the same location, without interfering with one another.
Abstract: A speakerphone accessory (100) is provided for use with a portable telephone (102). The speakerphone accessory (100) has a high-level speaker assembly (222) that includes a transducer (500) coupled to first and second resonators (502, 504). The first and second resonators (502, 504) are designed so that the first resonator (502) is most efficient at a high frequency while the second resonator (504) has maximum response at a low frequency. The frequency ranges produced by the first and second resonators (502, 504) overlap to reproduce frequencies between the high and low frequencies. The high-level speaker assembly (222), therefore, provides an acoustic bandpass having improved response when compared to the response of the transducer (500) alone.
Type:
Grant
Filed:
October 15, 1999
Date of Patent:
October 21, 2003
Assignee:
Motorola, Inc.
Inventors:
Robert A. Zurek, Thomas Gitzinger, Michael Charlier
Abstract: Apparatus (10) for transmitting sound between a user and a communications device such as a mobile telephone comprises a non-electrical ear piece (13) connected via a tube (12) to an adaptor (22) which can be electrically connected to the mobile phone. A further tube (14) has a voice entry end (14b) and is coupled at the other end (14a) to the adaptor. Sound travels up tube (12) from the phone to the user and down tube (14) from the user to the phone.
Abstract: A telephone having a hands-free mode of operation. The telephone includes a pair of microphones spaced apart from each other. Each microphone receives sound in hands-free mode of operation and provides audio signals representative of received sounds. The audio signals from each microphone may be converted to digital audio signals. The digital audio signals are presented to a fixed delay path and a variable delay path. Audio signals from both paths are combined and filtered in an adjustable filter to remove noise based upon a prior determination of the noise source location and the voice spectrum derived from the digital audio signals.
Abstract: A digital circuit for driving an audio transducer that provides consistent tonal quality over a range of volume levels, without requiring a variable gain analog amplifier. A fixed amplitude ringer tone is multiplied, or amplitude modulated, by a higher frequency digital pulse train to produce a transducer driving signal. The timbre of the transducer driving signal is similar to that of the fixed amplitude ringer tone, but the volume of the sound produced by the transducer varies with the mark-space ratio of the pulse train.
Abstract: Echo is canceled in a communication device including at least one microphone and a loudspeaker. Signals directed to the loudspeaker are filtered through at least one first filter. The filtered signals are subtracted from signals received by the microphone. At least one second filter filters the subtracted results to produce a signal corresponding to speech signals received by the microphone. At least one third filter filters the subtracted results to produce a signal corresponding to echo from the loudspeaker. The output from the third filter is subtracted from the output from the second filter to produce an echo-canceled signal. The device may include a number of microphones, and the first, second, and third filters may each include a number of filters equal to the number of microphones. The outputs from the second filter may be added, the outputs from the third filter may be added, and the sum of the outputs from the third filter may be subtracted from the sum of the outputs of the second filter.
Abstract: A voice-switching device for use in a telephone terminal which applies an optimal amount of switched loss to a plurality of distinct receive transducers. The voice-switching device detects speech signals on the transmit and receive paths to determine a mode of operation. Based on the mode of operation the device applies optimal amount of loss to the transmit path and each output path separately. This system eliminates excessive attenuation of non-dominant receive transducers as well as optimizes the transition time from active to quiescent mode for all receive transducers.
Type:
Grant
Filed:
September 4, 1998
Date of Patent:
March 18, 2003
Assignee:
Nortel Networks Limited
Inventors:
Paul Provencal, Christopher M Forrester
Abstract: An equalizer for a loudspeaker telephone includes a signal generator for generating a reference signal for broadcast by a loudspeaker of the loudspeaker telephone. A filter alters the frequency response of signals prior to broadcast by the loudspeaker. During set-up, a processor processes the reference signal picked up by a microphone remote from the loudspeaker that has been convolved with the acoustical response of the environment to determine coefficients for the filter that are necessary to compensate for the environment acoustics and applies the coefficients to the filter.
Abstract: A device and method of eliminating the interference to a voice command from the sound from the speaker, thereby improving the success rate of speech recognition even in the presence of the direct and echoed sound.
Abstract: In a speakerphone system, the signal that is directed to a loudspeaker is filtered by a dynamically controllable highpass filter. The filter characteristics are adjusted on the basis of the power of the signal and the degree of recent microphone activity. The filtered signal is a far-talker signal, while the microphone activity is represented as a near-talker signal that includes an echo component as a consequence of loudspeaker-to-microphone coupling. The degree of microphone activity is determined by a subband analyzer. A noise floor estimator may be used to determine the degree to which the near-talker signal is comprised of background noise, so that the noise can be eliminated in the process of controlling the highpass filter.
Type:
Application
Filed:
January 8, 2002
Publication date:
October 24, 2002
Inventors:
Hugh J. McLaughlin, Tamara L. Logan, Jeff L. Schenck
Abstract: There is provided a mobile telephone in which a class D amplifier is used to amplify the signals supplied to a loudspeaker. The baseband receiver circuitry, which remains switched on while the transmitter is active, processes only digital signals, thereby avoiding the “bumble bee” effect caused by signals from the transmitter interfering with signals in the receiver circuitry.
Abstract: A wireless user terminal operates in a mode that is selected based on an indication of the orientation of the wireless user terminal from a gravitational sensor in the wireless user terminal. When the wireless user terminal is in a first orientation, the wireless user terminal operates in a speakerphone mode. When the wireless user terminal is in a second orientation, the wireless user terminal operates in a handset mode. The mode of operation of the wireless user terminal transitions based on movement of the wireless user terminal from the first orientation to the second orientation. For example, when the wireless user terminal placed in a horizontal orientation, one of the speakerphone modes is selected. When the user picks up the wireless user terminal, the wireless user terminal transitions to handset mode.
Abstract: The conventional speakerphone using the voice switch, when combined with the echo canceler, involves a problem that the threshold for switching the transmit/receive state of the voice switch cannot comply smoothly and stably with the performance variation of the echo canceler, thereby obstructing echo canceling in a manner that the voice switch cooperates with the echo canceler.
The speakerphone of the invention estimates the performance variation of the adaptive filter by using the integrated value of a power of the receive signal or the transmit signal referred in the past when the adaptive filter learns, and varies the threshold in accordance with the performance variation. Thus, the speakerphone of the invention achieves a stable communication system that approaches to the fill duplex with the voice switching system and the echo canceling system combined.
Abstract: In a facsimile machine 1, when the speakerphone key 14 is depressed (S1: YES) and a call signal from some other telephone is not arriving to the facsimile machine 1(S2: NO), a speaker 26 is made ON and the microphone M is made OFF (S3). When at least one key of a dial key group 11 is depressed (S4: YES) and the telephone line 25 is connected to a telephone of the other end (S5: YES), the microphone M is made ON while the speaker 26 is maintained as being ON (S6).
Abstract: A method and apparatus for communicating both voice and control data between a communication device (such as a cellular phone) and an external accessory (such as a hands-free kit) over a data bus. The method includes formatting a sequence of bits into a repeating sequence of first time slots and second time slots, transmitting the voice data in the first time slot, and transmitting the control data in the second time slot. Notably, a first bit of each of the second time slots comprises a clock bit that alternates between a high value and a low value (e.g. a ‘1’ or a ‘0’) as between consecutive second time slots.
Type:
Grant
Filed:
February 8, 1999
Date of Patent:
December 11, 2001
Assignee:
Qualcomm Incorporated
Inventors:
Chienchung Chang, Way-Shing Lee, Robert Opalsky, George Pan, Karthick Chinnaswami, Hanchi David Huang, Steven C. Den Beste, James Hutchison
Abstract: An external speakerphone accessory system for a telephone, such as a LAN telephone, which enables the telephone user to place the speakerphone in a location that is convenient for multiple users, and which is not necessarily co-located with the attached telephone itself. The speakerphone accessory obtains all of its power, audio, and control signals from an attached telephone unit through a modified Universal Serial Bus (USB) interface, and includes a power supply system which enables it to obtain a relatively high power level from the telephone to which it is attached. The speakerphone accessory uses a delay detected between when an audio signal is received at a left hand side microphone and when that audio signal is received at a corresponding right hand side microphone to define zones of maximum sensitivity with regard to users of the speakerphone accessory.