Abstract: In a data processing system, a method includes receiving samples of a send path input signal (Sin) and samples of a receive path input signal (Rin); in response to receiving each sample of the send path input signal, providing a double-talk indicator to indicate whether or not double-talk has been detected; using the double-talk indicator to determine a double-talk density within a predetermined number of samples of the send path input signal or the receive path input signal; and using the double-talk density to provide a heavy double-talk indicator which indicates whether or not heavy double-talk has been detected. In response to the heavy double-talk indicator indicating that heavy double-talk has been detected, a heavy double-talk mode may be entered.

Abstract: The present invention provides a method and apparatus for finding an estimate of the delay of a signal travelling between two points. A quantity is evaluated from the signal at a final number of time instants, at both a reference point and a reception point. The values are quantized by comparison with a threshold adapted to a typical magnitude of the quantity. If the quantized values from the reception point are shifted back by the true delay with respect to the quantized values from the referenced point, then certain co-occurrences of quantized values have very low probability. Hence, the best delay estimate is that shift which yields the least number of low-probability co-occurrences.

Abstract: A method is disclosed that enables the managing of the overall sound level in an enterprise environment where telephones are used. A data-processing system such as a private branch exchange monitors whether one or more telephones are in use. Based on detecting when a first endpoint is in use and, therefore, when the endpoint's user is present, the private branch exchange controls one or more characteristics of the loudspeaker volume at a second endpoint. By accounting for other considerations such as the spatial closeness between the endpoints, which can be determined from office dimensions stored in a database, the private branch exchange of the illustrative embodiment is able to determine the degree of sound that is coupling over from one endpoint location to another. On a larger scale, the exchange is able to control the loudspeaker volumes of all of the endpoints in the workplace area. In doing so, the exchange manages the overall acoustic noise present.

Abstract: An incoming call prompting method is provided. The method includes the step of: providing a data storage for storing a note table, a melodic companion table, and an audio clip database; receiving an incoming call and sending the incoming call to a decoder to phone number of the incoming call; storing the phone number in the data storage; obtaining a predetermined digit set from the incoming call phone number stored in the data storage; obtaining the assigned note of each digit of the digit set, and piecing together a melodious sequence of note based upon a pre-determined algorithm guiding selection and sequencing of a combination of the primary notes and/or companion notes of the melodic companion table to edit a melody; and obtaining and outputting audio clip of each note of the edited melody in turn to generate a music.

Abstract: In a system having an adaptive filter block for receiving a reference signal and adapting to an input signal, and a block for detecting corruption of the input signal by an interference signal and in response limiting adaptation of the adaptive filter block, the improvement comprising delaying and applying the input signal to the adaptive filter block while applying the input signal to the block for detecting corruption without delay.

Abstract: An incoming call prompting method is provided. The method receiving an incoming call decoding the phone number of the incoming call so as to obtain a digit set from the phone number stored in the data storage system. The method further includes determining a call number character of the incoming call and assigning musical notes associated with the numbers of the digit set according to the musical note table. Furthermore, the method includes obtaining corresponding audio clip of each note from an audio clip database and outputting the audio clips with the corresponding rhythm of the call number character of the incoming call through a music output apparatus.

Abstract: A hand-held device that provides communication services and symbol processing is described. The device includes a case having a front side comprising a display and a lid having a transparent portion through which the display can be seen in a lid-closed state. In one version, the hand-held device comprises a QWERTY keyboard including an arrangement of digits as in a touch-tone area telephone keypad arrangement. In another version, the display includes a handwriting area for data entry using a stylus or other screen data entry device. In a closed position, the lid extends over the display and the keyboard, leaving another user input device physically accessible to touch for receiving input wherein the device performs processing responsive to receiving input from the physically accessible user input device in the lid closed state.

Abstract: An echo canceller apparatus comprises a receive side attenuator coupled in a receive side signal path that is configured to couple from a conference call bridge to a caller; a convolution processor coupled to the receive side signal path at a convolution processor pick-off point; a double-talk detector coupled to the receive side signal path and to a sending side signal path that is configured to couple from the caller to the conference call bridge; and logic coupled to the receive side attenuator which when executed is responsive to a double-talk condition detected by the double-talk detector and operable to determine a level of echo canceled by the convolution processor, to determine an additional amount of attenuation to introduce, and to activate the receive side attenuator to introduce the additional attenuation.

Abstract: An enhanced sidetone system is disclosed which provides the user of a telecommunications terminal, while speaking, with immediate audio feedback that corresponds to what the far-end party is probably hearing. The sidetone system continuously samples the input speech signal from the user and also obtains signal quality statistics of the transmission path. These statistics can include descriptions of network quality-of-service characteristics (e.g., packet loss rate, etc.) and/or media quality characteristics (e.g., audio distortion due to echo cancellation, etc.). These statistics enable the disclosed technique to determine whether the transmitted signal quality is acceptable. When an unacceptable condition in transmit-path signal quality is detected, the technique modifies the traditional (main) sidetone signal.

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: A system includes microphone devices, echo cancellation mechanism, and speaker devices. Each echo cancellation mechanism corresponds to one of the microphone devices. Each speaker device corresponds to a sound source signal. Each sound source signal is spatially distributed to all the speaker devices, and each echo cancellation mechanism receives as a reference signal a combination of all the sound source signals.

Abstract: In an echo processing method and device which can detect an accurate echo section without effects of a far end signal, an echo delay, and a reduction of an echo cancellation amount, a signal of a specified frequency band is generated in conformity with a near end signal, and the signal of the specified frequency band is added to the near end signal to form a transmitting signal. Receiving signals are separated into the signal of the specified frequency band and a signal of a band other than the specified frequency band. An echo section is detected based on the signal of the specified frequency band separated. An echo component in the signal of the band other than the specified frequency band is removed and a level of the echo component is detected based on the near end signal in the echo section.

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: A multichannel echo canceller according to the present invention includes an echo cancellation section which receives loudspeaker input signals sp1, sp2 including a second acoustic signal to be respectively inputted to a plurality of loudspeakers 10, 20 provided in a first location and detection signals m1, m2 detected by a plurality of microphones 11, 21 provided in the first location; separates a first acoustic signal and a second acoustic signal, which are included in the detection signals m1, m2, by performing signal processing based on an independent component analysis; cancels as an echo the second acoustic signal, which is included in the detection signals m1, m2, by outputting only the separated first acoustic signal to a plurality of loudspeakers 30, 40 provided in a second location.

Abstract: In a speech communication device capable of bidirectional communication, a clock deviation between the speaker side (the receiving side) and the microphone side (the transmitting side) is detected, and based on the above deviation, either a first frequency signal (speaker signal (reference signal)) or a second frequency signal (microphone signal) is frequency-shifted in the frequency domain, and therefore, an influence caused by the clock deviation can simply be corrected without an increased processing amount, and thus, an echo can be suppressed effectively.

Abstract: A system and a method for canceling an echo is disclosed. In accordance with the system and the method, a plurality of independently and variably delayed adaptive algorithm blocks are selectively applied to a delayed feedback signal to generate a plurality of echo components in parallel, thereby canceling the echo component from an input signal.

Abstract: In a particular embodiment, a circuit device includes a first circuit having a first plurality of serial terminals including a first data receive terminal and a first data transmit terminal. The first plurality of serial terminals is communicatively coupled to a particular circuit via isolation circuitry to communicate first serial data. The circuit device further includes a second circuit having a second plurality of serial terminals including a second data receive terminal coupled to the first data transmit terminal and including a second data transmit terminal coupled to the first data receive terminal to communicate second serial data to the particular circuit via the first data receive and transmit terminals.

Abstract: An echo canceler 10 generates an echo elimination signal by filtering through adaptive filters 101 and 102 reference signals input from sound sources causing echoes. It includes a sound source number detecting unit 103 for detecting the number of the sound sources causing echoes from the reference signals, and a control unit 105 for making the number of taps of the adaptive filters 101 and 102 variable in accordance with the number of the sound sources detected by the sound source number detecting unit 103.

Abstract: Signal detectors are described herein. By way of example, a system for detecting signals can include a microphone signal detector, a loudspeaker signal detector, a signal discriminator and a decision component. When the microphone signal detector detects the presence of a microphone signal, the loudspeaker signal detector detects the presence of a loudspeaker signal and the signal discriminator determines that near-end speech dominates loudspeaker echo, the decision component can confirm the presence of doubletalk. When the microphone signal detector detects the presence of a microphone signal and the signal discriminator determines that near-end speech dominates loudspeaker echo, the decision component confirms the presence of near-end signal.

Abstract: An echo canceller which can respond to a sudden change in echo characteristics in real time and does not require an alteration outside the echo canceller, includes a smoothed sending-speech signal calculation means for calculating a smoothed sending-speech signal from the sending-speech signal, the smoothed sending-speech signal being obtained by smoothing the sending-speech signal; a smoothed receiving-speech signal calculation means for calculating a smoothed receiving-speech signal from the receiving-speech signal, the smoothed receiving-speech signal being obtained by smoothing the receiving-speech signal; a delay time information generation means for obtaining delay time information reflecting delay characteristics of an echo path in accordance with a correlation between the smoothed sending-speech signal and the smoothed receiving-speech signal; and an update information generation means for obtaining update information indicating execution or suspension of updating of the tap coefficients of the adapti