Abstract: A three-dimension active silencer includes a measuring unit for detecting a signal relating to a sound radiated from a target sound source which is located in a closed box. A plurality of additional sound sources are arranged around the target sound source. A microphone is arranged at a predetermined position for detecting a sound power at the position. A controller is adapted to control respective amplitudes of the additional sound sources which can be different from an amplitude of the target sound source and respective phases of the additional sound sources, based on the signal detected by the measuring unit, in such a manner that the microphone detects the lowest sound power. The additional sound sources are arranged in such a manner that a position xp of the target sound source, respective relative positions di (i=1, 2, . . .

Abstract: A audio amplifier employs microchannel plate for multiplying input electrons to produce output electrons at a rate corresponding to an audio input signal. The MCP may have a segment input source, segmented input and output electrodes or a segmented anode for allowing independent amplification of multiple input channels.

Abstract: Disclosed herein is a three-way call detection system and method for detecting the addition of a third party to a pre-existing telephonic connection between a first party and a second party. The system comprises a meter for measuring the amplitude of a line voltage of a telephonic connection between a first party and a second party.

Abstract: Resistance in parallel with inductors in a series leg of the low-pass filter facilitates changing input and output resistance of the filter with little or no change in the reactance of the inductors. Furthermore, the reactance of the capacitors in the shunt legs of the filter will be substantially unaffected. This assists the designer in matching the impedance of the filter in the pass-band while still providing substantial impedance mismatching in the stop-band without substantially affecting the characteristics of the filter. Facilitating impedance matching in the pass-band and impedance mismatching in the stop-band is accomplished without the need for more complex active components. POTS splitters making use of such filters can facilitate impedance matching of line and load termination in a telecommunications system in response to signals within the frequencies of typical analog telephony service while providing impedance mismatching in response to signals within the frequencies of typical xDSL service.

Abstract: A multifunctional analog TRK circuit board is designed so as to serve as both loop, ground start, DID, E&M, ring down and both way TRK. Thus, the various analog TRK circuits are embodied in a single TRK board, thereby increasing productivity compared to the conventional system requiring separate TRK circuit boards. The multifunctional TRK circuit is selectively controlled to provide desired TRK functions.

Abstract: A computer system includes an audio chip to generate audio signals at a target volume level to be emitted as audio output by speakers. Also included in the computer system may be a volume control routine to limit the overall volume level of the system. A maximum volume level may be set by a user through interacting with a software routine. The software routine may include password protection, requiring a user to enter a correct password in order to modify the maximum volume level. If the target volume level is above the maximum volume level, then the volume control routine may set the overall volume level of the system at the maximum volume level. Otherwise the volume control routine may set the overall volume level at the target volume level.

Abstract: In an active noise control apparatus in which an audio signal and a noise control signal are mixed to thereby sound a loudspeaker inside a motor vehicle, a general-purpose audio head unit cannot conventionally be freely changed for the one already mounted on the motor vehicle. As a solution, an active noise control unit is provided with a buffer amplifier and a mixer. An audio signal amplified by a power amplifier in an audio head unit is adjusted in level by the buffer amplifier such that a level of an audio signal outputted after amplification by the first amplifier becomes substantially equal to a level of the audio signal outputted from the audio head unit. Then, at the mixer, a noise control signal outputted from an active noise controller is added to an audio signal outputted from the buffer amplifier. Therefore, any audio head unit can be freely changed for the one already mounted on the vehicle as long as the audio head unit contains therein a power amplifier.

Abstract: The present invention provides an echo canceller having a holding portion which holds at least a communication frame length's worth of reception signals; a power calculation portion which calculates power for each communication frame based on the reception signals held in this holding portion; a divider which divides, by the power calculated for each communication frame, the echo elimination residue signal for each sample of the communication frame output from a subtractor; and an update portion which updates the tap coefficients of an adaptive filter according to the output signals from the divider.

Abstract: The invention relates to an SLIC circuit (1), which is connected directly to a telephone line (2), for joint transmission of voice and data via the telephone line (2), with adequate echo attenuation achieved even at high data frequencies in that a first current which corresponds to the sum of the transmission and received currents (Ia, Ib), and a second current (Id) which corresponds to the current produced by the transmission power level (Vtx) at the data frequency, are produced, and the difference between the first current and the second current (Id) is emitted.

Abstract: Crosstalk between subscriber loops used to transmit different broadband services through the same bundled telephone cable is a significant limitation to providing digital subscriber line services. A method for estimating the crosstalk, identifying the sources of the crosstalk and predicting additional sources of crosstalk are disclosed. The crosstalk sources are identified in the frequency domain by maximizing the correlation with a “basis set” of received crosstalk PSDs which consist of the cascade of a finite set of known transmit PSDs types times a representative set of crosstalk couplings. Multiple crosstalk types are identified with a technique of successive spectral subtraction. Once a type is identified the crosstalk disturber is compared against all other members of the complete set for that type. Additionally, Mutliple Regression (MR) techniques and a Matching Pursuit (MP) algorithm are used to increase the ability of the system and method to identify various crosstalk disturbers.

Abstract: A multi-channel filter (20) includes S inputs for receiving S input signals (x1 . . . xS) and at least S filters (14, 28) coupled to the S inputs for filtering the input signals (x1 . . . xS) or auxiliary signals (y1 . . . yS) derived therefrom. The multi-channel filter (20) further includes a signal combiner (16) for combining the filtered signals into an output signal (ê) and a further signal combiner (22) coupled in between at least part of the inputs and at least part of the filters (14, 28). The further signal combiner (22) is arranged for combining at least part of the input signals (x1 . . . xS) into at least one auxiliary signal (y1 . . . yS) The multi-channel filter (20) is asymmetric in that at least one of the filters (28) has a filter length which is substantially larger than the filter lengths of the other filters (14).

Abstract: A method and an apparatus for controlling a sound function of a computer system based on whether or not a headphone plug is inserted into a monitor is disclosed. A switching section receives a sound signal from a main body of the computer system and outputs the sound signal to a speaker or to an inserted headphone plug based on whether or not the headphone plug is inserted by a user. A signal detecting section detects an insertion voltage signal by monitoring an insertion of the headphone plug from the switching section. A control section compares the insertion voltage signal from the signal detecting section with a predetermined reference voltage signal, and controls a speaker installed in the main body according to the compared result. Accordingly, the user can enjoy sound, such as music, regardless of the time.

Abstract: A method and audio converter for generating further audio signals (u, ul, ur, uc, us) from initial audio signal (x, xl, xr), wherein optionally an information signal (in means 23) is derived from said initial audio signals (x). On basis of the initial audio signal (x, xr, xl), a dominant signal y(k) and a residue signal (or signals) q(k), substantially transverse to each other, are determined (means 21 and 22). In at least two frequency ranges frequency components of the dominant signal are analysed (means 24), and a difference signal yr ({y(k)?yb(k)) corresponding to the dominant signal minus a frequency range component of the dominant signal in one or more frequency ranges (yb(k)) is formed. The difference audio signal yr and the residue signal q(k) are transformed into said further audio signal u(means 25), i.e. u = T ? ( y r ? ( k ) q ? ( k ) ) .

Abstract: The device for detecting telephone line seizure performs a measurement of voltage of the telephone line across a first and a second optocoupler connected in series to the telephone line and arranged head-to-tail. The first optocoupler makes it possible to discharge a capacitor connected to the telephone line, the second optocoupler converts the discharge current into an image current in a measurement circuit. This device comprises means ensuring negative feedback of the second optocoupler to the first optocoupler. With this layout, the influence of the transfer coefficients of the optocouplers on the voltage measurement is decreased, so as to increase the accuracy of measurement.

Abstract: A system and method for determining DTMF coefficients stores a plurality of reference templates, at least some of said reference templates representative of likelihood ratios. A plurality of LPC coefficients is received. The LPC coefficients are representative of an input signal. The system and method determines a plurality of current likelihood ratios based, at least in part, upon the LPC coefficients. The system and method also determines an initial tone based, at least in part, on the minimum of the current likelihood ratios and the minimum of the reference likelihood ratios. A plurality of LSF coefficients is received. The LSF coefficients are determined, at least in part, upon the input signal. The system and method verifies the validity of the initial tone based, at least in part, upon the LSF coefficients.

Abstract: A circuit where the same amplifiers and the same volume adjustment circuitry are used for the ringer mode as well as for the audio mode of a mobile phone is achieved. The volume adjustment in the audio and ringer mode is provided by a precise amplitude setting via the gain control stages of an inverting voltage amplifier used in a bridge circuit through a current-voltage conversion. This volume adjustment circuit avoids the high power dissipation of a volume control through pulse width modulation (PWM) and avoids the risk of over-and undershooting of the amplifier's output signal due to a high slew rate in combination with the inductance of the loudspeaker. The current-voltage conversion is performed by a series of resistors activated by a series of correspondent switches. High impedance current less sense paths are eliminating the parasitic effect of the resistance of low cost standard switches to adjust the volume of the loudspeaker in the audio and the ringer mode.

Abstract: An audio signal processing device is described for deriving auxiliary audio signals, such as audio direction sensing signals or a center audio signal from first and second audio signals through first and second filter paths, each of which comprises a first adaptive filter, and a first summing means is provided for coupled to the first adaptive filters for providing a summed audio signal at its summing output. Each filter path further comprises a second adaptive filter coupled to said summing output, whose respective adaptive filter coefficients are transferred to the first adaptive filters and are adapted in response to respective comparisons of the first and second audio signals with filtered sums of the first and second audio signals. Therewith correlated and uncorrelated parts of the input audio signals are processed effectively.

Abstract: A method of repeating a non-voice signal such as a dual tone, multiple frequency (DTMF) signal, by inserting a delay sequence of data values into an output data sequence of data values, a portion of the output data sequence following the delay sequence being the same as a corresponding portion of an input sequence of decoded data obtained from a low bit-rate speech decoder. The input sequence has at least one distorted non-voice sequence. The method provides for inserting a substantially undistorted non-voice sequence into the output sequence, the undistorted sequence being at least of substantially the same length as the distorted sequence, a portion of the output sequence following the undistorted sequence being the same as a corresponding portion of the input sequence, and the output sequence being substantially free of the distorted non-voice signal.

Abstract: A digitized input signal is sampled, to obtain a reference input signal. The reference input signal is spuriously whitened, to obtain a pseudo whitened signal. An echo replica signal is generated on the basis of the pseudo whitened signal and an adaptive filter coefficient. A cancellation error signal is obtained on the basis of an echo signal and the echo replica signal. The adaptive filter coefficient is updated on the basis of the cancellation error signal, the adaptive filter coefficient, and the pseudo whitened signal.

Abstract: An echo canceller has a high-pass filter 11, a speaker 3, an A/D converter 6, and an echo canceller 7. The high-pass filter 11 suppresses a low-frequency component in a received signal. The speaker 3 outputs an acoustic sound of a low-frequency suppressed received signal passed through the high-pass filter 11. The A/D converter 6 converts an acoustic echo from a microphone 4 into a transmission signal of a digital form. The echo canceller 7 generates a pseudo echo signal based on the low-frequency suppressed received signal passed through the high-pass filter 11, and eliminates the acoustic echo to be inputted to the microphone 4 from the speaker 3 by subtracting the pseudo echo signal from the digital signal converted by the A/D converter 6.