Apparatus for reducing acoustic background noise for use with a telephone handset (10) or a boom microphone device (100) or a boom headset (401) or the like. The apparatus includes first (12) and second (14) microphones which are arranged such that the first microphone (12) receives a desired speech input and the background noise present in the vicinity of the speech, and the second microphone (14) receives substantially only the background noise. The background noise from the second microphone (14) is converted into a corresponding electrical signal and substracted (16) from a signal corresponding to the speech and background noise obtained from the first microphone (12) so as to produce a signal representing substantially the speech.

Noise cancellation apparatus

A personal communications system is described which includes components of software and hardware operating in conjunction with a personal computer. The user interface control software operates on a personal computer, preferably within the Microsoft Windows® environment. The software control system communicates with hardware components linked to the software through the personal computer serial communications port. The hardware components include telephone communication equipment, digital signal processors, and hardware to enable voice, fax and data communication with a remote site connected through a standard telephone line. The functions of the hardware components are controlled by control software operating within the hardware component and from the software components operating within the personal computer. The major functions of the system are a telephone function, a voice mail function, a fax manager function, a multi-media mail function, a show and tell function, a terminal function and an address book function. The telephone function allows the present system to operate, from the users perspective, as a conventional telephone using either hands-free, headset or handset operation. The telephone function is more sophisticated than a standard telephone in that the present system converts the voice into a digital signal which can be processed with echo cancellation, compressed, stored as digital data for later retrieval and transmitted as digital voice data concurrent with the transfer of digital information data.

Computer-based multifunction personal communications system.

The present invention includes software and hardware components to enable digital data communication over standard telephone lines. The present invention converts analog voice signals to digital data, compresses that data and places the compressed speech data into packets for transfer over the telephone lines to a remote site. A voice control digital signal processor (DSP) operates to use one of a plurality of speech compression algorithms which produce a scaleable amount of compression. The rate of compression is inversely proportional to the quality of the speech the compression algorithm is able to reproduce. The higher the compression, the lower the reproduction quality. The selection of the rate of compression is dependant on such factors as the speed or data bandwidth on the communications connection between the two sites, the data demand between the sites and amount of silence detected in the speech signal. The voice compression rate is dynamically changed as the aforementioned factors change. A negotiation handshake protocol is described which enables the two sites to negotiate the compression rate based on such factors.

Dynamic selection of compression rate for a voice compression algorithm in a voice over data modem

Active noise control (ANC) has been widely applied in industry to reduce environmental noise and equipment vibrations. Most available control algorithms require the identification of the secondary path, which increases the control system complexity, contributes to an increased residual noise power, and can even cause the control system to fail if the identified secondary path is not sufficiently close to the actual path. In this paper, based on the geometric analysis and the strict positive real (SPR) property of the filtered-x LMS algorithm, we introduce a new ANC algorithm suitable for single-tone noises as well as some specific narrowband noises that does not require the identification of the secondary path, though its convergence can be very slow in some special cases. We are able to extend the developed ANC algorithm to the case of active control of broadband noises through our use of a subband implementation of the ANC algorithm. Compared to other available control algorithms that do not require secondary path identification, our developed method is simple to implement, yields good performance, and converges quickly. Simulation results confirm the effectiveness of our proposed algorithm

A New Active Noise Control Algorithm That Requires No Secondary Path Identification Based on the SPR Property

A personal communications system enables the operator to simultaneously transmit voice and data communication to a remote site. The personal communications system is equipped with two telephone line interfaces to allow connection between two remote sites. The connection between the first remote site and the local site may operate in a voice over data communications mode to simultaneously send compressed voice and data. A digital transmission protocol which is consistent with current packet standards is used to create an independent channel through use of a modified supervisory packet for negotiating communication parameters, including the speech compression algorithm, the speech compression ratio, the communication multiplex scheme, and other operations needed for control of remote hardware interfaces.

Voice over data modem with selectable voice compression

An active noise control apparatus for transmitting, from a loud speaker, a secondary noise synthesized so as to have the same amplitude as and the opposite phase to a primary noise and for canceling the noise by acoustically overlapping the secondary noise. An overall system filter for simulating a characteristic of an overall system leading to an error detecting microphone from a noise detecting microphone is provided for the first and the second overall system filter. A second and a first noise control filter are connected to the first and the second overall system filter in cascade to form a noise control filter. A coefficient of an estimating noise transfer system filter, obtained when the difference between the differential output of both obtained when a white noise is applied to the circuit of cascade connections and the response difference of the first and the second overall system filter from a differential overall system filter becomes a minimum, is made the coefficient of the noise control filter.

Active noise control apparatus

ADAPTIVE DIGITAL FILTER INCLUDING LOW-PASS FILTER ABSTRACT OF THE DISCLOSURE An adaptive digital filter for estimating a response characteristic of a signal path by monitoring a sampled output signal of the signal path, estimating a predetermined number of filter coefficients which represent the response characteristic of the signal path, and generating an estimated output signal of the signal path using a plurality of successive sampled input signals of the signal path and the estimated filter coefficients, where the estimation is carried out so that a difference between the output signal and the estimated output signal is reduced. Each of the filter coefficients is extracted by a low-pass filter where the low-pass filter coefficient in the low-pass filter can be set to a constant. Normalization can be carried out in either the input side or the output side of the low-pass filter. Otherwise, the low-pass filter coefficient may be set to 1-K?Xj(m)2/r, where r is set equal to a norm of the sampled input signals in the beginning, and is then set to an integrated power of the sampled input signals.

Adaptive digital filter including low-pass filter

An acoustic echo canceler and a side-tone echo canceler provided in a hands-free telephone suppress an acoustic echo and a side-tone echo respectively with few error by providing automatic gain controllers and/or limiters in the hands-free telephone so that the acoustic echo canceler and the side-tone echo canceler operate in linear, and a directional characteristic of a microphone used in the hands-free telephone for reducing the acoustic echo is controlled by a microphone direction controller so that the microphone operates as an omnidirectional microphone when a level of a received signal of the hands-free telephone is less than a designated level and as a bidirectional microphone when the level exceeds the designated level, further gains of an output of the microphone increases in a low frequency range.

Hands-free telephone set

A noise suppression apparatus is realized by at least one linear prediction analyzing circuit. Each linear prediction analyzing circuit includes: an adaptive filter which produces a linear prediction signal based on a first speech signal on which noise is superimposed, and outputs the linear prediction signal as a second speech signal in which the noise is suppressed; a subtraction unit which obtains a difference between the linear prediction signal and the first speech signal, and outputs the difference as a prediction error; and a coefficient updating unit which updates coefficients of the adaptive filter based on the first speech signal and the prediction error so as to minimize the prediction error. The noise suppression apparatus may includes a cascade connection of a plurality of linear prediction analyzing circuits each having the above construction. Alternatively, the linear prediction analyzing circuit may include: a lattice filter which produces a linear prediction signal based on a first speech signal on which noise is superimposed; and a subtraction unit which subtracts the linear prediction signal from the first speech signal, and outputs a remainder after subtraction, as a second speech signal in which the noise is suppressed.

Noise suppression apparatus realized by linear prediction analyzing circuit

In an active noise control system that uses the Filtered-x method as an adaptive algorithm, the impulse response on the error path is observed and the result is assigned as a coefficient of an error path filter before the system is started. Such an impulse response can obviously change after the system is started. The change can increase the difference between the assigned coefficient and the inherent coefficient of the error path filter, and thus render the operation of noise control unstable. This paper proposes a noise control filter coefficient renewal method that does not require the calculation of an error path filter coefficient by focusing on the fact that, when a set of two different coefficients is assigned to a noise control filter, the system consisting of components from a noise detection microphone to an error detection microphone establishes two independent equations whose variables are impulse responses for the feedforward path and the error path system. The coefficient of the noise control filer can be renewed by solving the simultaneous equations by iteration. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 85(12): 101–108, 2002; Published online in Wiley InterScience (www.interscience.wiley. com). DOI 10.1002/ecjc.1132

Juro Ohga

Papers

Active noise control apparatus

Adaptive digital filter including low-pass filter

Hands-free telephone set

Noise suppression apparatus realized by linear prediction analyzing circuit

Active noise control system using the simultaneous equation method without the estimation of error path filter coefficients