U.S. Philips Corporation

Tekade Felten & Guilleaume Fernmeldeanlagen Gmbh

Peter Vary

Bernd Selbach

Karl Hellwig

Rudolf Hofmann

Wolfgang Brox

Ulrich Wellens

An echo canceller is described comprising an adaptive filter (15). The adaptation rate of the adaptive filter (15) is controlled by means of a control unit (29) determining the appropriate control signal in response to its input signals (x(k), u(f), e(k)). If the described echo canceller is inserted, for example, in the subscriber set of a telephone network, the adaptation process is to be effected only when the far-end subscriber is talking and the near-end subscriber is silent. If such a decision is automatically made with the associated signals (x(k), u(k)), an erroneous decision may at least temporarily considerably degrade the quality of the telephone communication. For this reason, the adaptation rate of the adaptive filter (15) is provided to be controlled in a gradual manner by means of the control unit (29) in response to suitably chosen mean values of its three input signals (x(k), u(k), e(k)), when the far-end subscriber is talking.

Echo canceller

Coupling a plurality of transmitters operating with different carrier frequencies to a single aerial is often effected by multiplexing the output signals by means of band-pass filters in the output frequency band. The band-pass filters have one end connected to a start network and from there to the aerial via an aerial transmission line, and the other end to the transmitters via circulators. For an optimal use of the available frequency band, for making less stringent requirements on the band-pass filters and for providing a dynamic channel assignment feature, the transmitter coupler according to the invention has a first stage in the form of a digital coupler and a second stage in which multiplexing is effected, using band-pass filters. The transmitter output signals are combined in the digital coupler, in the pre-group phase, by digital signal processing, preferably using the GTFM method. In the second stage the output signals of the parallel-arranged digital coupler are subjected to a digital-to-analog conversion in the group-phase and to frequency conversion.

Multi-stage transmitter aerial coupling device

A digital analyzing and synthesizing filter bank in which a digital polyphase network in the analyzing filter bank provides sampling rate reduction by dividing an input signal into M complex sub-band signals, and the sub-band signals are subjected to a further staggered sampling rate reduction of the real and imaginary components thereof. This achieves maximum sampling reduction while permitting aliasing distortions to be compensated in the synthesizing filter bank.

Digital analyzing and synthesizing filter bank with maximum sampling

Method of recognizing pauses in a speech signal when a slowly varying noise signal is superposed on the speech signal. For the purpose of pause recognition so-called short-time mean values connected with a clock pulse are continuously determined from the samples of the disturbed speech signal, which short time mean values are a measure of the average power of approximately 100 ms long sections of the disturbed speech signals. The sequence of these short-time mean values is then smoothed by linear filtration or by means of a median filter. In parallel with the smoothing operation an estimate for the noise signal power averaged over a few seconds is taken from the sequence of short-time mean values. If the smoothed short-time mean value is once or several times less than a threshold which is proportional to the above-mentioned estimate, then it is decided that there is a speech pause.

Method of recognizing speech pauses

A method of recognizing speech pauses in a speech signal even when the signal is disturbed by a slowly varying noise signal superposed thereon. Mean values which are an approximate measure of the average power of successive sections of the disturbed signal are determined from the short-time Fourier coefficients of the disturbed speech signal. The sequential short-time mean values are then smoothed by a linear digital filter or a median filter. An estimate of the noise signal power averaged over a few seconds is also recovered from the sequence of short-time mean values. A speech pause is signified when the smoothed short-time mean value (output of GL) more than once falls to a threshold which is proportional to the estimated noise power (output of PA).

In order to upgrade input signals for further processing, one or more of the input signals is split up by a decimating technique into spectral component signals which are processed according to the desired purpose and recombined by means of an interpolation technique into an output signal. In this manner a digital filtering device can be selectively employed for achieving an arbitrary combination of simultaneous filtering, signal improvement, echo or feedback compensation, or signal masking of the input signals.

Method of processing electrical signals by means of Fourier

During algebraic summing of code words from conference participants to form a partial sum-code word overflow of the values may occur. Such an overflow can be prevented when prior to the adding operations the numerical range is extended and an adjustment of weighting factors is effected in dependence on the measured speech intensity of the partial sum-code word. By multiplying by these (attenuation) factors the outgoing subscriber-code word is mapped again on the original numerical range in the attenuation circuit. To reduce a noise level originating, for example, from line interferences, quantization errors, etc. subscribers which are currently not speaking can be attenuated in a further attenuation control circuit of the conference arrangement.

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Herzogenaurach-Niederndorf, Germany

Peter Vary