A processing method in a wireless telecommunications receiver receiving a digitally modulated single-carrier signal includes, between a matched filter, in the time domain, operating at a frequency drx×B and a frequency equalizer, operating at the frequency B, a decimation step comprising: i/extracting, from a filtered signal frame, a first sequence of samples for aiding the decimation and having the same power; and a second sequence of payload samples intended to be equalized; ii/estimating the variance in the power of each of the drx decimation phases of the first sequence and identifying the n.sup.th decimation phase associated with the minimum variance; iii/decimating the second sequence by selecting the n.sup.th decimation phase of the second sequence and supplying the decimation phase at the input of the frequency equalizer.

A digital interpolation filter delivering a series of output samples approximating a signal x(t) at sampling instants of the form (n+d)T s based on a series of input samples of the signal x(t) taken at sampling instants of the form nT s. Such a filter implements a transfer function in the Z-transform domain, H c<i/>d (Z−1), expressed as a linear combination between: a first transfer function H 1 d<i/>(Z−1) representing a Lagrange polynomial interpolation of the input samples implemented according to a Newton structure (100); and a second transfer function H 2 d (Z−1) representing another polynomial interpolation of the input samples implemented according to another structure comprising at least the Newton structure; the linear combination being a function of at least one real combination parameter c.

A method and device for resampling an audio frequency signal in an audio frequency signal coding or decoding. The method includes the following acts for each signal block to be resampled: determining, by adaptive linear prediction, a number of future signal samples, this number being defined as a function of a chosen resampling delay; constructing a resampling support vector from at least samples of the current block and determined future signal samples; applying a resampling filter to the samples of the resampling support vector.

An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes n analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.

An upper limit of a frequency range of audio indicated by input audio data is detected. A representative point extraction unit downsamples the input audio data to a sampling rate set to be less than or equal to twice the detected upper limit to obtain representative-point audio data. An interpolation processing unit upsamples the representative-point audio data by using a fractal interpolation function (FIF) that uses a mapping function calculated by a mapping function calculation unit, while using the input audio data, if necessary, to generate high-frequency interpolated audio data.

A method of increasing the sample rate of a digital signal by creating intermediate sample points between adjacent neighbouring sample points comprising the step of populating each of the intermediate sample points depending on a weighted influence of a predetermined number of the neighbouring sample points, the weighted influence being calculated by representing the digital signal or filter at the predetermined number of sample points at least in part by its cosine components, which are each represented by absolute values of a cosine function in the time domain substantially limited to half a waveform cycle at its mid-point; combining the aforementioned cosine components at each of the neighbouring sample points to obtain waveforms at each of the neighboring sample points; determining values for each of the waveforms at the intermediate sample points and combining the determined values at the intermediate sample point to derive the weighted influence.

A method of processing a digital signal includes providing a digital filter including neighbouring sample points and performing a sample rate increase on the digital filter to provide intermediate sample points between adjacent neighbouring sample points, said intermediate points being populated dependent on a weighted influence determined in the time domain of a predetermined number of the neighbouring sample points. The digital filter is applied to the signal where: i) one of the neighbouring sample points of the filter is applied to a corresponding sample point of the signal; ii) offset and neighbouring sample points of the signal are defined either side of the corresponding sample point, said offset points being offset in the time domain relative to the respective neighbouring sample points of the filter; and iii) the neighbouring sample points of the filter are applied to respective of the offset and neighbouring sample points of the signal.

A signal acquisition circuit for acquiring data of an input signal comprising at least n acquisition units, wherein n is integer greater than one, the n acquisition units comprising k inputs, wherein k is integer greater than one, and wherein at least two inputs are assigned to one channel and the corresponding acquisition units run time interleaved, and at least one trigger unit, wherein the number 1 of the at least one trigger unit is integer and wherein 1 is smaller than k. Further, a single-housed device as well as a method of acquiring data of an input signal are described.

A method (500) for resampling an audio signal (110) is described. The method (500) comprising providing (501) a set of input subband signals (210) which is representative of a time domain audio signal. Furthermore, the method (500) comprises applying (502) a first ripple pre-emphasis gain (323) to a first input subband signal (210) of the set of input subband signals (210) to determine a corresponding first output subband signal (213) of a set of output subband signals (213). In addition, the method (500) comprises determining (503) a time domain input audio signal (110) from the set of output subband signals (213). The method (500) further comprises performing (504) time domain resampling of the input audio signal (110) to provide an output audio signal (113) using an anti-aliasing filter (102), wherein the first ripple pre-emphasis gain (323) is dependent on a frequency response (311) of the anti-aliasing filter (102), such that an amplitude ripple of the frequency response (311) of the anti-aliasing filter (102) is at least partially compensated by the first ripple pre-emphasis gain (323).

An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes n analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.