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 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.

As part of a signal processing event, the maximum frequency of an input signal can be determined with a processor. The maximum frequency can be compared to a value generated with a decimator/interpolator. Based on the comparison, the sampling rate for sampling the input signal with the processor can be set as part of the digital signal processing event. The sampling rate can be adjusted as the frequency of the input signal varies during the signal processing event.

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.

A method, a computer-readable medium, and an apparatus for resampling audio signal are provided. The apparatus resamples the audio signal in order to preserve the audio playback quality when dealing with audio playback overrun and underrun problem. The apparatus may receive a data block of the audio signal including a first number of samples. For each sample of the first number of samples, the apparatus may slice a portion of the audio signal corresponding to the sample into a particular number of sub-samples. The apparatus may resample the data block of the audio signal into a second number of samples based on the first number of samples and the particular number of sub-samples associated with each sample of the first number of samples. The apparatus may play back the resampled data block of the audio signal via an electroacoustic device.

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 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.

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 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 1st frequency reduction circuit of a filter of the invention downsamples the sampling rate of a signal source to a predetermined value to obtain a 1st PCM stream, a 1st frequency raising circuit raises the sampling rate of the 1st PCM stream to be the same as that of the signal source, a 1st delay circuit delays a stream of the signal source, such that its phase is the same as that of the 1st PCM stream, a 1st adder subtracts the frequency raised 1st PCM steam from the delayed stream of the signal source to obtain a passband 1, a j-th frequency reduction circuit downsamples the sampling rate of a (j1)-th PCM stream to a predetermined value to obtain a j-th PCM stream, wherein 2jn, a j-th frequency raising circuit raises the sampling rate of the j-th PCM stream to be the same as that of the (j1)-th PCM stream, a j-th delay circuit delays the (j1)-th PCM stream, such that its phase is the same as that of the j-th PCM stream, a j-th adder subtracts the frequency raised j-th PCM stream from the delayed (j1)-th PCM stream to obtain a passband j, and when j=n, the j-th PCM stream is a passband n+1.