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.

The present disclosure provides a resampling apparatus and a resampling method. The resampling apparatus includes a control unit, a memory device, a resolution identifier, a phase rate generator, a coefficient generator, and a resample filter. The control unit controls reading and writing operations of the resampling apparatus according to a control signal. The memory device transmits the control signal to the control unit. The resolution identifier sets a resolution bandwidth identity according to an interpolation/decimation (I/D) value of the control signal. The phase rate generator generates a phase select signal and a counter enable signal according to the resolution bandwidth identity. The coefficient generator generates a coefficient select signal according to the resolution bandwidth identity. The resample filter generates a resampled output data according to the phase select signal, the coefficient select signal, and an input data.

A sample rate converter (SRC) for implementing a rate conversion L/M is described wherein data is input to the SRC at an input rate (F.sub.in) and output from the SRC at an output rate (F.sub.out) equal to F.sub.in*L/M. The SRC includes a low pass filter (LPF) including P multiply-add instances, wherein P is a parallelization factor of the SRC; an input formatter for arranging samples received at the SRC in accordance with the rate conversion L/M and providing P*T.sub.pp input samples to the filter at a given time, wherein T.sub.pp is a number of taps per phase of the LPF; and a coefficient bank for storing a plurality of coefficients and for providing P*T.sub.pp of the coefficients to the LPF at a given time.

Provided are a filter unit and a filter array, and the filter unit includes: a control module (10) configured to analyze configuration information of the filter unit, and control selection of internal functions and flow direction of data in the filter unit; a filtering module (20) connected to the control module (10), and configured to perform an operation on internal data of the filter unit and perform a sampling rate conversion function under control of the control module (10); and an output selection module (30) connected to the control module (10) and the filtering module (20), and configured to perform a selection function on output data of the filtering module (20) under control of the control module.

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 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.

A radio frequency (RF) transmitter includes a set of input ports to receive baseband samples of a signal to be transmitted on a set of disjoint frequency bands, a set of filter banks, there is one filter bank for each input port, each filter bank includes a plurality of digital polyphase interpolation filters to sample a shifted phase of the corresponding sequence of baseband samples and to interpolate the sampled phases to produce a plurality of sequences of interpolated baseband phased samples with the shifted phase, and a set of oscillators banks, each oscillator bank includes a plurality of polyphase Digital Direct Synthesizer (DDS) corresponding to the plurality of digital polyphase interpolation filters to generate a plurality of sequences of samples of digital waveform. The RF transmitter includes a set of mixer banks to mix corresponding sequences of samples of digital waveform and interpolated baseband phased samples to up convert each sequence of interpolated baseband phased samples to the effective frequency, a parallel digital combiner to combine in-phase sequences of interpolated baseband phased samples of different frequency bands to produce a plurality of sequences of multiband upconverted samples, and a pulse encoder to modulate and encode the plurality of sequences of multiband upconverted samples to produce a plurality of encoded multi-band signals. The RF transmitter converts the plurality of encoded multi-band signals into a RF bitstream and radiate the RF bitstream as an analog signal.

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 radio frequency (RF) transmitter includes a set of input ports to receive baseband samples of a signal to be transmitted on a set of disjoint frequency bands, a set of filter banks, there is one filter bank for each input port, each filter bank includes a plurality of digital polyphase interpolation filters to sample a shifted phase of the corresponding sequence of baseband samples and to interpolate the sampled phases to produce a plurality of sequences of interpolated baseband phased samples with the shifted phase, and a set of oscillators banks, each oscillator bank includes a plurality of polyphase Digital Direct Synthesizer (DDS) corresponding to the plurality of digital polyphase interpolation filters to generate a plurality of sequences of samples of digital waveform. The RF transmitter includes a set of mixer banks to mix corresponding sequences of samples of digital waveform and interpolated baseband phased samples to up convert each sequence of interpolated baseband phased samples to the effective frequency, a parallel digital combiner to combine in-phase sequences of interpolated baseband phased samples of different frequency bands to produce a plurality of sequences of multiband upconverted samples, and a pulse encoder to modulate and encode the plurality of sequences of multiband upconverted samples to produce a plurality of encoded multi-band signals. The RF transmitter converts the plurality of encoded multi-band signals into a RF bitstream and radiate the RF bitstream as an analog signal.

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.