Abstract:
An analog-to-digital converter comprises a signal input (6) for receiving an analog input signal and a set of comparators (4). Each comparator (4) has a first input (21 ) connected to the signal input (6) and a second input (22) connected to a reference voltage (16). Each comparator generates an output based on the comparison of the signals at the first input (21 ) and second input (22). The reference voltage is the same for all comparators. The set of comparators (4) has a non-identical response to the reference voltage (16) and the input signal and is due to an internally arising offset. An adder (25) determines a sum of the outputs of the set of comparators and conversion logic (27) generates an output digital signal dependent on the determined sum. Multiple sets of comparators can be provided, each set having a different respective reference voltage.
Abstract:
A digital signal processing circuit comprises a band selector (14) for selecting at least one sub-band from a frequency spectrum of a digital sampled input signal. The band selector (14) comprises a plurality of processing branches corresponding to respective phases and an adder (28a, 28b) for adding branch signals from the branches. Each branch comprises a sub-sampler (20a,b) for sub-sampling sample values of the input signal at the phase corresponding to the branch, a filter (24a,b) with a first FIR filter (32, 34), applied alternatingly to sets of even and to sets of odd samples from the subsampler (20a,b) and a second FIR filter (36, 38) applied to further sets of odd and even samples from the subsampler (20a,b) when the first FIR filter is applied to the even and odd sets respectively. Output samples from the first and second FIR filter (24a,b) are combined to form the branch signals of the branch, according to a changing combination pattern that changes cyclically as a function of sample position and depends on a phase for which the branch is used.
Abstract:
An audio coding scheme allowing PCM signal to lossless DSD signal expansion for next generation optical disc formats. The method of encoding an input DSD signal includes up-sampling a corresponding PCM signal to the DSD sample rate. Then a set of loop filter parameters for a noise-shaping loop of a sigma-delta modulator are generated, either using a random starting condition of the sigma-delta modulator or including synchronization parameters. This will allow a decoder to regenerate an almost perfect signal, but still it needs a correction signal to be able to bit identically regenerate the DSD input signal. Therefore, a correction signal is generated based on a difference between a sigma- delta modulated version of the up-sampled PCM signal and the input DSD signal, wherein the sigma-delta modulated version of the up-sampled PCM signal is obtained using the set of loop filter parameters. The correction signal may be adapted to be applied to the low bit PCM signal, to the up-sampled PCM signal or to the output bit stream. Finally, an expansion bit stream is generated where an encoded version of the set of loop filter parameters and an encoded version of the correction signal are included. The decoder can reproduce the original DSD signal based on the already available PCM signal and the described expansion bit stream. Thus, the coding scheme enables top quality audio with minimal storage overhead since the already available PCM signal is used in combination with an expansion bit stream. Since only an additional data stream is required to be stored on a disc, e.g. as part of an MPEG stream, DSD functionality is added to existing systems without causing compatibility problems.
Abstract:
A device for receiving an RF input signal (2) and for processing the received RF input signal (2) is provided. The device comprises: an input (3) receiving the RF input signal (2); a clock circuit (34) generating a reference clock signal having a clock circuit specific reference frequency (fclock); and a frequency feedback control loop (17, 18, 20, 21, 22, 23; 35, 37, 38). The frequency feedback control loop is adapted to extract frequency information from the RF input signal (2), to put the clock circuit specific reference frequency (fclock) into relation with the extracted frequency information and to correct for inaccuracies of the clock circuit specific reference frequency based on this relation.
Abstract:
A device for receiving a RF signal (1) is provided. The device comprises an input (3) receiving a RF input signal (2); an analog pre-processing circuitry (11) pre- processing the RF input signal (2); an analog-digitalconverter (8) converting the pre- processed RF input signal to a digital signal (9); and a digital signal processing unit (10) digitallyprocessing the digital signal (9). The digital signal processing unit (10) is adapted to compensate signaldistortions introduced by the analog pre-processing circuitry (11).
Abstract:
An audio coding scheme allowing PCM signal to lossless DSD signal expansion for next generation optical disc formats. The method of encoding an input DSD signal includes up-sampling a corresponding PCM signal to the DSD sample rate. Then a set of loop filter parameters for a noise-shaping loop of a sigma-delta modulator are generated, either using a random starting condition of the sigma-delta modulator or including synchronization parameters. This will allow a decoder to regenerate an almost perfect signal, but still it needs a correction signal to be able to bit identically regenerate the DSD input signal. Therefore, a correction signal is generated based on a difference between a sigma- delta modulated version of the up-sampled PCM signal and the input DSD signal, wherein the sigma-delta modulated version of the up-sampled PCM signal is obtained using the set of loop filter parameters. The correction signal may be adapted to be applied to the low bit PCM signal, to the up-sampled PCM signal or to the output bit stream. Finally, an expansion bit stream is generated where an encoded version of the set of loop filter parameters and an encoded version of the correction signal are included. The decoder can reproduce the original DSD signal based on the already available PCM signal and the described expansion bit stream. Thus, the coding scheme enables top quality audio with minimal storage overhead since the already available PCM signal is used in combination with an expansion bit stream. Since only an additional data stream is required to be stored on a disc, e.g. as part of an MPEG stream, DSD functionality is added to existing systems without causing compatibility problems.
Abstract:
An adaptive filtering device and method where at least one adaptive filter receives an input signal and a metering device receives an output of the at least one adaptive filter, monitors a characteristic of the output, such as power of high-frequency components, and forwards a correction signal in a feedback loop to adjust the characteristic of the at least one filter in order to change the characteristic of the output. An adaptive filtering device (200) and method where two low-pass FIR filters (202, 204) receive an input signal, a weighted adder (206) receives outputs from the two low-pass FIR filters and changes a weighting of each to produce filtered output data, and a controller (208) that receives a cutoff frequency (203), supplies the cut-off frequency to the two low-pass FIR filters, and supplies a signal to the weighted adder for varying the weighting of each of the low-pass FIR filters to switch therebetween.
Abstract:
Sigma-delta modulation is provided, wherein an input signal is feeded to at least two parallel filters, a first one of the filters preferably being a lower order filter and a second one of the filters preferably being a higher order filter, wherein output of the filters are weighted and wherein the weighted output from the at least two filters is quantized, in order to enable a sigma-delta modulation with variable order.
Abstract:
A multi-channel receiver comprising an ADC and a multi-band, multi-channel selector. The ADC converts a broad-band multi-channel signal into a digital signal. The digital signal is then broken into sub-bands each containing a plurality of channels. A channel selector selects desired channels from the appropriate sub-band. The multi-channel receiver may deliver simultaneous channels equal to the number of channel selectors that have been implemented. The multi-channel receiver may be implemented on a single integrated circuit.
Abstract:
A time-interleaved signal converter (800) comprising a plurality of analogue-to- digital converters (ADCO-3), hereinafter termed ADCs, the ADCs being configured to sample an input signal at a common sampling rate and at differing phases to produce a corresponding plurality of digital signal outputs, the signal converter (800) being configured to produce a combined digital signal output from a combination of the plurality of digital signal outputs, wherein the signal converter (800) is configured to determine a sampling timing error (ΔT) between a pair of the ADCs by comparing an autocorrelation (710) of the combined digital signal output with a cross-correlation (720) of a respective pair of the plurality of digital signal outputs.