Abstract: A sigma-delta modulator (SDM) including n (n>_1) integrators in series, where a first of the n integrators receiving an input signal, at least one Q device, which acts as a quantizer when an absolute value of a signal input thereto is smaller and as a gain element (either with or without offset) when the absolute value of the signal input thereto is larger, and a device for quantizing an output of the unit. The SDM may be a feed back or feed forward SDM. The SDM may include a single or multiple Q devices. The single Q device may be positioned so that the signal input to the one Q device is an output of the last integrator and the output of the one device Q, is input to the device for quantizing and/or to the n integrators. For multiple Q devices, each of the Q devices may have different parameters set to improve stability, improve SNR, and/or reduce introduction of artifacts. The SDM may be part of an analog to digital converter and/or a digital to digital converter.

Abstract: A system (10) and method that generate bit-streams that result in higher compression gains. The system is akin to a normal 1-bit SDM. Internally, the system (10) tries to find the best possible bit sequence by tracing N possible solutions at every time instant. In an implementation, the system has N>I trellis path structures (20). Every path is used to track a possible output bitstream. The quality of a bitstream is determined by measuring the (frequency weighted) difference between input and output; it is this measure that is reduced or minimized.

Abstract: A method of concealing errors in a single bit bitstream comprises low pass filtering the bitstream, replacing the low pass filtered signal during an error by a low frequency approximation of the signal and subsequently converting the signal by a ??-modulator into a regenerated single bit bitstream. During the absence of an error the original bitstream may be outputted and during an error the regenerated bitstream, which is obtained from a ??-modulator, which is bit-synchronized to the original bitstream.

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: Sigma-delta modulation is provided, comprising feeding an input signal to a first SDM, subtracting the output of the first SDM from the input signal, filtering the output of the subtracting to obtain a filtered signal, delaying the input signal, adding the filtered signal to the delayed signal, feeding the output of the adder to a second SDM and providing the output of the second SDM. The first SDM in combination with the subtracting and filtering delivers a correction signal which, by adding it to the input signal, reduces the distortion in the second SDM, which second SDM performs in fact the sigma-delta modulation of the device.

Abstract: A mixing system for mixing a plurality of digital audio signals, at least one of which is a noise-shaped oversampled digital audio signal having a predetermined sampling frequency and bit resolution, said system comprising a summing unit for computing a sum signal of said plurality of input signals; a clipping unit having an input for receiving said sum signal, said clipping unit clipping said sum signal; further comprising a filter unit between the input terminals and the clipping unit, arranged to selectively suppress frequency components outside an audio frequency band from the sum signal; and a converter unit arranged to receive a clipped sum signal from the clipping unit and to convert said clipped sum signal into an output signal of said bit resolution, using noise-shaping, the clipping unit being arranged to limit the input values to a range of values that the converter is able to handle in a stable manner.

Abstract: Sigma-delta modulation is provided, comprising feeding an input signal to a first SDM, subtracting the output of the first SDM from the input signal, filtering the output of the subtracting to obtain a filtered signal, delaying the input signal, adding the filtered signal to the delayed signal, feeding the output of the adder to a second SDM and providing the output of the second SDM. The first SDM in combination with the subtracting and filtering delivers a correction signal which, by adding it to the input signal, reduces the distortion in the second SDM, which second SDM performs in fact the sigma-delta modulation of the device.

Abstract: Estimating a compression gain obtainable in compressing a given audio signal, comprising extracting a signal power in a selected frequency band of the given audio signal, and obtaining an estimation of the compression gain by correlation with the extracted signal power.

Abstract: Estimating a compression gain obtainable in compressing a given audio signal, comprising extracting a signal power in a selected frequency band of the given audio signal, and obtaining an estimation of the compression gain by correlation with the extracted signal power.

Abstract: A high-quality signal, for example, a unity-bit coded (DSD) audio signal having a 2.822 MHz bit rate (fs1) is converted to a lower sample rate (fs3) PCM signal by a sample rate converter (102). The watermark is embedded at the lower sample rate by a conventional watermark embedder (101) which is capable of handling signals at this lower rate. The watermark (WM) is not otherwise available. It is subsequently retrieved by subtracting (103) the unwatermarked signal from the watermarked signal, and up-sampled (104) to an intermediate sampling rate (fs2). The DSD signal is converted (110) to a PCM signal at said intermediate sample rate. The retrieved watermark is then added (107) to the PCM signal and the watermarked PCM signal is back-converted (120) to a unity-bit coded DSD signal. The arrangement preferably includes a compensation circuit (105) which compensates the information signal X′ for any (e.g.

Abstract: A method and arrangement for bit-synchronizing a &Sgr;&Dgr;-modulator uses a pre-filter to filter an incoming single bit bitstream. One or more of the integrator states of the &Sgr;&Dgr;-modulator are corrected by a signal which is calculated from the incoming bitstream and at least one of the pre-filtered input signal and the output bitstream of the &Sgr;&Dgr;-modulator.

Abstract: A robust watermark embedded into a Direct Stream Digital (DSD) audio signal including a flat frequency response in a specific frequency range which does not extend below 20 kHz or above 100 kHz. The watermark is therefore hidden in the noise spectrum of the DSD signal, such that the watermark is inaudible to a listener. Since the noise spectrum contains important information that helps provide the DSD signals with sharp transients and an accurate impulse response, the watermark cannot be removed from the DSD signal without causing significant degradation to the signal's audio quality.

Abstract: Processing an audio signal is provided, which processing comprises conversion of the audio signal into a digital signal by a noise-shaping modulation, compressive encoding of the digital signal at a predetermined sampling rate into a compressed digital signal, and supplying the compressed digital signal, wherein the noise-shaping modulation is adaptive in response to at least one parameter.

Abstract: Estimating a compression gain obtainable in compressing a given audio signal, comprising extracting a signal power in a selected frequency band of the given audio signal, and obtaining an estimation of the compression gain by correlation with the extracted signal power.

Abstract: A digital low pass filter of the FIR type stores the input signal into register means with a plurality of taps. These taps are connected to address a look up table and the filtered output signal is derived from the output of the look up table. The filter may e.g. be used in systems where a one-bit direct stream digital (DSD) signal has to be requantized.

Abstract: A method of concealing errors in a single bit bitstream comprises low pass filtering the bitstream, replacing the low pass filtered signal during an error by a low frequency approximation of the signal and subsequently converting the signal by a &Sgr;&Dgr;-modulator into a regenerated single bit bitstream. During the absence of an error the original bitstream may be outputted and during an error the regenerated bitstream, which is obtained from a &Sgr;&Dgr;-modulator, which is bit-synchronized to the original bitstream.

Abstract: In determining an intrinsic spectrum from a measured spectrum using the Maximum Entropy Algorithm, it is hardly or even not at all practical to determine the eigenvalues of an N×N matrix of large dimensions (N of the order of from 104 to 105). According to the invention such a large matrix is subdivided into a large number of much smaller partial matrices that are located on the diagonal or trace of the large matrix. The set of eigenvalues to be determined then consists of all eigenvalues of the partial matrices which can be determined much faster. Because of the Toeplitz-like character of the partial matrices, their eigenvalues can be determined very fast by Fourier transformation of a single row of such a matrix. Using the set of eigenvalues thus obtained, the intrinsic spectrum is determined by means of a minimizing algorithm.

Abstract: A method and arrangement for bit-synchronizing a &Sgr;&Dgr;-modulator with pre-filter to an incoming single bit bitstream. One or more of the integrator states of the &Sgr;&Dgr;-modulator are corrected by a signal which is calculated from the incoming bitstream and at least one of the pre-filtered input signal and the output bitstream of the &Sgr;&Dgr;-modulator.

Abstract: A high-quality signal, for example a unity-bit coded (DSD) audio signal having a 2.822 MHz bit rate (fs1) is converted to a lower sample rate (fs3) PCM signal by means of a sample rate converter (102). The watermark is embedded at the lower sample rate by a conventional watermark embedder (101) which is capable of handling signals at this lower rate. The watermark (WM) is not otherwise available. It is subsequently retrieved by subtracting (103) the unwatermarked signal from the watermarked signal, and up-sampled (104) to an intermediate sampling rate (fs2). The DSD signal is converted (110) to a PCM signal at said intermediate sample rate. The retrieved watermark is then added (107) to the PCM signal and the watermarked PCM signal is back-converted (120) to a unity-bit coded DSD signal.

Abstract: A method of determining the concentrations of the constituents in a mixture of substances by way of an X-ray diffractogram of the mixture. The fundamental difficulty that it is not possible to determine the entire power spectrum (PS) of the diffraction is avoided by making a suitable estimate of the PS on the basis of the diffractions that can be observed. Using an estimate of the dispersive power of the individual atoms in the unity cells of the constituents and the PS, the absolute intensities are determined from the relative intensities and on the basis thereof the concentrations of the constituents in the mixture are determined.