Abstract:
This invention relates to Analog to Digital Converters (ADC) and, inter alia, to Time Interleaved ADCs and Successive Approximation Register (SAR) ADC's. In a conventional Time Interleaved ADC employing SAR ADC units, the input signal is processed through a track-and-hold circuit (T/H), and then through a buffer circuit, before the SAR ADC unit. There, by means of a comparator, the signal is compared with a Digital-to-Analog Converter (DAC) signal from the SAR logic. The buffer reduces the influence of capacitive loading and physical layout design on the SAR ADC input, but typically has a non-linear response and thus introduces distortion to the input signal. This can limit the ADC linearity, particularly for high-speed ADCs operating with low-supply voltages. An objective of the invention is to reduce or eliminate the effect of the buffer non-linearity. This is done in some embodiments by routing both the signals to the comparator through the same buffer circuit. In another embodiment the DAC signal is routed through a separate second buffer circuit. By use of a single buffer circuit, or where there is ideal matching of the buffer circuits in the latter embodiment, the distortion effects are completely eliminated; however, for practical imperfectly matched buffer circuits according to the latter embodiment, the gain and off-set mismatches can be accommodated through calibration of the buffers or, in suitable applications, through the DAC calibration.
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 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 roll-journal bearing mounting is provided by a movable bearing housing from which pistons rigidly extend in opposite directions into cylinders rigidly fixed against displacement. Introduction of pressurized fluid to the cylinders moves the bearing housing either way in the direction of the pistons and cylinders. Consequently, a roll journaled by bearings in the bearing housings of two of the mountings can be moved towards and from a counter roll to apply roll nip pressure. At the same time the rigidly extending pistons in their rigidly fixed cylinders also function to hold the journal loading in a right angular direction, eliminating the need for the use of separate devices for that purpose.
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:
The invention relates to a circuit and method for receiving a signal of which—at the receiver end—the frequency is basically unknown. By sampling the data and deriving the frequency of the signal (or actually: the data rate of the data carried by the signal) and setting a phase locked loop in the receiver to the derived—estimated—circuit, the receiver can very quickly tune in to the frequency of the signal. Hence, no embedded or accompanying clock is required for the signal. Oversampling of the signal by the receiver front end is preferred, though.
Abstract:
A device for receiving a RF signal (1; 21) with loop-through output (16) is provided. The device comprises: an input (3) receiving a RF input signal (2); an analog-digital converter (8) converting the RF input signal (2) to a digital signal (9); a digital signal processing unit (10) digitally processing the digital signal (9); a digital-analog converter (14) converting the processed digital signal (13) to a loop-through RF signal (15) corresponding to the RF input signal (2); and a loop-through output (16) outputting the loop-through RF signal (15).
Abstract:
A data communication system has a transmitter with a first clock-generation circuit, and a receiver with a second clock generation circuit. At least a specific one of the clock-generation circuits is powered-down between consecutive data bursts. The system expedites the starting up of operational use of the system upon a power-down of the specific clock-generation circuit. The system presets at a predetermined value an operational quantity of the specific clock-generation circuit at the starting up.
Abstract:
Various exemplary embodiments relate to a tracking system and method. The system includes a transistor switch having a gate node and a source node, a power source circuit connected to the gate node, and a bootstrapping circuit connected to the source node and to the gate node. The power source circuit charges the switch during a first tracking phase, and the bootstrapping circuit charges the switch during a second tracking phase.
Abstract:
A non-binary successive approximation analogue to digital converter, for converting using successive conversion steps, is operable in first and second modes. The first and second modes have different noise properties and the converter is switched between the modes during the conversion process.