Abstract: The present invention relates to the use of a new polymeric aqueous emulsion for surface treating particles of mineral matter.

Abstract: A magnetic recording device includes a preprocessor, an interpolator and a slicer. The preprocessor receives at least n saturated input signals including an nth saturated input signal The preprocessor is configured to process each of the n saturated input signals to produce a corresponding nth output signal. The n output signals include an nm output signal from the nth saturated input signal. The interpolator processes the nth output signal to determine an nth interpolator output. The slicer determines an nth slicer output for the nh output signal. The nth slicer output is at one of three different levels. The preprocessor can receive and process an n +1th saturated input signal to produce an n +1th output signal that is based on a difference between the nth interpolator output and the level of the nth slicer output.

Abstract: The present invention relates to the use of a new polymeric aqueous emulsion for surface treating particles of mineral matter.

Abstract: In one embodiment, a read channel comprises: a preprocessor for receiving a first signal and producing a second signal from the first signal using current values of a positive coefficient, a zero coefficient, and a negative coefficient; an interpolator for producing a third signal based on the second signal; and a slicer for producing a fourth signal from the third signal by estimating a level for the third signal. The fourth signal is at one of three levels consisting of a positive level, a zero level, and a negative level. For every n first signals received by the preprocessor, the current value of one of the positive coefficient, the zero coefficient, and the negative coefficient is adjusted depending on which of the three levels the fourth signal is at.

Abstract: In one embodiment, a read channel comprises: a preprocessor for receiving a first signal and producing a second signal from the first signal using current values of a positive coefficient, a zero coefficient, and a negative coefficient; an interpolator for producing a third signal based on the second signal; and a slicer for producing a fourth signal from the third signal by estimating a level for the third signal. The fourth signal is at one of three levels consisting of a positive level, a zero level, and a negative level. For every n first signals received by the preprocessor, the current value of one of the positive coefficient, the zero coefficient, and the negative coefficient is adjusted depending on which of the three levels the fourth signal is at.

Abstract: A digital data recovery system for converting a suboptimal signal into a converted signal that closely approximates an original signal includes a first data filter, a first interpolator and a second interpolator. The first data filter filters the suboptimal signal to generate a first filtered signal. The first interpolator receives the first filtered signal and generates a first interpolated signal. Substantially concurrently, the second interpolator receives the suboptimal signal and generates a second interpolated signal. The digital data recovery system may further comprise a second data filter that receives the second interpolated signal and generates a second filtered signal. Further, the first data filter can include a set of first coefficients and the second data filter can include a set of second coefficients. Moreover, the second coefficients can be updated and subsequently transformed in order to update the first coefficients.

Abstract: In one embodiment, a read channel comprises: a preprocessor for receiving a first signal and producing a second signal from the first signal using current values of a positive coefficient, a zero coefficient, and a negative coefficient; an interpolator for producing a third signal based on the second signal; and a slicer for producing a fourth signal from the third signal by estimating a level for the third signal. The fourth signal is at one of three levels consisting of a positive level, a zero level, and a negative level. For every n first signals received by the preprocessor, the current value of one of the positive coefficient, the zero coefficient, and the negative coefficient is adjusted depending on which of the three levels the fourth signal is at.

Abstract: In one embodiment, an apparatus comprises an adaptive filter, a timing recovery unit, and a reverse interpolation filter. The adaptive filter has adaptive filter coefficients that are adjusted based on a first error signal at a first sample rate and filters a first signal at the first sample rate to obtain a second signal at the first sample rate. The timing recovery unit interpolates the second signal at the first sample rate to obtain a third signal at a second sample rate; and estimates a partial response signal at the second sample rate corresponding to the third signal. The a reverse interpolation filter interpolates a second error signal at the first sample rate, which is a difference between the third signal and the partial response signal, to obtain the first error signal at the first sample rate for feeding back to the adaptive filter.

Abstract: A digital data recovery system (14) for converting a suboptimal signal (18) into a converted signal (20) that closely approximates an original signal (16) includes a first data filter (22), a first interpolator (26) and a second interpolator (36). The first data filter (22) filters the suboptimal signal (18) to generate a first filtered signal (48). The first interpolator (26) receives the first filtered signal (48) and generates a first interpolated signal (52). Substantially concurrently, the second interpolator (36) receives the suboptimal signal (18) and generates a second interpolated signal (64). The digital data recovery system (14) may further comprise a second data filter (38) that receives the second interpolated signal (64) and generates a second filtered signal (66). Further, the first data filter (22) can include a set of first coefficients (50) and the second data filter (38) can include a set of second coefficients (68).

Abstract: In one embodiment, an apparatus comprises an adaptive filter, a timing recovery unit, and a reverse interpolation filter. The adaptive filter has adaptive filter coefficients that are adjusted based on a first error signal at a first sample rate and filters a first signal at the first sample rate to obtain a second signal at the first sample rate. The timing recovery unit interpolates the second signal at the first sample rate to obtain a third signal at a second sample rate; and estimates a partial response signal at the second sample rate corresponding to the third signal. The a reverse interpolation filter interpolates a second error signal at the first sample rate, which is a difference between the third signal and the partial response signal, to obtain the first error signal at the first sample rate for feeding back to the adaptive filter.

Abstract: In one embodiment, an apparatus comprises two or more slave channels, wherein each one of the slave channels comprises a non-adaptive filter operable to filter an input signal to the slave channel using filter coefficients received from a master channel; and the master channel coupled to each one of the slave channels, wherein the master channel comprises an adaptive filter operable to: for each one of one or more of the slave channels, determine the filter coefficients for the slave channel using the input signal to the slave channel; and send the filter coefficients to the slave channel.

Abstract: Timing recovery in partial-response-based magnetic recording systems customarily employs the “decision-directed” method wherein phase error is recovered from the differences between the noise-corrupted received signal samples and their estimated ideal (noise and phase error free) values. The filtered phase error drives a numerically-controlled oscillator which determines the instants at which the signal is resampled, attempting to place said instants at the ideal sampling times. The resampled signal contains errors due to mistiming as well as to the original corrupting noise, and these errors directly influence the success of subsequent detection. However, the noise can be reduced using adaptive linear prediction, having the effect of reducing the output error for a given noise input, or maintaining the same error for a larger noise input.

Abstract: A circuit for a high-density data recording channel includes a first data detector, a second data detector, one or more multiplexers and a sequence identifier. The first data detector generates a first data detector output, and the second data detector generates a second data detector output. The multiplexers change between a first mode and a second mode to alternately receive the first data detector output and the second data detector output. The sequence identifier receives a data sequence including at least one of a first data sequence, such as VFO data, and a second data sequence, such as random data. The second data sequence includes a greater number of signal levels than the first data sequence. The sequence identifier changes the multiplexers between the first mode and the second mode based on whether the data sequence is the first data sequence or the second data sequence. The data sequence includes a plurality of timing stages.

Abstract: The present invention, in particular embodiments, is directed to methods, apparatuses and systems that provide global timing error information derived from timing error information of each data channel. This is achieved, in part, by summing the timing error information from all the data channels and integrating and scaling the resulting sum. The integrated, scaled sum is then added to the proportional and integral timing information of each individual data channel. By doing so, incorrect timing error estimates are averaged out. Additionally, when severe noise and dropouts (loss of data signal) at an individual data channel occur, that channel may rely on the global timing error information. In some implementations, that individual data channel's timing error information contribution can be excluded from the global timing error information.

Abstract: An estimator of the noiseless output of a noisy partial response channel is described. The estimator operates recursively. In each iteration, the estimator processes a window of the N most recently received noisy channel outputs to compare output level metrics for all possible channel output level, and selects a noiseless output level with maximal posterior probability.

Abstract: A method and device for detecting a peak which is substantially the same as the actual peak are disclosed. In one embodiment, the device includes a filter, a shift register, a controller and a digital interpolator. The filter is configured to receive a plurality of signal samples and the shift register, which is coupled with the filter, has multiple registers. The shift register is configured to receive the plurality of signal samples and to shift the plurality of signal samples through the registers. The controller is coupled with the shifter register and is configured to detect a zero-crossing event in the signal samples. The digital interpolator is coupled with the controller and configured to perform a binary search to identify a peak substantially the same as the actual peak.

Abstract: The claimed embodiments provide methods, apparatuses and systems directed to run-length limited (RLL) coding of data. In one implementation, concatenatable RLL codes with run lengths of zeroes not exceeding k are constructed for any rate N/(N+1) where N?2k?2+k?1. As code rates increase, the value of k departs from the minimum possible value more slowly than that of many other codes. Further, occurrences of k-bit run lengths occur only at the juncture of two codewords. Due to this, the codes are mostly k?1. This quality makes the codes ideal for parity bit insertion applications such as LDPC channels. The method, in one implementation, places the bit addresses of violating sequences in a table at the beginning of the codeword, and the user data, occupying the locations where the table entries are placed, are moved into the locations of the violating sequences. This is done iteratively and in a way which provides for cases in which the violating sequence is inside the address table itself.

Abstract: Timing recovery in partial-response-based magnetic recording systems customarily employs the “decision-directed” method wherein phase error is recovered from the differences between the noise-corrupted received signal samples and their estimated ideal (noise and phase error free) values. The filtered phase error drives a numerically-controlled oscillator which determines the instants at which the signal is resampled, attempting to place said instants at the ideal sampling times. The resampled signal contains errors due to mistiming as well as to the original corrupting noise, and these errors directly influence the success of subsequent detection. However, the noise can be reduced using adaptive linear prediction, having the effect of reducing the output error for a given noise input, or maintaining the same error for a larger noise input.

Abstract: An estimator of the noiseless output of a noisy partial response channel is described. The estimator operates recursively. In each iteration, the estimator processes a window of the N most recently received noisy channel outputs to compare output level metrics for all possible channel output level, and selects a noiseless output level with maximal posterior probability.

Abstract: The claimed embodiments provide methods, apparatuses and systems directed to run-length limited (RLL) coding of data. In one implementation, concatenatable RLL codes with run lengths of zeroes not exceeding k are constructed for any rate N/(N+1) where N?2k?2+k?1. As code rates increase, the value of k departs from the minimum possible value more slowly than that of many other codes. Further, occurrences of k-bit run lengths occur only at the juncture of two codewords. Due to this, the codes are mostly k?1. This quality makes the codes ideal for parity bit insertion applications such as LDPC channels. The method, in one implementation, places the bit addresses of violating sequences in a table at the beginning of the codeword, and the user data, occupying the locations where the table entries are placed, are moved into the locations of the violating sequences. This is done iteratively and in a way which provides for cases in which the violating sequence is inside the address table itself.