摘要:
A method of recovering data from one or more failed data sectors includes estimating a reader offset position from a first or a second read attempt of the one or more failed data sectors at a current set of channel parameters and basing the estimated reader offset position on, at least in part, a position error signal generated during the first or second read attempt. At least one read is performed on the one or more failed data sectors at the estimated reader offset position to obtain one or more samples. The one or more samples are processed to obtain a processed sample. Iterative outer code recovery is performed on the processed sample.
摘要:
Systems and techniques to interpret signals on a noisy channel. A described system includes a filter, buffer, detector, controller, and averager. The buffer can store a group of signals, including a filtered digital signal and previous signal(s). The controller can determine whether first discrete values are adequately indicated and initiate a retry mode when the first discrete values are not adequately indicated. The averager can produce a new signal, in the retry mode, based on an average of at least a portion of the group of signals. The detector can interpret the new signal as second discrete values. The controller can determine whether the second discrete values are adequately indicated based on a measurement of differences between hard decisions indicated by the new signal and hard decisions indicated by the filtered digital signal. The controller can selectively exclude a signal of the group of signals from the average.
摘要:
Systems and techniques to interpret signals on a noisy channel. In general, in one implementation, the technique includes: interpreting an input signal as discrete values, and in response to an inadequate signal, averaging multiple signals to improve interpretation of the input signal. The input signal can be a read signal from a storage medium, such as those found in disk drives. A read channel can include a buffer and an averaging circuit capable of different signal averaging approaches in a retry mode, including making signal averaging decisions based on a signal quality measure. Buffering read signals can be done in alternative locations in the read channel and can involve buffering of many prior read signals and/or buffering of an averaged read signal.
摘要:
A method of detecting a cycle slip in a data string read from a bit patterned media and adjusting the data string to compensate for the cycle slip is disclosed. A system including a writeable data storage medium and a compensator configured to adjust data read from the data storage medium to compensate for a cycle slip during the writing of the data onto the storage medium is also disclosed.
摘要:
An embodiment of a read channel includes a filter, an interpolator, a recovery circuit, an error detector, a reverse interpolator, and a filter calibrator. The filter is operable to receive a raw sample of a signal and a coefficient-correction value, generate a filtered sample from the raw sample and a pre-established coefficient, and change the coefficient in response to the coefficient-correction value. The interpolator is operable to interpolate the filtered sample, and the recovery circuit is operable to generate a data symbol from the interpolated sample. The error detector is operable to generate an ideal sample from the data symbol and to generate a difference between the ideal sample and the interpolated sample, and the reverse interpolator is operable to reverse interpolate the difference. The filter calibrator is operable to receive the raw sample and to generate the coefficient-correction value from the raw sample and the reverse-interpolated difference.
摘要:
A recording target error rate is selected for one or more of data storage devices, and for each data storage device, a dither value is determined for each read/write head in the data storage device, wherein for each head, using a dither value for writing data, essentially provides the selected recording target error for all the heads.
摘要:
An audio signal processing method for repairing an anomalous state such as noise, a discontinuity, and a break of sound, comprising detecting the anomalous state of an audio signal, deleting the audio signal in the anomalous segment, deducing the correct audio signal by referring to the waveform of the audio signal before and after the deleted segment, generating a repair signal for repairing the signal in the deleted segment based on the deduced result, inserting the repair signal into the deleted segment, and connecting it to the audio signal before and after the deleted segment.
摘要:
An audio signal processing method for repairing an anomalous state such as noise, a discontinuity, and a break of sound, comprising detecting the anomalous state of an audio signal, deleting the audio signal in the anomalous segment, deducing the correct audio signal by referring to the waveform of the audio signal before and after the deleted segment, generating a repair signal for repairing the signal in the deleted segment based on the deduced result, inserting the repair signal into the deleted segment, and connecting it to the audio signal before and after the deleted segment.
摘要:
An audio signal processing method for repairing an anomalous state such as noise, a discontinuity, and a break of sound, comprising detecting the anomalous state of an audio signal, deleting the audio signal in the anomalous segment, deducing the correct audio signal by referring to the waveform of the audio signal before and after the deleted segment, generating a repair signal for repairing the signal in the deleted segment based on the deduced result, inserting the repair signal into the deleted segment, and connecting it to the audio signal before and after the deleted segment.
摘要:
Data to be compressed is sampled at a time interval of a sampled point where an error is at a desired value or smaller between a data value on a straight line for connecting two pieces of sampled data and a sampled data value corresponding to the data value, and a pair of discrete amplitude data D1, D6, . . . , on each sample point and timing data indicative of a time interval between sample points is obtained as compressed data. Hence, from a large number of pieces of sampled data D1, D2, . . . , included in data to be compressed, only a pair of amplitude data and timing data can be obtained as compressed data on a sample point where an error from original data is not large even when linear interpolation is performed during expansion. Thus, it is possible to remarkably improve the quality of reproduced data while achieving high compressibility.