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Digital Signal Processing
Published in Richard L. Shell, Ernest L. Hall, Handbook of Industrial Automation, 2000
One of the important functions that a digital signal processing system can serve is that of sample rate conversion. A sample-rate converter changes a system’s sample rate from a value of fin samples per second, to a rate of fout samples per second. Systems which contain multiple sample rates are called multirate systems. If a time series x[k] is accepted at a sample rate fin and exported at a rate fout such that fin > fout, then the signal is said to be decimated by M where M is an integer satisfying () M=foutfin
Digital audio principles
Published in Francis Rumsey, Desktop Audio Technology, 2003
Sample rate conversion is necessary whenever audio is to be transferred between systems operating at different rates. The aim is to convert the audio to the new rate without any change in pitch or addition of distortion or noise. These days sample rate conversion can be a very high-quality process, although it is never an entirely transparent process because it involves modifying the sample values and timings. As with requantising algorithms, it is fairly common to encounter poorly implemented sample rate conversion on low-cost digital audio workstations, often depending very much on the specific software application rather than the hardware involved.
Digital Representation
Published in Eddy B. Brixen, Audio Metering, 2020
Sampling frequencies of 32 kHz, 44.1 kHz, and 48 kHz have long been the standard for quality audio for things like CD or broadcast audio tracks. However, higher sampling frequencies of 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz widely apply to production environments, and high-quality audio delivery formats (see oversampling below). If the final delivery format is specified to have a lower sampling rate, down-sampling is performed (sample rate conversion).
Reconfigurable radio receiver with fractional sample rate converter and multi-rate ADC based on LO-derived sampling clock
Published in International Journal of Electronics, 2018
Sungkyung Park, Chester Sungchung Park
Sample rate conversion is usually done with the combination of digital decimation and zero-stuffing interpolation. For fractional sample rate conversion, polynomial interpolation is typically used for more configurability in converting a host of bit rates to a fixed data rate. The well-known linear interpolator can be viewed as a minimal polynomial interpolator. Linear or first-order interpolator can be extended to cubic or third-order interpolator using mathematical schemes like Lagrange method to improve accuracy or reduce the interpolation error. A cubic interpolator is chosen for our scenario in view of the trade-off between performance and complexity. A FSRC is modelled and simulated to process the multi-rate delta-sigma ADC output signal. The signals in the FSRC based on the polynomial interpolator are viewed in the spectral domain to give insight into the FSRC effect on signal aliasing, nonlinearity and SNR.