The DOI matches its Crossref record; 12 of the 12 references it cites were found in a registry or an index.
A novel structure for real-time digital sample-rate converters with finite precision error analysis
Abstract
A real-time implementation of a sampling rate conversion technique between two arbitrarily chosen sampling rates is discussed. A method to generate fractionally sampled data from discrete-domain (integer) data samples is analyzed and implemented on the Motorola DSP56001 24-bit digital signal processor. This real-time implementation can be accomplished with a DSP56001 and 5k of data memory. As an example, the conversion from 44.1 kHz (compact disc sampling rate) to 48 kHz (digital audio tape sampling rate) is discussed. The assembly code for the main routine and the filtering method to generate interpolated samples are included.>
Overview
Integrity signals
Registry
Cites
Statistics
Verification
Claims the paper makes
Citations
Cites
Probability, Random Variables, and Stochastic Processes.
Abrahams, J., & Papoulis, A. (1984) · Journal of the American Statistical Association · doi:10.2307/2288754 · Cited by 16,301
Time delay estimation using the LMS adaptive filter--Static behavior
Reed, F., Feintuch, P., & Bershad, N. (1981) · IEEE Transactions on Acoustics, Speech, and Signal Processing · doi:10.1109/tassp.1981.1163614 · Cited by 232
Handbook of digital signal processing: engineering applications
(1988) · Choice Reviews Online · doi:10.5860/choice.26-0940 · Cited by 164
Signals, systems and communication
Lathi, B. P. (1965) · CERN Document Server (European Organization for Nuclear Research) · Cited by 118
A flexible sampling-rate conversion method
Smith, J., & Gossett, P. (2005) · ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing · doi:10.1109/icassp.1984.1172555 · Cited by 102
On using the LMS algorithm for time delay estimation
Youn, D. H., Ahmed, N., & Carter, G. (1982) · IEEE Transactions on Acoustics Speech and Signal Processing · doi:10.1109/tassp.1982.1163961 · Cited by 95
Discrete-Time Signals and Systems
Barker, H. A. (1984) · Electronics and Power · doi:10.1049/ep.1984.0082 · Cited by 53
A universal, digital sampling frequency converter for digital audio
Lagadec, R., & Kunz, H. (2005) · IEEE International Conference on Acoustics Speech and Signal Processing · doi:10.1109/icassp.1981.1171149 · Cited by 17
Digital Techniques for Changing the Sampling Rate of a Signal
Rabiner, L. R. (1982) · Cited by 9
Park, S. (2002) · International Conference on Acoustics, Speech, and Signal Processing · doi:10.1109/icassp.1990.116045 · Cited by 8
Self-citationA Real-Time Method for Sample-Rate Conversion from CD to DAT
Park, S. (1990) · International Conference on Consumer Electronics · doi:10.1109/icce.1990.665993 · Cited by 7
Self-citationDigital Sampling Frequency Conversion
Lagadec, R. (1982) · Cited by 4
Flagged first, then by citations
Cited by
Sorted by citations
Splitting the unit delay [FIR/all pass filters design]
Laakso, T. I., Valimaki, V., Karjalainen, M., & Laine, U. K. (1996) · IEEE Signal Processing Magazine · doi:10.1109/79.482137 · Cited by 1,168
Fractional Delay Filters—Design and Applications
Välimäki, V., & Laakso, T. I. (2001) · Information Technology: Transmission, Processing, and Storage · doi:10.1007/978-1-4615-1229-5_20 · Cited by 32
$H^{\infty}$-Optimal Fractional Delay Filters
Nagahara, M., & Yamamoto, Y. (2013) · IEEE Transactions on Signal Processing · doi:10.1109/tsp.2013.2265678 · Cited by 15
“Such filters have wide applications in signal processing, including sampling rate conversion [1], [2], [3], nonuniform sampling [4], [5], wavelet transform [6], [7], digital modeling of musical instruments [8], [9], to name a few.”
Background
Metadata
- Authors
- Park, S., Hillman, G., & Robles, R.
- Affiliations
- Motorola (United States) (US)
- Venue
- [Proceedings] ICASSP 91: 1991 International Conference on Acoustics, Speech, and Signal Processing
- Publisher
- IEEE
- Published
- 1991
- Pages
- 3613-3616 vol.5
- ISSN
- 1520-6149 (ISSN-L)
- OpenAlex
- W2166592878
- Cited by
- 19
- References
- 12
- Citations by year
- 2022: 1 · 2020: 1 · 2018: 1 · 2017: 1 · 2015: 1 · 2013: 1
- FWCI
- 0.00
- Type
- conference paper
- Language
- en
- Open access
- closed
- Locations
- IEEE International Conference on Acoustics Speech and Signal Processing (publishedVersion)
- Topics
- Physical Sciences › Computer Science › Signal Processing › Digital Filter Design and Implementation
- Physical Sciences › Engineering › Electrical and Electronic Engineering › Analog and Mixed-Signal Circuit Design
- Physical Sciences › Computer Science › Signal Processing › Speech and Audio Processing
- Keywords
- sampling rate conversion, real-time digital signal processing, digital signal processor implementation, assembly language
About these checks
Registry, retraction, Cites and statistics results are computed, not judged, with one exception: a printed reference that no index lists is read once by a model before it counts as a citation error. Claims are extracted by a model and kept only when their quote appears verbatim in the text; each cited source is then checked, with a model’s help, for whether it says what the claim relies on. Those verdicts can be wrong — read the claim’s quote and the cited source before relying on one.
Statistics and arithmetic are recomputed, not judged.
Confidence bands are shown only for fields calibrated on our annotated test set. None is calibrated yet, so no band is shown.
What counts as a claim ›How we measure confidence ›How we check this ›
Are you an author, or do you see a problem with this page? Request a review
