Method for processing a subband encoded audio data stream

Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission

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704230, 704211, 36476001, G10L 302, G10L 900

Patent

active

057941815

ABSTRACT:
A data processing system (10) is disclosed which comprises a microprocessor host (12) coupled to a decoding system (14). A host interface block (18) receives a bit stream and passes bit stream on to a system decoder block (20). The system decoder block (20) extracts the appropriate data from the bit stream and loads an input buffer (24) or an optional external buffer (26). An audio decoder block (28) retrieves the data from the input buffer (24) and generates scale factor indices, bit per code word values and subband samples which are stored in an arithmetic unit buffer (30). A hardware filter arithmetic unit block (32) retrieves the information from the arithmetic unit buffer (30) and dequantizes, transforms and filters the data to generate PCM output data which is loaded into a PCM buffer (34). The data within the PCM buffer (34) is output by a PCM output block (36) to a digital-to-analog converter (16).

REFERENCES:
Konstantinides, K., Fast Subband Filtering in MPEG Audio Coding, IEEE Signal Processing Letters, vol. 1, No. 2 Feb., 1994, pp. 26-28.
Princen, J..P., et. al., Analysis/Synthesis Filter Bank Design Based on Time-Domain Aliasing Cancellation, IEEE Tr. Acoustics, Speech and Signal Processing, vol. 34, 1986, pp. 1153-1161.

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