2002•Unpublished venueRequires access

Hybrid LPC and discrete wavelet transform audio coding with a novel bit allocation algorithm

S. Boland, Mohamed A. Deriche

Open publisher page 7 citations

Abstract

This paper examines a new method for coding high quality digital audio signals based on a combination of linear predictive coding (LPC) and the discrete wavelet transform (DWT). In this method, a linear predictor is first used to model each audio frame. Then, the prediction error is analyzed using the DWT. The LPC coefficients and DWT coefficients are quantized using a novel bit allocation scheme which minimizes the overall quantization error with respect to the masking threshold. The proposed coder is capable of delivering near-transparent audio signal quality at encoding bit rates of around 90-96 kb/s. Objective and subjective results suggest that the proposed coder operating at 90-96 kb/s has a performance comparable to that of the MPEG layer II codec operating at 128 kb/s.

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What this paper is about

This paper examines a new method for coding high quality digital audio signals based on a combination of linear predictive coding (LPC) and the discrete wavelet transform (DWT). In this method, a linear predictor is first used to model each audio frame. Then, the prediction error is analyzed using the DWT. The LPC coefficients and DWT coefficients are quantized using a novel bit allocation scheme which minimizes the overall quantization error with respect to the masking threshold. The proposed coder is capable of delivering near-transparent audio signal quality at encoding bit rates of around 90-96 kb/s. Objective and subjective results suggest that the proposed coder operating at 90-96 kb/s has a performance comparable to that of the MPEG layer II codec operating at 128 kb/s.

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OpenAlex reports 7 citations for this work. Citation counts describe recorded attention and do not establish research quality.

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Available abstract

This paper examines a new method for coding high quality digital audio signals based on a combination of linear predictive coding (LPC) and the discrete wavelet transform (DWT). In this method, a linear predictor is first used to model each audio frame. Then, the prediction error is analyzed using the DWT. The LPC coefficients and DWT coefficients are quantized using a novel bit allocation scheme which minimizes the overall quantization error with respect to the masking threshold. The proposed coder is capable of delivering near-transparent audio signal quality at encoding bit rates of around 90-96 kb/s. Objective and subjective results suggest that the proposed coder operating at 90-96 kb/s has a performance comparable to that of the MPEG layer II codec operating at 128 kb/s.

Key concepts: Computer science, Adaptive Multi-Rate audio codec, Speech coding, Codec, Sub-band coding, Quantization (signal processing), Algorithm, Speech recognition

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