TY - GEN
T1 - Bandwidth extension of audio based on partial loudness criteria
AU - Berisha, Visar
AU - Spanias, Andreas
PY - 2006/1/1
Y1 - 2006/1/1
N2 - Most modern speech coders operate on a limited bandwidth. This tends to decrease the naturalness of the synthesized audio and often also affects the intelligibility of certain sounds. While a few wideband speech coders have been standardized, implementing them in existing systems would require significant changes to the infrastructure. One solution is to use bandwidth extension techniques that predict the high-frequency band based on low-band features. Problems arise however when the correlation between the low and the high band is insufficient for an adequate representation of the wideband signal. In this paper, we propose a novel source-filter bandwidth extension algorithm that makes use of psychoacoustic concepts to determine the perceptual benefits that a particular audio frame gains from a more exact representation of the high band. Preliminary results indicate that the proposed system performs at a lower average bit rate when compared to other similar algorithms without compromising the audio quality.
AB - Most modern speech coders operate on a limited bandwidth. This tends to decrease the naturalness of the synthesized audio and often also affects the intelligibility of certain sounds. While a few wideband speech coders have been standardized, implementing them in existing systems would require significant changes to the infrastructure. One solution is to use bandwidth extension techniques that predict the high-frequency band based on low-band features. Problems arise however when the correlation between the low and the high band is insufficient for an adequate representation of the wideband signal. In this paper, we propose a novel source-filter bandwidth extension algorithm that makes use of psychoacoustic concepts to determine the perceptual benefits that a particular audio frame gains from a more exact representation of the high band. Preliminary results indicate that the proposed system performs at a lower average bit rate when compared to other similar algorithms without compromising the audio quality.
UR - http://www.scopus.com/inward/record.url?scp=34250724837&partnerID=8YFLogxK
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U2 - 10.1109/MMSP.2006.285286
DO - 10.1109/MMSP.2006.285286
M3 - Conference contribution
AN - SCOPUS:34250724837
SN - 0780397517
SN - 9780780397514
T3 - 2006 IEEE 8th Workshop on Multimedia Signal Processing, MMSP 2006
SP - 146
EP - 149
BT - 2006 IEEE 8th Workshop on Multimedia Signal Processing, MMSP 2006
PB - IEEE Computer Society
T2 - 2006 IEEE 8th Workshop on Multimedia Signal Processing, MMSP 2006
Y2 - 3 October 2006 through 6 October 2006
ER -