Details
| Original language | English |
|---|---|
| Title of host publication | 131st Audio Engineering Society Convention 2011 |
| Pages | 187-192 |
| Number of pages | 6 |
| Volume | 1 |
| Publication status | Published - 2011 |
| Event | 131st Audio Engineering Society Convention 2011 - New York, United States Duration: 20 Oct 2011 → 23 Oct 2011 |
Abstract
In this paper we present an approach for error robust audio coding at a medium data rate of about 176 kbps (mono, 44.1 kHz sampling rate). By combining a delay-free Adaptive Differential Pulse Code Modulation (ADPCM) coding-scheme and a numerically optimized low delay filter bank we achieve a very low algorithmic coding delay of only about 0.5 ms. The structure of the codec also allows for a high robustness against random single bit errors and even supports error resilience. Implementation structure, results of a listening test and PEAQ (Perceptual Evaluation of Audio Quality) based objective audio quality evaluation as well as tests of random single bit error performance are given. The presented coding-scheme provides a very good audio quality for vocals and speech. For most of the critical signals the audio quality can still be denoted as acceptable. Tests of random single bit error performance show good results for error rates up to 10 -4.
ASJC Scopus subject areas
- Mathematics(all)
- Modelling and Simulation
- Physics and Astronomy(all)
- Acoustics and Ultrasonics
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131st Audio Engineering Society Convention 2011. Vol. 1 2011. p. 187-192.
Research output: Chapter in book/report/conference proceeding › Conference contribution › Research › peer review
}
TY - GEN
T1 - Error robust low delay audio coding based on subband ADPCM
AU - Preihs, Stephan
AU - Ostermann, Jörn
PY - 2011
Y1 - 2011
N2 - In this paper we present an approach for error robust audio coding at a medium data rate of about 176 kbps (mono, 44.1 kHz sampling rate). By combining a delay-free Adaptive Differential Pulse Code Modulation (ADPCM) coding-scheme and a numerically optimized low delay filter bank we achieve a very low algorithmic coding delay of only about 0.5 ms. The structure of the codec also allows for a high robustness against random single bit errors and even supports error resilience. Implementation structure, results of a listening test and PEAQ (Perceptual Evaluation of Audio Quality) based objective audio quality evaluation as well as tests of random single bit error performance are given. The presented coding-scheme provides a very good audio quality for vocals and speech. For most of the critical signals the audio quality can still be denoted as acceptable. Tests of random single bit error performance show good results for error rates up to 10 -4.
AB - In this paper we present an approach for error robust audio coding at a medium data rate of about 176 kbps (mono, 44.1 kHz sampling rate). By combining a delay-free Adaptive Differential Pulse Code Modulation (ADPCM) coding-scheme and a numerically optimized low delay filter bank we achieve a very low algorithmic coding delay of only about 0.5 ms. The structure of the codec also allows for a high robustness against random single bit errors and even supports error resilience. Implementation structure, results of a listening test and PEAQ (Perceptual Evaluation of Audio Quality) based objective audio quality evaluation as well as tests of random single bit error performance are given. The presented coding-scheme provides a very good audio quality for vocals and speech. For most of the critical signals the audio quality can still be denoted as acceptable. Tests of random single bit error performance show good results for error rates up to 10 -4.
UR - http://www.scopus.com/inward/record.url?scp=84866324530&partnerID=8YFLogxK
M3 - Conference contribution
AN - SCOPUS:84866324530
SN - 9781618393968
VL - 1
SP - 187
EP - 192
BT - 131st Audio Engineering Society Convention 2011
T2 - 131st Audio Engineering Society Convention 2011
Y2 - 20 October 2011 through 23 October 2011
ER -