Packet Speech Transmission on ATM Networks Using a Variable Rate Embedded ADPCM Coding Scheme
Summary :
Subjective quality tests have proven that embedded adaptive differential PCM (ADPCM), known to tolerate information loss through bit dropping, does not maintain sufficient speech quality when directly applied to asynchronous transfer mode (ATM) due to the fixed-length cell transmission scheme unique to ATM. We propose a coding and transmission scheme which enhances the performance by adjusting the embedded ADPCM coding rate according to input speech characteristics, thereby taking advantage of the ATM environment, where the transmission of variable rate sources is feasible. By varying the number of code bits of an embedded ADPCM coder from 6bits per sample, or 48kbps, for blocks of speech with a high prediction gain, to 2bits, or 16kbps, for silent blocks, a good compromise between coding bit rate and speech quality with gradual degradation due to information loss is achieved. The results of subjective evaluation tests showed the speech quality of the proposed scheme to be over 3.5 mean opinion score (MOS) on a scale of 1 to 5 at a cell loss rate of 10%. A prototype of the codec and the ATM cell assembly/disassembly functions were also fabricated using 3 conventional digital signal processors (DSPs) for real-time conversation tests.
- Publication
- IEICE TRANSACTIONS on Communications Vol.E76-B No.4 pp.420-430
- Publication Date
- 1993/04/25
- Publicized
- Online ISSN
- DOI
- Type of Manuscript
- PAPER
- Category
- Communication Systems and Transmission Equipment
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Kazuhiro KONDO, Masashi OHNO, "Packet Speech Transmission on ATM Networks Using a Variable Rate Embedded ADPCM Coding Scheme" in IEICE TRANSACTIONS on Communications,
vol. E76-B, no. 4, pp. 420-430, April 1993, doi: .
Abstract: Subjective quality tests have proven that embedded adaptive differential PCM (ADPCM), known to tolerate information loss through bit dropping, does not maintain sufficient speech quality when directly applied to asynchronous transfer mode (ATM) due to the fixed-length cell transmission scheme unique to ATM. We propose a coding and transmission scheme which enhances the performance by adjusting the embedded ADPCM coding rate according to input speech characteristics, thereby taking advantage of the ATM environment, where the transmission of variable rate sources is feasible. By varying the number of code bits of an embedded ADPCM coder from 6bits per sample, or 48kbps, for blocks of speech with a high prediction gain, to 2bits, or 16kbps, for silent blocks, a good compromise between coding bit rate and speech quality with gradual degradation due to information loss is achieved. The results of subjective evaluation tests showed the speech quality of the proposed scheme to be over 3.5 mean opinion score (MOS) on a scale of 1 to 5 at a cell loss rate of 10%. A prototype of the codec and the ATM cell assembly/disassembly functions were also fabricated using 3 conventional digital signal processors (DSPs) for real-time conversation tests.
URL: https://global.ieice.org/en_transactions/communications/10.1587/e76-b_4_420/_p
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@ARTICLE{e76-b_4_420,
author={Kazuhiro KONDO, Masashi OHNO, },
journal={IEICE TRANSACTIONS on Communications},
title={Packet Speech Transmission on ATM Networks Using a Variable Rate Embedded ADPCM Coding Scheme},
year={1993},
volume={E76-B},
number={4},
pages={420-430},
abstract={Subjective quality tests have proven that embedded adaptive differential PCM (ADPCM), known to tolerate information loss through bit dropping, does not maintain sufficient speech quality when directly applied to asynchronous transfer mode (ATM) due to the fixed-length cell transmission scheme unique to ATM. We propose a coding and transmission scheme which enhances the performance by adjusting the embedded ADPCM coding rate according to input speech characteristics, thereby taking advantage of the ATM environment, where the transmission of variable rate sources is feasible. By varying the number of code bits of an embedded ADPCM coder from 6bits per sample, or 48kbps, for blocks of speech with a high prediction gain, to 2bits, or 16kbps, for silent blocks, a good compromise between coding bit rate and speech quality with gradual degradation due to information loss is achieved. The results of subjective evaluation tests showed the speech quality of the proposed scheme to be over 3.5 mean opinion score (MOS) on a scale of 1 to 5 at a cell loss rate of 10%. A prototype of the codec and the ATM cell assembly/disassembly functions were also fabricated using 3 conventional digital signal processors (DSPs) for real-time conversation tests.},
keywords={},
doi={},
ISSN={},
month={April},}
Copy
TY - JOUR
TI - Packet Speech Transmission on ATM Networks Using a Variable Rate Embedded ADPCM Coding Scheme
T2 - IEICE TRANSACTIONS on Communications
SP - 420
EP - 430
AU - Kazuhiro KONDO
AU - Masashi OHNO
PY - 1993
DO -
JO - IEICE TRANSACTIONS on Communications
SN -
VL - E76-B
IS - 4
JA - IEICE TRANSACTIONS on Communications
Y1 - April 1993
AB - Subjective quality tests have proven that embedded adaptive differential PCM (ADPCM), known to tolerate information loss through bit dropping, does not maintain sufficient speech quality when directly applied to asynchronous transfer mode (ATM) due to the fixed-length cell transmission scheme unique to ATM. We propose a coding and transmission scheme which enhances the performance by adjusting the embedded ADPCM coding rate according to input speech characteristics, thereby taking advantage of the ATM environment, where the transmission of variable rate sources is feasible. By varying the number of code bits of an embedded ADPCM coder from 6bits per sample, or 48kbps, for blocks of speech with a high prediction gain, to 2bits, or 16kbps, for silent blocks, a good compromise between coding bit rate and speech quality with gradual degradation due to information loss is achieved. The results of subjective evaluation tests showed the speech quality of the proposed scheme to be over 3.5 mean opinion score (MOS) on a scale of 1 to 5 at a cell loss rate of 10%. A prototype of the codec and the ATM cell assembly/disassembly functions were also fabricated using 3 conventional digital signal processors (DSPs) for real-time conversation tests.
ER -
English
