The Experts below are selected from a list of 1029 Experts worldwide ranked by ideXlab platform
T.e. Tremain - One of the best experts on this subject based on the ideXlab platform.
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Progress towards a new government standard 2400 bps Voice Coder
1995 International Conference on Acoustics Speech and Signal Processing, 1995Co-Authors: M A Kohler, L.m. Supplee, T.e. TremainAbstract:In order to support the need for higher quality low rate Voice communications for government, industry, and military customers, the United States Government is conducting a search for a new Voice compression algorithm at 2400 bits per second (bps). The United States Department of Defense Digital Voice Processing Consortium (DDVPC), consisting of members from civilian and military branches of the US Government, is directing the testing and evaluation of several candidate 2400 bps algorithms. The goal of the DDVPC is to select a new algorithm which meets or exceeds the published requirements by mid 1996. The selected algorithm, to become the new standard, should be implementable in a small, low powered device by 1997. This paper describes the status of the testing and evaluation process from its beginning in early 1993 through the end of 1994.
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ICASSP - Progress towards a new government standard 2400 bps Voice Coder
1995 International Conference on Acoustics Speech and Signal Processing, 1995Co-Authors: M A Kohler, L.m. Supplee, T.e. TremainAbstract:In order to support the need for higher quality low rate Voice communications for government, industry, and military customers, the United States Government is conducting a search for a new Voice compression algorithm at 2400 bits per second (bps). The United States Department of Defense Digital Voice Processing Consortium (DDVPC), consisting of members from civilian and military branches of the US Government, is directing the testing and evaluation of several candidate 2400 bps algorithms. The goal of the DDVPC is to select a new algorithm which meets or exceeds the published requirements by mid 1996. The selected algorithm, to become the new standard, should be implementable in a small, low powered device by 1997. This paper describes the status of the testing and evaluation process from its beginning in early 1993 through the end of 1994.
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ICASSP - An expandable error-protected 4800 bps CELP Coder (US Federal Standard 4800 bps Voice Coder)
International Conference on Acoustics Speech and Signal Processing, 1Co-Authors: J. P. Campbell, V.c. Welch, T.e. TremainAbstract:The authors describe a code-excited linear predictive (CELP) Coder that was recently selected as the US government standard 4800-b/s Voice Coder. The Coder has the capability for future expansion and is robust to channel errors and noisy environments. Listening tests and DRT (diagnostic rhyme test) and DAM (diagnostic acceptability measure) scores show that the deCoder exceeds the performance of all government standard speech Coders operating at rates below 16 kb/s and is comparable to 32-kb/s continuously-variable-slope delta modulated Coders. >
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ICASSP - CELP coding for land mobile radio applications
International Conference on Acoustics Speech and Signal Processing, 1Co-Authors: D.j. Rahikka, V.c. Welch, T.e. Tremain, J. P. CampbellAbstract:The performance of the Proposed Federal Standard 1024 8000-b/s system for digital land mobile radio on a 12.5- or 6.25-kHz channel is described. The system contains the Proposed Federal Standard 1016 4800-b/s code excited linear predictive (CELP) Voice Coder developed by the US DOD and AT&T Bell Laboratories. The system is error protected with 2400 b/s and has 800 b/s for overhead signaling. The system is intended to replace the 12 kHz continuous variable slope deltamod (CVSD) Voice Coder which is used in Federal Standard 1025. The error protection technique used is Golay coding with hard or soft decoding. The performance of this error protection over simulated fading radio channels is described. >
S. Dimolitsas - One of the best experts on this subject based on the ideXlab platform.
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Correlation between headphone and telephone-handset listener opinion scores for single-stimulus Voice Coder performance assessments
IEEE Signal Processing Letters, 1995Co-Authors: S. Dimolitsas, F.l. Corcoran, Chanasandra RavishankarAbstract:It has often been suggested that digital Voice Coder performance may depend on the type of listening set used in listener opinion assessments. In this letter, the quantitative impact of using headphones and telephone handsets is presented for single-stimulus MOS assessments. From this analysis, it is derived that there are significant numerical differences between tests employing these types of listening instruments and that cardinal differences in performance may be consistent and thus predictable. This study is extended to an ordinal analysis from which it is derived that although ranked performance does vary, depending on which of two listening instruments is employed, the two rankings are highly correlated. >
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Estimation of digital low-rate encoded Voice link performance from instrumental measurements
IEEE Transactions on Instrumentation and Measurement, 1993Co-Authors: S. Dimolitsas, F.l. Corcoran, J.g. Phipps, M. BaranieckiAbstract:The correlation between subjective and instrumental assessments of low-rate digital Voice Coder transparency to Voice and nonVoice signals is quantified. The instrumental, or objective, performance and subjective Voice quality of eight low-rate digital Voice processors were compared by correlating the results obtained from a set of multilingual subjective assessments with the results obtained from a series of instrumental measurements employing readily available, analog interface-based techniques. The results show that meaningful information on link performance with Voice signals can be obtained over a wide range of engineered link quality using instrumental technology employing nonVoice signals. >
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Characterization of low-rate digital Voice Coder performance with non-Voice signals
Speech Communication, 1993Co-Authors: S. DimolitsasAbstract:Abstract This paper describes the test methodology employed for the performance evaluation of the CCITT 16 kbit/sLD-CELP Coder (Recommendation G.728) with non-Voice signals. This methodology is sufficiently general that it can be employed for the evaluation of other low transmission-rate digital speech Coders. The types of non-Voice signals considered in this paper include Voiceband data, network signaling, circuit continuity tones and dual-tone multi-frequency signaling.
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A 1200 b/s Voice Coder Based on Split Vector Quantization of Line Spectral Frequencies
Proceedings. IEEE Workshop on Speech Coding for Telecommunications, 1Co-Authors: C.s. Ravishankar, B.r.u. Bhaskar, S. DimolitsasAbstract:This paper describes the current status of an on-going research and development effort whose objective is a 1200 b/s Voice Coder with a high degree of speech intelligibility and natural Voice quality. The 1200 b/s Voice Coder is based on tenth-order linear prediction analysis, split vector quantization of line spectral frequencies, differential pitch and gain quantization, and adaptive postfiltering. An improved version of the 1200 b/s Coder which employs harmonic deviations based excitation generation is also described, which operates at 1800 b/s during Voiced frames and 1200 bts during unVoiced and silent frames. These Voice coding technologies are critical components in the development of satellite or terrestrial based mobile and portable communications using miniature handheld transceivers.
M A Kohler - One of the best experts on this subject based on the ideXlab platform.
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Progress towards a new government standard 2400 bps Voice Coder
1995 International Conference on Acoustics Speech and Signal Processing, 1995Co-Authors: M A Kohler, L.m. Supplee, T.e. TremainAbstract:In order to support the need for higher quality low rate Voice communications for government, industry, and military customers, the United States Government is conducting a search for a new Voice compression algorithm at 2400 bits per second (bps). The United States Department of Defense Digital Voice Processing Consortium (DDVPC), consisting of members from civilian and military branches of the US Government, is directing the testing and evaluation of several candidate 2400 bps algorithms. The goal of the DDVPC is to select a new algorithm which meets or exceeds the published requirements by mid 1996. The selected algorithm, to become the new standard, should be implementable in a small, low powered device by 1997. This paper describes the status of the testing and evaluation process from its beginning in early 1993 through the end of 1994.
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ICASSP - Progress towards a new government standard 2400 bps Voice Coder
1995 International Conference on Acoustics Speech and Signal Processing, 1995Co-Authors: M A Kohler, L.m. Supplee, T.e. TremainAbstract:In order to support the need for higher quality low rate Voice communications for government, industry, and military customers, the United States Government is conducting a search for a new Voice compression algorithm at 2400 bits per second (bps). The United States Department of Defense Digital Voice Processing Consortium (DDVPC), consisting of members from civilian and military branches of the US Government, is directing the testing and evaluation of several candidate 2400 bps algorithms. The goal of the DDVPC is to select a new algorithm which meets or exceeds the published requirements by mid 1996. The selected algorithm, to become the new standard, should be implementable in a small, low powered device by 1997. This paper describes the status of the testing and evaluation process from its beginning in early 1993 through the end of 1994.
J. P. Campbell - One of the best experts on this subject based on the ideXlab platform.
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ICASSP - An expandable error-protected 4800 bps CELP Coder (US Federal Standard 4800 bps Voice Coder)
International Conference on Acoustics Speech and Signal Processing, 1Co-Authors: J. P. Campbell, V.c. Welch, T.e. TremainAbstract:The authors describe a code-excited linear predictive (CELP) Coder that was recently selected as the US government standard 4800-b/s Voice Coder. The Coder has the capability for future expansion and is robust to channel errors and noisy environments. Listening tests and DRT (diagnostic rhyme test) and DAM (diagnostic acceptability measure) scores show that the deCoder exceeds the performance of all government standard speech Coders operating at rates below 16 kb/s and is comparable to 32-kb/s continuously-variable-slope delta modulated Coders. >
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ICASSP - CELP coding for land mobile radio applications
International Conference on Acoustics Speech and Signal Processing, 1Co-Authors: D.j. Rahikka, V.c. Welch, T.e. Tremain, J. P. CampbellAbstract:The performance of the Proposed Federal Standard 1024 8000-b/s system for digital land mobile radio on a 12.5- or 6.25-kHz channel is described. The system contains the Proposed Federal Standard 1016 4800-b/s code excited linear predictive (CELP) Voice Coder developed by the US DOD and AT&T Bell Laboratories. The system is error protected with 2400 b/s and has 800 b/s for overhead signaling. The system is intended to replace the 12 kHz continuous variable slope deltamod (CVSD) Voice Coder which is used in Federal Standard 1025. The error protection technique used is Golay coding with hard or soft decoding. The performance of this error protection over simulated fading radio channels is described. >
L.m. Supplee - One of the best experts on this subject based on the ideXlab platform.
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Progress towards a new government standard 2400 bps Voice Coder
1995 International Conference on Acoustics Speech and Signal Processing, 1995Co-Authors: M A Kohler, L.m. Supplee, T.e. TremainAbstract:In order to support the need for higher quality low rate Voice communications for government, industry, and military customers, the United States Government is conducting a search for a new Voice compression algorithm at 2400 bits per second (bps). The United States Department of Defense Digital Voice Processing Consortium (DDVPC), consisting of members from civilian and military branches of the US Government, is directing the testing and evaluation of several candidate 2400 bps algorithms. The goal of the DDVPC is to select a new algorithm which meets or exceeds the published requirements by mid 1996. The selected algorithm, to become the new standard, should be implementable in a small, low powered device by 1997. This paper describes the status of the testing and evaluation process from its beginning in early 1993 through the end of 1994.
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ICASSP - Progress towards a new government standard 2400 bps Voice Coder
1995 International Conference on Acoustics Speech and Signal Processing, 1995Co-Authors: M A Kohler, L.m. Supplee, T.e. TremainAbstract:In order to support the need for higher quality low rate Voice communications for government, industry, and military customers, the United States Government is conducting a search for a new Voice compression algorithm at 2400 bits per second (bps). The United States Department of Defense Digital Voice Processing Consortium (DDVPC), consisting of members from civilian and military branches of the US Government, is directing the testing and evaluation of several candidate 2400 bps algorithms. The goal of the DDVPC is to select a new algorithm which meets or exceeds the published requirements by mid 1996. The selected algorithm, to become the new standard, should be implementable in a small, low powered device by 1997. This paper describes the status of the testing and evaluation process from its beginning in early 1993 through the end of 1994.
