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Gerhard G. Langenbucher

dblp:271/2473 · DBLP profile ↗
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1ranked-venue papers
1as first author
0since 2021 · last 1982
—ORCID · none

Domains — the database's venue-derived domains; a paper can count in several

Computer networks · 1 · 1 first-author

Expertise — from the expertise taxonomy: the topics of the expert's papers under the CCF categories. A weight counts papers with recency: 1 for a paper about the topic, 0.3 when the topic is its context, halved every five years.

Computer graphics and multimedia
1 paper
Audio and music processing · 100%
Theoretical computer science
1 paper
Coding theory · 100%

Topics — the 5 heaviest of 5, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Audio and music processing › speech coding
adaptive differential pulse code modulation
0.011982
Efficient Coding and Speech Interpolation: Principles and Performance Characterization · IEEE Trans. Commun. 1982
Audio and music processing › speech coding
digital speech interpolation
0.011982
Efficient Coding and Speech Interpolation: Principles and Performance Characterization · IEEE Trans. Commun. 1982
Audio and music processing
speech coding
0.011982
Efficient Coding and Speech Interpolation: Principles and Performance Characterization · IEEE Trans. Commun. 1982
Coding theory
source coding
0.011982
Efficient Coding and Speech Interpolation: Principles and Performance Characterization · IEEE Trans. Commun. 1982
Coding theory › source coding › rate-distortion theory
variable-rate coding
0.011982
Efficient Coding and Speech Interpolation: Principles and Performance Characterization · IEEE Trans. Commun. 1982

Methods — techniques the papers use, named apart from their topics

performance characterization · 0.0SNR estimation · 0.0
YearPublicationVenuePosition
1982 Efficient Coding and Speech Interpolation: Principles and Performance Characterization
abstract
The BPA (block priority assignment) system combines the principle of speech interpolation with waveform coding methods for the active speech signal. Coding with variable wordlength permits adapting the demand of theNspeech sources to the capacity of the channel. The system emphasized in this paper not only varies the bits per source according to the number of active talkers, but also assigns the number of bits to the source that needs it most based on estimates of the SNR over the block. When ADPCM and DSI are combined to give a 4:1 compression ratio over standard PCM (data rate 16 kbits/s) no distortions are audible. The influence of transmission errors and the use of various waveform coders is considered.
Gerhard G. Langenbucher
IEEE Trans. Commun.1