libgsm 1.0.19-3 source package in Ubuntu
Changelog
libgsm (1.0.19-3) unstable; urgency=medium * adopt package (Closes: #1009975) * debian/control: move maintenance to the mobcom team * debian/control: bump Standards-Version to 4.6.1 * debian/control: use VCS URLs from the mobcom team * Thanks to Marriott NZ <email address hidden> for the report about the quotes in the mailcap entry. This was already fixed in 1.0.19-1. (Closes: #987404) -- Thorsten Alteholz <email address hidden> Fri, 01 Jul 2022 22:03:02 +0200
Upload details
- Uploaded by:
- Debian Mobcom Maintainers
- Uploaded to:
- Sid
- Original maintainer:
- Debian Mobcom Maintainers
- Architectures:
- any
- Section:
- devel
- Urgency:
- Medium Urgency
See full publishing history Publishing
Series | Published | Component | Section | |
---|---|---|---|---|
Kinetic | release | universe | devel |
Downloads
File | Size | SHA-256 Checksum |
---|---|---|
libgsm_1.0.19-3.dsc | 2.1 KiB | 7cc73dd9c913bf7d6765643b2135dcc9af2bc7d9e961d95bb59c941755a0d689 |
libgsm_1.0.19.orig.tar.gz | 63.1 KiB | 4903652f68a8c04d0041f0d19b1eb713ddcd2aa011c5e595b3b8bca2755270f6 |
libgsm_1.0.19-3.debian.tar.xz | 10.4 KiB | ffbc332ee354533e51bf86e5664bd94aa492db9b42cbe5f8305c71fd212cb7f7 |
Available diffs
- diff from 1.0.19-2 to 1.0.19-3 (864 bytes)
No changes file available.
Binary packages built by this source
- libgsm-tools: User binaries for a GSM speech compressor
This package contains user binaries for libgsm, an implementation of
the European GSM 06.10 provisional standard for full-rate speech
transcoding, prI-ETS 300 036, which uses RPE/LTP (residual pulse
excitation/long term prediction) coding at 13 kbit/s.
.
GSM 06.10 compresses frames of 160 13-bit samples (8 kHz sampling
rate, i.e. a frame rate of 50 Hz) into 260 bits; for compatibility
with typical UNIX applications, this implementation turns frames of
160 16-bit linear samples into 33-byte frames (1650 Bytes/s).
The quality of the algorithm is good enough for reliable speaker
recognition; even music often survives transcoding in recognizable
form (given the bandwidth limitations of 8 kHz sampling rate).
.
The interfaces offered are a front end modelled after compress(1), and
a library API. Compression and decompression run faster than realtime
on most SPARCstations. The implementation has been verified against the
ETSI standard test patterns.
- libgsm-tools-dbgsym: debug symbols for libgsm-tools
- libgsm1: Shared libraries for GSM speech compressor
This package contains runtime shared libraries for libgsm, an
implementation of the European GSM 06.10 provisional standard for
full-rate speech transcoding, prI-ETS 300 036, which uses RPE/LTP
(residual pulse excitation/long term prediction) coding at 13 kbit/s.
.
GSM 06.10 compresses frames of 160 13-bit samples (8 kHz sampling
rate, i.e. a frame rate of 50 Hz) into 260 bits; for compatibility
with typical UNIX applications, this implementation turns frames of
160 16-bit linear samples into 33-byte frames (1650 Bytes/s).
The quality of the algorithm is good enough for reliable speaker
recognition; even music often survives transcoding in recognizable
form (given the bandwidth limitations of 8 kHz sampling rate).
.
The interfaces offered are a front end modelled after compress(1), and
a library API. Compression and decompression run faster than realtime
on most SPARCstations. The implementation has been verified against the
ETSI standard test patterns.
- libgsm1-dbgsym: debug symbols for libgsm1
- libgsm1-dev: Development libraries for a GSM speech compressor
This package contains header files and development libraries for
libgsm, an implementation of the European GSM 06.10 provisional
standard for full-rate speech transcoding, prI-ETS 300 036, which
uses RPE/LTP (residual pulse excitation/long term prediction) coding
at 13 kbit/s.
.
GSM 06.10 compresses frames of 160 13-bit samples (8 kHz sampling
rate, i.e. a frame rate of 50 Hz) into 260 bits; for compatibility
with typical UNIX applications, this implementation turns frames of
160 16-bit linear samples into 33-byte frames (1650 Bytes/s).
The quality of the algorithm is good enough for reliable speaker
recognition; even music often survives transcoding in recognizable
form (given the bandwidth limitations of 8 kHz sampling rate).
.
The interfaces offered are a front end modelled after compress(1), and
a library API. Compression and decompression run faster than realtime
on most SPARCstations. The implementation has been verified against the
ETSI standard test patterns.