File: control

package info (click to toggle)
libgsm 1.0.23-2
  • links: PTS, VCS
  • area: main
  • in suites: forky, sid
  • size: 632 kB
  • sloc: ansic: 6,258; makefile: 311
file content (87 lines) | stat: -rw-r--r-- 4,005 bytes parent folder | download
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
Source: libgsm
Section: libs
Priority: optional
Maintainer: Debian Mobcom Maintainers <Debian-mobcom-maintainers@lists.alioth.debian.org>
Uploaders: Thorsten Alteholz <debian@alteholz.de>
Standards-Version: 4.7.2
Rules-Requires-Root: no
Build-Depends:
 debhelper-compat (= 13)
Homepage: https://www.quut.com/gsm/
Vcs-Browser: https://salsa.debian.org/debian-mobcom-team/libgsm
Vcs-Git: https://salsa.debian.org/debian-mobcom-team/libgsm.git

Package: libgsm1
Architecture: any
Pre-Depends: ${misc:Pre-Depends}
Depends: ${shlibs:Depends}, ${misc:Depends}
Conflicts: libgsm-dev
Multi-Arch: same
Description: 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.

Package: libgsm1-dev
Section: libdevel
Architecture: any
Depends: libgsm1 (= ${binary:Version}), ${misc:Depends}
Replaces: libgsm-dev
Multi-Arch: same
Description: 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.

Package: libgsm-tools
Replaces: libgsm-bin
Section: sound
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}
Description: 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.