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Accepting request 342885 from multimedia:libs

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OBS-URL: https://build.opensuse.org/request/show/342885
OBS-URL: https://build.opensuse.org/package/show/openSUSE:Factory/libgsm?expand=0&rev=24
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Stephan Kulow 2015-11-22 09:58:07 +00:00 committed by Git OBS Bridge
commit 3b3eace9d2
2 changed files with 38 additions and 61 deletions
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libgsm.changes
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@ -1,3 +1,9 @@
-------------------------------------------------------------------
Sat Nov 7 09:23:43 UTC 2015 - jengelh@inai.de
- Make description more concise. SPARCstations are no longer a
meaningful metric.
-------------------------------------------------------------------
Sat Oct 10 14:06:43 UTC 2015 - p.drouand@gmail.com

93
libgsm.spec
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@ -1,7 +1,7 @@
#
# spec file for package libgsm
#
# Copyright (c) 2015 SUSE LINUX Products GmbH, Nuernberg, Germany.
# Copyright (c) 2015 SUSE LINUX GmbH, Nuernberg, Germany.
#
# All modifications and additions to the file contributed by third parties
# remain the property of their copyright owners, unless otherwise agreed
@ -35,21 +35,10 @@ Patch3: libgsm-strict-aliasing.patch
BuildRoot: %{_tmppath}/%{name}-%{version}-build
%description
Contains libraries and binaries for a GSM speech compressor. libgsm
contains a standard 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, which is a frame rate of 50 Hz) into 260 bits.
For compatibility with typical UNIX applications, our 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 modeled after compress(1)
and a library API. Compression and decompression run faster than
real-time on most SPARC stations. The implementation has been verified
against the ETSI standard test patterns.
libgsm implements the European GSM 06.10 provisional standard for
full-rate speech transcoding, prI-ETS 300 036, which uses RPE/LTP
coding at 13 kbit/s. GSM 06.10 compresses frames of 160 13-bit
samples (8 kHz sampling rate) into 260 bits.
%package -n libgsm1
Summary: GSM 06.10 Lossy Speech Compressor Library and Utilities
@ -59,21 +48,13 @@ Provides: %{name} = %{version}
Obsoletes: %{name} < %{version}
%description -n libgsm1
Contains libraries and binaries for a GSM speech compressor. libgsm
contains a standard 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, which is a frame rate of 50 Hz) into 260 bits.
For compatibility with typical UNIX applications, our 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 modeled after compress(1)
and a library API. Compression and decompression run faster than
real-time on most SPARC stations. The implementation has been verified
against the ETSI standard test patterns.
Contains the library for a GSM speech compressor.
libgsm implements 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) into 260 bits.
%package utils
Summary: GSM 06.10 Lossy Speech Compressor Library and Utilities
@ -82,21 +63,16 @@ Group: Productivity/Multimedia/Sound/Editors and Convertors
Provides: %{name}:%{_bindir}/toast
%description utils
Contains libraries and binaries for a GSM speech compressor. libgsm
contains a standard 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, which is a frame rate of 50 Hz) into 260 bits.
For compatibility with typical UNIX applications, our 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 modeled after compress(1)
and a library API. Compression and decompression run faster than
real-time on most SPARC stations. The implementation has been verified
against the ETSI standard test patterns.
Contains binaries for a GSM speech compressor, verified against the
ETSI standard test patterns.
libgsm implements 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) into 260 bits.
The front-end is modeled after the historic compress(1) utility.
%package devel
Summary: GSM 06.10 Lossy Speech Compressor Library and Utilities
@ -104,21 +80,16 @@ Group: Productivity/Multimedia/Sound/Editors and Convertors
Requires: libgsm1 = %{version}
%description devel
Contains libraries and binaries for a GSM speech compressor. libgsm
contains a standard 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, which is a frame rate of 50 Hz) into 260 bits.
For compatibility with typical UNIX applications, our 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 modeled after compress(1)
and a library API. Compression and decompression run faster than
real-time on most SPARC stations. The implementation has been verified
against the ETSI standard test patterns.
Contains the development kit for the libgsm speech compressor.
libgsm implements the European GSM 06.10 provisional standard for
full-rate speech transcoding, prI-ETS 300 036, which uses RPE/LTP
coding at 13 kbit/s. GSM 06.10 compresses frames of 160 13-bit
samples (8 kHz sampling rate) into 260 bits.
This implementation turns frames of 160 16-bit linear samples into
33-byte frames (1650 bytes/s) and has been verified against the ETSI
standard test patterns.
%prep
%setup -n %{_name}-%{_version}