Revisions of webrtc-audio-processing
buildservice-autocommit
accepted
request 1148307
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Takashi Iwai (tiwai)
(revision 28)
baserev update by copy to link target
Takashi Iwai (tiwai)
accepted
request 1148151
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Dominique Leuenberger (dimstar)
(revision 27)
Prepare for RPM 4.20
buildservice-autocommit
accepted
request 1121246
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Antonio Larrosa (alarrosa)
(revision 26)
baserev update by copy to link target
Antonio Larrosa (alarrosa)
accepted
request 1121245
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Antonio Larrosa (alarrosa)
(revision 25)
- ExcludeArch s390, s390x and ppc64 since big endian support is not implemented.
buildservice-autocommit
accepted
request 1112519
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Antonio Larrosa (alarrosa)
(revision 24)
baserev update by copy to link target
Antonio Larrosa (alarrosa)
accepted
request 1112518
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Antonio Larrosa (alarrosa)
(revision 23)
- Remove the tar.xz file. Having the obscpio file is enough
Antonio Larrosa (alarrosa)
accepted
request 1112514
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Antonio Larrosa (alarrosa)
(revision 22)
Really do what the changelog said I did
Antonio Larrosa (alarrosa)
accepted
request 1112511
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Antonio Larrosa (alarrosa)
(revision 21)
- Use also dashes instead of underscores in the manual Requires
Antonio Larrosa (alarrosa)
accepted
request 1112499
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Antonio Larrosa (alarrosa)
(revision 20)
- Rename the generated library package names to add a dash between the name and soname (libwebrtc*-1-3 instead of libwebrtc*1-3) - Rename the generated packages to use dashes instead of underscores - Change baselibs.conf accordingly - Add patch to reduce the required meson version so the package builds in Leap 15.4/15.5: * reduce-meson-dep.patch
Takashi Iwai (tiwai)
accepted
request 1111520
from
Antonio Larrosa (alarrosa)
(revision 19)
- Update to version 1.3: * build: Bump version to 1.3 * meson: Fix generation of pkgconfig files * build: Bump version to 1.2 * meson: Update minimum version based on what abseil wrap needs * build: Expose absl as a dependency of webrtc-audio-processing * meson: Update to latest wrap, install required absl headers * doc: Update tarball generation process * file_utils.h: Fix build with gcc-13 * meson: Fixes for MSVC build * meson: Ensure that abseil is built with c++17 too * More changes not listed by upstream. Check the following link to see them: https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/commits/v1.3 - Add patch that fixes some compiler "control reaches end of non-void function" errors: * fix-build.patch - Add patch that fixes i586 build: * fix-i586.patch - Disable patches until they're rebased to the current codebase: * big_endian_support.patch * big_endian_support_2.patch - Rebased patches: * webrtc-ppc64.patch * webrtc-s390x.patch
buildservice-autocommit
accepted
request 828514
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Takashi Iwai (tiwai)
(revision 18)
baserev update by copy to link target
Takashi Iwai (tiwai)
accepted
request 827376
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Dirk Mueller (dirkmueller)
(revision 17)
- update to 0.3.1: * doc: file invalid reference to pulseaudio mailing list * various build system fixes - spec-cleaner run
buildservice-autocommit
accepted
request 720522
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Tomáš Chvátal (scarabeus_iv)
(revision 16)
baserev update by copy to link target
Tomáš Chvátal (scarabeus_iv)
accepted
request 720509
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Martin Liška (marxin)
(revision 15)
Use FAT LTO objects in order to provide proper static library.
buildservice-autocommit
accepted
request 451393
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Olaf Hering (olh)
(revision 14)
baserev update by copy to link target
Olaf Hering (olh)
accepted
request 451390
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Olaf Hering (olh)
(revision 13)
- Add baselibs.conf for gstreamer-plugins-bad-32bit
buildservice-autocommit
accepted
request 404795
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Takashi Iwai (tiwai)
(revision 12)
baserev update by copy to link target
Takashi Iwai (tiwai)
accepted
request 404777
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Ondrej Holecek (oholecek)
(revision 11)
- Remove webrtc-aarch64.patch, no longer needed - Adapt the rest of webrtc- patches to new arch naming - Remove unneeded explicit version dependency for automake - Update to 0.3 * build: enforce linking with --no-undefined, add explicit -lpthread * build: Make sure files with SSE2 code are compiled with -msse2 - Remove no-undefined.patch - Remove webrtc-audio-processing-0.2-x86_msse2.patch - Add no-undefined.patch patch https://cgit.freedesktop.org/pulseaudio/webrtc-audio-processing/patch/?id=d58164e4d87854233564b59e76259b72e21507f6 - Add big_endian_support_2.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738 - Adapt webrtc-audio-processing-0.2-x86_msse2.patch to new version - Adapt big_endian_support.patch to new version - Add webrtc-audio-processing-0.2-x86_msse2.patch patch fixing 386 build https://lists.freedesktop.org/archives/pulseaudio-discuss/2016-May/026294.html - Add big_endian_support.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738 - New automake version dependency >= 1.5 - Update to 0.2: Contains API breaking changes. Upstream changes include: * Rewritten AGC and voice activity detection * Intelligibility enhancer * Extended AEC filter * Beamformer * Transient suppressor * ARM, NEON and MIPS optimisations (MIPS optimisations are not hooked up) API changes: * We no longer include a top-level audio_processing.h. The webrtc tree format is used, so use webrtc/modules/audio_processing/include/audio_processing.h * The top-level module_common_types.h has also been moved to webrtc/modules/interface/module_common_types.h * C++11 support is now required while compiling client code * AudioProcessing::Create() does not take any arguments any more * AudioProcessing::Destroy() is gone, use standard C++ "delete" instead * Stream parameters are now configured via StreamConfig and ProcessingConfig rather than set_sample_rate(), set_num_channels(), etc. * AudioFrame field names have changed * Use config API for newer audio processing options * Use ProcessReverseStream() instead of AnalyzeReverseStream(), particularly when using the intelligibility enhancer * GainControl::set_analog_level_limits() is broken. The AGC implementation hard codes 0-255 as the volume range Other notes: * The new audio processing parameters are not all tested, and a few are not enabled upstream (in Chromium) either * The rewritten AGC appears to be less sensitive, and it might make sense to initialise the capture volume to something reasonable (33% or 50%, for example) to make sure there is sufficient energy in the stream to trigger the AGC mechanism - Adapted all 3 arch patches
buildservice-autocommit
accepted
request 157852
from
Ismail Dönmez (namtrac)
(revision 10)
baserev update by copy to link target
Ismail Dönmez (namtrac)
committed
(revision 9)
- Add patch webrtc-aarch64.patch from algraf to add aarch64 support
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