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home:alarrosa:branches:multimedia:libs:webrtc-audio-processing-0.3
webrtc-audio-processing
webrtc-audio-processing.changes
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File webrtc-audio-processing.changes of Package webrtc-audio-processing
------------------------------------------------------------------- Tue Feb 20 15:14:05 UTC 2024 - Dominique Leuenberger <dimstar@opensuse.org> - Use %patch -P N instead of deprecated %patchN. ------------------------------------------------------------------- Mon Oct 30 16:42:04 UTC 2023 - Antonio Larrosa <alarrosa@suse.com> - ExcludeArch s390, s390x and ppc64 since big endian support is not implemented. ------------------------------------------------------------------- Wed Sep 20 09:49:19 UTC 2023 - Antonio Larrosa <alarrosa@suse.com> - Remove the tar.xz file. Having the obscpio file is enough ------------------------------------------------------------------- Wed Sep 20 09:38:21 UTC 2023 - Antonio Larrosa <alarrosa@suse.com> - Use also dashes instead of underscores in the manual Requires ------------------------------------------------------------------- Wed Sep 20 09:04:13 UTC 2023 - Antonio Larrosa <alarrosa@suse.com> - 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 ------------------------------------------------------------------- Fri Sep 08 10:40:12 UTC 2023 - alarrosa@suse.com - 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 ------------------------------------------------------------------- Mon Aug 17 15:30:03 UTC 2020 - Dirk Mueller <dmueller@suse.com> - update to 0.3.1: * doc: file invalid reference to pulseaudio mailing list * various build system fixes - spec-cleaner run ------------------------------------------------------------------- Fri Aug 2 08:23:00 UTC 2019 - Martin Liška <mliska@suse.cz> - Use FAT LTO objects in order to provide proper static library. ------------------------------------------------------------------- Thu Jan 12 08:32:04 UTC 2017 - olaf@aepfle.de - Add baselibs.conf for gstreamer-plugins-bad-32bit ------------------------------------------------------------------- Sat Jun 25 10:39:08 UTC 2016 - oholecek@suse.com - Remove webrtc-aarch64.patch, no longer needed - Adapt the rest of webrtc- patches to new arch naming ------------------------------------------------------------------- Thu Jun 23 13:31:14 UTC 2016 - oholecek@suse.com - Remove unneeded explicit version dependency for automake ------------------------------------------------------------------- Wed Jun 22 11:55:11 UTC 2016 - oholecek@suse.com - 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 ------------------------------------------------------------------- Mon Jun 20 13:02:06 UTC 2016 - oholecek@suse.com - 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 ------------------------------------------------------------------- Mon May 30 09:00:51 UTC 2016 - oholecek@suse.com - 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 ------------------------------------------------------------------- Thu May 26 21:19:28 UTC 2016 - oholecek@suse.com - 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 ------------------------------------------------------------------- Thu Mar 7 13:51:31 UTC 2013 - idonmez@suse.com - Add patch webrtc-aarch64.patch from algraf to add aarch64 support ------------------------------------------------------------------- Wed Dec 19 10:39:23 CET 2012 - ro@suse.de - add s390 and s390x to known platforms by adding webrtc-s390x.patch ------------------------------------------------------------------- Tue Jul 3 15:00:06 UTC 2012 - dvaleev@suse.com - add ppc64 to known platforms ------------------------------------------------------------------- Tue May 15 10:40:38 CET 2012 - pascal.bleser@opensuse.org - initial version (0.1)
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