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)
