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webrtc-audio-processing-0
webrtc-audio-processing-0.changes
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File webrtc-audio-processing-0.changes of Package webrtc-audio-processing-0
------------------------------------------------------------------- Tue Feb 20 15:18:06 UTC 2024 - Dominique Leuenberger <dimstar@opensuse.org> - Use %patch -P N instead of deprecated %patchN. ------------------------------------------------------------------- Thu Sep 28 09:56:45 UTC 2023 - Antonio Larrosa <alarrosa@suse.com> - Rename the 0.3.1 version of the package to webrtc-audio-processing-0 so we can keep it around while all applications are ported to version 1.x (like baresip and dino). There's no need to rename the devel package since the new version uses dashes instead of underscores in the package name. ------------------------------------------------------------------- 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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