Changelog for libwebrtc_audio_processing1-0.3.1-2.1.x86_64.rpm :

* Mon Aug 17 2020 Dirk Mueller - update to 0.3.1:
* doc: file invalid reference to pulseaudio mailing list
* various build system fixes- spec-cleaner run
* Fri Aug 02 2019 Martin Liška - Use FAT LTO objects in order to provide proper static library.
* Thu Jan 12 2017 olafAATTaepfle.de- Add baselibs.conf for gstreamer-plugins-bad-32bit
* Sat Jun 25 2016 oholecekAATTsuse.com- Remove webrtc-aarch64.patch, no longer needed- Adapt the rest of webrtc- patches to new arch naming
* Thu Jun 23 2016 oholecekAATTsuse.com- Remove unneeded explicit version dependency for automake
* Wed Jun 22 2016 oholecekAATTsuse.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 2016 oholecekAATTsuse.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 2016 oholecekAATTsuse.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 2016 oholecekAATTsuse.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 07 2013 idonmezAATTsuse.com- Add patch webrtc-aarch64.patch from algraf to add aarch64 support