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| Name: libwebrtc_audio_processing-devel | Distribution: openSUSE:Factory:zSystems |
| Version: 0.3.1 | Vendor: openSUSE |
| Release: 1.1 | Build date: Sat Sep 30 07:39:22 2023 |
| Group: Development/Libraries/C and C++ | Build host: s390zl21 |
| Size: 128821 | Source RPM: webrtc-audio-processing-0-0.3.1-1.1.src.rpm |
| Packager: https://bugs.opensuse.org | |
| Url: https://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/ | |
| Summary: Real-Time Communication Library for Web Browsers | |
WebRTC is an open source project that enables web browsers with Real-Time Communications (RTC) capabilities via simple Javascript APIs. The WebRTC components have been optimized to best serve this purpose. WebRTC implements the W3C's proposal for video conferencing on the web. This is a compatibility package which should only be used by applications that haven't be updated yet to the newer 1.x version.
BSD-3-Clause
* Thu Sep 28 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 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 02 2019 Martin Liška <mliska@suse.cz>
- Use FAT LTO objects in order to provide proper static library.
* Thu Jan 12 2017 olaf@aepfle.de
- Add baselibs.conf for gstreamer-plugins-bad-32bit
* Sat Jun 25 2016 oholecek@suse.com
- Remove webrtc-aarch64.patch, no longer needed
- Adapt the rest of webrtc- patches to new arch naming
* Thu Jun 23 2016 oholecek@suse.com
- Remove unneeded explicit version dependency for automake
* Wed Jun 22 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 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 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 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
/usr/include/webrtc_audio_processing /usr/include/webrtc_audio_processing/webrtc /usr/include/webrtc_audio_processing/webrtc/base /usr/include/webrtc_audio_processing/webrtc/base/arraysize.h /usr/include/webrtc_audio_processing/webrtc/base/basictypes.h /usr/include/webrtc_audio_processing/webrtc/base/checks.h /usr/include/webrtc_audio_processing/webrtc/base/constructormagic.h /usr/include/webrtc_audio_processing/webrtc/base/maybe.h /usr/include/webrtc_audio_processing/webrtc/base/platform_file.h /usr/include/webrtc_audio_processing/webrtc/common.h /usr/include/webrtc_audio_processing/webrtc/common_types.h /usr/include/webrtc_audio_processing/webrtc/modules /usr/include/webrtc_audio_processing/webrtc/modules/audio_processing /usr/include/webrtc_audio_processing/webrtc/modules/audio_processing/beamformer /usr/include/webrtc_audio_processing/webrtc/modules/audio_processing/beamformer/array_util.h /usr/include/webrtc_audio_processing/webrtc/modules/audio_processing/include /usr/include/webrtc_audio_processing/webrtc/modules/audio_processing/include/audio_processing.h /usr/include/webrtc_audio_processing/webrtc/modules/interface /usr/include/webrtc_audio_processing/webrtc/modules/interface/module_common_types.h /usr/include/webrtc_audio_processing/webrtc/system_wrappers /usr/include/webrtc_audio_processing/webrtc/system_wrappers/include /usr/include/webrtc_audio_processing/webrtc/system_wrappers/include/trace.h /usr/include/webrtc_audio_processing/webrtc/typedefs.h /usr/lib64/libwebrtc_audio_processing.so /usr/lib64/pkgconfig/webrtc-audio-processing.pc
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