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| Name: libwebrtc-audio-coding-devel-static | Distribution: SUSE Linux Enterprise 15 |
| Version: 1.3 | Vendor: SUSE LLC <https://www.suse.com/> |
| Release: 150600.1.2 | Build date: Fri Mar 8 20:53:47 2024 |
| Group: Development/Libraries/C and C++ | Build host: h01-ch4c |
| Size: 12546458 | Source RPM: webrtc-audio-processing-1.3-150600.1.2.src.rpm |
| Packager: https://www.suse.com/ | |
| 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.
BSD-3-Clause
* Mon Oct 30 2023 alarrosa@suse.com
- ExcludeArch s390, s390x and ppc64 since big endian support is
not implemented.
* Wed Sep 20 2023 alarrosa@suse.com
- Remove the tar.xz file. Having the obscpio file is enough
* Wed Sep 20 2023 alarrosa@suse.com
- Use also dashes instead of underscores in the manual Requires
* Wed Sep 20 2023 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 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 2020 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 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/lib64/libwebrtc-audio-coding-1.a
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