libwebrtc-audio-coding-devel-1.3-150600.1.3 >  A fO ?/BX͛ ii0<Ҝ5J̪._O|jk(f1 ?x 0 3"շmr-J~ۋfn%_k,bv"/~iI6Qh$,Ƌsgst%|jf*)$Q;xX^T- O $M ]@R6_'[ǴdN=艠xGTEZnIb5r*~>) {:eMP0V \H&+Dͨe *cݰxo *P}J6MFHi-3)L@/TZ#2,k `*+jo']}&!t.DH^ju= 8Z 0{@2y!1Ui Q۹@R ܈\ X<*.f3NM]X[zuzP9276352065e85919ce9107273badf0a9293dfdd8588df697a20f2e0d553d941eedd22b39fc49b880789bfbd6433e08deac8a9aaf2 @fO ?7{m-SryFj[ #]? r',4UB"S .c/Hb86E7CDHIۖ_MHVKSF ~_Z8 0=n |BA] פUwtpVaE%N\W b\aͳwx0 O" )qn1~Ax9 3"-?w@/?L`޸$R adAliF"n~BF/kMFO%u}Rf͇MY~S&F'"QU / gfP]fTI[sT6C#gOƐ7M1:_?XKQ Ol6IaPm uvT.|l>p>?xd# . _x|  RX`h l p x  (0   (8 9D::FGHIXY \D]L^bcLdefluvwxyz(,2tClibwebrtc-audio-coding-devel1.3150600.1.3Real-Time Communication Library for Web BrowsersWebRTC 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.fah04-armsrv1SUSE Linux Enterprise 15SUSE LLC BSD-3-Clausehttps://www.suse.com/Development/Libraries/C and C++https://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/linuxaarch64f^f46ef131e540b066cee96523840963287533e6b484c809503703c1577b7826ce25libwebrtc-audio-coding-1.so.3rootrootrootrootwebrtc-audio-processing-1.3-150600.1.3.src.rpmlibwebrtc-audio-coding-devellibwebrtc-audio-coding-devel(aarch-64)pkgconfig(webrtc-audio-coding-1)@@@@@@    /usr/bin/pkg-configlibwebrtc-audio-coding-1-3pkgconfig(absl_bad_optional_access)pkgconfig(absl_base)pkgconfig(absl_flags)pkgconfig(absl_strings)pkgconfig(absl_synchronization)rpmlib(CompressedFileNames)rpmlib(FileDigests)rpmlib(PayloadFilesHavePrefix)rpmlib(PayloadIsXz)1.33.0.4-14.6.0-14.0-15.2-14.14.3e?e e e d _:q@]D%XwoWnr@Wk@Wj}WgWL+@WF@alarrosa@suse.comalarrosa@suse.comalarrosa@suse.comalarrosa@suse.comalarrosa@suse.comdmueller@suse.commliska@suse.czolaf@aepfle.deoholecek@suse.comoholecek@suse.comoholecek@suse.comoholecek@suse.comoholecek@suse.comoholecek@suse.com- ExcludeArch s390, s390x and ppc64 since big endian support is not implemented.- Remove the tar.xz file. Having the obscpio file is enough- Use also dashes instead of underscores in the manual Requires- 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- 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- update to 0.3.1: * doc: file invalid reference to pulseaudio mailing list * various build system fixes - spec-cleaner run- Use FAT LTO objects in order to provide proper static library.- Add baselibs.conf for gstreamer-plugins-bad-32bit- Remove webrtc-aarch64.patch, no longer needed - Adapt the rest of webrtc- patches to new arch naming- Remove unneeded explicit version dependency for automake- 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- 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- 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- 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 patchesh04-armsrv1 17127552971.3-150600.1.31.3-150600.1.31.3libwebrtc-audio-coding-1.sowebrtc-audio-coding-1.pc/usr/lib64//usr/lib64/pkgconfig/-fmessage-length=0 -grecord-gcc-switches -O2 -Wall -D_FORTIFY_SOURCE=2 -fstack-protector-strong -funwind-tables -fasynchronous-unwind-tables -fstack-clash-protection -gobs://build.suse.de/SUSE:SLE-15-SP6:GA/standard/3359b25363aa46aa70faf001800fca6a-webrtc-audio-processingcpioxz5aarch64-suse-linuxpkgconfig filePRRRRRRPutf-806526ead130c290b938f228b86288eb72c96f0807815d0095a2b42e2f2957fad? 7zXZ !t/kp] cr$x#omЃ6VyO&@. `(pSjbiȅ^>v(3^@C6^ДbJښ.˻džACԏp2qhCm~//||_Q?ogzcwX'q=J{{I]!.*bq,0Aۓ@H pN<ڵ&a-SʍRs0\ӄC[ uҪS [AlNFKDМ>dޤ7z_@fZ8YY*KzpN\ί3.\re+\s2k.{P8hjc?'\ YZ