libwebrtc_audio_processing-devel-0.3-150000.3.2.1<>,($eEp9|XM+VpvdZ${PBSv7s6St"w+jڔ%. >`79J!VvW 5Wbdt xh16`߲(boDP;eH ń:_̽z|dY{Ĕ9 jdWn2X#$ }Ē ݝ,&>̥@{xV?o2tвh)m `e?xǐ{>>?d#' 4 e|  U[d  ,  ) hI \   w( 8 9 : p FGHTIXY\]l^ bUcdefluvwPxyz 04:|Clibwebrtc_audio_processing-devel0.3150000.3.2.1Real-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.eEh04-ch1aSUSE Linux Enterprise 15SUSE LLC BSD-3-Clausehttps://www.suse.com/Development/Libraries/C and C++http://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/linuxx86_64)   l iz#0AAA큤AAA큤A큤A큤AA큤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_audio_processing.so.1.0.0rootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootrootwebrtc-audio-processing-0.3-150000.3.2.1.src.rpmlibwebrtc_audio_processing-devellibwebrtc_audio_processing-devel(x86-64)pkgconfig(webrtc-audio-processing)@    /usr/bin/pkg-configlibwebrtc_audio_processing1rpmlib(CompressedFileNames)rpmlib(FileDigests)rpmlib(PayloadFilesHavePrefix)rpmlib(PayloadIsXz)0.33.0.4-14.6.0-14.0-15.2-14.14.1XwoWnr@Wk@Wj}WgWL+@WF@Q8@PѬ@Oolaf@aepfle.deoholecek@suse.comoholecek@suse.comoholecek@suse.comoholecek@suse.comoholecek@suse.comoholecek@suse.comidonmez@suse.comro@suse.dedvaleev@suse.com- 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 patches- Add patch webrtc-aarch64.patch from algraf to add aarch64 support- add s390 and s390x to known platforms by adding webrtc-s390x.patch- add ppc64 to known platformsh04-ch1a 1699025340 0.3-150000.3.2.10.3-150000.3.2.10.3  webrtc_audio_processingwebrtcbasearraysize.hbasictypes.hchecks.hconstructormagic.hmaybe.hplatform_file.hcommon.hcommon_types.hmodulesaudio_processingbeamformerarray_util.hincludeaudio_processing.hinterfacemodule_common_types.hsystem_wrappersincludetrace.htypedefs.hlibwebrtc_audio_processing.sowebrtc-audio-processing.pc/usr/include//usr/include/webrtc_audio_processing//usr/include/webrtc_audio_processing/webrtc//usr/include/webrtc_audio_processing/webrtc/base//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/include//usr/include/webrtc_audio_processing/webrtc/modules/interface//usr/include/webrtc_audio_processing/webrtc/system_wrappers//usr/include/webrtc_audio_processing/webrtc/system_wrappers/include//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:Maintenance:30992/SUSE_SLE-15_Update/917280e4040e95932bbd51051f98f158-webrtc-audio-processing.SUSE_SLE-15_Updatedrpmxz5x86_64-suse-linuxdirectoryC source, ASCII textC++ source, ASCII textpkgconfig filePRȡJ_rutf-86c1b0a06c6bfdc06cd78ccbc1e54c657ca435dfc76671f00b3710c603646a539? 7zXZ !t/?I]"k%Ǜ#|8lq{NJ8iYr6\ Hч mDh WwJ{|̎lNY?T}IPϕs @6<&I(>u|$-l A,ax$% W ҺZ(@~}PO!x*!e$Uz iq7\49mW"XrIE9 M&loWTQW+7ȹ{XL 矮ӝ\Ox_7![>*~9@wBF!LBM?J'CSyDڳp##]~Vi@K=e'_1>!%}4Ɍw$X /bh3V5 }~\GeRؼZv+~񔖳K?k{?JuB²vKG.S]lIj~9u\ߡ] ?8yؠ×]"n[i"j9OPK5î3>L{ >ot֊)+wϤpߜ8XoTLtbxG7}瞉7֗+mh;Ӡ؁햲at9Q Qԃ qhЃ)m̶ mQoxM;^J!]_fM㖡^ƙ; =֬.POl w