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b U c d e f l u v w P x y z 0 4 : | C libwebrtc_audio_processing-devel 0.3 150000.3.2.1 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. eEs390zl32 SUSE Linux Enterprise 15 SUSE LLC BSD-3-Clause https://www.suse.com/ Development/Libraries/C and C++ http://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/ linux s390x )
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i z # 0AAA큤AAA큤A큤A큤AA큤 eEeEeEeEeEeEeEeEeEeEeEeEeEeEeEeEeEeEeEeEeEeEeEeEeE db2ab77edf46f6fd0963ed5829dee900ba832e549f3bca67cb3df9f23da3b806 1ae445c4c7247a3356d82af1d7e7cadc9ea137ad61b75c1bdb425451e5cfa144 a6cae1a2d2f05690a0db4c718bae29c2732e8fb187aa257b7644ae49da4bc0e1 63d04a9881213ec4b22a19e6dd35d26348b65120589d5bb6aa461e3df35f57e6 1fef86a982240ae0129c6e73b9873e002c0162a9dd7279cada6a9756ec7390bb bb14a1dc1b1c95d40b6478068803cff9da88b74e02d34c16c59c5784036e395f d70c64ece34181560db6e77c1b1fc29ada3b7dc499d2b2ef43cce13cb0127019 52e7653cbab9a18abc4e17892dc6a7744f11ba696d4127e4d4444d88b3c7d7cd fd3dbe641f1768a3c9ec1fb9d7083f8427c79b26df5e9452de8e0404e06f59e1 ce442ba4d40642933d20417bac4b858c533851e3504f2ce3ab962cb124cb99a8 d29f6010ba74970b4c6ca33c9beaae3f771ba7ae09203d26599ca754b1adf0d5 12e2a069d0cebed270bd3e6bc846013e8b845786c2509088a753ad5236c3f3a0 20c27e5204f54cd7117f37a68256a8e27f5f7ae5b51c1bbd314b25de96c17d9a 3b6f6c36baecb08b4ae5dd2e699ad3d2c4ac6584247dcd64aeb6774494bbc664 libwebrtc_audio_processing.so.1.0.0 root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root root webrtc-audio-processing-0.3-150000.3.2.1.src.rpm libwebrtc_audio_processing-devel libwebrtc_audio_processing-devel(s390-64) pkgconfig(webrtc-audio-processing) @
/usr/bin/pkg-config libwebrtc_audio_processing1 rpmlib(CompressedFileNames) rpmlib(FileDigests) rpmlib(PayloadFilesHavePrefix) rpmlib(PayloadIsXz) 0.3 3.0.4-1 4.6.0-1 4.0-1 5.2-1 4.14.1 XwoWnr@Wk@Wj}WgWL+@WF@Q8@PѬ@Oolaf@aepfle.de oholecek@suse.com oholecek@suse.com oholecek@suse.com oholecek@suse.com oholecek@suse.com oholecek@suse.com idonmez@suse.com ro@suse.de dvaleev@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 platforms s390zl32 1699025427
0.3-150000.3.2.1 0.3-150000.3.2.1 0.3
webrtc_audio_processing webrtc base arraysize.h basictypes.h checks.h constructormagic.h maybe.h platform_file.h common.h common_types.h modules audio_processing beamformer array_util.h include audio_processing.h interface module_common_types.h system_wrappers include trace.h typedefs.h libwebrtc_audio_processing.so webrtc-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 -g obs://build.suse.de/SUSE:Maintenance:30992/SUSE_SLE-15_Update/917280e4040e95932bbd51051f98f158-webrtc-audio-processing.SUSE_SLE-15_Update drpm xz 5 s390x-suse-linux directory C source, ASCII text C++ source, ASCII text pkgconfig file P R }ps07Öy' utf-8 d9b27da931e90035d5436af9bec6fcfa6846356de3df19e1a373c06a26da6765 ? 7zXZ
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