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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. eE�h04-ch1a �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 x86_64 � � )
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/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 Xwo�Wnr@Wk�@Wj}�Wg��WL+@WF�@Q8�@PѬ@O���olaf@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 h04-ch1a 1699025340
�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 x86_64-suse-linux directory C source, ASCII text C++ source, ASCII text pkgconfig file P R �ȡ�� J _�r� �� utf-8 6c1b0a06c6bfdc06cd78ccbc1e54c657ca435dfc76671f00b3710c603646a539 ? ��� �7zXZ
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