libwebrtc_audio_processing1-32bit-0.3-1.35 RPM for x86_64

From OpenSuSE Leap 15.4 for x86_64

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.

Provides

• libwebrtc_audio_processing1-32bit
• libwebrtc_audio_processing.so.1
• libwebrtc_audio_processing1-32bit(x86-32)

Requires

• /bin/sh
• libc.so.6
• libc.so.6(GLIBC_2.0)
• libc.so.6(GLIBC_2.1)
• libc.so.6(GLIBC_2.1.3)
• libc.so.6(GLIBC_2.17)
• libc.so.6(GLIBC_2.3.4)
• libc.so.6(GLIBC_2.4)
• libgcc_s.so.1
• libgcc_s.so.1(GCC_3.0)
• libgcc_s.so.1(GCC_4.0.0)
• libgcc_s.so.1(GLIBC_2.0)
• libm.so.6
• libm.so.6(GLIBC_2.0)
• libm.so.6(GLIBC_2.1)
• libpthread.so.0
• libpthread.so.0(GLIBC_2.0)
• libpthread.so.0(GLIBC_2.1)
• libpthread.so.0(GLIBC_2.3.2)
• libpthread.so.0(GLIBC_2.3.3)
• libstdc++.so.6
• libstdc++.so.6(CXXABI_1.3)
• libstdc++.so.6(CXXABI_1.3.8)
• libstdc++.so.6(GLIBCXX_3.4)
• libstdc++.so.6(GLIBCXX_3.4.11)
• libstdc++.so.6(GLIBCXX_3.4.15)
• libstdc++.so.6(GLIBCXX_3.4.20)
• libstdc++.so.6(GLIBCXX_3.4.21)
• libstdc++.so.6(GLIBCXX_3.4.9)
• rpmlib(CompressedFileNames) <= 3.0.4-1
• rpmlib(FileDigests) <= 4.6.0-1
• rpmlib(PayloadFilesHavePrefix) <= 4.0-1
• rpmlib(PayloadIsXz) <= 5.2-1

License

BSD-3-Clause

Changelog

* 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
* Thu Mar 07 2013 idonmez@suse.com
  - Add patch webrtc-aarch64.patch from algraf to add aarch64 support
* Wed Dec 19 2012 ro@suse.de
  - add s390 and s390x to known platforms
    by adding webrtc-s390x.patch
* Tue Jul 03 2012 dvaleev@suse.com
  - add ppc64 to known platforms
* Tue May 15 2012 pascal.bleser@opensuse.org
  - initial version (0.1)

Files

/usr/lib/libwebrtc_audio_processing.so.1
/usr/lib/libwebrtc_audio_processing.so.1.0.0

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Fabrice Bellet, Tue Apr 9 17:13:04 2024