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libwebrtc_audio_processing1-32bit-0.3-1.35 RPM for x86_64

From OpenSuSE Leap 15.4 for x86_64

Name: libwebrtc_audio_processing1-32bit Distribution: SUSE Linux Enterprise 15
Version: 0.3 Vendor: SUSE LLC <>
Release: 1.35 Build date: Fri May 25 20:09:13 2018
Group: System/Libraries Build host: sheep23
Size: 718432 Source RPM: webrtc-audio-processing-0.3-1.35.src.rpm
Summary: 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.






* Thu Jan 12 2017
  - Add baselibs.conf for gstreamer-plugins-bad-32bit
* Sat Jun 25 2016
  - Remove webrtc-aarch64.patch, no longer needed
  - Adapt the rest of webrtc- patches to new arch naming
* Thu Jun 23 2016
  - Remove unneeded explicit version dependency for automake
* Wed Jun 22 2016
  - 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
  - Add no-undefined.patch patch
  - Add big_endian_support_2.patch
  - Adapt webrtc-audio-processing-0.2-x86_msse2.patch to new version
  - Adapt big_endian_support.patch to new version
* Mon May 30 2016
  - Add webrtc-audio-processing-0.2-x86_msse2.patch patch fixing 386 build
  - Add big_endian_support.patch
  - New automake version dependency >= 1.5
* Thu May 26 2016
  - 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
    * 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
  - Add patch webrtc-aarch64.patch from algraf to add aarch64 support
* Wed Dec 19 2012
  - add s390 and s390x to known platforms
    by adding webrtc-s390x.patch
* Tue Jul 03 2012
  - add ppc64 to known platforms
* Tue May 15 2012
  - initial version (0.1)



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Fabrice Bellet, Tue Jul 9 16:00:25 2024