libwebrtc_audio_processing-devel-static - Real-Time Communication Library for Web Browsers

Distribution: openSUSE 42.1
Repository: Packman all
Package name: libwebrtc_audio_processing-devel-static
Package version: 0.3
Package release: 3.1
Package architecture: x86_64
Package type: rpm
Installed size: 13.72 MB
Download size: 2.14 MB
Official Mirror:
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.


  • libwebrtc_audio_processing-devel-static = 0.3-3.1
  • libwebrtc_audio_processing-devel-static(x86-64) = 0.3-3.1


    Source package: webrtc-audio-processing-0.3-3.1.src.rpm

    Install Howto

    1. Add the Packman repository:
      # zypper addrepo packman
    2. Install libwebrtc_audio_processing-devel-static rpm package:
      # zypper install libwebrtc_audio_processing-devel-static


    • /usr/lib64/libwebrtc_audio_processing.a


    2016-06-25 - - Remove webrtc-aarch64.patch, no longer needed - Adapt the rest of webrtc- patches to new arch naming

    2016-06-23 - - Remove unneeded explicit version dependency for automake

    2016-06-22 - - 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

    2016-06-20 - - 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

    2016-05-30 - - Add webrtc-audio-processing-0.2-x86_msse2.patch patch fixing 386 build - Add big_endian_support.patch - New automake version dependency >= 1.5

    2016-05-26 - - 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

    2013-03-07 - - Add patch webrtc-aarch64.patch from algraf to add aarch64 support

    2012-12-19 - - add s390 and s390x to known platforms by adding webrtc-s390x.patch

    2012-07-03 - - add ppc64 to known platforms

    2012-05-15 - - initial version (0.1)