libwebrtc_audio_processing1-0.3-3.1.x86_64.rpm


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Description

libwebrtc_audio_processing1 - Real-Time Communication Library for Web Browsers

Distribution: openSUSE 42.1
Repository: Packman all
Package name: libwebrtc_audio_processing1
Package version: 0.3
Package release: 3.1
Package architecture: x86_64
Package type: rpm
Installed size: 738.12 KB
Download size: 272.96 KB
Official Mirror: packman.inode.at
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.

Alternatives

Provides

  • libwebrtc_audio_processing.so.1()(64bit)
  • libwebrtc_audio_processing1 = 0.3-3.1
  • libwebrtc_audio_processing1(x86-64) = 0.3-3.1

    Download

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

    Install Howto

    1. Add the Packman repository:
      # zypper addrepo http://packman.inode.at/suse/openSUSE_Leap_42.1/ packman
    2. Install libwebrtc_audio_processing1 rpm package:
      # zypper install libwebrtc_audio_processing1

    Files

    • /usr/lib64/libwebrtc_audio_processing.so.1
    • /usr/lib64/libwebrtc_audio_processing.so.1.0.0
    • /usr/share/doc/packages/libwebrtc_audio_processing1/AUTHORS
    • /usr/share/doc/packages/libwebrtc_audio_processing1/COPYING
    • /usr/share/doc/packages/libwebrtc_audio_processing1/NEWS
    • /usr/share/doc/packages/libwebrtc_audio_processing1/README.md
    • /usr/share/doc/packages/libwebrtc_audio_processing1/UPDATING.md

    Changelog

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

    2016-06-23 - oholecek@suse.com - Remove unneeded explicit version dependency for automake

    2016-06-22 - 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

    2016-06-20 - 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

    2016-05-30 - 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

    2016-05-26 - 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

    2013-03-07 - idonmez@suse.com - Add patch webrtc-aarch64.patch from algraf to add aarch64 support

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

    2012-07-03 - dvaleev@suse.com - add ppc64 to known platforms

    2012-05-15 - pascal.bleser@opensuse.org - initial version (0.1)

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