Takashi Iwai
d47a474aa6
- 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 OBS-URL: https://build.opensuse.org/request/show/404777 OBS-URL: https://build.opensuse.org/package/show/multimedia:libs/webrtc-audio-processing?expand=0&rev=11
129 lines
4.7 KiB
RPMSpec
129 lines
4.7 KiB
RPMSpec
# vim: set sw=4 ts=4 et nu:
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#
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# spec file for package webrtc-audio-processing
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#
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# Copyright (c) 2016 SUSE LINUX GmbH, Nuernberg, Germany.
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# Copyright (c) 2012 Pascal Bleser <pascal.bleser@opensuse.org>
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#
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# All modifications and additions to the file contributed by third parties
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# remain the property of their copyright owners, unless otherwise agreed
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# upon. The license for this file, and modifications and additions to the
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# file, is the same license as for the pristine package itself (unless the
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# license for the pristine package is not an Open Source License, in which
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# case the license is the MIT License). An "Open Source License" is a
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# license that conforms to the Open Source Definition (Version 1.9)
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# published by the Open Source Initiative.
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# Please submit bugfixes or comments via http://bugs.opensuse.org/
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#
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%define soname 1
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# Please submit bugfixes or comments via http://bugs.opensuse.org/
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Name: webrtc-audio-processing
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Version: 0.3
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Release: 0
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Summary: Real-Time Communication Library for Web Browsers
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License: BSD-3-Clause
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Group: System/Libraries
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Url: http://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/
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Source: http://freedesktop.org/software/pulseaudio/webrtc-audio-processing/webrtc-audio-processing-%{version}.tar.xz
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# PATCH-FIX-UPSTREAN big_endian_support.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738
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Patch1: big_endian_support.patch
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# PATCH-FIX-UPSTREAN big_endian_support.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738
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Patch2: big_endian_support_2.patch
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# PATCH-FIX-OPENSUSE webrtc-(ppc64|s390x|aarch64).patch
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Patch100: webrtc-ppc64.patch
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Patch101: webrtc-s390x.patch
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BuildRequires: autoconf
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BuildRequires: automake
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BuildRequires: gcc-c++
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BuildRequires: glibc-devel
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BuildRequires: libtool
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BuildRequires: make
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BuildRequires: pkgconfig
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BuildRequires: xz
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BuildRoot: %{_tmppath}/%{name}-%{version}-build
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%description
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WebRTC is an open source project that enables web browsers with Real-Time
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Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
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components have been optimized to best serve this purpose.
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WebRTC implements the W3C's proposal for video conferencing on the web.
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%package -n libwebrtc_audio_processing%{soname}
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Summary: Real-Time Communication Library for Web Browsers
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Group: System/Libraries
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%description -n libwebrtc_audio_processing%{soname}
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WebRTC is an open source project that enables web browsers with Real-Time
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Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
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components have been optimized to best serve this purpose.
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WebRTC implements the W3C's proposal for video conferencing on the web.
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%package -n libwebrtc_audio_processing-devel
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Summary: Real-Time Communication Library for Web Browsers
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Group: Development/Libraries/C and C++
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Requires: libwebrtc_audio_processing%{soname} = %{version}
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%description -n libwebrtc_audio_processing-devel
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WebRTC is an open source project that enables web browsers with Real-Time
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Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
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components have been optimized to best serve this purpose.
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WebRTC implements the W3C's proposal for video conferencing on the web.
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%package -n libwebrtc_audio_processing-devel-static
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Summary: Real-Time Communication Library for Web Browsers
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Group: Development/Libraries/C and C++
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Requires: libwebrtc_audio_processing-devel = %{version}
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%description -n libwebrtc_audio_processing-devel-static
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WebRTC is an open source project that enables web browsers with Real-Time
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Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
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components have been optimized to best serve this purpose.
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WebRTC implements the W3C's proposal for video conferencing on the web.
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%prep
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%setup -q -T -c "%{name}-%{version}"
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xz --decompress --stdout "%{SOURCE0}" | tar xf - --strip-components=1
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sed -i 's/\r$//' AUTHORS
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%patch1 -p1
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%patch2 -p1
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%patch100
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%patch101
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%build
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%configure
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make %{?_smp_mflags} V=1
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%install
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%makeinstall
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rm -f "%{buildroot}%{_libdir}"/*.la
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%post -n libwebrtc_audio_processing%{soname} -p /sbin/ldconfig
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%postun -n libwebrtc_audio_processing%{soname} -p /sbin/ldconfig
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%files -n libwebrtc_audio_processing%{soname}
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%defattr(-,root,root)
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%doc AUTHORS COPYING NEWS README.md UPDATING.md
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%{_libdir}/libwebrtc_audio_processing.so.%{soname}
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%{_libdir}/libwebrtc_audio_processing.so.%{soname}.*.*
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%files -n libwebrtc_audio_processing-devel
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%defattr(-,root,root)
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%{_includedir}/webrtc_audio_processing
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%{_libdir}/libwebrtc_audio_processing.so
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%{_libdir}/pkgconfig/webrtc-audio-processing.pc
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%files -n libwebrtc_audio_processing-devel-static
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%defattr(-,root,root)
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%{_libdir}/libwebrtc_audio_processing.a
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%changelog
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