2012-05-15 12:42:12 +02:00
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# 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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2023-09-18 14:05:03 +02:00
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# Copyright (c) 2023 SUSE LLC
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2012-05-15 12:42:12 +02:00
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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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2019-08-02 10:26:30 +02:00
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# Please submit bugfixes or comments via https://bugs.opensuse.org/
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2012-05-15 12:42:12 +02:00
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#
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2023-09-18 14:05:03 +02:00
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%define pkg_soname 1-3
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%define soname 3
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2012-05-15 12:42:12 +02:00
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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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2023-09-18 14:05:03 +02:00
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Version: 1.3
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2012-05-15 12:42:12 +02:00
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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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2020-08-21 15:46:29 +02:00
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URL: https://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/
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2023-09-18 14:05:03 +02:00
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Source: webrtc-audio-processing-%{version}.tar.xz
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2017-01-19 14:38:47 +01:00
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Source1: baselibs.conf
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2023-09-18 14:05:03 +02:00
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# PATCH-FIX-UPSTREAM fix-build.patch alarrosa@suse.com -- Fix a number of "control reaches end of non-void function" errors
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Patch0: fix-build.patch
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Accepting request 404777 from home:oholecek:branches:multimedia:libs
- 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
2016-06-25 18:50:49 +02:00
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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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2023-09-18 14:05:03 +02:00
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Patch3: fix-i586.patch
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Accepting request 404777 from home:oholecek:branches:multimedia:libs
- 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
2016-06-25 18:50:49 +02:00
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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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2023-09-20 11:12:33 +02:00
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# PATCH-FIX-OPENSUSE reduce-meson-dep.patch
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Patch102: reduce-meson-dep.patch
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2023-09-18 14:05:03 +02:00
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BuildRequires: cmake
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2012-05-15 12:42:12 +02:00
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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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2023-09-20 11:12:33 +02:00
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BuildRequires: meson >= 0.59.4
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2012-05-15 12:42:12 +02:00
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BuildRequires: pkgconfig
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BuildRequires: xz
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2023-09-18 14:05:03 +02:00
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BuildRequires: cmake(absl)
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2023-10-30 17:45:37 +01:00
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ExcludeArch: s390 s390x ppc64
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2012-05-15 12:42:12 +02:00
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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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2023-09-20 11:12:33 +02:00
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%package -n libwebrtc-audio-processing-%{pkg_soname}
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2012-05-15 12:42:12 +02:00
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Summary: Real-Time Communication Library for Web Browsers
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Group: System/Libraries
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2023-09-20 11:12:33 +02:00
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%description -n libwebrtc-audio-processing-%{pkg_soname}
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2012-05-15 12:42:12 +02:00
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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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2023-09-20 11:12:33 +02:00
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%package -n libwebrtc-audio-processing-devel
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2012-05-15 12:42:12 +02:00
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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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2023-09-20 11:46:15 +02:00
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Requires: libwebrtc-audio-processing-%{pkg_soname} = %{version}
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2012-05-15 12:42:12 +02:00
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2023-09-20 11:12:33 +02:00
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%description -n libwebrtc-audio-processing-devel
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2012-05-15 12:42:12 +02:00
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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
|
|
|
|
|
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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2023-09-20 11:12:33 +02:00
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%package -n libwebrtc-audio-processing-devel-static
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2012-05-15 12:42:12 +02:00
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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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2023-09-20 11:39:27 +02:00
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Requires: libwebrtc-audio-processing-devel = %{version}
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2012-05-15 12:42:12 +02:00
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2023-09-20 11:12:33 +02:00
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%description -n libwebrtc-audio-processing-devel-static
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2012-05-15 12:42:12 +02:00
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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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2023-09-20 11:12:33 +02:00
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%package -n libwebrtc-audio-coding-%{pkg_soname}
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2023-09-18 14:05:03 +02:00
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Summary: Real-Time Communication Library for Web Browsers
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Group: System/Libraries
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2023-09-20 11:12:33 +02:00
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%description -n libwebrtc-audio-coding-%{pkg_soname}
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2023-09-18 14:05:03 +02:00
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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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2023-09-20 11:12:33 +02:00
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%package -n libwebrtc-audio-coding-devel
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2023-09-18 14:05:03 +02:00
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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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2023-09-20 11:46:15 +02:00
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Requires: libwebrtc-audio-coding-%{pkg_soname} = %{version}
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2023-09-18 14:05:03 +02:00
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2023-09-20 11:12:33 +02:00
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%description -n libwebrtc-audio-coding-devel
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2023-09-18 14:05:03 +02:00
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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
|
|
|
|
|
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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2023-09-20 11:12:33 +02:00
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%package -n libwebrtc-audio-coding-devel-static
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2023-09-18 14:05:03 +02:00
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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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2023-09-20 11:39:27 +02:00
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Requires: libwebrtc-audio-coding-devel = %{version}
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2023-09-18 14:05:03 +02:00
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2023-09-20 11:12:33 +02:00
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%description -n libwebrtc-audio-coding-devel-static
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2023-09-18 14:05:03 +02:00
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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.
|
|
|
|
|
|
|
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WebRTC implements the W3C's proposal for video conferencing on the web.
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2012-05-15 12:42:12 +02:00
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%prep
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2023-09-18 14:05:03 +02:00
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%autosetup -p1 -N
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Accepting request 404777 from home:oholecek:branches:multimedia:libs
- 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
2016-06-25 18:50:49 +02:00
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sed -i 's/\r$//' AUTHORS
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2024-02-20 18:21:56 +01:00
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%patch -P 0 -p1
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#%%patch -P 1 -p1
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#%%patch -P 2 -p1
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%patch -P 3 -p1
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%patch -P 100 -p1
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%patch -P 101 -p1
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%patch -P 102 -p1
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2012-05-15 12:42:12 +02:00
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%build
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2019-08-02 10:26:30 +02:00
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%global _lto_cflags %{_lto_cflags} -ffat-lto-objects
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2023-09-18 14:05:03 +02:00
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%meson \
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2023-09-20 11:12:33 +02:00
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-Dc_std=gnu11 \
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2023-09-18 14:05:03 +02:00
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-Dcpp_std=gnu++17 \
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-Ddefault_library=both \
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-Dc_args="${CFLAGS} ${LDFLAGS}" \
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-Dcpp_args="${CXXFLAGS} ${LDFLAGS}" \
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%{nil}
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%meson_build
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2012-05-15 12:42:12 +02:00
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%install
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2023-09-18 14:05:03 +02:00
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%meson_install
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2012-05-15 12:42:12 +02:00
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2020-08-21 15:46:29 +02:00
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find %{buildroot} -type f -name "*.la" -delete -print
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2012-05-15 12:42:12 +02:00
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2023-09-20 11:12:33 +02:00
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%post -n libwebrtc-audio-processing-%{pkg_soname} -p /sbin/ldconfig
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%postun -n libwebrtc-audio-processing-%{pkg_soname} -p /sbin/ldconfig
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%post -n libwebrtc-audio-coding-%{pkg_soname} -p /sbin/ldconfig
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%postun -n libwebrtc-audio-coding-%{pkg_soname} -p /sbin/ldconfig
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2012-05-15 12:42:12 +02:00
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2023-09-20 11:12:33 +02:00
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%files -n libwebrtc-audio-processing-%{pkg_soname}
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2020-08-21 15:46:29 +02:00
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%license COPYING
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%doc AUTHORS NEWS README.md UPDATING.md
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2023-09-18 14:05:03 +02:00
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%{_libdir}/libwebrtc-audio-processing-1.so.%{soname}*
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2012-05-15 12:42:12 +02:00
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2023-09-20 11:12:33 +02:00
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%files -n libwebrtc-audio-processing-devel
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2023-09-18 14:05:03 +02:00
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%{_includedir}/webrtc-audio-processing-1
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%{_libdir}/libwebrtc-audio-processing-1.so
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%{_libdir}/pkgconfig/webrtc-audio-processing-1.pc
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2012-05-15 12:42:12 +02:00
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2023-09-20 11:12:33 +02:00
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%files -n libwebrtc-audio-processing-devel-static
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2023-09-18 14:05:03 +02:00
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%{_libdir}/libwebrtc-audio-processing-1.a
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2023-09-20 11:12:33 +02:00
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%files -n libwebrtc-audio-coding-%{pkg_soname}
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2023-09-18 14:05:03 +02:00
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%license COPYING
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%doc AUTHORS NEWS README.md UPDATING.md
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%{_libdir}/libwebrtc-audio-coding-1.so.%{soname}*
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2023-09-20 11:12:33 +02:00
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%files -n libwebrtc-audio-coding-devel
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2023-09-18 14:05:03 +02:00
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%{_libdir}/libwebrtc-audio-coding-1.so
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%{_libdir}/pkgconfig/webrtc-audio-coding-1.pc
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2023-09-20 11:12:33 +02:00
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%files -n libwebrtc-audio-coding-devel-static
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2023-09-18 14:05:03 +02:00
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%{_libdir}/libwebrtc-audio-coding-1.a
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2012-05-15 12:42:12 +02:00
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%changelog
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