From cc396a0dcc4645b5e428232b563bcc8317dd1e180d265ffe06c881c0da7d5b14 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Adrian=20Schr=C3=B6ter?= Date: Sat, 4 May 2024 01:52:29 +0200 Subject: [PATCH] Sync from SUSE:SLFO:Main webrtc-audio-processing-0 revision 50bed665ff886f41a6b8a88c0523b1c9 --- .gitattributes | 23 +++++ baselibs.conf | 1 + big_endian_support.patch | 90 +++++++++++++++++ big_endian_support_2.patch | 24 +++++ webrtc-audio-processing-0.3.1.tar.xz | 3 + webrtc-audio-processing-0.changes | 125 ++++++++++++++++++++++++ webrtc-audio-processing-0.spec | 138 +++++++++++++++++++++++++++ webrtc-ppc64.patch | 17 ++++ webrtc-s390x.patch | 15 +++ 9 files changed, 436 insertions(+) create mode 100644 .gitattributes create mode 100644 baselibs.conf create mode 100644 big_endian_support.patch create mode 100644 big_endian_support_2.patch create mode 100644 webrtc-audio-processing-0.3.1.tar.xz create mode 100644 webrtc-audio-processing-0.changes create mode 100644 webrtc-audio-processing-0.spec create mode 100644 webrtc-ppc64.patch create mode 100644 webrtc-s390x.patch diff --git a/.gitattributes b/.gitattributes new file mode 100644 index 0000000..9b03811 --- /dev/null +++ b/.gitattributes @@ -0,0 +1,23 @@ +## Default LFS +*.7z filter=lfs diff=lfs merge=lfs -text +*.bsp filter=lfs diff=lfs merge=lfs -text +*.bz2 filter=lfs diff=lfs merge=lfs -text +*.gem filter=lfs diff=lfs merge=lfs -text +*.gz filter=lfs diff=lfs merge=lfs -text +*.jar filter=lfs diff=lfs merge=lfs -text +*.lz filter=lfs diff=lfs merge=lfs -text +*.lzma filter=lfs diff=lfs merge=lfs -text +*.obscpio filter=lfs diff=lfs merge=lfs -text +*.oxt filter=lfs diff=lfs merge=lfs -text +*.pdf filter=lfs diff=lfs merge=lfs -text +*.png filter=lfs diff=lfs merge=lfs -text +*.rpm filter=lfs diff=lfs merge=lfs -text +*.tbz filter=lfs diff=lfs merge=lfs -text +*.tbz2 filter=lfs diff=lfs merge=lfs -text +*.tgz filter=lfs diff=lfs merge=lfs -text +*.ttf filter=lfs diff=lfs merge=lfs -text +*.txz filter=lfs diff=lfs merge=lfs -text +*.whl filter=lfs diff=lfs merge=lfs -text +*.xz filter=lfs diff=lfs merge=lfs -text +*.zip filter=lfs diff=lfs merge=lfs -text +*.zst filter=lfs diff=lfs merge=lfs -text diff --git a/baselibs.conf b/baselibs.conf new file mode 100644 index 0000000..9e92647 --- /dev/null +++ b/baselibs.conf @@ -0,0 +1 @@ +libwebrtc_audio_processing1 diff --git a/big_endian_support.patch b/big_endian_support.patch new file mode 100644 index 0000000..26850f7 --- /dev/null +++ b/big_endian_support.patch @@ -0,0 +1,90 @@ +diff -up webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc.than webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc +--- webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc.than 2016-05-24 08:28:45.749940095 -0400 ++++ webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc 2016-05-24 08:50:30.361020010 -0400 +@@ -64,9 +64,6 @@ WavReader::~WavReader() { + } + + size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) { +-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN +-#error "Need to convert samples to big-endian when reading from WAV file" +-#endif + // There could be metadata after the audio; ensure we don't read it. + num_samples = std::min(rtc::checked_cast(num_samples), + num_samples_remaining_); +@@ -76,6 +73,12 @@ size_t WavReader::ReadSamples(size_t num + RTC_CHECK(read == num_samples || feof(file_handle_)); + RTC_CHECK_LE(read, num_samples_remaining_); + num_samples_remaining_ -= rtc::checked_cast(read); ++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN ++ //convert to big-endian ++ for(size_t idx = 0; idx < num_samples; idx++) { ++ samples[idx] = (samples[idx]<<8) | (samples[idx]>>8); ++ } ++#endif + return read; + } + +@@ -120,10 +123,17 @@ WavWriter::~WavWriter() { + + void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) { + #ifndef WEBRTC_ARCH_LITTLE_ENDIAN +-#error "Need to convert samples to little-endian when writing to WAV file" +-#endif ++ int16_t * le_samples = new int16_t[num_samples]; ++ for(size_t idx = 0; idx < num_samples; idx++) { ++ le_samples[idx] = (samples[idx]<<8) | (samples[idx]>>8); ++ } ++ const size_t written = ++ fwrite(le_samples, sizeof(*le_samples), num_samples, file_handle_); ++ delete []le_samples; ++#else + const size_t written = + fwrite(samples, sizeof(*samples), num_samples, file_handle_); ++#endif + RTC_CHECK_EQ(num_samples, written); + num_samples_ += static_cast(written); + RTC_CHECK(written <= std::numeric_limits::max() || +diff -up webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc.than webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc +--- webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc.than 2016-05-24 08:50:52.591379263 -0400 ++++ webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc 2016-05-24 08:52:08.552606848 -0400 +@@ -129,7 +129,39 @@ static inline std::string ReadFourCC(uin + return std::string(reinterpret_cast(&x), 4); + } + #else +-#error "Write be-to-le conversion functions" ++static inline void WriteLE16(uint16_t* f, uint16_t x) { ++ *f = ((x << 8) & 0xff00) | ( ( x >> 8) & 0x00ff); ++} ++ ++static inline void WriteLE32(uint32_t* f, uint32_t x) { ++ *f = ( (x & 0x000000ff) << 24 ) ++ | ((x & 0x0000ff00) << 8) ++ | ((x & 0x00ff0000) >> 8) ++ | ((x & 0xff000000) >> 24 ); ++} ++ ++static inline void WriteFourCC(uint32_t* f, char a, char b, char c, char d) { ++ *f = (static_cast(a) << 24 ) ++ | (static_cast(b) << 16) ++ | (static_cast(c) << 8) ++ | (static_cast(d) ); ++} ++ ++static inline uint16_t ReadLE16(uint16_t x) { ++ return (( x & 0x00ff) << 8 )| ((x & 0xff00)>>8); ++} ++ ++static inline uint32_t ReadLE32(uint32_t x) { ++ return ( (x & 0x000000ff) << 24 ) ++ | ( (x & 0x0000ff00) << 8 ) ++ | ( (x & 0x00ff0000) >> 8) ++ | ( (x & 0xff000000) >> 24 ); ++} ++ ++static inline std::string ReadFourCC(uint32_t x) { ++ x = ReadLE32(x); ++ return std::string(reinterpret_cast(&x), 4); ++} + #endif + + static inline uint32_t RiffChunkSize(uint32_t bytes_in_payload) { diff --git a/big_endian_support_2.patch b/big_endian_support_2.patch new file mode 100644 index 0000000..f38262a --- /dev/null +++ b/big_endian_support_2.patch @@ -0,0 +1,24 @@ +diff -up webrtc-audio-processing-0.2/webrtc/typedefs.h.typedef webrtc-audio-processing-0.2/webrtc/typedefs.h +--- webrtc-audio-processing-0.2/webrtc/typedefs.h.typedef 2016-05-12 09:08:53.885000410 -0500 ++++ webrtc-audio-processing-0.2/webrtc/typedefs.h 2016-05-12 09:12:38.006851953 -0500 +@@ -48,7 +48,19 @@ + #define WEBRTC_ARCH_32_BITS + #define WEBRTC_ARCH_LITTLE_ENDIAN + #else +-#error Please add support for your architecture in typedefs.h ++/* instead of failing, use typical unix defines... */ ++#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__ ++#define WEBRTC_ARCH_LITTLE_ENDIAN ++#elif __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__ ++#define WEBRTC_ARCH_BIG_ENDIAN ++#else ++#error __BYTE_ORDER__ is not defined ++#endif ++#if defined(__LP64__) ++#define WEBRTC_ARCH_64_BITS ++#else ++#define WEBRTC_ARCH_32_BITS ++#endif + #endif + + #if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN)) diff --git a/webrtc-audio-processing-0.3.1.tar.xz b/webrtc-audio-processing-0.3.1.tar.xz new file mode 100644 index 0000000..d35a51c --- /dev/null +++ b/webrtc-audio-processing-0.3.1.tar.xz @@ -0,0 +1,3 @@ +version https://git-lfs.github.com/spec/v1 +oid sha256:a0fdd938fd85272d67e81572c5a4d9e200a0c104753cb3c209ded175ce3c5dbf +size 695920 diff --git a/webrtc-audio-processing-0.changes b/webrtc-audio-processing-0.changes new file mode 100644 index 0000000..1fd9714 --- /dev/null +++ b/webrtc-audio-processing-0.changes @@ -0,0 +1,125 @@ +------------------------------------------------------------------- +Thu Sep 28 09:56:45 UTC 2023 - Antonio Larrosa + +- Rename the 0.3.1 version of the package to + webrtc-audio-processing-0 so we can keep it around while all + applications are ported to version 1.x (like baresip and dino). + There's no need to rename the devel package since the new version + uses dashes instead of underscores in the package name. + +------------------------------------------------------------------- +Mon Aug 17 15:30:03 UTC 2020 - Dirk Mueller + +- update to 0.3.1: + * doc: file invalid reference to pulseaudio mailing list + * various build system fixes +- spec-cleaner run + +------------------------------------------------------------------- +Fri Aug 2 08:23:00 UTC 2019 - Martin Liška + +- Use FAT LTO objects in order to provide proper static library. + +------------------------------------------------------------------- +Thu Jan 12 08:32:04 UTC 2017 - olaf@aepfle.de + +- Add baselibs.conf for gstreamer-plugins-bad-32bit + +------------------------------------------------------------------- +Sat Jun 25 10:39:08 UTC 2016 - oholecek@suse.com + +- Remove webrtc-aarch64.patch, no longer needed +- Adapt the rest of webrtc- patches to new arch naming + +------------------------------------------------------------------- +Thu Jun 23 13:31:14 UTC 2016 - oholecek@suse.com + +- Remove unneeded explicit version dependency for automake + +------------------------------------------------------------------- +Wed Jun 22 11:55:11 UTC 2016 - 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 +------------------------------------------------------------------- +Mon Jun 20 13:02:06 UTC 2016 - 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 + +------------------------------------------------------------------- +Mon May 30 09:00:51 UTC 2016 - 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 + +------------------------------------------------------------------- +Thu May 26 21:19:28 UTC 2016 - 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 + +------------------------------------------------------------------- +Thu Mar 7 13:51:31 UTC 2013 - idonmez@suse.com + +- Add patch webrtc-aarch64.patch from algraf to add aarch64 support + +------------------------------------------------------------------- +Wed Dec 19 10:39:23 CET 2012 - ro@suse.de + +- add s390 and s390x to known platforms + by adding webrtc-s390x.patch + +------------------------------------------------------------------- +Tue Jul 3 15:00:06 UTC 2012 - dvaleev@suse.com + +- add ppc64 to known platforms + +------------------------------------------------------------------- +Tue May 15 10:40:38 CET 2012 - pascal.bleser@opensuse.org + +- initial version (0.1) + diff --git a/webrtc-audio-processing-0.spec b/webrtc-audio-processing-0.spec new file mode 100644 index 0000000..3d41993 --- /dev/null +++ b/webrtc-audio-processing-0.spec @@ -0,0 +1,138 @@ +# vim: set sw=4 ts=4 et nu: +# +# spec file for package webrtc-audio-processing +# +# Copyright (c) 2020 SUSE LLC +# Copyright (c) 2012 Pascal Bleser +# +# All modifications and additions to the file contributed by third parties +# remain the property of their copyright owners, unless otherwise agreed +# upon. The license for this file, and modifications and additions to the +# file, is the same license as for the pristine package itself (unless the +# license for the pristine package is not an Open Source License, in which +# case the license is the MIT License). An "Open Source License" is a +# license that conforms to the Open Source Definition (Version 1.9) +# published by the Open Source Initiative. + +# Please submit bugfixes or comments via https://bugs.opensuse.org/ +# + + +%define soname 1 +# Please submit bugfixes or comments via http://bugs.opensuse.org/ +Name: webrtc-audio-processing-0 +Version: 0.3.1 +Release: 0 +Summary: Real-Time Communication Library for Web Browsers +License: BSD-3-Clause +Group: System/Libraries +URL: https://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/ +Source: http://freedesktop.org/software/pulseaudio/webrtc-audio-processing/webrtc-audio-processing-%{version}.tar.xz +Source1: baselibs.conf +# PATCH-FIX-UPSTREAN big_endian_support.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738 +Patch1: big_endian_support.patch +# PATCH-FIX-UPSTREAN big_endian_support.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738 +Patch2: big_endian_support_2.patch +# PATCH-FIX-OPENSUSE webrtc-(ppc64|s390x|aarch64).patch +Patch100: webrtc-ppc64.patch +Patch101: webrtc-s390x.patch +BuildRequires: autoconf +BuildRequires: automake +BuildRequires: gcc-c++ +BuildRequires: glibc-devel +BuildRequires: libtool +BuildRequires: make +BuildRequires: pkgconfig +BuildRequires: xz + +%description +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. + +This is a compatibility package which should only be used by applications +that haven't be updated yet to the newer 1.x version. + +%package -n libwebrtc_audio_processing%{soname} +Summary: Real-Time Communication Library for Web Browsers +Group: System/Libraries + +%description -n libwebrtc_audio_processing%{soname} +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. + +This is a compatibility package which should only be used by applications +that haven't be updated yet to the newer 1.x version. + +%package -n libwebrtc_audio_processing-devel +Summary: Real-Time Communication Library for Web Browsers +Group: Development/Libraries/C and C++ +Requires: libwebrtc_audio_processing%{soname} = %{version} + +%description -n libwebrtc_audio_processing-devel +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. + +This is a compatibility package which should only be used by applications +that haven't be updated yet to the newer 1.x version. + +%package -n libwebrtc_audio_processing-devel-static +Summary: Real-Time Communication Library for Web Browsers +Group: Development/Libraries/C and C++ +Requires: libwebrtc_audio_processing-devel = %{version} + +%description -n libwebrtc_audio_processing-devel-static +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. + +This is a compatibility package which should only be used by applications +that haven't be updated yet to the newer 1.x version. + +%prep +%setup -q -T -c "%{name}-%{version}" +xz --decompress --stdout "%{SOURCE0}" | tar xf - --strip-components=1 +sed -i 's/\r$//' AUTHORS +%patch1 -p1 +%patch2 -p1 +%patch100 +%patch101 + +%build +%global _lto_cflags %{_lto_cflags} -ffat-lto-objects +%configure +%make_build + +%install +%make_install + +find %{buildroot} -type f -name "*.la" -delete -print + +%post -n libwebrtc_audio_processing%{soname} -p /sbin/ldconfig +%postun -n libwebrtc_audio_processing%{soname} -p /sbin/ldconfig + +%files -n libwebrtc_audio_processing%{soname} +%license COPYING +%doc AUTHORS NEWS README.md UPDATING.md +%{_libdir}/libwebrtc_audio_processing.so.%{soname} +%{_libdir}/libwebrtc_audio_processing.so.%{soname}.*.* + +%files -n libwebrtc_audio_processing-devel +%{_includedir}/webrtc_audio_processing +%{_libdir}/libwebrtc_audio_processing.so +%{_libdir}/pkgconfig/webrtc-audio-processing.pc + +%files -n libwebrtc_audio_processing-devel-static +%{_libdir}/libwebrtc_audio_processing.a + +%changelog diff --git a/webrtc-ppc64.patch b/webrtc-ppc64.patch new file mode 100644 index 0000000..28dad72 --- /dev/null +++ b/webrtc-ppc64.patch @@ -0,0 +1,17 @@ +Index: webrtc/typedefs.h +=================================================================== +--- webrtc/typedefs.h.org ++++ webrtc/typedefs.h +@@ -47,6 +47,12 @@ + #elif defined(__pnacl__) + #define WEBRTC_ARCH_32_BITS + #define WEBRTC_ARCH_LITTLE_ENDIAN ++#elif defined(__powerpc64__) ++#define WEBRTC_ARCH_BIG_ENDIAN ++#define WEBRTC_ARCH_64_BITS ++#elif defined(__powerpc__) ++#define WEBRTC_ARCH_BIG_ENDIAN ++#define WEBRTC_ARCH_32_BITS + #else + /* instead of failing, use typical unix defines... */ + #if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__ diff --git a/webrtc-s390x.patch b/webrtc-s390x.patch new file mode 100644 index 0000000..1ae3523 --- /dev/null +++ b/webrtc-s390x.patch @@ -0,0 +1,15 @@ +--- webrtc/typedefs.h ++++ webrtc/typedefs.h +@@ -53,6 +53,12 @@ + #elif defined(__powerpc__) + #define WEBRTC_ARCH_BIG_ENDIAN + #define WEBRTC_ARCH_32_BITS ++#elif defined(__s390x__) ++#define WEBRTC_ARCH_BIG_ENDIAN ++#define WEBRTC_ARCH_64_BITS ++#elif defined(__s390__) ++#define WEBRTC_ARCH_BIG_ENDIAN ++#define WEBRTC_ARCH_32_BITS + #else + /* instead of failing, use typical unix defines... */ + #if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__