Sync from SUSE:SLFO:Main webrtc-audio-processing-0 revision 50bed665ff886f41a6b8a88c0523b1c9
This commit is contained in:
commit
cc396a0dcc
23
.gitattributes
vendored
Normal file
23
.gitattributes
vendored
Normal file
@ -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
|
1
baselibs.conf
Normal file
1
baselibs.conf
Normal file
@ -0,0 +1 @@
|
||||
libwebrtc_audio_processing1
|
90
big_endian_support.patch
Normal file
90
big_endian_support.patch
Normal file
@ -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<uint32_t>(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<uint32_t>(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<uint32_t>(written);
|
||||
RTC_CHECK(written <= std::numeric_limits<uint32_t>::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<char*>(&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<uint32_t>(a) << 24 )
|
||||
+ | (static_cast<uint32_t>(b) << 16)
|
||||
+ | (static_cast<uint32_t>(c) << 8)
|
||||
+ | (static_cast<uint32_t>(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<char*>(&x), 4);
|
||||
+}
|
||||
#endif
|
||||
|
||||
static inline uint32_t RiffChunkSize(uint32_t bytes_in_payload) {
|
24
big_endian_support_2.patch
Normal file
24
big_endian_support_2.patch
Normal file
@ -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))
|
BIN
webrtc-audio-processing-0.3.1.tar.xz
(Stored with Git LFS)
Normal file
BIN
webrtc-audio-processing-0.3.1.tar.xz
(Stored with Git LFS)
Normal file
Binary file not shown.
125
webrtc-audio-processing-0.changes
Normal file
125
webrtc-audio-processing-0.changes
Normal file
@ -0,0 +1,125 @@
|
||||
-------------------------------------------------------------------
|
||||
Thu Sep 28 09:56:45 UTC 2023 - Antonio Larrosa <alarrosa@suse.com>
|
||||
|
||||
- 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 <dmueller@suse.com>
|
||||
|
||||
- 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 <mliska@suse.cz>
|
||||
|
||||
- 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)
|
||||
|
138
webrtc-audio-processing-0.spec
Normal file
138
webrtc-audio-processing-0.spec
Normal file
@ -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 <pascal.bleser@opensuse.org>
|
||||
#
|
||||
# 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
|
17
webrtc-ppc64.patch
Normal file
17
webrtc-ppc64.patch
Normal file
@ -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__
|
15
webrtc-s390x.patch
Normal file
15
webrtc-s390x.patch
Normal file
@ -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__
|
Loading…
Reference in New Issue
Block a user