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Hello community,
here is the log from the commit of package webrtc-audio-processing for
openSUSE:Factory checked in at 2023-09-21 22:13:13
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Comparing /work/SRC/openSUSE:Factory/webrtc-audio-processing (Old)
and /work/SRC/openSUSE:Factory/.webrtc-audio-processing.new.1770 (New)
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Package is "webrtc-audio-processing"
Thu Sep 21 22:13:13 2023 rev:13 rq:1112519 version:1.3
Changes:
--------
---
/work/SRC/openSUSE:Factory/webrtc-audio-processing/webrtc-audio-processing.changes
2020-09-01 20:01:42.640433654 +0200
+++
/work/SRC/openSUSE:Factory/.webrtc-audio-processing.new.1770/webrtc-audio-processing.changes
2023-09-21 22:13:30.316085079 +0200
@@ -1,0 +2,50 @@
+Wed Sep 20 09:49:19 UTC 2023 - Antonio Larrosa <alarr...@suse.com>
+
+- Remove the tar.xz file. Having the obscpio file is enough
+
+-------------------------------------------------------------------
+Wed Sep 20 09:38:21 UTC 2023 - Antonio Larrosa <alarr...@suse.com>
+
+- Use also dashes instead of underscores in the manual Requires
+
+-------------------------------------------------------------------
+Wed Sep 20 09:04:13 UTC 2023 - Antonio Larrosa <alarr...@suse.com>
+
+- Rename the generated library package names to add a dash between
+ the name and soname (libwebrtc*-1-3 instead of libwebrtc*1-3)
+- Rename the generated packages to use dashes instead of underscores
+- Change baselibs.conf accordingly
+- Add patch to reduce the required meson version so the package
+ builds in Leap 15.4/15.5:
+ * reduce-meson-dep.patch
+
+-------------------------------------------------------------------
+Fri Sep 08 10:40:12 UTC 2023 - alarr...@suse.com
+
+- Update to version 1.3:
+ * build: Bump version to 1.3
+ * meson: Fix generation of pkgconfig files
+ * build: Bump version to 1.2
+ * meson: Update minimum version based on what abseil wrap needs
+ * build: Expose absl as a dependency of webrtc-audio-processing
+ * meson: Update to latest wrap, install required absl headers
+ * doc: Update tarball generation process
+ * file_utils.h: Fix build with gcc-13
+ * meson: Fixes for MSVC build
+ * meson: Ensure that abseil is built with c++17 too
+ * More changes not listed by upstream. Check
+ the following link to see them:
+
https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/commits/v1.3
+- Add patch that fixes some compiler "control reaches end of
+ non-void function" errors:
+ * fix-build.patch
+- Add patch that fixes i586 build:
+ * fix-i586.patch
+- Disable patches until they're rebased to the current codebase:
+ * big_endian_support.patch
+ * big_endian_support_2.patch
+- Rebased patches:
+ * webrtc-ppc64.patch
+ * webrtc-s390x.patch
+
+-------------------------------------------------------------------
Old:
----
webrtc-audio-processing-0.3.1.tar.xz
New:
----
_service
fix-build.patch
fix-i586.patch
reduce-meson-dep.patch
webrtc-audio-processing-1.3.obscpio
webrtc-audio-processing.obsinfo
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Other differences:
------------------
++++++ webrtc-audio-processing.spec ++++++
--- /var/tmp/diff_new_pack.b5G3qX/_old 2023-09-21 22:13:32.032147360 +0200
+++ /var/tmp/diff_new_pack.b5G3qX/_new 2023-09-21 22:13:32.036147506 +0200
@@ -2,7 +2,7 @@
#
# spec file for package webrtc-audio-processing
#
-# Copyright (c) 2020 SUSE LLC
+# Copyright (c) 2023 SUSE LLC
# Copyright (c) 2012 Pascal Bleser <pascal.ble...@opensuse.org>
#
# All modifications and additions to the file contributed by third parties
@@ -18,32 +18,39 @@
#
-%define soname 1
+%define pkg_soname 1-3
+%define soname 3
# Please submit bugfixes or comments via http://bugs.opensuse.org/
Name: webrtc-audio-processing
-Version: 0.3.1
+Version: 1.3
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
+Source: webrtc-audio-processing-%{version}.tar.xz
Source1: baselibs.conf
+# PATCH-FIX-UPSTREAM fix-build.patch alarr...@suse.com -- Fix a number of
"control reaches end of non-void function" errors
+Patch0: fix-build.patch
# 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
+Patch3: fix-i586.patch
# PATCH-FIX-OPENSUSE webrtc-(ppc64|s390x|aarch64).patch
Patch100: webrtc-ppc64.patch
Patch101: webrtc-s390x.patch
-BuildRequires: autoconf
-BuildRequires: automake
+# PATCH-FIX-OPENSUSE reduce-meson-dep.patch
+Patch102: reduce-meson-dep.patch
+BuildRequires: cmake
BuildRequires: gcc-c++
BuildRequires: glibc-devel
BuildRequires: libtool
BuildRequires: make
+BuildRequires: meson >= 0.59.4
BuildRequires: pkgconfig
BuildRequires: xz
+BuildRequires: cmake(absl)
%description
WebRTC is an open source project that enables web browsers with Real-Time
@@ -52,35 +59,70 @@
WebRTC implements the W3C's proposal for video conferencing on the web.
-%package -n libwebrtc_audio_processing%{soname}
+%package -n libwebrtc-audio-processing-%{pkg_soname}
Summary: Real-Time Communication Library for Web Browsers
Group: System/Libraries
-%description -n libwebrtc_audio_processing%{soname}
+%description -n libwebrtc-audio-processing-%{pkg_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.
-%package -n libwebrtc_audio_processing-devel
+%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}
+Requires: libwebrtc-audio-processing-%{pkg_soname} = %{version}
-%description -n libwebrtc_audio_processing-devel
+%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.
-%package -n libwebrtc_audio_processing-devel-static
+%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}
+Requires: libwebrtc-audio-processing-devel = %{version}
-%description -n libwebrtc_audio_processing-devel-static
+%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.
+
+%package -n libwebrtc-audio-coding-%{pkg_soname}
+Summary: Real-Time Communication Library for Web Browsers
+Group: System/Libraries
+
+%description -n libwebrtc-audio-coding-%{pkg_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.
+
+%package -n libwebrtc-audio-coding-devel
+Summary: Real-Time Communication Library for Web Browsers
+Group: Development/Libraries/C and C++
+Requires: libwebrtc-audio-coding-%{pkg_soname} = %{version}
+
+%description -n libwebrtc-audio-coding-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.
+
+%package -n libwebrtc-audio-coding-devel-static
+Summary: Real-Time Communication Library for Web Browsers
+Group: Development/Libraries/C and C++
+Requires: libwebrtc-audio-coding-devel = %{version}
+
+%description -n libwebrtc-audio-coding-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.
@@ -88,38 +130,59 @@
WebRTC implements the W3C's proposal for video conferencing on the web.
%prep
-%setup -q -T -c "%{name}-%{version}"
-xz --decompress --stdout "%{SOURCE0}" | tar xf - --strip-components=1
+%autosetup -p1 -N
sed -i 's/\r$//' AUTHORS
-%patch1 -p1
-%patch2 -p1
-%patch100
-%patch101
+%patch0 -p1
+#%%patch1 -p1
+#%%patch2 -p1
+%patch3 -p1
+%patch100 -p1
+%patch101 -p1
+%patch102 -p1
%build
%global _lto_cflags %{_lto_cflags} -ffat-lto-objects
-%configure
-%make_build
+%meson \
+ -Dc_std=gnu11 \
+ -Dcpp_std=gnu++17 \
+ -Ddefault_library=both \
+ -Dc_args="${CFLAGS} ${LDFLAGS}" \
+ -Dcpp_args="${CXXFLAGS} ${LDFLAGS}" \
+ %{nil}
+%meson_build
%install
-%make_install
+%meson_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
+%post -n libwebrtc-audio-processing-%{pkg_soname} -p /sbin/ldconfig
+%postun -n libwebrtc-audio-processing-%{pkg_soname} -p /sbin/ldconfig
+%post -n libwebrtc-audio-coding-%{pkg_soname} -p /sbin/ldconfig
+%postun -n libwebrtc-audio-coding-%{pkg_soname} -p /sbin/ldconfig
+
+%files -n libwebrtc-audio-processing-%{pkg_soname}
+%license COPYING
+%doc AUTHORS NEWS README.md UPDATING.md
+%{_libdir}/libwebrtc-audio-processing-1.so.%{soname}*
+
+%files -n libwebrtc-audio-processing-devel
+%{_includedir}/webrtc-audio-processing-1
+%{_libdir}/libwebrtc-audio-processing-1.so
+%{_libdir}/pkgconfig/webrtc-audio-processing-1.pc
+
+%files -n libwebrtc-audio-processing-devel-static
+%{_libdir}/libwebrtc-audio-processing-1.a
-%files -n libwebrtc_audio_processing%{soname}
+%files -n libwebrtc-audio-coding-%{pkg_soname}
%license COPYING
%doc AUTHORS NEWS README.md UPDATING.md
-%{_libdir}/libwebrtc_audio_processing.so.%{soname}
-%{_libdir}/libwebrtc_audio_processing.so.%{soname}.*.*
+%{_libdir}/libwebrtc-audio-coding-1.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-coding-devel
+%{_libdir}/libwebrtc-audio-coding-1.so
+%{_libdir}/pkgconfig/webrtc-audio-coding-1.pc
-%files -n libwebrtc_audio_processing-devel-static
-%{_libdir}/libwebrtc_audio_processing.a
+%files -n libwebrtc-audio-coding-devel-static
+%{_libdir}/libwebrtc-audio-coding-1.a
++++++ _service ++++++
<?xml version="1.0"?>
<services>
<service name="obs_scm" mode="manual">
<param name="scm">git</param>
<param
name="url">https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing.git</param>
<param name="revision">v1.3</param>
<param name="versionformat">1.3</param>
<!--
<param name="revision">master</param>
<param name="versionformat">@PARENT_TAG@+git%cd.%h</param>
-->
</service>
<service name="tar" mode="buildtime"/>
<service name="recompress" mode="buildtime">
<param name="file">*.tar</param>
<param name="compression">xz</param>
</service>
<service name="set_version" mode="manual" />
</services>
++++++ baselibs.conf ++++++
--- /var/tmp/diff_new_pack.b5G3qX/_old 2023-09-21 22:13:32.072148812 +0200
+++ /var/tmp/diff_new_pack.b5G3qX/_new 2023-09-21 22:13:32.076148957 +0200
@@ -1,2 +1,3 @@
-libwebrtc_audio_processing1
+libwebrtc-audio-processing-1-3
+libwebrtc-audio-coding-1-3
++++++ big_endian_support.patch ++++++
--- /var/tmp/diff_new_pack.b5G3qX/_old 2023-09-21 22:13:32.088149393 +0200
+++ /var/tmp/diff_new_pack.b5G3qX/_new 2023-09-21 22:13:32.092149538 +0200
@@ -2,26 +2,26 @@
--- 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) {
+ size_t WavReader::ReadSamples(const size_t num_samples,
+ int16_t* const 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_);
+
+ size_t num_samples_left_to_read = num_samples;
+ size_t next_chunk_start = 0;
@@ -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);
+ num_samples_left_to_read -= num_samples_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;
+ return num_samples - num_samples_left_to_read;
}
@@ -120,10 +123,17 @@ WavWriter::~WavWriter() {
++++++ fix-build.patch ++++++
Index:
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
===================================================================
---
webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
+++
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
@@ -39,6 +39,7 @@ float GetLevel(const VadLevelAnalyzer::R
return vad_level.rms_dbfs;
break;
case LevelEstimatorType::kPeak:
+ default:
return vad_level.peak_dbfs;
break;
}
Index:
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/audio_processing_impl.cc
===================================================================
---
webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/audio_processing_impl.cc
+++
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -112,6 +112,7 @@ GainControl::Mode Agc1ConfigModeToInterf
case Agc1Config::kAdaptiveDigital:
return GainControl::kAdaptiveDigital;
case Agc1Config::kFixedDigital:
+ default:
return GainControl::kFixedDigital;
}
}
@@ -1852,6 +1853,7 @@ void AudioProcessingImpl::InitializeNois
return NsConfig::SuppressionLevel::k21dB;
default:
RTC_NOTREACHED();
+ return NsConfig::SuppressionLevel::k21dB; // Just to keep the
compiler happy
}
};
Index:
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/include/audio_processing.cc
===================================================================
---
webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/include/audio_processing.cc
+++
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/include/audio_processing.cc
@@ -26,6 +26,7 @@ std::string NoiseSuppressionLevelToStrin
case AudioProcessing::Config::NoiseSuppression::Level::kHigh:
return "High";
case AudioProcessing::Config::NoiseSuppression::Level::kVeryHigh:
+ default:
return "VeryHigh";
}
}
@@ -38,6 +39,7 @@ std::string GainController1ModeToString(
case AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital:
return "AdaptiveDigital";
case AudioProcessing::Config::GainController1::Mode::kFixedDigital:
+ default:
return "FixedDigital";
}
}
@@ -48,6 +50,7 @@ std::string GainController2LevelEstimato
case AudioProcessing::Config::GainController2::LevelEstimator::kRms:
return "Rms";
case AudioProcessing::Config::GainController2::LevelEstimator::kPeak:
+ default:
return "Peak";
}
}
++++++ fix-i586.patch ++++++
Index: webrtc-audio-processing-1.3/webrtc/third_party/pffft/src/pffft.c
===================================================================
--- webrtc-audio-processing-1.3.orig/webrtc/third_party/pffft/src/pffft.c
+++ webrtc-audio-processing-1.3/webrtc/third_party/pffft/src/pffft.c
@@ -131,7 +131,7 @@ inline v4sf ld_ps1(const float *p) { v4s
/*
SSE1 support macros
*/
-#elif !defined(PFFFT_SIMD_DISABLE) && (defined(__x86_64__) || defined(_M_X64)
|| defined(i386) || defined(__i386__) || defined(_M_IX86))
+#elif !defined(PFFFT_SIMD_DISABLE) && (defined(__x86_64__) || defined(_M_X64)
|| defined(i386) || defined(__i386__) || defined(_M_IX86)) && defined(__SSE2__)
#include <xmmintrin.h>
typedef __m128 v4sf;
Index:
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
===================================================================
---
webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
+++
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
@@ -88,6 +88,7 @@ void ComputeFrequencyResponse_Neon(
#if defined(WEBRTC_ARCH_X86_FAMILY)
// Computes and stores the frequency response of the filter.
+__attribute__((target("sse2")))
void ComputeFrequencyResponse_Sse2(
size_t num_partitions,
const std::vector<std::vector<FftData>>& H,
@@ -207,9 +208,10 @@ void AdaptPartitions_Neon(const RenderBu
} while (p < lim2);
}
#endif
-
+
#if defined(WEBRTC_ARCH_X86_FAMILY)
// Adapts the filter partitions. (SSE2 variant)
+__attribute__((target("sse2")))
void AdaptPartitions_Sse2(const RenderBuffer& render_buffer,
const FftData& G,
size_t num_partitions,
@@ -375,6 +377,7 @@ void ApplyFilter_Neon(const RenderBuffer
#if defined(WEBRTC_ARCH_X86_FAMILY)
// Produces the filter output (SSE2 variant).
+__attribute__((target("sse2")))
void ApplyFilter_Sse2(const RenderBuffer& render_buffer,
size_t num_partitions,
const std::vector<std::vector<FftData>>& H,
Index:
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/matched_filter.cc
===================================================================
---
webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/aec3/matched_filter.cc
+++
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/matched_filter.cc
@@ -143,7 +143,7 @@ void MatchedFilterCore_NEON(size_t x_sta
#endif
#if defined(WEBRTC_ARCH_X86_FAMILY)
-
+__attribute__((target("sse2")))
void MatchedFilterCore_SSE2(size_t x_start_index,
float x2_sum_threshold,
float smoothing,
Index:
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/fft_data.h
===================================================================
---
webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/aec3/fft_data.h
+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/fft_data.h
@@ -48,7 +48,7 @@ struct FftData {
rtc::ArrayView<float> power_spectrum) const {
RTC_DCHECK_EQ(kFftLengthBy2Plus1, power_spectrum.size());
switch (optimization) {
-#if defined(WEBRTC_ARCH_X86_FAMILY)
+#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
case Aec3Optimization::kSse2: {
constexpr int kNumFourBinBands = kFftLengthBy2 / 4;
constexpr int kLimit = kNumFourBinBands * 4;
Index:
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/vector_math.h
===================================================================
---
webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/aec3/vector_math.h
+++
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/vector_math.h
@@ -43,7 +43,7 @@ class VectorMath {
void SqrtAVX2(rtc::ArrayView<float> x);
void Sqrt(rtc::ArrayView<float> x) {
switch (optimization_) {
-#if defined(WEBRTC_ARCH_X86_FAMILY)
+#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
case Aec3Optimization::kSse2: {
const int x_size = static_cast<int>(x.size());
const int vector_limit = x_size >> 2;
@@ -123,7 +123,7 @@ class VectorMath {
RTC_DCHECK_EQ(z.size(), x.size());
RTC_DCHECK_EQ(z.size(), y.size());
switch (optimization_) {
-#if defined(WEBRTC_ARCH_X86_FAMILY)
+#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
case Aec3Optimization::kSse2: {
const int x_size = static_cast<int>(x.size());
const int vector_limit = x_size >> 2;
@@ -173,7 +173,7 @@ class VectorMath {
void Accumulate(rtc::ArrayView<const float> x, rtc::ArrayView<float> z) {
RTC_DCHECK_EQ(z.size(), x.size());
switch (optimization_) {
-#if defined(WEBRTC_ARCH_X86_FAMILY)
+#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
case Aec3Optimization::kSse2: {
const int x_size = static_cast<int>(x.size());
const int vector_limit = x_size >> 2;
Index:
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
===================================================================
---
webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
+++
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
@@ -229,6 +229,7 @@ void ComputeFullyConnectedLayerOutput(
#if defined(WEBRTC_ARCH_X86_FAMILY)
// Fully connected layer SSE2 implementation.
+__attribute__((target("sse2")))
void ComputeFullyConnectedLayerOutputSse2(
size_t input_size,
size_t output_size,
Index:
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
===================================================================
---
webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
+++
webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
@@ -57,6 +57,7 @@ void ErlComputer_NEON(
#if defined(WEBRTC_ARCH_X86_FAMILY)
// Computes and stores the echo return loss estimate of the filter, which is
the
// sum of the partition frequency responses.
+__attribute__((target("sse2")))
void ErlComputer_SSE2(
const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
rtc::ArrayView<float> erl) {
++++++ reduce-meson-dep.patch ++++++
Index: webrtc-audio-processing-1.3/meson.build
===================================================================
--- webrtc-audio-processing-1.3.orig/meson.build
+++ webrtc-audio-processing-1.3/meson.build
@@ -1,6 +1,6 @@
project('webrtc-audio-processing', 'c', 'cpp',
version : '1.3',
- meson_version : '>= 0.63',
+ meson_version : '>= 0.59.4',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized',
'c_std=c11',
++++++ webrtc-audio-processing.obsinfo ++++++
name: webrtc-audio-processing
version: 1.3
mtime: 1693927187
commit: 8e258a1933d405073c9e6465628a69ac7d2a1f13
++++++ webrtc-ppc64.patch ++++++
--- /var/tmp/diff_new_pack.b5G3qX/_old 2023-09-21 22:13:32.164152151 +0200
+++ /var/tmp/diff_new_pack.b5G3qX/_new 2023-09-21 22:13:32.164152151 +0200
@@ -1,11 +1,17 @@
Index: webrtc/typedefs.h
===================================================================
---- webrtc/typedefs.h.org
-+++ webrtc/typedefs.h
-@@ -47,6 +47,12 @@
- #elif defined(__pnacl__)
+--- a/webrtc/rtc_base/system/arch.h.orig
++++ b/webrtc/rtc_base/system/arch.h
+@@ -57,6 +57,15 @@
+# #elif defined(__pnacl__)
+# #define WEBRTC_ARCH_32_BITS
+# #define WEBRTC_ARCH_LITTLE_ENDIAN
+ #elif defined(__EMSCRIPTEN__)
#define WEBRTC_ARCH_32_BITS
#define WEBRTC_ARCH_LITTLE_ENDIAN
++#elif defined(__powerpc64__) && defined(__LITTLE_ENDIAN__)
++#define WEBRTC_ARCH_LITTLE_ENDIAN
++#define WEBRTC_ARCH_64_BITS
+#elif defined(__powerpc64__)
+#define WEBRTC_ARCH_BIG_ENDIAN
+#define WEBRTC_ARCH_64_BITS
@@ -13,6 +19,9 @@
+#define WEBRTC_ARCH_BIG_ENDIAN
+#define WEBRTC_ARCH_32_BITS
#else
- /* instead of failing, use typical unix defines... */
- #if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
+ #error Please add support for your architecture in rtc_base/system/arch.h
+ #endif
+# #else
+# /* instead of failing, use typical unix defines... */
+# #if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
++++++ webrtc-s390x.patch ++++++
--- /var/tmp/diff_new_pack.b5G3qX/_old 2023-09-21 22:13:32.176152587 +0200
+++ /var/tmp/diff_new_pack.b5G3qX/_new 2023-09-21 22:13:32.180152732 +0200
@@ -1,6 +1,6 @@
---- webrtc/typedefs.h
-+++ webrtc/typedefs.h
-@@ -53,6 +53,12 @@
+--- a/webrtc/rtc_base/system/arch.h.orig
++++ b/webrtc/rtc_base/system/arch.h
+@@ -63,6 +63,12 @@
#elif defined(__powerpc__)
#define WEBRTC_ARCH_BIG_ENDIAN
#define WEBRTC_ARCH_32_BITS
@@ -11,6 +11,9 @@
+#define WEBRTC_ARCH_BIG_ENDIAN
+#define WEBRTC_ARCH_32_BITS
#else
- /* instead of failing, use typical unix defines... */
- #if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
+ #error Please add support for your architecture in rtc_base/system/arch.h
+ #endif
+# #else
+# /* instead of failing, use typical unix defines... */
+# #if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
