- webrtc-audio-processing updated to 1.0 (parallel installable with webrtc-audio...

- added abseil patch to fix/adjust abseil library name
- added nosimd patch to fix build for x86 without SSE2

diff --git a/webrtc-audio-processing1-abseil.patch b/webrtc-audio-processing1-abseil.patch
new file mode 100644 (file)
index 0000000..4978c24
--- /dev/null
+++ b/webrtc-audio-processing1-abseil.patch
@@ -0,0 +1,11 @@
+--- webrtc-audio-processing-1.0/meson.build.orig       2020-11-27 20:30:53.000000000 +0100
++++ webrtc-audio-processing-1.0/meson.build    2021-03-14 12:53:34.720849883 +0100
+@@ -46,7 +46,7 @@
+   cpp.find_library('absl_flags_internal'),
+   cpp.find_library('absl_flags_marshalling'),
+   cpp.find_library('absl_flags_parse'),
+-  cpp.find_library('absl_flags_registry'),
++  cpp.find_library('absl_flags_reflection'),
+   cpp.find_library('absl_flags_usage_internal'),
+   cpp.find_library('absl_raw_logging_internal'),
+   cpp.find_library('absl_strings'),

diff --git a/webrtc-audio-processing1-nosimd.patch b/webrtc-audio-processing1-nosimd.patch
new file mode 100644 (file)
index 0000000..92bb26e
--- /dev/null
+++ b/webrtc-audio-processing1-nosimd.patch
@@ -0,0 +1,219 @@
+--- webrtc-audio-processing-1.0/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc.orig       2020-11-27 20:30:53.000000000 +0100
++++ webrtc-audio-processing-1.0/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc    2021-03-14 13:27:15.929900062 +0100
+@@ -16,7 +16,7 @@
+ #if defined(WEBRTC_HAS_NEON)
+ #include <arm_neon.h>
+ #endif
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+ #include <emmintrin.h>
+ #endif
+ #include <math.h>
+@@ -86,7 +86,7 @@
+ }
+ #endif
+ 
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+ // Computes and stores the frequency response of the filter.
+ void ComputeFrequencyResponse_Sse2(
+     size_t num_partitions,
+@@ -208,7 +208,7 @@
+ }
+ #endif
+ 
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+ // Adapts the filter partitions. (SSE2 variant)
+ void AdaptPartitions_Sse2(const RenderBuffer& render_buffer,
+                           const FftData& G,
+@@ -373,7 +373,7 @@
+ }
+ #endif
+ 
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+ // Produces the filter output (SSE2 variant).
+ void ApplyFilter_Sse2(const RenderBuffer& render_buffer,
+                       size_t num_partitions,
+@@ -552,7 +552,7 @@
+                                FftData* S) const {
+   RTC_DCHECK(S);
+   switch (optimization_) {
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+     case Aec3Optimization::kSse2:
+       aec3::ApplyFilter_Sse2(render_buffer, current_size_partitions_, H_, S);
+       break;
+@@ -596,7 +596,7 @@
+   H2->resize(current_size_partitions_);
+ 
+   switch (optimization_) {
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+     case Aec3Optimization::kSse2:
+       aec3::ComputeFrequencyResponse_Sse2(current_size_partitions_, H_, H2);
+       break;
+@@ -621,7 +621,7 @@
+ 
+   // Adapt the filter.
+   switch (optimization_) {
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+     case Aec3Optimization::kSse2:
+       aec3::AdaptPartitions_Sse2(render_buffer, G, current_size_partitions_,
+                                  &H_);
+--- webrtc-audio-processing-1.0/webrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc.orig   2020-11-27 20:30:53.000000000 +0100
++++ webrtc-audio-processing-1.0/webrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc        2021-03-14 13:27:55.803017384 +0100
+@@ -16,7 +16,7 @@
+ #if defined(WEBRTC_HAS_NEON)
+ #include <arm_neon.h>
+ #endif
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+ #include <emmintrin.h>
+ #endif
+ 
+@@ -54,7 +54,7 @@
+ }
+ #endif
+ 
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+ // Computes and stores the echo return loss estimate of the filter, which is the
+ // sum of the partition frequency responses.
+ void ErlComputer_SSE2(
+@@ -81,7 +81,7 @@
+   RTC_DCHECK_EQ(kFftLengthBy2Plus1, erl.size());
+   // Update the frequency response and echo return loss for the filter.
+   switch (optimization) {
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+     case Aec3Optimization::kSse2:
+       aec3::ErlComputer_SSE2(H2, erl);
+       break;
+--- webrtc-audio-processing-1.0/webrtc/modules/audio_processing/aec3/fft_data.h.orig   2020-11-27 20:30:53.000000000 +0100
++++ webrtc-audio-processing-1.0/webrtc/modules/audio_processing/aec3/fft_data.h        2021-03-14 13:28:55.999357940 +0100
+@@ -14,7 +14,7 @@
+ // Defines WEBRTC_ARCH_X86_FAMILY, used below.
+ #include "rtc_base/system/arch.h"
+ 
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+ #include <emmintrin.h>
+ #endif
+ #include <algorithm>
+@@ -48,7 +48,7 @@
+                 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;
+--- webrtc-audio-processing-1.0/webrtc/modules/audio_processing/aec3/vector_math.h.orig        2020-11-27 20:30:53.000000000 +0100
++++ webrtc-audio-processing-1.0/webrtc/modules/audio_processing/aec3/vector_math.h     2021-03-14 13:29:18.652568550 +0100
+@@ -17,7 +17,7 @@
+ #if defined(WEBRTC_HAS_NEON)
+ #include <arm_neon.h>
+ #endif
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+ #include <emmintrin.h>
+ #endif
+ #include <math.h>
+@@ -43,7 +43,7 @@
+   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 @@
+     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 @@
+   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;
+--- webrtc-audio-processing-1.0/webrtc/modules/audio_processing/aec3/matched_filter.cc.orig    2020-11-27 20:30:53.000000000 +0100
++++ webrtc-audio-processing-1.0/webrtc/modules/audio_processing/aec3/matched_filter.cc 2021-03-14 13:29:40.592449692 +0100
+@@ -15,7 +15,7 @@
+ #if defined(WEBRTC_HAS_NEON)
+ #include <arm_neon.h>
+ #endif
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+ #include <emmintrin.h>
+ #endif
+ #include <algorithm>
+@@ -142,7 +142,7 @@
+ 
+ #endif
+ 
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+ 
+ void MatchedFilterCore_SSE2(size_t x_start_index,
+                             float x2_sum_threshold,
+@@ -358,7 +358,7 @@
+         render_buffer.buffer.size();
+ 
+     switch (optimization_) {
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+       case Aec3Optimization::kSse2:
+         aec3::MatchedFilterCore_SSE2(x_start_index, x2_sum_threshold,
+                                      smoothing_, render_buffer.buffer, y,
+--- webrtc-audio-processing-1.0/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc.orig       2020-11-27 20:30:53.000000000 +0100
++++ webrtc-audio-processing-1.0/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc    2021-03-14 13:42:56.104806702 +0100
+@@ -16,7 +16,7 @@
+ #if defined(WEBRTC_HAS_NEON)
+ #include <arm_neon.h>
+ #endif
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+ #include <emmintrin.h>
+ #endif
+ #include <algorithm>
+@@ -227,7 +227,7 @@
+   }
+ }
+ 
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+ // Fully connected layer SSE2 implementation.
+ void ComputeFullyConnectedLayerOutputSse2(
+     size_t input_size,
+@@ -295,7 +295,7 @@
+ 
+ void FullyConnectedLayer::ComputeOutput(rtc::ArrayView<const float> input) {
+   switch (optimization_) {
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+     case Optimization::kSse2:
+       ComputeFullyConnectedLayerOutputSse2(input_size_, output_size_, input,
+                                            bias_, weights_,
+@@ -355,7 +355,7 @@
+ 
+ void GatedRecurrentLayer::ComputeOutput(rtc::ArrayView<const float> input) {
+   switch (optimization_) {
+-#if defined(WEBRTC_ARCH_X86_FAMILY)
++#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
+     case Optimization::kSse2:
+       // TODO(bugs.chromium.org/10480): Handle Optimization::kSse2.
+       ComputeGruLayerOutput(input_size_, output_size_, input, weights_,

diff --git a/webrtc-audio-processing.spec b/webrtc-audio-processing1.spec
similarity index 63%
rename from webrtc-audio-processing.spec
rename to webrtc-audio-processing1.spec
index 0fab2344f8d70beddd01f745aace046e042cd448..fd57a136f6050e058912528df1956b427e4789a7 100644 (file)
--- a/webrtc-audio-processing.spec
+++ b/webrtc-audio-processing1.spec
@@ -5,28 +5,32 @@
 # [1] http://code.google.com/p/webrtc/
 #
 # Conditional build:
-%bcond_without neon            # without ARM NEON instructions
+%bcond_without neon            # ARM NEON instructions
+%bcond_with    sse2            # SSE2 instructions
 
 %ifnarch armv7l armv7hl armv7hnl armv8l armv8hl armv8hnl armv8hcnl aarch64
 %undefine      with_neon
 %endif
+%ifarch pentium4 %{x8664} x32
+%define                with_sse2       1
+%endif
 
 Summary:       WebRTC Audio Processing library
 Summary(pl.UTF-8):     Biblioteka WebRTC Audio Processing
-Name:          webrtc-audio-processing
-Version:       0.3.1
+Name:          webrtc-audio-processing1
+Version:       1.0
 Release:       1
 License:       BSD
 Group:         Libraries
-Source0:       https://freedesktop.org/software/pulseaudio/webrtc-audio-processing/%{name}-%{version}.tar.xz
-# Source0-md5: 6e10724ca34bcbc715a4c208273acb0c
+Source0:       https://freedesktop.org/software/pulseaudio/webrtc-audio-processing/webrtc-audio-processing-%{version}.tar.gz
+# Source0-md5: 8ee1b2f3e615c6c2024951c559a9913a
+Patch0:                %{name}-abseil.patch
+Patch1:                %{name}-nosimd.patch
 URL:           https://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/
-BuildRequires: autoconf >= 2.50
-BuildRequires: automake >= 1:1.11
-BuildRequires: libstdc++-devel >= 6:4.7
-BuildRequires: libtool
-BuildRequires: tar >= 1:1.22
-BuildRequires: xz
+BuildRequires: abseil-cpp-devel >= 20200923
+BuildRequires: libstdc++-devel >= 6:5
+BuildRequires: meson >= 0.54
+BuildRequires: ninja >= 1.5
 BuildRoot:     %{tmpdir}/%{name}-%{version}-root-%(id -u -n)
 
 %description
@@ -48,7 +52,8 @@ Summary:      Header files for WebRTC Audio Processing library
 Summary(pl.UTF-8):     Pliki nagłówkowe biblioteki WebRTC Audio Processing
 Group:         Development/Libraries
 Requires:      %{name} = %{version}-%{release}
-Requires:      libstdc++-devel
+Requires:      abseil-cpp-devel >= 20200923
+Requires:      libstdc++-devel >= 6:5
 
 %description devel
 This package contains the header files needed to develop programs
@@ -71,25 +76,31 @@ Static WebRTC Audio Processing library.
 Biblioteka statyczna WebRTC Audio Processing.
 
 %prep
-%setup -q
+%setup -q -n webrtc-audio-processing-%{version}
+%patch0 -p1
+%patch1 -p1
+
+%ifarch %{ix86}
+%if %{without sse2}
+# add -DPFFFT_SIMD_DISABLE
+%{__sed} -i -e 's/have_arm and not have_neon.*/& or true/' webrtc/third_party/pffft/meson.build
+%endif
+%endif
 
 %build
-%{__libtoolize}
-%{__aclocal}
-%{__autoconf}
-%{__automake}
-%configure \
-       %{!?with_neon:--disable-neon} \
-       --disable-silent-rules
-%{__make}
+%if %{with sse2}
+CFLAGS="%{rpmcflags} -msse2"
+CXXFLAGS="%{rpmcxxflags} -msse2"
+%endif
+%meson build \
+       -Dneon=%{?with_neon:runtime}%{!?with_neon:no}
+
+%ninja_build -C build
 
 %install
 rm -rf $RPM_BUILD_ROOT
-%{__make} install \
-       DESTDIR=$RPM_BUILD_ROOT
 
-# obsoleted by pkg-config
-%{__rm} $RPM_BUILD_ROOT%{_libdir}/*.la
+%ninja_install -C build
 
 %clean
 rm -rf $RPM_BUILD_ROOT
@@ -100,15 +111,18 @@ rm -rf $RPM_BUILD_ROOT
 %files
 %defattr(644,root,root,755)
 %doc AUTHORS COPYING NEWS README.md webrtc/PATENTS
-%attr(755,root,root) %{_libdir}/libwebrtc_audio_processing.so.*.*.*
-%attr(755,root,root) %ghost %{_libdir}/libwebrtc_audio_processing.so.1
+%attr(755,root,root) %{_libdir}/libwebrtc-audio-coding-1.so.0
+%attr(755,root,root) %{_libdir}/libwebrtc-audio-processing-1.so.0
 
 %files devel
 %defattr(644,root,root,755)
-%attr(755,root,root) %{_libdir}/libwebrtc_audio_processing.so
-%{_includedir}/webrtc_audio_processing
-%{_pkgconfigdir}/webrtc-audio-processing.pc
+%attr(755,root,root) %{_libdir}/libwebrtc-audio-coding-1.so
+%attr(755,root,root) %{_libdir}/libwebrtc-audio-processing-1.so
+%{_includedir}/webrtc-audio-processing-1
+%{_pkgconfigdir}/webrtc-audio-coding-1.pc
+%{_pkgconfigdir}/webrtc-audio-processing-1.pc
 
 %files static
 %defattr(644,root,root,755)
-%{_libdir}/libwebrtc_audio_processing.a
+%{_libdir}/libwebrtc-audio-coding-1.a
+%{_libdir}/libwebrtc-audio-processing-1.a