[packages/webrtc-audio-processing2] - updated to 2.1
qboosh
qboosh at pld-linux.org
Mon Jul 20 19:07:10 CEST 2026
commit 393295f28fb88bd14db4b112073cd86858b41e75
Author: Jakub Bogusz <qboosh at pld-linux.org>
Date: Mon Jul 20 19:07:17 2026 +0200
- updated to 2.1
webrtc-audio-processing-absl.patch | 173 ++++++++++++++++
webrtc-audio-processing-includes.patch | 10 +
webrtc-audio-processing1-nosimd.patch | 219 ---------------------
...ocessing1.spec => webrtc-audio-processing2.spec | 45 ++---
4 files changed, 200 insertions(+), 247 deletions(-)
---
diff --git a/webrtc-audio-processing1.spec b/webrtc-audio-processing2.spec
similarity index 76%
rename from webrtc-audio-processing1.spec
rename to webrtc-audio-processing2.spec
index a3de48a..637b252 100644
--- a/webrtc-audio-processing1.spec
+++ b/webrtc-audio-processing2.spec
@@ -17,17 +17,18 @@
Summary: WebRTC Audio Processing library
Summary(pl.UTF-8): Biblioteka WebRTC Audio Processing
-Name: webrtc-audio-processing1
-Version: 1.3
+Name: webrtc-audio-processing2
+Version: 2.1
Release: 1
License: BSD
Group: Libraries
Source0: https://freedesktop.org/software/pulseaudio/webrtc-audio-processing/webrtc-audio-processing-%{version}.tar.gz
-# Source0-md5: 743bf4f5c7132560b1af33aaa0129228
-Patch0: %{name}-nosimd.patch
+# Source0-md5: 3b91a60162744925731b29834fd56197
+Patch0: webrtc-audio-processing-absl.patch
+Patch1: webrtc-audio-processing-includes.patch
URL: https://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/
-BuildRequires: abseil-cpp-devel >= 20200923
-BuildRequires: libstdc++-devel >= 6:5
+BuildRequires: abseil-cpp-devel >= 20240722
+BuildRequires: libstdc++-devel >= 6:7
BuildRequires: meson >= 0.63
BuildRequires: ninja >= 1.5
BuildRequires: pkgconfig
@@ -53,7 +54,7 @@ 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: abseil-cpp-devel >= 20200923
+Requires: abseil-cpp-devel >= 20240722
Requires: libstdc++-devel >= 6:5
%description devel
@@ -79,21 +80,13 @@ Biblioteka statyczna WebRTC Audio Processing.
%prep
%setup -q -n webrtc-audio-processing-%{version}
%patch -P0 -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
+%patch -P1 -p1
%build
-%if %{with sse2}
-CFLAGS="%{rpmcflags} -msse2"
-CXXFLAGS="%{rpmcxxflags} -msse2"
-%endif
%meson \
- -Dneon=%{?with_neon:yes}%{!?with_neon:no}
+ -Dgnustl=disabled \
+ -Dneon=%{__enabled_disabled neon} \
+ -Dinline-sse=%{__true_false sse2}
%meson_build
@@ -111,18 +104,14 @@ rm -rf $RPM_BUILD_ROOT
%files
%defattr(644,root,root,755)
%doc AUTHORS COPYING NEWS README.md webrtc/PATENTS
-%{_libdir}/libwebrtc-audio-coding-1.so.3
-%{_libdir}/libwebrtc-audio-processing-1.so.3
+%{_libdir}/libwebrtc-audio-processing-2.so.1
%files devel
%defattr(644,root,root,755)
-%{_libdir}/libwebrtc-audio-coding-1.so
-%{_libdir}/libwebrtc-audio-processing-1.so
-%{_includedir}/webrtc-audio-processing-1
-%{_pkgconfigdir}/webrtc-audio-coding-1.pc
-%{_pkgconfigdir}/webrtc-audio-processing-1.pc
+%{_libdir}/libwebrtc-audio-processing-2.so
+%{_includedir}/webrtc-audio-processing-2
+%{_pkgconfigdir}/webrtc-audio-processing-2.pc
%files static
%defattr(644,root,root,755)
-%{_libdir}/libwebrtc-audio-coding-1.a
-%{_libdir}/libwebrtc-audio-processing-1.a
+%{_libdir}/libwebrtc-audio-processing-2.a
diff --git a/webrtc-audio-processing-absl.patch b/webrtc-audio-processing-absl.patch
new file mode 100644
index 0000000..881b097
--- /dev/null
+++ b/webrtc-audio-processing-absl.patch
@@ -0,0 +1,173 @@
+--- webrtc-audio-processing-2.1/webrtc/api/scoped_refptr.h.orig 2025-01-22 23:32:18.000000000 +0100
++++ webrtc-audio-processing-2.1/webrtc/api/scoped_refptr.h 2026-07-20 17:22:57.126469558 +0200
+@@ -79,7 +79,7 @@ class ABSL_NULLABILITY_COMPATIBLE scoped
+ scoped_refptr() : ptr_(nullptr) {}
+ scoped_refptr(std::nullptr_t) : ptr_(nullptr) {} // NOLINT(runtime/explicit)
+
+- explicit scoped_refptr(absl::Nullable<T*> p) : ptr_(p) {
++ explicit scoped_refptr(T* absl_nullable p) : ptr_(p) {
+ if (ptr_)
+ ptr_->AddRef();
+ }
+@@ -122,7 +122,7 @@ class ABSL_NULLABILITY_COMPATIBLE scoped
+ return retVal;
+ }
+
+- scoped_refptr<T>& operator=(absl::Nullable<T*> p) {
++ scoped_refptr<T>& operator=(T* absl_nullable p) {
+ // AddRef first so that self assignment should work
+ if (p)
+ p->AddRef();
+@@ -152,7 +152,7 @@ class ABSL_NULLABILITY_COMPATIBLE scoped
+ return *this;
+ }
+
+- void swap(absl::Nonnull<T**> pp) noexcept {
++ void swap(T** absl_nonnull pp) noexcept {
+ T* p = ptr_;
+ ptr_ = *pp;
+ *pp = p;
+--- webrtc-audio-processing-2.1/webrtc/api/make_ref_counted.h.orig 2025-01-22 23:32:18.000000000 +0100
++++ webrtc-audio-processing-2.1/webrtc/api/make_ref_counted.h 2026-07-20 17:31:44.236947289 +0200
+@@ -86,7 +86,7 @@ template <
+ typename std::enable_if<std::is_convertible_v<T*, RefCountInterface*> &&
+ std::is_abstract_v<T>,
+ T>::type* = nullptr>
+-absl::Nonnull<scoped_refptr<T>> make_ref_counted(Args&&... args) {
++scoped_refptr<T> absl_nonnull make_ref_counted(Args&&... args) {
+ return scoped_refptr<T>(new RefCountedObject<T>(std::forward<Args>(args)...));
+ }
+
+@@ -99,7 +99,7 @@ template <
+ !std::is_convertible_v<T*, RefCountInterface*> &&
+ webrtc_make_ref_counted_internal::HasAddRefAndRelease<T>::value,
+ T>::type* = nullptr>
+-absl::Nonnull<scoped_refptr<T>> make_ref_counted(Args&&... args) {
++scoped_refptr<T> absl_nonnull make_ref_counted(Args&&... args) {
+ return scoped_refptr<T>(new T(std::forward<Args>(args)...));
+ }
+
+@@ -113,7 +113,7 @@ template <
+ !webrtc_make_ref_counted_internal::HasAddRefAndRelease<T>::value,
+
+ T>::type* = nullptr>
+-absl::Nonnull<scoped_refptr<FinalRefCountedObject<T>>> make_ref_counted(
++scoped_refptr<FinalRefCountedObject<T>> absl_nonnull make_ref_counted(
+ Args&&... args) {
+ return scoped_refptr<FinalRefCountedObject<T>>(
+ new FinalRefCountedObject<T>(std::forward<Args>(args)...));
+--- webrtc-audio-processing-2.1/webrtc/api/audio/audio_processing.h.orig 2025-01-22 23:32:18.000000000 +0100
++++ webrtc-audio-processing-2.1/webrtc/api/audio/audio_processing.h 2026-07-20 17:46:52.272028040 +0200
+@@ -635,11 +635,11 @@ class RTC_EXPORT AudioProcessing : publi
+ virtual bool CreateAndAttachAecDump(
+ absl::string_view file_name,
+ int64_t max_log_size_bytes,
+- absl::Nonnull<TaskQueueBase*> worker_queue) = 0;
++ TaskQueueBase* absl_nonnull worker_queue) = 0;
+ virtual bool CreateAndAttachAecDump(
+- absl::Nonnull<FILE*> handle,
++ FILE* absl_nonnull handle,
+ int64_t max_log_size_bytes,
+- absl::Nonnull<TaskQueueBase*> worker_queue) = 0;
++ TaskQueueBase* absl_nonnull worker_queue) = 0;
+
+ // TODO(webrtc:5298) Deprecated variant.
+ // Attaches provided webrtc::AecDump for recording debugging
+--- webrtc-audio-processing-2.1/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h.orig 2025-01-22 23:32:18.000000000 +0100
++++ webrtc-audio-processing-2.1/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h 2026-07-20 17:50:01.377670234 +0200
+@@ -29,18 +29,18 @@ class RTC_EXPORT AecDumpFactory {
+ // The AecDump takes responsibility for `handle` and closes it in the
+ // destructor. A non-null return value indicates that the file has been
+ // sucessfully opened.
+- static absl::Nullable<std::unique_ptr<AecDump>> Create(
++ static std::unique_ptr<AecDump> absl_nullable Create(
+ FileWrapper file,
+ int64_t max_log_size_bytes,
+- absl::Nonnull<TaskQueueBase*> worker_queue);
+- static absl::Nullable<std::unique_ptr<AecDump>> Create(
++ TaskQueueBase* absl_nonnull worker_queue);
++ static std::unique_ptr<AecDump> absl_nullable Create(
+ absl::string_view file_name,
+ int64_t max_log_size_bytes,
+- absl::Nonnull<TaskQueueBase*> worker_queue);
+- static absl::Nullable<std::unique_ptr<AecDump>> Create(
+- absl::Nonnull<FILE*> handle,
++ TaskQueueBase* absl_nonnull worker_queue);
++ static std::unique_ptr<AecDump> absl_nullable Create(
++ FILE* absl_nonnull handle,
+ int64_t max_log_size_bytes,
+- absl::Nonnull<TaskQueueBase*> worker_queue);
++ TaskQueueBase* absl_nonnull worker_queue);
+ };
+
+ } // namespace webrtc
+--- webrtc-audio-processing-2.1/webrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc.orig 2025-01-22 23:32:18.000000000 +0100
++++ webrtc-audio-processing-2.1/webrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc 2026-07-20 17:52:02.093682925 +0200
+@@ -16,24 +16,24 @@
+
+ namespace webrtc {
+
+-absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
++std::unique_ptr<AecDump> absl_nullable AecDumpFactory::Create(
+ FileWrapper file,
+ int64_t max_log_size_bytes,
+- absl::Nonnull<TaskQueueBase*> worker_queue) {
++ TaskQueueBase* absl_nonnull worker_queue) {
+ return nullptr;
+ }
+
+-absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
++std::unique_ptr<AecDump> absl_nullable AecDumpFactory::Create(
+ absl::string_view file_name,
+ int64_t max_log_size_bytes,
+- absl::Nonnull<TaskQueueBase*> worker_queue) {
++ TaskQueueBase* absl_nonnull worker_queue) {
+ return nullptr;
+ }
+
+-absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
+- absl::Nonnull<FILE*> handle,
++std::unique_ptr<AecDump> absl_nullable AecDumpFactory::Create(
++ FILE* absl_nonnull handle,
+ int64_t max_log_size_bytes,
+- absl::Nonnull<TaskQueueBase*> worker_queue) {
++ TaskQueueBase* absl_nonnull worker_queue) {
+ return nullptr;
+ }
+ } // namespace webrtc
+--- webrtc-audio-processing-2.1/webrtc/modules/audio_processing/audio_processing_impl.h.orig 2025-01-22 23:32:18.000000000 +0100
++++ webrtc-audio-processing-2.1/webrtc/modules/audio_processing/audio_processing_impl.h 2026-07-20 17:54:09.712991552 +0200
+@@ -74,11 +74,11 @@ class AudioProcessingImpl : public Audio
+ bool CreateAndAttachAecDump(
+ absl::string_view file_name,
+ int64_t max_log_size_bytes,
+- absl::Nonnull<TaskQueueBase*> worker_queue) override;
++ TaskQueueBase* absl_nonnull worker_queue) override;
+ bool CreateAndAttachAecDump(
+ FILE* handle,
+ int64_t max_log_size_bytes,
+- absl::Nonnull<TaskQueueBase*> worker_queue) override;
++ TaskQueueBase* absl_nonnull worker_queue) override;
+ // TODO(webrtc:5298) Deprecated variant.
+ void AttachAecDump(std::unique_ptr<AecDump> aec_dump) override;
+ void DetachAecDump() override;
+--- webrtc-audio-processing-2.1/webrtc/modules/audio_processing/audio_processing_impl.cc.orig 2025-01-22 23:32:18.000000000 +0100
++++ webrtc-audio-processing-2.1/webrtc/modules/audio_processing/audio_processing_impl.cc 2026-07-20 17:55:07.636011089 +0200
+@@ -1787,7 +1787,7 @@ void AudioProcessingImpl::UpdateRecommen
+ bool AudioProcessingImpl::CreateAndAttachAecDump(
+ absl::string_view file_name,
+ int64_t max_log_size_bytes,
+- absl::Nonnull<TaskQueueBase*> worker_queue) {
++ TaskQueueBase* absl_nonnull worker_queue) {
+ std::unique_ptr<AecDump> aec_dump =
+ AecDumpFactory::Create(file_name, max_log_size_bytes, worker_queue);
+ if (!aec_dump) {
+@@ -1801,7 +1801,7 @@ bool AudioProcessingImpl::CreateAndAttac
+ bool AudioProcessingImpl::CreateAndAttachAecDump(
+ FILE* handle,
+ int64_t max_log_size_bytes,
+- absl::Nonnull<TaskQueueBase*> worker_queue) {
++ TaskQueueBase* absl_nonnull worker_queue) {
+ std::unique_ptr<AecDump> aec_dump =
+ AecDumpFactory::Create(handle, max_log_size_bytes, worker_queue);
+ if (!aec_dump) {
diff --git a/webrtc-audio-processing-includes.patch b/webrtc-audio-processing-includes.patch
new file mode 100644
index 0000000..ce9c3fe
--- /dev/null
+++ b/webrtc-audio-processing-includes.patch
@@ -0,0 +1,10 @@
+--- webrtc-audio-processing-2.1/webrtc/rtc_base/trace_event.h.orig 2025-01-22 23:32:18.000000000 +0100
++++ webrtc-audio-processing-2.1/webrtc/rtc_base/trace_event.h 2026-07-20 17:29:07.627795714 +0200
+@@ -27,6 +27,7 @@
+ // IWYU pragma: end_exports
+
+ #if !defined(RTC_USE_PERFETTO)
++#include <cstdint>
+ #include <string>
+
+ #include "rtc_base/event_tracer.h"
diff --git a/webrtc-audio-processing1-nosimd.patch b/webrtc-audio-processing1-nosimd.patch
deleted file mode 100644
index 92bb26e..0000000
--- a/webrtc-audio-processing1-nosimd.patch
+++ /dev/null
@@ -1,219 +0,0 @@
---- 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_,
================================================================
---- gitweb:
http://git.pld-linux.org/gitweb.cgi/packages/webrtc-audio-processing2.git/commitdiff/393295f28fb88bd14db4b112073cd86858b41e75
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