Takashi Iwai
e822930c68
which provides webrtc-audio-processing-1 for packages that haven't updated to the breaking changes in w-a-p-2 yet. - Update to version 2.1: * Build-system fixups to install more headers * add a missing absl dependency * forward port some missing patches to fix Windows builds. - Update to version 2.0: * Bump to code from WebRTC M131 version. * Minor (breaking) API changes upstream * Various improvements to the AEC implementation * Transient suppression is removed * ExperimentalAgc and ExperimentalNs are removed * iSAC and the webrtc-audio-coding library were removed * abseil-cpp dependency bumped to 20240722 * NEON runtime detection dropped following upstream * Fixes for building on i686 and MIPS * Support for BSDs is added * Other build-system cleanups * Patches to upstream are now also tracked in patches/ - Do not generate libwebrtc-audio-coding-* subpackages since the library was removed by upstream. - Drop patches that aren't needed anymore: * big_endian_support.patch * big_endian_support_2.patch * fix-i586.patch * reduce-meson-dep.patch * webrtc-ppc64.patch * webrtc-s390x.patch - Rebase patch to fix build with the new sources: * fix-build.patch OBS-URL: https://build.opensuse.org/package/show/multimedia:libs/webrtc-audio-processing?expand=0&rev=29
127 lines
6.2 KiB
Diff
127 lines
6.2 KiB
Diff
Index: webrtc-audio-processing-1.3/webrtc/third_party/pffft/src/pffft.c
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===================================================================
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--- webrtc-audio-processing-1.3.orig/webrtc/third_party/pffft/src/pffft.c
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+++ webrtc-audio-processing-1.3/webrtc/third_party/pffft/src/pffft.c
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@@ -131,7 +131,7 @@ inline v4sf ld_ps1(const float *p) { v4s
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/*
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SSE1 support macros
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*/
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-#elif !defined(PFFFT_SIMD_DISABLE) && (defined(__x86_64__) || defined(_M_X64) || defined(i386) || defined(__i386__) || defined(_M_IX86))
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+#elif !defined(PFFFT_SIMD_DISABLE) && (defined(__x86_64__) || defined(_M_X64) || defined(i386) || defined(__i386__) || defined(_M_IX86)) && defined(__SSE2__)
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#include <xmmintrin.h>
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typedef __m128 v4sf;
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Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
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===================================================================
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--- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
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+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
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@@ -88,6 +88,7 @@ void ComputeFrequencyResponse_Neon(
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#if defined(WEBRTC_ARCH_X86_FAMILY)
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// Computes and stores the frequency response of the filter.
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+__attribute__((target("sse2")))
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void ComputeFrequencyResponse_Sse2(
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size_t num_partitions,
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const std::vector<std::vector<FftData>>& H,
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@@ -207,9 +208,10 @@ void AdaptPartitions_Neon(const RenderBu
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} while (p < lim2);
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}
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#endif
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-
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+
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#if defined(WEBRTC_ARCH_X86_FAMILY)
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// Adapts the filter partitions. (SSE2 variant)
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+__attribute__((target("sse2")))
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void AdaptPartitions_Sse2(const RenderBuffer& render_buffer,
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const FftData& G,
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size_t num_partitions,
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@@ -375,6 +377,7 @@ void ApplyFilter_Neon(const RenderBuffer
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#if defined(WEBRTC_ARCH_X86_FAMILY)
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// Produces the filter output (SSE2 variant).
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+__attribute__((target("sse2")))
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void ApplyFilter_Sse2(const RenderBuffer& render_buffer,
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size_t num_partitions,
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const std::vector<std::vector<FftData>>& H,
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Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/matched_filter.cc
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===================================================================
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--- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/aec3/matched_filter.cc
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+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/matched_filter.cc
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@@ -143,7 +143,7 @@ void MatchedFilterCore_NEON(size_t x_sta
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#endif
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#if defined(WEBRTC_ARCH_X86_FAMILY)
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-
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+__attribute__((target("sse2")))
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void MatchedFilterCore_SSE2(size_t x_start_index,
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float x2_sum_threshold,
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float smoothing,
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Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/fft_data.h
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===================================================================
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--- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/aec3/fft_data.h
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+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/fft_data.h
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@@ -48,7 +48,7 @@ struct FftData {
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rtc::ArrayView<float> power_spectrum) const {
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RTC_DCHECK_EQ(kFftLengthBy2Plus1, power_spectrum.size());
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switch (optimization) {
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-#if defined(WEBRTC_ARCH_X86_FAMILY)
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+#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
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case Aec3Optimization::kSse2: {
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constexpr int kNumFourBinBands = kFftLengthBy2 / 4;
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constexpr int kLimit = kNumFourBinBands * 4;
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Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/vector_math.h
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===================================================================
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--- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/aec3/vector_math.h
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+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/vector_math.h
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@@ -43,7 +43,7 @@ class VectorMath {
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void SqrtAVX2(rtc::ArrayView<float> x);
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void Sqrt(rtc::ArrayView<float> x) {
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switch (optimization_) {
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-#if defined(WEBRTC_ARCH_X86_FAMILY)
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+#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
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case Aec3Optimization::kSse2: {
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const int x_size = static_cast<int>(x.size());
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const int vector_limit = x_size >> 2;
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@@ -123,7 +123,7 @@ class VectorMath {
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RTC_DCHECK_EQ(z.size(), x.size());
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RTC_DCHECK_EQ(z.size(), y.size());
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switch (optimization_) {
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-#if defined(WEBRTC_ARCH_X86_FAMILY)
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+#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
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case Aec3Optimization::kSse2: {
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const int x_size = static_cast<int>(x.size());
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const int vector_limit = x_size >> 2;
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@@ -173,7 +173,7 @@ class VectorMath {
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void Accumulate(rtc::ArrayView<const float> x, rtc::ArrayView<float> z) {
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RTC_DCHECK_EQ(z.size(), x.size());
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switch (optimization_) {
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-#if defined(WEBRTC_ARCH_X86_FAMILY)
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+#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
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case Aec3Optimization::kSse2: {
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const int x_size = static_cast<int>(x.size());
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const int vector_limit = x_size >> 2;
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Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
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===================================================================
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--- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
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+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
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@@ -229,6 +229,7 @@ void ComputeFullyConnectedLayerOutput(
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#if defined(WEBRTC_ARCH_X86_FAMILY)
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// Fully connected layer SSE2 implementation.
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+__attribute__((target("sse2")))
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void ComputeFullyConnectedLayerOutputSse2(
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size_t input_size,
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size_t output_size,
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Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
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===================================================================
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--- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
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+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
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@@ -57,6 +57,7 @@ void ErlComputer_NEON(
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#if defined(WEBRTC_ARCH_X86_FAMILY)
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// Computes and stores the echo return loss estimate of the filter, which is the
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// sum of the partition frequency responses.
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+__attribute__((target("sse2")))
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void ErlComputer_SSE2(
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const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
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rtc::ArrayView<float> erl) {
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