commit e9f4ce4dd8bc4cc86c2c81d99219821ee21839f9 Author: MSVSphere Packaging Team Date: Tue Nov 26 19:41:00 2024 +0300 import webrtc-audio-processing-1.3-4.el10 diff --git a/.gitignore b/.gitignore new file mode 100644 index 0000000..6294bfc --- /dev/null +++ b/.gitignore @@ -0,0 +1,3 @@ +SOURCES/abseil-cpp-20230125.1.tar.gz +SOURCES/abseil-cpp_20230125.1-4_patch.zip +SOURCES/webrtc-audio-processing-1.3.tar.xz diff --git a/.webrtc-audio-processing.metadata b/.webrtc-audio-processing.metadata new file mode 100644 index 0000000..6121fae --- /dev/null +++ b/.webrtc-audio-processing.metadata @@ -0,0 +1,3 @@ +609817274bf327339efc8488bb773d5f633ada62 SOURCES/abseil-cpp-20230125.1.tar.gz +11b3e50fd05da4c77f270ae7167a44539ab34e59 SOURCES/abseil-cpp_20230125.1-4_patch.zip +f7cd684db4429722e7f8d8ed6559094aaf4dfc18 SOURCES/webrtc-audio-processing-1.3.tar.xz diff --git a/SOURCES/65f002e.patch b/SOURCES/65f002e.patch new file mode 100644 index 0000000..f9098ab --- /dev/null +++ b/SOURCES/65f002e.patch @@ -0,0 +1,313 @@ +--- a/webrtc/common_audio/wav_file.cc ++++ b/webrtc/common_audio/wav_file.cc +@@ -10,6 +10,7 @@ + + #include "common_audio/wav_file.h" + ++#include + #include + + #include +@@ -34,6 +35,38 @@ bool FormatSupported(WavFormat format) { + format == WavFormat::kWavFormatIeeeFloat; + } + ++template ++void TranslateEndianness(T* destination, const T* source, size_t length) { ++ static_assert(sizeof(T) == 2 || sizeof(T) == 4 || sizeof(T) == 8, ++ "no converter, use integral types"); ++ if (sizeof(T) == 2) { ++ const uint16_t* src = reinterpret_cast(source); ++ uint16_t* dst = reinterpret_cast(destination); ++ for (size_t index = 0; index < length; index++) { ++ dst[index] = bswap_16(src[index]); ++ } ++ } ++ if (sizeof(T) == 4) { ++ const uint32_t* src = reinterpret_cast(source); ++ uint32_t* dst = reinterpret_cast(destination); ++ for (size_t index = 0; index < length; index++) { ++ dst[index] = bswap_32(src[index]); ++ } ++ } ++ if (sizeof(T) == 8) { ++ const uint64_t* src = reinterpret_cast(source); ++ uint64_t* dst = reinterpret_cast(destination); ++ for (size_t index = 0; index < length; index++) { ++ dst[index] = bswap_64(src[index]); ++ } ++ } ++} ++ ++template ++void TranslateEndianness(T* buffer, size_t length) { ++ TranslateEndianness(buffer, buffer, length); ++} ++ + // Doesn't take ownership of the file handle and won't close it. + class WavHeaderFileReader : public WavHeaderReader { + public: +@@ -89,10 +122,6 @@ void WavReader::Reset() { + + 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 +- + size_t num_samples_left_to_read = num_samples; + size_t next_chunk_start = 0; + while (num_samples_left_to_read > 0 && num_unread_samples_ > 0) { +@@ -105,6 +134,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, + num_bytes_read = file_.Read(samples_to_convert.data(), + chunk_size * sizeof(samples_to_convert[0])); + num_samples_read = num_bytes_read / sizeof(samples_to_convert[0]); ++#ifdef WEBRTC_ARCH_BIG_ENDIAN ++ TranslateEndianness(samples_to_convert.data(), num_samples_read); ++#endif + + for (size_t j = 0; j < num_samples_read; ++j) { + samples[next_chunk_start + j] = FloatToS16(samples_to_convert[j]); +@@ -114,6 +146,10 @@ size_t WavReader::ReadSamples(const size_t num_samples, + num_bytes_read = file_.Read(&samples[next_chunk_start], + chunk_size * sizeof(samples[0])); + num_samples_read = num_bytes_read / sizeof(samples[0]); ++ ++#ifdef WEBRTC_ARCH_BIG_ENDIAN ++ TranslateEndianness(&samples[next_chunk_start], num_samples_read); ++#endif + } + RTC_CHECK(num_samples_read == 0 || (num_bytes_read % num_samples_read) == 0) + << "Corrupt file: file ended in the middle of a sample."; +@@ -129,10 +165,6 @@ size_t WavReader::ReadSamples(const size_t num_samples, + } + + size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) { +-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN +-#error "Need to convert samples to big-endian when reading from WAV file" +-#endif +- + size_t num_samples_left_to_read = num_samples; + size_t next_chunk_start = 0; + while (num_samples_left_to_read > 0 && num_unread_samples_ > 0) { +@@ -145,6 +177,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) { + num_bytes_read = file_.Read(samples_to_convert.data(), + chunk_size * sizeof(samples_to_convert[0])); + num_samples_read = num_bytes_read / sizeof(samples_to_convert[0]); ++#ifdef WEBRTC_ARCH_BIG_ENDIAN ++ TranslateEndianness(samples_to_convert.data(), num_samples_read); ++#endif + + for (size_t j = 0; j < num_samples_read; ++j) { + samples[next_chunk_start + j] = +@@ -155,6 +190,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) { + num_bytes_read = file_.Read(&samples[next_chunk_start], + chunk_size * sizeof(samples[0])); + num_samples_read = num_bytes_read / sizeof(samples[0]); ++#ifdef WEBRTC_ARCH_BIG_ENDIAN ++ TranslateEndianness(&samples[next_chunk_start], num_samples_read); ++#endif + + for (size_t j = 0; j < num_samples_read; ++j) { + samples[next_chunk_start + j] = +@@ -213,24 +251,32 @@ WavWriter::WavWriter(FileWrapper file, + } + + 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 +- + for (size_t i = 0; i < num_samples; i += kMaxChunksize) { + const size_t num_remaining_samples = num_samples - i; + const size_t num_samples_to_write = + std::min(kMaxChunksize, num_remaining_samples); + + if (format_ == WavFormat::kWavFormatPcm) { ++#ifndef WEBRTC_ARCH_BIG_ENDIAN + RTC_CHECK( + file_.Write(&samples[i], num_samples_to_write * sizeof(samples[0]))); ++#else ++ std::array converted_samples; ++ TranslateEndianness(converted_samples.data(), &samples[i], ++ num_samples_to_write); ++ RTC_CHECK( ++ file_.Write(converted_samples.data(), ++ num_samples_to_write * sizeof(converted_samples[0]))); ++#endif + } else { + RTC_CHECK_EQ(format_, WavFormat::kWavFormatIeeeFloat); + std::array converted_samples; + for (size_t j = 0; j < num_samples_to_write; ++j) { + converted_samples[j] = S16ToFloat(samples[i + j]); + } ++#ifdef WEBRTC_ARCH_BIG_ENDIAN ++ TranslateEndianness(converted_samples.data(), num_samples_to_write); ++#endif + RTC_CHECK( + file_.Write(converted_samples.data(), + num_samples_to_write * sizeof(converted_samples[0]))); +@@ -243,10 +289,6 @@ void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) { + } + + void WavWriter::WriteSamples(const float* samples, size_t num_samples) { +-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN +-#error "Need to convert samples to little-endian when writing to WAV file" +-#endif +- + for (size_t i = 0; i < num_samples; i += kMaxChunksize) { + const size_t num_remaining_samples = num_samples - i; + const size_t num_samples_to_write = +@@ -257,6 +299,9 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) { + for (size_t j = 0; j < num_samples_to_write; ++j) { + converted_samples[j] = FloatS16ToS16(samples[i + j]); + } ++#ifdef WEBRTC_ARCH_BIG_ENDIAN ++ TranslateEndianness(converted_samples.data(), num_samples_to_write); ++#endif + RTC_CHECK( + file_.Write(converted_samples.data(), + num_samples_to_write * sizeof(converted_samples[0]))); +@@ -266,6 +311,9 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) { + for (size_t j = 0; j < num_samples_to_write; ++j) { + converted_samples[j] = FloatS16ToFloat(samples[i + j]); + } ++#ifdef WEBRTC_ARCH_BIG_ENDIAN ++ TranslateEndianness(converted_samples.data(), num_samples_to_write); ++#endif + RTC_CHECK( + file_.Write(converted_samples.data(), + num_samples_to_write * sizeof(converted_samples[0]))); +--- a/webrtc/common_audio/wav_header.cc ++++ b/webrtc/common_audio/wav_header.cc +@@ -14,6 +14,8 @@ + + #include "common_audio/wav_header.h" + ++#include ++ + #include + #include + #include +@@ -26,10 +28,6 @@ + namespace webrtc { + namespace { + +-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN +-#error "Code not working properly for big endian platforms." +-#endif +- + #pragma pack(2) + struct ChunkHeader { + uint32_t ID; +@@ -174,6 +172,8 @@ bool FindWaveChunk(ChunkHeader* chunk_header, + if (readable->Read(chunk_header, sizeof(*chunk_header)) != + sizeof(*chunk_header)) + return false; // EOF. ++ chunk_header->Size = le32toh(chunk_header->Size); ++ + if (ReadFourCC(chunk_header->ID) == sought_chunk_id) + return true; // Sought chunk found. + // Ignore current chunk by skipping its payload. +@@ -187,6 +187,13 @@ bool ReadFmtChunkData(FmtPcmSubchunk* fmt_subchunk, WavHeaderReader* readable) { + if (readable->Read(&(fmt_subchunk->AudioFormat), kFmtPcmSubchunkSize) != + kFmtPcmSubchunkSize) + return false; ++ fmt_subchunk->AudioFormat = le16toh(fmt_subchunk->AudioFormat); ++ fmt_subchunk->NumChannels = le16toh(fmt_subchunk->NumChannels); ++ fmt_subchunk->SampleRate = le32toh(fmt_subchunk->SampleRate); ++ fmt_subchunk->ByteRate = le32toh(fmt_subchunk->ByteRate); ++ fmt_subchunk->BlockAlign = le16toh(fmt_subchunk->BlockAlign); ++ fmt_subchunk->BitsPerSample = le16toh(fmt_subchunk->BitsPerSample); ++ + const uint32_t fmt_size = fmt_subchunk->header.Size; + if (fmt_size != kFmtPcmSubchunkSize) { + // There is an optional two-byte extension field permitted to be present +@@ -214,19 +221,22 @@ void WritePcmWavHeader(size_t num_channels, + auto header = rtc::MsanUninitialized({}); + const size_t bytes_in_payload = bytes_per_sample * num_samples; + +- header.riff.header.ID = PackFourCC('R', 'I', 'F', 'F'); +- header.riff.header.Size = RiffChunkSize(bytes_in_payload, *header_size); +- header.riff.Format = PackFourCC('W', 'A', 'V', 'E'); +- header.fmt.header.ID = PackFourCC('f', 'm', 't', ' '); +- header.fmt.header.Size = kFmtPcmSubchunkSize; +- header.fmt.AudioFormat = MapWavFormatToHeaderField(WavFormat::kWavFormatPcm); +- header.fmt.NumChannels = static_cast(num_channels); +- header.fmt.SampleRate = sample_rate; +- header.fmt.ByteRate = ByteRate(num_channels, sample_rate, bytes_per_sample); +- header.fmt.BlockAlign = BlockAlign(num_channels, bytes_per_sample); +- header.fmt.BitsPerSample = static_cast(8 * bytes_per_sample); +- header.data.header.ID = PackFourCC('d', 'a', 't', 'a'); +- header.data.header.Size = static_cast(bytes_in_payload); ++ header.riff.header.ID = htole32(PackFourCC('R', 'I', 'F', 'F')); ++ header.riff.header.Size = ++ htole32(RiffChunkSize(bytes_in_payload, *header_size)); ++ header.riff.Format = htole32(PackFourCC('W', 'A', 'V', 'E')); ++ header.fmt.header.ID = htole32(PackFourCC('f', 'm', 't', ' ')); ++ header.fmt.header.Size = htole32(kFmtPcmSubchunkSize); ++ header.fmt.AudioFormat = ++ htole16(MapWavFormatToHeaderField(WavFormat::kWavFormatPcm)); ++ header.fmt.NumChannels = htole16(num_channels); ++ header.fmt.SampleRate = htole32(sample_rate); ++ header.fmt.ByteRate = ++ htole32(ByteRate(num_channels, sample_rate, bytes_per_sample)); ++ header.fmt.BlockAlign = htole16(BlockAlign(num_channels, bytes_per_sample)); ++ header.fmt.BitsPerSample = htole16(8 * bytes_per_sample); ++ header.data.header.ID = htole32(PackFourCC('d', 'a', 't', 'a')); ++ header.data.header.Size = htole32(bytes_in_payload); + + // Do an extra copy rather than writing everything to buf directly, since buf + // might not be correctly aligned. +@@ -245,24 +255,26 @@ void WriteIeeeFloatWavHeader(size_t num_channels, + auto header = rtc::MsanUninitialized({}); + const size_t bytes_in_payload = bytes_per_sample * num_samples; + +- header.riff.header.ID = PackFourCC('R', 'I', 'F', 'F'); +- header.riff.header.Size = RiffChunkSize(bytes_in_payload, *header_size); +- header.riff.Format = PackFourCC('W', 'A', 'V', 'E'); +- header.fmt.header.ID = PackFourCC('f', 'm', 't', ' '); +- header.fmt.header.Size = kFmtIeeeFloatSubchunkSize; ++ header.riff.header.ID = htole32(PackFourCC('R', 'I', 'F', 'F')); ++ header.riff.header.Size = ++ htole32(RiffChunkSize(bytes_in_payload, *header_size)); ++ header.riff.Format = htole32(PackFourCC('W', 'A', 'V', 'E')); ++ header.fmt.header.ID = htole32(PackFourCC('f', 'm', 't', ' ')); ++ header.fmt.header.Size = htole32(kFmtIeeeFloatSubchunkSize); + header.fmt.AudioFormat = +- MapWavFormatToHeaderField(WavFormat::kWavFormatIeeeFloat); +- header.fmt.NumChannels = static_cast(num_channels); +- header.fmt.SampleRate = sample_rate; +- header.fmt.ByteRate = ByteRate(num_channels, sample_rate, bytes_per_sample); +- header.fmt.BlockAlign = BlockAlign(num_channels, bytes_per_sample); +- header.fmt.BitsPerSample = static_cast(8 * bytes_per_sample); +- header.fmt.ExtensionSize = 0; +- header.fact.header.ID = PackFourCC('f', 'a', 'c', 't'); +- header.fact.header.Size = 4; +- header.fact.SampleLength = static_cast(num_channels * num_samples); +- header.data.header.ID = PackFourCC('d', 'a', 't', 'a'); +- header.data.header.Size = static_cast(bytes_in_payload); ++ htole16(MapWavFormatToHeaderField(WavFormat::kWavFormatIeeeFloat)); ++ header.fmt.NumChannels = htole16(num_channels); ++ header.fmt.SampleRate = htole32(sample_rate); ++ header.fmt.ByteRate = ++ htole32(ByteRate(num_channels, sample_rate, bytes_per_sample)); ++ header.fmt.BlockAlign = htole16(BlockAlign(num_channels, bytes_per_sample)); ++ header.fmt.BitsPerSample = htole16(8 * bytes_per_sample); ++ header.fmt.ExtensionSize = htole16(0); ++ header.fact.header.ID = htole32(PackFourCC('f', 'a', 'c', 't')); ++ header.fact.header.Size = htole32(4); ++ header.fact.SampleLength = htole32(num_channels * num_samples); ++ header.data.header.ID = htole32(PackFourCC('d', 'a', 't', 'a')); ++ header.data.header.Size = htole32(bytes_in_payload); + + // Do an extra copy rather than writing everything to buf directly, since buf + // might not be correctly aligned. +@@ -391,6 +403,7 @@ bool ReadWavHeader(WavHeaderReader* readable, + return false; + if (ReadFourCC(header.riff.Format) != "WAVE") + return false; ++ header.riff.header.Size = le32toh(header.riff.header.Size); + + // Find "fmt " and "data" chunks. While the official Wave file specification + // does not put requirements on the chunks order, it is uncommon to find the diff --git a/SOURCES/arches.patch b/SOURCES/arches.patch new file mode 100644 index 0000000..eaf43df --- /dev/null +++ b/SOURCES/arches.patch @@ -0,0 +1,86 @@ +--- webrtc-audio-processing-1.3/webrtc/rtc_base/system/arch.h 2023-09-05 10:19:47.000000000 -0500 ++++ webrtc-audio-processing-1.3/webrtc/rtc_base/system/arch.h 2024-02-12 10:04:12.114812565 -0600 +@@ -15,8 +15,9 @@ + #define RTC_BASE_SYSTEM_ARCH_H_ + + // Processor architecture detection. For more info on what's defined, see: +-// http://msdn.microsoft.com/en-us/library/b0084kay.aspx +-// http://www.agner.org/optimize/calling_conventions.pdf ++// https://docs.microsoft.com/en-us/cpp/preprocessor/predefined-macros ++// https://www.agner.org/optimize/calling_conventions.pdf ++// https://sourceforge.net/p/predef/wiki/Architectures/ + // or with gcc, run: "echo | gcc -E -dM -" + #if defined(_M_X64) || defined(__x86_64__) + #define WEBRTC_ARCH_X86_FAMILY +@@ -27,29 +28,50 @@ + #define WEBRTC_ARCH_ARM_FAMILY + #define WEBRTC_ARCH_64_BITS + #define WEBRTC_ARCH_LITTLE_ENDIAN +-#elif defined(__riscv) || defined(__riscv__) +-#define WEBRTC_ARCH_LITTLE_ENDIAN +-#if __riscv_xlen == 64 +-#define WEBRTC_ARCH_64_BITS +-#else +-#define WEBRTC_ARCH_32_BITS +-#endif + #elif defined(_M_IX86) || defined(__i386__) + #define WEBRTC_ARCH_X86_FAMILY + #define WEBRTC_ARCH_X86 + #define WEBRTC_ARCH_32_BITS + #define WEBRTC_ARCH_LITTLE_ENDIAN +-#elif defined(__ARMEL__) ++#elif defined(_M_ARM) || defined(__ARMEL__) + #define WEBRTC_ARCH_ARM_FAMILY + #define WEBRTC_ARCH_32_BITS + #define WEBRTC_ARCH_LITTLE_ENDIAN +-#elif defined(__MIPSEL__) ++#elif defined(__MIPSEL__) || defined(__MIPSEB__) + #define WEBRTC_ARCH_MIPS_FAMILY + #if defined(__LP64__) + #define WEBRTC_ARCH_64_BITS + #else + #define WEBRTC_ARCH_32_BITS + #endif ++#if defined(__MIPSEL__) ++#define WEBRTC_ARCH_LITTLE_ENDIAN ++#else ++#define WEBRTC_ARCH_BIG_ENDIAN ++#endif ++#elif defined(__PPC__) ++#if defined(__PPC64__) ++#define WEBRTC_ARCH_64_BITS ++#else ++#define WEBRTC_ARCH_32_BITS ++#endif ++#if defined(__LITTLE_ENDIAN__) ++#define WEBRTC_ARCH_LITTLE_ENDIAN ++#else ++#define WEBRTC_ARCH_BIG_ENDIAN ++#endif ++#elif defined(__sparc) || defined(__sparc__) ++#if __SIZEOF_LONG__ == 8 ++#define WEBRTC_ARCH_64_BITS ++#else ++#define WEBRTC_ARCH_32_BITS ++#endif ++#define WEBRTC_ARCH_BIG_ENDIAN ++#elif defined(__riscv) && __riscv_xlen == 64 ++#define WEBRTC_ARCH_64_BITS ++#define WEBRTC_ARCH_LITTLE_ENDIAN ++#elif defined(__riscv) && __riscv_xlen == 32 ++#define WEBRTC_ARCH_32_BITS + #define WEBRTC_ARCH_LITTLE_ENDIAN + #elif defined(__pnacl__) + #define WEBRTC_ARCH_32_BITS +--- webrtc-audio-processing-1.3/webrtc/rtc_base/system/arch.h~ 2024-02-12 10:14:11.277835532 -0600 ++++ webrtc-audio-processing-1.3/webrtc/rtc_base/system/arch.h 2024-02-12 10:25:11.558554823 -0600 +@@ -79,6 +79,9 @@ + #elif defined(__EMSCRIPTEN__) + #define WEBRTC_ARCH_32_BITS + #define WEBRTC_ARCH_LITTLE_ENDIAN ++#elif defined(__s390x__) ++#define WEBRTC_ARCH_64_BITS ++#define WEBRTC_ARCH_BIG_ENDIAN + #else + #error Please add support for your architecture in rtc_base/system/arch.h + #endif diff --git a/SOURCES/build-abseil-cpp-from-local-tarbal.patch b/SOURCES/build-abseil-cpp-from-local-tarbal.patch new file mode 100644 index 0000000..f8c1ddd --- /dev/null +++ b/SOURCES/build-abseil-cpp-from-local-tarbal.patch @@ -0,0 +1,17 @@ +diff -ru webrtc-audio-processing-1.3.old/subprojects/abseil-cpp.wrap webrtc-audio-processing-1.3/subprojects/abseil-cpp.wrap +--- webrtc-audio-processing-1.3.old/subprojects/abseil-cpp.wrap 2023-09-05 17:19:47.000000000 +0200 ++++ webrtc-audio-processing-1.3/subprojects/abseil-cpp.wrap 2024-07-12 17:45:02.377170755 +0200 +@@ -1,13 +1,7 @@ + [wrap-file] + directory = abseil-cpp-20230125.1 +-source_url = https://github.com/abseil/abseil-cpp/archive/20230125.1.tar.gz + source_filename = abseil-cpp-20230125.1.tar.gz +-source_hash = 81311c17599b3712069ded20cca09a62ab0bf2a89dfa16993786c8782b7ed145 + patch_filename = abseil-cpp_20230125.1-4_patch.zip +-patch_url = https://wrapdb.mesonbuild.com/v2/abseil-cpp_20230125.1-4/get_patch +-patch_hash = 112ee72052049d930396c2778fc1c6e184137905dd75d60a97dcfc386426610d +-source_fallback_url = https://github.com/mesonbuild/wrapdb/releases/download/abseil-cpp_20230125.1-4/abseil-cpp-20230125.1.tar.gz +-wrapdb_version = 20230125.1-4 + + [provide] + absl_base = absl_base_dep diff --git a/SPECS/webrtc-audio-processing.spec b/SPECS/webrtc-audio-processing.spec new file mode 100644 index 0000000..f459260 --- /dev/null +++ b/SPECS/webrtc-audio-processing.spec @@ -0,0 +1,207 @@ +Name: webrtc-audio-processing +Version: 1.3 +Release: 4%{?dist} +Summary: Library for echo cancellation + +License: BSD-3-Clause +URL: http://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/ +Source0: http://freedesktop.org/software/pulseaudio/webrtc-audio-processing/%{name}-%{version}.tar.xz +Source1: abseil-cpp-20230125.1.tar.gz +Source2: abseil-cpp_20230125.1-4_patch.zip + + +Patch0: arches.patch +Patch1: 65f002e.patch +Patch2: build-abseil-cpp-from-local-tarbal.patch + +BuildRequires: meson +BuildRequires: gcc gcc-c++ +#BuildRequires: abseil-cpp-devel +#BuildRequires: neon-devel + +%description +%{name} is a library derived from Google WebRTC project that +provides echo cancellation functionality. This library is used by for example +PulseAudio to provide echo cancellation. + +%package devel +Summary: Development files for %{name} +Requires: %{name}%{?_isa} = %{version}-%{release} + +%description devel +The %{name}-devel package contains libraries and header +files for developing applications that use %{name}. + +%prep +%autosetup -p1 + +mkdir subprojects/packagefiles +cp %{SOURCE1} subprojects/packagefiles/ +cp %{SOURCE2} subprojects/packagefiles/ + +%build +%meson +%meson_build \ +#%%ifarch %%{arm} aarch64 +# -Dneon=no \ +#%endif + +%install +%meson_install + +%ldconfig_scriptlets + +%files +%doc NEWS AUTHORS README.md +%license COPYING +%{_libdir}/libwebrtc-audio-coding-1.so.3* +%{_libdir}/libwebrtc-audio-processing-1.so.3* + +%files devel +%{_libdir}/libwebrtc-audio-coding-1.so +%{_libdir}/libwebrtc-audio-processing-1.so +%{_libdir}/pkgconfig/webrtc-audio-coding-1.pc +%{_libdir}/pkgconfig/webrtc-audio-processing-1.pc +%{_includedir}/webrtc-audio-processing-1/ +%{_includedir}/absl/ + +%changelog +* Tue Nov 26 2024 MSVSphere Packaging Team - 1.3-4 +- Rebuilt for MSVSphere 10 + +* Mon Jul 15 2024 Wim Taymans - 1.3-4 +- Install abseil-cpp headers as well +- Resolves: RHEL-28928 + +* Fri Jul 12 2024 Wim Taymans - 1.3-3 +- Build abseil-cpp from tarbal +- Resolves: RHEL-28928 + +* Mon Jun 24 2024 Troy Dawson - 1.3-2 +- Bump release for June 2024 mass rebuild + +* Fri Feb 09 2024 Gwyn Ciesla - 1.3-1 +- 1.3 + +* Sat Jan 27 2024 Fedora Release Engineering - 0.3.1-12 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild + +* Sat Jul 22 2023 Fedora Release Engineering - 0.3.1-11 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild + +* Sat Jan 21 2023 Fedora Release Engineering - 0.3.1-10 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild + +* Sat Jul 23 2022 Fedora Release Engineering - 0.3.1-9 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_37_Mass_Rebuild + +* Sat Jan 22 2022 Fedora Release Engineering - 0.3.1-8 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild + +* Fri Jul 23 2021 Fedora Release Engineering - 0.3.1-7 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_35_Mass_Rebuild + +* Wed Jan 27 2021 Fedora Release Engineering - 0.3.1-6 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild + +* Wed Jul 29 2020 Fedora Release Engineering - 0.3.1-5 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_33_Mass_Rebuild + +* Fri Jan 31 2020 Fedora Release Engineering - 0.3.1-4 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild + +* Sat Jul 27 2019 Fedora Release Engineering - 0.3.1-3 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild + +* Sun Feb 03 2019 Fedora Release Engineering - 0.3.1-2 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild + +* Wed Jul 25 2018 Rex Dieter - 0.3.1-1 +- 0.3.1 +- use %%make_build %%make_install %%ldconfig_scriptlets + +* Mon Jul 23 2018 Debarshi Ray 0.3-9 +- Update License + +* Sat Jul 14 2018 Fedora Release Engineering - 0.3-8 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild + +* Sat Mar 3 2018 Peter Robinson 0.3-7 +- Add gcc/gcc-c++ build requires +- Add aarch64 to NEON exception + +* Fri Feb 09 2018 Fedora Release Engineering - 0.3-6 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild + +* Mon Jan 22 2018 Rex Dieter - 0.3-5 +- pull in upstream fixes, use %%autosetup + +* Thu Aug 03 2017 Fedora Release Engineering - 0.3-4 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild + +* Thu Jul 27 2017 Fedora Release Engineering - 0.3-3 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild + +* Sat Feb 11 2017 Fedora Release Engineering - 0.3-2 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild + +* Mon Jul 18 2016 Rex Dieter - 0.3-1 +- 0.3 + +* Fri May 27 2016 Rex Dieter - 0.2-7 +- better/upstreamable x86_msse2.patch + +* Fri May 27 2016 Rex Dieter - 0.2-6 +- simpler/upstreamable no_undefined.patch (fdo#96244) + +* Wed May 25 2016 Than Ngo - 0.2-5 +- add url to upstream bug report + +* Tue May 24 2016 Than Ngo - 0.2-4 +- add support big endian + +* Mon May 16 2016 Rex Dieter - 0.2-3 +- ExclusiveArch primary archs, FTBFS on big endian arches (#1336466) + +* Mon May 16 2016 Rex Dieter - 0.2-2 +- link w/ --no-undefined +- fix x86 sse2 runtime detection + +* Thu May 12 2016 Rex Dieter - 0.2-1 +- webrtc-audio-processing-0.2 (#1335536) +- %%files: track ABI/API closer + +* Fri Feb 05 2016 Fedora Release Engineering - 0.1-11 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild + +* Fri Jun 19 2015 Fedora Release Engineering - 0.1-10 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild + +* Sat May 02 2015 Kalev Lember - 0.1-9 +- Rebuilt for GCC 5 C++11 ABI change + +* Mon Aug 18 2014 Fedora Release Engineering - 0.1-8 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild + +* Sun Jun 08 2014 Fedora Release Engineering - 0.1-7 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild + +* Tue Jan 28 2014 Kyle McMartin - 0.1-6 +- webrtc-fix-typedefs-on-other-arches.patch: fix ftbfs on non-x86/arm due to + a build #error in typedefs.h, however, the defines are not used anywhere in + the code. Fixes build on ppc{,64}, s390x, and aarch64. + +* Sun Aug 04 2013 Fedora Release Engineering - 0.1-5 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild + +* Thu Jul 11 2013 Debarshi Ray 0.1-4 +- Rebuilt to fix broken binary possibly caused by broken toolchain + +* Fri Feb 15 2013 Fedora Release Engineering - 0.1-3 +- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild + +* Tue Oct 9 2012 Dan Horák 0.1-2 +- set ExclusiveArch x86 and ARM for now + +* Fri Oct 5 2012 Christian Schaller 0.1-1 +- Initial Fedora spec.