From 5eee525eccdadaa49c43473111a9fe093a3467c9ef56eea0e5501f002df619b8 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Adrian=20Schr=C3=B6ter?= Date: Sat, 4 May 2024 01:52:13 +0200 Subject: [PATCH] Sync from SUSE:SLFO:Main webrtc-audio-processing revision 3359b25363aa46aa70faf001800fca6a --- .gitattributes | 23 ++++ _service | 20 +++ baselibs.conf | 2 + big_endian_support.patch | 90 +++++++++++++ big_endian_support_2.patch | 24 ++++ fix-build.patch | 60 +++++++++ fix-i586.patch | 126 ++++++++++++++++++ reduce-meson-dep.patch | 12 ++ webrtc-audio-processing-1.3.obscpio | 3 + webrtc-audio-processing.changes | 172 +++++++++++++++++++++++++ webrtc-audio-processing.obsinfo | 4 + webrtc-audio-processing.spec | 190 ++++++++++++++++++++++++++++ webrtc-ppc64.patch | 26 ++++ webrtc-s390x.patch | 18 +++ 14 files changed, 770 insertions(+) create mode 100644 .gitattributes create mode 100644 _service create mode 100644 baselibs.conf create mode 100644 big_endian_support.patch create mode 100644 big_endian_support_2.patch create mode 100644 fix-build.patch create mode 100644 fix-i586.patch create mode 100644 reduce-meson-dep.patch create mode 100644 webrtc-audio-processing-1.3.obscpio create mode 100644 webrtc-audio-processing.changes create mode 100644 webrtc-audio-processing.obsinfo create mode 100644 webrtc-audio-processing.spec create mode 100644 webrtc-ppc64.patch create mode 100644 webrtc-s390x.patch diff --git a/.gitattributes b/.gitattributes new file mode 100644 index 0000000..9b03811 --- /dev/null +++ b/.gitattributes @@ -0,0 +1,23 @@ +## Default LFS +*.7z filter=lfs diff=lfs merge=lfs -text +*.bsp filter=lfs diff=lfs merge=lfs -text +*.bz2 filter=lfs diff=lfs merge=lfs -text +*.gem filter=lfs diff=lfs merge=lfs -text +*.gz filter=lfs diff=lfs merge=lfs -text +*.jar filter=lfs diff=lfs merge=lfs -text +*.lz filter=lfs diff=lfs merge=lfs -text +*.lzma filter=lfs diff=lfs merge=lfs -text +*.obscpio filter=lfs diff=lfs merge=lfs -text +*.oxt filter=lfs diff=lfs merge=lfs -text +*.pdf filter=lfs diff=lfs merge=lfs -text +*.png filter=lfs diff=lfs merge=lfs -text +*.rpm filter=lfs diff=lfs merge=lfs -text +*.tbz filter=lfs diff=lfs merge=lfs -text +*.tbz2 filter=lfs diff=lfs merge=lfs -text +*.tgz filter=lfs diff=lfs merge=lfs -text +*.ttf filter=lfs diff=lfs merge=lfs -text +*.txz filter=lfs diff=lfs merge=lfs -text +*.whl filter=lfs diff=lfs merge=lfs -text +*.xz filter=lfs diff=lfs merge=lfs -text +*.zip filter=lfs diff=lfs merge=lfs -text +*.zst filter=lfs diff=lfs merge=lfs -text diff --git a/_service b/_service new file mode 100644 index 0000000..f27b2a3 --- /dev/null +++ b/_service @@ -0,0 +1,20 @@ + + + + git + https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing.git + v1.3 + 1.3 + + + + + *.tar + xz + + + + diff --git a/baselibs.conf b/baselibs.conf new file mode 100644 index 0000000..bd93661 --- /dev/null +++ b/baselibs.conf @@ -0,0 +1,2 @@ +libwebrtc-audio-processing-1-3 +libwebrtc-audio-coding-1-3 diff --git a/big_endian_support.patch b/big_endian_support.patch new file mode 100644 index 0000000..2a58630 --- /dev/null +++ b/big_endian_support.patch @@ -0,0 +1,90 @@ +diff -up webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc.than webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc +--- 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(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; +@@ -76,6 +73,12 @@ size_t WavReader::ReadSamples(size_t num + 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 num_samples - num_samples_left_to_read; + } + +@@ -120,10 +123,17 @@ WavWriter::~WavWriter() { + + 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 ++ int16_t * le_samples = new int16_t[num_samples]; ++ for(size_t idx = 0; idx < num_samples; idx++) { ++ le_samples[idx] = (samples[idx]<<8) | (samples[idx]>>8); ++ } ++ const size_t written = ++ fwrite(le_samples, sizeof(*le_samples), num_samples, file_handle_); ++ delete []le_samples; ++#else + const size_t written = + fwrite(samples, sizeof(*samples), num_samples, file_handle_); ++#endif + RTC_CHECK_EQ(num_samples, written); + num_samples_ += static_cast(written); + RTC_CHECK(written <= std::numeric_limits::max() || +diff -up webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc.than webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc +--- webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc.than 2016-05-24 08:50:52.591379263 -0400 ++++ webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc 2016-05-24 08:52:08.552606848 -0400 +@@ -129,7 +129,39 @@ static inline std::string ReadFourCC(uin + return std::string(reinterpret_cast(&x), 4); + } + #else +-#error "Write be-to-le conversion functions" ++static inline void WriteLE16(uint16_t* f, uint16_t x) { ++ *f = ((x << 8) & 0xff00) | ( ( x >> 8) & 0x00ff); ++} ++ ++static inline void WriteLE32(uint32_t* f, uint32_t x) { ++ *f = ( (x & 0x000000ff) << 24 ) ++ | ((x & 0x0000ff00) << 8) ++ | ((x & 0x00ff0000) >> 8) ++ | ((x & 0xff000000) >> 24 ); ++} ++ ++static inline void WriteFourCC(uint32_t* f, char a, char b, char c, char d) { ++ *f = (static_cast(a) << 24 ) ++ | (static_cast(b) << 16) ++ | (static_cast(c) << 8) ++ | (static_cast(d) ); ++} ++ ++static inline uint16_t ReadLE16(uint16_t x) { ++ return (( x & 0x00ff) << 8 )| ((x & 0xff00)>>8); ++} ++ ++static inline uint32_t ReadLE32(uint32_t x) { ++ return ( (x & 0x000000ff) << 24 ) ++ | ( (x & 0x0000ff00) << 8 ) ++ | ( (x & 0x00ff0000) >> 8) ++ | ( (x & 0xff000000) >> 24 ); ++} ++ ++static inline std::string ReadFourCC(uint32_t x) { ++ x = ReadLE32(x); ++ return std::string(reinterpret_cast(&x), 4); ++} + #endif + + static inline uint32_t RiffChunkSize(uint32_t bytes_in_payload) { diff --git a/big_endian_support_2.patch b/big_endian_support_2.patch new file mode 100644 index 0000000..f38262a --- /dev/null +++ b/big_endian_support_2.patch @@ -0,0 +1,24 @@ +diff -up webrtc-audio-processing-0.2/webrtc/typedefs.h.typedef webrtc-audio-processing-0.2/webrtc/typedefs.h +--- webrtc-audio-processing-0.2/webrtc/typedefs.h.typedef 2016-05-12 09:08:53.885000410 -0500 ++++ webrtc-audio-processing-0.2/webrtc/typedefs.h 2016-05-12 09:12:38.006851953 -0500 +@@ -48,7 +48,19 @@ + #define WEBRTC_ARCH_32_BITS + #define WEBRTC_ARCH_LITTLE_ENDIAN + #else +-#error Please add support for your architecture in typedefs.h ++/* instead of failing, use typical unix defines... */ ++#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__ ++#define WEBRTC_ARCH_LITTLE_ENDIAN ++#elif __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__ ++#define WEBRTC_ARCH_BIG_ENDIAN ++#else ++#error __BYTE_ORDER__ is not defined ++#endif ++#if defined(__LP64__) ++#define WEBRTC_ARCH_64_BITS ++#else ++#define WEBRTC_ARCH_32_BITS ++#endif + #endif + + #if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN)) diff --git a/fix-build.patch b/fix-build.patch new file mode 100644 index 0000000..10f559f --- /dev/null +++ b/fix-build.patch @@ -0,0 +1,60 @@ +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"; + } + } diff --git a/fix-i586.patch b/fix-i586.patch new file mode 100644 index 0000000..50f568f --- /dev/null +++ b/fix-i586.patch @@ -0,0 +1,126 @@ +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 + 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>& 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>& 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 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 x); + void Sqrt(rtc::ArrayView 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(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(x.size()); + const int vector_limit = x_size >> 2; +@@ -173,7 +173,7 @@ class VectorMath { + void Accumulate(rtc::ArrayView x, rtc::ArrayView 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(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>& H2, + rtc::ArrayView erl) { diff --git a/reduce-meson-dep.patch b/reduce-meson-dep.patch new file mode 100644 index 0000000..b190dff --- /dev/null +++ b/reduce-meson-dep.patch @@ -0,0 +1,12 @@ +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', diff --git a/webrtc-audio-processing-1.3.obscpio b/webrtc-audio-processing-1.3.obscpio new file mode 100644 index 0000000..7416e78 --- /dev/null +++ b/webrtc-audio-processing-1.3.obscpio @@ -0,0 +1,3 @@ +version https://git-lfs.github.com/spec/v1 +oid sha256:d95e27e348b777c26f66b06842269ae418ccb6cd41330d3007ae6f876114d58a +size 4396556 diff --git a/webrtc-audio-processing.changes b/webrtc-audio-processing.changes new file mode 100644 index 0000000..d7444a2 --- /dev/null +++ b/webrtc-audio-processing.changes @@ -0,0 +1,172 @@ +------------------------------------------------------------------- +Mon Oct 30 16:42:04 UTC 2023 - Antonio Larrosa + +- ExcludeArch s390, s390x and ppc64 since big endian support is + not implemented. + +------------------------------------------------------------------- +Wed Sep 20 09:49:19 UTC 2023 - Antonio Larrosa + +- Remove the tar.xz file. Having the obscpio file is enough + +------------------------------------------------------------------- +Wed Sep 20 09:38:21 UTC 2023 - Antonio Larrosa + +- Use also dashes instead of underscores in the manual Requires + +------------------------------------------------------------------- +Wed Sep 20 09:04:13 UTC 2023 - Antonio Larrosa + +- 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 - alarrosa@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 + +------------------------------------------------------------------- +Mon Aug 17 15:30:03 UTC 2020 - Dirk Mueller + +- update to 0.3.1: + * doc: file invalid reference to pulseaudio mailing list + * various build system fixes +- spec-cleaner run + +------------------------------------------------------------------- +Fri Aug 2 08:23:00 UTC 2019 - Martin Liška + +- Use FAT LTO objects in order to provide proper static library. + +------------------------------------------------------------------- +Thu Jan 12 08:32:04 UTC 2017 - olaf@aepfle.de + +- Add baselibs.conf for gstreamer-plugins-bad-32bit + +------------------------------------------------------------------- +Sat Jun 25 10:39:08 UTC 2016 - oholecek@suse.com + +- Remove webrtc-aarch64.patch, no longer needed +- Adapt the rest of webrtc- patches to new arch naming + +------------------------------------------------------------------- +Thu Jun 23 13:31:14 UTC 2016 - oholecek@suse.com + +- Remove unneeded explicit version dependency for automake + +------------------------------------------------------------------- +Wed Jun 22 11:55:11 UTC 2016 - oholecek@suse.com + +- Update to 0.3 + * build: enforce linking with --no-undefined, add explicit -lpthread + * build: Make sure files with SSE2 code are compiled with -msse2 +- Remove no-undefined.patch +- Remove webrtc-audio-processing-0.2-x86_msse2.patch +------------------------------------------------------------------- +Mon Jun 20 13:02:06 UTC 2016 - oholecek@suse.com + +- Add no-undefined.patch patch + https://cgit.freedesktop.org/pulseaudio/webrtc-audio-processing/patch/?id=d58164e4d87854233564b59e76259b72e21507f6 +- Add big_endian_support_2.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738 +- Adapt webrtc-audio-processing-0.2-x86_msse2.patch to new version +- Adapt big_endian_support.patch to new version + +------------------------------------------------------------------- +Mon May 30 09:00:51 UTC 2016 - oholecek@suse.com + +- Add webrtc-audio-processing-0.2-x86_msse2.patch patch fixing 386 build + https://lists.freedesktop.org/archives/pulseaudio-discuss/2016-May/026294.html +- Add big_endian_support.patch + https://bugs.freedesktop.org/show_bug.cgi?id=95738 +- New automake version dependency >= 1.5 + +------------------------------------------------------------------- +Thu May 26 21:19:28 UTC 2016 - oholecek@suse.com + +- Update to 0.2: + Contains API breaking changes. + + Upstream changes include: + * Rewritten AGC and voice activity detection + * Intelligibility enhancer + * Extended AEC filter + * Beamformer + * Transient suppressor + * ARM, NEON and MIPS optimisations (MIPS optimisations are not hooked up) + + API changes: + * We no longer include a top-level audio_processing.h. The webrtc tree format + is used, so use webrtc/modules/audio_processing/include/audio_processing.h + * The top-level module_common_types.h has also been moved to + webrtc/modules/interface/module_common_types.h + * C++11 support is now required while compiling client code + * AudioProcessing::Create() does not take any arguments any more + * AudioProcessing::Destroy() is gone, use standard C++ "delete" instead + * Stream parameters are now configured via StreamConfig and ProcessingConfig + rather than set_sample_rate(), set_num_channels(), etc. + * AudioFrame field names have changed + * Use config API for newer audio processing options + * Use ProcessReverseStream() instead of AnalyzeReverseStream(), particularly + when using the intelligibility enhancer + * GainControl::set_analog_level_limits() is broken. The AGC implementation + hard codes 0-255 as the volume range + + Other notes: + * The new audio processing parameters are not all tested, and a few are not + enabled upstream (in Chromium) either + * The rewritten AGC appears to be less sensitive, and it might make sense to + initialise the capture volume to something reasonable (33% or 50%, for + example) to make sure there is sufficient energy in the stream to trigger + the AGC mechanism +- Adapted all 3 arch patches + +------------------------------------------------------------------- +Thu Mar 7 13:51:31 UTC 2013 - idonmez@suse.com + +- Add patch webrtc-aarch64.patch from algraf to add aarch64 support + +------------------------------------------------------------------- +Wed Dec 19 10:39:23 CET 2012 - ro@suse.de + +- add s390 and s390x to known platforms + by adding webrtc-s390x.patch + +------------------------------------------------------------------- +Tue Jul 3 15:00:06 UTC 2012 - dvaleev@suse.com + +- add ppc64 to known platforms + +------------------------------------------------------------------- +Tue May 15 10:40:38 CET 2012 - pascal.bleser@opensuse.org + +- initial version (0.1) + diff --git a/webrtc-audio-processing.obsinfo b/webrtc-audio-processing.obsinfo new file mode 100644 index 0000000..6847668 --- /dev/null +++ b/webrtc-audio-processing.obsinfo @@ -0,0 +1,4 @@ +name: webrtc-audio-processing +version: 1.3 +mtime: 1693927187 +commit: 8e258a1933d405073c9e6465628a69ac7d2a1f13 diff --git a/webrtc-audio-processing.spec b/webrtc-audio-processing.spec new file mode 100644 index 0000000..320ffc8 --- /dev/null +++ b/webrtc-audio-processing.spec @@ -0,0 +1,190 @@ +# vim: set sw=4 ts=4 et nu: +# +# spec file for package webrtc-audio-processing +# +# Copyright (c) 2023 SUSE LLC +# Copyright (c) 2012 Pascal Bleser +# +# All modifications and additions to the file contributed by third parties +# remain the property of their copyright owners, unless otherwise agreed +# upon. The license for this file, and modifications and additions to the +# file, is the same license as for the pristine package itself (unless the +# license for the pristine package is not an Open Source License, in which +# case the license is the MIT License). An "Open Source License" is a +# license that conforms to the Open Source Definition (Version 1.9) +# published by the Open Source Initiative. + +# Please submit bugfixes or comments via https://bugs.opensuse.org/ +# + + +%define pkg_soname 1-3 +%define soname 3 +# Please submit bugfixes or comments via http://bugs.opensuse.org/ +Name: webrtc-audio-processing +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: webrtc-audio-processing-%{version}.tar.xz +Source1: baselibs.conf +# PATCH-FIX-UPSTREAM fix-build.patch alarrosa@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 +# 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) +ExcludeArch: s390 s390x ppc64 + +%description +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-%{pkg_soname} +Summary: Real-Time Communication Library for Web Browsers +Group: System/Libraries + +%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 +Summary: Real-Time Communication Library for Web Browsers +Group: Development/Libraries/C and C++ +Requires: libwebrtc-audio-processing-%{pkg_soname} = %{version} + +%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 +Summary: Real-Time Communication Library for Web Browsers +Group: Development/Libraries/C and C++ +Requires: libwebrtc-audio-processing-devel = %{version} + +%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. + +WebRTC implements the W3C's proposal for video conferencing on the web. + +%prep +%autosetup -p1 -N +sed -i 's/\r$//' AUTHORS +%patch0 -p1 +#%%patch1 -p1 +#%%patch2 -p1 +%patch3 -p1 +%patch100 -p1 +%patch101 -p1 +%patch102 -p1 + +%build +%global _lto_cflags %{_lto_cflags} -ffat-lto-objects +%meson \ + -Dc_std=gnu11 \ + -Dcpp_std=gnu++17 \ + -Ddefault_library=both \ + -Dc_args="${CFLAGS} ${LDFLAGS}" \ + -Dcpp_args="${CXXFLAGS} ${LDFLAGS}" \ + %{nil} +%meson_build + +%install +%meson_install + +find %{buildroot} -type f -name "*.la" -delete -print + +%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-coding-%{pkg_soname} +%license COPYING +%doc AUTHORS NEWS README.md UPDATING.md +%{_libdir}/libwebrtc-audio-coding-1.so.%{soname}* + +%files -n libwebrtc-audio-coding-devel +%{_libdir}/libwebrtc-audio-coding-1.so +%{_libdir}/pkgconfig/webrtc-audio-coding-1.pc + +%files -n libwebrtc-audio-coding-devel-static +%{_libdir}/libwebrtc-audio-coding-1.a + +%changelog diff --git a/webrtc-ppc64.patch b/webrtc-ppc64.patch new file mode 100644 index 0000000..7dab59c --- /dev/null +++ b/webrtc-ppc64.patch @@ -0,0 +1,26 @@ +Index: webrtc/typedefs.h +=================================================================== +--- 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 ++#elif defined(__powerpc__) ++#define WEBRTC_ARCH_BIG_ENDIAN ++#define WEBRTC_ARCH_32_BITS + #else + #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__ diff --git a/webrtc-s390x.patch b/webrtc-s390x.patch new file mode 100644 index 0000000..3dc7f44 --- /dev/null +++ b/webrtc-s390x.patch @@ -0,0 +1,18 @@ +--- 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 ++#elif defined(__s390x__) ++#define WEBRTC_ARCH_BIG_ENDIAN ++#define WEBRTC_ARCH_64_BITS ++#elif defined(__s390__) ++#define WEBRTC_ARCH_BIG_ENDIAN ++#define WEBRTC_ARCH_32_BITS + #else + #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__