Accepting request 1242817 from multimedia:libs

OBS-URL: https://build.opensuse.org/request/show/1242817
OBS-URL: https://build.opensuse.org/package/show/openSUSE:Factory/webrtc-audio-processing?expand=0&rev=16

_service

@@ -3,8 +3,8 @@
   <service name="obs_scm" mode="manual">
     <param name="scm">git</param>
     <param name="url">https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing.git</param>
-    <param name="revision">v1.3</param>
+    <param name="revision">v2.1</param>
-    <param name="versionformat">1.3</param>
+    <param name="versionformat">2.1</param>
 <!--
     <param name="revision">master</param>
     <param name="versionformat">@PARENT_TAG@+git%cd.%h</param>

baselibs.conf

@@ -1,2 +1 @@
-libwebrtc-audio-processing-1-3
+libwebrtc-audio-processing-2-1
-libwebrtc-audio-coding-1-3

big_endian_support.patch

@@ -1,90 +0,0 @@
-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<uint32_t>(written);
-   RTC_CHECK(written <= std::numeric_limits<uint32_t>::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<char*>(&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<uint32_t>(a) << 24 )
-+      |  (static_cast<uint32_t>(b) << 16)
-+      |  (static_cast<uint32_t>(c) << 8)
-+      |  (static_cast<uint32_t>(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<char*>(&x), 4);
-+}
- #endif
- 
- static inline uint32_t RiffChunkSize(uint32_t bytes_in_payload) {

big_endian_support_2.patch

@@ -1,24 +0,0 @@
-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))

fix-build.patch

@@ -1,60 +1,48 @@
-Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
+Index: webrtc-audio-processing-2.1/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
 ===================================================================
---- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
+--- webrtc-audio-processing-2.1.orig/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
-+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
++++ webrtc-audio-processing-2.1/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
-@@ -39,6 +39,7 @@ float GetLevel(const VadLevelAnalyzer::R
+@@ -58,6 +58,7 @@ rtc::FunctionView<float(float)> GetActiv
-       return vad_level.rms_dbfs;
+     case ActivationFunction::kTansigApproximated:
-       break;
+       return ::rnnoise::TansigApproximated;
-     case LevelEstimatorType::kPeak:
+     case ActivationFunction::kSigmoidApproximated:
 +    default:
-       return vad_level.peak_dbfs;
+       return ::rnnoise::SigmoidApproximated;
-       break;
    }
-Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/audio_processing_impl.cc
+ }
+Index: webrtc-audio-processing-2.1/webrtc/modules/audio_processing/agc2/input_volume_stats_reporter.cc
 ===================================================================
---- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/audio_processing_impl.cc
+--- webrtc-audio-processing-2.1.orig/webrtc/modules/audio_processing/agc2/input_volume_stats_reporter.cc
-+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/audio_processing_impl.cc
++++ webrtc-audio-processing-2.1/webrtc/modules/audio_processing/agc2/input_volume_stats_reporter.cc
-@@ -112,6 +112,7 @@ GainControl::Mode Agc1ConfigModeToInterf
+@@ -46,6 +46,7 @@ constexpr absl::string_view MetricNamePr
-     case Agc1Config::kAdaptiveDigital:
+     case InputVolumeType::kApplied:
-       return GainControl::kAdaptiveDigital;
+       return "WebRTC.Audio.Apm.AppliedInputVolume.";
-     case Agc1Config::kFixedDigital:
+     case InputVolumeType::kRecommended:
-+    default:
++     default:
-       return GainControl::kFixedDigital;
+       return "WebRTC.Audio.Apm.RecommendedInputVolume.";
    }
  }
-@@ -1852,6 +1853,7 @@ void AudioProcessingImpl::InitializeNois
+Index: webrtc-audio-processing-2.1/webrtc/modules/audio_processing/agc2/clipping_predictor.cc
-               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-2.1.orig/webrtc/modules/audio_processing/agc2/clipping_predictor.cc
-+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/include/audio_processing.cc
++++ webrtc-audio-processing-2.1/webrtc/modules/audio_processing/agc2/clipping_predictor.cc
-@@ -26,6 +26,7 @@ std::string NoiseSuppressionLevelToStrin
+@@ -373,6 +373,7 @@ std::unique_ptr<ClippingPredictor> Creat
-     case AudioProcessing::Config::NoiseSuppression::Level::kHigh:
+           config.reference_window_delay, config.clipping_threshold,
-       return "High";
+           /*adaptive_step_estimation=*/true);
-     case AudioProcessing::Config::NoiseSuppression::Level::kVeryHigh:
+     case ClippingPredictorMode::kFixedStepClippingPeakPrediction:
 +    default:
-       return "VeryHigh";
+       return std::make_unique<ClippingPeakPredictor>(
-   }
+           num_channels, config.window_length, config.reference_window_length,
- }
+           config.reference_window_delay, config.clipping_threshold,
-@@ -38,6 +39,7 @@ std::string GainController1ModeToString(
+Index: webrtc-audio-processing-2.1/webrtc/modules/audio_processing/audio_processing_impl.cc
-     case AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital:
+===================================================================
-       return "AdaptiveDigital";
+--- webrtc-audio-processing-2.1.orig/webrtc/modules/audio_processing/audio_processing_impl.cc
-     case AudioProcessing::Config::GainController1::Mode::kFixedDigital:
++++ webrtc-audio-processing-2.1/webrtc/modules/audio_processing/audio_processing_impl.cc
+@@ -165,6 +165,7 @@ int AudioFormatValidityToErrorCode(Audio
+     case AudioFormatValidity::kInvalidSampleRate:
+       return AudioProcessing::kBadSampleRateError;
+     case AudioFormatValidity::kInvalidChannelCount:
 +    default:
-       return "FixedDigital";
+       return AudioProcessing::kBadNumberChannelsError;
    }
- }
+   RTC_DCHECK(false);
-@@ -48,6 +50,7 @@ std::string GainController2LevelEstimato
-     case AudioProcessing::Config::GainController2::LevelEstimator::kRms:
-       return "Rms";
-     case AudioProcessing::Config::GainController2::LevelEstimator::kPeak:
-+    default:
-       return "Peak";
-   }
- }

fix-i586.patch

@@ -1,126 +0,0 @@
-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 <xmmintrin.h>
- 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<std::vector<FftData>>& 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<std::vector<FftData>>& 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<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;
-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<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 @@ 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<int>(x.size());
-         const int vector_limit = x_size >> 2;
-@@ -173,7 +173,7 @@ class VectorMath {
-   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;
-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<std::array<float, kFftLengthBy2Plus1>>& H2,
-     rtc::ArrayView<float> erl) {

reduce-meson-dep.patch

@@ -1,12 +0,0 @@
-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',

webrtc-audio-processing-2.1.obscpio

@@ -0,0 +1,3 @@
+version https://git-lfs.github.com/spec/v1
+oid sha256:ea446ba793ebb2f66a3f045e42b5c71863aedf954ac781b47051392884e2d757
+size 4075020

webrtc-audio-processing.changes

@@ -1,3 +1,38 @@
+-------------------------------------------------------------------
+Fri Jan 31 10:21:07 UTC 2025 - Antonio Larrosa <alarrosa@suse.com>
+
+- 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
+
 -------------------------------------------------------------------
 Tue Feb 20 15:14:05 UTC 2024 - Dominique Leuenberger <dimstar@opensuse.org>
 

webrtc-audio-processing.obsinfo

@@ -1,4 +1,4 @@
 name: webrtc-audio-processing
-version: 1.3
+version: 2.1
-mtime: 1693927187
+mtime: 1737585138
-commit: 8e258a1933d405073c9e6465628a69ac7d2a1f13
+commit: 846fe90a289f58b7c9303a635142aa2c7caa93e5

webrtc-audio-processing.spec

@@ -2,7 +2,7 @@
 #
 # spec file for package webrtc-audio-processing
 #
-# Copyright (c) 2023 SUSE LLC
+# Copyright (c) 2025 SUSE LLC
 # Copyright (c) 2012 Pascal Bleser <pascal.bleser@opensuse.org>
 #
 # All modifications and additions to the file contributed by third parties
@@ -18,11 +18,12 @@
 #
 
 
-%define pkg_soname  1-3
+%define major_version 2
-%define soname      3
+%define soname      1
+%define pkg_soname  %{major_version}-%{soname}
 # Please submit bugfixes or comments via http://bugs.opensuse.org/
 Name:           webrtc-audio-processing
-Version:        1.3
+Version:        2.1
 Release:        0
 Summary:        Real-Time Communication Library for Web Browsers
 License:        BSD-3-Clause
@@ -32,16 +33,6 @@ 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
@@ -50,7 +41,7 @@ BuildRequires:  make
 BuildRequires:  meson >= 0.59.4
 BuildRequires:  pkgconfig
 BuildRequires:  xz
-BuildRequires:  cmake(absl)
+BuildRequires:  cmake(absl) >= 20240722
 ExcludeArch:    s390 s390x ppc64
 
 %description
@@ -95,51 +86,9 @@ 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
+%autosetup -p1
 sed -i 's/\r$//' AUTHORS
-%patch -P 0 -p1
-#%%patch -P 1 -p1
-#%%patch -P 2 -p1
-%patch -P 3 -p1
-%patch -P 100 -p1
-%patch -P 101 -p1
-%patch -P 102 -p1
 
 %build
 %global _lto_cflags %{_lto_cflags} -ffat-lto-objects
@@ -149,6 +98,14 @@ sed -i 's/\r$//' AUTHORS
 	-Ddefault_library=both \
 	-Dc_args="${CFLAGS} ${LDFLAGS}" \
 	-Dcpp_args="${CXXFLAGS} ${LDFLAGS}" \
+%ifarch aarch64
+	-Dneon=enabled \
+%else
+	-Dneon=disabled \
+%endif
+%ifarch i586
+	-Dinline-sse=false \
+%endif
 	%{nil}
 %meson_build
 
@@ -159,32 +116,18 @@ 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}*
+%{_libdir}/libwebrtc-audio-processing-%{major_version}.so.%{soname}*
 
 %files -n libwebrtc-audio-processing-devel
-%{_includedir}/webrtc-audio-processing-1
+%{_includedir}/webrtc-audio-processing-%{major_version}
-%{_libdir}/libwebrtc-audio-processing-1.so
+%{_libdir}/libwebrtc-audio-processing-%{major_version}.so
-%{_libdir}/pkgconfig/webrtc-audio-processing-1.pc
+%{_libdir}/pkgconfig/webrtc-audio-processing-%{major_version}.pc
 
 %files -n libwebrtc-audio-processing-devel-static
-%{_libdir}/libwebrtc-audio-processing-1.a
+%{_libdir}/libwebrtc-audio-processing-%{major_version}.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

webrtc-ppc64.patch

@@ -1,26 +0,0 @@
-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__

webrtc-s390x.patch

@@ -1,18 +0,0 @@
---- 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__