gentoo auto-resync : 29:11:2023 - 10:51:05

4 files changed, 392 insertions, 0 deletions

diff --git a/media-libs/webrtc-audio-processing/Manifest b/media-libs/webrtc-audio-processing/Manifest
index 0cbe5496ae79..25e1d2cbe99f 100644
--- a/media-libs/webrtc-audio-processing/Manifest
+++ b/media-libs/webrtc-audio-processing/Manifest
@@ -1,9 +1,12 @@
 AUX webrtc-audio-processing-0.3-Add-generic-byte-order-and-pointer-size-detection.patch 1125 BLAKE2B 5f9935ad20888444ab07cd4758d7ff0705a1fd14ea95c12e95a716cb3aa916ff21f3c6917acb24f8122051876838df9971cbd6936bdae757d55db9da57beed69 SHA512 a6f8ac5d178f2c27c0450c10948352eef5f0f9bb953f27f13904d16f7b6d5047c6ce6f2676f8727ea53aef0d834161da524929e872f18bb30c19dbc67dc35e6e
 AUX webrtc-audio-processing-0.3-big-endian-support.patch 3773 BLAKE2B 24170b27885261d397b07e603ed12aaaf5afd28d86150d62d452a78976cd791b7e40606effb4c125a7170784de6ebe7d6487fd1f8de72a6038c8c4a440d2d3c7 SHA512 65e68a26ef8953d901b96f4d48e6f976f8f18d4a48dafe5d065502910c611596b364519761db15187f3eb3c34256e6b237fcb2ae8e9c249e56f744fcf642f600
 AUX webrtc-audio-processing-0.3-proper_detection_cxxabi_execinfo.patch 2251 BLAKE2B f534e6f58d86693d257fa653d92ca07db2c1d34b88fda3b112237cff99eed81252f4111d64070f96a4265dc2655896843dbbfa01785ed18c6ea10b0eec74aeb9 SHA512 592345960101a9538c6e1197bc8cb296ca0fb0f8c6b9b64f1f4fd906ae4a9c7e9e92de740bf6e1e38cc4efca7cab8265b59542ffcad5d04bb6f1abb399851f0f
+AUX webrtc-audio-processing-1.3-Add-generic-byte-order-and-pointer-size-detection.patch 1105 BLAKE2B cc177fb92c5eba5276b8eb056dbbbb271f676b07f52ff78ff8b5d333c471b67f10d72c4d8a0a39463d6a7135da43f3056d29e9f0f9d324d1f6849f18e464498c SHA512 12670922dfaea74d168150fff1aef8ef7a2590751ba8f053a2c8ece9ef1d61bd6664fece1e37ccad07fe51fc40d3278fdca0b94bde952f28eb34ca7dfb428d4a
+AUX webrtc-audio-processing-1.3-big-endian-support.patch 13688 BLAKE2B e2f6ee383f1b93e120c378590c7b72655d2320c79ffe8a876d072f4f4647ccc7f7da90a3a8247387108ab655c99aaf48becde8524b143b7cb8a412589d640f27 SHA512 fab640e876acfa2b015d160f6727cc6d4fe0eeeca8f6efa63af7c031f179231986f2abd6b12ae67bb1300736eef8ca222019dfe7228a91b4b71d91bf237f905c
 DIST webrtc-audio-processing-0.3.1.tar.xz 695920 BLAKE2B 833c6d12b358918d95dee5b165308c8cc382f98264349fd38649bfe478557765b85d9112a35194676ee52a8ef297fb7cb7e3a570d9c2295785b6fb97d35be948 SHA512 1c7a2d16f7f6c03cf6d60405d0dcd224caae6e80c9c4d43f8373bad2446affcdf49a02efb0085387328289aa79c8981dcaedff876cde55be9602dbde9c3f440b
 DIST webrtc-audio-processing-1.3.tar.gz 879768 BLAKE2B 3bf61e5b9eadde824deb26f0591a10651d3a593ad89d3c71408655a12407bf7ecf422fdef58e651fc31245f3b0d575869e3abb0abfcfcb2e1aea21c03cd79e82 SHA512 4f56cc0acfd93b5ae432bdf681151e91344cea3388107a3eee5f9b17261cf0f09779f8b0bb67b4d35582f1f54dadc236d059802e69447e994dd588506cac95df
 EBUILD webrtc-audio-processing-0.3.1-r1.ebuild 920 BLAKE2B b3e67c3488d2f9f90d3bee1c001f5aeaaa523e4ffb9b12abb451d86dc331e172f6e9c70f7b0832a237b05040a097d553d670633080d7e467ad008193cd7a596d SHA512 607beee8f25696c4e74af1bed53c6d30da692f395e1cd3fe88d864ae5d3cf16260f5636882a6206632e08e971fc9e9f284981262fa998f17b06e850f96479628
 EBUILD webrtc-audio-processing-0.3.1.ebuild 780 BLAKE2B 9f39495c7189d7d35d7d246cd7a635d355b0fc65438c8a187b2529a2f54e1775fdad3fff0d2a61ceab3abdb4550342b007dc5e48758d20feaa8c7877cbb69f35 SHA512 b209145b656265c5b016b8ef2c22cbe3bf031230e6a3a84cdeb785c963c62e7cb4e144440c88c5c7b18d91b1ead8133752da5d40e86e3a015e4147e56f4aa0de
 EBUILD webrtc-audio-processing-1.3-r1.ebuild 709 BLAKE2B 9a92484f54423df99d622f3697258006777ed4cc915667816e42184973fa84bd0ff181fc4c5119b1a769b68e9c24e39a178edf7db0b1d357eb7f92fd03dcbe1e SHA512 8312eeeaf010a6dc7e23a5d12c34dcd913228d058df8f8da04b698245d850cde52977136c93df7af110e9f6bb6eb99971b71f897c574ba51bceea3511285dd4f
+EBUILD webrtc-audio-processing-1.3-r2.ebuild 840 BLAKE2B 62028c589e644a60c8640818f216eb128fdd043abc96a4df18bdcd950645ae635bdefce9f4c626252920efe01a06b54310dd0be9a1962da69d2d893335437894 SHA512 6e72621e3617d2017972a15959b2be75cb927d8aeb9bf27d8a47e6c5ad5e8851564cce22f1fd973fb67a82b59b3dce99a142e43bfa8389917fadf82a4a9fad27
 MISC metadata.xml 356 BLAKE2B 8852456f2e40daf7f1c67ba75e3df0f26512439b0bf1c56f85c648deeb62537b24600d49705c05f5f2afa9856d6f6d1accb5615e1a04a1a1a12bc035def7ac6b SHA512 1dbcf128eb2c1a714a822a953e05f4061fa3dc257a0bae2d8e8e9720085c5e9535f4f3373f025725f1c5ae088ab508ac97fb09e996c9fbeb5188196d15f82d3a
diff --git a/media-libs/webrtc-audio-processing/files/webrtc-audio-processing-1.3-Add-generic-byte-order-and-pointer-size-detection.patch b/media-libs/webrtc-audio-processing/files/webrtc-audio-processing-1.3-Add-generic-byte-order-and-pointer-size-detection.patch
new file mode 100644
index 000000000000..e2d974afd976
--- /dev/null
+++ b/media-libs/webrtc-audio-processing/files/webrtc-audio-processing-1.3-Add-generic-byte-order-and-pointer-size-detection.patch
@@ -0,0 +1,32 @@
+https://bugs.gentoo.org/917493
+https://sources.debian.org/src/webrtc-audio-processing/1.0-0.2/debian/patches/Add-generic-byte-order-and-pointer-size-detection.patch/
+
+Description: Add generic byte order and pointer size detection
+Author: Than <than@redhat.com>
+Origin: https://bugs.freedesktop.org/show_bug.cgi?id=95738#c4
+Last-Update: 2022-02-01
+---
+This patch header follows DEP-3: http://dep.debian.net/deps/dep3/
+--- a/webrtc/rtc_base/system/arch.h
++++ b/webrtc/rtc_base/system/arch.h
+@@ -58,7 +58,19 @@
+ #define WEBRTC_ARCH_32_BITS
+ #define WEBRTC_ARCH_LITTLE_ENDIAN
+ #else
+-#error Please add support for your architecture in rtc_base/system/arch.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/media-libs/webrtc-audio-processing/files/webrtc-audio-processing-1.3-big-endian-support.patch b/media-libs/webrtc-audio-processing/files/webrtc-audio-processing-1.3-big-endian-support.patch
new file mode 100644
index 000000000000..3984cf70124c
--- /dev/null
+++ b/media-libs/webrtc-audio-processing/files/webrtc-audio-processing-1.3-big-endian-support.patch
@@ -0,0 +1,324 @@
+https://bugs.gentoo.org/917493
+https://sources.debian.org/src/webrtc-audio-processing/1.0-0.2/debian/patches/big-endian-support.patch/
+
+Description: big endian support
+ Provide endianness converters before writing or after reading WAV
+Author: Nicholas Guriev <nicholas@guriev.su>
+Bug-telegram: https://github.com/desktop-app/tg_owt/pull/46
+Origin: https://github.com/desktop-app/tg_owt/commit/65f002e
+Last-Update: 2022-02-01
+---
+This patch header follows DEP-3: http://dep.debian.net/deps/dep3/
+--- 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 <byteswap.h>
+ #include <errno.h>
+ 
+ #include <algorithm>
+@@ -34,6 +35,38 @@
+          format == WavFormat::kWavFormatIeeeFloat;
+ }
+ 
++template <typename T>
++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<const uint16_t*>(source);
++    uint16_t* dst = reinterpret_cast<uint16_t*>(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<const uint32_t*>(source);
++    uint32_t* dst = reinterpret_cast<uint32_t*>(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<const uint64_t*>(source);
++    uint64_t* dst = reinterpret_cast<uint64_t*>(destination);
++    for (size_t index = 0; index < length; index++) {
++      dst[index] = bswap_64(src[index]);
++    }
++  }
++}
++
++template <typename T>
++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 @@
+ 
+ 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 @@
+       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 @@
+       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, 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 @@
+       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 @@
+       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 @@
+ }
+ 
+ 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<int16_t, kMaxChunksize> 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<float, kMaxChunksize> 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 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 @@
+       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 @@
+       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 <endian.h>
++
+ #include <cstring>
+ #include <limits>
+ #include <string>
+@@ -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;
+@@ -172,6 +170,8 @@
+     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.
+@@ -185,6 +185,13 @@
+   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
+@@ -212,19 +219,22 @@
+   auto header = rtc::MsanUninitialized<WavHeaderPcm>({});
+   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<uint16_t>(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<uint16_t>(8 * bytes_per_sample);
+-  header.data.header.ID = PackFourCC('d', 'a', 't', 'a');
+-  header.data.header.Size = static_cast<uint32_t>(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.
+@@ -243,24 +253,26 @@
+   auto header = rtc::MsanUninitialized<WavHeaderIeeeFloat>({});
+   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<uint16_t>(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<uint16_t>(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<uint32_t>(num_channels * num_samples);
+-  header.data.header.ID = PackFourCC('d', 'a', 't', 'a');
+-  header.data.header.Size = static_cast<uint32_t>(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.
+@@ -389,6 +401,7 @@
+     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/media-libs/webrtc-audio-processing/webrtc-audio-processing-1.3-r2.ebuild b/media-libs/webrtc-audio-processing/webrtc-audio-processing-1.3-r2.ebuild
new file mode 100644
index 000000000000..939e470e913f
--- /dev/null
+++ b/media-libs/webrtc-audio-processing/webrtc-audio-processing-1.3-r2.ebuild
@@ -0,0 +1,33 @@
+# Copyright 1999-2023 Gentoo Authors
+# Distributed under the terms of the GNU General Public License v2
+
+EAPI=7
+
+inherit meson-multilib
+
+DESCRIPTION="AudioProcessing library from the webrtc.org codebase"
+HOMEPAGE="https://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/"
+SRC_URI="https://freedesktop.org/software/pulseaudio/${PN}/${P}.tar.gz"
+
+LICENSE="BSD"
+SLOT="1"
+KEYWORDS="~amd64 ~x86 ~amd64-linux"
+IUSE="cpu_flags_arm_neon"
+
+RDEPEND="dev-cpp/abseil-cpp:=[${MULTILIB_USEDEP}]"
+DEPEND="${RDEPEND}"
+BDEPEND="virtual/pkgconfig"
+
+PATCHES=(
+	"${FILESDIR}/${PN}-1.3-Add-generic-byte-order-and-pointer-size-detection.patch"
+	"${FILESDIR}/${PN}-1.3-big-endian-support.patch"
+)
+
+DOCS=( AUTHORS NEWS README.md )
+
+multilib_src_configure() {
+	local emesonargs=(
+		-Dneon=$(usex cpu_flags_arm_neon yes no)
+	)
+	meson_src_configure
+}