Subject: [bug#60214] [PATCH core-updates] gnu: webrtc-audio-processing: Update to 1.0.

From: Adam Faiz <[email-masked]>
Date: Tue, 20 Dec 2022 16:19:21 +0800

From 64c8e00eab770ca697c39feabfbc60927e4ccd57 Mon Sep 17 00:00:00 2001
From: AwesomeAdam54321 <[email-masked]>
Date: Tue, 20 Dec 2022 15:18:58 +0800
Subject: [PATCH] gnu: webrtc-audio-processing: Update to 1.0.

* gnu/packages/audio.scm (webrtc-audio-processing): Update to 1.0.
[source]: Add snippet that fixes building on riscv and powerpc 
architectures.
[arguments]: Remove patch-source phase.
[build-system]: Update to meson-build-system.
[inputs]: Add abseil-cpp as input.
* gnu/packages/patches/webrtc-audio-processing-big-endian.patch: Update 
to 1.0.
* 
gnu/packages/patches/webrtc-audio-processing-wav-header-big-endian.patch: 
New file.
* gnu/local.mk (dist_patch_DATA): Add it.
---
  gnu/local.mk                                  |   1 +
  gnu/packages/audio.scm                        |  77 +++---
  .../webrtc-audio-processing-big-endian.patch  | 249 ++++++++++++------
  ...dio-processing-wav-header-big-endian.patch | 152 +++++++++++
  4 files changed, 357 insertions(+), 122 deletions(-)
  create mode 100644 
gnu/packages/patches/webrtc-audio-processing-wav-header-big-endian.patch

diff --git a/gnu/local.mk b/gnu/local.mk
index af570125bd..4dfc03acce 100644
--- a/gnu/local.mk
+++ b/gnu/local.mk
@@ -1996,6 +1996,7 @@ dist_patch_DATA =						\
    %D%/packages/patches/warsow-qfusion-fix-bool-return-type.patch	\
    %D%/packages/patches/webkitgtk-adjust-bubblewrap-paths.patch	\
    %D%/packages/patches/webrtc-audio-processing-big-endian.patch	\
+ 
%D%/packages/patches/webrtc-audio-processing-wav-header-big-endian.patch	\
 
%D%/packages/patches/webrtc-for-telegram-desktop-fix-gcc12-cstdint.patch   \
    %D%/packages/patches/websocketpp-fix-for-cmake-3.15.patch	\
    %D%/packages/patches/widelands-add-missing-map-include.patch	\
diff --git a/gnu/packages/audio.scm b/gnu/packages/audio.scm
index e480ce9e58..140eeb2e1d 100644
--- a/gnu/packages/audio.scm
+++ b/gnu/packages/audio.scm
@@ -67,6 +67,7 @@ (define-module (gnu packages audio)
    #:use-module (gnu packages boost)
    #:use-module (gnu packages check)
    #:use-module (gnu packages compression)
+  #:use-module (gnu packages cpp)
    #:use-module (gnu packages curl)
    #:use-module (gnu packages dbm)
    #:use-module (gnu packages documentation)
@@ -249,55 +250,45 @@ (define-public wildmidi
  (define-public webrtc-audio-processing
    (package
      (name "webrtc-audio-processing")
-    (version "0.3.1")
+    (version "1.0")
      (source
       (origin
         (method url-fetch)
         (uri
          (string-append "http://freedesktop.org/software/pulseaudio/"
-                       name "/" name "-" version ".tar.xz"))
+                       name "/" name "-" version ".tar.gz"))
         (sha256
-        (base32 "1gsx7k77blfy171b6g3m0k0s0072v6jcawhmx1kjs9w5zlwdkzd0"))))
-    (build-system gnu-build-system)
-    (arguments
-     ;; TODO: Move this to a snippet/patch or remove with the upgrade 
to 1.0.
-     (if (or (target-riscv64?)
-             (target-powerpc?))
-       (list
-         #:phases
-         #~(modify-phases %standard-phases
-             (add-after 'unpack 'patch-source
-               (lambda* (#:key inputs #:allow-other-keys)
-                 (let ((patch-file
-                        #$(local-file
-                           (search-patch
-                             "webrtc-audio-processing-big-endian.patch"))))
-                   (invoke "patch" "--force" "-p1" "-i" patch-file)
-                   (substitute* "webrtc/typedefs.h"
-                     (("defined\\(__aarch64__\\)" all)
-                      (string-append
-                        ;; powerpc-linux
-                        "(defined(__PPC__) && __SIZEOF_SIZE_T__ == 4)\n"
-                        "#define WEBRTC_ARCH_32_BITS\n"
-                        "#define WEBRTC_ARCH_BIG_ENDIAN\n"
-                        ;; powerpc64-linux
-                        "#elif (defined(__PPC64__) && 
defined(_BIG_ENDIAN))\n"
-                        "#define WEBRTC_ARCH_64_BITS\n"
-                        "#define WEBRTC_ARCH_BIG_ENDIAN\n"
-                        ;; aarch64-linux
-                        "#elif " all
-                        ;; riscv64-linux
-                        " || (defined(__riscv) && __riscv_xlen == 64)"
-                        ;; powerpc64le-linux
-                        " || (defined(__PPC64__) && 
defined(_LITTLE_ENDIAN))"))))))))
-       '()))
-    (native-inputs
-     (if (or (target-riscv64?)
-             (target-powerpc?))
-       (list
-         (local-file (search-patch 
"webrtc-audio-processing-big-endian.patch"))
-         patch)
-       '()))
+        (base32 "0vwkw5xw8l37f5vbzbkipjsf03r7b8nnrfbfbhab8bkvf79306j4"))
+       (modules '((guix build utils)))
+       (snippet
+        #~(begin
+            ;; 
https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/issues/4
+            (substitute* "meson.build"
+              (("absl_flags_registry") "absl_flags_reflection"))
+            (substitute* "webrtc/rtc_base/system/arch.h"
+              (("defined\\(__aarch64__\\)" all)
+               (string-append
+                ;; powerpc-linux
+                "(defined(__PPC__) && __SIZEOF_SIZE_T__ == 4)\n"
+                "#define WEBRTC_ARCH_32_BITS\n"
+                "#define WEBRTC_ARCH_BIG_ENDIAN\n"
+                ;; powerpc64-linux
+                "#elif (defined(__PPC64__) && defined(_BIG_ENDIAN))\n"
+                "#define WEBRTC_ARCH_64_BITS\n"
+                "#define WEBRTC_ARCH_BIG_ENDIAN\n"
+                ;; aarch64-linux
+                "#elif " all
+                ;; riscv64-linux
+                " || (defined(__riscv) && __riscv_xlen == 64)"
+                ;; powerpc64le-linux
+                " || (defined(__PPC64__) && defined(_LITTLE_ENDIAN))")))))
+       (patches
+        (search-patches
+         "webrtc-audio-processing-big-endian.patch"
+         "webrtc-audio-processing-wav-header-big-endian.patch"))))
+    (build-system meson-build-system)
+    (inputs
+     (list abseil-cpp))
      (synopsis "WebRTC's Audio Processing Library")
      (description "WebRTC-Audio-Processing library based on Google's
  implementation of WebRTC.")
diff --git 
a/gnu/packages/patches/webrtc-audio-processing-big-endian.patch 
b/gnu/packages/patches/webrtc-audio-processing-big-endian.patch
index 78333fe7b7..c77fd87c1f 100644
--- a/gnu/packages/patches/webrtc-audio-processing-big-endian.patch
+++ b/gnu/packages/patches/webrtc-audio-processing-big-endian.patch
@@ -1,93 +1,184 @@
  https://bugs.freedesktop.org/show_bug.cgi?id=95738
-https://bugs.freedesktop.org/attachment.cgi?id=124025
+https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/127
+https://github.com/desktop-app/tg_owt/commit/65f002e

-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() {
+--- webrtc-audio-processing-1.0/webrtc/common_audio/wav_file.cc 
2020-11-28 03:30:53.000000000 +0800
++++ webrtc-audio-processing-1.0/webrtc/common_audio/wav_file.cc 
2022-12-20 12:54:22.058539694 +0800
+@@ -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;
   }
-
- size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) {
+
++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:
+@@ -90,10 +123,6 @@
+
+ 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
-   // There could be metadata after the audio; ensure we don't read it.
-   num_samples = std::min(rtc::checked_cast<uint32_t>(num_samples),
-                          num_samples_remaining_);
-@@ -76,6 +73,12 @@ size_t WavReader::ReadSamples(size_t num
-   RTC_CHECK(read == num_samples || feof(file_handle_));
-   RTC_CHECK_LE(read, num_samples_remaining_);
-   num_samples_remaining_ -= rtc::checked_cast<uint32_t>(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);
-+  }
+-
+   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
-   return read;
+     }
+     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 @@
   }
-
-@@ -120,10 +123,17 @@ WavWriter::~WavWriter() {
-
+
+ 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
+-#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;
+-
+   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
-   const size_t written =
-       fwrite(samples, sizeof(*samples), num_samples, file_handle_);
++      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_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);
+       RTC_CHECK(
+           file_.Write(converted_samples.data(),
+                       num_samples_to_write * 
sizeof(converted_samples[0])));
+@@ -243,10 +289,6 @@
   }
- #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) {
+
+ 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])));
diff --git 
a/gnu/packages/patches/webrtc-audio-processing-wav-header-big-endian.patch 
b/gnu/packages/patches/webrtc-audio-processing-wav-header-big-endian.patch
new file mode 100644
index 0000000000..b25c2e364f
--- /dev/null
+++ 
b/gnu/packages/patches/webrtc-audio-processing-wav-header-big-endian.patch
@@ -0,0 +1,152 @@
+https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/127
+https://github.com/desktop-app/tg_owt/commit/65f002e
+
+--- webrtc-audio-processing-1.0/webrtc/common_audio/wav_header.cc 
2020-11-28 03:30:53.000000000 +0800
++++ webrtc-audio-processing-1.0/webrtc/common_audio/wav_header.cc 
2022-12-19 20:28:31.511187824 +0800
+@@ -15,6 +15,8 @@
+
+ #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;
+@@ -80,6 +78,7 @@
+ // read audio samples.
+ #pragma pack(2)
+ struct WavHeaderPcm {
++  WavHeaderPcm() = default;
+   WavHeaderPcm(const WavHeaderPcm&) = default;
+   WavHeaderPcm& operator=(const WavHeaderPcm&) = default;
+   RiffHeader riff;
+@@ -95,6 +94,7 @@
+ // WAV implementation.
+ #pragma pack(2)
+ struct WavHeaderIeeeFloat {
++  WavHeaderIeeeFloat() = default;
+   WavHeaderIeeeFloat(const WavHeaderIeeeFloat&) = default;
+   WavHeaderIeeeFloat& operator=(const WavHeaderIeeeFloat&) = default;
+   RiffHeader riff;
+@@ -172,6 +172,8 @@
+     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 +187,13 @@
+   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 +221,22 @@
+   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 +255,26 @@
+   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 +403,7 @@
+     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
-- 
2.38.1