gnu: webrtc-audio-processing: Fix build.

* gnu/packages/audio.scm (webrtc-audio-processing)
[source]: Drop patch and snippet.
[native-inputs]: Add pkg-config.
* gnu/packages/patches/webrtc-audio-processing-big-endian.patch: Delete file.
* gnu/local.mk (dist_patch_DATA): De-register it.

Change-Id: I3340371a8d484a0ad1faddedc911169e29957281

2 files changed, 2 insertions, 358 deletions

diff --git a/gnu/packages/audio.scm b/gnu/packages/audio.scm
index 0c974c078f..1c5f3f653f 100644
--- a/gnu/packages/audio.scm
+++ b/gnu/packages/audio.scm
@@ -313,34 +313,9 @@ displays a histogram of the roundtrip time jitter.")
         (string-append "http://freedesktop.org/software/pulseaudio/"
                        name "/" name "-" version ".tar.gz"))
        (sha256
-        (base32 "0xfvq5lxg612vfzk3zk6896zcb4cgrrb7fq76w9h40magz0jymcm"))
-       (modules '((guix build utils)))
-       (snippet
-        #~(begin
-            ;; See:
-            ;; <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"))))
+        (base32 "0xfvq5lxg612vfzk3zk6896zcb4cgrrb7fq76w9h40magz0jymcm"))))
     (build-system meson-build-system)
+    (native-inputs (list pkg-config))
     (inputs (list abseil-cpp))
     (synopsis "WebRTC's Audio Processing Library")
     (description "WebRTC-Audio-Processing library based on Google's
diff --git a/gnu/packages/patches/webrtc-audio-processing-big-endian.patch b/gnu/packages/patches/webrtc-audio-processing-big-endian.patch
deleted file mode 100644
index 1690597025..0000000000
--- a/gnu/packages/patches/webrtc-audio-processing-big-endian.patch
+++ /dev/null
@@ -1,331 +0,0 @@
-https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/127
-https://github.com/desktop-app/tg_owt/commit/65f002e
-
-From 65f002eeda1d97ddc70c8c49ec563987203c76f5 Mon Sep 17 00:00:00 2001
-From: Nicholas Guriev <nicholas@guriev.su>
-Date: Thu, 28 Jan 2021 20:54:06 +0300
-Subject: [PATCH] Provide endianness converters before writing or after reading
- WAV
-
----
- src/common_audio/wav_file.cc   | 80 ++++++++++++++++++++++++++-------
- src/common_audio/wav_header.cc | 81 ++++++++++++++++++++--------------
- 2 files changed, 111 insertions(+), 50 deletions(-)
-
-diff --git a/src/common_audio/wav_file.cc b/src/common_audio/wav_file.cc
-index e49126f1..b5292668 100644
---- 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 @@ bool FormatSupported(WavFormat format) {
-          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 @@ void WavReader::Reset() {
- 
- size_t WavReader::ReadSamples(const size_t num_samples,
-                               int16_t* const samples) {
--#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
--#error "Need to convert samples to big-endian when reading from WAV file"
--#endif
--
-   size_t num_samples_left_to_read = num_samples;
-   size_t next_chunk_start = 0;
-   while (num_samples_left_to_read > 0 && num_unread_samples_ > 0) {
-@@ -105,6 +134,9 @@ size_t WavReader::ReadSamples(const size_t num_samples,
-       num_bytes_read = file_.Read(samples_to_convert.data(),
-                                   chunk_size * sizeof(samples_to_convert[0]));
-       num_samples_read = num_bytes_read / sizeof(samples_to_convert[0]);
-+#ifdef WEBRTC_ARCH_BIG_ENDIAN
-+      TranslateEndianness(samples_to_convert.data(), num_samples_read);
-+#endif
- 
-       for (size_t j = 0; j < num_samples_read; ++j) {
-         samples[next_chunk_start + j] = FloatToS16(samples_to_convert[j]);
-@@ -114,6 +146,10 @@ size_t WavReader::ReadSamples(const size_t num_samples,
-       num_bytes_read = file_.Read(&samples[next_chunk_start],
-                                   chunk_size * sizeof(samples[0]));
-       num_samples_read = num_bytes_read / sizeof(samples[0]);
-+
-+#ifdef WEBRTC_ARCH_BIG_ENDIAN
-+      TranslateEndianness(&samples[next_chunk_start], num_samples_read);
-+#endif
-     }
-     RTC_CHECK(num_samples_read == 0 || (num_bytes_read % num_samples_read) == 0)
-         << "Corrupt file: file ended in the middle of a sample.";
-@@ -129,10 +165,6 @@ size_t WavReader::ReadSamples(const size_t num_samples,
- }
- 
- size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) {
--#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
--#error "Need to convert samples to big-endian when reading from WAV file"
--#endif
--
-   size_t num_samples_left_to_read = num_samples;
-   size_t next_chunk_start = 0;
-   while (num_samples_left_to_read > 0 && num_unread_samples_ > 0) {
-@@ -145,6 +177,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) {
-       num_bytes_read = file_.Read(samples_to_convert.data(),
-                                   chunk_size * sizeof(samples_to_convert[0]));
-       num_samples_read = num_bytes_read / sizeof(samples_to_convert[0]);
-+#ifdef WEBRTC_ARCH_BIG_ENDIAN
-+      TranslateEndianness(samples_to_convert.data(), num_samples_read);
-+#endif
- 
-       for (size_t j = 0; j < num_samples_read; ++j) {
-         samples[next_chunk_start + j] =
-@@ -155,6 +190,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) {
-       num_bytes_read = file_.Read(&samples[next_chunk_start],
-                                   chunk_size * sizeof(samples[0]));
-       num_samples_read = num_bytes_read / sizeof(samples[0]);
-+#ifdef WEBRTC_ARCH_BIG_ENDIAN
-+      TranslateEndianness(&samples[next_chunk_start], num_samples_read);
-+#endif
- 
-       for (size_t j = 0; j < num_samples_read; ++j) {
-         samples[next_chunk_start + j] =
-@@ -213,24 +251,32 @@ WavWriter::WavWriter(FileWrapper file,
- }
- 
- void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
--#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
--#error "Need to convert samples to little-endian when writing to WAV file"
--#endif
--
-   for (size_t i = 0; i < num_samples; i += kMaxChunksize) {
-     const size_t num_remaining_samples = num_samples - i;
-     const size_t num_samples_to_write =
-         std::min(kMaxChunksize, num_remaining_samples);
- 
-     if (format_ == WavFormat::kWavFormatPcm) {
-+#ifndef WEBRTC_ARCH_BIG_ENDIAN
-       RTC_CHECK(
-           file_.Write(&samples[i], num_samples_to_write * sizeof(samples[0])));
-+#else
-+      std::array<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 int16_t* samples, size_t num_samples) {
- }
- 
- void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
--#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
--#error "Need to convert samples to little-endian when writing to WAV file"
--#endif
--
-   for (size_t i = 0; i < num_samples; i += kMaxChunksize) {
-     const size_t num_remaining_samples = num_samples - i;
-     const size_t num_samples_to_write =
-@@ -257,6 +299,9 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
-       for (size_t j = 0; j < num_samples_to_write; ++j) {
-         converted_samples[j] = FloatS16ToS16(samples[i + j]);
-       }
-+#ifdef WEBRTC_ARCH_BIG_ENDIAN
-+      TranslateEndianness(converted_samples.data(), num_samples_to_write);
-+#endif
-       RTC_CHECK(
-           file_.Write(converted_samples.data(),
-                       num_samples_to_write * sizeof(converted_samples[0])));
-@@ -266,6 +311,9 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
-       for (size_t j = 0; j < num_samples_to_write; ++j) {
-         converted_samples[j] = FloatS16ToFloat(samples[i + j]);
-       }
-+#ifdef WEBRTC_ARCH_BIG_ENDIAN
-+      TranslateEndianness(converted_samples.data(), num_samples_to_write);
-+#endif
-       RTC_CHECK(
-           file_.Write(converted_samples.data(),
-                       num_samples_to_write * sizeof(converted_samples[0])));
-diff --git a/webrtc/common_audio/wav_header.cc b/webrtc/common_audio/wav_header.cc
-index 1ccbffca..98264a5c 100644
---- a/src/common_audio/wav_header.cc
-+++ b/src/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;
-@@ -174,6 +172,8 @@ bool FindWaveChunk(ChunkHeader* chunk_header,
-     if (readable->Read(chunk_header, sizeof(*chunk_header)) !=
-         sizeof(*chunk_header))
-       return false;  // EOF.
-+    chunk_header->Size = le32toh(chunk_header->Size);
-+
-     if (ReadFourCC(chunk_header->ID) == sought_chunk_id)
-       return true;  // Sought chunk found.
-     // Ignore current chunk by skipping its payload.
-@@ -187,6 +187,13 @@ bool ReadFmtChunkData(FmtPcmSubchunk* fmt_subchunk, WavHeaderReader* readable) {
-   if (readable->Read(&(fmt_subchunk->AudioFormat), kFmtPcmSubchunkSize) !=
-       kFmtPcmSubchunkSize)
-     return false;
-+  fmt_subchunk->AudioFormat = le16toh(fmt_subchunk->AudioFormat);
-+  fmt_subchunk->NumChannels = le16toh(fmt_subchunk->NumChannels);
-+  fmt_subchunk->SampleRate = le32toh(fmt_subchunk->SampleRate);
-+  fmt_subchunk->ByteRate = le32toh(fmt_subchunk->ByteRate);
-+  fmt_subchunk->BlockAlign = le16toh(fmt_subchunk->BlockAlign);
-+  fmt_subchunk->BitsPerSample = le16toh(fmt_subchunk->BitsPerSample);
-+
-   const uint32_t fmt_size = fmt_subchunk->header.Size;
-   if (fmt_size != kFmtPcmSubchunkSize) {
-     // There is an optional two-byte extension field permitted to be present
-@@ -214,19 +221,22 @@ void WritePcmWavHeader(size_t num_channels,
-   auto header = rtc::MsanUninitialized<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.
-@@ -245,24 +255,26 @@ void WriteIeeeFloatWavHeader(size_t num_channels,
-   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.
-@@ -391,6 +403,7 @@ bool ReadWavHeader(WavHeaderReader* readable,
-     return false;
-   if (ReadFourCC(header.riff.Format) != "WAVE")
-     return false;
-+  header.riff.header.Size = le32toh(header.riff.header.Size);
- 
-   // Find "fmt " and "data" chunks. While the official Wave file specification
-   // does not put requirements on the chunks order, it is uncommon to find the