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01/02: gnu: webrtc-audio-processing: Fix build.
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From: |
guix-commits |
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Subject: |
01/02: gnu: webrtc-audio-processing: Fix build. |
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Date: |
Wed, 24 Jan 2024 12:19:23 -0500 (EST) |
apteryx pushed a commit to branch core-updates
in repository guix.
commit c1369d1b277be895d7c7f9e1defa8465e3349abe
Author: Maxim Cournoyer <maxim.cournoyer@gmail.com>
AuthorDate: Wed Jan 24 12:07:58 2024 -0500
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
---
gnu/local.mk | 1 -
gnu/packages/audio.scm | 29 +-
.../webrtc-audio-processing-big-endian.patch | 331 ---------------------
3 files changed, 2 insertions(+), 359 deletions(-)
diff --git a/gnu/local.mk b/gnu/local.mk
index 4f41e14867..f193163f14 100644
--- a/gnu/local.mk
+++ b/gnu/local.mk
@@ -2201,7 +2201,6 @@ dist_patch_DATA =
\
%D%/packages/patches/wcstools-extend-makefiles.patch \
%D%/packages/patches/wdl-link-libs-and-fix-jnetlib.patch \
%D%/packages/patches/webkitgtk-adjust-bubblewrap-paths.patch \
- %D%/packages/patches/webrtc-audio-processing-big-endian.patch \
%D%/packages/patches/webrtc-for-telegram-desktop-unbundle-libsrtp.patch \
%D%/packages/patches/websocketpp-fix-for-cmake-3.15.patch \
%D%/packages/patches/wmctrl-64-fix.patch \
diff --git a/gnu/packages/audio.scm b/gnu/packages/audio.scm
index 88e4dd2f3c..1c8ceb11a9 100644
--- a/gnu/packages/audio.scm
+++ b/gnu/packages/audio.scm
@@ -273,34 +273,9 @@ softsynth library that can be used with other
applications.")
(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