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gnu: webrtc-audio-processing: Fix build.

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* 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
This commit is contained in:
Maxim Cournoyer 2024-01-24 12:07:58 -05:00 committed by Ludovic Courtès
parent 1cf91238a2
commit 0ed04bbfe2
No known key found for this signature in database
GPG key ID: 090B11993D9AEBB5
3 changed files with 2 additions and 359 deletions
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1
gnu/local.mk
View file

@ -2274,7 +2274,6 @@ dist_patch_DATA = \
%D%/packages/patches/wcstools-extend-makefiles.patch \ %D%/packages/patches/wcstools-extend-makefiles.patch \
%D%/packages/patches/wdl-link-libs-and-fix-jnetlib.patch \ %D%/packages/patches/wdl-link-libs-and-fix-jnetlib.patch \
%D%/packages/patches/webkitgtk-adjust-bubblewrap-paths.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/webrtc-for-telegram-desktop-unbundle-libsrtp.patch \
%D%/packages/patches/websocketpp-fix-for-cmake-3.15.patch \ %D%/packages/patches/websocketpp-fix-for-cmake-3.15.patch \
%D%/packages/patches/wlroots-hwdata-fallback.patch \ %D%/packages/patches/wlroots-hwdata-fallback.patch \

29
gnu/packages/audio.scm
View file

@ -313,34 +313,9 @@ (define-public webrtc-audio-processing
(string-append "http://freedesktop.org/software/pulseaudio/" (string-append "http://freedesktop.org/software/pulseaudio/"
name "/" name "-" version ".tar.gz")) name "/" name "-" version ".tar.gz"))
(sha256 (sha256
(base32 "0xfvq5lxg612vfzk3zk6896zcb4cgrrb7fq76w9h40magz0jymcm")) (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"))))
(build-system meson-build-system) (build-system meson-build-system)
(native-inputs (list pkg-config))
(inputs (list abseil-cpp)) (inputs (list abseil-cpp))
(synopsis "WebRTC's Audio Processing Library") (synopsis "WebRTC's Audio Processing Library")
(description "WebRTC-Audio-Processing library based on Google's (description "WebRTC-Audio-Processing library based on Google's

331
gnu/packages/patches/webrtc-audio-processing-big-endian.patch
View file

@ -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