Adam Faiz wrote 2 years ago
(address . guix-patches@gnu.org)(address . paren@disroot.org)
06c3f1db-b41b-2314-b597-0d16c549605e@disroot.org
From 64c8e00eab770ca697c39feabfbc60927e4ccd57 Mon Sep 17 00:00:00 2001
From: AwesomeAdam54321 <adam.faiz@disroot.org>
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
Toggle diff (490 lines)
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 @@
+
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