* update webrtc-audio-processing to 2.1-1

webrtc-audio-processing/.SRCINFO

@@ -1,20 +1,22 @@
 pkgbase = webrtc-audio-processing
 	pkgdesc = AudioProcessing library based on Google's implementation of WebRTC
-	pkgver = 0.3.1
+	pkgver = 2.1
-	pkgrel = 5.1
+	pkgrel = 1
 	url = https://freedesktop.org/software/pulseaudio/webrtc-audio-processing/
 	arch = x86_64
 	arch = powerpc64le
 	arch = powerpc64
 	arch = powerpc
 	arch = riscv64
-	license = custom
+	license = BSD-3-Clause
 	makedepends = git
+	makedepends = meson
+	depends = abseil-cpp
 	depends = gcc-libs
-	provides = libwebrtc_audio_processing.so
+	depends = glibc
-	source = git+https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing.git#commit=e882a5442ac22c93648e12837248d651d18b9247
+	provides = libwebrtc-audio-processing-2.so
-	source = big-endian.patch
+	source = git+https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing.git?signed#tag=v2.1
-	b2sums = 5ebffa25cd06b1ebefb84d05e7698b0f1cd1f25355480953742bd95b13bb6348e6f0fa2aa019ce5017783eec6919debf7a98a5425bfa6e403dc8c0a4c5773b98
+	validpgpkeys = 52DFA7B8BAC74687C8A88EF48165E3D1987E2132
-	b2sums = 7c2826cf485c5c6b548adaafbdcf40d9a6e2d867f16085ef8ec0c4372d657fce0f739f42904a31a6c939e603231dbd1093d7fa2ca6372e20f782d89c3d709cff
+	b2sums = 835d68d138f6081eee657d7d454c0c3fbb6389638da5d011020a25a20002ec6e1bc8a3fd756bdfd24c0060323cd52e99e9fece4f069852139aa2d2f274f93689
 
 pkgname = webrtc-audio-processing

webrtc-audio-processing/.nvchecker.toml

@@ -0,0 +1,4 @@
+[webrtc-audio-processing]
+source = "git"
+git = "https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing.git"
+prefix = "v"

webrtc-audio-processing/PKGBUILD

@@ -2,56 +2,54 @@
 # Maintainer: Jan Alexander Steffens (heftig) <heftig@archlinux.org>
 
 pkgname=webrtc-audio-processing
-pkgver=0.3.1
+pkgver=2.1
-pkgrel=5.1
+pkgrel=1
 pkgdesc="AudioProcessing library based on Google's implementation of WebRTC"
 url="https://freedesktop.org/software/pulseaudio/webrtc-audio-processing/"
 arch=(x86_64 powerpc64le powerpc64 powerpc riscv64)
-license=(custom)
+license=(BSD-3-Clause)
-depends=(gcc-libs)
+depends=(
-makedepends=(git)
+  abseil-cpp
-provides=(libwebrtc_audio_processing.so)
+  gcc-libs
-_commit=e882a5442ac22c93648e12837248d651d18b9247  # tags/v0.3.1^0
+  glibc
-source=("git+https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing.git#commit=$_commit"
+)
-        big-endian.patch)
+makedepends=(
-b2sums=('5ebffa25cd06b1ebefb84d05e7698b0f1cd1f25355480953742bd95b13bb6348e6f0fa2aa019ce5017783eec6919debf7a98a5425bfa6e403dc8c0a4c5773b98'
+  git
-        '7c2826cf485c5c6b548adaafbdcf40d9a6e2d867f16085ef8ec0c4372d657fce0f739f42904a31a6c939e603231dbd1093d7fa2ca6372e20f782d89c3d709cff')
+  meson
-
+)
-pkgver() {
+provides=(
-  cd $pkgname
+  libwebrtc-audio-processing-2.so
-  git describe --tags | sed 's/^v//;s/[^-]*-g/r&/;s/-/+/g'
+)
-}
+source=("git+https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing.git?signed#tag=v$pkgver")
+b2sums=('835d68d138f6081eee657d7d454c0c3fbb6389638da5d011020a25a20002ec6e1bc8a3fd756bdfd24c0060323cd52e99e9fece4f069852139aa2d2f274f93689')
+validpgpkeys=(
+  52DFA7B8BAC74687C8A88EF48165E3D1987E2132 # Arun Raghavan <arun@asymptotic.io>
+)
 
 prepare() {
-  cd $pkgname
+  cd webrtc-audio-processing
-
-  patch -Np0 -i ${srcdir}/big-endian.patch
-
-  NOCONFIGURE=1 ./autogen.sh
 }
 
 build() {
-  local configure_options=(
+  local meson_options=(
-    --build=${CHOST}
+    # must match abseil
-    --prefix /usr
+    -D cpp_std=c++17
-    --disable-static
+
+    -D neon=disabled
   )
 
-  cd $pkgname
+  arch-meson webrtc-audio-processing build "${meson_options[@]}"
-  ./configure "${configure_options[@]}"
+  meson compile -C build
-  sed -i -e 's/ -shared / -Wl,-O1,--as-needed\0/g' libtool
-  make
 }
 
 check() {
-  cd $pkgname
+  meson test -C build --print-errorlogs
-  make check
 }
 
 package() {
-  cd $pkgname
+  meson install -C build --destdir "$pkgdir"
-  make DESTDIR="$pkgdir" install
+  install -Dm644 webrtc-audio-processing/COPYING \
-  install -Dt "$pkgdir/usr/share/licenses/$pkgname" -m644 COPYING
+    -t "$pkgdir/usr/share/licenses/$pkgname"
 }
 
 # vim:set sw=2 sts=-1 et:

webrtc-audio-processing/big-endian.patch

@@ -1,113 +0,0 @@
-https://bugs.freedesktop.org/show_bug.cgi?id=95738
-
---- webrtc/common_audio/wav_file.cc.orig	2018-07-23 14:02:57 UTC
-+++ webrtc/common_audio/wav_file.cc
-@@ -64,9 +64,6 @@ WavReader::~WavReader() {
- }
- 
- size_t WavReader::ReadSamples(size_t num_samples, int16_t* 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_samples, int1
-   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);
-+  }
-+#endif
-   return read;
- }
- 
-@@ -120,10 +123,17 @@ WavWriter::~WavWriter() {
- 
- 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
-+  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;
-+#else
-+  const size_t written =
-       fwrite(samples, sizeof(*samples), num_samples, file_handle_);
-+#endif
-   RTC_CHECK_EQ(num_samples, written);
-   num_samples_ += static_cast<uint32_t>(written);
-   RTC_CHECK(written <= std::numeric_limits<uint32_t>::max() ||
---- webrtc/common_audio/wav_header.cc.orig	2018-07-23 14:02:57 UTC
-+++ webrtc/common_audio/wav_header.cc
-@@ -129,7 +129,39 @@ static inline std::string ReadFourCC(uint32_t x) {
-   return std::string(reinterpret_cast<char*>(&x), 4);
- }
- #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) {
---- webrtc/typedefs.h.orig	2018-07-23 14:02:57 UTC
-+++ webrtc/typedefs.h
-@@ -48,7 +48,19 @@
- #define WEBRTC_ARCH_32_BITS
- #define WEBRTC_ARCH_LITTLE_ENDIAN
- #else
--#error Please add support for your architecture in typedefs.h
-+/* instead of failing, use typical unix defines... */
-+#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
-+#define WEBRTC_ARCH_LITTLE_ENDIAN
-+#elif __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__
-+#define WEBRTC_ARCH_BIG_ENDIAN
-+#else
-+#error __BYTE_ORDER__ is not defined
-+#endif
-+#if defined(__LP64__)
-+#define WEBRTC_ARCH_64_BITS
-+#else
-+#define WEBRTC_ARCH_32_BITS
-+#endif
- #endif
- 
- #if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))

webrtc-audio-processing/keys/pgp/52DFA7B8BAC74687C8A88EF48165E3D1987E2132.asc

@@ -0,0 +1,51 @@
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