audio/webrtc-audio-processing: update to 1.0

Changes:	https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/compare/v0.3.1...v1.0
Reported by:	Repology

9 files changed, 46 insertions, 303 deletions

diff --git a/audio/webrtc-audio-processing/Makefile b/audio/webrtc-audio-processing/Makefile
index bcb652681cbf..bbde55f43dde 100644
--- a/audio/webrtc-audio-processing/Makefile
+++ b/audio/webrtc-audio-processing/Makefile
@@ -1,19 +1,27 @@
 # $FreeBSD$
 
 PORTNAME=	webrtc-audio-processing
-PORTVERSION=	0.3.1
-PORTREVISION=	2
+DISTVERSION=	1.0
 CATEGORIES=	audio
 MASTER_SITES=	https://freedesktop.org/software/pulseaudio/${PORTNAME}/
 
+PATCH_SITES=	https://gitlab.freedesktop.org/pulseaudio/${PORTNAME}/-/commit/
+PATCHFILES+=	2083c9a5dd34.diff:-p1 # https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/6
+
 MAINTAINER=	jbeich@FreeBSD.org
 COMMENT=	AudioProcessing module from WebRTC project
 
 LICENSE=	BSD3CLAUSE
 LICENSE_FILE=	${WRKSRC}/COPYING
 
-USES=		autoreconf compiler:c++11-lib libtool pkgconfig tar:xz
+LIB_DEPENDS=	libabsl_strings.so:devel/abseil
+
+USES=		compiler:c++14-lang localbase:ldflags meson
 USE_LDCONFIG=	yes
-GNU_CONFIGURE=	yes
+CFLAGS_i386=	-msse # https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/issues/5
+
+post-patch:
+# https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/issues/4
+	@${REINPLACE_CMD} '/absl/s/registry/reflection/' ${WRKSRC}/meson.build
 
 .include <bsd.port.mk>
diff --git a/audio/webrtc-audio-processing/distinfo b/audio/webrtc-audio-processing/distinfo
index f0b4010ef1ef..0d1441cafdb6 100644
--- a/audio/webrtc-audio-processing/distinfo
+++ b/audio/webrtc-audio-processing/distinfo
@@ -1,3 +1,5 @@
-TIMESTAMP = 1532354746
-SHA256 (webrtc-audio-processing-0.3.1.tar.xz) = a0fdd938fd85272d67e81572c5a4d9e200a0c104753cb3c209ded175ce3c5dbf
-SIZE (webrtc-audio-processing-0.3.1.tar.xz) = 695920
+TIMESTAMP = 1606505453
+SHA256 (webrtc-audio-processing-1.0.tar.gz) = 441a30d2717b2eb4145c6eb96c2d5a270fe0b4bc71aebf76716750c47be1936f
+SIZE (webrtc-audio-processing-1.0.tar.gz) = 856721
+SHA256 (2083c9a5dd34.diff) = e61660be88366621317924a3bbd6a852e07ee0912e96cbcbe332bf76a57761b9
+SIZE (2083c9a5dd34.diff) = 2243
diff --git a/audio/webrtc-audio-processing/files/patch-big-endian b/audio/webrtc-audio-processing/files/patch-big-endian
deleted file mode 100644
index 57daadce1870..000000000000
--- a/audio/webrtc-audio-processing/files/patch-big-endian
+++ /dev/null
@@ -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))
diff --git a/audio/webrtc-audio-processing/files/patch-configure.ac b/audio/webrtc-audio-processing/files/patch-configure.ac
deleted file mode 100644
index b41e30d1a20a..000000000000
--- a/audio/webrtc-audio-processing/files/patch-configure.ac
+++ /dev/null
@@ -1,18 +0,0 @@
-- Add WEBRTC_BSD as it's closer to WEBRTC_LINUX than WEBRTC_MAC
-
---- configure.ac.orig	2018-07-23 14:02:57 UTC
-+++ configure.ac
-@@ -63,6 +63,13 @@ AS_CASE(["${host}"],
-          OS_LDFLAGS="-lrt -lpthread"
-          HAVE_POSIX=1
-         ],
-+    [*-*dragonfly*|*-*bsd*],
-+        [
-+         OS_CFLAGS="-DWEBRTC_BSD -DWEBRTC_THREAD_RR"
-+         PLATFORM_CFLAGS="-DWEBRTC_POSIX"
-+         OS_LDFLAGS="-lpthread"
-+         HAVE_POSIX=1
-+        ],
-     [*-*darwin*],
-         [
-          OS_CFLAGS="-DWEBRTC_MAC -DWEBRTC_THREAD_RR -DWEBRTC_CLOCK_TYPE_REALTIME"
diff --git a/audio/webrtc-audio-processing/files/patch-webrtc_base_checks.cc b/audio/webrtc-audio-processing/files/patch-webrtc_base_checks.cc
deleted file mode 100644
index 82ffcf49c7ec..000000000000
--- a/audio/webrtc-audio-processing/files/patch-webrtc_base_checks.cc
+++ /dev/null
@@ -1,70 +0,0 @@
-- Drop unnecessary dependency on libexecinfo for GCC build
-  https://chromium.googlesource.com/external/webrtc/+/7c4dedade158%5E!/
-
---- webrtc/base/checks.cc.orig	2018-07-23 14:02:57 UTC
-+++ webrtc/base/checks.cc
-@@ -11,16 +11,10 @@
- // Most of this was borrowed (with minor modifications) from V8's and Chromium's
- // src/base/logging.cc.
- 
--// Use the C++ version to provide __GLIBCXX__.
- #include <cstdarg>
- #include <cstdio>
- #include <cstdlib>
- 
--#if defined(__GLIBCXX__) && !defined(__UCLIBC__)
--#include <cxxabi.h>
--#include <execinfo.h>
--#endif
--
- #if defined(WEBRTC_ANDROID)
- #define LOG_TAG "rtc"
- #include <android/log.h>  // NOLINT
-@@ -51,39 +45,6 @@ void PrintError(const char* format, ...) {
-   va_end(args);
- }
- 
--// TODO(ajm): This works on Mac (although the parsing fails) but I don't seem
--// to get usable symbols on Linux. This is copied from V8. Chromium has a more
--// advanced stace trace system; also more difficult to copy.
--void DumpBacktrace() {
--#if defined(__GLIBCXX__) && !defined(__UCLIBC__)
--  void* trace[100];
--  int size = backtrace(trace, sizeof(trace) / sizeof(*trace));
--  char** symbols = backtrace_symbols(trace, size);
--  PrintError("\n==== C stack trace ===============================\n\n");
--  if (size == 0) {
--    PrintError("(empty)\n");
--  } else if (symbols == NULL) {
--    PrintError("(no symbols)\n");
--  } else {
--    for (int i = 1; i < size; ++i) {
--      char mangled[201];
--      if (sscanf(symbols[i], "%*[^(]%*[(]%200[^)+]", mangled) == 1) {  // NOLINT
--        PrintError("%2d: ", i);
--        int status;
--        size_t length;
--        char* demangled = abi::__cxa_demangle(mangled, NULL, &length, &status);
--        PrintError("%s\n", demangled != NULL ? demangled : mangled);
--        free(demangled);
--      } else {
--        // If parsing failed, at least print the unparsed symbol.
--        PrintError("%s\n", symbols[i]);
--      }
--    }
--  }
--  free(symbols);
--#endif
--}
--
- FatalMessage::FatalMessage(const char* file, int line) {
-   Init(file, line);
- }
-@@ -99,7 +60,6 @@ NO_RETURN FatalMessage::~FatalMessage() {
-   fflush(stderr);
-   stream_ << std::endl << "#" << std::endl;
-   PrintError(stream_.str().c_str());
--  DumpBacktrace();
-   fflush(stderr);
-   abort();
- }
diff --git a/audio/webrtc-audio-processing/files/patch-webrtc_base_platform__thread.cc b/audio/webrtc-audio-processing/files/patch-webrtc_base_platform__thread.cc
deleted file mode 100644
index 23840052102e..000000000000
--- a/audio/webrtc-audio-processing/files/patch-webrtc_base_platform__thread.cc
+++ /dev/null
@@ -1,42 +0,0 @@
-- Implement CurrentThreadId() using global thread ID
-- Implement SetCurrentThreadName()
-
---- webrtc/base/platform_thread.cc.orig	2018-07-23 14:02:57 UTC
-+++ webrtc/base/platform_thread.cc
-@@ -19,6 +19,12 @@
- #include <sys/syscall.h>
- #endif
- 
-+#if defined(__DragonFly__) || defined(__FreeBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
-+#include <pthread_np.h>
-+#elif defined(__NetBSD__) // WEBRTC_BSD
-+#include <lwp.h>
-+#endif
-+
- namespace rtc {
- 
- PlatformThreadId CurrentThreadId() {
-@@ -32,6 +38,12 @@ PlatformThreadId CurrentThreadId() {
-   ret =  syscall(__NR_gettid);
- #elif defined(WEBRTC_ANDROID)
-   ret = gettid();
-+#elif defined(__DragonFly__) || defined(__FreeBSD__) // WEBRTC_BSD
-+  ret = pthread_getthreadid_np();
-+#elif defined(__NetBSD__) // WEBRTC_BSD
-+  ret = _lwp_self();
-+#elif defined(__OpenBSD__) // WEBRTC_BSD
-+  ret = getthrid();
- #else
-   // Default implementation for nacl and solaris.
-   ret = reinterpret_cast<pid_t>(pthread_self());
-@@ -76,6 +88,10 @@ void SetCurrentThreadName(const char* name) {
-   prctl(PR_SET_NAME, reinterpret_cast<unsigned long>(name));
- #elif defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
-   pthread_setname_np(name);
-+#elif defined(__DragonFly__) || defined(__FreeBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
-+  pthread_set_name_np(pthread_self(), name);
-+#elif defined(__NetBSD__) // WEBRTC_BSD
-+  pthread_setname_np(pthread_self(), "%s", (void*)name);
- #endif
- }
- 
diff --git a/audio/webrtc-audio-processing/files/patch-webrtc_base_stringutils.h b/audio/webrtc-audio-processing/files/patch-webrtc_base_stringutils.h
deleted file mode 100644
index c01ec8a2c14b..000000000000
--- a/audio/webrtc-audio-processing/files/patch-webrtc_base_stringutils.h
+++ /dev/null
@@ -1,13 +0,0 @@
-- BSD macro (in sys/param.h) is an archaic of the (University of California) past
-
---- webrtc/base/stringutils.h.orig	2018-07-23 14:02:57 UTC
-+++ webrtc/base/stringutils.h
-@@ -23,7 +23,7 @@
- #endif  // WEBRTC_WIN 
- 
- #if defined(WEBRTC_POSIX)
--#ifdef BSD
-+#ifdef WEBRTC_BSD
- #include <stdlib.h>
- #else  // BSD
- #include <alloca.h>
diff --git a/audio/webrtc-audio-processing/files/patch-webrtc_system__wrappers_source_condition__variable.cc b/audio/webrtc-audio-processing/files/patch-webrtc_system__wrappers_source_condition__variable.cc
deleted file mode 100644
index b1c7dd0e2d06..000000000000
--- a/audio/webrtc-audio-processing/files/patch-webrtc_system__wrappers_source_condition__variable.cc
+++ /dev/null
@@ -1,22 +0,0 @@
-- Match conditional in webrtc/system_wrappers/Makefile.am
-
---- webrtc/system_wrappers/source/condition_variable.cc.orig	2018-07-23 14:02:57 UTC
-+++ webrtc/system_wrappers/source/condition_variable.cc
-@@ -14,7 +14,7 @@
- #include <windows.h>
- #include "webrtc/system_wrappers/source/condition_variable_event_win.h"
- #include "webrtc/system_wrappers/source/condition_variable_native_win.h"
--#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
-+#elif defined(WEBRTC_POSIX)
- #include <pthread.h>
- #include "webrtc/system_wrappers/source/condition_variable_posix.h"
- #endif
-@@ -31,7 +31,7 @@ ConditionVariableWrapper* ConditionVariableWrapper::Cr
-     ret_val = new ConditionVariableEventWin();
-   }
-   return ret_val;
--#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
-+#elif defined(WEBRTC_POSIX)
-   return ConditionVariablePosix::Create();
- #else
-   return NULL;
diff --git a/audio/webrtc-audio-processing/pkg-plist b/audio/webrtc-audio-processing/pkg-plist
index 0cb8228850fb..f03f4397d672 100644
--- a/audio/webrtc-audio-processing/pkg-plist
+++ b/audio/webrtc-audio-processing/pkg-plist
@@ -1,18 +1,29 @@
-include/webrtc_audio_processing/webrtc/base/arraysize.h
-include/webrtc_audio_processing/webrtc/base/basictypes.h
-include/webrtc_audio_processing/webrtc/base/checks.h
-include/webrtc_audio_processing/webrtc/base/constructormagic.h
-include/webrtc_audio_processing/webrtc/base/maybe.h
-include/webrtc_audio_processing/webrtc/base/platform_file.h
-include/webrtc_audio_processing/webrtc/common.h
-include/webrtc_audio_processing/webrtc/common_types.h
-include/webrtc_audio_processing/webrtc/modules/audio_processing/beamformer/array_util.h
-include/webrtc_audio_processing/webrtc/modules/audio_processing/include/audio_processing.h
-include/webrtc_audio_processing/webrtc/modules/interface/module_common_types.h
-include/webrtc_audio_processing/webrtc/system_wrappers/include/trace.h
-include/webrtc_audio_processing/webrtc/typedefs.h
-lib/libwebrtc_audio_processing.a
-lib/libwebrtc_audio_processing.so
-lib/libwebrtc_audio_processing.so.1
-lib/libwebrtc_audio_processing.so.1.0.0
-libdata/pkgconfig/webrtc-audio-processing.pc
+include/webrtc-audio-processing-1/api/array_view.h
+include/webrtc-audio-processing-1/api/audio/echo_canceller3_config.h
+include/webrtc-audio-processing-1/api/audio/echo_control.h
+include/webrtc-audio-processing-1/api/scoped_refptr.h
+include/webrtc-audio-processing-1/modules/audio_coding/codecs/isac/bandwidth_info.h
+include/webrtc-audio-processing-1/modules/audio_coding/codecs/isac/main/include/isac.h
+include/webrtc-audio-processing-1/modules/audio_coding/codecs/isac/main/source/settings.h
+include/webrtc-audio-processing-1/modules/audio_processing/include/audio_processing.h
+include/webrtc-audio-processing-1/modules/audio_processing/include/audio_processing_statistics.h
+include/webrtc-audio-processing-1/modules/audio_processing/include/config.h
+include/webrtc-audio-processing-1/rtc_base/arraysize.h
+include/webrtc-audio-processing-1/rtc_base/checks.h
+include/webrtc-audio-processing-1/rtc_base/constructor_magic.h
+include/webrtc-audio-processing-1/rtc_base/deprecation.h
+include/webrtc-audio-processing-1/rtc_base/numerics/safe_compare.h
+include/webrtc-audio-processing-1/rtc_base/ref_count.h
+include/webrtc-audio-processing-1/rtc_base/system/file_wrapper.h
+include/webrtc-audio-processing-1/rtc_base/system/inline.h
+include/webrtc-audio-processing-1/rtc_base/system/rtc_export.h
+include/webrtc-audio-processing-1/rtc_base/type_traits.h
+include/webrtc-audio-processing-1/system_wrappers/include/cpu_features_wrapper.h
+include/webrtc-audio-processing-1/system_wrappers/include/metrics.h
+include/webrtc-audio-processing-1/system_wrappers/include/sleep.h
+lib/libwebrtc-audio-coding-1.so
+lib/libwebrtc-audio-coding-1.so.0
+lib/libwebrtc-audio-processing-1.so
+lib/libwebrtc-audio-processing-1.so.0
+libdata/pkgconfig/webrtc-audio-coding-1.pc
+libdata/pkgconfig/webrtc-audio-processing-1.pc