gnu: Add audiofile.

Patches should fix all CVEs reported by `guix lint`: CVE-2015-7747; CVE-2017-6827, CVE-2017-6828, CVE-2017-6829, CVE-2017-6830, CVE-2017-6831, CVE-2017-6832, CVE-2017-6833, CVE-2017-6834, CVE-2017-6835, CVE-2017-6836, CVE-2017-6837, CVE-2017-6838, CVE-2017-6839; CVE-2018-13440; CVE-2018-17095 Since the patches do not reference to CVEs, it's a bit hard to tell which patch actually closes which CVE. Debian reports all these to be closed by the patches below and NixPkgs provides references. * gnu/packages/audio.scm (audiofile): New variable. * gnu/packages/patches/audiofile-fix-datatypes-in-tests.patch, gnu/packages/patches/audiofile-fix-sign-conversion.patch, gnu/packages/patches/audiofile-CVE-2015-7747.patch, gnu/packages/patches/audiofile-CVE-2018-13440.patch, gnu/packages/patches/audiofile-CVE-2018-17095.patch, gnu/packages/patches/audiofile-Check-the-number-of-coefficients.patch, gnu/packages/patches/audiofile-Fail-on-error-in-parseFormat.patch, gnu/packages/patches/audiofile-Fix-index-overflow-in-IMA.cpp.patch, gnu/packages/patches/audiofile-Fix-multiply-overflow-sfconvert.patch, gnu/packages/patches/audiofile-Fix-overflow-in-MSADPCM-decodeSam.patch, gnu/packages/patches/audiofile-division-by-zero-BlockCodec-runPull.patch, gnu/packages/patches/audiofile-hurd.patch, gnu/packages/patches/audiofile-signature-of-multiplyCheckOverflow.patch: New files. * gnu/local.mk: Add them.
2024-11-07 07:26:13 -05:00 · 2019-12-07 13:22:04 +01:00 · 2019-12-07 13:22:04 +01:00 · a8e149434e
commit a8e149434e
parent 9d25a4548c
15 changed files with 1070 additions and 0 deletions
--- a/gnu/local.mk
+++ b/gnu/local.mk
@ -714,6 +714,19 @@ dist_patch_DATA =						\
  %D%/packages/patches/ath9k-htc-firmware-gcc.patch		\
  %D%/packages/patches/ath9k-htc-firmware-objcopy.patch		\
  %D%/packages/patches/audacity-build-with-system-portaudio.patch \
+  %D%/packages/patches/audiofile-fix-datatypes-in-tests.patch	\
+  %D%/packages/patches/audiofile-fix-sign-conversion.patch	\
+  %D%/packages/patches/audiofile-CVE-2015-7747.patch		\
+  %D%/packages/patches/audiofile-CVE-2018-13440.patch		\
+  %D%/packages/patches/audiofile-CVE-2018-17095.patch		\
+  %D%/packages/patches/audiofile-Check-the-number-of-coefficients.patch \
+  %D%/packages/patches/audiofile-Fail-on-error-in-parseFormat.patch \
+  %D%/packages/patches/audiofile-Fix-index-overflow-in-IMA.cpp.patch \
+  %D%/packages/patches/audiofile-Fix-multiply-overflow-sfconvert.patch \
+  %D%/packages/patches/audiofile-Fix-overflow-in-MSADPCM-decodeSam.patch \
+  %D%/packages/patches/audiofile-division-by-zero-BlockCodec-runPull.patch \
+  %D%/packages/patches/audiofile-hurd.patch \
+  %D%/packages/patches/audiofile-signature-of-multiplyCheckOverflow.patch \
  %D%/packages/patches/automake-skip-amhello-tests.patch	\
  %D%/packages/patches/avahi-CVE-2018-1000845.patch		\
  %D%/packages/patches/avahi-localstatedir.patch		\
--- a/gnu/packages/audio.scm
+++ b/gnu/packages/audio.scm
@ -26,6 +26,7 @@
 ;;; Copyright © 2019 Alexandros Theodotou <alex@zrythm.org>
 ;;; Copyright © 2019 Christopher Lemmer Webber <cwebber@dustycloud.org>
 ;;; Copyright © 2019 Jan Wielkiewicz <tona_kosmicznego_smiecia@interia.pl>
+;;; Copyright © 2019 Hartmt Goebel <h.goebel@crazy-compilers.com>
 ;;;
 ;;; This file is part of GNU Guix.
 ;;;
@ -467,6 +468,54 @@ (define-public audacity
 tools.")
    (license license:gpl2+)))

+(define-public audiofile
+  (package
+    (name "audiofile")
+    (version "0.3.6")
+    (source
+     (origin
+       (method url-fetch)
+       (uri (string-append
+             "https://audiofile.68k.org/audiofile-" version ".tar.gz"))
+       (sha256
+        (base32 "0rb927zknk9kmhprd8rdr4azql4gn2dp75a36iazx2xhkbqhvind"))
+       (patches
+        ;; CVE references according to nixpgs
+        (search-patches
+         "audiofile-fix-datatypes-in-tests.patch"
+         "audiofile-fix-sign-conversion.patch"
+         "audiofile-hurd.patch"
+         "audiofile-CVE-2015-7747.patch"
+         ;; CVE-2017-6829:
+         "audiofile-Fix-index-overflow-in-IMA.cpp.patch"
+         ;; CVE-2017-6827, CVE-2017-6828, CVE-2017-6832, CVE-2017-6835,
+         ;; CVE-2017-6837:
+         "audiofile-Check-the-number-of-coefficients.patch"
+         ;; CVE-2017-6839:
+         "audiofile-Fix-overflow-in-MSADPCM-decodeSam.patch"
+         ;; CVE-2017-6830, CVE-2017-6834, CVE-2017-6836, CVE-2017-6838:
+         "audiofile-Fix-multiply-overflow-sfconvert.patch"
+         "audiofile-signature-of-multiplyCheckOverflow.patch"
+         ;; CVE-2017-6831:
+         "audiofile-Fail-on-error-in-parseFormat.patch"
+         ;; CVE-2017-6833:
+         "audiofile-division-by-zero-BlockCodec-runPull.patch"
+         "audiofile-CVE-2018-13440.patch"
+         "audiofile-CVE-2018-17095.patch"))))
+    (build-system gnu-build-system)
+    (inputs
+     `(("alsa-lib" ,alsa-lib)))
+    (home-page "https://audiofile.68k.org/")
+    (synopsis "Library to handle various audio file formats")
+    (description "This is an open-source version of SGI's audiofile library.
+It provides a uniform programming interface for processing of audio data to
+and from audio files of many common formats.
+
+Currently supported file formats include AIFF/AIFF-C, WAVE, and NeXT/Sun
+.snd/.au, BICS, and raw data.  Supported compression formats are currently
+G.711 mu-law and A-law.")
+    (license license:lgpl2.1+)))
+
 (define-public autotalent
  (package
    (name "autotalent")
--- a/gnu/packages/patches/audiofile-CVE-2015-7747.patch
+++ b/gnu/packages/patches/audiofile-CVE-2015-7747.patch
@ -0,0 +1,156 @@
+Description: fix buffer overflow when changing both sample format and
+ number of channels
+Origin: https://github.com/mpruett/audiofile/pull/25
+Bug-Ubuntu: https://bugs.launchpad.net/ubuntu/+source/audiofile/+bug/1502721
+Bug-Debian: https://bugs.debian.org/801102
+
+--- a/libaudiofile/modules/ModuleState.cpp
+++ b/libaudiofile/modules/ModuleState.cpp
+@@ -402,7 +402,7 @@ status ModuleState::arrange(AFfilehandle
+ 		addModule(new Transform(outfc, in.pcm, out.pcm));
+ 
+ 	if (in.channelCount != out.channelCount)
+-		addModule(new ApplyChannelMatrix(infc, isReading,
+		addModule(new ApplyChannelMatrix(outfc, isReading,
+ 			in.channelCount, out.channelCount,
+ 			in.pcm.minClip, in.pcm.maxClip,
+ 			track->channelMatrix));
+--- a/test/Makefile.am
+++ b/test/Makefile.am
+@@ -26,6 +26,7 @@ TESTS = \
+ 	VirtualFile \
+ 	floatto24 \
+ 	query2 \
+	sixteen-stereo-to-eight-mono \
+ 	sixteen-to-eight \
+ 	testchannelmatrix \
+ 	testdouble \
+@@ -139,6 +140,7 @@ printmarkers_SOURCES = printmarkers.c
+ printmarkers_LDADD = $(LIBAUDIOFILE) -lm
+ 
+ sixteen_to_eight_SOURCES = sixteen-to-eight.c TestUtilities.cpp TestUtilities.h
+sixteen_stereo_to_eight_mono_SOURCES = sixteen-stereo-to-eight-mono.c TestUtilities.cpp TestUtilities.h
+ 
+ testchannelmatrix_SOURCES = testchannelmatrix.c TestUtilities.cpp TestUtilities.h
+ 
+--- /dev/null
+++ b/test/sixteen-stereo-to-eight-mono.c
+@@ -0,0 +1,118 @@
+/*
+	Audio File Library
+
+	Copyright 2000, Silicon Graphics, Inc.
+
+	This program is free software; you can redistribute it and/or modify
+	it under the terms of the GNU General Public License as published by
+	the Free Software Foundation; either version 2 of the License, or
+	(at your option) any later version.
+
+	This program is distributed in the hope that it will be useful,
+	but WITHOUT ANY WARRANTY; without even the implied warranty of
+	MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+	GNU General Public License for more details.
+
+	You should have received a copy of the GNU General Public License along
+	with this program; if not, write to the Free Software Foundation, Inc.,
+	51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+*/
+
+/*
+	sixteen-stereo-to-eight-mono.c
+
+	This program tests the conversion from 2-channel 16-bit integers to
+	1-channel 8-bit	integers.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdint.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <unistd.h>
+#include <limits.h>
+
+#include <audiofile.h>
+
+#include "TestUtilities.h"
+
+int main (int argc, char **argv)
+{
+	AFfilehandle file;
+	AFfilesetup setup;
+	int16_t frames16[] = {14298, 392, 3923, -683, 958, -1921};
+	int8_t frames8[] = {28, 6, -2};
+	int i, frameCount = 3;
+	int8_t byte;
+	AFframecount result;
+
+	setup = afNewFileSetup();
+
+	afInitFileFormat(setup, AF_FILE_WAVE);
+
+	afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
+	afInitChannels(setup, AF_DEFAULT_TRACK, 2);
+
+	char *testFileName;
+	if (!createTemporaryFile("sixteen-to-eight", &testFileName))
+	{
+		fprintf(stderr, "Could not create temporary file.\n");
+		exit(EXIT_FAILURE);
+	}
+
+	file = afOpenFile(testFileName, "w", setup);
+	if (file == AF_NULL_FILEHANDLE)
+	{
+		fprintf(stderr, "could not open file for writing\n");
+		exit(EXIT_FAILURE);
+	}
+
+	afFreeFileSetup(setup);
+
+	afWriteFrames(file, AF_DEFAULT_TRACK, frames16, frameCount);
+
+	afCloseFile(file);
+
+	file = afOpenFile(testFileName, "r", AF_NULL_FILESETUP);
+	if (file == AF_NULL_FILEHANDLE)
+	{
+		fprintf(stderr, "could not open file for reading\n");
+		exit(EXIT_FAILURE);
+	}
+
+	afSetVirtualSampleFormat(file, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 8);
+	afSetVirtualChannels(file, AF_DEFAULT_TRACK, 1);
+
+	for (i=0; i<frameCount; i++)
+	{
+		/* Read one frame. */
+		result = afReadFrames(file, AF_DEFAULT_TRACK, &byte, 1);
+
+		if (result != 1)
+			break;
+
+		/* Compare the byte read with its precalculated value. */
+		if (memcmp(&byte, &frames8[i], 1) != 0)
+		{
+			printf("error\n");
+			printf("expected %d, got %d\n", frames8[i], byte);
+			exit(EXIT_FAILURE);
+		}
+		else
+		{
+#ifdef DEBUG
+			printf("got what was expected: %d\n", byte);
+#endif
+		}
+	}
+
+	afCloseFile(file);
+	unlink(testFileName);
+	free(testFileName);
+
+	exit(EXIT_SUCCESS);
+}
--- a/gnu/packages/patches/audiofile-CVE-2018-13440.patch
+++ b/gnu/packages/patches/audiofile-CVE-2018-13440.patch
@ -0,0 +1,28 @@
+From fde6d79fb8363c4a329a184ef0b107156602b225 Mon Sep 17 00:00:00 2001
+From: Wim Taymans <wtaymans@redhat.com>
+Date: Thu, 27 Sep 2018 10:48:45 +0200
+Subject: [PATCH] ModuleState: handle compress/decompress init failure
+
+When the unit initcompress or initdecompress function fails,
+m_fileModule is NULL. Return AF_FAIL in that case instead of
+causing NULL pointer dereferences later.
+
+Fixes #49
+---
+ libaudiofile/modules/ModuleState.cpp | 3 +++
+ 1 file changed, 3 insertions(+)
+
+diff --git a/libaudiofile/modules/ModuleState.cpp b/libaudiofile/modules/ModuleState.cpp
+index 0c29d7a..070fd9b 100644
+--- a/libaudiofile/modules/ModuleState.cpp
+++ b/libaudiofile/modules/ModuleState.cpp
+@@ -75,6 +75,9 @@ status ModuleState::initFileModule(AFfilehandle file, Track *track)
+ 		m_fileModule = unit->initcompress(track, file->m_fh, file->m_seekok,
+ 			file->m_fileFormat == AF_FILE_RAWDATA, &chunkFrames);
+ 
+	if (!m_fileModule)
+		return AF_FAIL;
+
+ 	if (unit->needsRebuffer)
+ 	{
+ 		assert(unit->nativeSampleFormat == AF_SAMPFMT_TWOSCOMP);
--- a/gnu/packages/patches/audiofile-CVE-2018-17095.patch
+++ b/gnu/packages/patches/audiofile-CVE-2018-17095.patch
@ -0,0 +1,26 @@
+From 822b732fd31ffcb78f6920001e9b1fbd815fa712 Mon Sep 17 00:00:00 2001
+From: Wim Taymans <wtaymans@redhat.com>
+Date: Thu, 27 Sep 2018 12:11:12 +0200
+Subject: [PATCH] SimpleModule: set output chunk framecount after pull
+
+After pulling the data, set the output chunk to the amount of
+frames we pulled so that the next module in the chain has the correct
+frame count.
+
+Fixes #50 and #51
+---
+ libaudiofile/modules/SimpleModule.cpp | 1 +
+ 1 file changed, 1 insertion(+)
+
+diff --git a/libaudiofile/modules/SimpleModule.cpp b/libaudiofile/modules/SimpleModule.cpp
+index 2bae1eb..e87932c 100644
+--- a/libaudiofile/modules/SimpleModule.cpp
+++ b/libaudiofile/modules/SimpleModule.cpp
+@@ -26,6 +26,7 @@
+ void SimpleModule::runPull()
+ {
+ 	pull(m_outChunk->frameCount);
+	m_outChunk->frameCount = m_inChunk->frameCount;
+ 	run(*m_inChunk, *m_outChunk);
+ }
+ 
--- a/gnu/packages/patches/audiofile-Check-the-number-of-coefficients.patch
+++ b/gnu/packages/patches/audiofile-Check-the-number-of-coefficients.patch
@ -0,0 +1,30 @@
+From: Antonio Larrosa <larrosa@kde.org>
+Date: Mon, 6 Mar 2017 12:51:22 +0100
+Subject: Always check the number of coefficients
+
+When building the library with NDEBUG, asserts are eliminated
+so it's better to always check that the number of coefficients
+is inside the array range.
+
+This fixes the 00191-audiofile-indexoob issue in #41
+---
+ libaudiofile/WAVE.cpp | 6 ++++++
+ 1 file changed, 6 insertions(+)
+
+diff --git a/libaudiofile/WAVE.cpp b/libaudiofile/WAVE.cpp
+index 9dd8511..0fc48e8 100644
+--- a/libaudiofile/WAVE.cpp
+++ b/libaudiofile/WAVE.cpp
+@@ -281,6 +281,12 @@ status WAVEFile::parseFormat(const Tag &id, uint32_t size)
+ 
+ 			/* numCoefficients should be at least 7. */
+ 			assert(numCoefficients >= 7 && numCoefficients <= 255);
+			if (numCoefficients < 7 || numCoefficients > 255)
+			{
+				_af_error(AF_BAD_HEADER,
+						"Bad number of coefficients");
+				return AF_FAIL;
+			}
+ 
+ 			m_msadpcmNumCoefficients = numCoefficients;
+ 
--- a/gnu/packages/patches/audiofile-Fail-on-error-in-parseFormat.patch
+++ b/gnu/packages/patches/audiofile-Fail-on-error-in-parseFormat.patch
@ -0,0 +1,36 @@
+From: Antonio Larrosa <larrosa@kde.org>
+Date: Mon, 6 Mar 2017 18:59:26 +0100
+Subject: Actually fail when error occurs in parseFormat
+
+When there's an unsupported number of bits per sample or an invalid
+number of samples per block, don't only print an error message using
+the error handler, but actually stop parsing the file.
+
+This fixes #35 (also reported at
+https://bugzilla.opensuse.org/show_bug.cgi?id=1026983 and
+https://blogs.gentoo.org/ago/2017/02/20/audiofile-heap-based-buffer-overflow-in-imadecodeblockwave-ima-cpp/
+)
+---
+ libaudiofile/WAVE.cpp | 2 ++
+ 1 file changed, 2 insertions(+)
+
+diff --git a/libaudiofile/WAVE.cpp b/libaudiofile/WAVE.cpp
+index 0fc48e8..d04b796 100644
+--- a/libaudiofile/WAVE.cpp
+++ b/libaudiofile/WAVE.cpp
+@@ -332,6 +332,7 @@ status WAVEFile::parseFormat(const Tag &id, uint32_t size)
+ 			{
+ 				_af_error(AF_BAD_NOT_IMPLEMENTED,
+ 					"IMA ADPCM compression supports only 4 bits per sample");
+				return AF_FAIL;
+ 			}
+ 
+ 			int bytesPerBlock = (samplesPerBlock + 14) / 8 * 4 * channelCount;
+@@ -339,6 +340,7 @@ status WAVEFile::parseFormat(const Tag &id, uint32_t size)
+ 			{
+ 				_af_error(AF_BAD_CODEC_CONFIG,
+ 					"Invalid samples per block for IMA ADPCM compression");
+				return AF_FAIL;
+ 			}
+ 
+ 			track->f.sampleWidth = 16;
--- a/gnu/packages/patches/audiofile-Fix-index-overflow-in-IMA.cpp.patch
+++ b/gnu/packages/patches/audiofile-Fix-index-overflow-in-IMA.cpp.patch
@ -0,0 +1,33 @@
+From: Antonio Larrosa <larrosa@kde.org>
+Date: Mon, 6 Mar 2017 18:02:31 +0100
+Subject: clamp index values to fix index overflow in IMA.cpp
+
+This fixes #33
+(also reported at https://bugzilla.opensuse.org/show_bug.cgi?id=1026981
+and https://blogs.gentoo.org/ago/2017/02/20/audiofile-global-buffer-overflow-in-decodesample-ima-cpp/)
+---
+ libaudiofile/modules/IMA.cpp | 4 ++--
+ 1 file changed, 2 insertions(+), 2 deletions(-)
+
+diff --git a/libaudiofile/modules/IMA.cpp b/libaudiofile/modules/IMA.cpp
+index 7476d44..df4aad6 100644
+--- a/libaudiofile/modules/IMA.cpp
+++ b/libaudiofile/modules/IMA.cpp
+@@ -169,7 +169,7 @@ int IMA::decodeBlockWAVE(const uint8_t *encoded, int16_t *decoded)
+ 		if (encoded[1] & 0x80)
+ 			m_adpcmState[c].previousValue -= 0x10000;
+ 
+-		m_adpcmState[c].index = encoded[2];
+		m_adpcmState[c].index = clamp(encoded[2], 0, 88);
+ 
+ 		*decoded++ = m_adpcmState[c].previousValue;
+ 
+@@ -210,7 +210,7 @@ int IMA::decodeBlockQT(const uint8_t *encoded, int16_t *decoded)
+ 			predictor -= 0x10000;
+ 
+ 		state.previousValue = clamp(predictor, MIN_INT16, MAX_INT16);
+-		state.index = encoded[1] & 0x7f;
+		state.index = clamp(encoded[1] & 0x7f, 0, 88);
+ 		encoded += 2;
+ 
+ 		for (int n=0; n<m_framesPerPacket; n+=2)
--- a/gnu/packages/patches/audiofile-Fix-multiply-overflow-sfconvert.patch
+++ b/gnu/packages/patches/audiofile-Fix-multiply-overflow-sfconvert.patch
@ -0,0 +1,66 @@
+From: Antonio Larrosa <larrosa@kde.org>
+Date: Mon, 6 Mar 2017 13:54:52 +0100
+Subject: Check for multiplication overflow in sfconvert
+
+Checks that a multiplication doesn't overflow when
+calculating the buffer size, and if it overflows,
+reduce the buffer size instead of failing.
+
+This fixes the 00192-audiofile-signintoverflow-sfconvert case
+in #41
+---
+ sfcommands/sfconvert.c | 34 ++++++++++++++++++++++++++++++++--
+ 1 file changed, 32 insertions(+), 2 deletions(-)
+
+diff --git a/sfcommands/sfconvert.c b/sfcommands/sfconvert.c
+index 80a1bc4..970a3e4 100644
+--- a/sfcommands/sfconvert.c
+++ b/sfcommands/sfconvert.c
+@@ -45,6 +45,33 @@ void printusage (void);
+ void usageerror (void);
+ bool copyaudiodata (AFfilehandle infile, AFfilehandle outfile, int trackid);
+ 
+int firstBitSet(int x)
+{
+        int position=0;
+        while (x!=0)
+        {
+                x>>=1;
+                ++position;
+        }
+        return position;
+}
+
+#ifndef __has_builtin
+#define __has_builtin(x) 0
+#endif
+
+int multiplyCheckOverflow(int a, int b, int *result)
+{
+#if (defined __GNUC__ && __GNUC__ >= 5) || ( __clang__ && __has_builtin(__builtin_mul_overflow))
+	return __builtin_mul_overflow(a, b, result);
+#else
+	if (firstBitSet(a)+firstBitSet(b)>31) // int is signed, so we can't use 32 bits
+		return true;
+	*result = a * b;
+	return false;
+#endif
+}
+
+ int main (int argc, char **argv)
+ {
+ 	if (argc == 2)
+@@ -323,8 +350,11 @@ bool copyaudiodata (AFfilehandle infile, AFfilehandle outfile, int trackid)
+ {
+ 	int frameSize = afGetVirtualFrameSize(infile, trackid, 1);
+ 
+-	const int kBufferFrameCount = 65536;
+-	void *buffer = malloc(kBufferFrameCount * frameSize);
+	int kBufferFrameCount = 65536;
+	int bufferSize;
+	while (multiplyCheckOverflow(kBufferFrameCount, frameSize, &bufferSize))
+		kBufferFrameCount /= 2;
+	void *buffer = malloc(bufferSize);
+ 
+ 	AFframecount totalFrames = afGetFrameCount(infile, AF_DEFAULT_TRACK);
+ 	AFframecount totalFramesWritten = 0;
--- a/gnu/packages/patches/audiofile-Fix-overflow-in-MSADPCM-decodeSam.patch
+++ b/gnu/packages/patches/audiofile-Fix-overflow-in-MSADPCM-decodeSam.patch
@ -0,0 +1,116 @@
+From: Antonio Larrosa <larrosa@kde.org>
+Date: Mon, 6 Mar 2017 13:43:53 +0100
+Subject: Check for multiplication overflow in MSADPCM decodeSample
+
+Check for multiplication overflow (using __builtin_mul_overflow
+if available) in MSADPCM.cpp decodeSample and return an empty
+decoded block if an error occurs.
+
+This fixes the 00193-audiofile-signintoverflow-MSADPCM case of #41
+---
+ libaudiofile/modules/BlockCodec.cpp |  5 ++--
+ libaudiofile/modules/MSADPCM.cpp    | 47 +++++++++++++++++++++++++++++++++----
+ 2 files changed, 46 insertions(+), 6 deletions(-)
+
+diff --git a/libaudiofile/modules/BlockCodec.cpp b/libaudiofile/modules/BlockCodec.cpp
+index 45925e8..4731be1 100644
+--- a/libaudiofile/modules/BlockCodec.cpp
+++ b/libaudiofile/modules/BlockCodec.cpp
+@@ -52,8 +52,9 @@ void BlockCodec::runPull()
+ 	// Decompress into m_outChunk.
+ 	for (int i=0; i<blocksRead; i++)
+ 	{
+-		decodeBlock(static_cast<const uint8_t *>(m_inChunk->buffer) + i * m_bytesPerPacket,
+-			static_cast<int16_t *>(m_outChunk->buffer) + i * m_framesPerPacket * m_track->f.channelCount);
+		if (decodeBlock(static_cast<const uint8_t *>(m_inChunk->buffer) + i * m_bytesPerPacket,
+			static_cast<int16_t *>(m_outChunk->buffer) + i * m_framesPerPacket * m_track->f.channelCount)==0)
+			break;
+ 
+ 		framesRead += m_framesPerPacket;
+ 	}
+diff --git a/libaudiofile/modules/MSADPCM.cpp b/libaudiofile/modules/MSADPCM.cpp
+index 8ea3c85..ef9c38c 100644
+--- a/libaudiofile/modules/MSADPCM.cpp
+++ b/libaudiofile/modules/MSADPCM.cpp
+@@ -101,24 +101,60 @@ static const int16_t adaptationTable[] =
+ 	768, 614, 512, 409, 307, 230, 230, 230
+ };
+ 
+int firstBitSet(int x)
+{
+        int position=0;
+        while (x!=0)
+        {
+                x>>=1;
+                ++position;
+        }
+        return position;
+}
+
+#ifndef __has_builtin
+#define __has_builtin(x) 0
+#endif
+
+int multiplyCheckOverflow(int a, int b, int *result)
+{
+#if (defined __GNUC__ && __GNUC__ >= 5) || ( __clang__ && __has_builtin(__builtin_mul_overflow))
+	return __builtin_mul_overflow(a, b, result);
+#else
+	if (firstBitSet(a)+firstBitSet(b)>31) // int is signed, so we can't use 32 bits
+		return true;
+	*result = a * b;
+	return false;
+#endif
+}
+
+
+ // Compute a linear PCM value from the given differential coded value.
+ static int16_t decodeSample(ms_adpcm_state &state,
+-	uint8_t code, const int16_t *coefficient)
+	uint8_t code, const int16_t *coefficient, bool *ok=NULL)
+ {
+ 	int linearSample = (state.sample1 * coefficient[0] +
+ 		state.sample2 * coefficient[1]) >> 8;
+	int delta;
+ 
+ 	linearSample += ((code & 0x08) ? (code - 0x10) : code) * state.delta;
+ 
+ 	linearSample = clamp(linearSample, MIN_INT16, MAX_INT16);
+ 
+-	int delta = (state.delta * adaptationTable[code]) >> 8;
+	if (multiplyCheckOverflow(state.delta, adaptationTable[code], &delta))
+	{
+                if (ok) *ok=false;
+		_af_error(AF_BAD_COMPRESSION, "Error decoding sample");
+		return 0;
+	}
+	delta >>= 8;
+ 	if (delta < 16)
+ 		delta = 16;
+ 
+ 	state.delta = delta;
+ 	state.sample2 = state.sample1;
+ 	state.sample1 = linearSample;
+	if (ok) *ok=true;
+ 
+ 	return static_cast<int16_t>(linearSample);
+ }
+@@ -212,13 +248,16 @@ int MSADPCM::decodeBlock(const uint8_t *encoded, int16_t *decoded)
+ 	{
+ 		uint8_t code;
+ 		int16_t newSample;
+		bool ok;
+ 
+ 		code = *encoded >> 4;
+-		newSample = decodeSample(*state[0], code, coefficient[0]);
+		newSample = decodeSample(*state[0], code, coefficient[0], &ok);
+		if (!ok) return 0;
+ 		*decoded++ = newSample;
+ 
+ 		code = *encoded & 0x0f;
+-		newSample = decodeSample(*state[1], code, coefficient[1]);
+		newSample = decodeSample(*state[1], code, coefficient[1], &ok);
+		if (!ok) return 0;
+ 		*decoded++ = newSample;
+ 
+ 		encoded++;
--- a/gnu/packages/patches/audiofile-division-by-zero-BlockCodec-runPull.patch
+++ b/gnu/packages/patches/audiofile-division-by-zero-BlockCodec-runPull.patch
@ -0,0 +1,21 @@
+From: Antonio Larrosa <larrosa@kde.org>
+Date: Thu, 9 Mar 2017 10:21:18 +0100
+Subject: Check for division by zero in BlockCodec::runPull
+
+---
+ libaudiofile/modules/BlockCodec.cpp | 2 +-
+ 1 file changed, 1 insertion(+), 1 deletion(-)
+
+diff --git a/libaudiofile/modules/BlockCodec.cpp b/libaudiofile/modules/BlockCodec.cpp
+index 4731be1..eb2fb4d 100644
+--- a/libaudiofile/modules/BlockCodec.cpp
+++ b/libaudiofile/modules/BlockCodec.cpp
+@@ -47,7 +47,7 @@ void BlockCodec::runPull()
+ 
+ 	// Read the compressed data.
+ 	ssize_t bytesRead = read(m_inChunk->buffer, m_bytesPerPacket * blockCount);
+-	int blocksRead = bytesRead >= 0 ? bytesRead / m_bytesPerPacket : 0;
+	int blocksRead = (bytesRead >= 0 && m_bytesPerPacket > 0) ? bytesRead / m_bytesPerPacket : 0;
+ 
+ 	// Decompress into m_outChunk.
+ 	for (int i=0; i<blocksRead; i++)
--- a/gnu/packages/patches/audiofile-fix-datatypes-in-tests.patch
+++ b/gnu/packages/patches/audiofile-fix-datatypes-in-tests.patch
@ -0,0 +1,54 @@
+Based on (hunks for changelog and Identy.cpp removed)
+From ecbc07f0ed336187cc9a67c3363f89681b8b8f52 Mon Sep 17 00:00:00 2001
+From: Michael Pruett <michael@68k.org>
+Date: Tue, 5 Jul 2016 23:26:16 -0500
+Subject: [PATCH] Fix type of test data arrays.
+
+
+
+
+---
+ ChangeLog         | 6 ++++++
+ test/Identify.cpp | 3 ++-
+ test/NeXT.cpp     | 7 ++++---
+ 3 files changed, 12 insertions(+), 4 deletions(-)
+
+diff --git a/test/NeXT.cpp b/test/NeXT.cpp
+index 7e39850..29af877 100644
+--- a/test/NeXT.cpp
+++ b/test/NeXT.cpp
+@@ -30,6 +30,7 @@
+ #include <audiofile.h>
+ #include <fcntl.h>
+ #include <gtest/gtest.h>
+#include <stdint.h>
+ #include <sys/stat.h>
+ #include <sys/types.h>
+ #include <unistd.h>
+@@ -37,7 +38,7 @@
+ 
+ #include "TestUtilities.h"
+ 
+-const char kDataUnspecifiedLength[] =
+const uint8_t kDataUnspecifiedLength[] =
+ {
+ 	'.', 's', 'n', 'd',
+ 	0, 0, 0, 24, // offset of 24 bytes
+@@ -57,7 +58,7 @@ const char kDataUnspecifiedLength[] =
+ 	0, 55
+ };
+ 
+-const char kDataTruncated[] =
+const uint8_t kDataTruncated[] =
+ {
+ 	'.', 's', 'n', 'd',
+ 	0, 0, 0, 24, // offset of 24 bytes
+@@ -152,7 +153,7 @@ TEST(NeXT, Truncated)
+ 	ASSERT_EQ(::unlink(testFileName.c_str()), 0);
+ }
+ 
+-const char kDataZeroChannels[] =
+const uint8_t kDataZeroChannels[] =
+ {
+ 	'.', 's', 'n', 'd',
+ 	0, 0, 0, 24, // offset of 24 bytes
--- a/gnu/packages/patches/audiofile-fix-sign-conversion.patch
+++ b/gnu/packages/patches/audiofile-fix-sign-conversion.patch
@ -0,0 +1,26 @@
+Based on (hunk for changelog removed)
+From b62c902dd258125cac86cd2df21fc898035a43d3 Mon Sep 17 00:00:00 2001
+From: Michael Pruett <michael@68k.org>
+Date: Mon, 29 Aug 2016 23:08:26 -0500
+Subject: [PATCH] Fix undefined behavior in sign conversion.
+
+
+---
+ ChangeLog                           | 5 +++++
+ libaudiofile/modules/SimpleModule.h | 3 ++-
+ 2 files changed, 7 insertions(+), 1 deletion(-)
+
+diff --git a/libaudiofile/modules/SimpleModule.h b/libaudiofile/modules/SimpleModule.h
+index 03c6c69..bad85ad 100644
+--- a/libaudiofile/modules/SimpleModule.h
+++ b/libaudiofile/modules/SimpleModule.h
+@@ -123,7 +123,8 @@ struct signConverter
+ 	typedef typename IntTypes<Format>::UnsignedType UnsignedType;
+ 
+ 	static const int kScaleBits = (Format + 1) * CHAR_BIT - 1;
+-	static const int kMinSignedValue = -1 << kScaleBits;
+	static const int kMaxSignedValue = (((1 << (kScaleBits - 1)) - 1) << 1) + 1;
+	static const int kMinSignedValue = -kMaxSignedValue - 1;
+ 
+ 	struct signedToUnsigned : public std::unary_function<SignedType, UnsignedType>
+ 	{
--- a/gnu/packages/patches/audiofile-hurd.patch
+++ b/gnu/packages/patches/audiofile-hurd.patch
@ -0,0 +1,381 @@
+Description: Remove usage of PATH_MAX in tests to fix FTBFS on Hurd.
+ jcowgill: Removed Changelog changes
+Author: Pino Toscano <toscano.pino@tiscali.it>
+Origin: backport, https://github.com/mpruett/audiofile/commit/34c261034f1193a783196618f0052112e00fbcfe
+Bug: https://github.com/mpruett/audiofile/pull/17
+Bug-Debian: https://bugs.debian.org/762595
+---
+This patch header follows DEP-3: http://dep.debian.net/deps/dep3/
+
+--- a/test/TestUtilities.cpp
+++ b/test/TestUtilities.cpp
+@@ -21,8 +21,8 @@
+ #include "TestUtilities.h"
+ 
+ #include <limits.h>
+-#include <stdio.h>
+ #include <stdlib.h>
+#include <string.h>
+ #include <unistd.h>
+ 
+ bool createTemporaryFile(const std::string &prefix, std::string *path)
+@@ -35,12 +35,12 @@ bool createTemporaryFile(const std::stri
+ 	return true;
+ }
+ 
+-bool createTemporaryFile(const char *prefix, char *path)
+bool createTemporaryFile(const char *prefix, char **path)
+ {
+-	snprintf(path, PATH_MAX, "/tmp/%s-XXXXXX", prefix);
+-	int fd = ::mkstemp(path);
+-	if (fd < 0)
+-		return false;
+-	::close(fd);
+-	return true;
+	*path = NULL;
+	std::string pathString;
+	bool result = createTemporaryFile(prefix, &pathString);
+	if (result)
+		*path = ::strdup(pathString.c_str());
+	return result;
+ }
+--- a/test/TestUtilities.h
+++ b/test/TestUtilities.h
+@@ -53,7 +53,7 @@ extern "C" {
+ 
+ #include <stdbool.h>
+ 
+-bool createTemporaryFile(const char *prefix, char *path);
+bool createTemporaryFile(const char *prefix, char **path);
+ 
+ #ifdef __cplusplus
+ }
+--- a/test/floatto24.c
+++ b/test/floatto24.c
+@@ -86,8 +86,8 @@ int main (int argc, char **argv)
+ 	afInitChannels(setup, AF_DEFAULT_TRACK, 1);
+ 	afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_FLOAT, 32);
+ 
+-	char testFileName[PATH_MAX];
+-	if (!createTemporaryFile("floatto24", testFileName))
+	char *testFileName;
+	if (!createTemporaryFile("floatto24", &testFileName))
+ 	{
+ 		fprintf(stderr, "Could not create temporary file.\n");
+ 		exit(EXIT_FAILURE);
+@@ -182,6 +182,7 @@ int main (int argc, char **argv)
+ 	}
+ 
+ 	unlink(testFileName);
+	free(testFileName);
+ 
+ 	exit(EXIT_SUCCESS);
+ }
+--- a/test/sixteen-to-eight.c
+++ b/test/sixteen-to-eight.c
+@@ -57,8 +57,8 @@ int main (int argc, char **argv)
+ 	afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_UNSIGNED, 8);
+ 	afInitChannels(setup, AF_DEFAULT_TRACK, 1);
+ 
+-	char testFileName[PATH_MAX];
+-	if (!createTemporaryFile("sixteen-to-eight", testFileName))
+	char *testFileName;
+	if (!createTemporaryFile("sixteen-to-eight", &testFileName))
+ 	{
+ 		fprintf(stderr, "Could not create temporary file.\n");
+ 		exit(EXIT_FAILURE);
+@@ -113,6 +113,7 @@ int main (int argc, char **argv)
+ 
+ 	afCloseFile(file);
+ 	unlink(testFileName);
+	free(testFileName);
+ 
+ 	exit(EXIT_SUCCESS);
+ }
+--- a/test/testchannelmatrix.c
+++ b/test/testchannelmatrix.c
+@@ -39,7 +39,7 @@
+ 
+ #include "TestUtilities.h"
+ 
+-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
+ 
+ const short samples[] = {300, -300, 515, -515, 2315, -2315, 9154, -9154};
+ #define SAMPLE_COUNT (sizeof (samples) / sizeof (short))
+@@ -47,7 +47,11 @@ const short samples[] = {300, -300, 515,
+ 
+ void cleanup (void)
+ {
+-	unlink(sTestFileName);
+	if (sTestFileName)
+	{
+		unlink(sTestFileName);
+		free(sTestFileName);
+	}
+ }
+ 
+ void ensure (int condition, const char *message)
+@@ -76,7 +80,7 @@ int main (void)
+ 	afInitFileFormat(setup, AF_FILE_AIFFC);
+ 
+ 	/* Write stereo data to test file. */
+-	ensure(createTemporaryFile("testchannelmatrix", sTestFileName),
+	ensure(createTemporaryFile("testchannelmatrix", &sTestFileName),
+ 		"could not create temporary file");
+ 	file = afOpenFile(sTestFileName, "w", setup);
+ 	ensure(file != AF_NULL_FILEHANDLE, "could not open file for writing");
+--- a/test/testdouble.c
+++ b/test/testdouble.c
+@@ -38,7 +38,7 @@
+ 
+ #include "TestUtilities.h"
+ 
+-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
+ 
+ const double samples[] =
+ 	{1.0, 0.6, -0.3, 0.95, 0.2, -0.6, 0.9, 0.4, -0.22, 0.125, 0.1, -0.4};
+@@ -48,7 +48,11 @@ void testdouble (int fileFormat);
+ 
+ void cleanup (void)
+ {
+-	unlink(sTestFileName);
+	if (sTestFileName)
+	{
+		unlink(sTestFileName);
+		free(sTestFileName);
+	}
+ }
+ 
+ void ensure (int condition, const char *message)
+@@ -96,7 +100,7 @@ void testdouble (int fileFormat)
+ 	afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_DOUBLE, 64);
+ 	afInitChannels(setup, AF_DEFAULT_TRACK, 2);
+ 
+-	ensure(createTemporaryFile("testdouble", sTestFileName),
+	ensure(createTemporaryFile("testdouble", &sTestFileName),
+ 		"could not create temporary file");
+ 	file = afOpenFile(sTestFileName, "w", setup);
+ 	ensure(file != AF_NULL_FILEHANDLE, "could not open file for writing");
+--- a/test/testfloat.c
+++ b/test/testfloat.c
+@@ -38,7 +38,7 @@
+ 
+ #include "TestUtilities.h"
+ 
+-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
+ 
+ const float samples[] =
+ 	{1.0, 0.6, -0.3, 0.95, 0.2, -0.6, 0.9, 0.4, -0.22, 0.125, 0.1, -0.4};
+@@ -48,7 +48,11 @@ void testfloat (int fileFormat);
+ 
+ void cleanup (void)
+ {
+-	unlink(sTestFileName);
+	if (sTestFileName)
+	{
+		unlink(sTestFileName);
+		free(sTestFileName);
+	}
+ }
+ 
+ void ensure (int condition, const char *message)
+@@ -96,7 +100,7 @@ void testfloat (int fileFormat)
+ 	afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_FLOAT, 32);
+ 	afInitChannels(setup, AF_DEFAULT_TRACK, 2);
+ 
+-	ensure(createTemporaryFile("testfloat", sTestFileName),
+	ensure(createTemporaryFile("testfloat", &sTestFileName),
+ 		"could not create temporary file");
+ 	file = afOpenFile(sTestFileName, "w", setup);
+ 	ensure(file != AF_NULL_FILEHANDLE, "could not open file for writing");
+--- a/test/testmarkers.c
+++ b/test/testmarkers.c
+@@ -32,15 +32,19 @@
+ 
+ #include "TestUtilities.h"
+ 
+-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
+ 
+ #define FRAME_COUNT 200
+ 
+ void cleanup (void)
+ {
+	if (sTestFileName)
+	{
+ #ifndef DEBUG
+-	unlink(sTestFileName);
+		unlink(sTestFileName);
+ #endif
+		free(sTestFileName);
+	}
+ }
+ 
+ void ensure (int condition, const char *message)
+@@ -127,7 +131,7 @@ int testmarkers (int fileformat)
+ 
+ int main (void)
+ {
+-	ensure(createTemporaryFile("testmarkers", sTestFileName),
+	ensure(createTemporaryFile("testmarkers", &sTestFileName),
+ 		"could not create temporary file");
+ 
+ 	testmarkers(AF_FILE_AIFF);
+--- a/test/twentyfour.c
+++ b/test/twentyfour.c
+@@ -71,8 +71,8 @@ int main (int argc, char **argv)
+ 	afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 24);
+ 	afInitChannels(setup, AF_DEFAULT_TRACK, 1);
+ 
+-	char testFileName[PATH_MAX];
+-	if (!createTemporaryFile("twentyfour", testFileName))
+	char *testFileName;
+	if (!createTemporaryFile("twentyfour", &testFileName))
+ 	{
+ 		fprintf(stderr, "could not create temporary file\n");
+ 		exit(EXIT_FAILURE);
+@@ -239,6 +239,7 @@ int main (int argc, char **argv)
+ 		exit(EXIT_FAILURE);
+ 	}
+ 	unlink(testFileName);
+	free(testFileName);
+ 
+ 	exit(EXIT_SUCCESS);
+ }
+--- a/test/twentyfour2.c
+++ b/test/twentyfour2.c
+@@ -45,15 +45,19 @@
+ 
+ #include "TestUtilities.h"
+ 
+-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
+ 
+ #define FRAME_COUNT 10000
+ 
+ void cleanup (void)
+ {
+	if (sTestFileName)
+	{
+ #ifndef DEBUG
+-	unlink(sTestFileName);
+		unlink(sTestFileName);
+ #endif
+		free(sTestFileName);
+	}
+ }
+ 
+ void ensure (int condition, const char *message)
+@@ -78,7 +82,7 @@ int main (void)
+ 	afInitChannels(setup, AF_DEFAULT_TRACK, 1);
+ 	afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 24);
+ 
+-	ensure(createTemporaryFile("twentyfour2", sTestFileName),
+	ensure(createTemporaryFile("twentyfour2", &sTestFileName),
+ 		"could not create temporary file");
+ 	file = afOpenFile(sTestFileName, "w", setup);
+ 	ensure(file != NULL, "could not open test file for writing");
+--- a/test/writealaw.c
+++ b/test/writealaw.c
+@@ -53,7 +53,7 @@
+ 
+ #include "TestUtilities.h"
+ 
+-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
+ 
+ #define FRAME_COUNT 16
+ #define SAMPLE_COUNT FRAME_COUNT
+@@ -62,9 +62,13 @@ void testalaw (int fileFormat);
+ 
+ void cleanup (void)
+ {
+	if (sTestFileName)
+	{
+ #ifndef DEBUG
+-	unlink(sTestFileName);
+		unlink(sTestFileName);
+ #endif
+		free(sTestFileName);
+	}
+ }
+ 
+ void ensure (int condition, const char *message)
+@@ -113,7 +117,7 @@ void testalaw (int fileFormat)
+ 	afInitFileFormat(setup, fileFormat);
+ 	afInitChannels(setup, AF_DEFAULT_TRACK, 1);
+ 
+-	ensure(createTemporaryFile("writealaw", sTestFileName),
+	ensure(createTemporaryFile("writealaw", &sTestFileName),
+ 		"could not create temporary file");
+ 	file = afOpenFile(sTestFileName, "w", setup);
+ 	afFreeFileSetup(setup);
+--- a/test/writeraw.c
+++ b/test/writeraw.c
+@@ -44,13 +44,17 @@
+ 
+ #include "TestUtilities.h"
+ 
+-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
+ 
+ void cleanup (void)
+ {
+	if (sTestFileName)
+	{
+ #ifndef DEBUG
+-	unlink(sTestFileName);
+		unlink(sTestFileName);
+ #endif
+		free(sTestFileName);
+	}
+ }
+ 
+ void ensure (int condition, const char *message)
+@@ -84,7 +88,7 @@ int main (int argc, char **argv)
+ 	afInitChannels(setup, AF_DEFAULT_TRACK, 1);
+ 	afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
+ 
+-	ensure(createTemporaryFile("writeraw", sTestFileName),
+	ensure(createTemporaryFile("writeraw", &sTestFileName),
+ 		"could not create temporary file");
+ 	file = afOpenFile(sTestFileName, "w", setup);
+ 	ensure(file != AF_NULL_FILEHANDLE, "unable to open file for writing");
+--- a/test/writeulaw.c
+++ b/test/writeulaw.c
+@@ -53,7 +53,7 @@
+ 
+ #include "TestUtilities.h"
+ 
+-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
+ 
+ #define FRAME_COUNT 16
+ #define SAMPLE_COUNT FRAME_COUNT
+@@ -62,9 +62,13 @@ void testulaw (int fileFormat);
+ 
+ void cleanup (void)
+ {
+	if (sTestFileName)
+	{
+ #ifndef DEBUG
+-	unlink(sTestFileName);
+		unlink(sTestFileName);
+ #endif
+		free(sTestFileName);
+	}
+ }
+ 
+ void ensure (int condition, const char *message)
+@@ -113,7 +117,7 @@ void testulaw (int fileFormat)
+ 	afInitFileFormat(setup, fileFormat);
+ 	afInitChannels(setup, AF_DEFAULT_TRACK, 1);
+ 
+-	ensure(createTemporaryFile("writeulaw", sTestFileName),
+	ensure(createTemporaryFile("writeulaw", &sTestFileName),
+ 		"could not create temporary file");
+ 	file = afOpenFile(sTestFileName, "w", setup);
+ 	afFreeFileSetup(setup);
--- a/gnu/packages/patches/audiofile-signature-of-multiplyCheckOverflow.patch
+++ b/gnu/packages/patches/audiofile-signature-of-multiplyCheckOverflow.patch
@ -0,0 +1,35 @@
+From: Antonio Larrosa <larrosa@kde.org>
+Date: Fri, 10 Mar 2017 15:40:02 +0100
+Subject: Fix signature of multiplyCheckOverflow. It returns a bool, not an int
+
+---
+ libaudiofile/modules/MSADPCM.cpp | 2 +-
+ sfcommands/sfconvert.c           | 2 +-
+ 2 files changed, 2 insertions(+), 2 deletions(-)
+
+diff --git a/libaudiofile/modules/MSADPCM.cpp b/libaudiofile/modules/MSADPCM.cpp
+index ef9c38c..d8c9553 100644
+--- a/libaudiofile/modules/MSADPCM.cpp
+++ b/libaudiofile/modules/MSADPCM.cpp
+@@ -116,7 +116,7 @@ int firstBitSet(int x)
+ #define __has_builtin(x) 0
+ #endif
+ 
+-int multiplyCheckOverflow(int a, int b, int *result)
+bool multiplyCheckOverflow(int a, int b, int *result)
+ {
+ #if (defined __GNUC__ && __GNUC__ >= 5) || ( __clang__ && __has_builtin(__builtin_mul_overflow))
+ 	return __builtin_mul_overflow(a, b, result);
+diff --git a/sfcommands/sfconvert.c b/sfcommands/sfconvert.c
+index 970a3e4..367f7a5 100644
+--- a/sfcommands/sfconvert.c
+++ b/sfcommands/sfconvert.c
+@@ -60,7 +60,7 @@ int firstBitSet(int x)
+ #define __has_builtin(x) 0
+ #endif
+ 
+-int multiplyCheckOverflow(int a, int b, int *result)
+bool multiplyCheckOverflow(int a, int b, int *result)
+ {
+ #if (defined __GNUC__ && __GNUC__ >= 5) || ( __clang__ && __has_builtin(__builtin_mul_overflow))
+ 	return __builtin_mul_overflow(a, b, result);