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.

15 files changed, 1070 insertions, 0 deletions

diff --git a/gnu/local.mk b/gnu/local.mk
index 2f1b3c4baf..e4eaa4e848 100644
--- 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		\
diff --git a/gnu/packages/audio.scm b/gnu/packages/audio.scm
index 87d6947657..76c59f33cc 100644
--- 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 @@ and editing digital audio.  It features digital effects and spectrum analysis
 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")
diff --git a/gnu/packages/patches/audiofile-CVE-2015-7747.patch b/gnu/packages/patches/audiofile-CVE-2015-7747.patch
new file mode 100644
index 0000000000..3325639591
--- /dev/null
+++ 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);
++}
diff --git a/gnu/packages/patches/audiofile-CVE-2018-13440.patch b/gnu/packages/patches/audiofile-CVE-2018-13440.patch
new file mode 100644
index 0000000000..ffd65b43b0
--- /dev/null
+++ 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);
diff --git a/gnu/packages/patches/audiofile-CVE-2018-17095.patch b/gnu/packages/patches/audiofile-CVE-2018-17095.patch
new file mode 100644
index 0000000000..231021b9fc
--- /dev/null
+++ 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);
+ }
+ 
diff --git a/gnu/packages/patches/audiofile-Check-the-number-of-coefficients.patch b/gnu/packages/patches/audiofile-Check-the-number-of-coefficients.patch
new file mode 100644
index 0000000000..f9427cbe61
--- /dev/null
+++ 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;
+ 
diff --git a/gnu/packages/patches/audiofile-Fail-on-error-in-parseFormat.patch b/gnu/packages/patches/audiofile-Fail-on-error-in-parseFormat.patch
new file mode 100644
index 0000000000..50cd3dc9a3
--- /dev/null
+++ 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;
diff --git a/gnu/packages/patches/audiofile-Fix-index-overflow-in-IMA.cpp.patch b/gnu/packages/patches/audiofile-Fix-index-overflow-in-IMA.cpp.patch
new file mode 100644
index 0000000000..c1047af06c
--- /dev/null
+++ 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)
diff --git a/gnu/packages/patches/audiofile-Fix-multiply-overflow-sfconvert.patch b/gnu/packages/patches/audiofile-Fix-multiply-overflow-sfconvert.patch
new file mode 100644
index 0000000000..0f17140d6b
--- /dev/null
+++ 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;
diff --git a/gnu/packages/patches/audiofile-Fix-overflow-in-MSADPCM-decodeSam.patch b/gnu/packages/patches/audiofile-Fix-overflow-in-MSADPCM-decodeSam.patch
new file mode 100644
index 0000000000..2be930b924
--- /dev/null
+++ 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++;
diff --git a/gnu/packages/patches/audiofile-division-by-zero-BlockCodec-runPull.patch b/gnu/packages/patches/audiofile-division-by-zero-BlockCodec-runPull.patch
new file mode 100644
index 0000000000..e001133916
--- /dev/null
+++ 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++)
diff --git a/gnu/packages/patches/audiofile-fix-datatypes-in-tests.patch b/gnu/packages/patches/audiofile-fix-datatypes-in-tests.patch
new file mode 100644
index 0000000000..00e0f3c4a3
--- /dev/null
+++ 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
diff --git a/gnu/packages/patches/audiofile-fix-sign-conversion.patch b/gnu/packages/patches/audiofile-fix-sign-conversion.patch
new file mode 100644
index 0000000000..648161d620
--- /dev/null
+++ 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>
+ 	{
diff --git a/gnu/packages/patches/audiofile-hurd.patch b/gnu/packages/patches/audiofile-hurd.patch
new file mode 100644
index 0000000000..b5941dcf44
--- /dev/null
+++ 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);
diff --git a/gnu/packages/patches/audiofile-signature-of-multiplyCheckOverflow.patch b/gnu/packages/patches/audiofile-signature-of-multiplyCheckOverflow.patch
new file mode 100644
index 0000000000..35627d3869
--- /dev/null
+++ 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);