[arch-commits] Commit in audiofile/repos (14 files)

Evangelos Foutras foutrelis at archlinux.org
Sat May 16 10:19:36 UTC 2020


    Date: Saturday, May 16, 2020 @ 10:19:36
  Author: foutrelis
Revision: 384075

archrelease: copy trunk to staging-x86_64

Added:
  audiofile/repos/staging-x86_64/
  audiofile/repos/staging-x86_64/01_gcc6.patch
    (from rev 384073, audiofile/trunk/01_gcc6.patch)
  audiofile/repos/staging-x86_64/02_hurd.patch
    (from rev 384073, audiofile/trunk/02_hurd.patch)
  audiofile/repos/staging-x86_64/03_CVE-2015-7747.patch
    (from rev 384073, audiofile/trunk/03_CVE-2015-7747.patch)
  audiofile/repos/staging-x86_64/04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
    (from rev 384073, audiofile/trunk/04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch)
  audiofile/repos/staging-x86_64/05_Always-check-the-number-of-coefficients.patch
    (from rev 384073, audiofile/trunk/05_Always-check-the-number-of-coefficients.patch)
  audiofile/repos/staging-x86_64/06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch
    (from rev 384073, audiofile/trunk/06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch)
  audiofile/repos/staging-x86_64/07_Check-for-multiplication-overflow-in-sfconvert.patch
    (from rev 384073, audiofile/trunk/07_Check-for-multiplication-overflow-in-sfconvert.patch)
  audiofile/repos/staging-x86_64/08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch
    (from rev 384073, audiofile/trunk/08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch)
  audiofile/repos/staging-x86_64/09_Actually-fail-when-error-occurs-in-parseFormat.patch
    (from rev 384073, audiofile/trunk/09_Actually-fail-when-error-occurs-in-parseFormat.patch)
  audiofile/repos/staging-x86_64/10_Check-for-division-by-zero-in-BlockCodec-runPull.patch
    (from rev 384073, audiofile/trunk/10_Check-for-division-by-zero-in-BlockCodec-runPull.patch)
  audiofile/repos/staging-x86_64/11_CVE-2018-13440.patch
    (from rev 384073, audiofile/trunk/11_CVE-2018-13440.patch)
  audiofile/repos/staging-x86_64/12_CVE-2018-17095.patch
    (from rev 384073, audiofile/trunk/12_CVE-2018-17095.patch)
  audiofile/repos/staging-x86_64/PKGBUILD
    (from rev 384073, audiofile/trunk/PKGBUILD)

-----------------------------------------------------------------+
 01_gcc6.patch                                                   |  102 ++
 02_hurd.patch                                                   |  381 ++++++++++
 03_CVE-2015-7747.patch                                          |  156 ++++
 04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch    |   33 
 05_Always-check-the-number-of-coefficients.patch                |   30 
 06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch |  116 +++
 07_Check-for-multiplication-overflow-in-sfconvert.patch         |   66 +
 08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch |   35 
 09_Actually-fail-when-error-occurs-in-parseFormat.patch         |   36 
 10_Check-for-division-by-zero-in-BlockCodec-runPull.patch       |   21 
 11_CVE-2018-13440.patch                                         |   28 
 12_CVE-2018-17095.patch                                         |   26 
 PKGBUILD                                                        |   70 +
 13 files changed, 1100 insertions(+)

Copied: audiofile/repos/staging-x86_64/01_gcc6.patch (from rev 384073, audiofile/trunk/01_gcc6.patch)
===================================================================
--- staging-x86_64/01_gcc6.patch	                        (rev 0)
+++ staging-x86_64/01_gcc6.patch	2020-05-16 10:19:36 UTC (rev 384075)
@@ -0,0 +1,102 @@
+Description: Fix FTBFS with GCC 6
+Author: Michael Schwendt <mschwendt at fedoraproject.org>
+Origin: vendor, https://github.com/mpruett/audiofile/pull/27
+Bug-Debian: https://bugs.debian.org/812055
+---
+This patch header follows DEP-3: http://dep.debian.net/deps/dep3/
+
+--- a/libaudiofile/modules/SimpleModule.h
++++ b/libaudiofile/modules/SimpleModule.h
+@@ -123,7 +123,7 @@ 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 kMinSignedValue = 0-(1U<<kScaleBits);
+ 
+ 	struct signedToUnsigned : public std::unary_function<SignedType, UnsignedType>
+ 	{
+--- a/test/FloatToInt.cpp
++++ b/test/FloatToInt.cpp
+@@ -115,7 +115,7 @@ TEST_F(FloatToIntTest, Int16)
+ 		EXPECT_EQ(readData[i], expectedData[i]);
+ }
+ 
+-static const int32_t kMinInt24 = -1<<23;
++static const int32_t kMinInt24 = 0-(1U<<23);
+ static const int32_t kMaxInt24 = (1<<23) - 1;
+ 
+ TEST_F(FloatToIntTest, Int24)
+--- a/test/IntToFloat.cpp
++++ b/test/IntToFloat.cpp
+@@ -117,7 +117,7 @@ TEST_F(IntToFloatTest, Int16)
+ 		EXPECT_EQ(readData[i], expectedData[i]);
+ }
+ 
+-static const int32_t kMinInt24 = -1<<23;
++static const int32_t kMinInt24 = 0-(1U<<23);
+ static const int32_t kMaxInt24 = (1<<23) - 1;
+ 
+ TEST_F(IntToFloatTest, Int24)
+--- a/test/NeXT.cpp
++++ b/test/NeXT.cpp
+@@ -37,13 +37,13 @@
+ 
+ #include "TestUtilities.h"
+ 
+-const char kDataUnspecifiedLength[] =
++const signed char kDataUnspecifiedLength[] =
+ {
+ 	'.', 's', 'n', 'd',
+ 	0, 0, 0, 24, // offset of 24 bytes
+-	0xff, 0xff, 0xff, 0xff, // unspecified length
++	-1, -1, -1, -1, // unspecified length
+ 	0, 0, 0, 3, // 16-bit linear
+-	0, 0, 172, 68, // 44100 Hz
++	0, 0, -84, 68, // 44100 Hz (0xAC44)
+ 	0, 0, 0, 1, // 1 channel
+ 	0, 1,
+ 	0, 1,
+@@ -57,13 +57,13 @@ const char kDataUnspecifiedLength[] =
+ 	0, 55
+ };
+ 
+-const char kDataTruncated[] =
++const signed char kDataTruncated[] =
+ {
+ 	'.', 's', 'n', 'd',
+ 	0, 0, 0, 24, // offset of 24 bytes
+ 	0, 0, 0, 20, // length of 20 bytes
+ 	0, 0, 0, 3, // 16-bit linear
+-	0, 0, 172, 68, // 44100 Hz
++	0, 0, -84, 68, // 44100 Hz (0xAC44)
+ 	0, 0, 0, 1, // 1 channel
+ 	0, 1,
+ 	0, 1,
+@@ -152,13 +152,13 @@ TEST(NeXT, Truncated)
+ 	ASSERT_EQ(::unlink(testFileName.c_str()), 0);
+ }
+ 
+-const char kDataZeroChannels[] =
++const signed char kDataZeroChannels[] =
+ {
+ 	'.', 's', 'n', 'd',
+ 	0, 0, 0, 24, // offset of 24 bytes
+ 	0, 0, 0, 2, // 2 bytes
+ 	0, 0, 0, 3, // 16-bit linear
+-	0, 0, 172, 68, // 44100 Hz
++	0, 0, -84, 68, // 44100 Hz (0xAC44)
+ 	0, 0, 0, 0, // 0 channels
+ 	0, 1
+ };
+--- a/test/Sign.cpp
++++ b/test/Sign.cpp
+@@ -116,7 +116,7 @@ TEST_F(SignConversionTest, Int16)
+ 		EXPECT_EQ(readData[i], expectedData[i]);
+ }
+ 
+-static const int32_t kMinInt24 = -1<<23;
++static const int32_t kMinInt24 = 0-(1U<<23);
+ static const int32_t kMaxInt24 = (1<<23) - 1;
+ static const uint32_t kMaxUInt24 = (1<<24) - 1;
+ 

Copied: audiofile/repos/staging-x86_64/02_hurd.patch (from rev 384073, audiofile/trunk/02_hurd.patch)
===================================================================
--- staging-x86_64/02_hurd.patch	                        (rev 0)
+++ staging-x86_64/02_hurd.patch	2020-05-16 10:19:36 UTC (rev 384075)
@@ -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 at 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);

Copied: audiofile/repos/staging-x86_64/03_CVE-2015-7747.patch (from rev 384073, audiofile/trunk/03_CVE-2015-7747.patch)
===================================================================
--- staging-x86_64/03_CVE-2015-7747.patch	                        (rev 0)
+++ staging-x86_64/03_CVE-2015-7747.patch	2020-05-16 10:19:36 UTC (rev 384075)
@@ -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);
++}

Copied: audiofile/repos/staging-x86_64/04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch (from rev 384073, audiofile/trunk/04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch)
===================================================================
--- staging-x86_64/04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch	                        (rev 0)
+++ staging-x86_64/04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch	2020-05-16 10:19:36 UTC (rev 384075)
@@ -0,0 +1,33 @@
+From: Antonio Larrosa <larrosa at 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)

Copied: audiofile/repos/staging-x86_64/05_Always-check-the-number-of-coefficients.patch (from rev 384073, audiofile/trunk/05_Always-check-the-number-of-coefficients.patch)
===================================================================
--- staging-x86_64/05_Always-check-the-number-of-coefficients.patch	                        (rev 0)
+++ staging-x86_64/05_Always-check-the-number-of-coefficients.patch	2020-05-16 10:19:36 UTC (rev 384075)
@@ -0,0 +1,30 @@
+From: Antonio Larrosa <larrosa at 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;
+ 

Copied: audiofile/repos/staging-x86_64/06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch (from rev 384073, audiofile/trunk/06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch)
===================================================================
--- staging-x86_64/06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch	                        (rev 0)
+++ staging-x86_64/06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch	2020-05-16 10:19:36 UTC (rev 384075)
@@ -0,0 +1,116 @@
+From: Antonio Larrosa <larrosa at 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++;

Copied: audiofile/repos/staging-x86_64/07_Check-for-multiplication-overflow-in-sfconvert.patch (from rev 384073, audiofile/trunk/07_Check-for-multiplication-overflow-in-sfconvert.patch)
===================================================================
--- staging-x86_64/07_Check-for-multiplication-overflow-in-sfconvert.patch	                        (rev 0)
+++ staging-x86_64/07_Check-for-multiplication-overflow-in-sfconvert.patch	2020-05-16 10:19:36 UTC (rev 384075)
@@ -0,0 +1,66 @@
+From: Antonio Larrosa <larrosa at 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;

Copied: audiofile/repos/staging-x86_64/08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch (from rev 384073, audiofile/trunk/08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch)
===================================================================
--- staging-x86_64/08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch	                        (rev 0)
+++ staging-x86_64/08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch	2020-05-16 10:19:36 UTC (rev 384075)
@@ -0,0 +1,35 @@
+From: Antonio Larrosa <larrosa at 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);

Copied: audiofile/repos/staging-x86_64/09_Actually-fail-when-error-occurs-in-parseFormat.patch (from rev 384073, audiofile/trunk/09_Actually-fail-when-error-occurs-in-parseFormat.patch)
===================================================================
--- staging-x86_64/09_Actually-fail-when-error-occurs-in-parseFormat.patch	                        (rev 0)
+++ staging-x86_64/09_Actually-fail-when-error-occurs-in-parseFormat.patch	2020-05-16 10:19:36 UTC (rev 384075)
@@ -0,0 +1,36 @@
+From: Antonio Larrosa <larrosa at 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;

Copied: audiofile/repos/staging-x86_64/10_Check-for-division-by-zero-in-BlockCodec-runPull.patch (from rev 384073, audiofile/trunk/10_Check-for-division-by-zero-in-BlockCodec-runPull.patch)
===================================================================
--- staging-x86_64/10_Check-for-division-by-zero-in-BlockCodec-runPull.patch	                        (rev 0)
+++ staging-x86_64/10_Check-for-division-by-zero-in-BlockCodec-runPull.patch	2020-05-16 10:19:36 UTC (rev 384075)
@@ -0,0 +1,21 @@
+From: Antonio Larrosa <larrosa at 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++)

Copied: audiofile/repos/staging-x86_64/11_CVE-2018-13440.patch (from rev 384073, audiofile/trunk/11_CVE-2018-13440.patch)
===================================================================
--- staging-x86_64/11_CVE-2018-13440.patch	                        (rev 0)
+++ staging-x86_64/11_CVE-2018-13440.patch	2020-05-16 10:19:36 UTC (rev 384075)
@@ -0,0 +1,28 @@
+From fde6d79fb8363c4a329a184ef0b107156602b225 Mon Sep 17 00:00:00 2001
+From: Wim Taymans <wtaymans at 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);

Copied: audiofile/repos/staging-x86_64/12_CVE-2018-17095.patch (from rev 384073, audiofile/trunk/12_CVE-2018-17095.patch)
===================================================================
--- staging-x86_64/12_CVE-2018-17095.patch	                        (rev 0)
+++ staging-x86_64/12_CVE-2018-17095.patch	2020-05-16 10:19:36 UTC (rev 384075)
@@ -0,0 +1,26 @@
+From 822b732fd31ffcb78f6920001e9b1fbd815fa712 Mon Sep 17 00:00:00 2001
+From: Wim Taymans <wtaymans at 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);
+ }
+ 

Copied: audiofile/repos/staging-x86_64/PKGBUILD (from rev 384073, audiofile/trunk/PKGBUILD)
===================================================================
--- staging-x86_64/PKGBUILD	                        (rev 0)
+++ staging-x86_64/PKGBUILD	2020-05-16 10:19:36 UTC (rev 384075)
@@ -0,0 +1,70 @@
+# Maintainer: David Runge <dave at sleepmap.de>
+# Contributor: Ray Rashif <schiv at archlinux.org>
+# Contributor: dorphell <dorphell at archlinux.org>
+
+pkgname=audiofile
+pkgver=0.3.6
+pkgrel=6
+pkgdesc="Silicon Graphics Audio File Library"
+arch=('x86_64')
+url="https://audiofile.68k.org/"
+license=('GPL2' 'LGPL2.1')
+depends=('gcc-libs' 'alsa-lib' 'flac')
+provides=('libaudiofile.so')
+source=("https://audiofile.68k.org/$pkgname-$pkgver.tar.gz"
+        01_gcc6.patch
+        02_hurd.patch
+        03_CVE-2015-7747.patch
+        04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
+        05_Always-check-the-number-of-coefficients.patch
+        06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch
+        07_Check-for-multiplication-overflow-in-sfconvert.patch
+        08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch
+        09_Actually-fail-when-error-occurs-in-parseFormat.patch
+        10_Check-for-division-by-zero-in-BlockCodec-runPull.patch
+        11_CVE-2018-13440.patch
+        12_CVE-2018-17095.patch)
+sha512sums=('f9a1182d93e405c21eba79c5cc40962347bff13f1b3b732d9a396e3d1675297515188bd6eb43033aaa00e9bde74ff4628c1614462456529cabba464f03c1d5fa'
+            'ae11735970eaddb664251614743cb46ae029b4073f4f8ea7cd4570d50c0f4b7f7b426399901b011d1ea799bb99d4ac648e76be97f13a51e32d7a63f97b38a89f'
+            '76ce5a29beaa394f3a24e7db7c40864f26119857e78087b6780853d06d4f44e80656c418b2c99d95224d29b69c23c51c54a4c8edac5dbaa4038a9d6c1ef7be06'
+            '7673ab3fafdb0dac514a42622f53ea17aa56836c76413e5680c475537e195c53df21f26da1bd4e7941df2dc8b33a471ab52d539dabffbaef8bc95ee59951e7fe'
+            'e7afe1a27566fb593ea53176256df23e447a2ee842cb4168930dec365fdabe7f2f43512d81bca5f14336ef0c756f6006c24948a3c2d79baafb0042ed8a145aae'
+            '187fb02a0d23390a62507756918c6f0b149570d7361bfe18944ea182adb966bb2bece93ed25eb6b38b61e252347cb68372c39ea948e094be7afea126d38115c0'
+            '2a81cd1e87976b0123de0638fe4a20a644bc3292f938def3f1de205296f86c0dc7dfbb78a7c8d75c9b9e771c2dc96708f45d9766cf25be2a11bac61285e7de7f'
+            '65e46f7c7e5c994d98e15ed6e94b9512650cf30d4a7fb213f27a177e38defdb0575faa74712d2ef1c3541db069f98b10f7f365ebb01304a0bcdc92552114d701'
+            '7c81e9dda0fc996a0c7a32da3f7480ddcb5cb30b1fd08c36d485021d699ab886732430271ac5a458c1d43dfb11fd0e97a4a9d7608c7f414eb23de59384b81a80'
+            '51c92ce66e987ae1d4bda65247134097705ef45cf7670401af7943bf6bbfc674089bcfafa49983046b10573ea72900adb96c296739c234d5e98539098eebe022'
+            '234b0b520eebccc8e7782735615ad8fb2f7c03937da2b7dec0b091ca35b8a542d4e5c7ad22ed6715f019cdb36992838d7458ef58980bfb4fa80062e764d18ae2'
+            'e29ab46b2edcbbeb048a7d9e6210d0faac8b75d9a48a663f62b37881e03d34fa97ffaa05d61da53b49404f60f0cadfcbbbb58438ae82af40dd37d0117bf8c631'
+            'ace83995606f900543f65ce6199fe1a69c757b7b37e92561be1c49c2f827676f888e36132ab3fedf3b9f77d4382ea933480fe326859c092aa95ba2c24e777363')
+
+prepare() {
+  cd "$pkgname-$pkgver"
+  local filename
+  for filename in "${source[@]}"; do
+    if [[ "$filename" =~ \.patch$ ]]; then
+      echo "Applying patch ${filename##*/}"
+      patch -p1 -N -i "$srcdir/${filename##*/}"
+    fi
+  done
+  autoreconf -vfi
+}
+
+build() {
+  cd "$pkgname-$pkgver"
+  ./configure --prefix=/usr
+  make
+}
+
+check() {
+  cd "$pkgname-$pkgver"
+  make -k check
+}
+
+package() {
+  cd "$pkgname-$pkgver"
+  make DESTDIR="$pkgdir" install
+  install -vDm 644 {AUTHORS,ChangeLog,NEWS,NOTES,README,TODO} \
+    -t "${pkgdir}/usr/share/doc/${pkgname}"
+}
+# vim:set ts=2 sw=2 et:


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