Date: Monday, September 2, 2019 @ 22:01:16
Author: dvzrv
Revision: 361508
archrelease: copy trunk to extra-x86_64
Added:
audiofile/repos/extra-x86_64/01_gcc6.patch
(from rev 361507, audiofile/trunk/01_gcc6.patch)
audiofile/repos/extra-x86_64/02_hurd.patch
(from rev 361507, audiofile/trunk/02_hurd.patch)
audiofile/repos/extra-x86_64/03_CVE-2015-7747.patch
(from rev 361507, audiofile/trunk/03_CVE-2015-7747.patch)
audiofile/repos/extra-x86_64/04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
(from rev 361507,
audiofile/trunk/04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch)
audiofile/repos/extra-x86_64/05_Always-check-the-number-of-coefficients.patch
(from rev 361507,
audiofile/trunk/05_Always-check-the-number-of-coefficients.patch)
audiofile/repos/extra-x86_64/06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch
(from rev 361507,
audiofile/trunk/06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch)
audiofile/repos/extra-x86_64/07_Check-for-multiplication-overflow-in-sfconvert.patch
(from rev 361507,
audiofile/trunk/07_Check-for-multiplication-overflow-in-sfconvert.patch)
audiofile/repos/extra-x86_64/08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch
(from rev 361507,
audiofile/trunk/08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch)
audiofile/repos/extra-x86_64/09_Actually-fail-when-error-occurs-in-parseFormat.patch
(from rev 361507,
audiofile/trunk/09_Actually-fail-when-error-occurs-in-parseFormat.patch)
audiofile/repos/extra-x86_64/10_Check-for-division-by-zero-in-BlockCodec-runPull.patch
(from rev 361507,
audiofile/trunk/10_Check-for-division-by-zero-in-BlockCodec-runPull.patch)
audiofile/repos/extra-x86_64/11_CVE-2018-13440.patch
(from rev 361507, audiofile/trunk/11_CVE-2018-13440.patch)
audiofile/repos/extra-x86_64/12_CVE-2018-17095.patch
(from rev 361507, audiofile/trunk/12_CVE-2018-17095.patch)
audiofile/repos/extra-x86_64/PKGBUILD
(from rev 361507, audiofile/trunk/PKGBUILD)
Deleted:
audiofile/repos/extra-x86_64/01_gcc6.patch
audiofile/repos/extra-x86_64/03_CVE-2015-7747.patch
audiofile/repos/extra-x86_64/04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
audiofile/repos/extra-x86_64/05_Always-check-the-number-of-coefficients.patch
audiofile/repos/extra-x86_64/06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch
audiofile/repos/extra-x86_64/07_Check-for-multiplication-overflow-in-sfconvert.patch
audiofile/repos/extra-x86_64/08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch
audiofile/repos/extra-x86_64/09_Actually-fail-when-error-occurs-in-parseFormat.patch
audiofile/repos/extra-x86_64/10_Check-for-division-by-zero-in-BlockCodec-runPull.patch
audiofile/repos/extra-x86_64/PKGBUILD
-----------------------------------------------------------------+
01_gcc6.patch | 204 ++---
02_hurd.patch | 381
++++++++++
03_CVE-2015-7747.patch | 312 ++++----
04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch | 66 -
05_Always-check-the-number-of-coefficients.patch | 60 -
06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch | 232 +++---
07_Check-for-multiplication-overflow-in-sfconvert.patch | 132 +--
08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch | 70 -
09_Actually-fail-when-error-occurs-in-parseFormat.patch | 72 -
10_Check-for-division-by-zero-in-BlockCodec-runPull.patch | 42 -
11_CVE-2018-13440.patch | 28
12_CVE-2018-17095.patch | 26
PKGBUILD | 130 +--
13 files changed, 1100 insertions(+), 655 deletions(-)
Deleted: 01_gcc6.patch
===================================================================
--- 01_gcc6.patch 2019-09-02 22:01:03 UTC (rev 361507)
+++ 01_gcc6.patch 2019-09-02 22:01:16 UTC (rev 361508)
@@ -1,102 +0,0 @@
-Description: Fix FTBFS with GCC 6
-Author: Michael Schwendt <mschwe...@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/extra-x86_64/01_gcc6.patch (from rev 361507,
audiofile/trunk/01_gcc6.patch)
===================================================================
--- 01_gcc6.patch (rev 0)
+++ 01_gcc6.patch 2019-09-02 22:01:16 UTC (rev 361508)
@@ -0,0 +1,102 @@
+Description: Fix FTBFS with GCC 6
+Author: Michael Schwendt <mschwe...@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/extra-x86_64/02_hurd.patch (from rev 361507,
audiofile/trunk/02_hurd.patch)
===================================================================
--- 02_hurd.patch (rev 0)
+++ 02_hurd.patch 2019-09-02 22:01:16 UTC (rev 361508)
@@ -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.p...@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);
Deleted: 03_CVE-2015-7747.patch
===================================================================
--- 03_CVE-2015-7747.patch 2019-09-02 22:01:03 UTC (rev 361507)
+++ 03_CVE-2015-7747.patch 2019-09-02 22:01:16 UTC (rev 361508)
@@ -1,156 +0,0 @@
-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/extra-x86_64/03_CVE-2015-7747.patch (from rev 361507,
audiofile/trunk/03_CVE-2015-7747.patch)
===================================================================
--- 03_CVE-2015-7747.patch (rev 0)
+++ 03_CVE-2015-7747.patch 2019-09-02 22:01:16 UTC (rev 361508)
@@ -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);
++}
Deleted: 04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
===================================================================
--- 04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
2019-09-02 22:01:03 UTC (rev 361507)
+++ 04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
2019-09-02 22:01:16 UTC (rev 361508)
@@ -1,33 +0,0 @@
-From: Antonio Larrosa <larr...@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/extra-x86_64/04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
(from rev 361507,
audiofile/trunk/04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch)
===================================================================
--- 04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
(rev 0)
+++ 04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
2019-09-02 22:01:16 UTC (rev 361508)
@@ -0,0 +1,33 @@
+From: Antonio Larrosa <larr...@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)
Deleted: 05_Always-check-the-number-of-coefficients.patch
===================================================================
--- 05_Always-check-the-number-of-coefficients.patch 2019-09-02 22:01:03 UTC
(rev 361507)
+++ 05_Always-check-the-number-of-coefficients.patch 2019-09-02 22:01:16 UTC
(rev 361508)
@@ -1,30 +0,0 @@
-From: Antonio Larrosa <larr...@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/extra-x86_64/05_Always-check-the-number-of-coefficients.patch
(from rev 361507,
audiofile/trunk/05_Always-check-the-number-of-coefficients.patch)
===================================================================
--- 05_Always-check-the-number-of-coefficients.patch
(rev 0)
+++ 05_Always-check-the-number-of-coefficients.patch 2019-09-02 22:01:16 UTC
(rev 361508)
@@ -0,0 +1,30 @@
+From: Antonio Larrosa <larr...@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;
+
Deleted: 06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch
===================================================================
--- 06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch
2019-09-02 22:01:03 UTC (rev 361507)
+++ 06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch
2019-09-02 22:01:16 UTC (rev 361508)
@@ -1,116 +0,0 @@
-From: Antonio Larrosa <larr...@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/extra-x86_64/06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch
(from rev 361507,
audiofile/trunk/06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch)
===================================================================
--- 06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch
(rev 0)
+++ 06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch
2019-09-02 22:01:16 UTC (rev 361508)
@@ -0,0 +1,116 @@
+From: Antonio Larrosa <larr...@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++;
Deleted: 07_Check-for-multiplication-overflow-in-sfconvert.patch
===================================================================
--- 07_Check-for-multiplication-overflow-in-sfconvert.patch 2019-09-02
22:01:03 UTC (rev 361507)
+++ 07_Check-for-multiplication-overflow-in-sfconvert.patch 2019-09-02
22:01:16 UTC (rev 361508)
@@ -1,66 +0,0 @@
-From: Antonio Larrosa <larr...@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/extra-x86_64/07_Check-for-multiplication-overflow-in-sfconvert.patch
(from rev 361507,
audiofile/trunk/07_Check-for-multiplication-overflow-in-sfconvert.patch)
===================================================================
--- 07_Check-for-multiplication-overflow-in-sfconvert.patch
(rev 0)
+++ 07_Check-for-multiplication-overflow-in-sfconvert.patch 2019-09-02
22:01:16 UTC (rev 361508)
@@ -0,0 +1,66 @@
+From: Antonio Larrosa <larr...@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;
Deleted: 08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch
===================================================================
--- 08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch
2019-09-02 22:01:03 UTC (rev 361507)
+++ 08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch
2019-09-02 22:01:16 UTC (rev 361508)
@@ -1,35 +0,0 @@
-From: Antonio Larrosa <larr...@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/extra-x86_64/08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch
(from rev 361507,
audiofile/trunk/08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch)
===================================================================
--- 08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch
(rev 0)
+++ 08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch
2019-09-02 22:01:16 UTC (rev 361508)
@@ -0,0 +1,35 @@
+From: Antonio Larrosa <larr...@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);
Deleted: 09_Actually-fail-when-error-occurs-in-parseFormat.patch
===================================================================
--- 09_Actually-fail-when-error-occurs-in-parseFormat.patch 2019-09-02
22:01:03 UTC (rev 361507)
+++ 09_Actually-fail-when-error-occurs-in-parseFormat.patch 2019-09-02
22:01:16 UTC (rev 361508)
@@ -1,36 +0,0 @@
-From: Antonio Larrosa <larr...@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/extra-x86_64/09_Actually-fail-when-error-occurs-in-parseFormat.patch
(from rev 361507,
audiofile/trunk/09_Actually-fail-when-error-occurs-in-parseFormat.patch)
===================================================================
--- 09_Actually-fail-when-error-occurs-in-parseFormat.patch
(rev 0)
+++ 09_Actually-fail-when-error-occurs-in-parseFormat.patch 2019-09-02
22:01:16 UTC (rev 361508)
@@ -0,0 +1,36 @@
+From: Antonio Larrosa <larr...@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;
Deleted: 10_Check-for-division-by-zero-in-BlockCodec-runPull.patch
===================================================================
--- 10_Check-for-division-by-zero-in-BlockCodec-runPull.patch 2019-09-02
22:01:03 UTC (rev 361507)
+++ 10_Check-for-division-by-zero-in-BlockCodec-runPull.patch 2019-09-02
22:01:16 UTC (rev 361508)
@@ -1,21 +0,0 @@
-From: Antonio Larrosa <larr...@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/extra-x86_64/10_Check-for-division-by-zero-in-BlockCodec-runPull.patch
(from rev 361507,
audiofile/trunk/10_Check-for-division-by-zero-in-BlockCodec-runPull.patch)
===================================================================
--- 10_Check-for-division-by-zero-in-BlockCodec-runPull.patch
(rev 0)
+++ 10_Check-for-division-by-zero-in-BlockCodec-runPull.patch 2019-09-02
22:01:16 UTC (rev 361508)
@@ -0,0 +1,21 @@
+From: Antonio Larrosa <larr...@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/extra-x86_64/11_CVE-2018-13440.patch (from rev 361507,
audiofile/trunk/11_CVE-2018-13440.patch)
===================================================================
--- 11_CVE-2018-13440.patch (rev 0)
+++ 11_CVE-2018-13440.patch 2019-09-02 22:01:16 UTC (rev 361508)
@@ -0,0 +1,28 @@
+From fde6d79fb8363c4a329a184ef0b107156602b225 Mon Sep 17 00:00:00 2001
+From: Wim Taymans <wtaym...@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/extra-x86_64/12_CVE-2018-17095.patch (from rev 361507,
audiofile/trunk/12_CVE-2018-17095.patch)
===================================================================
--- 12_CVE-2018-17095.patch (rev 0)
+++ 12_CVE-2018-17095.patch 2019-09-02 22:01:16 UTC (rev 361508)
@@ -0,0 +1,26 @@
+From 822b732fd31ffcb78f6920001e9b1fbd815fa712 Mon Sep 17 00:00:00 2001
+From: Wim Taymans <wtaym...@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);
+ }
+
Deleted: PKGBUILD
===================================================================
--- PKGBUILD 2019-09-02 22:01:03 UTC (rev 361507)
+++ PKGBUILD 2019-09-02 22:01:16 UTC (rev 361508)
@@ -1,60 +0,0 @@
-# $Id$
-# Maintainer: Ray Rashif <sc...@archlinux.org>
-# Contributor: dorphell <dorph...@archlinux.org>
-
-pkgname=audiofile
-pkgver=0.3.6
-pkgrel=4
-pkgdesc="Silicon Graphics Audio File Library"
-arch=('i686' 'x86_64')
-url="https://audiofile.68k.org/";
-license=('LGPL')
-depends=('gcc-libs' 'alsa-lib' 'flac')
-source=("https://audiofile.68k.org/$pkgname-$pkgver.tar.gz";
- 01_gcc6.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)
-sha256sums=('cdc60df19ab08bfe55344395739bb08f50fc15c92da3962fac334d3bff116965'
- 'a1904603c0292e76530f635dfc1828fb4e0d9d13555581cad33c0200640f7a27'
- 'bcfc180708d089b5abe0ae1439809b5a4306a08917b0212c3d135e5ec56711f2'
- '540c517828d5573ba7bc3fd9b3811f39f4ea0132011d348d22bdfc545e865a8e'
- '1b55abeb867d66b7d3b7c34585e77e6d3656c6317b582c99f3280d37523c7718'
- '7a464eb7521ae8deb67516309bb396caa93135dc62fbad7351e67923b1766423'
- '2ed5cc3b57394ea33ad466ca9844b766e4cb91dd7b1e2b71deaf15cf881dbf51'
- '257f157cf2cc8947e0f5be4bff2c4afddbe73643e9e39a83171dbea02f5d52f4'
- '48deaaa07bfade35208edb9e22b4fe78f91470012414ddb26cd68f684c95e33d'
- 'f31d51ebd8f8e0bd076cd1bce34b210c4dbbd959ca9b87693ad86a6399c492a3')
-
-prepare() {
- cd $pkgname-$pkgver
- patch -Np1 -i ../01_gcc6.patch
- patch -Np1 -i ../03_CVE-2015-7747.patch
- patch -Np1 -i ../04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
- patch -Np1 -i ../05_Always-check-the-number-of-coefficients.patch
- patch -Np1 -i
../06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch
- patch -Np1 -i ../07_Check-for-multiplication-overflow-in-sfconvert.patch
- patch -Np1 -i
../08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch
- patch -Np1 -i ../09_Actually-fail-when-error-occurs-in-parseFormat.patch
- patch -Np1 -i ../10_Check-for-division-by-zero-in-BlockCodec-runPull.patch
-}
-
-build() {
- cd "$srcdir/$pkgname-$pkgver"
-
- ./configure --prefix=/usr
- make
-}
-
-package() {
- cd "$srcdir/$pkgname-$pkgver"
-
- make DESTDIR="$pkgdir" install
-}
-
-# vim:set ts=2 sw=2 et:
Copied: audiofile/repos/extra-x86_64/PKGBUILD (from rev 361507,
audiofile/trunk/PKGBUILD)
===================================================================
--- PKGBUILD (rev 0)
+++ PKGBUILD 2019-09-02 22:01:16 UTC (rev 361508)
@@ -0,0 +1,70 @@
+# Maintainer: David Runge <d...@sleepmap.de>
+# Contributor: Ray Rashif <sc...@archlinux.org>
+# Contributor: dorphell <dorph...@archlinux.org>
+
+pkgname=audiofile
+pkgver=0.3.6
+pkgrel=5
+pkgdesc="Silicon Graphics Audio File Library"
+arch=('x86_64')
+url="https://audiofile.68k.org/";
+license=('GPL2' 'LGPL2.1')
+depends=('gcc-libs' 'alsa-lib' 'flac')
+provides=('libaudiofile.so')
+source=("https://audiofile.68k.org/$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:
