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media-libs/audiofile: Add security patches

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Dropping the system-gtest patch is necessary to make the tests run, as
mentioned here: https://bugs.gentoo.org/680482#c8

The three closed bugs are reported test failures fixed by dropping the
aforementioned patch and a slight repair of src_test. Because we're not
using system gtest anymore, we can drop the test dependency on
dev-cpp/gtest, and by extension the IUSE=test boilerplate.

Bug: https://bugs.gentoo.org/614046
Bug: https://bugs.gentoo.org/687766
Closes: https://bugs.gentoo.org/680482
Closes: https://bugs.gentoo.org/715192
Closes: https://bugs.gentoo.org/720836
Package-Manager: Portage-2.3.100, Repoman-2.3.22
Signed-off-by: John Helmert III <jchelmert3@posteo.net>
Closes: https://github.com/gentoo/gentoo/pull/16141
Signed-off-by: Sam James <sam@gentoo.org>
This commit is contained in:
John Helmert III
2020-07-19 18:28:17 +00:00
committed by Sam James
parent 3c8aa03578
commit f2bb2dc35e
3 changed files with 516 additions and 0 deletions
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media-libs/audiofile/audiofile-0.3.6-r4.ebuild
+55
View File
@@ -0,0 +1,55 @@
# Copyright 1999-2020 Gentoo Authors
# Distributed under the terms of the GNU General Public License v2
EAPI=6
inherit autotools gnome.org multilib-minimal
DESCRIPTION="An elegant API for accessing audio files"
HOMEPAGE="http://www.68k.org/~michael/audiofile/"
LICENSE="GPL-2 LGPL-2.1"
SLOT="0/1" # subslot = soname major version
KEYWORDS="~alpha ~amd64 ~arm ~arm64 ~hppa ~ia64 ~mips ~ppc ~ppc64 ~sparc ~x86 ~amd64-linux ~x86-linux ~ppc-macos ~x64-macos ~x86-macos ~sparc-solaris ~x86-solaris"
IUSE="flac"
RDEPEND="flac? ( >=media-libs/flac-1.2.1[${MULTILIB_USEDEP}] )"
DEPEND="${RDEPEND}
virtual/pkgconfig"
PATCHES=(
"${FILESDIR}"/${PN}-0.3.6-gcc6-build-fixes.patch
"${FILESDIR}"/${PN}-0.3.6-CVE-2015-7747.patch
"${FILESDIR}"/${PN}-0.3.6-mingw32.patch
"${FILESDIR}"/${PN}-0.3.6-CVE-2017-68xx.patch
"${FILESDIR}"/${PN}-0.3.6-CVE-2018-13440-CVE-2018-17095.patch
)
src_prepare() {
default
eautoreconf
}
multilib_src_configure() {
# Tests depend on statically compiled binaries to work, so we'll have to
# delete them later rather than not compile them at all
local myconf=(
--enable-largefile
--disable-werror
--disable-examples
$(use_enable flac)
)
ECONF_SOURCE="${S}" econf "${myconf[@]}"
}
multilib_src_test() {
emake check
}
multilib_src_install_all() {
einstalldocs
# package provides .pc file
find "${ED}" -name '*.la' -delete || die
find "${ED}" -name '*.a' -delete || die
}
media-libs/audiofile/files/audiofile-0.3.6-CVE-2017-68xx.patch
+379
View File
@@ -0,0 +1,379 @@
Debian patchset for CVE-2017-68{29..38} and two other vulnerabilities:
https://salsa.debian.org/multimedia-team/audiofile/commit/242f019#a064ca928f514268d4bae308e2e3990138341b76:
* Address several vulnerabilities (Closes: #857651)
- Always check the number of coefficients (CVE-2017-6827 CVE-2017-6828
CVE-2017-6832 CVE-2017-6833 CVE-2017-6835 CVE-2017-6837)
- clamp index values to fix index overflow in IMA.cpp (CVE-2017-6829)
- Check for multiplication overflow in sfconvert (CVE-2017-6830
CVE-2017-6834 CVE-2017-6836 CVE-2017-6838)
- Actually fail when error occurs in parseFormat (CVE-2017-6831)
- Check for multiplication overflow in MSADPCM decodeSample
(CVE-2017-6839)
* Fix signature of multiplyCheckOverflow. It returns a bool, not an int
* Check for division by zero in BlockCodec::runPull
From a2e9eab8ea87c4ffc494d839ebb4ea145eb9f2e6 Mon Sep 17 00:00:00 2001
From: Antonio Larrosa <larrosa@kde.org>
Date: Mon, 6 Mar 2017 18:59:26 +0100
Subject: [PATCH] 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 0e81cf7..d762249 100644
--- a/libaudiofile/WAVE.cpp
+++ b/libaudiofile/WAVE.cpp
@@ -326,6 +326,7 @@ 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;
@@ -333,6 +334,7 @@ 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;
--
2.11.0
From c48e4c6503f7dabd41f11d4c9c7b7f8960e7f2c0 Mon Sep 17 00:00:00 2001
From: Antonio Larrosa <larrosa@kde.org>
Date: Mon, 6 Mar 2017 12:51:22 +0100
Subject: [PATCH] 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 0e81cf7..61f9541 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;
--
2.11.0
From: Antonio Larrosa <larrosa@kde.org>
Date: Thu, 9 Mar 2017 10:21:18 +0100
Subject: Check for division by zero in BlockCodec::runPull
---
libaudiofile/modules/BlockCodec.cpp | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/libaudiofile/modules/BlockCodec.cpp b/libaudiofile/modules/BlockCodec.cpp
index 4731be1..eb2fb4d 100644
--- a/libaudiofile/modules/BlockCodec.cpp
+++ b/libaudiofile/modules/BlockCodec.cpp
@@ -47,7 +47,7 @@ void BlockCodec::runPull()
// Read the compressed data.
ssize_t bytesRead = read(m_inChunk->buffer, m_bytesPerPacket * blockCount);
- int blocksRead = bytesRead >= 0 ? bytesRead / m_bytesPerPacket : 0;
+ int blocksRead = (bytesRead >= 0 && m_bytesPerPacket > 0) ? bytesRead / m_bytesPerPacket : 0;
// Decompress into m_outChunk.
for (int i=0; i<blocksRead; i++)
From beacc44eb8cdf6d58717ec1a5103c5141f1b37f9 Mon Sep 17 00:00:00 2001
From: Antonio Larrosa <larrosa@kde.org>
Date: Mon, 6 Mar 2017 13:43:53 +0100
Subject: [PATCH] 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++;
--
2.11.0
From 7d65f89defb092b63bcbc5d98349fb222ca73b3c Mon Sep 17 00:00:00 2001
From: Antonio Larrosa <larrosa@kde.org>
Date: Mon, 6 Mar 2017 13:54:52 +0100
Subject: [PATCH] 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;
--
2.11.0
From 25eb00ce913452c2e614548d7df93070bf0d066f Mon Sep 17 00:00:00 2001
From: Antonio Larrosa <larrosa@kde.org>
Date: Mon, 6 Mar 2017 18:02:31 +0100
Subject: [PATCH] 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)
--
2.11.0
From ce536d707b8e2a26baca77320398c45238224ca7 Mon Sep 17 00:00:00 2001
From: Antonio Larrosa <larrosa@kde.org>
Date: Fri, 10 Mar 2017 15:40:02 +0100
Subject: [PATCH] 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);
--
2.11.0
media-libs/audiofile/files/audiofile-0.3.6-CVE-2018-13440-CVE-2018-17095.patch
+82
View File
@@ -0,0 +1,82 @@
Fixes for CVE-2018-13440 and CVE-2018-17095 from here:
https://github.com/mpruett/audiofile/pull/52
These are the same used in Fedora.
From fde6d79fb8363c4a329a184ef0b107156602b225 Mon Sep 17 00:00:00 2001
From: Wim Taymans <wtaymans@redhat.com>
Date: Thu, 27 Sep 2018 10:48:45 +0200
Subject: [PATCH 1/3] 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);
From 941774c8c0e79007196d7f1e7afdc97689f869b3 Mon Sep 17 00:00:00 2001
From: Wim Taymans <wtaymans@redhat.com>
Date: Thu, 27 Sep 2018 12:09:45 +0200
Subject: [PATCH 2/3] ALAC: set chunk frameCount to 0 on short read
---
libaudiofile/modules/ALAC.cpp | 1 +
1 file changed, 1 insertion(+)
diff --git a/libaudiofile/modules/ALAC.cpp b/libaudiofile/modules/ALAC.cpp
index 7593c11..478e2af 100644
--- a/libaudiofile/modules/ALAC.cpp
+++ b/libaudiofile/modules/ALAC.cpp
@@ -240,6 +240,7 @@ void ALAC::runPull()
if (read(m_inChunk->buffer, bytesPerPacket) < bytesPerPacket)
{
reportReadError(0, m_track->f.framesPerPacket);
+ m_outChunk->frameCount = 0;
return;
}
From 822b732fd31ffcb78f6920001e9b1fbd815fa712 Mon Sep 17 00:00:00 2001
From: Wim Taymans <wtaymans@redhat.com>
Date: Thu, 27 Sep 2018 12:11:12 +0200
Subject: [PATCH 3/3] 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);
}