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audiofile: add security patch for CVE-2017-6839

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Integer overflow in modules/MSADPCM.cpp in Audio File Library (aka
audiofile) 0.3.6 allows remote attackers to cause a denial of service
(crash) via a crafted file.

https://blogs.gentoo.org/ago/2017/02/20/audiofile-multiple-ubsan-crashes/
https://github.com/mpruett/audiofile/issues/41

Signed-off-by: Peter Korsgaard <peter@korsgaard.com>
This commit is contained in:
Peter Korsgaard 2017-03-30 23:03:35 +02:00
parent bd5f84d301
commit 844a7c6281
1 changed files with 122 additions and 0 deletions
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122
package/audiofile/0007-Check-for-multiplication-overflow-in-MSADPCM-decodeS.patch Normal file
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@ -0,0 +1,122 @@
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
Signed-off-by: Peter Korsgaard <peter@korsgaard.com>
---
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