@@ -17,6 +17,7 @@ SRC_URI = " \
file://0005-clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch \
file://0006-Check-for-multiplication-overflow-in-sfconvert.patch \
file://0007-Actually-fail-when-error-occurs-in-parseFormat.patch \
+ file://0008-Check-for-multiplication-overflow-in-MSADPCM-decodeS.patch \
"
SRC_URI[sha256sum] = "ea2449ad3f201ec590d811db9da6d02ffc5e87a677d06b92ab15363d8cb59782"
new file mode 100644
@@ -0,0 +1,126 @@
+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>
+
+CVE: CVE-2017-6839
+Upstream-Status: Inactive-Upstream [lastrelease: 2013]
+Signed-off-by: Peter Marko <peter.marko@siemens.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
+