audiofile: patch CVE-2017-6839

Use patch from buildroot:
https://github.com/buildroot/buildroot/commit/844a7c6281eb442881330a5d36d5a0719f2870bf

Signed-off-by: Peter Marko <peter.marko@siemens.com>
Signed-off-by: Khem Raj <raj.khem@gmail.com>

2 files changed, 127 insertions, 0 deletions

diff --git a/meta-oe/recipes-multimedia/audiofile/audiofile_0.3.6.bb b/meta-oe/recipes-multimedia/audiofile/audiofile_0.3.6.bb
index 08709a35e3..50df31c7b9 100644
--- a/meta-oe/recipes-multimedia/audiofile/audiofile_0.3.6.bb
+++ b/meta-oe/recipes-multimedia/audiofile/audiofile_0.3.6.bb
@@ -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"
 
diff --git a/meta-oe/recipes-multimedia/audiofile/files/0008-Check-for-multiplication-overflow-in-MSADPCM-decodeS.patch b/meta-oe/recipes-multimedia/audiofile/files/0008-Check-for-multiplication-overflow-in-MSADPCM-decodeS.patch
new file mode 100644
index 0000000000..857ed78c59
--- /dev/null
+++ b/meta-oe/recipes-multimedia/audiofile/files/0008-Check-for-multiplication-overflow-in-MSADPCM-decodeS.patch
@@ -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
+