diff -Nur audiofile-0.3.6/libaudiofile/modules/BlockCodec.cpp audiofile-0.3.6-pull42/libaudiofile/modules/BlockCodec.cpp

--- audiofile-0.3.6/libaudiofile/modules/BlockCodec.cpp	2013-03-06 06:30:03.000000000 +0100

+++ audiofile-0.3.6-pull42/libaudiofile/modules/BlockCodec.cpp	2017-03-10 15:40:02.000000000 +0100

@@ -52,8 +52,9 @@

 	// Decompress into m_outChunk.

 	for (int i=0; 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 -Nur audiofile-0.3.6/libaudiofile/modules/MSADPCM.cpp audiofile-0.3.6-pull42/libaudiofile/modules/MSADPCM.cpp

--- audiofile-0.3.6/libaudiofile/modules/MSADPCM.cpp	2013-03-06 06:30:03.000000000 +0100

+++ audiofile-0.3.6-pull42/libaudiofile/modules/MSADPCM.cpp	2017-03-10 15:40:02.000000000 +0100

@@ -101,24 +101,60 @@

 	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

+

+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);

+#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 @@

 	{

 		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++;

diff -Nur audiofile-0.3.6/libaudiofile/WAVE.cpp audiofile-0.3.6-pull42/libaudiofile/WAVE.cpp

--- audiofile-0.3.6/libaudiofile/WAVE.cpp	2013-03-06 06:30:03.000000000 +0100

+++ audiofile-0.3.6-pull42/libaudiofile/WAVE.cpp	2017-03-10 15:40:02.000000000 +0100

@@ -281,6 +281,12 @@

 			/* 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;

@@ -834,6 +840,8 @@

 	}

 	TrackSetup *track = setup->getTrack();

+	if (!track)

+		return AF_NULL_FILESETUP;

 	if (track->f.isCompressed())

 	{

diff -Nur audiofile-0.3.6/sfcommands/sfconvert.c audiofile-0.3.6-pull42/sfcommands/sfconvert.c

--- audiofile-0.3.6/sfcommands/sfconvert.c	2013-03-06 06:30:03.000000000 +0100

+++ audiofile-0.3.6-pull42/sfcommands/sfconvert.c	2017-03-10 15:40:02.000000000 +0100

@@ -45,6 +45,33 @@

 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

+

+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);

+#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 @@

 {

 	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;