[patch] audiofile-0.3.6-cve-2015.patch

audiofile-0.3.6-cve-2015.patch (text/plain), 9.29 KB, created by John Helmert III on 2020-06-09 06:29:22 UTC

From 25eb00ce913452c2e614548d7df93070bf0d066f Mon Sep 17 00:00:00 2001
From: Antonio Larrosa <[e-mail]>
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)
From 7d65f89defb092b63bcbc5d98349fb222ca73b3c Mon Sep 17 00:00:00 2001
From: Antonio Larrosa <[e-mail]>
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;
From a2e9eab8ea87c4ffc494d839ebb4ea145eb9f2e6 Mon Sep 17 00:00:00 2001
From: Antonio Larrosa <[e-mail]>
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;
From beacc44eb8cdf6d58717ec1a5103c5141f1b37f9 Mon Sep 17 00:00:00 2001
From: Antonio Larrosa <[e-mail]>
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++;
From c48e4c6503f7dabd41f11d4c9c7b7f8960e7f2c0 Mon Sep 17 00:00:00 2001
From: Antonio Larrosa <[e-mail]>
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;