!19 [sync] PR-14: 去除编译选项rpath

From: @openeuler-sync-bot 
Reviewed-by: @dillon_chen 
Signed-off-by: @dillon_chen

audiofile-0.3.6-CVE-2015-7747.patch

@@ -1,12 +0,0 @@
-diff -Nurb --strip-trailing-cr audiofile-0.3.6-orig/libaudiofile/modules/ModuleState.cpp audiofile-0.3.6/libaudiofile/modules/ModuleState.cpp
---- audiofile-0.3.6-orig/libaudiofile/modules/ModuleState.cpp	2013-03-06 06:30:03.000000000 +0100
-+++ audiofile-0.3.6/libaudiofile/modules/ModuleState.cpp	2015-10-08 11:29:51.846082162 +0200
-@@ -402,7 +402,7 @@
- 		addModule(new Transform(outfc, in.pcm, out.pcm));
- 
- 	if (in.channelCount != out.channelCount)
--		addModule(new ApplyChannelMatrix(infc, isReading,
-+		addModule(new ApplyChannelMatrix(outfc, isReading,
- 			in.channelCount, out.channelCount,
- 			in.pcm.minClip, in.pcm.maxClip,
- 			track->channelMatrix));

audiofile-0.3.6-left-shift-neg.patch

@@ -1,48 +0,0 @@
-diff -Nurb --strip-trailing-cr audiofile-0.3.6-orig/libaudiofile/modules/SimpleModule.h audiofile-0.3.6/libaudiofile/modules/SimpleModule.h
---- audiofile-0.3.6-orig/libaudiofile/modules/SimpleModule.h	2013-03-06 06:30:03.000000000 +0100
-+++ audiofile-0.3.6/libaudiofile/modules/SimpleModule.h	2016-02-03 21:19:43.065454454 +0100
-@@ -123,7 +123,7 @@
- 	typedef typename IntTypes<Format>::UnsignedType UnsignedType;
- 
- 	static const int kScaleBits = (Format + 1) * CHAR_BIT - 1;
--	static const int kMinSignedValue = -1 << kScaleBits;
-+	static const int kMinSignedValue = 0-(1U<<kScaleBits);
- 
- 	struct signedToUnsigned : public std::unary_function<SignedType, UnsignedType>
- 	{
-diff -Nurb --strip-trailing-cr audiofile-0.3.6-orig/test/FloatToInt.cpp audiofile-0.3.6/test/FloatToInt.cpp
---- audiofile-0.3.6-orig/test/FloatToInt.cpp	2013-02-11 18:23:26.000000000 +0100
-+++ audiofile-0.3.6/test/FloatToInt.cpp	2016-02-03 21:21:14.714510229 +0100
-@@ -115,7 +115,7 @@
- 		EXPECT_EQ(readData[i], expectedData[i]);
- }
- 
--static const int32_t kMinInt24 = -1<<23;
-+static const int32_t kMinInt24 = 0-(1U<<23);
- static const int32_t kMaxInt24 = (1<<23) - 1;
- 
- TEST_F(FloatToIntTest, Int24)
-diff -Nurb --strip-trailing-cr audiofile-0.3.6-orig/test/IntToFloat.cpp audiofile-0.3.6/test/IntToFloat.cpp
---- audiofile-0.3.6-orig/test/IntToFloat.cpp	2013-02-11 18:23:26.000000000 +0100
-+++ audiofile-0.3.6/test/IntToFloat.cpp	2016-02-03 21:20:57.380445355 +0100
-@@ -117,7 +117,7 @@
- 		EXPECT_EQ(readData[i], expectedData[i]);
- }
- 
--static const int32_t kMinInt24 = -1<<23;
-+static const int32_t kMinInt24 = 0-(1U<<23);
- static const int32_t kMaxInt24 = (1<<23) - 1;
- 
- TEST_F(IntToFloatTest, Int24)
-diff -Nurb --strip-trailing-cr audiofile-0.3.6-orig/test/Sign.cpp audiofile-0.3.6/test/Sign.cpp
---- audiofile-0.3.6-orig/test/Sign.cpp	2013-02-11 18:23:26.000000000 +0100
-+++ audiofile-0.3.6/test/Sign.cpp	2016-02-03 21:20:38.742450826 +0100
-@@ -116,7 +116,7 @@
- 		EXPECT_EQ(readData[i], expectedData[i]);
- }
- 
--static const int32_t kMinInt24 = -1<<23;
-+static const int32_t kMinInt24 = 0-(1U<<23);
- static const int32_t kMaxInt24 = (1<<23) - 1;
- static const uint32_t kMaxUInt24 = (1<<24) - 1;
- 

audiofile-0.3.6-narrowing.patch

@@ -1,52 +0,0 @@
-diff -Nur audiofile-0.3.6-orig/test/NeXT.cpp audiofile-0.3.6/test/NeXT.cpp
---- audiofile-0.3.6-orig/test/NeXT.cpp	2013-02-11 18:23:26.000000000 +0100
-+++ audiofile-0.3.6/test/NeXT.cpp	2016-02-04 10:37:32.457140823 +0100
-@@ -37,13 +37,13 @@
- 
- #include "TestUtilities.h"
- 
--const char kDataUnspecifiedLength[] =
-+const signed char kDataUnspecifiedLength[] =
- {
- 	'.', 's', 'n', 'd',
- 	0, 0, 0, 24, // offset of 24 bytes
--	0xff, 0xff, 0xff, 0xff, // unspecified length
-+	-1, -1, -1, -1, // unspecified length
- 	0, 0, 0, 3, // 16-bit linear
--	0, 0, 172, 68, // 44100 Hz
-+	0, 0, -84, 68, // 44100 Hz (0xAC44)
- 	0, 0, 0, 1, // 1 channel
- 	0, 1,
- 	0, 1,
-@@ -57,13 +57,13 @@
- 	0, 55
- };
- 
--const char kDataTruncated[] =
-+const signed char kDataTruncated[] =
- {
- 	'.', 's', 'n', 'd',
- 	0, 0, 0, 24, // offset of 24 bytes
- 	0, 0, 0, 20, // length of 20 bytes
- 	0, 0, 0, 3, // 16-bit linear
--	0, 0, 172, 68, // 44100 Hz
-+	0, 0, -84, 68, // 44100 Hz (0xAC44)
- 	0, 0, 0, 1, // 1 channel
- 	0, 1,
- 	0, 1,
-@@ -152,13 +152,13 @@
- 	ASSERT_EQ(::unlink(testFileName.c_str()), 0);
- }
- 
--const char kDataZeroChannels[] =
-+const signed char kDataZeroChannels[] =
- {
- 	'.', 's', 'n', 'd',
- 	0, 0, 0, 24, // offset of 24 bytes
- 	0, 0, 0, 2, // 2 bytes
- 	0, 0, 0, 3, // 16-bit linear
--	0, 0, 172, 68, // 44100 Hz
-+	0, 0, -84, 68, // 44100 Hz (0xAC44)
- 	0, 0, 0, 0, // 0 channels
- 	0, 1
- };

audiofile-0.3.6-pull42.patch

@@ -1,176 +0,0 @@
-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<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 -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;

audiofile-0.3.6-pull43.patch

@@ -1,21 +0,0 @@
-diff -Nur audiofile-0.3.6/libaudiofile/modules/IMA.cpp audiofile-0.3.6-pull43/libaudiofile/modules/IMA.cpp
---- audiofile-0.3.6/libaudiofile/modules/IMA.cpp	2013-03-06 06:30:03.000000000 +0100
-+++ audiofile-0.3.6-pull43/libaudiofile/modules/IMA.cpp	2017-03-06 18:06:35.000000000 +0100
-@@ -169,7 +169,7 @@
- 		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 @@
- 			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)

audiofile-0.3.6-pull44.patch

@@ -1,31 +0,0 @@
-diff -Nur audiofile-0.3.6/libaudiofile/modules/BlockCodec.cpp audiofile-0.3.6-pull44/libaudiofile/modules/BlockCodec.cpp
---- audiofile-0.3.6/libaudiofile/modules/BlockCodec.cpp	2013-03-06 06:30:03.000000000 +0100
-+++ audiofile-0.3.6-pull44/libaudiofile/modules/BlockCodec.cpp	2017-03-09 10:21:18.000000000 +0100
-@@ -47,7 +47,7 @@
- 
- 	// Read the compressed data.
- 	ssize_t bytesRead = read(m_inChunk->buffer, m_bytesPerPacket * blockCount);
--	int blocksRead = bytesRead >= 0 ? bytesRead / m_bytesPerPacket : 0;
-+	int blocksRead = (bytesRead >= 0 && m_bytesPerPacket > 0) ? bytesRead / m_bytesPerPacket : 0;
- 
- 	// Decompress into m_outChunk.
- 	for (int i=0; i<blocksRead; i++)
-diff -Nur audiofile-0.3.6/libaudiofile/WAVE.cpp audiofile-0.3.6-pull44/libaudiofile/WAVE.cpp
---- audiofile-0.3.6/libaudiofile/WAVE.cpp	2013-03-06 06:30:03.000000000 +0100
-+++ audiofile-0.3.6-pull44/libaudiofile/WAVE.cpp	2017-03-09 10:21:18.000000000 +0100
-@@ -326,6 +326,7 @@
- 			{
- 				_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 @@
- 			{
- 				_af_error(AF_BAD_CODEC_CONFIG,
- 					"Invalid samples per block for IMA ADPCM compression");
-+				return AF_FAIL;
- 			}
- 
- 			track->f.sampleWidth = 16;

audiofile-CVE-2015-7747.patch

@@ -0,0 +1,13 @@
+diff --git a/libaudiofile/modules/ModuleState.cpp b/libaudiofile/modules/ModuleState.cpp
+index f76c495..0c29d7a 100644
+--- a/libaudiofile/modules/ModuleState.cpp
++++ b/libaudiofile/modules/ModuleState.cpp
+@@ -402,7 +402,7 @@ status ModuleState::arrange(AFfilehandle file, Track *track)
+ 		addModule(new Transform(outfc, in.pcm, out.pcm));
+ 
+ 	if (in.channelCount != out.channelCount)
+-		addModule(new ApplyChannelMatrix(infc, isReading,
++		addModule(new ApplyChannelMatrix(outfc, isReading,
+ 			in.channelCount, out.channelCount,
+ 			in.pcm.minClip, in.pcm.maxClip,
+ 			track->channelMatrix));

audiofile-fix-gcc6-compile-error.patch

@@ -0,0 +1,53 @@
+diff --git a/libaudiofile/modules/SimpleModule.h b/libaudiofile/modules/SimpleModule.h
+index 03c6c69..8e453a7 100644
+--- a/libaudiofile/modules/SimpleModule.h
++++ b/libaudiofile/modules/SimpleModule.h
+@@ -123,7 +123,7 @@ struct signConverter
+ 	typedef typename IntTypes<Format>::UnsignedType UnsignedType;
+ 
+ 	static const int kScaleBits = (Format + 1) * CHAR_BIT - 1;
+-	static const int kMinSignedValue = -1 << kScaleBits;
++	static const int kMinSignedValue = -1U << kScaleBits;
+ 
+ 	struct signedToUnsigned : public std::unary_function<SignedType, UnsignedType>
+ 	{
+diff --git a/test/FloatToInt.cpp b/test/FloatToInt.cpp
+index 0d179a8..624be25 100644
+--- a/test/FloatToInt.cpp
++++ b/test/FloatToInt.cpp
+@@ -115,7 +115,7 @@ TEST_F(FloatToIntTest, Int16)
+ 		EXPECT_EQ(readData[i], expectedData[i]);
+ }
+ 
+-static const int32_t kMinInt24 = -1<<23;
++static const int32_t kMinInt24 = -1U<<23;
+ static const int32_t kMaxInt24 = (1<<23) - 1;
+ 
+ TEST_F(FloatToIntTest, Int24)
+diff --git a/test/IntToFloat.cpp b/test/IntToFloat.cpp
+index b716635..b619a2c 100644
+--- a/test/IntToFloat.cpp
++++ b/test/IntToFloat.cpp
+@@ -117,7 +117,7 @@ TEST_F(IntToFloatTest, Int16)
+ 		EXPECT_EQ(readData[i], expectedData[i]);
+ }
+ 
+-static const int32_t kMinInt24 = -1<<23;
++static const int32_t kMinInt24 = -1U<<23;
+ static const int32_t kMaxInt24 = (1<<23) - 1;
+ 
+ TEST_F(IntToFloatTest, Int24)
+diff --git a/test/Sign.cpp b/test/Sign.cpp
+index 7275399..b3609ca 100644
+--- a/test/Sign.cpp
++++ b/test/Sign.cpp
+@@ -116,7 +116,7 @@ TEST_F(SignConversionTest, Int16)
+ 		EXPECT_EQ(readData[i], expectedData[i]);
+ }
+ 
+-static const int32_t kMinInt24 = -1<<23;
++static const int32_t kMinInt24 = -1U<<23;
+ static const int32_t kMaxInt24 = (1<<23) - 1;
+ static const uint32_t kMaxUInt24 = (1<<24) - 1;
+ 
+

audiofile-fix-test-compile-err.patch

@@ -0,0 +1,47 @@
+From ea3b8e88065b9698357b3636363b78c2bb74e240 Mon Sep 17 00:00:00 2001
+From: Shouping Wang<wangshouping@huawei.com>
+Date: Sat, 21 Mar 2020 18:54:32 +0800
+Subject: [PATCH] audiofile-fix-test-compile-err
+
+Signed-off-by: Shouping Wang<wangshouping@huawei.com>
+---
+ test/NeXT.cpp | 8 ++++----
+ 1 file changed, 4 insertions(+), 4 deletions(-)
+
+diff --git a/test/NeXT.cpp b/test/NeXT.cpp
+index 7e39850..b8829a9 100644
+--- a/test/NeXT.cpp
++++ b/test/NeXT.cpp
+@@ -41,9 +41,9 @@ const char kDataUnspecifiedLength[] =
+ {
+ 	'.', 's', 'n', 'd',
+ 	0, 0, 0, 24, // offset of 24 bytes
+-	0xff, 0xff, 0xff, 0xff, // unspecified length
++	(char)0xff, (char)0xff, (char)0xff, (char)0xff, // unspecified length
+ 	0, 0, 0, 3, // 16-bit linear
+-	0, 0, 172, 68, // 44100 Hz
++	0, 0, (char)172, 68, // 44100 Hz
+ 	0, 0, 0, 1, // 1 channel
+ 	0, 1,
+ 	0, 1,
+@@ -63,7 +63,7 @@ const char kDataTruncated[] =
+ 	0, 0, 0, 24, // offset of 24 bytes
+ 	0, 0, 0, 20, // length of 20 bytes
+ 	0, 0, 0, 3, // 16-bit linear
+-	0, 0, 172, 68, // 44100 Hz
++	0, 0, (char)172, 68, // 44100 Hz
+ 	0, 0, 0, 1, // 1 channel
+ 	0, 1,
+ 	0, 1,
+@@ -158,7 +158,7 @@ const char kDataZeroChannels[] =
+ 	0, 0, 0, 24, // offset of 24 bytes
+ 	0, 0, 0, 2, // 2 bytes
+ 	0, 0, 0, 3, // 16-bit linear
+-	0, 0, 172, 68, // 44100 Hz
++	0, 0, (char)172, 68, // 44100 Hz
+ 	0, 0, 0, 0, // 0 channels
+ 	0, 1
+ };
+-- 
+1.8.3.1
+

audiofile.spec

@@ -1,18 +1,20 @@
 Name: audiofile
 Version: 0.3.6
-Release: 22
+Release: 27
 Summary: Library for reading and writing audio files in many common formats
 License: LGPLv2+ and GPLv2+
 URL: http://audiofile.68k.org/
 Source0: http://audiofile.68k.org/%{name}-%{version}.tar.gz
-Patch0: audiofile-0.3.6-CVE-2015-7747.patch
-Patch1: audiofile-0.3.6-left-shift-neg.patch
-Patch2: audiofile-0.3.6-narrowing.patch
-Patch3: audiofile-0.3.6-pull42.patch
-Patch4: audiofile-0.3.6-pull43.patch
-Patch5: audiofile-0.3.6-pull44.patch
+Patch0: audiofile-CVE-2015-7747.patch
+Patch1: audiofile-fix-gcc6-compile-error.patch
+Patch2: audiofile-fix-test-compile-err.patch
+Patch3: backport-CVE-2017-6828.patch
+Patch4: backport-CVE-2017-6829.patch
+Patch5: backport-CVE-2017-6831.patch
+Patch6: backport-CVE-2017-6838.patch
+Patch7: backport-CVE-2017-6839.patch
 
-BuildRequires: gcc-c++ libtool alsa-lib-devel flac-devel
+BuildRequires: gcc-c++ libtool alsa-lib-devel flac-devel chrpath
 
 %description
 The Audio File Library is a C-based library for reading and writing audio files in many
@@ -46,6 +48,8 @@ other resources you can use to develop Audio File applications.
 rm -rf $RPM_BUILD_ROOT
 %make_install
 rm -f $RPM_BUILD_ROOT%{_libdir}/libaudiofile.la
+chrpath --delete %{buildroot}%{_bindir}/sfinfo
+chrpath --delete %{buildroot}%{_bindir}/sfconvert
 
 %check
 make check
@@ -78,6 +82,33 @@ make check
 %{_mandir}/man3/*
 
 %changelog
+* Mon Sep 6 2021 Hongxun Ren<renhongxun@huawei.com> - 0.3.6-27
+- Type:enhanence
+- ID:NA
+- SUG:NA
+- DESC:remove rpath
+
+* Thu Jul 22 2021 wuchaochao <wuchaochao4@huawei.com> - 0.3.6-26
+- Remove BuildRequires gdb
+
+* Fri Feb 19 2021 shixuantong<shixuantong@huawei.com> - 0.3.6-25
+- Type:cves
+- ID:CVE-2017-6828 CVE-2017-6829 CVE-2017-6831 CVE-2017-6838 CVE-2017-6839
+- SUG:NA
+- DESC:fix CVE-2017-6828 CVE-2017-6829 CVE-2017-6831 CVE-2017-6838 CVE-2017-6839
+
+* Sat Mar 21 2020 Shouping Wang<wangshouping@huawei.com> - 0.3.6-24
+- Type:bugfix
+- ID:NA
+- SUG:NA
+- DESC:fix test compile err
+
+* Thu Jan  9 2020 JeanLeo<liujianliu.liu@huawei.com> - 0.3.6-23
+- Type:bugfix
+- ID:NA
+- SUG:NA
+- DESC:update software package
+
 * Fri Sep 27 2019 chengquan<chengquan3@huawei.com> - 0.3.6-22
 - Type:bugfix
 - ID:NA

backport-CVE-2017-6828.patch

@@ -0,0 +1,31 @@
+From c48e4c6503f7dabd41f11d4c9c7b7f8960e7f2c0 Mon Sep 17 00:00:00 2001
+From: Antonio Larrosa <larrosa@kde.org>
+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;
+ 

backport-CVE-2017-6829.patch

@@ -0,0 +1,34 @@
+From 25eb00ce913452c2e614548d7df93070bf0d066f Mon Sep 17 00:00:00 2001
+From: Antonio Larrosa <larrosa@kde.org>
+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)

backport-CVE-2017-6831.patch

@@ -0,0 +1,37 @@
+From a2e9eab8ea87c4ffc494d839ebb4ea145eb9f2e6 Mon Sep 17 00:00:00 2001
+From: Antonio Larrosa <larrosa@kde.org>
+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;

backport-CVE-2017-6838.patch

@@ -0,0 +1,67 @@
+From 7d65f89defb092b63bcbc5d98349fb222ca73b3c Mon Sep 17 00:00:00 2001
+From: Antonio Larrosa <larrosa@kde.org>
+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;

backport-CVE-2017-6839.patch

@@ -0,0 +1,117 @@
+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
+---
+ 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++;