Package init

2019-09-30 10:31:49 -04:00 · 2019-09-30 10:31:49 -04:00 · ba3c548d39
commit ba3c548d39
8 changed files with 435 additions and 0 deletions
--- a/audiofile-0.3.6-CVE-2015-7747.patch
+++ b/audiofile-0.3.6-CVE-2015-7747.patch
@ -0,0 +1,12 @@
+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));
--- a/audiofile-0.3.6-left-shift-neg.patch
+++ b/audiofile-0.3.6-left-shift-neg.patch
@ -0,0 +1,48 @@
+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;
+ 
--- a/audiofile-0.3.6-narrowing.patch
+++ b/audiofile-0.3.6-narrowing.patch
@ -0,0 +1,52 @@
+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
+ };
--- a/audiofile-0.3.6-pull42.patch
+++ b/audiofile-0.3.6-pull42.patch
@ -0,0 +1,176 @@
+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;
--- a/audiofile-0.3.6-pull43.patch
+++ b/audiofile-0.3.6-pull43.patch
@ -0,0 +1,21 @@
+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)
--- a/audiofile-0.3.6-pull44.patch
+++ b/audiofile-0.3.6-pull44.patch
@ -0,0 +1,31 @@
+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;
--- a/audiofile-0.3.6.tar.gz
+++ b/audiofile-0.3.6.tar.gz
--- a/audiofile.spec
+++ b/audiofile.spec
@ -0,0 +1,95 @@
+Name: audiofile
+Version: 0.3.6
+Release: 22
+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
+
+BuildRequires: gcc-c++ libtool alsa-lib-devel flac-devel
+
+%description
+The Audio File Library is a C-based library for reading and writing audio files in many
+common formats.
+
+The Audio File Library provides a uniform API which abstracts away details of file formats
+and data formats. The same calls for opening a file, accessing and manipulating audio
+metadata (e.g. sample rate, sample format, textual information, MIDI parameters), and
+reading and writing sample data will work with any supported audio file format.
+
+%package devel
+Summary: Development files for Audio File applications
+Requires: %{name}%{?_isa} = %{version}-%{release}
+Provides: audiofile-static
+Obsoletes: audiofile-static
+
+%description devel
+The audiofile-devel package contains libraries, include files, and
+other resources you can use to develop Audio File applications.
+
+%package_help
+
+%prep
+%autosetup -n %{name}-%{version} -p1
+
+%build
+%configure
+%make_build
+
+%install
+rm -rf $RPM_BUILD_ROOT
+%make_install
+rm -f $RPM_BUILD_ROOT%{_libdir}/libaudiofile.la
+
+%check
+make check
+
+%pre
+
+%preun
+
+%post
+
+%postun
+
+%ldconfig_scriptlets
+
+%files
+%doc ACKNOWLEDGEMENTS AUTHORS NEWS NOTES README TODO COPYING*
+%{_bindir}/sfconvert
+%{_bindir}/sfinfo
+%{_libdir}/lib*.so.1*
+
+%files devel
+%doc ChangeLog docs/*.3.txt
+%{_libdir}/libaudiofile.so
+%{_libdir}/pkgconfig/audiofile.pc
+%{_includedir}/*
+%{_libdir}/libaudiofile.a
+
+%files help
+%{_mandir}/man1/*
+%{_mandir}/man3/*
+
+%changelog
+* Fri Sep 27 2019 chengquan<chengquan3@huawei.com> - 0.3.6-22
+- Type:bugfix
+- ID:NA
+- SUG:NA
+- DESC:add help package and merge devel package
+
+* Tue Sep 17 2019 shenyangyang<shenyangyang4@huawei.com> - 0.3.6-21
+- Type:bugfix
+- ID:NA
+- SUG:NA
+- DESC:and % before isa of devel
+
+* Tue Aug 13 2019 openEuler Buildteam <buildteam@openeuler.org> - 0.3.6-20
+- Package init
+