audiofile.spec

@@ -1,6 +1,6 @@
 Name: audiofile
 Version: 0.3.6
-Release: 28
+Release: 24
 Summary: Library for reading and writing audio files in many common formats
 License: LGPLv2+ and GPLv2+
 URL: http://audiofile.68k.org/
@@ -8,13 +8,8 @@ Source0: http://audiofile.68k.org/%{name}-%{version}.tar.gz
 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 chrpath
+BuildRequires: gcc-c++ libtool alsa-lib-devel flac-devel gdb
 
 %description
 The Audio File Library is a C-based library for reading and writing audio files in many
@@ -48,11 +43,9 @@ 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
+%check
-#make check
+make check
 
 %pre
 
@@ -82,24 +75,6 @@ chrpath --delete %{buildroot}%{_bindir}/sfconvert
 %{_mandir}/man3/*
 
 %changelog
-* Tue Nov 15 2022 dillon chen <dillon.chen@gmail.com> - 0.3.6-28
-- Remove check when flac >= 1.3.4
-
-* 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

audiofile.yaml

@@ -1,4 +0,0 @@
-version_control: github
-src_repo: mpruett/audiofile
-tag_prefix: ^audiofile- 
-seperator: . 

backport-CVE-2017-6828.patch

@@ -1,31 +0,0 @@
-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

@@ -1,34 +0,0 @@
-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

@@ -1,37 +0,0 @@
-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

@@ -1,67 +0,0 @@
-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

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