Commit d800dc2e authored by Luca Giambonini's avatar Luca Giambonini

gcc rebuild, group2

parent 6a419170
pkgbase=atkmm
pkgname=(atkmm atkmm-docs)
pkgver=2.24.2+1+gf30b47f
pkgrel=1
pkgrel=2
pkgdesc="C++ bindings for ATK"
url="http://www.gtkmm.org/"
arch=('x86_64')
......
Description: Fix FTBFS with GCC 6
Author: Michael Schwendt <mschwendt@fedoraproject.org>
Origin: vendor, https://github.com/mpruett/audiofile/pull/27
Bug-Debian: https://bugs.debian.org/812055
---
This patch header follows DEP-3: http://dep.debian.net/deps/dep3/
--- 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 = 0-(1U<<kScaleBits);
struct signedToUnsigned : public std::unary_function<SignedType, UnsignedType>
{
--- 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 = 0-(1U<<23);
static const int32_t kMaxInt24 = (1<<23) - 1;
TEST_F(FloatToIntTest, Int24)
--- 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 = 0-(1U<<23);
static const int32_t kMaxInt24 = (1<<23) - 1;
TEST_F(IntToFloatTest, Int24)
--- a/test/NeXT.cpp
+++ b/test/NeXT.cpp
@@ -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 @@ const char kDataUnspecifiedLength[] =
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 @@ TEST(NeXT, Truncated)
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/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 = 0-(1U<<23);
static const int32_t kMaxInt24 = (1<<23) - 1;
static const uint32_t kMaxUInt24 = (1<<24) - 1;
Description: fix buffer overflow when changing both sample format and
number of channels
Origin: https://github.com/mpruett/audiofile/pull/25
Bug-Ubuntu: https://bugs.launchpad.net/ubuntu/+source/audiofile/+bug/1502721
Bug-Debian: https://bugs.debian.org/801102
--- a/libaudiofile/modules/ModuleState.cpp
+++ b/libaudiofile/modules/ModuleState.cpp
@@ -402,7 +402,7 @@ status ModuleState::arrange(AFfilehandle
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/test/Makefile.am
+++ b/test/Makefile.am
@@ -26,6 +26,7 @@ TESTS = \
VirtualFile \
floatto24 \
query2 \
+ sixteen-stereo-to-eight-mono \
sixteen-to-eight \
testchannelmatrix \
testdouble \
@@ -139,6 +140,7 @@ printmarkers_SOURCES = printmarkers.c
printmarkers_LDADD = $(LIBAUDIOFILE) -lm
sixteen_to_eight_SOURCES = sixteen-to-eight.c TestUtilities.cpp TestUtilities.h
+sixteen_stereo_to_eight_mono_SOURCES = sixteen-stereo-to-eight-mono.c TestUtilities.cpp TestUtilities.h
testchannelmatrix_SOURCES = testchannelmatrix.c TestUtilities.cpp TestUtilities.h
--- /dev/null
+++ b/test/sixteen-stereo-to-eight-mono.c
@@ -0,0 +1,118 @@
+/*
+ Audio File Library
+
+ Copyright 2000, Silicon Graphics, Inc.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2 of the License, or
+ (at your option) any later version.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License along
+ with this program; if not, write to the Free Software Foundation, Inc.,
+ 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+*/
+
+/*
+ sixteen-stereo-to-eight-mono.c
+
+ This program tests the conversion from 2-channel 16-bit integers to
+ 1-channel 8-bit integers.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdint.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <unistd.h>
+#include <limits.h>
+
+#include <audiofile.h>
+
+#include "TestUtilities.h"
+
+int main (int argc, char **argv)
+{
+ AFfilehandle file;
+ AFfilesetup setup;
+ int16_t frames16[] = {14298, 392, 3923, -683, 958, -1921};
+ int8_t frames8[] = {28, 6, -2};
+ int i, frameCount = 3;
+ int8_t byte;
+ AFframecount result;
+
+ setup = afNewFileSetup();
+
+ afInitFileFormat(setup, AF_FILE_WAVE);
+
+ afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
+ afInitChannels(setup, AF_DEFAULT_TRACK, 2);
+
+ char *testFileName;
+ if (!createTemporaryFile("sixteen-to-eight", &testFileName))
+ {
+ fprintf(stderr, "Could not create temporary file.\n");
+ exit(EXIT_FAILURE);
+ }
+
+ file = afOpenFile(testFileName, "w", setup);
+ if (file == AF_NULL_FILEHANDLE)
+ {
+ fprintf(stderr, "could not open file for writing\n");
+ exit(EXIT_FAILURE);
+ }
+
+ afFreeFileSetup(setup);
+
+ afWriteFrames(file, AF_DEFAULT_TRACK, frames16, frameCount);
+
+ afCloseFile(file);
+
+ file = afOpenFile(testFileName, "r", AF_NULL_FILESETUP);
+ if (file == AF_NULL_FILEHANDLE)
+ {
+ fprintf(stderr, "could not open file for reading\n");
+ exit(EXIT_FAILURE);
+ }
+
+ afSetVirtualSampleFormat(file, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 8);
+ afSetVirtualChannels(file, AF_DEFAULT_TRACK, 1);
+
+ for (i=0; i<frameCount; i++)
+ {
+ /* Read one frame. */
+ result = afReadFrames(file, AF_DEFAULT_TRACK, &byte, 1);
+
+ if (result != 1)
+ break;
+
+ /* Compare the byte read with its precalculated value. */
+ if (memcmp(&byte, &frames8[i], 1) != 0)
+ {
+ printf("error\n");
+ printf("expected %d, got %d\n", frames8[i], byte);
+ exit(EXIT_FAILURE);
+ }
+ else
+ {
+#ifdef DEBUG
+ printf("got what was expected: %d\n", byte);
+#endif
+ }
+ }
+
+ afCloseFile(file);
+ unlink(testFileName);
+ free(testFileName);
+
+ exit(EXIT_SUCCESS);
+}
From: Antonio Larrosa <larrosa@kde.org>
Date: Mon, 6 Mar 2017 18:02:31 +0100
Subject: 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: Antonio Larrosa <larrosa@kde.org>
Date: Mon, 6 Mar 2017 12:51:22 +0100
Subject: 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 9dd8511..0fc48e8 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;
From: Antonio Larrosa <larrosa@kde.org>
Date: Mon, 6 Mar 2017 13:43:53 +0100
Subject: 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: Antonio Larrosa <larrosa@kde.org>
Date: Mon, 6 Mar 2017 13:54:52 +0100
Subject: 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: Antonio Larrosa <larrosa@kde.org>
Date: Fri, 10 Mar 2017 15:40:02 +0100
Subject: Fix signature of multiplyCheckOverflow. It returns a bool, not an int
---
libaudiofile/modules/MSADPCM.cpp | 2 +-
sfcommands/sfconvert.c | 2 +-
2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/libaudiofile/modules/MSADPCM.cpp b/libaudiofile/modules/MSADPCM.cpp
index ef9c38c..d8c9553 100644
--- a/libaudiofile/modules/MSADPCM.cpp
+++ b/libaudiofile/modules/MSADPCM.cpp
@@ -116,7 +116,7 @@ int firstBitSet(int x)
#define __has_builtin(x) 0
#endif
-int multiplyCheckOverflow(int a, int b, int *result)
+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);
diff --git a/sfcommands/sfconvert.c b/sfcommands/sfconvert.c
index 970a3e4..367f7a5 100644
--- a/sfcommands/sfconvert.c
+++ b/sfcommands/sfconvert.c
@@ -60,7 +60,7 @@ int firstBitSet(int x)
#define __has_builtin(x) 0
#endif
-int multiplyCheckOverflow(int a, int b, int *result)
+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);
From: Antonio Larrosa <larrosa@kde.org>
Date: Mon, 6 Mar 2017 18:59:26 +0100
Subject: 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 0fc48e8..d04b796 100644
--- a/libaudiofile/WAVE.cpp
+++ b/libaudiofile/WAVE.cpp
@@ -332,6 +332,7 @@ 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;
@@ -339,6 +340,7 @@ 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: Antonio Larrosa <larrosa@kde.org>
Date: Thu, 9 Mar 2017 10:21:18 +0100
Subject: Check for division by zero in BlockCodec::runPull
---
libaudiofile/modules/BlockCodec.cpp | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/libaudiofile/modules/BlockCodec.cpp b/libaudiofile/modules/BlockCodec.cpp
index 4731be1..eb2fb4d 100644
--- a/libaudiofile/modules/BlockCodec.cpp
+++ b/libaudiofile/modules/BlockCodec.cpp
@@ -47,7 +47,7 @@ void BlockCodec::runPull()
// 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++)
pkgname=audiofile
pkgver=0.3.6
pkgrel=2
pkgrel=3
pkgdesc="Silicon Graphics Audio File Library"
arch=('x86_64')
url="http://www.68k.org/~michael/audiofile/"
license=('LGPL')
depends=('gcc-libs' 'alsa-lib' 'flac')
source=("http://audiofile.68k.org/$pkgname-$pkgver.tar.gz")
md5sums=('2731d79bec0acef3d30d2fc86b0b72fd')
source=("https://audiofile.68k.org/$pkgname-$pkgver.tar.gz"
01_gcc6.patch
03_CVE-2015-7747.patch
04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
05_Always-check-the-number-of-coefficients.patch
06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch
07_Check-for-multiplication-overflow-in-sfconvert.patch
08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch
09_Actually-fail-when-error-occurs-in-parseFormat.patch
10_Check-for-division-by-zero-in-BlockCodec-runPull.patch)
sha256sums=('cdc60df19ab08bfe55344395739bb08f50fc15c92da3962fac334d3bff116965'
'a1904603c0292e76530f635dfc1828fb4e0d9d13555581cad33c0200640f7a27'
'bcfc180708d089b5abe0ae1439809b5a4306a08917b0212c3d135e5ec56711f2'
'540c517828d5573ba7bc3fd9b3811f39f4ea0132011d348d22bdfc545e865a8e'
'1b55abeb867d66b7d3b7c34585e77e6d3656c6317b582c99f3280d37523c7718'
'7a464eb7521ae8deb67516309bb396caa93135dc62fbad7351e67923b1766423'
'2ed5cc3b57394ea33ad466ca9844b766e4cb91dd7b1e2b71deaf15cf881dbf51'
'257f157cf2cc8947e0f5be4bff2c4afddbe73643e9e39a83171dbea02f5d52f4'
'48deaaa07bfade35208edb9e22b4fe78f91470012414ddb26cd68f684c95e33d'
'f31d51ebd8f8e0bd076cd1bce34b210c4dbbd959ca9b87693ad86a6399c492a3')
prepare() {
cd $pkgname-$pkgver
patch -Np1 -i ../01_gcc6.patch
patch -Np1 -i ../03_CVE-2015-7747.patch
patch -Np1 -i ../04_clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
patch -Np1 -i ../05_Always-check-the-number-of-coefficients.patch
patch -Np1 -i ../06_Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch
patch -Np1 -i ../07_Check-for-multiplication-overflow-in-sfconvert.patch
patch -Np1 -i ../08_Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch
patch -Np1 -i ../09_Actually-fail-when-error-occurs-in-parseFormat.patch
patch -Np1 -i ../10_Check-for-division-by-zero-in-BlockCodec-runPull.patch
}
build() {
cd "$srcdir/$pkgname-$pkgver"
......
#
# Platform Packages for Chakra, part of chakra-project.org
#
# maintainer abveritas[at]chakra-project[dot]org>
pkgname=autopano-sift-c
pkgver=2.5.1
pkgrel=5
pkgrel=6
pkgdesc="Identify key feature points within arbitrary images"
arch=('x86_64')
url="http://hugin.sourceforge.net/"
......
pkgname=cdrdao
pkgver=1.2.3
pkgrel=12
pkgrel=13
arch=('x86_64')
license=('GPL')
url="http://cdrdao.sourceforge.net/"
pkgdesc='Records audio/data CD-Rs in disk-at-once (DAO) mode'
depends=('gcc-libs' 'lame' 'libmad' 'libvorbis' 'libao')
makedepends=('gcc-libs' 'lame' 'libmad' 'libvorbis' 'libao' 'libsigc++2.0')
source=("http://downloads.sourceforge.net/${pkgname}/${pkgname}-${pkgver}.tar.bz2"
'cdrdao-1.2.3-autoconf-update.patch'
'cdrdao-1.2.3-k3b.patch'
'cdrdao-1.2.3-stat.patch')
cdrdao-1.2.3-autoconf-update.patch
cdrdao-1.2.3-k3b.patch
cdrdao-1.2.3-stat.patch
cdrdao-1.2.3-format_security.patch
cdrdao-1.2.3-narrowing.patch)
md5sums=('8d15ba6280bb7ba2f4d6be31d28b3c0c'
'8e53dfc174f7c0882194caa05e68b85e'
'696f6ca01e1eeb9b6a5be88e535d9398'
'0fce05542ebad283f36fa1c4d62992a0')
'0fce05542ebad283f36fa1c4d62992a0'
'9c3dc231b2b1e862b8b95d7778bc439e'
'9344989788e013a16f06556bde282d09')
build() {
cd ${srcdir}/${pkgbase}-${pkgver}
patch -p1 -i "${srcdir}/cdrdao-1.2.3-autoconf-update.patch"
patch -p1 -i "${srcdir}/cdrdao-1.2.3-k3b.patch"
patch -p1 -i "${srcdir}/cdrdao-1.2.3-stat.patch"
prepare() {
cd ${pkgname}-${pkgver}
./configure --prefix=/usr \
--mandir=/usr/share/man \
--sysconfdir=/etc \
--with-xdao --with-lame \
--with-ogg-support --with-mp3-support
make
patch -Np1 -i ../cdrdao-1.2.3-autoconf-update.patch
patch -Np1 -i ../cdrdao-1.2.3-k3b.patch
patch -Np1 -i ../cdrdao-1.2.3-stat.patch
patch -Np1 -i ../cdrdao-1.2.3-format_security.patch
patch -Np1