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rebuilt with debug package, -tools subpackage and patches from arch linux [release 0.3.6-2mamba;Sat Dec 12 2020]

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This commit is contained in:
Silvan Calarco 2024-01-06 03:49:58 +01:00
parent e2dca5c4ea
commit 8ad7b84bca
13 changed files with 1087 additions and 17 deletions
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36
libaudiofile-0.3.6-Actually-fail-when-error-occurs-in-parseFormat.patch Normal file
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@ -0,0 +1,36 @@
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;

30
libaudiofile-0.3.6-Always-check-the-number-of-coefficients.patch Normal file
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@ -0,0 +1,30 @@
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;

156
libaudiofile-0.3.6-CVE-2015-7747.patch Normal file
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@ -0,0 +1,156 @@
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);
+}

28
libaudiofile-0.3.6-CVE-2018-13440.patch Normal file
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@ -0,0 +1,28 @@
From fde6d79fb8363c4a329a184ef0b107156602b225 Mon Sep 17 00:00:00 2001
From: Wim Taymans <wtaymans@redhat.com>
Date: Thu, 27 Sep 2018 10:48:45 +0200
Subject: [PATCH] ModuleState: handle compress/decompress init failure
When the unit initcompress or initdecompress function fails,
m_fileModule is NULL. Return AF_FAIL in that case instead of
causing NULL pointer dereferences later.
Fixes #49
---
libaudiofile/modules/ModuleState.cpp | 3 +++
1 file changed, 3 insertions(+)
diff --git a/libaudiofile/modules/ModuleState.cpp b/libaudiofile/modules/ModuleState.cpp
index 0c29d7a..070fd9b 100644
--- a/libaudiofile/modules/ModuleState.cpp
+++ b/libaudiofile/modules/ModuleState.cpp
@@ -75,6 +75,9 @@ status ModuleState::initFileModule(AFfilehandle file, Track *track)
m_fileModule = unit->initcompress(track, file->m_fh, file->m_seekok,
file->m_fileFormat == AF_FILE_RAWDATA, &chunkFrames);
+ if (!m_fileModule)
+ return AF_FAIL;
+
if (unit->needsRebuffer)
{
assert(unit->nativeSampleFormat == AF_SAMPFMT_TWOSCOMP);

26
libaudiofile-0.3.6-CVE-2018-17095.patch Normal file
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@ -0,0 +1,26 @@
From 822b732fd31ffcb78f6920001e9b1fbd815fa712 Mon Sep 17 00:00:00 2001
From: Wim Taymans <wtaymans@redhat.com>
Date: Thu, 27 Sep 2018 12:11:12 +0200
Subject: [PATCH] SimpleModule: set output chunk framecount after pull
After pulling the data, set the output chunk to the amount of
frames we pulled so that the next module in the chain has the correct
frame count.
Fixes #50 and #51
---
libaudiofile/modules/SimpleModule.cpp | 1 +
1 file changed, 1 insertion(+)
diff --git a/libaudiofile/modules/SimpleModule.cpp b/libaudiofile/modules/SimpleModule.cpp
index 2bae1eb..e87932c 100644
--- a/libaudiofile/modules/SimpleModule.cpp
+++ b/libaudiofile/modules/SimpleModule.cpp
@@ -26,6 +26,7 @@
void SimpleModule::runPull()
{
pull(m_outChunk->frameCount);
+ m_outChunk->frameCount = m_inChunk->frameCount;
run(*m_inChunk, *m_outChunk);
}

21
libaudiofile-0.3.6-Check-for-division-by-zero-in-BlockCodec-runPull.patch Normal file
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@ -0,0 +1,21 @@
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++)

116
libaudiofile-0.3.6-Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch Normal file
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@ -0,0 +1,116 @@
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++;

66
libaudiofile-0.3.6-Check-for-multiplication-overflow-in-sfconvert.patch Normal file
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@ -0,0 +1,66 @@
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;

35
libaudiofile-0.3.6-Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch Normal file
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@ -0,0 +1,35 @@
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);

33
libaudiofile-0.3.6-clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch Normal file
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@ -0,0 +1,33 @@
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)

102
libaudiofile-0.3.6-gcc6.patch Normal file
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@ -0,0 +1,102 @@
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;

381
libaudiofile-0.3.6-hurd.patch Normal file
View File

@ -0,0 +1,381 @@
Description: Remove usage of PATH_MAX in tests to fix FTBFS on Hurd.
jcowgill: Removed Changelog changes
Author: Pino Toscano <toscano.pino@tiscali.it>
Origin: backport, https://github.com/mpruett/audiofile/commit/34c261034f1193a783196618f0052112e00fbcfe
Bug: https://github.com/mpruett/audiofile/pull/17
Bug-Debian: https://bugs.debian.org/762595
---
This patch header follows DEP-3: http://dep.debian.net/deps/dep3/
--- a/test/TestUtilities.cpp
+++ b/test/TestUtilities.cpp
@@ -21,8 +21,8 @@
#include "TestUtilities.h"
#include <limits.h>
-#include <stdio.h>
#include <stdlib.h>
+#include <string.h>
#include <unistd.h>
bool createTemporaryFile(const std::string &prefix, std::string *path)
@@ -35,12 +35,12 @@ bool createTemporaryFile(const std::stri
return true;
}
-bool createTemporaryFile(const char *prefix, char *path)
+bool createTemporaryFile(const char *prefix, char **path)
{
- snprintf(path, PATH_MAX, "/tmp/%s-XXXXXX", prefix);
- int fd = ::mkstemp(path);
- if (fd < 0)
- return false;
- ::close(fd);
- return true;
+ *path = NULL;
+ std::string pathString;
+ bool result = createTemporaryFile(prefix, &pathString);
+ if (result)
+ *path = ::strdup(pathString.c_str());
+ return result;
}
--- a/test/TestUtilities.h
+++ b/test/TestUtilities.h
@@ -53,7 +53,7 @@ extern "C" {
#include <stdbool.h>
-bool createTemporaryFile(const char *prefix, char *path);
+bool createTemporaryFile(const char *prefix, char **path);
#ifdef __cplusplus
}
--- a/test/floatto24.c
+++ b/test/floatto24.c
@@ -86,8 +86,8 @@ int main (int argc, char **argv)
afInitChannels(setup, AF_DEFAULT_TRACK, 1);
afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_FLOAT, 32);
- char testFileName[PATH_MAX];
- if (!createTemporaryFile("floatto24", testFileName))
+ char *testFileName;
+ if (!createTemporaryFile("floatto24", &testFileName))
{
fprintf(stderr, "Could not create temporary file.\n");
exit(EXIT_FAILURE);
@@ -182,6 +182,7 @@ int main (int argc, char **argv)
}
unlink(testFileName);
+ free(testFileName);
exit(EXIT_SUCCESS);
}
--- a/test/sixteen-to-eight.c
+++ b/test/sixteen-to-eight.c
@@ -57,8 +57,8 @@ int main (int argc, char **argv)
afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_UNSIGNED, 8);
afInitChannels(setup, AF_DEFAULT_TRACK, 1);
- char testFileName[PATH_MAX];
- if (!createTemporaryFile("sixteen-to-eight", testFileName))
+ char *testFileName;
+ if (!createTemporaryFile("sixteen-to-eight", &testFileName))
{
fprintf(stderr, "Could not create temporary file.\n");
exit(EXIT_FAILURE);
@@ -113,6 +113,7 @@ int main (int argc, char **argv)
afCloseFile(file);
unlink(testFileName);
+ free(testFileName);
exit(EXIT_SUCCESS);
}
--- a/test/testchannelmatrix.c
+++ b/test/testchannelmatrix.c
@@ -39,7 +39,7 @@
#include "TestUtilities.h"
-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
const short samples[] = {300, -300, 515, -515, 2315, -2315, 9154, -9154};
#define SAMPLE_COUNT (sizeof (samples) / sizeof (short))
@@ -47,7 +47,11 @@ const short samples[] = {300, -300, 515,
void cleanup (void)
{
- unlink(sTestFileName);
+ if (sTestFileName)
+ {
+ unlink(sTestFileName);
+ free(sTestFileName);
+ }
}
void ensure (int condition, const char *message)
@@ -76,7 +80,7 @@ int main (void)
afInitFileFormat(setup, AF_FILE_AIFFC);
/* Write stereo data to test file. */
- ensure(createTemporaryFile("testchannelmatrix", sTestFileName),
+ ensure(createTemporaryFile("testchannelmatrix", &sTestFileName),
"could not create temporary file");
file = afOpenFile(sTestFileName, "w", setup);
ensure(file != AF_NULL_FILEHANDLE, "could not open file for writing");
--- a/test/testdouble.c
+++ b/test/testdouble.c
@@ -38,7 +38,7 @@
#include "TestUtilities.h"
-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
const double samples[] =
{1.0, 0.6, -0.3, 0.95, 0.2, -0.6, 0.9, 0.4, -0.22, 0.125, 0.1, -0.4};
@@ -48,7 +48,11 @@ void testdouble (int fileFormat);
void cleanup (void)
{
- unlink(sTestFileName);
+ if (sTestFileName)
+ {
+ unlink(sTestFileName);
+ free(sTestFileName);
+ }
}
void ensure (int condition, const char *message)
@@ -96,7 +100,7 @@ void testdouble (int fileFormat)
afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_DOUBLE, 64);
afInitChannels(setup, AF_DEFAULT_TRACK, 2);
- ensure(createTemporaryFile("testdouble", sTestFileName),
+ ensure(createTemporaryFile("testdouble", &sTestFileName),
"could not create temporary file");
file = afOpenFile(sTestFileName, "w", setup);
ensure(file != AF_NULL_FILEHANDLE, "could not open file for writing");
--- a/test/testfloat.c
+++ b/test/testfloat.c
@@ -38,7 +38,7 @@
#include "TestUtilities.h"
-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
const float samples[] =
{1.0, 0.6, -0.3, 0.95, 0.2, -0.6, 0.9, 0.4, -0.22, 0.125, 0.1, -0.4};
@@ -48,7 +48,11 @@ void testfloat (int fileFormat);
void cleanup (void)
{
- unlink(sTestFileName);
+ if (sTestFileName)
+ {
+ unlink(sTestFileName);
+ free(sTestFileName);
+ }
}
void ensure (int condition, const char *message)
@@ -96,7 +100,7 @@ void testfloat (int fileFormat)
afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_FLOAT, 32);
afInitChannels(setup, AF_DEFAULT_TRACK, 2);
- ensure(createTemporaryFile("testfloat", sTestFileName),
+ ensure(createTemporaryFile("testfloat", &sTestFileName),
"could not create temporary file");
file = afOpenFile(sTestFileName, "w", setup);
ensure(file != AF_NULL_FILEHANDLE, "could not open file for writing");
--- a/test/testmarkers.c
+++ b/test/testmarkers.c
@@ -32,15 +32,19 @@
#include "TestUtilities.h"
-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
#define FRAME_COUNT 200
void cleanup (void)
{
+ if (sTestFileName)
+ {
#ifndef DEBUG
- unlink(sTestFileName);
+ unlink(sTestFileName);
#endif
+ free(sTestFileName);
+ }
}
void ensure (int condition, const char *message)
@@ -127,7 +131,7 @@ int testmarkers (int fileformat)
int main (void)
{
- ensure(createTemporaryFile("testmarkers", sTestFileName),
+ ensure(createTemporaryFile("testmarkers", &sTestFileName),
"could not create temporary file");
testmarkers(AF_FILE_AIFF);
--- a/test/twentyfour.c
+++ b/test/twentyfour.c
@@ -71,8 +71,8 @@ int main (int argc, char **argv)
afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 24);
afInitChannels(setup, AF_DEFAULT_TRACK, 1);
- char testFileName[PATH_MAX];
- if (!createTemporaryFile("twentyfour", testFileName))
+ char *testFileName;
+ if (!createTemporaryFile("twentyfour", &testFileName))
{
fprintf(stderr, "could not create temporary file\n");
exit(EXIT_FAILURE);
@@ -239,6 +239,7 @@ int main (int argc, char **argv)
exit(EXIT_FAILURE);
}
unlink(testFileName);
+ free(testFileName);
exit(EXIT_SUCCESS);
}
--- a/test/twentyfour2.c
+++ b/test/twentyfour2.c
@@ -45,15 +45,19 @@
#include "TestUtilities.h"
-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
#define FRAME_COUNT 10000
void cleanup (void)
{
+ if (sTestFileName)
+ {
#ifndef DEBUG
- unlink(sTestFileName);
+ unlink(sTestFileName);
#endif
+ free(sTestFileName);
+ }
}
void ensure (int condition, const char *message)
@@ -78,7 +82,7 @@ int main (void)
afInitChannels(setup, AF_DEFAULT_TRACK, 1);
afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 24);
- ensure(createTemporaryFile("twentyfour2", sTestFileName),
+ ensure(createTemporaryFile("twentyfour2", &sTestFileName),
"could not create temporary file");
file = afOpenFile(sTestFileName, "w", setup);
ensure(file != NULL, "could not open test file for writing");
--- a/test/writealaw.c
+++ b/test/writealaw.c
@@ -53,7 +53,7 @@
#include "TestUtilities.h"
-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
#define FRAME_COUNT 16
#define SAMPLE_COUNT FRAME_COUNT
@@ -62,9 +62,13 @@ void testalaw (int fileFormat);
void cleanup (void)
{
+ if (sTestFileName)
+ {
#ifndef DEBUG
- unlink(sTestFileName);
+ unlink(sTestFileName);
#endif
+ free(sTestFileName);
+ }
}
void ensure (int condition, const char *message)
@@ -113,7 +117,7 @@ void testalaw (int fileFormat)
afInitFileFormat(setup, fileFormat);
afInitChannels(setup, AF_DEFAULT_TRACK, 1);
- ensure(createTemporaryFile("writealaw", sTestFileName),
+ ensure(createTemporaryFile("writealaw", &sTestFileName),
"could not create temporary file");
file = afOpenFile(sTestFileName, "w", setup);
afFreeFileSetup(setup);
--- a/test/writeraw.c
+++ b/test/writeraw.c
@@ -44,13 +44,17 @@
#include "TestUtilities.h"
-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
void cleanup (void)
{
+ if (sTestFileName)
+ {
#ifndef DEBUG
- unlink(sTestFileName);
+ unlink(sTestFileName);
#endif
+ free(sTestFileName);
+ }
}
void ensure (int condition, const char *message)
@@ -84,7 +88,7 @@ int main (int argc, char **argv)
afInitChannels(setup, AF_DEFAULT_TRACK, 1);
afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
- ensure(createTemporaryFile("writeraw", sTestFileName),
+ ensure(createTemporaryFile("writeraw", &sTestFileName),
"could not create temporary file");
file = afOpenFile(sTestFileName, "w", setup);
ensure(file != AF_NULL_FILEHANDLE, "unable to open file for writing");
--- a/test/writeulaw.c
+++ b/test/writeulaw.c
@@ -53,7 +53,7 @@
#include "TestUtilities.h"
-static char sTestFileName[PATH_MAX];
+static char *sTestFileName;
#define FRAME_COUNT 16
#define SAMPLE_COUNT FRAME_COUNT
@@ -62,9 +62,13 @@ void testulaw (int fileFormat);
void cleanup (void)
{
+ if (sTestFileName)
+ {
#ifndef DEBUG
- unlink(sTestFileName);
+ unlink(sTestFileName);
#endif
+ free(sTestFileName);
+ }
}
void ensure (int condition, const char *message)
@@ -113,7 +117,7 @@ void testulaw (int fileFormat)
afInitFileFormat(setup, fileFormat);
afInitChannels(setup, AF_DEFAULT_TRACK, 1);
- ensure(createTemporaryFile("writeulaw", sTestFileName),
+ ensure(createTemporaryFile("writeulaw", &sTestFileName),
"could not create temporary file");
file = afOpenFile(sTestFileName, "w", setup);
afFreeFileSetup(setup);

72
libaudiofile.spec
View File

@ -1,6 +1,6 @@
Name: libaudiofile
Version: 0.3.6
Release: 1mamba
Release: 2mamba
Summary: A uniform programming interface to standard digital audio file formats
Group: System/Kernel and Hardware
Vendor: openmamba
@ -9,9 +9,26 @@ Packager: Silvan Calarco <silvan.calarco@mambasoft.it>
URL: http://www.68k.org/~michael/audiofile
Source: http://www.68k.org/~michael/audiofile/audiofile-%{version}.tar.gz
Patch: %{name}-0.2.6-m4_underquoted_warning.patch
Patch1: libaudiofile-0.3.6-gcc6.patch
Patch2: libaudiofile-0.3.6-hurd.patch
Patch3: libaudiofile-0.3.6-CVE-2015-7747.patch
Patch4: libaudiofile-0.3.6-clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
Patch5: libaudiofile-0.3.6-Always-check-the-number-of-coefficients.patch
Patch6: libaudiofile-0.3.6-Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch
Patch7: libaudiofile-0.3.6-Check-for-multiplication-overflow-in-sfconvert.patch
Patch8: libaudiofile-0.3.6-Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch
Patch9: libaudiofile-0.3.6-Actually-fail-when-error-occurs-in-parseFormat.patch
Patch10: libaudiofile-0.3.6-Check-for-division-by-zero-in-BlockCodec-runPull.patch
Patch11: libaudiofile-0.3.6-CVE-2018-13440.patch
Patch12: libaudiofile-0.3.6-CVE-2018-17095.patch
License: LGPL
## AUTOBUILDREQ-BEGIN
BuildRequires: glibc-devel
BuildRequires: ldconfig
BuildRequires: libflac-devel
BuildRequires: libgcc
BuildRequires: libogg-devel
BuildRequires: libstdc++6-devel
## AUTOBUILDREQ-END
BuildRoot: %{_tmppath}/%{name}-%{version}-root
@ -29,22 +46,38 @@ Group: Development/Libraries
Requires: %{name} = %{version}
%description devel
The Silicon Graphics Audio File Library provides a uniform programming interface to standard digital audio file formats.
Currently supported sound file formats include AIFF/AIFF-C, WAVE, NeXT/Sun .snd/.au, and Berkeley/IRCAM/CARL.
Supported compression formats are currently G.711 mu-law and A-law and IMA and MS ADPCM.
Key goals of the Audio File Library are file format transparency and data format transparency.
The same calls for opening a file, accessing and manipulating audio metadata (e.g., sample rate, sample format, textual information, MIDI parameters), and reading/writing sample data will work with any supported audio file format.
Likewise, the format of the audio data presented to the application need not be tied to the format of the data contained in the file.
This package contains static libraries and header files needed for development with %{name}.
This package contains static libraries and header files need for development.
%package tools
Summary: Tools provided with %{name}
Group: Applications/Multimedia
Requires: %{name} = %{version}
%description tools
This package contains sthe tools provided with %{name}.
%debug_package
%prep
%setup -q -n audiofile-%{version}
#%patch -p1
%patch1 -p1
%patch2 -p1
%patch3 -p1
%patch4 -p1
%patch5 -p1
%patch6 -p1
%patch7 -p1
%patch8 -p1
%patch9 -p1
%patch10 -p1
%patch11 -p1
%patch12 -p1
autoreconf -vfi
%build
%configure \
--with-kernel=/usr/src/linux-`uname -r`
%configure
%make
%install
@ -59,23 +92,30 @@ This package contains static libraries and header files need for development.
%files
%defattr(-,root,root)
%{_bindir}/*
%{_libdir}/*.so.*
%{_mandir}/man1/*
%{_libdir}/libaudiofile.so.*
%doc AUTHORS COPYING
%files devel
%defattr(-,root,root)
%{_libdir}/*.a
%{_libdir}/*.la
%{_libdir}/*.so
%{_libdir}/libaudiofile.a
%{_libdir}/libaudiofile.la
%{_libdir}/libaudiofile.so
%{_includedir}/*.h
%{_libdir}/pkgconfig/*
%{_mandir}/man3/*
#%{_datadir}/aclocal/*
%doc ACKNOWLEDGEMENTS ChangeLog NEWS NOTES README TODO
%files tools
%defattr(-,root,root)
%{_bindir}/sfconvert
%{_bindir}/sfinfo
%{_mandir}/man1/*
%changelog
* Sat Dec 12 2020 Silvan Calarco <silvan.calarco@mambasoft.it> 0.3.6-2mamba
- rebuilt with debug package, -tools subpackage and patches from arch linux
* Fri Mar 08 2013 Automatic Build System <autodist@mambasoft.it> 0.3.6-1mamba
- automatic version update by autodist