@ -1,73 +0,0 @@ | |||
# | |||
# This is free software, licensed under the GNU General Public License v2. | |||
# See /LICENSE for more information. | |||
# | |||
include $(TOPDIR)/rules.mk | |||
PKG_NAME:=audiofile | |||
PKG_VERSION:=0.3.6 | |||
PKG_RELEASE:=5 | |||
PKG_SOURCE:=$(PKG_NAME)-$(PKG_VERSION).tar.xz | |||
PKG_SOURCE_URL:=@GNOME/$(PKG_NAME)/0.3 | |||
PKG_HASH:=ea2449ad3f201ec590d811db9da6d02ffc5e87a677d06b92ab15363d8cb59782 | |||
PKG_MAINTAINER:=Ted Hess <thess@kitschensync.net> | |||
PKG_CPE_ID:=cpe:/a:audiofile:audiofile | |||
PKG_FIXUP:=autoreconf | |||
PKG_INSTALL=1 | |||
include $(INCLUDE_DIR)/uclibc++.mk | |||
include $(INCLUDE_DIR)/package.mk | |||
define Package/libaudiofile | |||
SECTION:=libs | |||
CATEGORY:=Libraries | |||
TITLE:=Audio File library | |||
URL:=http://audiofile.68k.org/ | |||
DEPENDS:=$(CXX_DEPENDS) +libflac | |||
endef | |||
define Package/libaudiofile/description | |||
The audiofile library allows the processing of audio data to and from audio | |||
files of many common formats (currently AIFF, AIFF-C, WAVE, NeXT/Sun, BICS, | |||
FLAC, ALAC, and raw data). | |||
endef | |||
CONFIGURE_ARGS+= \ | |||
--enable-shared \ | |||
--enable-static \ | |||
--disable-docs \ | |||
--disable-coverage \ | |||
--disable-examples | |||
TARGET_CFLAGS+= $(FPIC) | |||
define Build/InstallDev | |||
$(INSTALL_DIR) $(1)/usr/include | |||
$(CP) \ | |||
$(PKG_INSTALL_DIR)/usr/include/{af_vfs,audiofile,aupvlist}.h \ | |||
$(1)/usr/include/ | |||
$(INSTALL_DIR) $(1)/usr/lib | |||
$(CP) \ | |||
$(PKG_INSTALL_DIR)/usr/lib/libaudiofile.{la,a,so*} \ | |||
$(1)/usr/lib/ | |||
$(INSTALL_DIR) $(1)/usr/lib/pkgconfig | |||
$(CP) \ | |||
$(PKG_INSTALL_DIR)/usr/lib/pkgconfig/audiofile.pc \ | |||
$(1)/usr/lib/pkgconfig/ | |||
endef | |||
define Package/libaudiofile/install | |||
$(INSTALL_DIR) $(1)/usr/lib | |||
$(CP) \ | |||
$(PKG_INSTALL_DIR)/usr/lib/libaudiofile.so.* \ | |||
$(1)/usr/lib/ | |||
endef | |||
$(eval $(call BuildPackage,libaudiofile)) |
@ -1,18 +0,0 @@ | |||
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> | |||
{ |
@ -1,25 +0,0 @@ | |||
--- a/configure.ac | |||
+++ b/configure.ac | |||
@@ -159,12 +159,8 @@ AC_CONFIG_FILES([ | |||
audiofile.pc | |||
audiofile-uninstalled.pc | |||
sfcommands/Makefile | |||
- test/Makefile | |||
- gtest/Makefile | |||
- examples/Makefile | |||
libaudiofile/Makefile | |||
libaudiofile/alac/Makefile | |||
libaudiofile/modules/Makefile | |||
- docs/Makefile | |||
Makefile]) | |||
AC_OUTPUT | |||
--- a/Makefile.am | |||
+++ b/Makefile.am | |||
@@ -1,6 +1,6 @@ | |||
## Process this file with automake to produce Makefile.in | |||
-SUBDIRS = gtest libaudiofile sfcommands test examples docs | |||
+SUBDIRS = libaudiofile sfcommands | |||
EXTRA_DIST = \ | |||
ACKNOWLEDGEMENTS \ |
@ -1,19 +0,0 @@ | |||
Description: fix buffer overflow when changing both sample format and | |||
number of channels | |||
Origin: backport, https://github.com/mpruett/audiofile/pull/25 | |||
Bug-Ubuntu: https://bugs.launchpad.net/ubuntu/+source/audiofile/+bug/1502721 | |||
Bug-Debian: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=801102 | |||
Index: audiofile-0.3.6/libaudiofile/modules/ModuleState.cpp | |||
=================================================================== | |||
--- audiofile-0.3.6.orig/libaudiofile/modules/ModuleState.cpp 2015-10-20 08:00:58.036128202 -0400 | |||
+++ audiofile-0.3.6/libaudiofile/modules/ModuleState.cpp 2015-10-20 08:00:58.036128202 -0400 | |||
@@ -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)); |
@ -1,34 +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; | |||
-- | |||
2.11.0 | |||
@ -1,37 +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) | |||
-- | |||
2.11.0 | |||
@ -1,70 +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; | |||
-- | |||
2.11.0 | |||
@ -1,35 +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(+) | |||
--- a/libaudiofile/WAVE.cpp | |||
+++ b/libaudiofile/WAVE.cpp | |||
@@ -332,6 +332,7 @@ status WAVEFile::parseFormat(const Tag & | |||
{ | |||
_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 & | |||
{ | |||
_af_error(AF_BAD_CODEC_CONFIG, | |||
"Invalid samples per block for IMA ADPCM compression"); | |||
+ return AF_FAIL; | |||
} | |||
track->f.sampleWidth = 16; |
@ -1,120 +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++; | |||
-- | |||
2.11.0 | |||
@ -1,40 +0,0 @@ | |||
From ce536d707b8e2a26baca77320398c45238224ca7 Mon Sep 17 00:00:00 2001 | |||
From: Antonio Larrosa <larrosa@kde.org> | |||
Date: Fri, 10 Mar 2017 15:40:02 +0100 | |||
Subject: [PATCH] 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); | |||
-- | |||
2.11.0 | |||
@ -1,21 +0,0 @@ | |||
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++) |