+++ /dev/null
-#
-# 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))
+++ /dev/null
-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>
- {
+++ /dev/null
---- 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 \
+++ /dev/null
-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));
+++ /dev/null
-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
-
+++ /dev/null
-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
-
+++ /dev/null
-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
-
+++ /dev/null
-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;
+++ /dev/null
-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
-
+++ /dev/null
-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
-
+++ /dev/null
-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++)
git.99rst.org / openwrt-packages.git / commitdiff