#
-# Copyright (C) 2006-2016 OpenWrt.org
-#
# This is free software, licensed under the GNU General Public License v2.
# See /LICENSE for more information.
#
PKG_NAME:=audiofile
PKG_VERSION:=0.3.6
-PKG_RELEASE:=3
+PKG_RELEASE:=4
PKG_SOURCE:=$(PKG_NAME)-$(PKG_VERSION).tar.xz
PKG_SOURCE_URL:=@GNOME/$(PKG_NAME)/0.3
CONFIGURE_ARGS+= \
--enable-shared \
--enable-static \
- --disable-examples \
- --with-build-cc="$(HOSTCC)"
+ --disable-docs \
+ --disable-coverage \
+ --disable-examples
+
TARGET_CFLAGS+= $(FPIC)
--- /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++)
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