user/audiofile: multiple CVE patches. fixes #124.

8 files changed, 543 insertions, 2 deletions

diff --git a/user/audiofile/APKBUILD b/user/audiofile/APKBUILD
index 66b00d0c7..904fea5d0 100644
--- a/user/audiofile/APKBUILD
+++ b/user/audiofile/APKBUILD
@@ -2,7 +2,7 @@
 # Maintainer: A. Wilcox <awilfox@adelielinux.org>
 pkgname=audiofile
 pkgver=0.3.6
-pkgrel=0
+pkgrel=1
 pkgdesc="Library for reading and writing audio files in many formats"
 url="https://audiofile.68k.org"
 arch="all"
@@ -13,6 +13,14 @@ makedepends="$depends_dev alsa-lib-dev"
 subpackages="$pkgname-dev $pkgname-doc"
 source="https://audiofile.68k.org/audiofile-$pkgver.tar.gz
 	tests-unsigned-gcc6.patch
+
+	CVE-2015-7747.patch
+	CVE-2017-6827,6828,6832,6833,6835,6837.patch
+	CVE-2017-6829.patch
+	CVE-2017-6830,6834,6836,6838.patch
+	CVE-2017-6831.patch
+	CVE-2017-6839.patch
+	CVE-2018-13440,17095.patch
 	"
 
 prepare() {
@@ -41,4 +49,11 @@ package() {
 }
 
 sha512sums="f9a1182d93e405c21eba79c5cc40962347bff13f1b3b732d9a396e3d1675297515188bd6eb43033aaa00e9bde74ff4628c1614462456529cabba464f03c1d5fa  audiofile-0.3.6.tar.gz
-b85546eaccf0911fb2f5ba26c323ea2c0b18cae7d7476d5ac9e821c90097f331e3391b787f390d130382fffb6b777887344f914489ca1bc0a3974f1ebe16e328  tests-unsigned-gcc6.patch"
+b85546eaccf0911fb2f5ba26c323ea2c0b18cae7d7476d5ac9e821c90097f331e3391b787f390d130382fffb6b777887344f914489ca1bc0a3974f1ebe16e328  tests-unsigned-gcc6.patch
+7673ab3fafdb0dac514a42622f53ea17aa56836c76413e5680c475537e195c53df21f26da1bd4e7941df2dc8b33a471ab52d539dabffbaef8bc95ee59951e7fe  CVE-2015-7747.patch
+880392c89e0f5982ceb3f56b32da7f28217df64dedf588eb2a72616367f77813b34f3f12092d0b05818247da8e411de8fded336fe09413e81184e98546489b3a  CVE-2017-6827,6828,6832,6833,6835,6837.patch
+e7afe1a27566fb593ea53176256df23e447a2ee842cb4168930dec365fdabe7f2f43512d81bca5f14336ef0c756f6006c24948a3c2d79baafb0042ed8a145aae  CVE-2017-6829.patch
+3b870602fab366015de6dc2f3cf83eae90a4113e2c18e843494c3a75744730f41aefd99547ea758cef194d921a7fc66024b1b9ffbf0bbe6614693453cce09e99  CVE-2017-6830,6834,6836,6838.patch
+51c92ce66e987ae1d4bda65247134097705ef45cf7670401af7943bf6bbfc674089bcfafa49983046b10573ea72900adb96c296739c234d5e98539098eebe022  CVE-2017-6831.patch
+88603061ffe607910d0c73b88c2a305134d6ea4d51dadb6a33163d24c7dd3ab091e7f32c8549fec442c4898a2d6ba91da209973d002c47342645d82d3957a0f0  CVE-2017-6839.patch
+b90684b8e8082acd84f40ec8da83a6f2a2280e71be873055829d4555377454797446f4f5f77c7a9cda7aa4450f1647370a16e2284b5b7777eb86da4ff7d1e336  CVE-2018-13440,17095.patch"
diff --git a/user/audiofile/CVE-2015-7747.patch b/user/audiofile/CVE-2015-7747.patch
new file mode 100644
index 000000000..332563959
--- /dev/null
+++ b/user/audiofile/CVE-2015-7747.patch
@@ -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);
++}
diff --git a/user/audiofile/CVE-2017-6827,6828,6832,6833,6835,6837.patch b/user/audiofile/CVE-2017-6827,6828,6832,6833,6835,6837.patch
new file mode 100644
index 000000000..0465a1f59
--- /dev/null
+++ b/user/audiofile/CVE-2017-6827,6828,6832,6833,6835,6837.patch
@@ -0,0 +1,34 @@
+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 1/3] 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;
+ 
+-- 
+2.36.1
+
diff --git a/user/audiofile/CVE-2017-6829.patch b/user/audiofile/CVE-2017-6829.patch
new file mode 100644
index 000000000..c1047af06
--- /dev/null
+++ b/user/audiofile/CVE-2017-6829.patch
@@ -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)
diff --git a/user/audiofile/CVE-2017-6830,6834,6836,6838.patch b/user/audiofile/CVE-2017-6830,6834,6836,6838.patch
new file mode 100644
index 000000000..eae11b27b
--- /dev/null
+++ b/user/audiofile/CVE-2017-6830,6834,6836,6838.patch
@@ -0,0 +1,70 @@
+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 3/3] 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;
+-- 
+2.36.1
+
diff --git a/user/audiofile/CVE-2017-6831.patch b/user/audiofile/CVE-2017-6831.patch
new file mode 100644
index 000000000..50cd3dc9a
--- /dev/null
+++ b/user/audiofile/CVE-2017-6831.patch
@@ -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;
diff --git a/user/audiofile/CVE-2017-6839.patch b/user/audiofile/CVE-2017-6839.patch
new file mode 100644
index 000000000..ff0cdc2b3
--- /dev/null
+++ b/user/audiofile/CVE-2017-6839.patch
@@ -0,0 +1,120 @@
+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 2/3] 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++;
+-- 
+2.36.1
+
diff --git a/user/audiofile/CVE-2018-13440,17095.patch b/user/audiofile/CVE-2018-13440,17095.patch
new file mode 100644
index 000000000..b35ce980e
--- /dev/null
+++ b/user/audiofile/CVE-2018-13440,17095.patch
@@ -0,0 +1,77 @@
+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 1/3] 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);
+
+From 941774c8c0e79007196d7f1e7afdc97689f869b3 Mon Sep 17 00:00:00 2001
+From: Wim Taymans <wtaymans@redhat.com>
+Date: Thu, 27 Sep 2018 12:09:45 +0200
+Subject: [PATCH 2/3] ALAC: set chunk frameCount to 0 on short read
+
+---
+ libaudiofile/modules/ALAC.cpp | 1 +
+ 1 file changed, 1 insertion(+)
+
+diff --git a/libaudiofile/modules/ALAC.cpp b/libaudiofile/modules/ALAC.cpp
+index 7593c11..478e2af 100644
+--- a/libaudiofile/modules/ALAC.cpp
++++ b/libaudiofile/modules/ALAC.cpp
+@@ -240,6 +240,7 @@ void ALAC::runPull()
+ 	if (read(m_inChunk->buffer, bytesPerPacket) < bytesPerPacket)
+ 	{
+ 		reportReadError(0, m_track->f.framesPerPacket);
++		m_outChunk->frameCount = 0;
+ 		return;
+ 	}
+ 
+
+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 3/3] 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);
+ }
+