media-libs/audiofile: Add security patches

Dropping the system-gtest patch is necessary to make the tests run, as
mentioned here: https://bugs.gentoo.org/680482#c8

The three closed bugs are reported test failures fixed by dropping the
aforementioned patch and a slight repair of src_test. Because we're not
using system gtest anymore, we can drop the test dependency on
dev-cpp/gtest, and by extension the IUSE=test boilerplate.

Bug: https://bugs.gentoo.org/614046
Bug: https://bugs.gentoo.org/687766
Closes: https://bugs.gentoo.org/680482
Closes: https://bugs.gentoo.org/715192
Closes: https://bugs.gentoo.org/720836
Package-Manager: Portage-2.3.100, Repoman-2.3.22
Signed-off-by: John Helmert III <jchelmert3@posteo.net>
Closes: https://github.com/gentoo/gentoo/pull/16141
Signed-off-by: Sam James <sam@gentoo.org>

3 files changed, 516 insertions, 0 deletions

diff --git a/media-libs/audiofile/audiofile-0.3.6-r4.ebuild b/media-libs/audiofile/audiofile-0.3.6-r4.ebuild
new file mode 100644
index 000000000000..402fd444e5be
--- /dev/null
+++ b/media-libs/audiofile/audiofile-0.3.6-r4.ebuild
@@ -0,0 +1,55 @@
+# Copyright 1999-2020 Gentoo Authors
+# Distributed under the terms of the GNU General Public License v2
+
+EAPI=6
+
+inherit autotools gnome.org multilib-minimal
+
+DESCRIPTION="An elegant API for accessing audio files"
+HOMEPAGE="http://www.68k.org/~michael/audiofile/"
+
+LICENSE="GPL-2 LGPL-2.1"
+SLOT="0/1" # subslot = soname major version
+KEYWORDS="~alpha ~amd64 ~arm ~arm64 ~hppa ~ia64 ~mips ~ppc ~ppc64 ~sparc ~x86 ~amd64-linux ~x86-linux ~ppc-macos ~x64-macos ~x86-macos ~sparc-solaris ~x86-solaris"
+IUSE="flac"
+
+RDEPEND="flac? ( >=media-libs/flac-1.2.1[${MULTILIB_USEDEP}] )"
+DEPEND="${RDEPEND}
+	virtual/pkgconfig"
+
+PATCHES=(
+	"${FILESDIR}"/${PN}-0.3.6-gcc6-build-fixes.patch
+	"${FILESDIR}"/${PN}-0.3.6-CVE-2015-7747.patch
+	"${FILESDIR}"/${PN}-0.3.6-mingw32.patch
+	"${FILESDIR}"/${PN}-0.3.6-CVE-2017-68xx.patch
+	"${FILESDIR}"/${PN}-0.3.6-CVE-2018-13440-CVE-2018-17095.patch
+)
+
+src_prepare() {
+	default
+	eautoreconf
+}
+
+multilib_src_configure() {
+	# Tests depend on statically compiled binaries to work, so we'll have to
+	# delete them later rather than not compile them at all
+	local myconf=(
+		--enable-largefile
+		--disable-werror
+		--disable-examples
+		$(use_enable flac)
+	)
+	ECONF_SOURCE="${S}" econf "${myconf[@]}"
+}
+
+multilib_src_test() {
+	emake check
+}
+
+multilib_src_install_all() {
+	einstalldocs
+
+	# package provides .pc file
+	find "${ED}" -name '*.la' -delete || die
+	find "${ED}" -name '*.a' -delete || die
+}
diff --git a/media-libs/audiofile/files/audiofile-0.3.6-CVE-2017-68xx.patch b/media-libs/audiofile/files/audiofile-0.3.6-CVE-2017-68xx.patch
new file mode 100644
index 000000000000..99473d7e22ed
--- /dev/null
+++ b/media-libs/audiofile/files/audiofile-0.3.6-CVE-2017-68xx.patch
@@ -0,0 +1,379 @@
+Debian patchset for CVE-2017-68{29..38} and two other vulnerabilities:
+
+https://salsa.debian.org/multimedia-team/audiofile/commit/242f019#a064ca928f514268d4bae308e2e3990138341b76:
+
+* Address several vulnerabilities (Closes: #857651)
+  - Always check the number of coefficients (CVE-2017-6827 CVE-2017-6828
+    CVE-2017-6832 CVE-2017-6833 CVE-2017-6835 CVE-2017-6837)
+  - clamp index values to fix index overflow in IMA.cpp (CVE-2017-6829)
+  - Check for multiplication overflow in sfconvert (CVE-2017-6830
+    CVE-2017-6834 CVE-2017-6836 CVE-2017-6838)
+  - Actually fail when error occurs in parseFormat (CVE-2017-6831)
+  - Check for multiplication overflow in MSADPCM decodeSample
+    (CVE-2017-6839)
+* Fix signature of multiplyCheckOverflow. It returns a bool, not an int
+* Check for division by zero in BlockCodec::runPull
+
+
+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(+)
+
+diff --git a/libaudiofile/WAVE.cpp b/libaudiofile/WAVE.cpp
+index 0e81cf7..d762249 100644
+--- a/libaudiofile/WAVE.cpp
++++ b/libaudiofile/WAVE.cpp
+@@ -326,6 +326,7 @@ 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;
+@@ -333,6 +334,7 @@ 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;
+-- 
+2.11.0
+
+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
+
+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++)
+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
+
+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
+
+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
+
+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
+
diff --git a/media-libs/audiofile/files/audiofile-0.3.6-CVE-2018-13440-CVE-2018-17095.patch b/media-libs/audiofile/files/audiofile-0.3.6-CVE-2018-13440-CVE-2018-17095.patch
new file mode 100644
index 000000000000..0d356fb072a2
--- /dev/null
+++ b/media-libs/audiofile/files/audiofile-0.3.6-CVE-2018-13440-CVE-2018-17095.patch
@@ -0,0 +1,82 @@
+Fixes for CVE-2018-13440 and CVE-2018-17095 from here:
+https://github.com/mpruett/audiofile/pull/52
+
+These are the same used in Fedora.
+
+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);
+ }
+