diff options
| -rw-r--r-- | user/audiofile/APKBUILD | 19 | ||||
| -rw-r--r-- | user/audiofile/CVE-2015-7747.patch | 156 | ||||
| -rw-r--r-- | user/audiofile/CVE-2017-6827,6828,6832,6833,6835,6837.patch | 34 | ||||
| -rw-r--r-- | user/audiofile/CVE-2017-6829.patch | 33 | ||||
| -rw-r--r-- | user/audiofile/CVE-2017-6830,6834,6836,6838.patch | 70 | ||||
| -rw-r--r-- | user/audiofile/CVE-2017-6831.patch | 36 | ||||
| -rw-r--r-- | user/audiofile/CVE-2017-6839.patch | 120 | ||||
| -rw-r--r-- | user/audiofile/CVE-2018-13440,17095.patch | 77 |
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); + } + |