diff options
Diffstat (limited to 'gnu/packages')
14 files changed, 1057 insertions, 0 deletions
diff --git a/gnu/packages/audio.scm b/gnu/packages/audio.scm index 87d6947657..76c59f33cc 100644 --- a/gnu/packages/audio.scm +++ b/gnu/packages/audio.scm @@ -26,6 +26,7 @@ ;;; Copyright © 2019 Alexandros Theodotou <alex@zrythm.org> ;;; Copyright © 2019 Christopher Lemmer Webber <cwebber@dustycloud.org> ;;; Copyright © 2019 Jan Wielkiewicz <tona_kosmicznego_smiecia@interia.pl> +;;; Copyright © 2019 Hartmt Goebel <h.goebel@crazy-compilers.com> ;;; ;;; This file is part of GNU Guix. ;;; @@ -467,6 +468,54 @@ and editing digital audio. It features digital effects and spectrum analysis tools.") (license license:gpl2+))) +(define-public audiofile + (package + (name "audiofile") + (version "0.3.6") + (source + (origin + (method url-fetch) + (uri (string-append + "https://audiofile.68k.org/audiofile-" version ".tar.gz")) + (sha256 + (base32 "0rb927zknk9kmhprd8rdr4azql4gn2dp75a36iazx2xhkbqhvind")) + (patches + ;; CVE references according to nixpgs + (search-patches + "audiofile-fix-datatypes-in-tests.patch" + "audiofile-fix-sign-conversion.patch" + "audiofile-hurd.patch" + "audiofile-CVE-2015-7747.patch" + ;; CVE-2017-6829: + "audiofile-Fix-index-overflow-in-IMA.cpp.patch" + ;; CVE-2017-6827, CVE-2017-6828, CVE-2017-6832, CVE-2017-6835, + ;; CVE-2017-6837: + "audiofile-Check-the-number-of-coefficients.patch" + ;; CVE-2017-6839: + "audiofile-Fix-overflow-in-MSADPCM-decodeSam.patch" + ;; CVE-2017-6830, CVE-2017-6834, CVE-2017-6836, CVE-2017-6838: + "audiofile-Fix-multiply-overflow-sfconvert.patch" + "audiofile-signature-of-multiplyCheckOverflow.patch" + ;; CVE-2017-6831: + "audiofile-Fail-on-error-in-parseFormat.patch" + ;; CVE-2017-6833: + "audiofile-division-by-zero-BlockCodec-runPull.patch" + "audiofile-CVE-2018-13440.patch" + "audiofile-CVE-2018-17095.patch")))) + (build-system gnu-build-system) + (inputs + `(("alsa-lib" ,alsa-lib))) + (home-page "https://audiofile.68k.org/") + (synopsis "Library to handle various audio file formats") + (description "This is an open-source version of SGI's audiofile library. +It provides a uniform programming interface for processing of audio data to +and from audio files of many common formats. + +Currently supported file formats include AIFF/AIFF-C, WAVE, and NeXT/Sun +.snd/.au, BICS, and raw data. Supported compression formats are currently +G.711 mu-law and A-law.") + (license license:lgpl2.1+))) + (define-public autotalent (package (name "autotalent") diff --git a/gnu/packages/patches/audiofile-CVE-2015-7747.patch b/gnu/packages/patches/audiofile-CVE-2015-7747.patch new file mode 100644 index 0000000000..3325639591 --- /dev/null +++ b/gnu/packages/patches/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/gnu/packages/patches/audiofile-CVE-2018-13440.patch b/gnu/packages/patches/audiofile-CVE-2018-13440.patch new file mode 100644 index 0000000000..ffd65b43b0 --- /dev/null +++ b/gnu/packages/patches/audiofile-CVE-2018-13440.patch @@ -0,0 +1,28 @@ +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] 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); diff --git a/gnu/packages/patches/audiofile-CVE-2018-17095.patch b/gnu/packages/patches/audiofile-CVE-2018-17095.patch new file mode 100644 index 0000000000..231021b9fc --- /dev/null +++ b/gnu/packages/patches/audiofile-CVE-2018-17095.patch @@ -0,0 +1,26 @@ +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] 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); + } + diff --git a/gnu/packages/patches/audiofile-Check-the-number-of-coefficients.patch b/gnu/packages/patches/audiofile-Check-the-number-of-coefficients.patch new file mode 100644 index 0000000000..f9427cbe61 --- /dev/null +++ b/gnu/packages/patches/audiofile-Check-the-number-of-coefficients.patch @@ -0,0 +1,30 @@ +From: Antonio Larrosa <larrosa@kde.org> +Date: Mon, 6 Mar 2017 12:51:22 +0100 +Subject: 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 9dd8511..0fc48e8 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; + diff --git a/gnu/packages/patches/audiofile-Fail-on-error-in-parseFormat.patch b/gnu/packages/patches/audiofile-Fail-on-error-in-parseFormat.patch new file mode 100644 index 0000000000..50cd3dc9a3 --- /dev/null +++ b/gnu/packages/patches/audiofile-Fail-on-error-in-parseFormat.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/gnu/packages/patches/audiofile-Fix-index-overflow-in-IMA.cpp.patch b/gnu/packages/patches/audiofile-Fix-index-overflow-in-IMA.cpp.patch new file mode 100644 index 0000000000..c1047af06c --- /dev/null +++ b/gnu/packages/patches/audiofile-Fix-index-overflow-in-IMA.cpp.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/gnu/packages/patches/audiofile-Fix-multiply-overflow-sfconvert.patch b/gnu/packages/patches/audiofile-Fix-multiply-overflow-sfconvert.patch new file mode 100644 index 0000000000..0f17140d6b --- /dev/null +++ b/gnu/packages/patches/audiofile-Fix-multiply-overflow-sfconvert.patch @@ -0,0 +1,66 @@ +From: Antonio Larrosa <larrosa@kde.org> +Date: Mon, 6 Mar 2017 13:54:52 +0100 +Subject: 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; diff --git a/gnu/packages/patches/audiofile-Fix-overflow-in-MSADPCM-decodeSam.patch b/gnu/packages/patches/audiofile-Fix-overflow-in-MSADPCM-decodeSam.patch new file mode 100644 index 0000000000..2be930b924 --- /dev/null +++ b/gnu/packages/patches/audiofile-Fix-overflow-in-MSADPCM-decodeSam.patch @@ -0,0 +1,116 @@ +From: Antonio Larrosa <larrosa@kde.org> +Date: Mon, 6 Mar 2017 13:43:53 +0100 +Subject: 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++; diff --git a/gnu/packages/patches/audiofile-division-by-zero-BlockCodec-runPull.patch b/gnu/packages/patches/audiofile-division-by-zero-BlockCodec-runPull.patch new file mode 100644 index 0000000000..e001133916 --- /dev/null +++ b/gnu/packages/patches/audiofile-division-by-zero-BlockCodec-runPull.patch @@ -0,0 +1,21 @@ +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++) diff --git a/gnu/packages/patches/audiofile-fix-datatypes-in-tests.patch b/gnu/packages/patches/audiofile-fix-datatypes-in-tests.patch new file mode 100644 index 0000000000..00e0f3c4a3 --- /dev/null +++ b/gnu/packages/patches/audiofile-fix-datatypes-in-tests.patch @@ -0,0 +1,54 @@ +Based on (hunks for changelog and Identy.cpp removed) +From ecbc07f0ed336187cc9a67c3363f89681b8b8f52 Mon Sep 17 00:00:00 2001 +From: Michael Pruett <michael@68k.org> +Date: Tue, 5 Jul 2016 23:26:16 -0500 +Subject: [PATCH] Fix type of test data arrays. + + + + +--- + ChangeLog | 6 ++++++ + test/Identify.cpp | 3 ++- + test/NeXT.cpp | 7 ++++--- + 3 files changed, 12 insertions(+), 4 deletions(-) + +diff --git a/test/NeXT.cpp b/test/NeXT.cpp +index 7e39850..29af877 100644 +--- a/test/NeXT.cpp ++++ b/test/NeXT.cpp +@@ -30,6 +30,7 @@ + #include <audiofile.h> + #include <fcntl.h> + #include <gtest/gtest.h> ++#include <stdint.h> + #include <sys/stat.h> + #include <sys/types.h> + #include <unistd.h> +@@ -37,7 +38,7 @@ + + #include "TestUtilities.h" + +-const char kDataUnspecifiedLength[] = ++const uint8_t kDataUnspecifiedLength[] = + { + '.', 's', 'n', 'd', + 0, 0, 0, 24, // offset of 24 bytes +@@ -57,7 +58,7 @@ const char kDataUnspecifiedLength[] = + 0, 55 + }; + +-const char kDataTruncated[] = ++const uint8_t kDataTruncated[] = + { + '.', 's', 'n', 'd', + 0, 0, 0, 24, // offset of 24 bytes +@@ -152,7 +153,7 @@ TEST(NeXT, Truncated) + ASSERT_EQ(::unlink(testFileName.c_str()), 0); + } + +-const char kDataZeroChannels[] = ++const uint8_t kDataZeroChannels[] = + { + '.', 's', 'n', 'd', + 0, 0, 0, 24, // offset of 24 bytes diff --git a/gnu/packages/patches/audiofile-fix-sign-conversion.patch b/gnu/packages/patches/audiofile-fix-sign-conversion.patch new file mode 100644 index 0000000000..648161d620 --- /dev/null +++ b/gnu/packages/patches/audiofile-fix-sign-conversion.patch @@ -0,0 +1,26 @@ +Based on (hunk for changelog removed) +From b62c902dd258125cac86cd2df21fc898035a43d3 Mon Sep 17 00:00:00 2001 +From: Michael Pruett <michael@68k.org> +Date: Mon, 29 Aug 2016 23:08:26 -0500 +Subject: [PATCH] Fix undefined behavior in sign conversion. + + +--- + ChangeLog | 5 +++++ + libaudiofile/modules/SimpleModule.h | 3 ++- + 2 files changed, 7 insertions(+), 1 deletion(-) + +diff --git a/libaudiofile/modules/SimpleModule.h b/libaudiofile/modules/SimpleModule.h +index 03c6c69..bad85ad 100644 +--- a/libaudiofile/modules/SimpleModule.h ++++ b/libaudiofile/modules/SimpleModule.h +@@ -123,7 +123,8 @@ 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 kMaxSignedValue = (((1 << (kScaleBits - 1)) - 1) << 1) + 1; ++ static const int kMinSignedValue = -kMaxSignedValue - 1; + + struct signedToUnsigned : public std::unary_function<SignedType, UnsignedType> + { diff --git a/gnu/packages/patches/audiofile-hurd.patch b/gnu/packages/patches/audiofile-hurd.patch new file mode 100644 index 0000000000..b5941dcf44 --- /dev/null +++ b/gnu/packages/patches/audiofile-hurd.patch @@ -0,0 +1,381 @@ +Description: Remove usage of PATH_MAX in tests to fix FTBFS on Hurd. + jcowgill: Removed Changelog changes +Author: Pino Toscano <toscano.pino@tiscali.it> +Origin: backport, https://github.com/mpruett/audiofile/commit/34c261034f1193a783196618f0052112e00fbcfe +Bug: https://github.com/mpruett/audiofile/pull/17 +Bug-Debian: https://bugs.debian.org/762595 +--- +This patch header follows DEP-3: http://dep.debian.net/deps/dep3/ + +--- a/test/TestUtilities.cpp ++++ b/test/TestUtilities.cpp +@@ -21,8 +21,8 @@ + #include "TestUtilities.h" + + #include <limits.h> +-#include <stdio.h> + #include <stdlib.h> ++#include <string.h> + #include <unistd.h> + + bool createTemporaryFile(const std::string &prefix, std::string *path) +@@ -35,12 +35,12 @@ bool createTemporaryFile(const std::stri + return true; + } + +-bool createTemporaryFile(const char *prefix, char *path) ++bool createTemporaryFile(const char *prefix, char **path) + { +- snprintf(path, PATH_MAX, "/tmp/%s-XXXXXX", prefix); +- int fd = ::mkstemp(path); +- if (fd < 0) +- return false; +- ::close(fd); +- return true; ++ *path = NULL; ++ std::string pathString; ++ bool result = createTemporaryFile(prefix, &pathString); ++ if (result) ++ *path = ::strdup(pathString.c_str()); ++ return result; + } +--- a/test/TestUtilities.h ++++ b/test/TestUtilities.h +@@ -53,7 +53,7 @@ extern "C" { + + #include <stdbool.h> + +-bool createTemporaryFile(const char *prefix, char *path); ++bool createTemporaryFile(const char *prefix, char **path); + + #ifdef __cplusplus + } +--- a/test/floatto24.c ++++ b/test/floatto24.c +@@ -86,8 +86,8 @@ int main (int argc, char **argv) + afInitChannels(setup, AF_DEFAULT_TRACK, 1); + afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_FLOAT, 32); + +- char testFileName[PATH_MAX]; +- if (!createTemporaryFile("floatto24", testFileName)) ++ char *testFileName; ++ if (!createTemporaryFile("floatto24", &testFileName)) + { + fprintf(stderr, "Could not create temporary file.\n"); + exit(EXIT_FAILURE); +@@ -182,6 +182,7 @@ int main (int argc, char **argv) + } + + unlink(testFileName); ++ free(testFileName); + + exit(EXIT_SUCCESS); + } +--- a/test/sixteen-to-eight.c ++++ b/test/sixteen-to-eight.c +@@ -57,8 +57,8 @@ int main (int argc, char **argv) + afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_UNSIGNED, 8); + afInitChannels(setup, AF_DEFAULT_TRACK, 1); + +- char testFileName[PATH_MAX]; +- if (!createTemporaryFile("sixteen-to-eight", testFileName)) ++ char *testFileName; ++ if (!createTemporaryFile("sixteen-to-eight", &testFileName)) + { + fprintf(stderr, "Could not create temporary file.\n"); + exit(EXIT_FAILURE); +@@ -113,6 +113,7 @@ int main (int argc, char **argv) + + afCloseFile(file); + unlink(testFileName); ++ free(testFileName); + + exit(EXIT_SUCCESS); + } +--- a/test/testchannelmatrix.c ++++ b/test/testchannelmatrix.c +@@ -39,7 +39,7 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + const short samples[] = {300, -300, 515, -515, 2315, -2315, 9154, -9154}; + #define SAMPLE_COUNT (sizeof (samples) / sizeof (short)) +@@ -47,7 +47,11 @@ const short samples[] = {300, -300, 515, + + void cleanup (void) + { +- unlink(sTestFileName); ++ if (sTestFileName) ++ { ++ unlink(sTestFileName); ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -76,7 +80,7 @@ int main (void) + afInitFileFormat(setup, AF_FILE_AIFFC); + + /* Write stereo data to test file. */ +- ensure(createTemporaryFile("testchannelmatrix", sTestFileName), ++ ensure(createTemporaryFile("testchannelmatrix", &sTestFileName), + "could not create temporary file"); + file = afOpenFile(sTestFileName, "w", setup); + ensure(file != AF_NULL_FILEHANDLE, "could not open file for writing"); +--- a/test/testdouble.c ++++ b/test/testdouble.c +@@ -38,7 +38,7 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + const double samples[] = + {1.0, 0.6, -0.3, 0.95, 0.2, -0.6, 0.9, 0.4, -0.22, 0.125, 0.1, -0.4}; +@@ -48,7 +48,11 @@ void testdouble (int fileFormat); + + void cleanup (void) + { +- unlink(sTestFileName); ++ if (sTestFileName) ++ { ++ unlink(sTestFileName); ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -96,7 +100,7 @@ void testdouble (int fileFormat) + afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_DOUBLE, 64); + afInitChannels(setup, AF_DEFAULT_TRACK, 2); + +- ensure(createTemporaryFile("testdouble", sTestFileName), ++ ensure(createTemporaryFile("testdouble", &sTestFileName), + "could not create temporary file"); + file = afOpenFile(sTestFileName, "w", setup); + ensure(file != AF_NULL_FILEHANDLE, "could not open file for writing"); +--- a/test/testfloat.c ++++ b/test/testfloat.c +@@ -38,7 +38,7 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + const float samples[] = + {1.0, 0.6, -0.3, 0.95, 0.2, -0.6, 0.9, 0.4, -0.22, 0.125, 0.1, -0.4}; +@@ -48,7 +48,11 @@ void testfloat (int fileFormat); + + void cleanup (void) + { +- unlink(sTestFileName); ++ if (sTestFileName) ++ { ++ unlink(sTestFileName); ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -96,7 +100,7 @@ void testfloat (int fileFormat) + afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_FLOAT, 32); + afInitChannels(setup, AF_DEFAULT_TRACK, 2); + +- ensure(createTemporaryFile("testfloat", sTestFileName), ++ ensure(createTemporaryFile("testfloat", &sTestFileName), + "could not create temporary file"); + file = afOpenFile(sTestFileName, "w", setup); + ensure(file != AF_NULL_FILEHANDLE, "could not open file for writing"); +--- a/test/testmarkers.c ++++ b/test/testmarkers.c +@@ -32,15 +32,19 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + #define FRAME_COUNT 200 + + void cleanup (void) + { ++ if (sTestFileName) ++ { + #ifndef DEBUG +- unlink(sTestFileName); ++ unlink(sTestFileName); + #endif ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -127,7 +131,7 @@ int testmarkers (int fileformat) + + int main (void) + { +- ensure(createTemporaryFile("testmarkers", sTestFileName), ++ ensure(createTemporaryFile("testmarkers", &sTestFileName), + "could not create temporary file"); + + testmarkers(AF_FILE_AIFF); +--- a/test/twentyfour.c ++++ b/test/twentyfour.c +@@ -71,8 +71,8 @@ int main (int argc, char **argv) + afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 24); + afInitChannels(setup, AF_DEFAULT_TRACK, 1); + +- char testFileName[PATH_MAX]; +- if (!createTemporaryFile("twentyfour", testFileName)) ++ char *testFileName; ++ if (!createTemporaryFile("twentyfour", &testFileName)) + { + fprintf(stderr, "could not create temporary file\n"); + exit(EXIT_FAILURE); +@@ -239,6 +239,7 @@ int main (int argc, char **argv) + exit(EXIT_FAILURE); + } + unlink(testFileName); ++ free(testFileName); + + exit(EXIT_SUCCESS); + } +--- a/test/twentyfour2.c ++++ b/test/twentyfour2.c +@@ -45,15 +45,19 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + #define FRAME_COUNT 10000 + + void cleanup (void) + { ++ if (sTestFileName) ++ { + #ifndef DEBUG +- unlink(sTestFileName); ++ unlink(sTestFileName); + #endif ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -78,7 +82,7 @@ int main (void) + afInitChannels(setup, AF_DEFAULT_TRACK, 1); + afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 24); + +- ensure(createTemporaryFile("twentyfour2", sTestFileName), ++ ensure(createTemporaryFile("twentyfour2", &sTestFileName), + "could not create temporary file"); + file = afOpenFile(sTestFileName, "w", setup); + ensure(file != NULL, "could not open test file for writing"); +--- a/test/writealaw.c ++++ b/test/writealaw.c +@@ -53,7 +53,7 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + #define FRAME_COUNT 16 + #define SAMPLE_COUNT FRAME_COUNT +@@ -62,9 +62,13 @@ void testalaw (int fileFormat); + + void cleanup (void) + { ++ if (sTestFileName) ++ { + #ifndef DEBUG +- unlink(sTestFileName); ++ unlink(sTestFileName); + #endif ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -113,7 +117,7 @@ void testalaw (int fileFormat) + afInitFileFormat(setup, fileFormat); + afInitChannels(setup, AF_DEFAULT_TRACK, 1); + +- ensure(createTemporaryFile("writealaw", sTestFileName), ++ ensure(createTemporaryFile("writealaw", &sTestFileName), + "could not create temporary file"); + file = afOpenFile(sTestFileName, "w", setup); + afFreeFileSetup(setup); +--- a/test/writeraw.c ++++ b/test/writeraw.c +@@ -44,13 +44,17 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + void cleanup (void) + { ++ if (sTestFileName) ++ { + #ifndef DEBUG +- unlink(sTestFileName); ++ unlink(sTestFileName); + #endif ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -84,7 +88,7 @@ int main (int argc, char **argv) + afInitChannels(setup, AF_DEFAULT_TRACK, 1); + afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16); + +- ensure(createTemporaryFile("writeraw", sTestFileName), ++ ensure(createTemporaryFile("writeraw", &sTestFileName), + "could not create temporary file"); + file = afOpenFile(sTestFileName, "w", setup); + ensure(file != AF_NULL_FILEHANDLE, "unable to open file for writing"); +--- a/test/writeulaw.c ++++ b/test/writeulaw.c +@@ -53,7 +53,7 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + #define FRAME_COUNT 16 + #define SAMPLE_COUNT FRAME_COUNT +@@ -62,9 +62,13 @@ void testulaw (int fileFormat); + + void cleanup (void) + { ++ if (sTestFileName) ++ { + #ifndef DEBUG +- unlink(sTestFileName); ++ unlink(sTestFileName); + #endif ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -113,7 +117,7 @@ void testulaw (int fileFormat) + afInitFileFormat(setup, fileFormat); + afInitChannels(setup, AF_DEFAULT_TRACK, 1); + +- ensure(createTemporaryFile("writeulaw", sTestFileName), ++ ensure(createTemporaryFile("writeulaw", &sTestFileName), + "could not create temporary file"); + file = afOpenFile(sTestFileName, "w", setup); + afFreeFileSetup(setup); diff --git a/gnu/packages/patches/audiofile-signature-of-multiplyCheckOverflow.patch b/gnu/packages/patches/audiofile-signature-of-multiplyCheckOverflow.patch new file mode 100644 index 0000000000..35627d3869 --- /dev/null +++ b/gnu/packages/patches/audiofile-signature-of-multiplyCheckOverflow.patch @@ -0,0 +1,35 @@ +From: Antonio Larrosa <larrosa@kde.org> +Date: Fri, 10 Mar 2017 15:40:02 +0100 +Subject: 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); |