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package/audiofile: drop package

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The audiofile package is affected by multiple CVEs and is not maintained
anymore (no release since 2013):

  https://nvd.nist.gov/vuln/search/results?form_type=Advanced&results_type=overview&seach_type=all&query=cpe:2.3:a:audio_file_library_project:audio_file_library:0.3.6:*:*:*:*:*:*:*

Signed-off-by: Fabrice Fontaine <fontaine.fabrice@gmail.com>
Signed-off-by: Peter Korsgaard <peter@korsgaard.com>
This commit is contained in:
Fabrice Fontaine
2021-02-07 21:27:18 +01:00
committed by Peter Korsgaard
parent 4c69951791
commit 8d93a30066
16 changed files with 19 additions and 821 deletions
Download Patch File Download Diff File Expand all files Collapse all files

16
Config.in.legacy
View File

@@ -146,6 +146,22 @@ endif
comment "Legacy options removed in 2021.02" comment "Legacy options removed in 2021.02"
config BR2_PACKAGE_MPD_AUDIOFILE
bool "mpd audiofile support removed"
select BR2_LEGACY
help
The audiofile support was removed from mpd as audiofile is
affected by multiple CVEs and is not maintained anymore (no
release since 2013).
config BR2_PACKAGE_AUDIOFILE
bool "audiofile package removed"
select BR2_LEGACY
help
The audiofile package was removed as it is affected by
multiple CVEs and is not maintained anymore (no release since
2013).
config BR2_BINUTILS_VERSION_2_33_X config BR2_BINUTILS_VERSION_2_33_X
bool "binutils 2.33.x has been removed" bool "binutils 2.33.x has been removed"
select BR2_LEGACY select BR2_LEGACY

1
package/Config.in
View File

@@ -1281,7 +1281,6 @@ menu "Audio/Sound"
source "package/alsa-lib/Config.in" source "package/alsa-lib/Config.in"
source "package/alure/Config.in" source "package/alure/Config.in"
source "package/aubio/Config.in" source "package/aubio/Config.in"
source "package/audiofile/Config.in"
source "package/bcg729/Config.in" source "package/bcg729/Config.in"
source "package/caps/Config.in" source "package/caps/Config.in"
source "package/fdk-aac/Config.in" source "package/fdk-aac/Config.in"

56
package/audiofile/0001-Fix-pkg-config-for-static-linking.patch
View File

@@ -1,56 +0,0 @@
From 2abf7d2e5c533bf4d7407c2c8057a329cd49a3cd Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?J=C3=B6rg=20Krause?= <joerg.krause@embedded.rocks>
Date: Tue, 24 Nov 2015 21:57:27 +0100
Subject: [PATCH 1/1] Fix pkg-config for static linking
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Static linking userspace programs such as MPD against libaudiofile fails if
FLAC is available, because libaudiofile is linked against FLAC, but this isn't
expressed in the pkg-config file:
[..]
arm-buildroot-linux-uclibcgnueabi/sysroot/usr/lib/libaudiofile.a(FLAC.o): In function `FLACDecoder::reset2()':
FLAC.cpp:(.text+0x58): undefined reference to `FLAC__stream_decoder_seek_absolute'
/home/buildroot/build/instance-1/output/host/usr/arm-buildroot-linux-uclibcgnueabi/sysroot/usr/lib/libaudiofile.a(FLAC.o): In function `FLACEncoder::sync2()':
FLAC.cpp:(.text+0x88): undefined reference to `FLAC__stream_encoder_finish'
/home/buildroot/build/instance-1/output/host/usr/arm-buildroot-linux-uclibcgnueabi/sysroot/usr/lib/libaudiofile.a(FLAC.o): In function `FLACDecoder::~FLACDecoder()':
FLAC.cpp:(.text+0xc4): undefined reference to `FLAC__stream_decoder_delete'
/home/buildroot/build/instance-1/output/host/usr/arm-buildroot-linux-uclibcgnueabi/sysroot/usr/lib/libaudiofile.a(FLAC.o): In function `FLACEncoder::~FLACEncoder()':
FLAC.cpp:(.text+0x164): undefined reference to `FLAC__stream_encoder_delete'
/home/buildroot/build/instance-1/output/host/usr/arm-buildroot-linux-uclibcgnueabi/sysroot/usr/lib/libaudiofile.a(FLAC.o): In function `FLACDecoder::runPull()':
[..]
The Libs.private field is specifically designed for such usage:
From pkg-config documentation:
Libs.private:
This line should list any private libraries in use. Private
libraries are libraries which are not exposed through your
library, but are needed in the case of static linking.
Therefore, this patch adds a reference to FLAC as well as to lcov in the
Libs.private field of the pkg-config file.
Signed-off-by: Jörg Krause <joerg.krause@embedded.rocks>
---
audiofile.pc.in | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/audiofile.pc.in b/audiofile.pc.in
index ad5956a..d6055ef 100644
--- a/audiofile.pc.in
+++ b/audiofile.pc.in
@@ -8,5 +8,5 @@ Description: audiofile
Requires:
Version: @VERSION@
Libs: -L${libdir} -laudiofile
-Libs.private: -lm
+Libs.private: @FLAC_LIBS@ @COVERAGE_LIBS@ -lm
Cflags: -I${includedir}
--
2.6.2

28
package/audiofile/0002-cast-to-unsigned-gcc6.patch
View File

@@ -1,28 +0,0 @@
From 28cfdbbcb96a69087c3d21faf69b5eae7bcf6d69 Mon Sep 17 00:00:00 2001
From: Hodorgasm <nsane457@gmail.com>
Date: Wed, 11 May 2016 21:42:07 -0400
Subject: [PATCH] Cast to unsigned while left bit-shifting
GCC-6 now treats the left bitwise-shift of a negative integer as nonconformant so explicitly cast to an unsigned int while bit-shifting.
Downloaded from upstream PR:
https://github.com/mpruett/audiofile/pull/28
Signed-off-by: Bernd Kuhls <bernd.kuhls@t-online.de>
---
libaudiofile/modules/SimpleModule.h | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/libaudiofile/modules/SimpleModule.h b/libaudiofile/modules/SimpleModule.h
index 03c6c69..4014fb2 100644
--- a/libaudiofile/modules/SimpleModule.h
+++ b/libaudiofile/modules/SimpleModule.h
@@ -123,7 +123,7 @@ struct signConverter
typedef typename IntTypes<Format>::UnsignedType UnsignedType;
static const int kScaleBits = (Format + 1) * CHAR_BIT - 1;
- static const int kMinSignedValue = -1 << kScaleBits;
+ static const int kMinSignedValue = static_cast<signed>(static_cast<unsigned>(-1) << kScaleBits);;
struct signedToUnsigned : public std::unary_function<SignedType, UnsignedType>
{

36
package/audiofile/0003-Always-check-the-number-of-coefficients.patch
View File

@@ -1,36 +0,0 @@
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
Signed-off-by: Peter Korsgaard <peter@korsgaard.com>
---
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

39
package/audiofile/0004-clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
View File

@@ -1,39 +0,0 @@
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/)
Signed-off-by: Peter Korsgaard <peter@korsgaard.com>
---
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

72
package/audiofile/0005-Check-for-multiplication-overflow-in-sfconvert.patch
View File

@@ -1,72 +0,0 @@
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
Signed-off-by: Peter Korsgaard <peter@korsgaard.com>
---
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

42
package/audiofile/0006-Actually-fail-when-error-occurs-in-parseFormat.patch
View File

@@ -1,42 +0,0 @@
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/
)
Signed-off-by: Peter Korsgaard <peter@korsgaard.com>
---
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

122
package/audiofile/0007-Check-for-multiplication-overflow-in-MSADPCM-decodeS.patch
View File

@@ -1,122 +0,0 @@
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
Signed-off-by: Peter Korsgaard <peter@korsgaard.com>
---
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

161
package/audiofile/0008-CVE-2015-7747.patch
View File

@@ -1,161 +0,0 @@
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
Downloaded from
https://gitweb.gentoo.org/repo/gentoo.git/tree/media-libs/audiofile/files/audiofile-0.3.6-CVE-2015-7747.patch
Signed-off-by: Bernd Kuhls <bernd.kuhls@t-online.de>
--- 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);
+}

193
package/audiofile/0009-Fix-static-linking-with-libsndfile.patch
View File

@@ -1,193 +0,0 @@
From d89a938f48e97b5770509d53c5478c5c3008d6e8 Mon Sep 17 00:00:00 2001
From: Bernd Kuhls <bernd.kuhls@t-online.de>
Date: Sat, 27 May 2017 17:53:33 +0200
Subject: [PATCH 1/1] Fix static linking with libsndfile
libsndfile and audiofile both contain mixXX functions in their alac
code which lead to symbol name clashes when apps like mpd try to
statically link to both audiofile and libsndfile at the same time.
This patch renames these functions to avoid the problem which was
detected by the buildroot autobuilders:
http://autobuild.buildroot.net/results/799/7997ccd698f03885f98d00bd150dc3a578e4b161/
Patch sent upstream: https://github.com/mpruett/audiofile/pull/45
Signed-off-by: Bernd Kuhls <bernd.kuhls@t-online.de>
---
libaudiofile/alac/ALACEncoder.cpp | 28 ++++++++++++++--------------
libaudiofile/alac/matrix_enc.c | 8 ++++----
libaudiofile/alac/matrixlib.h | 8 ++++----
3 files changed, 22 insertions(+), 22 deletions(-)
diff --git a/libaudiofile/alac/ALACEncoder.cpp b/libaudiofile/alac/ALACEncoder.cpp
index da922c2..3d088cc 100644
--- a/libaudiofile/alac/ALACEncoder.cpp
+++ b/libaudiofile/alac/ALACEncoder.cpp
@@ -332,19 +332,19 @@ int32_t ALACEncoder::EncodeStereo( BitBuffer * bitstream, void * inputBuffer, ui
switch ( mBitDepth )
{
case 16:
- mix16( (int16_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples/dilate, mixBits, mixRes );
+ audiofile_alac_mix16( (int16_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples/dilate, mixBits, mixRes );
break;
case 20:
- mix20( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples/dilate, mixBits, mixRes );
+ audiofile_alac_mix20( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples/dilate, mixBits, mixRes );
break;
case 24:
// includes extraction of shifted-off bytes
- mix24( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples/dilate,
+ audiofile_alac_mix24( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples/dilate,
mixBits, mixRes, mShiftBufferUV, bytesShifted );
break;
case 32:
// includes extraction of shifted-off bytes
- mix32( (int32_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples/dilate,
+ audiofile_alac_mix32( (int32_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples/dilate,
mixBits, mixRes, mShiftBufferUV, bytesShifted );
break;
}
@@ -379,19 +379,19 @@ int32_t ALACEncoder::EncodeStereo( BitBuffer * bitstream, void * inputBuffer, ui
switch ( mBitDepth )
{
case 16:
- mix16( (int16_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples, mixBits, mixRes );
+ audiofile_alac_mix16( (int16_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples, mixBits, mixRes );
break;
case 20:
- mix20( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples, mixBits, mixRes );
+ audiofile_alac_mix20( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples, mixBits, mixRes );
break;
case 24:
// also extracts the shifted off bytes into the shift buffers
- mix24( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples,
+ audiofile_alac_mix24( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples,
mixBits, mixRes, mShiftBufferUV, bytesShifted );
break;
case 32:
// also extracts the shifted off bytes into the shift buffers
- mix32( (int32_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples,
+ audiofile_alac_mix32( (int32_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples,
mixBits, mixRes, mShiftBufferUV, bytesShifted );
break;
}
@@ -605,19 +605,19 @@ int32_t ALACEncoder::EncodeStereoFast( BitBuffer * bitstream, void * inputBuffer
switch ( mBitDepth )
{
case 16:
- mix16( (int16_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples, mixBits, mixRes );
+ audiofile_alac_mix16( (int16_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples, mixBits, mixRes );
break;
case 20:
- mix20( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples, mixBits, mixRes );
+ audiofile_alac_mix20( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples, mixBits, mixRes );
break;
case 24:
// also extracts the shifted off bytes into the shift buffers
- mix24( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples,
+ audiofile_alac_mix24( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples,
mixBits, mixRes, mShiftBufferUV, bytesShifted );
break;
case 32:
// also extracts the shifted off bytes into the shift buffers
- mix32( (int32_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples,
+ audiofile_alac_mix32( (int32_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples,
mixBits, mixRes, mShiftBufferUV, bytesShifted );
break;
}
@@ -756,7 +756,7 @@ int32_t ALACEncoder::EncodeStereoEscape( BitBuffer * bitstream, void * inputBuff
break;
case 20:
// mix20() with mixres param = 0 means de-interleave so use it to simplify things
- mix20( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples, 0, 0 );
+ audiofile_alac_mix20( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples, 0, 0 );
for ( index = 0; index < numSamples; index++ )
{
BitBufferWrite( bitstream, mMixBufferU[index], 20 );
@@ -765,7 +765,7 @@ int32_t ALACEncoder::EncodeStereoEscape( BitBuffer * bitstream, void * inputBuff
break;
case 24:
// mix24() with mixres param = 0 means de-interleave so use it to simplify things
- mix24( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples, 0, 0, mShiftBufferUV, 0 );
+ audiofile_alac_mix24( (uint8_t *) inputBuffer, stride, mMixBufferU, mMixBufferV, numSamples, 0, 0, mShiftBufferUV, 0 );
for ( index = 0; index < numSamples; index++ )
{
BitBufferWrite( bitstream, mMixBufferU[index], 24 );
diff --git a/libaudiofile/alac/matrix_enc.c b/libaudiofile/alac/matrix_enc.c
index e194330..8abd556 100644
--- a/libaudiofile/alac/matrix_enc.c
+++ b/libaudiofile/alac/matrix_enc.c
@@ -57,7 +57,7 @@
// 16-bit routines
-void mix16( int16_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples, int32_t mixbits, int32_t mixres )
+void audiofile_alac_mix16( int16_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples, int32_t mixbits, int32_t mixres )
{
int16_t * ip = in;
int32_t j;
@@ -95,7 +95,7 @@ void mix16( int16_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t num
// 20-bit routines
// - the 20 bits of data are left-justified in 3 bytes of storage but right-aligned for input/output predictor buffers
-void mix20( uint8_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples, int32_t mixbits, int32_t mixres )
+void audiofile_alac_mix20( uint8_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples, int32_t mixbits, int32_t mixres )
{
int32_t l, r;
uint8_t * ip = in;
@@ -140,7 +140,7 @@ void mix20( uint8_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t num
// 24-bit routines
// - the 24 bits of data are right-justified in the input/output predictor buffers
-void mix24( uint8_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples,
+void audiofile_alac_mix24( uint8_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples,
int32_t mixbits, int32_t mixres, uint16_t * shiftUV, int32_t bytesShifted )
{
int32_t l, r;
@@ -240,7 +240,7 @@ void mix24( uint8_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t num
// - otherwise, the calculations might overflow into the 33rd bit and be lost
// - therefore, these routines deal with the specified "unused lower" bytes in the "shift" buffers
-void mix32( int32_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples,
+void audiofile_alac_mix32( int32_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples,
int32_t mixbits, int32_t mixres, uint16_t * shiftUV, int32_t bytesShifted )
{
int32_t * ip = in;
diff --git a/libaudiofile/alac/matrixlib.h b/libaudiofile/alac/matrixlib.h
index 0a4f371..5728b6d 100644
--- a/libaudiofile/alac/matrixlib.h
+++ b/libaudiofile/alac/matrixlib.h
@@ -38,17 +38,17 @@ extern "C" {
#endif
// 16-bit routines
-void mix16( int16_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples, int32_t mixbits, int32_t mixres );
+void audiofile_alac_mix16( int16_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples, int32_t mixbits, int32_t mixres );
void unmix16( int32_t * u, int32_t * v, int16_t * out, uint32_t stride, int32_t numSamples, int32_t mixbits, int32_t mixres );
// 20-bit routines
-void mix20( uint8_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples, int32_t mixbits, int32_t mixres );
+void audiofile_alac_mix20( uint8_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples, int32_t mixbits, int32_t mixres );
void unmix20( int32_t * u, int32_t * v, uint8_t * out, uint32_t stride, int32_t numSamples, int32_t mixbits, int32_t mixres );
// 24-bit routines
// - 24-bit data sometimes compresses better by shifting off the bottom byte so these routines deal with
// the specified "unused lower bytes" in the combined "shift" buffer
-void mix24( uint8_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples,
+void audiofile_alac_mix24( uint8_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples,
int32_t mixbits, int32_t mixres, uint16_t * shiftUV, int32_t bytesShifted );
void unmix24( int32_t * u, int32_t * v, uint8_t * out, uint32_t stride, int32_t numSamples,
int32_t mixbits, int32_t mixres, uint16_t * shiftUV, int32_t bytesShifted );
@@ -57,7 +57,7 @@ void unmix24( int32_t * u, int32_t * v, uint8_t * out, uint32_t stride, int32_t
// - note that these really expect the internal data width to be < 32-bit but the arrays are 32-bit
// - otherwise, the calculations might overflow into the 33rd bit and be lost
// - therefore, these routines deal with the specified "unused lower" bytes in the combined "shift" buffer
-void mix32( int32_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples,
+void audiofile_alac_mix32( int32_t * in, uint32_t stride, int32_t * u, int32_t * v, int32_t numSamples,
int32_t mixbits, int32_t mixres, uint16_t * shiftUV, int32_t bytesShifted );
void unmix32( int32_t * u, int32_t * v, int32_t * out, uint32_t stride, int32_t numSamples,
int32_t mixbits, int32_t mixres, uint16_t * shiftUV, int32_t bytesShifted );
--
2.11.0

11
package/audiofile/Config.in
View File

@@ -1,11 +0,0 @@
config BR2_PACKAGE_AUDIOFILE
bool "audiofile"
depends on BR2_INSTALL_LIBSTDCPP
help
The Audio File Library handles reading and writing audio files
in many common formats.
http://www.68k.org/~michael/audiofile/
comment "audiofile needs a toolchain w/ C++"
depends on !BR2_INSTALL_LIBSTDCPP

4
package/audiofile/audiofile.hash
View File

@@ -1,4 +0,0 @@
# Locally calculated
sha256 cdc60df19ab08bfe55344395739bb08f50fc15c92da3962fac334d3bff116965 audiofile-0.3.6.tar.gz
sha256 dc626520dcd53a22f727af3ee42c770e56c97a64fe3adb063799d8ab032fe551 COPYING
sha256 8177f97513213526df2cf6184d8ff986c675afb514d4e68a404010521b880643 COPYING.GPL

41
package/audiofile/audiofile.mk
View File

@@ -1,41 +0,0 @@
################################################################################
#
# audiofile
#
################################################################################
AUDIOFILE_VERSION = 0.3.6
AUDIOFILE_SITE = http://audiofile.68k.org
AUDIOFILE_INSTALL_STAGING = YES
AUDIOFILE_CONF_ENV = ac_cv_prog_cc_c99='-std=gnu99'
AUDIOFILE_CONF_OPTS = --disable-examples
AUDIOFILE_DEPENDENCIES = host-pkgconf
# configure is outdated and has old bugs because of it
AUDIOFILE_AUTORECONF = YES
AUDIOFILE_LICENSE = GPL-2.0+, LGPL-2.1+
AUDIOFILE_LICENSE_FILES = COPYING COPYING.GPL
# 0003-Always-check-the-number-of-coefficients.patch
AUDIOFILE_IGNORE_CVES += \
CVE-2017-6827 CVE-2017-6828 CVE-2017-6832 \
CVE-2017-6833 CVE-2017-6835 CVE-2017-6837
# 0004-clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
AUDIOFILE_IGNORE_CVES += CVE-2017-6829
# 0005-Check-for-multiplication-overflow-in-sfconvert.patch
AUDIOFILE_IGNORE_CVES += \
CVE-2017-6830 CVE-2017-6834 CVE-2017-6836 CVE-2017-6838
# 0006-Actually-fail-when-error-occurs-in-parseFormat.patch
AUDIOFILE_IGNORE_CVES += CVE-2017-6831
# 0007-Check-for-multiplication-overflow-in-MSADPCM-decodeS.patch
AUDIOFILE_IGNORE_CVES += CVE-2017-6839
# 0008-CVE-2015-7747.patch
AUDIOFILE_IGNORE_CVES += CVE-2015-7747
ifeq ($(BR2_PACKAGE_FLAC),y)
AUDIOFILE_DEPENDENCIES += flac
AUDIOFILE_CONF_OPTS += --enable-flac
else
AUDIOFILE_CONF_OPTS += --disable-flac
endif
$(eval $(autotools-package))

7
package/mpd/Config.in
View File

@@ -84,13 +84,6 @@ config BR2_PACKAGE_MPD_LIBSOXR
comment "Decoder plugins" comment "Decoder plugins"
config BR2_PACKAGE_MPD_AUDIOFILE
bool "audiofile"
select BR2_PACKAGE_AUDIOFILE
help
Enable audiofile input/streaming support.
Select this if you want to play back WAV files.
config BR2_PACKAGE_MPD_DSD config BR2_PACKAGE_MPD_DSD
bool "dsd" bool "dsd"
help help

11
package/mpd/mpd.mk
View File

@@ -11,7 +11,9 @@ MPD_SITE = http://www.musicpd.org/download/mpd/$(MPD_VERSION_MAJOR)
MPD_DEPENDENCIES = host-pkgconf boost MPD_DEPENDENCIES = host-pkgconf boost
MPD_LICENSE = GPL-2.0+ MPD_LICENSE = GPL-2.0+
MPD_LICENSE_FILES = COPYING MPD_LICENSE_FILES = COPYING
MPD_CONF_OPTS = -Ddocumentation=disabled MPD_CONF_OPTS = \
-Daudiofile=disabled \
-Ddocumentation=disabled
# Zeroconf support depends on libdns_sd from avahi. # Zeroconf support depends on libdns_sd from avahi.
ifeq ($(BR2_PACKAGE_MPD_AVAHI_SUPPORT),y) ifeq ($(BR2_PACKAGE_MPD_AVAHI_SUPPORT),y)
@@ -43,13 +45,6 @@ else
MPD_CONF_OPTS += -Dao=disabled MPD_CONF_OPTS += -Dao=disabled
endif endif
ifeq ($(BR2_PACKAGE_MPD_AUDIOFILE),y)
MPD_DEPENDENCIES += audiofile
MPD_CONF_OPTS += -Daudiofile=enabled
else
MPD_CONF_OPTS += -Daudiofile=disabled
endif
ifeq ($(BR2_PACKAGE_MPD_BZIP2),y) ifeq ($(BR2_PACKAGE_MPD_BZIP2),y)
MPD_DEPENDENCIES += bzip2 MPD_DEPENDENCIES += bzip2
MPD_CONF_OPTS += -Dbzip2=enabled MPD_CONF_OPTS += -Dbzip2=enabled