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
bool "binutils 2.33.x has been removed"
select BR2_LEGACY
source "package/alsa-lib/Config.in"
source "package/alure/Config.in"
source "package/aubio/Config.in"
- source "package/audiofile/Config.in"
source "package/bcg729/Config.in"
source "package/caps/Config.in"
source "package/fdk-aac/Config.in"
+++ /dev/null
-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
-
+++ /dev/null
-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>
- {
+++ /dev/null
-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
-
+++ /dev/null
-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
-
+++ /dev/null
-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
-
+++ /dev/null
-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
-
+++ /dev/null
-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
-
+++ /dev/null
-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);
-+}
+++ /dev/null
-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
-
+++ /dev/null
-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
+++ /dev/null
-# Locally calculated
-sha256 cdc60df19ab08bfe55344395739bb08f50fc15c92da3962fac334d3bff116965 audiofile-0.3.6.tar.gz
-sha256 dc626520dcd53a22f727af3ee42c770e56c97a64fe3adb063799d8ab032fe551 COPYING
-sha256 8177f97513213526df2cf6184d8ff986c675afb514d4e68a404010521b880643 COPYING.GPL
+++ /dev/null
-################################################################################
-#
-# 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 $(autotools-package))
comment "Decoder plugins"
-config BR2_PACKAGE_MPD_AUDIOFILE
- bool "audiofile"
- select BR2_PACKAGE_AUDIOFILE
- help
- Enable audiofile input/streaming support.
- Select this if you want to play back WAV files.
-
config BR2_PACKAGE_MPD_DSD
bool "dsd"
help
MPD_DEPENDENCIES = host-pkgconf boost
MPD_LICENSE = GPL-2.0+
MPD_LICENSE_FILES = COPYING
-MPD_CONF_OPTS = -Ddocumentation=disabled
+MPD_CONF_OPTS = \
+ -Daudiofile=disabled \
+ -Ddocumentation=disabled
# Zeroconf support depends on libdns_sd from avahi.
ifeq ($(BR2_PACKAGE_MPD_AVAHI_SUPPORT),y)
MPD_CONF_OPTS += -Dao=disabled
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)
MPD_DEPENDENCIES += bzip2
MPD_CONF_OPTS += -Dbzip2=enabled
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