Update buildroot from 17.02.2 -> 17.02.3

2018-01-04 14:25:23 +01:00
parent 01d7c53e3e
commit d83c6ae740
285 changed files with 5286 additions and 876 deletions
--- a/bsp/buildroot/package/audiofile/0003-Always-check-the-number-of-coefficients.patch
+++ b/bsp/buildroot/package/audiofile/0003-Always-check-the-number-of-coefficients.patch
@@ -0,0 +1,36 @@
+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
+
--- a/bsp/buildroot/package/audiofile/0004-clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
+++ b/bsp/buildroot/package/audiofile/0004-clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch
@@ -0,0 +1,39 @@
+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
+
--- a/bsp/buildroot/package/audiofile/0005-Check-for-multiplication-overflow-in-sfconvert.patch
+++ b/bsp/buildroot/package/audiofile/0005-Check-for-multiplication-overflow-in-sfconvert.patch
@@ -0,0 +1,72 @@
+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
+
--- a/bsp/buildroot/package/audiofile/0006-Actually-fail-when-error-occurs-in-parseFormat.patch
+++ b/bsp/buildroot/package/audiofile/0006-Actually-fail-when-error-occurs-in-parseFormat.patch
@@ -0,0 +1,42 @@
+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
+
--- a/bsp/buildroot/package/audiofile/0007-Check-for-multiplication-overflow-in-MSADPCM-decodeS.patch
+++ b/bsp/buildroot/package/audiofile/0007-Check-for-multiplication-overflow-in-MSADPCM-decodeS.patch
@@ -0,0 +1,122 @@
+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
+
--- a/bsp/buildroot/package/audiofile/0008-CVE-2015-7747.patch
+++ b/bsp/buildroot/package/audiofile/0008-CVE-2015-7747.patch
@@ -0,0 +1,161 @@
+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);
+}
--- a/bsp/buildroot/package/audiofile/0009-Fix-static-linking-with-libsndfile.patch
+++ b/bsp/buildroot/package/audiofile/0009-Fix-static-linking-with-libsndfile.patch
@@ -0,0 +1,193 @@
+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
+