Sun Apr 14 00:46:05 PDT 2013

6 files changed, 44 insertions, 1052 deletions

diff --git a/community/csfml/PKGBUILD b/community/csfml/PKGBUILD
index 1f47c2772..4ed6778c8 100644
--- a/community/csfml/PKGBUILD
+++ b/community/csfml/PKGBUILD
@@ -1,59 +1,34 @@
-# $Id: PKGBUILD 70299 2012-05-04 03:01:09Z svenstaro $
+# $Id: PKGBUILD 88191 2013-04-13 15:25:43Z svenstaro $
 # Maintainer: Sven-Hendrik Haase <sh@lutzhaase.com>
 
 pkgname=csfml
-
-_git=true
-
-if [[ "${_git}" = "true" ]]; then
-    pkgver=1.99.git20120504
-fi
-
+pkgver=2.0
 pkgrel=1
 pkgdesc='C bindings for sfml'
 arch=('i686' 'x86_64')
 url='http://www.sfml-dev.org/'
 license=('zlib')
-depends=('sfml')
-makedepends=('git' 'cmake' 'doxygen')
-
-_gitroot='https://github.com/LaurentGomila/CSFML.git'
-_gitname='CSFML'
+depends=("sfml=${pkgver}")
+makedepends=('cmake' 'doxygen')
+source=("csfml-${pkgver}::https://github.com/LaurentGomila/CSFML/archive/2.0.tar.gz")
+md5sums=('d609c9e3b115d480d8c61911d774472c')
 
 build() {
-  cd "$srcdir"
-  msg "Connecting to GIT server...."
-
-  if [ -d $_gitname ] ; then
-    cd $_gitname && git pull origin
-    msg "The local files are updated."
-  else
-    git clone $_gitroot
-    cd $_gitname
-  fi
-
-  msg "GIT checkout done or server timeout"
-  msg "Starting make..."
-
-  rm -rf "$srcdir/$_gitname-build"
-  cp -r "$srcdir/$_gitname" "$srcdir/$_gitname-build"
-  cd "$srcdir/$_gitname-build"
+  cd "$srcdir"/CSFML-${pkgver}
 
   mkdir build && cd build
-  cmake -DCMAKE_INSTALL_PREFIX=/usr .. \
-        -DBUILD_DOC=true
+  cmake .. \
+      -DCMAKE_INSTALL_PREFIX=/usr \
+      -DBUILD_DOC=true
   make
   make doc
 }
 
 package() {
-  cd "$srcdir/$_gitname-build/build/"
+  cd "$srcdir"/CSFML-${pkgver}/build
 
   make DESTDIR="$pkgdir/" install
 
-  install -Dm644 ../license.txt \
-    ${pkgdir}/usr/share/licenses/${pkgname}/LICENSE
-
-  make clean
+  install -Dm644 ../license.txt ${pkgdir}/usr/share/licenses/${pkgname}/LICENSE
 }
 
diff --git a/community/ffmpegsource/PKGBUILD b/community/ffmpegsource/PKGBUILD
index eac86386e..e32088959 100755
--- a/community/ffmpegsource/PKGBUILD
+++ b/community/ffmpegsource/PKGBUILD
@@ -1,9 +1,9 @@
-# $Id: PKGBUILD 85215 2013-02-27 18:14:32Z alucryd $
+# $Id: PKGBUILD 88204 2013-04-13 20:20:33Z alucryd $
 # Maintainer: Maxime Gauduin <alucryd@gmail.com>
 
 pkgname=ffmpegsource
-pkgver=743
-pkgrel=2
+pkgver=753
+pkgrel=1
 pkgdesc="A libav/ffmpeg based source library and Avisynth plugin for easy frame accurate access"
 arch=('i686' 'x86_64')
 url="http://code.google.com/p/ffmpegsource/"
@@ -11,49 +11,35 @@ license=('MIT')
 depends=('ffmpeg')
 makedepends=('svn')
 options=('!libtool')
-source=('autoconf.patch' 'enable-libavresample.patch')
-sha256sums=('b09a7e9a08a16bdaf19d43c7ad8d3ec455f6fecec2f4f5ada417345343adda93'
-            '05f03515cc2405cdf8a8ba835f5adc2057f40054a4a1d9e493f0ad512c5de70d')
+source=("${pkgname}::svn+http://ffmpegsource.googlecode.com/svn/trunk/"
+        'autoconf.patch')
+sha256sums=('SKIP'
+            'b09a7e9a08a16bdaf19d43c7ad8d3ec455f6fecec2f4f5ada417345343adda93')
 
-_svntrunk=http://ffmpegsource.googlecode.com/svn/trunk/
-_svnmod=ffmpegsource
+pkgver() {
+  cd "${SRCDEST}"/${pkgname}
 
-build() {
-  cd "${srcdir}"
-
-# Checkout
-  msg "Connecting to SVN server...."
-
-  if [[ -d ${_svnmod}/.svn ]]; then
-    (cd ${_svnmod} && svn up -r ${pkgver})
-  else
-    svn co ${_svntrunk} --config-dir ./ -r ${pkgver} ${_svnmod}
-  fi
+  svnversion | tr -d [A-z]
+}
 
-  msg "SVN checkout done or server timeout"
-  msg "Starting build..."
+prepare() {
+  cd "${srcdir}"/${pkgname}
 
-  rm -rf "${srcdir}"/${_svnmod}-build
-# svn export "${srcdir}"/${_svnmod} "${srcdir}"/${_svnmod}-build
-  cp -R "${srcdir}"/${_svnmod} "${srcdir}"/${_svnmod}-build
-  cd "${srcdir}"/${_svnmod}-build
+  patch -Np1 -i ../autoconf.patch
+}
 
-# Patch
-  patch -Np1 -i "${srcdir}"/autoconf.patch
-  patch -Np1 -i "${srcdir}"/enable-libavresample.patch
+build() {
+  cd "${srcdir}"/${pkgname}
 
-# Build
   ./autogen.sh --prefix=/usr --enable-shared --disable-static
   make
 }
 
 package() {
-  cd "${srcdir}"/${_svnmod}-build
+  cd "${srcdir}"/${pkgname}
 
-# Install
   make DESTDIR="${pkgdir}" install
 
-# License
   install -dm 755 "${pkgdir}"/usr/share/licenses/ffmpegsource
   install -m 644 COPYING "${pkgdir}"/usr/share/licenses/ffmpegsource/LICENSE
 }
diff --git a/community/ffmpegsource/enable-libavresample.patch b/community/ffmpegsource/enable-libavresample.patch
deleted file mode 100755
index 56d435782..000000000
--- a/community/ffmpegsource/enable-libavresample.patch
+++ /dev/null
@@ -1,970 +0,0 @@
-# enable-libavresample.patch
-#
-# Adds libavresample support. Created by diffing Thomas Goyne's GIT repo
-# with official ffms SVN.
-#
-
-diff -ru ffmpegsource/configure.ac ffms2/configure.ac
---- ffmpegsource/configure.ac	2013-02-27 16:53:39.230691825 +0100
-+++ ffms2/configure.ac	2013-02-27 16:53:31.737713841 +0100
-@@ -181,6 +181,25 @@
-               AC_MSG_RESULT([no])
-             ])
- 
-+AC_ARG_ENABLE(avresample,
-+              AS_HELP_STRING([--enable-avresample],
-+                             [use libavresample for audio resampling]))
-+AS_IF([test x$enable_avresample != xno], [
-+  PKG_CHECK_MODULES(AVRESAMPLE, [libavresample >= 1.0.0], [enable_avresample=yes], [
-+    AS_IF([test x$enable_avresample = xyes],
-+          [AC_MSG_ERROR([--enable-avresample was specified, but avresample 1.0.0+ could not be found.])])
-+    enable_avresample=no
-+  ])
-+])
-+
-+AS_IF([test x$enable_avresample],
-+      [libavresample="libavresample"
-+       AC_DEFINE([WITH_AVRESAMPLE], [1], [Use avresample])])
-+
-+AC_SUBST([AVRESAMPLE_CFLAGS])
-+AC_SUBST([AVRESAMPLE_LIBS])
-+AC_SUBST([libavresample])
-+
- AC_MSG_CHECKING([whether -Wl,-Bsymbolic is needed])
- if test "$enable_shared" = yes; then
-     _LDFLAGS="$LDFLAGS"
-diff -ru ffmpegsource/ffms2.pc.in ffms2/ffms2.pc.in
---- ffmpegsource/ffms2.pc.in	2013-02-27 16:53:38.924039701 +0100
-+++ ffms2/ffms2.pc.in	2013-02-27 16:53:31.737713841 +0100
-@@ -7,7 +7,7 @@
- 
- Name: ffms2
- Description: The Fabulous FM Library 2
--Requires.private: libavformat libavcodec libswscale libavutil
-+Requires.private: libavformat libavcodec libswscale libavutil @libavresample@
- Version: @FFMS_VERSION@
- Libs.private: @ZLIB_LDFLAGS@ -lz
- Libs: -L${libdir} -lffms2
-diff -ru ffmpegsource/include/ffmscompat.h ffms2/include/ffmscompat.h
---- ffmpegsource/include/ffmscompat.h	2013-02-27 16:53:38.920706525 +0100
-+++ ffms2/include/ffmscompat.h	2013-02-27 16:53:31.737713841 +0100
-@@ -71,6 +71,15 @@
- #       define FFMS_CodecID AVCodecID
- #       undef CodecID
- #   endif
-+#   if VERSION_CHECK(LIBAVCODEC_VERSION_INT, <, 54, 28, 0, 54, 59, 100)
-+#       define avcodec_free_frame av_free
-+#   endif
-+#endif
-+
-+#ifdef LIBAVUTIL_VERSION_INT
-+#	if VERSION_CHECK(LIBAVUTIL_VERSION_INT, <, 51, 27, 0, 51, 46, 100)
-+#		define av_get_packed_sample_fmt(fmt) (fmt < AV_SAMPLE_FMT_U8P ? fmt : fmt - (AV_SAMPLE_FMT_U8P - AV_SAMPLE_FMT_U8))
-+#	endif
- #endif
- 
- #endif // FFMSCOMPAT_H
-diff -ru ffmpegsource/include/ffms.h ffms2/include/ffms.h
---- ffmpegsource/include/ffms.h	2013-02-27 16:53:38.920706525 +0100
-+++ ffms2/include/ffms.h	2013-02-27 16:53:31.737713841 +0100
-@@ -113,6 +113,7 @@
- 	FFMS_ERROR_TRACK,				// track handling
- 	FFMS_ERROR_WAVE_WRITER,			// WAVE64 file writer
- 	FFMS_ERROR_CANCELLED,			// operation aborted
-+	FFMS_ERROR_RESAMPLING,			// audio resampling (libavresample)
- 
- 	// Subtypes - what caused the error
- 	FFMS_ERROR_UNKNOWN = 20,		// unknown error
-@@ -237,6 +238,53 @@
- 	FFMS_CR_JPEG		= 2 // 2^n-1, or "fullrange"
- } FFMS_ColorRanges;
- 
-+typedef enum FFMS_MixingCoefficientType {
-+	FFMS_MIXING_COEFFICIENT_Q8  = 0,
-+	FFMS_MIXING_COEFFICIENT_Q15 = 1,
-+	FFMS_MIXING_COEFFICIENT_FLT = 2
-+} FFMS_MixingCoefficientType;
-+
-+typedef enum FFMS_MatrixEncoding {
-+	FFMS_MATRIX_ENCODING_NONE         = 0,
-+	FFMS_MATRIX_ENCODING_DOBLY        = 1,
-+	FFMS_MATRIX_ENCODING_PRO_LOGIC_II = 2
-+} FFMS_MatrixEncoding;
-+
-+typedef enum FFMS_ResampleFilterType {
-+	FFMS_RESAMPLE_FILTER_CUBIC  = 0,
-+	FFMS_RESAMPLE_FILTER_SINC   = 1,
-+	FFMS_RESAMPLE_FILTER_KAISER = 2
-+} FFMS_ResampleFilterType;
-+
-+typedef enum FFMS_AudioDitherMethod {
-+	FFMS_RESAMPLE_DITHER_NONE                    = 0,
-+	FFMS_RESAMPLE_DITHER_RECTANGULAR             = 1,
-+	FFMS_RESAMPLE_DITHER_TRIANGULAR              = 2,
-+	FFMS_RESAMPLE_DITHER_TRIANGULAR_HIGHPASS     = 3,
-+	FFMS_RESAMPLE_DITHER_TRIANGULAR_NOISESHAPING = 4
-+} FFMS_AudioDitherMethod;
-+
-+typedef struct FFMS_ResampleOptions {
-+	int64_t ChannelLayout;
-+	FFMS_SampleFormat SampleFormat;
-+	int SampleRate;
-+	FFMS_MixingCoefficientType MixingCoefficientType;
-+	double CenterMixLevel;
-+	double SurroundMixLevel;
-+	double LFEMixLevel;
-+	int Normalize;
-+	int ForceResample;
-+	int ResampleFilterSize;
-+	int ResamplePhaseShift;
-+	int LinearInterpolation;
-+	double CutoffFrequencyRatio;
-+	FFMS_MatrixEncoding MatrixedStereoEncoding;
-+	FFMS_ResampleFilterType FilterType;
-+	int KaiserBeta;
-+	FFMS_AudioDitherMethod DitherMethod;
-+} FFMS_ResampleOptions;
-+
-+
- typedef struct FFMS_Frame {
- 	uint8_t *Data[4];
- 	int Linesize[4];
-@@ -319,6 +367,9 @@
- FFMS_API(void) FFMS_ResetOutputFormatV(FFMS_VideoSource *V);
- FFMS_API(int) FFMS_SetInputFormatV(FFMS_VideoSource *V, int ColorSpace, int ColorRange, int Format, FFMS_ErrorInfo *ErrorInfo); /* Introduced in FFMS_VERSION ((2 << 24) | (17 << 16) | (1 << 8) | 0) */
- FFMS_API(void) FFMS_ResetInputFormatV(FFMS_VideoSource *V);
-+FFMS_API(FFMS_ResampleOptions *) FFMS_CreateResampleOptions(FFMS_AudioSource *A); /* Introduced in FFMS_VERSION ((2 << 24) | (15 << 16) | (4 << 8) | 0) */
-+FFMS_API(int) FFMS_SetOutputFormatA(FFMS_AudioSource *A, const FFMS_ResampleOptions*options, FFMS_ErrorInfo *ErrorInfo); /* Introduced in FFMS_VERSION ((2 << 24) | (15 << 16) | (4 << 8) | 0) */
-+FFMS_API(void) FFMS_DestroyResampleOptions(FFMS_ResampleOptions *options); /* Introduced in FFMS_VERSION ((2 << 24) | (15 << 16) | (4 << 8) | 0) */
- FFMS_API(void) FFMS_DestroyIndex(FFMS_Index *Index);
- FFMS_API(int) FFMS_GetSourceType(FFMS_Index *Index);
- FFMS_API(int) FFMS_GetSourceTypeI(FFMS_Indexer *Indexer);
-diff -ru ffmpegsource/Makefile.am ffms2/Makefile.am
---- ffmpegsource/Makefile.am	2013-02-27 16:53:39.310688030 +0100
-+++ ffms2/Makefile.am	2013-02-27 16:53:31.724381141 +0100
-@@ -9,7 +9,7 @@
- INCLUDES = -I. -I$(top_srcdir)/include -I$(top_srcdir)/src/config @LIBAV_CFLAGS@ @ZLIB_CPPFLAGS@ -include config.h
- 
- lib_LTLIBRARIES = src/core/libffms2.la
--src_core_libffms2_la_LIBADD = @LIBAV_LIBS@ @ZLIB_LDFLAGS@ -lz @LTUNDEF@
-+src_core_libffms2_la_LIBADD = @LIBAV_LIBS@ @AVRESAMPLE_LIBS@ @ZLIB_LDFLAGS@ -lz @LTUNDEF@
- src_core_libffms2_la_SOURCES = \
- 	src/core/audiosource.h \
- 	src/core/audiosource.cpp \
-diff -ru ffmpegsource/src/config/config.h.in ffms2/src/config/config.h.in
---- ffmpegsource/src/config/config.h.in	2013-02-27 16:53:39.017368608 +0100
-+++ ffms2/src/config/config.h.in	2013-02-27 16:53:31.744380192 +0100
-@@ -90,5 +90,8 @@
- /* Version number of package */
- #undef VERSION
- 
-+/* Use avresample */
-+#undef WITH_AVRESAMPLE
-+
- /* Define to `unsigned int' if <sys/types.h> does not define. */
- #undef size_t
-diff -ru ffmpegsource/src/config/libs.cpp ffms2/src/config/libs.cpp
---- ffmpegsource/src/config/libs.cpp	2013-02-27 16:53:39.017368608 +0100
-+++ ffms2/src/config/libs.cpp	2013-02-27 16:53:31.744380192 +0100
-@@ -45,6 +45,9 @@
- #pragma comment(lib, "libavcodec.a")
- #pragma comment(lib, "libavformat.a")
- #pragma comment(lib, "libswscale.a")
-+#ifdef WITH_AVRESAMPLE
-+#pragma comment(lib, "libavresample.a")
-+#endif
- 
- #ifdef WITH_OPENCORE_AMR_NB
- #ifdef WITH_GCC_LIBAV
-diff -ru ffmpegsource/src/core/audiosource.cpp ffms2/src/core/audiosource.cpp
---- ffmpegsource/src/core/audiosource.cpp	2013-02-27 16:53:39.137362917 +0100
-+++ ffms2/src/core/audiosource.cpp	2013-02-27 16:53:31.744380192 +0100
-@@ -23,17 +23,45 @@
- #include <algorithm>
- #include <cassert>
- 
-+namespace {
-+
-+	int64_t ChannelLayout;
-+	FFMS_SampleFormat SampleFormat;
-+	int SampleRate;
-+#define MAPPER(m, n) OptionMapper<FFMS_ResampleOptions>(n, &FFMS_ResampleOptions::m)
-+OptionMapper<FFMS_ResampleOptions> resample_options[] = {
-+	MAPPER(ChannelLayout,          "out_channel_layout"),
-+	MAPPER(SampleFormat,           "out_sample_fmt"),
-+	MAPPER(SampleRate,             "out_sample_rate"),
-+	MAPPER(MixingCoefficientType,  "mix_coeff_type"),
-+	MAPPER(CenterMixLevel,         "center_mix_level"),
-+	MAPPER(SurroundMixLevel,       "surround_mix_level"),
-+	MAPPER(LFEMixLevel,            "lfe_mix_level"),
-+	MAPPER(Normalize,              "normalize_mix_level"),
-+	MAPPER(ForceResample,          "force_resampling"),
-+	MAPPER(ResampleFilterSize,     "filter_size"),
-+	MAPPER(ResamplePhaseShift,     "phase_shift"),
-+	MAPPER(LinearInterpolation,    "linear_interp"),
-+	MAPPER(CutoffFrequencyRatio,   "cutoff"),
-+	MAPPER(MatrixedStereoEncoding, "matrix_encoding"),
-+	MAPPER(FilterType,             "filter_type"),
-+	MAPPER(KaiserBeta,             "kaiser_beta"),
-+	MAPPER(DitherMethod,           "dither_method")
-+};
-+#undef MAPPER
-+
-+}
-+
- FFMS_AudioSource::FFMS_AudioSource(const char *SourceFile, FFMS_Index &Index, int Track)
- : Delay(0)
- , MaxCacheBlocks(50)
- , BytesPerSample(0)
--, Decoded(0)
-+, NeedsResample(false)
- , CurrentSample(-1)
- , PacketNumber(0)
- , CurrentFrame(NULL)
- , TrackNumber(Track)
- , SeekOffset(0)
--, DecodingBuffer(AVCODEC_MAX_AUDIO_FRAME_SIZE * 10)
- , Index(Index)
- {
- 	if (Track < 0 || Track >= static_cast<int>(Index.size()))
-@@ -57,44 +85,14 @@
- 	Index.AddRef();
- }
- 
--
- #define EXCESSIVE_CACHE_SIZE 400
- 
- void FFMS_AudioSource::Init(const FFMS_Index &Index, int DelayMode) {
--	// The first packet after a seek is often decoded incorrectly, which
--	// makes it impossible to ever correctly seek back to the beginning, so
--	// store the first block now
--
--	// In addition, anything with the same PTS as the first packet can't be
--	// distinguished from the first packet and so can't be seeked to, so
--	// store those as well
--
--	// Some of LAVF's splitters don't like to seek to the beginning of the
--	// file (ts and?), so cache a few blocks even if PTSes are unique
--	// Packet 7 is the last packet I've had be unseekable to, so cache up to
--	// 10 for a bit of an extra buffer
--	CacheIterator end = Cache.end();
--	while (PacketNumber < Frames.size() &&
--		((Frames[0].PTS != ffms_av_nopts_value && Frames[PacketNumber].PTS == Frames[0].PTS) ||
--		 Cache.size() < 10)) {
--
--		// Vorbis in particular seems to like having 60+ packets at the start of the file with a PTS of 0,
--		// so we might need to expand the search range to account for that.
--		if (Cache.size() >= MaxCacheBlocks - 1) {
--			 if (MaxCacheBlocks >= EXCESSIVE_CACHE_SIZE)
--				 throw FFMS_Exception(FFMS_ERROR_DECODING, FFMS_ERROR_ALLOCATION_FAILED, "Exceeded the search range for an initial valid audio PTS");
--			MaxCacheBlocks *= 2;
--		}
--
-+	// Decode the first packet to ensure all properties are initialized
-+	// Don't cache it since it might be in the wrong format
-+	// Instead, leave it in DecodeFrame and it'll get cached later
-+	while (DecodeFrame->nb_samples == 0)
- 		DecodeNextBlock();
--		if (Decoded)
--			CacheBlock(end, CurrentSample, Decoded, &DecodingBuffer[0]);
--	}
--	// Store the iterator to the last element of the cache which is used for
--	// correctness rather than speed, so that when looking for one to delete
--	// we know how much to skip
--	CacheNoDelete = Cache.end();
--	--CacheNoDelete;
- 
- 	// Read properties of the audio which may not be available until the first
- 	// frame has been decoded
-@@ -104,6 +102,11 @@
- 		throw FFMS_Exception(FFMS_ERROR_DECODING, FFMS_ERROR_CODEC,
- 			"Codec returned zero size audio");
- 
-+	if (av_sample_fmt_is_planar(CodecContext->sample_fmt)) {
-+		std::auto_ptr<FFMS_ResampleOptions> opt(CreateResampleOptions());
-+		SetOutputFormat(opt.get());
-+	}
-+
- 	if (DelayMode < FFMS_DELAY_NO_SHIFT)
- 		throw FFMS_Exception(FFMS_ERROR_INDEX, FFMS_ERROR_INVALID_ARGUMENT,
- 			"Bad audio delay compensation mode");
-@@ -146,8 +149,133 @@
- 	AP.NumSamples += Delay;
- }
- 
--void FFMS_AudioSource::CacheBlock(CacheIterator &pos, int64_t Start, size_t Samples, uint8_t *SrcData) {
--	Cache.insert(pos, AudioBlock(Start, Samples, SrcData, Samples * BytesPerSample));
-+void FFMS_AudioSource::CacheBeginning() {
-+	// Nothing to do if the cache is already populated
-+	if (!Cache.empty()) return;
-+
-+	// The first frame is already decoded, so add it to the cache
-+	CacheBlock(Cache.end());
-+
-+	// The first packet after a seek is often decoded incorrectly, which
-+	// makes it impossible to ever correctly seek back to the beginning, so
-+	// store the first block now
-+
-+	// In addition, anything with the same PTS as the first packet can't be
-+	// distinguished from the first packet and so can't be seeked to, so
-+	// store those as well
-+
-+	// Some of LAVF's splitters don't like to seek to the beginning of the
-+	// file (ts and?), so cache a few blocks even if PTSes are unique
-+	// Packet 7 is the last packet I've had be unseekable to, so cache up to
-+	// 10 for a bit of an extra buffer
-+	CacheIterator end = Cache.end();
-+	while (PacketNumber < Frames.size() &&
-+		((Frames[0].PTS != ffms_av_nopts_value && Frames[PacketNumber].PTS == Frames[0].PTS) ||
-+		 Cache.size() < 10)) {
-+
-+		// Vorbis in particular seems to like having 60+ packets at the start
-+		// of the file with a PTS of 0, so we might need to expand the search
-+		// range to account for that.
-+		// Expanding slightly before it's strictly needed to ensure there's a
-+		// bit of space for an actual cache
-+		if (Cache.size() >= MaxCacheBlocks - 5) {
-+			 if (MaxCacheBlocks >= EXCESSIVE_CACHE_SIZE)
-+				throw FFMS_Exception(FFMS_ERROR_DECODING, FFMS_ERROR_ALLOCATION_FAILED,
-+					"Exceeded the search range for an initial valid audio PTS");
-+			MaxCacheBlocks *= 2;
-+		}
-+
-+		DecodeNextBlock(&end);
-+	}
-+	// Store the iterator to the last element of the cache which is used for
-+	// correctness rather than speed, so that when looking for one to delete
-+	// we know how much to skip
-+	CacheNoDelete = Cache.end();
-+	--CacheNoDelete;
-+}
-+
-+void FFMS_AudioSource::SetOutputFormat(const FFMS_ResampleOptions *opt) {
-+	if (!Cache.empty())
-+		throw FFMS_Exception(FFMS_ERROR_RESAMPLING, FFMS_ERROR_USER,
-+			"Cannot change the output format after audio decoding has begun");
-+
-+	BytesPerSample = av_get_bytes_per_sample(static_cast<AVSampleFormat>(opt->SampleFormat)) * av_get_channel_layout_nb_channels(opt->ChannelLayout);
-+
-+	NeedsResample =
-+		opt->SampleFormat != (int)CodecContext->sample_fmt ||
-+		opt->SampleRate != AP.SampleRate ||
-+		opt->ChannelLayout != AP.ChannelLayout ||
-+		opt->ForceResample;
-+	if (!NeedsResample) return;
-+
-+	if (opt->SampleRate != AP.SampleRate)
-+		throw FFMS_Exception(FFMS_ERROR_RESAMPLING, FFMS_ERROR_UNSUPPORTED,
-+			"Sample rate changes are currently unsupported.");
-+
-+#ifdef WITH_AVRESAMPLE
-+	if (opt->SampleRate != AP.SampleRate)
-+		throw FFMS_Exception(FFMS_ERROR_RESAMPLING, FFMS_ERROR_UNSUPPORTED,
-+			"Changing the audio sample rate is currently not supported");
-+
-+	std::auto_ptr<FFMS_ResampleOptions> oldOptions(ReadOptions(ResampleContext, resample_options));
-+	SetOptions(opt, ResampleContext, resample_options);
-+	av_opt_set_int(ResampleContext, "in_sample_rate", AP.SampleRate, 0);
-+	av_opt_set_int(ResampleContext, "in_sample_fmt", CodecContext->sample_fmt, 0);
-+	av_opt_set_int(ResampleContext, "in_channel_layout", AP.ChannelLayout, 0);
-+
-+	if (avresample_open(ResampleContext)) {
-+		SetOptions(oldOptions.get(), ResampleContext, resample_options);
-+		avresample_open(ResampleContext);
-+		throw FFMS_Exception(FFMS_ERROR_RESAMPLING, FFMS_ERROR_UNKNOWN,
-+			"Could not open avresample context");
-+	}
-+#else
-+	if (opt->SampleFormat != AP.SampleFormat || opt->SampleRate != AP.SampleRate || opt->ChannelLayout != AP.ChannelLayout)
-+		throw FFMS_Exception(FFMS_ERROR_RESAMPLING, FFMS_ERROR_UNSUPPORTED,
-+			"FFMS was not built with resampling enabled. The only supported conversion is interleaving planar audio.");
-+#endif
-+}
-+
-+FFMS_ResampleOptions *FFMS_AudioSource::CreateResampleOptions() const {
-+#ifdef WITH_AVRESAMPLE
-+	FFMS_ResampleOptions *ret = ReadOptions(ResampleContext, resample_options);
-+#else
-+	FFMS_ResampleOptions *ret = new FFMS_ResampleOptions;
-+	memset(ret, 0, sizeof(FFMS_ResampleOptions));
-+#endif
-+	ret->SampleRate = AP.SampleRate;
-+	ret->SampleFormat = static_cast<FFMS_SampleFormat>(AP.SampleFormat);
-+	ret->ChannelLayout = AP.ChannelLayout;
-+	return ret;
-+}
-+
-+void FFMS_AudioSource::ResampleAndCache(CacheIterator pos) {
-+	AudioBlock& block = *Cache.insert(pos, AudioBlock(CurrentSample, DecodeFrame->nb_samples));
-+	block.Data.reserve(DecodeFrame->nb_samples * BytesPerSample);
-+
-+#ifdef WITH_AVRESAMPLE
-+	block.Data.resize(block.Data.capacity());
-+
-+	uint8_t *OutPlanes[1] = { static_cast<uint8_t *>(&block.Data[0]) };
-+	avresample_convert(ResampleContext,
-+		OutPlanes, block.Data.size(), DecodeFrame->nb_samples,
-+		DecodeFrame->extended_data, DecodeFrame->nb_samples * av_get_bytes_per_sample(CodecContext->sample_fmt), DecodeFrame->nb_samples);
-+#else
-+	int width = av_get_bytes_per_sample(CodecContext->sample_fmt);
-+	uint8_t **Data = DecodeFrame->extended_data;
-+
-+	for (int s = 0; s < DecodeFrame->nb_samples; ++s) {
-+		for (int c = 0; c < CodecContext->channels; ++c)
-+			block.Data.insert(block.Data.end(), &Data[c][s * width], &Data[c][(s + 1) * width]);
-+	}
-+#endif
-+}
-+
-+void FFMS_AudioSource::CacheBlock(CacheIterator pos) {
-+	if (NeedsResample)
-+		ResampleAndCache(pos);
-+	else
-+		Cache.insert(pos, AudioBlock(CurrentSample, DecodeFrame->nb_samples, DecodeFrame->extended_data[0], DecodeFrame->nb_samples * BytesPerSample));
- 
- 	if (Cache.size() >= MaxCacheBlocks) {
- 		// Kill the oldest one
-@@ -162,45 +290,45 @@
- 	}
- }
- 
--void FFMS_AudioSource::DecodeNextBlock() {
--	if (BytesPerSample == 0) BytesPerSample = av_get_bytes_per_sample(CodecContext->sample_fmt) * CodecContext->channels;
--
-+void FFMS_AudioSource::DecodeNextBlock(CacheIterator *pos) {
- 	CurrentFrame = &Frames[PacketNumber];
- 
- 	AVPacket Packet;
- 	if (!ReadPacket(&Packet))
--		throw FFMS_Exception(FFMS_ERROR_PARSER, FFMS_ERROR_UNKNOWN, "ReadPacket unexpectedly failed to read a packet");
-+		throw FFMS_Exception(FFMS_ERROR_PARSER, FFMS_ERROR_UNKNOWN,
-+			"ReadPacket unexpectedly failed to read a packet");
- 
- 	// ReadPacket may have changed the packet number
- 	CurrentFrame = &Frames[PacketNumber];
- 	CurrentSample = CurrentFrame->SampleStart;
--	++PacketNumber;
- 
--	uint8_t *Buf = &DecodingBuffer[0];
-+	bool GotSamples = false;
- 	uint8_t *Data = Packet.data;
- 	while (Packet.size > 0) {
--		int TempOutputBufSize = AVCODEC_MAX_AUDIO_FRAME_SIZE * 10 - (Buf - &DecodingBuffer[0]);
--		int Ret = avcodec_decode_audio3(CodecContext, (int16_t *)Buf, &TempOutputBufSize, &Packet);
-+		DecodeFrame.reset();
-+		int GotFrame = 0;
-+		int Ret = avcodec_decode_audio4(CodecContext, DecodeFrame, &GotFrame, &Packet);
- 
- 		// Should only ever happen if the user chose to ignore decoding errors
- 		// during indexing, so continue to just ignore decoding errors
- 		if (Ret < 0) break;
- 
--		if (Ret > 0) {
-+		if (Ret > 0 && GotFrame) {
- 			Packet.size -= Ret;
- 			Packet.data += Ret;
--			Buf += TempOutputBufSize;
-+			if (DecodeFrame->nb_samples > 0) {
-+				GotSamples = true;
-+				if (pos)
-+					CacheBlock(*pos);
-+			}
- 		}
- 	}
- 	Packet.data = Data;
- 	FreePacket(&Packet);
- 
--	Decoded = (Buf - &DecodingBuffer[0]) / BytesPerSample;
--	if (Decoded == 0) {
--		// zero sample packets aren't included in the index so we didn't
--		// actually move to the next packet
--		--PacketNumber;
--	}
-+	// Zero sample packets aren't included in the index
-+	if (GotSamples)
-+		++PacketNumber;
- }
- 
- static bool SampleStartComp(const TFrameInfo &a, const TFrameInfo &b) {
-@@ -216,6 +344,8 @@
- 		throw FFMS_Exception(FFMS_ERROR_DECODING, FFMS_ERROR_INVALID_ARGUMENT,
- 			"Out of bounds audio samples requested");
- 
-+	CacheBeginning();
-+
- 	uint8_t *Dst = static_cast<uint8_t*>(Buf);
- 
- 	// Apply audio delay (if any) and fill any samples before the start time with zero
-@@ -253,10 +383,12 @@
- 		}
- 		// Decode another block
- 		else {
-+			CacheIterator cachePos = it; --cachePos;
-+
- 			if (Start < CurrentSample && SeekOffset == -1)
- 				throw FFMS_Exception(FFMS_ERROR_SEEKING, FFMS_ERROR_CODEC, "Audio stream is not seekable");
- 
--			if (SeekOffset >= 0 && (Start < CurrentSample || Start > CurrentSample + Decoded * 5)) {
-+			if (SeekOffset >= 0 && (Start < CurrentSample || Start > CurrentSample + DecodeFrame->nb_samples * 5)) {
- 				TFrameInfo f;
- 				f.SampleStart = Start;
- 				int NewPacketNumber = std::distance(Frames.begin(), std::lower_bound(Frames.begin(), Frames.end(), f, SampleStartComp));
-@@ -266,32 +398,22 @@
- 				// Only seek forward if it'll actually result in moving forward
- 				if (Start < CurrentSample || static_cast<size_t>(NewPacketNumber) > PacketNumber) {
- 					PacketNumber = NewPacketNumber;
--					Decoded = 0;
- 					CurrentSample = -1;
-+					DecodeFrame.reset();
- 					avcodec_flush_buffers(CodecContext);
- 					Seek();
- 				}
- 			}
- 
--			// Decode everything between the last keyframe and the block we want
-+			// Decode until we hit the block we want
- 			if (PacketNumber >= Frames.size())
- 				throw FFMS_Exception(FFMS_ERROR_SEEKING, FFMS_ERROR_CODEC, "Seeking is severely broken");
--			while (CurrentSample + Decoded <= Start && PacketNumber < Frames.size())
--				DecodeNextBlock();
-+			while (CurrentSample + DecodeFrame->nb_samples <= Start && PacketNumber < Frames.size())
-+				DecodeNextBlock(&it);
- 			if (CurrentSample > Start)
- 				throw FFMS_Exception(FFMS_ERROR_SEEKING, FFMS_ERROR_CODEC, "Seeking is severely broken");
- 
--			CacheBlock(it, CurrentSample, Decoded, &DecodingBuffer[0]);
--
--			size_t FirstSample = static_cast<size_t>(Start - CurrentSample);
--			size_t Samples = static_cast<size_t>(Decoded - FirstSample);
--			size_t Bytes = FFMIN(Samples, static_cast<size_t>(Count)) * BytesPerSample;
--
--			memcpy(Dst, &DecodingBuffer[FirstSample * BytesPerSample], Bytes);
--
--			Start += Samples;
--			Count -= Samples;
--			Dst += Bytes;
-+			it = cachePos;
- 		}
- 	}
- }
-diff -ru ffmpegsource/src/core/audiosource.h ffms2/src/core/audiosource.h
---- ffmpegsource/src/core/audiosource.h	2013-02-27 16:53:39.130696566 +0100
-+++ ffms2/src/core/audiosource.h	2013-02-27 16:53:31.744380192 +0100
-@@ -46,7 +46,6 @@
- #endif
- 
- struct FFMS_AudioSource {
--private:
- 	struct AudioBlock {
- 		int64_t Age;
- 		int64_t Start;
-@@ -54,9 +53,17 @@
- 		std::vector<uint8_t> Data;
- 
- 		AudioBlock(int64_t Start, int64_t Samples, uint8_t *SrcData, size_t SrcBytes)
--			: Start(Start)
--			, Samples(Samples)
--			, Data(SrcData, SrcData + SrcBytes)
-+		: Start(Start)
-+		, Samples(Samples)
-+		, Data(SrcData, SrcData + SrcBytes)
-+		{
-+			static int64_t Now = 0;
-+			Age = Now++;
-+		}
-+
-+		AudioBlock(int64_t Start, int64_t Samples)
-+		: Start(Start)
-+		, Samples(Samples)
- 		{
- 			static int64_t Now = 0;
- 			Age = Now++;
-@@ -74,11 +81,18 @@
- 	CacheIterator CacheNoDelete;
- 	// bytes per sample * number of channels
- 	size_t BytesPerSample;
--	// Number of samples stored in the decoding buffer
--	size_t Decoded;
- 
--	// Insert a block into the cache
--	void CacheBlock(CacheIterator &pos, int64_t Start, size_t Samples, uint8_t *SrcData);
-+	bool NeedsResample;
-+	FFResampleContext ResampleContext;
-+
-+	// Insert the current audio frame into the cache
-+	void CacheBlock(CacheIterator pos);
-+
-+	// Interleave the current audio frame and insert it into the cache
-+	void ResampleAndCache(CacheIterator pos);
-+
-+	// Cache the unseekable beginning of the file once the output format is set
-+	void CacheBeginning();
- 
- 	// Called after seeking
- 	virtual void Seek() { };
-@@ -99,13 +113,13 @@
- 	int SeekOffset;
- 
- 	// Buffer which audio is decoded into
--	AlignedBuffer<uint8_t> DecodingBuffer;
-+	ScopedFrame DecodeFrame;
- 	FFMS_Index &Index;
- 	FFMS_Track Frames;
- 	FFCodecContext CodecContext;
- 	FFMS_AudioProperties AP;
- 
--	void DecodeNextBlock();
-+	void DecodeNextBlock(CacheIterator *cachePos = 0);
- 	// Initialization which has to be done after the codec is opened
- 	void Init(const FFMS_Index &Index, int DelayMode);
- 
-@@ -116,6 +130,9 @@
- 	FFMS_Track *GetTrack() { return &Frames; }
- 	const FFMS_AudioProperties& GetAudioProperties() const { return AP; }
- 	void GetAudio(void *Buf, int64_t Start, int64_t Count);
-+
-+	FFMS_ResampleOptions *CreateResampleOptions() const;
-+	void SetOutputFormat(const FFMS_ResampleOptions *opt);
- };
- 
- class FFLAVFAudio : public FFMS_AudioSource {
-diff -ru ffmpegsource/src/core/ffms.cpp ffms2/src/core/ffms.cpp
---- ffmpegsource/src/core/ffms.cpp	2013-02-27 16:53:39.137362917 +0100
-+++ ffms2/src/core/ffms.cpp	2013-02-27 16:53:31.744380192 +0100
-@@ -256,6 +256,24 @@
- 	V->ResetInputFormat();
- }
- 
-+FFMS_API(FFMS_ResampleOptions *) FFMS_CreateResampleOptions(FFMS_AudioSource *A) {
-+	return A->CreateResampleOptions();
-+}
-+
-+FFMS_API(void) FFMS_DestroyResampleOptions(FFMS_ResampleOptions *options) {
-+	delete options;
-+}
-+
-+FFMS_API(int) FFMS_SetOutputFormatA(FFMS_AudioSource *A, const FFMS_ResampleOptions *options, FFMS_ErrorInfo *ErrorInfo) {
-+	ClearErrorInfo(ErrorInfo);
-+	try {
-+		A->SetOutputFormat(options);
-+	} catch (FFMS_Exception &e) {
-+		return e.CopyOut(ErrorInfo);
-+	}
-+	return FFMS_ERROR_SUCCESS;
-+}
-+
- FFMS_API(void) FFMS_DestroyIndex(FFMS_Index *Index) {
- 	assert(Index != NULL);
- 	if (Index == NULL)
-diff -ru ffmpegsource/src/core/indexing.cpp ffms2/src/core/indexing.cpp
---- ffmpegsource/src/core/indexing.cpp	2013-02-27 16:53:39.134029741 +0100
-+++ ffms2/src/core/indexing.cpp	2013-02-27 16:53:31.744380192 +0100
-@@ -693,7 +693,6 @@
- , ANC(0)
- , ANCPrivate(0)
- , SourceFile(Filename)
--, DecodingBuffer(AVCODEC_MAX_AUDIO_FRAME_SIZE * 10)
- {
- 	FFMS_Index::CalculateFileSignature(Filename, &Filesize, Digest);
- }
-@@ -702,9 +701,9 @@
- 
- }
- 
--void FFMS_Indexer::WriteAudio(SharedAudioContext &AudioContext, FFMS_Index *Index, int Track, int DBSize) {
-+void FFMS_Indexer::WriteAudio(SharedAudioContext &AudioContext, FFMS_Index *Index, int Track) {
- 	// Delay writer creation until after an audio frame has been decoded. This ensures that all parameters are known when writing the headers.
--	if (DBSize <= 0) return;
-+	if (DecodeFrame->nb_samples) return;
- 
- 	if (!AudioContext.W64Writer) {
- 		FFMS_AudioProperties AP;
-@@ -715,6 +714,8 @@
- 			return;
- 		}
- 
-+		int Format = av_get_packed_sample_fmt(AudioContext.CodecContext->sample_fmt);
-+
- 		std::vector<char> WName(FNSize);
- 		(*ANC)(SourceFile.c_str(), Track, &AP, &WName[0], FNSize, ANCPrivate);
- 		std::string WN(&WName[0]);
-@@ -724,14 +725,14 @@
- 					av_get_bytes_per_sample(AudioContext.CodecContext->sample_fmt),
- 					AudioContext.CodecContext->channels,
- 					AudioContext.CodecContext->sample_rate,
--					(AudioContext.CodecContext->sample_fmt == AV_SAMPLE_FMT_FLT) || (AudioContext.CodecContext->sample_fmt == AV_SAMPLE_FMT_DBL));
-+					(Format == AV_SAMPLE_FMT_FLT) || (Format == AV_SAMPLE_FMT_DBL));
- 		} catch (...) {
- 			throw FFMS_Exception(FFMS_ERROR_WAVE_WRITER, FFMS_ERROR_FILE_WRITE,
- 				"Failed to write wave data");
- 		}
- 	}
- 
--	AudioContext.W64Writer->WriteData(&DecodingBuffer[0], DBSize);
-+	AudioContext.W64Writer->WriteData(*DecodeFrame);
- }
- 
- int64_t FFMS_Indexer::IndexAudioPacket(int Track, AVPacket *Packet, SharedAudioContext &Context, FFMS_Index &TrackIndices) {
-@@ -739,8 +740,10 @@
- 	int64_t StartSample = Context.CurrentSample;
- 	int Read = 0;
- 	while (Packet->size > 0) {
--		int dbsize = AVCODEC_MAX_AUDIO_FRAME_SIZE*10;
--		int Ret = avcodec_decode_audio3(CodecContext, (int16_t *)&DecodingBuffer[0], &dbsize, Packet);
-+		DecodeFrame.reset();
-+
-+		int GotFrame = 0;
-+		int Ret = avcodec_decode_audio4(CodecContext, DecodeFrame, &GotFrame, Packet);
- 		if (Ret < 0) {
- 			if (ErrorHandling == FFMS_IEH_ABORT) {
- 				throw FFMS_Exception(FFMS_ERROR_CODEC, FFMS_ERROR_DECODING, "Audio decoding error");
-@@ -756,13 +759,14 @@
- 		Packet->data += Ret;
- 		Read += Ret;
- 
--		CheckAudioProperties(Track, CodecContext);
-+		if (GotFrame) {
-+			CheckAudioProperties(Track, CodecContext);
- 
--		if (dbsize > 0)
--			Context.CurrentSample += dbsize / (av_get_bytes_per_sample(CodecContext->sample_fmt) * CodecContext->channels);
-+			Context.CurrentSample += DecodeFrame->nb_samples;
- 
--		if (DumpMask & (1 << Track))
--			WriteAudio(Context, &TrackIndices, Track, dbsize);
-+			if (DumpMask & (1 << Track))
-+				WriteAudio(Context, &TrackIndices, Track);
-+		}
- 	}
- 	Packet->size += Read;
- 	Packet->data -= Read;
-diff -ru ffmpegsource/src/core/indexing.h ffms2/src/core/indexing.h
---- ffmpegsource/src/core/indexing.h	2013-02-27 16:53:39.127363391 +0100
-+++ ffms2/src/core/indexing.h	2013-02-27 16:53:31.744380192 +0100
-@@ -155,7 +155,6 @@
- };
- 
- struct FFMS_Indexer {
--private:
- 	std::map<int, FFMS_AudioProperties> LastAudioProperties;
- protected:
- 	int IndexMask;
-@@ -166,12 +165,12 @@
- 	TAudioNameCallback ANC;
- 	void *ANCPrivate;
- 	std::string SourceFile;
--	AlignedBuffer<uint8_t> DecodingBuffer;
-+	ScopedFrame DecodeFrame;
- 
- 	int64_t Filesize;
- 	uint8_t Digest[20];
- 
--	void WriteAudio(SharedAudioContext &AudioContext, FFMS_Index *Index, int Track, int DBSize);
-+	void WriteAudio(SharedAudioContext &AudioContext, FFMS_Index *Index, int Track);
- 	void CheckAudioProperties(int Track, AVCodecContext *Context);
- 	int64_t IndexAudioPacket(int Track, AVPacket *Packet, SharedAudioContext &Context, FFMS_Index &TrackIndices);
- 	void ParseVideoPacket(SharedVideoContext &VideoContext, AVPacket &pkt, int *RepeatPict, int *FrameType, bool *Invisible);
-diff -ru ffmpegsource/src/core/utils.cpp ffms2/src/core/utils.cpp
---- ffmpegsource/src/core/utils.cpp	2013-02-27 16:53:39.134029741 +0100
-+++ ffms2/src/core/utils.cpp	2013-02-27 16:53:31.744380192 +0100
-@@ -214,10 +214,32 @@
- 	pkt.size = 0;
- }
- 
-+extern "C" {
-+#if VERSION_CHECK(LIBAVUTIL_VERSION_INT, >=, 52, 2, 0, 52, 6, 100)
-+#include <libavutil/channel_layout.h>
-+#elif VERSION_CHECK(LIBAVUTIL_VERSION_INT, >=, 51, 26, 0, 51, 45, 100)
-+#include <libavutil/audioconvert.h>
-+#else
-+static int64_t av_get_default_channel_layout(int nb_channels) {
-+	switch(nb_channels) {
-+		case 1: return AV_CH_LAYOUT_MONO;
-+		case 2: return AV_CH_LAYOUT_STEREO;
-+		case 3: return AV_CH_LAYOUT_SURROUND;
-+		case 4: return AV_CH_LAYOUT_QUAD;
-+		case 5: return AV_CH_LAYOUT_5POINT0;
-+		case 6: return AV_CH_LAYOUT_5POINT1;
-+		case 7: return AV_CH_LAYOUT_6POINT1;
-+		case 8: return AV_CH_LAYOUT_7POINT1;
-+		default: return 0;
-+	}
-+}
-+#endif
-+}
-+
- void FillAP(FFMS_AudioProperties &AP, AVCodecContext *CTX, FFMS_Track &Frames) {
--	AP.SampleFormat = static_cast<FFMS_SampleFormat>(CTX->sample_fmt);
-+	AP.SampleFormat = static_cast<FFMS_SampleFormat>(av_get_packed_sample_fmt(CTX->sample_fmt));
- 	AP.BitsPerSample = av_get_bytes_per_sample(CTX->sample_fmt) * 8;
--	AP.Channels = CTX->channels;;
-+	AP.Channels = CTX->channels;
- 	AP.ChannelLayout = CTX->channel_layout;
- 	AP.SampleRate = CTX->sample_rate;
- 	if (!Frames.empty()) {
-@@ -225,6 +247,9 @@
- 		AP.FirstTime = ((Frames.front().PTS * Frames.TB.Num) / (double)Frames.TB.Den) / 1000;
- 		AP.LastTime = ((Frames.back().PTS * Frames.TB.Num) / (double)Frames.TB.Den) / 1000;
- 	}
-+
-+	if (AP.ChannelLayout == 0)
-+		AP.ChannelLayout = av_get_default_channel_layout(AP.Channels);
- }
- 
- #ifdef HAALISOURCE
-diff -ru ffmpegsource/src/core/utils.h ffms2/src/core/utils.h
---- ffmpegsource/src/core/utils.h	2013-02-27 16:53:39.127363391 +0100
-+++ ffms2/src/core/utils.h	2013-02-27 16:53:31.744380192 +0100
-@@ -31,9 +31,13 @@
- extern "C" {
- #include "stdiostream.h"
- #include <libavutil/mem.h>
-+#include <libavutil/opt.h>
- #include <libavformat/avformat.h>
- #include <libavcodec/avcodec.h>
- #include <libswscale/swscale.h>
-+#ifdef WITH_AVRESAMPLE
-+#include <libavresample/avresample.h>
-+#endif
- }
- 
- // must be included after ffmpeg headers
-@@ -133,6 +137,34 @@
- 	}
- };
- 
-+template<typename T, T *(*Alloc)(), void (*Del)(T **)>
-+class unknown_size {
-+	T *ptr;
-+
-+	unknown_size(unknown_size const&);
-+	unknown_size& operator=(unknown_size const&);
-+public:
-+	operator T*() const { return ptr; }
-+	operator void*() const { return ptr; }
-+	T *operator->() const { return ptr; }
-+
-+	unknown_size() : ptr(Alloc()) { }
-+	~unknown_size() { Del(&ptr); }
-+};
-+
-+class ScopedFrame : public unknown_size<AVFrame, avcodec_alloc_frame, avcodec_free_frame> {
-+public:
-+	void reset() {
-+		avcodec_get_frame_defaults(*this);
-+	}
-+};
-+
-+#ifdef WITH_AVRESAMPLE
-+typedef unknown_size<AVAudioResampleContext, avresample_alloc_context, avresample_free> FFResampleContext;
-+#else
-+typedef struct {} FFResampleContext;
-+#endif
-+
- inline void DeleteHaaliCodecContext(AVCodecContext *CodecContext) {
- 	av_freep(&CodecContext->extradata);
- 	av_freep(&CodecContext);
-@@ -228,4 +240,68 @@
- 
- void FlushBuffers(AVCodecContext *CodecContext);
- 
-+namespace optdetail {
-+	template<typename T>
-+	T get_av_opt(void *v, const char *name) {
-+		return static_cast<T>(av_get_int(v, name, 0));
-+	}
-+
-+	template<>
-+	inline double get_av_opt<double>(void *v, const char *name) {
-+		return av_get_double(v, name, 0);
-+	}
-+
-+	template<typename T>
-+	void set_av_opt(void *v, const char *name, T value) {
-+		av_opt_set_int(v, name, value, 0);
-+	}
-+
-+	template<>
-+	inline void set_av_opt<double>(void *v, const char *name, double value) {
-+		av_opt_set_double(v, name, value, 0);
-+	}
-+}
-+
-+template<typename FFMS_Struct>
-+class OptionMapper {
-+	struct OptionMapperBase {
-+		virtual void ToOpt(const FFMS_Struct *src, void *dst) const=0;
-+		virtual void FromOpt(FFMS_Struct *dst, void *src) const=0;
-+	};
-+
-+	template<typename T>
-+	class OptionMapperImpl : public OptionMapperBase {
-+		T (FFMS_Struct::*ptr);
-+		const char *name;
-+
-+	public:
-+		OptionMapperImpl(T (FFMS_Struct::*ptr), const char *name) : ptr(ptr), name(name) { }
-+		void ToOpt(const FFMS_Struct *src, void *dst) const { optdetail::set_av_opt(dst, name, src->*ptr); }
-+		void FromOpt(FFMS_Struct *dst, void *src) const { dst->*ptr = optdetail::get_av_opt<T>(src, name); }
-+	};
-+
-+	OptionMapperBase *impl;
-+
-+public:
-+	template<typename T>
-+	OptionMapper(const char *opt_name, T (FFMS_Struct::*member)) : impl(new OptionMapperImpl<T>(member, opt_name)) { }
-+
-+	void ToOpt(const FFMS_Struct *src, void *dst) const { impl->ToOpt(src, dst); }
-+	void FromOpt(FFMS_Struct *dst, void *src) const { impl->FromOpt(dst, src); }
-+};
-+
-+template<typename T, int N>
-+T *ReadOptions(void *opt, OptionMapper<T> (&options)[N]) {
-+	T *ret = new T;
-+	for (int i = 0; i < N; ++i)
-+		options[i].FromOpt(ret, opt);
-+	return ret;
-+}
-+
-+template<typename T, int N>
-+void SetOptions(const T* src, void *opt, OptionMapper<T> (&options)[N]) {
-+	for (int i = 0; i < N; ++i)
-+		options[i].ToOpt(src, opt);
-+}
-+
- #endif
-diff -ru ffmpegsource/src/core/wave64writer.cpp ffms2/src/core/wave64writer.cpp
---- ffmpegsource/src/core/wave64writer.cpp	2013-02-27 16:53:39.134029741 +0100
-+++ ffms2/src/core/wave64writer.cpp	2013-02-27 16:53:31.744380192 +0100
-@@ -106,7 +106,16 @@
- 		WavFile.seekp(CPos, std::ios::beg);
- }
- 
--void Wave64Writer::WriteData(void *Data, std::streamsize Length) {
--	WavFile.write(reinterpret_cast<char *>(Data), Length);
-+void Wave64Writer::WriteData(AVFrame const& Frame) {
-+	uint64_t Length = Frame.nb_samples * BytesPerSample * Channels;
-+	if (Channels > 1 && av_sample_fmt_is_planar(static_cast<AVSampleFormat>(Frame.format))) {
-+		for (int32_t sample = 0; sample < Frame.nb_samples; ++sample) {
-+			for (int32_t channel = 0; channel < Channels; ++channel)
-+				WavFile.write(reinterpret_cast<char *>(&Frame.extended_data[channel][sample * BytesPerSample]), BytesPerSample);
-+		}
-+	}
-+	else {
-+		WavFile.write(reinterpret_cast<char *>(Frame.extended_data[0]), Length);
-+	}
- 	BytesWritten += Length;
- }
-diff -ru /tmp/ffmpegsource/src/ffmpegsource/src/core/wave64writer.h ffms2/src/core/wave64writer.h
---- /tmp/ffmpegsource/src/ffmpegsource/src/core/wave64writer.h	2013-02-27 16:53:39.127363391 +0100
-+++ ffms2/src/core/wave64writer.h	2013-02-27 16:53:31.744380192 +0100
-@@ -28,8 +28,8 @@
- class Wave64Writer {
- public:
- 	Wave64Writer(const char *Filename, uint16_t BitsPerSample, uint16_t Channels, uint32_t SamplesPerSec, bool IsFloat);
- 	~Wave64Writer();
--	void WriteData(void *Data, std::streamsize Length);
-+	void WriteData(AVFrame const& Frame);
- private:
- 	ffms_fstream WavFile;
- 	int32_t BytesPerSample;
diff --git a/community/intellij-idea-libs/PKGBUILD b/community/intellij-idea-libs/PKGBUILD
index f2e0d699c..6cd5c12f5 100644
--- a/community/intellij-idea-libs/PKGBUILD
+++ b/community/intellij-idea-libs/PKGBUILD
@@ -1,16 +1,17 @@
-# $Id: PKGBUILD 87801 2013-04-06 19:27:42Z stativ $
+# $Id: PKGBUILD 88193 2013-04-13 15:39:41Z stativ $
 # Maintainer: Lukas Jirkovsky <l.jirkovsky@gmail.com>
 pkgname=intellij-idea-libs
-pkgver=12.1
-_pkgver=129.161
+pkgver=12.1.1
+_pkgver=129.239
 pkgrel=1
 pkgdesc="Architecture dependend libraries needed by the Intellij Idea IDE"
 arch=('i686' 'x86_64')
 url="http://www.jetbrains.org/"
 license=('apache')
 depends=('glibc')
+options=(!strip)
 source=(http://download.jetbrains.com/idea/ideaIC-$pkgver.tar.gz)
-md5sums=('687ba6aea524099f10605bb50569864d')
+md5sums=('279905aa380822f8a72a01e2b4955b44')
 
 package() {
   cd "$srcdir"
diff --git a/community/nodejs/PKGBUILD b/community/nodejs/PKGBUILD
index b466757ed..571fc58a8 100644
--- a/community/nodejs/PKGBUILD
+++ b/community/nodejs/PKGBUILD
@@ -1,4 +1,4 @@
-# $Id: PKGBUILD 87751 2013-04-06 09:18:35Z bpiotrowski $
+# $Id: PKGBUILD 88176 2013-04-13 10:02:06Z bpiotrowski $
 # Maintainer:  Bartłomiej Piotrowski <nospam@bpiotrowski.pl>
 # Contributor: Thomas Dziedzic < gostrc at gmail >
 # Contributor: James Campos <james.r.campos@gmail.com>
@@ -8,7 +8,7 @@
 # Contributor: TIanyi Cui <tianyicui@gmail.com>
 
 pkgname=nodejs
-pkgver=0.10.3
+pkgver=0.10.4
 pkgrel=1
 pkgdesc='Evented I/O for V8 javascript'
 arch=('i686' 'x86_64')
@@ -18,7 +18,7 @@ depends=('openssl' 'python2')
 checkdepends=('curl')
 options=('!emptydirs')
 source=(http://nodejs.org/dist/v${pkgver}/node-v${pkgver}.tar.gz)
-sha256sums=('bc8796ff6414231fa0603e0383404f14648dfd2fe9fb0fa4d4a6043dfddbb328')
+sha256sums=('1c960d2822447a9e4f7c46b832ff05e86743033c6643d644975af1cbf6a44fb8')
 
 build() {
   cd node-v${pkgver}
diff --git a/community/python-psutil/PKGBUILD b/community/python-psutil/PKGBUILD
index 4b00d7e8c..041dcacb4 100644
--- a/community/python-psutil/PKGBUILD
+++ b/community/python-psutil/PKGBUILD
@@ -1,16 +1,16 @@
-# $Id: PKGBUILD 78236 2012-10-17 10:26:22Z allan $
-# Maintainer: Sébastien Luttringer <seblu@aur.archlinux.org>
+# $Id: PKGBUILD 88178 2013-04-13 10:41:30Z seblu $
+# Maintainer: Sébastien Luttringer
 
 pkgbase=python-psutil
 pkgname=('python-psutil' 'python2-psutil')
-pkgver=0.6.1
-pkgrel=2
+pkgver=0.7.0
+pkgrel=1
 arch=('i686' 'x86_64')
 url='http://code.google.com/p/psutil/'
 license=('custom: BSD')
 makedepends=('python' 'python-distribute' 'python2' 'python2-distribute')
 source=("https://psutil.googlecode.com/files/psutil-$pkgver.tar.gz") 
-sha1sums=('f7a76e201601d8e06a1fdf434422f884888aac86')
+sha1sums=('e19bd1c050786616dd81a3b200c61b8147eb9985')
 
 build() {
   cd psutil-$pkgver
@@ -40,4 +40,4 @@ package_python2-psutil() {
   install -D -m 644 LICENSE "$pkgdir/"usr/share/licenses/$pkgname/LICENSE
 }
 
-# vim:set ts=2 sw=2 ft=sh et:
+# vim:set ts=2 sw=2 et: