From 8ec4fd7a9e4748c568b20fefcb2dec0f289dc213 Mon Sep 17 00:00:00 2001 From: root Date: Sun, 14 Apr 2013 00:46:08 -0700 Subject: Sun Apr 14 00:46:05 PDT 2013 --- community/csfml/PKGBUILD | 49 +- community/ffmpegsource/PKGBUILD | 50 +- community/ffmpegsource/enable-libavresample.patch | 970 ---------------------- community/intellij-idea-libs/PKGBUILD | 9 +- community/nodejs/PKGBUILD | 6 +- community/python-psutil/PKGBUILD | 12 +- 6 files changed, 44 insertions(+), 1052 deletions(-) delete mode 100755 community/ffmpegsource/enable-libavresample.patch (limited to 'community') 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 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 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 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 - #include - -+namespace { -+ -+ int64_t ChannelLayout; -+ FFMS_SampleFormat SampleFormat; -+ int SampleRate; -+#define MAPPER(m, n) OptionMapper(n, &FFMS_ResampleOptions::m) -+OptionMapper 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(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 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(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 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(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(&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(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(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(Start - CurrentSample); -- size_t Samples = static_cast(Decoded - FirstSample); -- size_t Bytes = FFMIN(Samples, static_cast(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 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 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 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 LastAudioProperties; - protected: - int IndexMask; -@@ -166,12 +165,12 @@ - TAudioNameCallback ANC; - void *ANCPrivate; - std::string SourceFile; -- AlignedBuffer 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 -+#elif VERSION_CHECK(LIBAVUTIL_VERSION_INT, >=, 51, 26, 0, 51, 45, 100) -+#include -+#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(CTX->sample_fmt); -+ AP.SampleFormat = static_cast(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 -+#include - #include - #include - #include -+#ifdef WITH_AVRESAMPLE -+#include -+#endif - } - - // must be included after ffmpeg headers -@@ -133,6 +137,34 @@ - } - }; - -+template -+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 { -+public: -+ void reset() { -+ avcodec_get_frame_defaults(*this); -+ } -+}; -+ -+#ifdef WITH_AVRESAMPLE -+typedef unknown_size 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 -+ T get_av_opt(void *v, const char *name) { -+ return static_cast(av_get_int(v, name, 0)); -+ } -+ -+ template<> -+ inline double get_av_opt(void *v, const char *name) { -+ return av_get_double(v, name, 0); -+ } -+ -+ template -+ 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(void *v, const char *name, double value) { -+ av_opt_set_double(v, name, value, 0); -+ } -+} -+ -+template -+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 -+ 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(src, name); } -+ }; -+ -+ OptionMapperBase *impl; -+ -+public: -+ template -+ OptionMapper(const char *opt_name, T (FFMS_Struct::*member)) : impl(new OptionMapperImpl(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 -+T *ReadOptions(void *opt, OptionMapper (&options)[N]) { -+ T *ret = new T; -+ for (int i = 0; i < N; ++i) -+ options[i].FromOpt(ret, opt); -+ return ret; -+} -+ -+template -+void SetOptions(const T* src, void *opt, OptionMapper (&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(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(Frame.format))) { -+ for (int32_t sample = 0; sample < Frame.nb_samples; ++sample) { -+ for (int32_t channel = 0; channel < Channels; ++channel) -+ WavFile.write(reinterpret_cast(&Frame.extended_data[channel][sample * BytesPerSample]), BytesPerSample); -+ } -+ } -+ else { -+ WavFile.write(reinterpret_cast(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 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 # Contributor: Thomas Dziedzic < gostrc at gmail > # Contributor: James Campos @@ -8,7 +8,7 @@ # Contributor: TIanyi Cui 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 +# $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: -- cgit v1.2.3-54-g00ecf