Date: Saturday, April 13, 2013 @ 22:20:12
Author: alucryd
Revision: 88203
upgpkg: ffmpegsource 753-1
Modified:
ffmpegsource/trunk/PKGBUILD
Deleted:
ffmpegsource/trunk/enable-libavresample.patch
----------------------------+
PKGBUILD | 48 --
enable-libavresample.patch | 970 -------------------------------------------
2 files changed, 17 insertions(+), 1001 deletions(-)
Modified: PKGBUILD
===================================================================
--- PKGBUILD 2013-04-13 19:43:05 UTC (rev 88202)
+++ PKGBUILD 2013-04-13 20:20:12 UTC (rev 88203)
@@ -2,8 +2,8 @@
# Maintainer: Maxime Gauduin <aluc...@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 @@
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}"
+ svnversion | tr -d [A-z]
+}
-# Checkout
- msg "Connecting to SVN server...."
+prepare() {
+ cd "${srcdir}"/${pkgname}
- if [[ -d ${_svnmod}/.svn ]]; then
- (cd ${_svnmod} && svn up -r ${pkgver})
- else
- svn co ${_svntrunk} --config-dir ./ -r ${pkgver} ${_svnmod}
- fi
+ patch -Np1 -i ../autoconf.patch
+}
- msg "SVN checkout done or server timeout"
- msg "Starting build..."
+build() {
+ cd "${srcdir}"/${pkgname}
- rm -rf "${srcdir}"/${_svnmod}-build
-# svn export "${srcdir}"/${_svnmod} "${srcdir}"/${_svnmod}-build
- cp -R "${srcdir}"/${_svnmod} "${srcdir}"/${_svnmod}-build
- cd "${srcdir}"/${_svnmod}-build
-
-# Patch
- patch -Np1 -i "${srcdir}"/autoconf.patch
- patch -Np1 -i "${srcdir}"/enable-libavresample.patch
-
-# Build
./autogen.sh --prefix=/usr --enable-shared --disable-static
make
}
package() {
- cd "${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
}
Deleted: enable-libavresample.patch
===================================================================
--- enable-libavresample.patch 2013-04-13 19:43:05 UTC (rev 88202)
+++ enable-libavresample.patch 2013-04-13 20:20:12 UTC (rev 88203)
@@ -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;
