[arch-commits] Commit in ffmpegsource/trunk (PKGBUILD enable-libavresample.patch)
Maxime Gauduin
alucryd at nymeria.archlinux.org
Wed Feb 27 18:14:13 UTC 2013
Date: Wednesday, February 27, 2013 @ 19:14:12
Author: alucryd
Revision: 85214
upgpkg: ffmpegsource 743-2
Added:
ffmpegsource/trunk/enable-libavresample.patch
Modified:
ffmpegsource/trunk/PKGBUILD
----------------------------+
PKGBUILD | 13
enable-libavresample.patch | 970 +++++++++++++++++++++++++++++++++++++++++++
2 files changed, 979 insertions(+), 4 deletions(-)
Modified: PKGBUILD
===================================================================
--- PKGBUILD 2013-02-27 18:06:27 UTC (rev 85213)
+++ PKGBUILD 2013-02-27 18:14:12 UTC (rev 85214)
@@ -3,7 +3,7 @@
pkgname=ffmpegsource
pkgver=743
-pkgrel=1
+pkgrel=2
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/"
@@ -11,14 +11,17 @@
depends=('ffmpeg')
makedepends=('svn')
options=('!libtool')
-source=('autoconf.patch')
-sha256sums=('b09a7e9a08a16bdaf19d43c7ad8d3ec455f6fecec2f4f5ada417345343adda93')
+source=('autoconf.patch' 'enable-libavresample.patch')
+sha256sums=('b09a7e9a08a16bdaf19d43c7ad8d3ec455f6fecec2f4f5ada417345343adda93'
+ '05f03515cc2405cdf8a8ba835f5adc2057f40054a4a1d9e493f0ad512c5de70d')
_svntrunk=http://ffmpegsource.googlecode.com/svn/trunk/
_svnmod=ffmpegsource
build() {
cd "${srcdir}"
+
+# Checkout
msg "Connecting to SVN server...."
if [[ -d ${_svnmod}/.svn ]]; then
@@ -31,11 +34,13 @@
msg "Starting build..."
rm -rf "${srcdir}"/${_svnmod}-build
- svn export "${srcdir}"/${_svnmod} "${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 "${srcdir}"/enable-libavresample.patch
# Build
./autogen.sh --prefix=/usr --enable-shared --disable-static
Added: enable-libavresample.patch
===================================================================
--- enable-libavresample.patch (rev 0)
+++ enable-libavresample.patch 2013-02-27 18:14:12 UTC (rev 85214)
@@ -0,0 +1,970 @@
+# enable-libavresample.patch
+#
+# Adds libavresample support. Created by diffing Thomas Goyne's GIT repo
+# with official ffms SVN.
+#
+
+diff -ru ffmpegsource/configure.ac ffms2/configure.ac
+--- ffmpegsource/configure.ac 2013-02-27 16:53:39.230691825 +0100
++++ ffms2/configure.ac 2013-02-27 16:53:31.737713841 +0100
+@@ -181,6 +181,25 @@
+ AC_MSG_RESULT([no])
+ ])
+
++AC_ARG_ENABLE(avresample,
++ AS_HELP_STRING([--enable-avresample],
++ [use libavresample for audio resampling]))
++AS_IF([test x$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;
Property changes on: ffmpegsource/trunk/enable-libavresample.patch
___________________________________________________________________
Added: svn:executable
## -0,0 +1 ##
+*
\ No newline at end of property
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