# 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;