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authorKostya Shishkov <kostya.shishkov@gmail.com>2007-09-13 03:22:47 +0000
committerKostya Shishkov <kostya.shishkov@gmail.com>2007-09-13 03:22:47 +0000
commitbf4a1f17ee9237b6efd4250cf894e274afc31a5f (patch)
tree0c0e605b516d2e9bb64181daa2ec0819d078eb34
parent48fe9238a0aec437aa9ab9a8912191d163feb519 (diff)
downloadffmpeg-bf4a1f17ee9237b6efd4250cf894e274afc31a5f.tar.gz
Monkey Audio decoder
Originally committed as revision 10484 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--Changelog1
-rw-r--r--doc/general.texi2
-rw-r--r--libavcodec/Makefile1
-rw-r--r--libavcodec/allcodecs.c1
-rw-r--r--libavcodec/allcodecs.h1
-rw-r--r--libavcodec/apedec.c922
-rw-r--r--libavcodec/avcodec.h5
-rw-r--r--libavformat/Makefile1
-rw-r--r--libavformat/allformats.c1
-rw-r--r--libavformat/allformats.h1
-rw-r--r--libavformat/ape.c392
-rw-r--r--libavformat/avformat.h4
12 files changed, 1328 insertions, 4 deletions
diff --git a/Changelog b/Changelog
index f746ca6785..97326f930f 100644
--- a/Changelog
+++ b/Changelog
@@ -94,6 +94,7 @@ version <next>
- NUT muxer (since r10052)
- Matroska muxer
- Slice-based parallel H.264 decoding
+- Monkey's Audio demuxer and decoder
version 0.4.9-pre1:
diff --git a/doc/general.texi b/doc/general.texi
index 13c67a5462..542233fbf6 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -116,6 +116,7 @@ different game cutscenes repacked for use with ScummVM.
@tab Used in some games from Bethesda Softworks.
@item CRYO APC @tab @tab X
@tab Audio format used in some games by CRYO Interactive Entertainment.
+@item Monkey's Audio @tab @tab X
@end multitable
@code{X} means that encoding (resp. decoding) is supported.
@@ -311,6 +312,7 @@ following image formats are supported:
@tab Only SV7 is supported
@item DT$ Coherent Audio @tab @tab X
@item ATRAC 3 @tab @tab X
+@item Monkey's Audio @tab @tab X @tab Only versions 3.97-3.99 are supported
@end multitable
@code{X} means that encoding (resp. decoding) is supported.
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 84742d4c06..3efe55295f 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -35,6 +35,7 @@ OBJS-$(CONFIG_AASC_DECODER) += aasc.o
OBJS-$(CONFIG_AC3_DECODER) += ac3dec.o ac3tab.o ac3.o mdct.o fft.o
OBJS-$(CONFIG_AC3_ENCODER) += ac3enc.o ac3tab.o ac3.o
OBJS-$(CONFIG_ALAC_DECODER) += alac.o
+OBJS-$(CONFIG_APE_DECODER) += apedec.o
OBJS-$(CONFIG_ASV1_DECODER) += asv1.o
OBJS-$(CONFIG_ASV1_ENCODER) += asv1.o
OBJS-$(CONFIG_ASV2_DECODER) += asv1.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 1362451d20..729710474a 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -168,6 +168,7 @@ void avcodec_register_all(void)
REGISTER_DECODER (MPEG4AAC, mpeg4aac);
REGISTER_ENCDEC (AC3, ac3);
REGISTER_DECODER (ALAC, alac);
+ REGISTER_DECODER (APE, ape);
REGISTER_DECODER (ATRAC3, atrac3);
REGISTER_DECODER (COOK, cook);
REGISTER_DECODER (DCA, dca);
diff --git a/libavcodec/allcodecs.h b/libavcodec/allcodecs.h
index 34efcb3bfa..4e94c37ccb 100644
--- a/libavcodec/allcodecs.h
+++ b/libavcodec/allcodecs.h
@@ -79,6 +79,7 @@ extern AVCodec zmbv_encoder;
extern AVCodec aasc_decoder;
extern AVCodec ac3_decoder;
extern AVCodec alac_decoder;
+extern AVCodec ape_decoder;
extern AVCodec asv1_decoder;
extern AVCodec asv2_decoder;
extern AVCodec atrac3_decoder;
diff --git a/libavcodec/apedec.c b/libavcodec/apedec.c
new file mode 100644
index 0000000000..68358e1868
--- /dev/null
+++ b/libavcodec/apedec.c
@@ -0,0 +1,922 @@
+/*
+ * Monkey's Audio lossless audio decoder
+ * Copyright (c) 2007 Benjamin Zores <ben@geexbox.org>
+ * based upon libdemac from Dave Chapman.
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#define ALT_BITSTREAM_READER_LE
+#include "avcodec.h"
+#include "dsputil.h"
+#include "bitstream.h"
+#include "bytestream.h"
+
+/**
+ * @file apedec.c
+ * Monkey's Audio lossless audio decoder
+ */
+
+#define BLOCKS_PER_LOOP 4608
+#define MAX_CHANNELS 2
+#define MAX_BYTESPERSAMPLE 3
+
+#define APE_FRAMECODE_MONO_SILENCE 1
+#define APE_FRAMECODE_STEREO_SILENCE 3
+#define APE_FRAMECODE_PSEUDO_STEREO 4
+
+#define HISTORY_SIZE 512
+#define PREDICTOR_ORDER 8
+/** Total size of all predictor histories */
+#define PREDICTOR_SIZE 50
+
+#define YDELAYA (18 + PREDICTOR_ORDER*4)
+#define YDELAYB (18 + PREDICTOR_ORDER*3)
+#define XDELAYA (18 + PREDICTOR_ORDER*2)
+#define XDELAYB (18 + PREDICTOR_ORDER)
+
+#define YADAPTCOEFFSA 18
+#define XADAPTCOEFFSA 14
+#define YADAPTCOEFFSB 10
+#define XADAPTCOEFFSB 5
+
+/**
+ * Possible compression levels
+ * @{
+ */
+enum APECompressionLevel {
+ COMPRESSION_LEVEL_FAST = 1000,
+ COMPRESSION_LEVEL_NORMAL = 2000,
+ COMPRESSION_LEVEL_HIGH = 3000,
+ COMPRESSION_LEVEL_EXTRA_HIGH = 4000,
+ COMPRESSION_LEVEL_INSANE = 5000
+};
+/** @} */
+
+#define APE_FILTER_LEVELS 3
+
+/** Filter orders depending on compression level */
+static const uint16_t ape_filter_orders[5][APE_FILTER_LEVELS] = {
+ { 0, 0, 0 },
+ { 16, 0, 0 },
+ { 64, 0, 0 },
+ { 32, 256, 0 },
+ { 16, 256, 1280 }
+};
+
+/** Filter fraction bits depending on compression level */
+static const uint16_t ape_filter_fracbits[5][APE_FILTER_LEVELS] = {
+ { 0, 0, 0 },
+ { 11, 0, 0 },
+ { 11, 0, 0 },
+ { 10, 13, 0 },
+ { 11, 13, 15 }
+};
+
+
+/** Filters applied to the decoded data */
+typedef struct APEFilter {
+ int16_t *coeffs; ///< actual coefficients used in filtering
+ int16_t *adaptcoeffs; ///< adaptive filter coefficients used for correcting of actual filter coefficients
+ int16_t *historybuffer; ///< filter memory
+ int16_t *delay; ///< filtered values
+
+ int avg;
+} APEFilter;
+
+typedef struct APERice {
+ uint32_t k;
+ uint32_t ksum;
+} APERice;
+
+typedef struct APERangecoder {
+ uint32_t low; ///< low end of interval
+ uint32_t range; ///< length of interval
+ uint32_t help; ///< bytes_to_follow resp. intermediate value
+ unsigned int buffer; ///< buffer for input/output
+} APERangecoder;
+
+/** Filter histories */
+typedef struct APEPredictor {
+ int32_t *buf;
+
+ int32_t lastA[2];
+
+ int32_t filterA[2];
+ int32_t filterB[2];
+
+ int32_t coeffsA[2][4]; ///< adaption coefficients
+ int32_t coeffsB[2][5]; ///< adaption coefficients
+ int32_t historybuffer[HISTORY_SIZE + PREDICTOR_SIZE];
+} APEPredictor;
+
+/** Decoder context */
+typedef struct APEContext {
+ AVCodecContext *avctx;
+ DSPContext dsp;
+ int channels;
+ int samples; ///< samples left to decode in current frame
+
+ int fileversion; ///< codec version, very important in decoding process
+ int compression_level; ///< compression levels
+ int fset; ///< which filter set to use (calculated from compression level)
+ int flags; ///< global decoder flags
+
+ uint32_t CRC; ///< frame CRC
+ int frameflags; ///< frame flags
+ int currentframeblocks; ///< samples (per channel) in current frame
+ int blocksdecoded; ///< count of decoded samples in current frame
+ APEPredictor predictor; ///< predictor used for final reconstruction
+
+ int32_t decoded0[BLOCKS_PER_LOOP]; ///< decoded data for the first channel
+ int32_t decoded1[BLOCKS_PER_LOOP]; ///< decoded data for the second channel
+
+ int16_t* filterbuf[APE_FILTER_LEVELS]; ///< filter memory
+
+ APERangecoder rc; ///< rangecoder used to decode actual values
+ APERice riceX; ///< rice code parameters for the second channel
+ APERice riceY; ///< rice code parameters for the first channel
+ APEFilter filters[APE_FILTER_LEVELS][2]; ///< filters used for reconstruction
+
+ uint8_t *data; ///< current frame data
+ uint8_t *data_end; ///< frame data end
+ uint8_t *ptr; ///< current position in frame data
+ uint8_t *last_ptr; ///< position where last 4608-sample block ended
+} APEContext;
+
+// TODO: dsputilize
+static inline void vector_add(int16_t * v1, int16_t * v2, int order)
+{
+ while (order--)
+ *v1++ += *v2++;
+}
+
+// TODO: dsputilize
+static inline void vector_sub(int16_t * v1, int16_t * v2, int order)
+{
+ while (order--)
+ *v1++ -= *v2++;
+}
+
+// TODO: dsputilize
+static inline int32_t scalarproduct(int16_t * v1, int16_t * v2, int order)
+{
+ int res = 0;
+
+ while (order--)
+ res += *v1++ * *v2++;
+
+ return res;
+}
+
+static int ape_decode_init(AVCodecContext * avctx)
+{
+ APEContext *s = avctx->priv_data;
+ int i;
+
+ if (avctx->extradata_size != 6) {
+ av_log(avctx, AV_LOG_ERROR, "Incorrect extradata\n");
+ return -1;
+ }
+ if (avctx->bits_per_sample != 16) {
+ av_log(avctx, AV_LOG_ERROR, "Only 16-bit samples are supported\n");
+ return -1;
+ }
+ if (avctx->channels > 2) {
+ av_log(avctx, AV_LOG_ERROR, "Only mono and stereo is supported\n");
+ return -1;
+ }
+ s->avctx = avctx;
+ s->channels = avctx->channels;
+ s->fileversion = AV_RL16(avctx->extradata);
+ s->compression_level = AV_RL16(avctx->extradata + 2);
+ s->flags = AV_RL16(avctx->extradata + 4);
+
+ av_log(avctx, AV_LOG_DEBUG, "Compression Level: %d - Flags: %d\n", s->compression_level, s->flags);
+ if (s->compression_level % 1000 || s->compression_level > COMPRESSION_LEVEL_INSANE) {
+ av_log(avctx, AV_LOG_ERROR, "Incorrect compression level %d\n", s->compression_level);
+ return -1;
+ }
+ s->fset = s->compression_level / 1000 - 1;
+ for (i = 0; i < APE_FILTER_LEVELS; i++) {
+ if (!ape_filter_orders[s->fset][i])
+ break;
+ s->filterbuf[i] = av_malloc((ape_filter_orders[s->fset][i] * 3 + HISTORY_SIZE) * 4);
+ }
+
+ dsputil_init(&s->dsp, avctx);
+ return 0;
+}
+
+static int ape_decode_close(AVCodecContext * avctx)
+{
+ APEContext *s = avctx->priv_data;
+ int i;
+
+ for (i = 0; i < APE_FILTER_LEVELS; i++)
+ av_freep(&s->filterbuf[i]);
+
+ return 0;
+}
+
+/**
+ * @defgroup rangecoder APE range decoder
+ * @{
+ */
+
+#define CODE_BITS 32
+#define TOP_VALUE ((unsigned int)1 << (CODE_BITS-1))
+#define SHIFT_BITS (CODE_BITS - 9)
+#define EXTRA_BITS ((CODE_BITS-2) % 8 + 1)
+#define BOTTOM_VALUE (TOP_VALUE >> 8)
+
+/** Start the decoder */
+static inline void range_start_decoding(APEContext * ctx)
+{
+ ctx->rc.buffer = bytestream_get_byte(&ctx->ptr);
+ ctx->rc.low = ctx->rc.buffer >> (8 - EXTRA_BITS);
+ ctx->rc.range = (uint32_t) 1 << EXTRA_BITS;
+}
+
+/** Perform normalization */
+static inline void range_dec_normalize(APEContext * ctx)
+{
+ while (ctx->rc.range <= BOTTOM_VALUE) {
+ ctx->rc.buffer = (ctx->rc.buffer << 8) | bytestream_get_byte(&ctx->ptr);
+ ctx->rc.low = (ctx->rc.low << 8) | ((ctx->rc.buffer >> 1) & 0xFF);
+ ctx->rc.range <<= 8;
+ }
+}
+
+/**
+ * Calculate culmulative frequency for next symbol. Does NO update!
+ * @param tot_f is the total frequency or (code_value)1<<shift
+ * @return the culmulative frequency
+ */
+static inline int range_decode_culfreq(APEContext * ctx, int tot_f)
+{
+ range_dec_normalize(ctx);
+ ctx->rc.help = ctx->rc.range / tot_f;
+ return ctx->rc.low / ctx->rc.help;
+}
+
+/**
+ * Decode value with given size in bits
+ * @param shift number of bits to decode
+ */
+static inline int range_decode_culshift(APEContext * ctx, int shift)
+{
+ range_dec_normalize(ctx);
+ ctx->rc.help = ctx->rc.range >> shift;
+ return ctx->rc.low / ctx->rc.help;
+}
+
+
+/**
+ * Update decoding state
+ * @param sy_f the interval length (frequency of the symbol)
+ * @param lt_f the lower end (frequency sum of < symbols)
+ */
+static inline void range_decode_update(APEContext * ctx, int sy_f, int lt_f)
+{
+ ctx->rc.low -= ctx->rc.help * lt_f;
+ ctx->rc.range = ctx->rc.help * sy_f;
+}
+
+/** Decode n bits (n <= 16) without modelling */
+static inline int range_decode_bits(APEContext * ctx, int n)
+{
+ int sym = range_decode_culshift(ctx, n);
+ range_decode_update(ctx, 1, sym);
+ return sym;
+}
+
+
+#define MODEL_ELEMENTS 64
+
+/**
+ * Fixed probabilities for symbols in Monkey Audio version 3.97
+ */
+static const uint32_t counts_3970[65] = {
+ 0, 14824, 28224, 39348, 47855, 53994, 58171, 60926,
+ 62682, 63786, 64463, 64878, 65126, 65276, 65365, 65419,
+ 65450, 65469, 65480, 65487, 65491, 65493, 65494, 65495,
+ 65496, 65497, 65498, 65499, 65500, 65501, 65502, 65503,
+ 65504, 65505, 65506, 65507, 65508, 65509, 65510, 65511,
+ 65512, 65513, 65514, 65515, 65516, 65517, 65518, 65519,
+ 65520, 65521, 65522, 65523, 65524, 65525, 65526, 65527,
+ 65528, 65529, 65530, 65531, 65532, 65533, 65534, 65535,
+ 65536
+};
+
+/**
+ * Probability ranges for symbols in Monkey Audio version 3.97
+ */
+static const uint16_t counts_diff_3970[64] = {
+ 14824, 13400, 11124, 8507, 6139, 4177, 2755, 1756,
+ 1104, 677, 415, 248, 150, 89, 54, 31,
+ 19, 11, 7, 4, 2, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1
+};
+
+/**
+ * Fixed probabilities for symbols in Monkey Audio version 3.98
+ */
+static const uint32_t counts_3980[65] = {
+ 0, 19578, 36160, 48417, 56323, 60899, 63265, 64435,
+ 64971, 65232, 65351, 65416, 65447, 65466, 65476, 65482,
+ 65485, 65488, 65490, 65491, 65492, 65493, 65494, 65495,
+ 65496, 65497, 65498, 65499, 65500, 65501, 65502, 65503,
+ 65504, 65505, 65506, 65507, 65508, 65509, 65510, 65511,
+ 65512, 65513, 65514, 65515, 65516, 65517, 65518, 65519,
+ 65520, 65521, 65522, 65523, 65524, 65525, 65526, 65527,
+ 65528, 65529, 65530, 65531, 65532, 65533, 65534, 65535,
+ 65536
+};
+
+/**
+ * Probability ranges for symbols in Monkey Audio version 3.98
+ */
+static const uint16_t counts_diff_3980[64] = {
+ 19578, 16582, 12257, 7906, 4576, 2366, 1170, 536,
+ 261, 119, 65, 31, 19, 10, 6, 3,
+ 3, 2, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1
+};
+
+/**
+ * Decode symbol
+ * @param counts probability range start position
+ * @param count_diffs probability range widths
+ */
+static inline int range_get_symbol(APEContext * ctx,
+ const uint32_t counts[],
+ const uint16_t counts_diff[])
+{
+ int symbol, cf;
+
+ cf = range_decode_culshift(ctx, 16);
+
+ /* figure out the symbol inefficiently; a binary search would be much better */
+ for (symbol = 0; counts[symbol + 1] <= cf; symbol++);
+
+ range_decode_update(ctx, counts_diff[symbol], counts[symbol]);
+
+ return symbol;
+}
+/** @} */ // group rangecoder
+
+static inline void update_rice(APERice *rice, int x)
+{
+ rice->ksum += ((x + 1) / 2) - ((rice->ksum + 16) >> 5);
+
+ if (rice->k == 0)
+ rice->k = 1;
+ else if (rice->ksum < (1 << (rice->k + 4)))
+ rice->k--;
+ else if (rice->ksum >= (1 << (rice->k + 5)))
+ rice->k++;
+}
+
+static inline int ape_decode_value(APEContext * ctx, APERice *rice)
+{
+ int x, overflow;
+
+ if (ctx->fileversion < 3980) {
+ int tmpk;
+
+ overflow = range_get_symbol(ctx, counts_3970, counts_diff_3970);
+
+ if (overflow == (MODEL_ELEMENTS - 1)) {
+ tmpk = range_decode_bits(ctx, 5);
+ overflow = 0;
+ } else
+ tmpk = (rice->k < 1) ? 0 : rice->k - 1;
+
+ if (tmpk <= 16)
+ x = range_decode_bits(ctx, tmpk);
+ else {
+ x = range_decode_bits(ctx, 16);
+ x |= (range_decode_bits(ctx, tmpk - 16) << 16);
+ }
+ x += overflow << tmpk;
+ } else {
+ int base, pivot;
+
+ pivot = rice->ksum >> 5;
+ if (pivot == 0)
+ pivot = 1;
+
+ overflow = range_get_symbol(ctx, counts_3980, counts_diff_3980);
+
+ if (overflow == (MODEL_ELEMENTS - 1)) {
+ overflow = range_decode_bits(ctx, 16) << 16;
+ overflow |= range_decode_bits(ctx, 16);
+ }
+
+ base = range_decode_culfreq(ctx, pivot);
+ range_decode_update(ctx, 1, base);
+
+ x = base + overflow * pivot;
+ }
+
+ update_rice(rice, x);
+
+ /* Convert to signed */
+ if (x & 1)
+ return (x >> 1) + 1;
+ else
+ return -(x >> 1);
+}
+
+static void entropy_decode(APEContext * ctx, int blockstodecode, int stereo)
+{
+ int32_t *decoded0 = ctx->decoded0;
+ int32_t *decoded1 = ctx->decoded1;
+
+ ctx->blocksdecoded = blockstodecode;
+
+ if (ctx->frameflags & APE_FRAMECODE_STEREO_SILENCE) {
+ /* We are pure silence, just memset the output buffer. */
+ memset(decoded0, 0, blockstodecode * sizeof(int32_t));
+ memset(decoded1, 0, blockstodecode * sizeof(int32_t));
+ } else {
+ while (blockstodecode--) {
+ *decoded0++ = ape_decode_value(ctx, &ctx->riceY);
+ if (stereo)
+ *decoded1++ = ape_decode_value(ctx, &ctx->riceX);
+ }
+ }
+
+ if (ctx->blocksdecoded == ctx->currentframeblocks)
+ range_dec_normalize(ctx); /* normalize to use up all bytes */
+}
+
+static void init_entropy_decoder(APEContext * ctx)
+{
+ /* Read the CRC */
+ ctx->CRC = bytestream_get_be32(&ctx->ptr);
+
+ /* Read the frame flags if they exist */
+ ctx->frameflags = 0;
+ if ((ctx->fileversion > 3820) && (ctx->CRC & 0x80000000)) {
+ ctx->CRC &= ~0x80000000;
+
+ ctx->frameflags = bytestream_get_be32(&ctx->ptr);
+ }
+
+ /* Keep a count of the blocks decoded in this frame */
+ ctx->blocksdecoded = 0;
+
+ /* Initialise the rice structs */
+ ctx->riceX.k = 10;
+ ctx->riceX.ksum = (1 << ctx->riceX.k) * 16;
+ ctx->riceY.k = 10;
+ ctx->riceY.ksum = (1 << ctx->riceY.k) * 16;
+
+ /* The first 8 bits of input are ignored. */
+ ctx->ptr++;
+
+ range_start_decoding(ctx);
+}
+
+static const int32_t initial_coeffs[4] = {
+ 360, 317, -109, 98
+};
+
+static void init_predictor_decoder(APEContext * ctx)
+{
+ APEPredictor *p = &ctx->predictor;
+
+ /* Zero the history buffers */
+ memset(p->historybuffer, 0, PREDICTOR_SIZE * sizeof(int32_t));
+ p->buf = p->historybuffer;
+
+ /* Initialise and zero the co-efficients */
+ memcpy(p->coeffsA[0], initial_coeffs, sizeof(initial_coeffs));
+ memcpy(p->coeffsA[1], initial_coeffs, sizeof(initial_coeffs));
+ memset(p->coeffsB, 0, sizeof(p->coeffsB));
+
+ p->filterA[0] = p->filterA[1] = 0;
+ p->filterB[0] = p->filterB[1] = 0;
+ p->lastA[0] = p->lastA[1] = 0;
+}
+
+/** Get inverse sign of integer (-1 for positive, 1 for negative and 0 for zero) */
+static inline int APESIGN(int32_t x) {
+ return (x < 0) - (x > 0);
+}
+
+static int predictor_update_filter(APEPredictor *p, const int decoded, const int filter, const int delayA, const int delayB, const int adaptA, const int adaptB)
+{
+ int32_t predictionA, predictionB;
+
+ p->buf[delayA] = p->lastA[filter];
+ p->buf[adaptA] = APESIGN(p->buf[delayA]);
+ p->buf[delayA - 1] = p->buf[delayA] - p->buf[delayA - 1];
+ p->buf[adaptA - 1] = APESIGN(p->buf[delayA - 1]);
+
+ predictionA = p->buf[delayA ] * p->coeffsA[filter][0] +
+ p->buf[delayA - 1] * p->coeffsA[filter][1] +
+ p->buf[delayA - 2] * p->coeffsA[filter][2] +
+ p->buf[delayA - 3] * p->coeffsA[filter][3];
+
+ /* Apply a scaled first-order filter compression */
+ p->buf[delayB] = p->filterA[filter ^ 1] - ((p->filterB[filter] * 31) >> 5);
+ p->buf[adaptB] = APESIGN(p->buf[delayB]);
+ p->buf[delayB - 1] = p->buf[delayB] - p->buf[delayB - 1];
+ p->buf[adaptB - 1] = APESIGN(p->buf[delayB - 1]);
+ p->filterB[filter] = p->filterA[filter ^ 1];
+
+ predictionB = p->buf[delayB ] * p->coeffsB[filter][0] +
+ p->buf[delayB - 1] * p->coeffsB[filter][1] +
+ p->buf[delayB - 2] * p->coeffsB[filter][2] +
+ p->buf[delayB - 3] * p->coeffsB[filter][3] +
+ p->buf[delayB - 4] * p->coeffsB[filter][4];
+
+ p->lastA[filter] = decoded + ((predictionA + (predictionB >> 1)) >> 10);
+ p->filterA[filter] = p->lastA[filter] + ((p->filterA[filter] * 31) >> 5);
+
+ if (!decoded) // no need updating filter coefficients
+ return p->filterA[filter];
+
+ if (decoded > 0) {
+ p->coeffsA[filter][0] -= p->buf[adaptA ];
+ p->coeffsA[filter][1] -= p->buf[adaptA - 1];
+ p->coeffsA[filter][2] -= p->buf[adaptA - 2];
+ p->coeffsA[filter][3] -= p->buf[adaptA - 3];
+
+ p->coeffsB[filter][0] -= p->buf[adaptB ];
+ p->coeffsB[filter][1] -= p->buf[adaptB - 1];
+ p->coeffsB[filter][2] -= p->buf[adaptB - 2];
+ p->coeffsB[filter][3] -= p->buf[adaptB - 3];
+ p->coeffsB[filter][4] -= p->buf[adaptB - 4];
+ } else {
+ p->coeffsA[filter][0] += p->buf[adaptA ];
+ p->coeffsA[filter][1] += p->buf[adaptA - 1];
+ p->coeffsA[filter][2] += p->buf[adaptA - 2];
+ p->coeffsA[filter][3] += p->buf[adaptA - 3];
+
+ p->coeffsB[filter][0] += p->buf[adaptB ];
+ p->coeffsB[filter][1] += p->buf[adaptB - 1];
+ p->coeffsB[filter][2] += p->buf[adaptB - 2];
+ p->coeffsB[filter][3] += p->buf[adaptB - 3];
+ p->coeffsB[filter][4] += p->buf[adaptB - 4];
+ }
+ return p->filterA[filter];
+}
+
+static void predictor_decode_stereo(APEContext * ctx, int count)
+{
+ int32_t predictionA, predictionB;
+ APEPredictor *p = &ctx->predictor;
+ int32_t *decoded0 = ctx->decoded0;
+ int32_t *decoded1 = ctx->decoded1;
+
+ while (count--) {
+ /* Predictor Y */
+ predictionA = predictor_update_filter(p, *decoded0, 0, YDELAYA, YDELAYB, YADAPTCOEFFSA, YADAPTCOEFFSB);
+ predictionB = predictor_update_filter(p, *decoded1, 1, XDELAYA, XDELAYB, XADAPTCOEFFSA, XADAPTCOEFFSB);
+ *(decoded0++) = predictionA;
+ *(decoded1++) = predictionB;
+
+ /* Combined */
+ p->buf++;
+
+ /* Have we filled the history buffer? */
+ if (p->buf == p->historybuffer + HISTORY_SIZE) {
+ memmove(p->historybuffer, p->buf, PREDICTOR_SIZE * sizeof(int32_t));
+ p->buf = p->historybuffer;
+ }
+ }
+}
+
+static void predictor_decode_mono(APEContext * ctx, int count)
+{
+ APEPredictor *p = &ctx->predictor;
+ int32_t *decoded0 = ctx->decoded0;
+ int32_t predictionA, currentA, A;
+
+ currentA = p->lastA[0];
+
+ while (count--) {
+ A = *decoded0;
+
+ p->buf[YDELAYA] = currentA;
+ p->buf[YDELAYA - 1] = p->buf[YDELAYA] - p->buf[YDELAYA - 1];
+
+ predictionA = p->buf[YDELAYA ] * p->coeffsA[0][0] +
+ p->buf[YDELAYA - 1] * p->coeffsA[0][1] +
+ p->buf[YDELAYA - 2] * p->coeffsA[0][2] +
+ p->buf[YDELAYA - 3] * p->coeffsA[0][3];
+
+ currentA = A + (predictionA >> 10);
+
+ p->buf[YADAPTCOEFFSA] = APESIGN(p->buf[YDELAYA ]);
+ p->buf[YADAPTCOEFFSA - 1] = APESIGN(p->buf[YDELAYA - 1]);
+
+ if (A > 0) {
+ p->coeffsA[0][0] -= p->buf[YADAPTCOEFFSA ];
+ p->coeffsA[0][1] -= p->buf[YADAPTCOEFFSA - 1];
+ p->coeffsA[0][2] -= p->buf[YADAPTCOEFFSA - 2];
+ p->coeffsA[0][3] -= p->buf[YADAPTCOEFFSA - 3];
+ } else if (A < 0) {
+ p->coeffsA[0][0] += p->buf[YADAPTCOEFFSA ];
+ p->coeffsA[0][1] += p->buf[YADAPTCOEFFSA - 1];
+ p->coeffsA[0][2] += p->buf[YADAPTCOEFFSA - 2];
+ p->coeffsA[0][3] += p->buf[YADAPTCOEFFSA - 3];
+ }
+
+ p->buf++;
+
+ /* Have we filled the history buffer? */
+ if (p->buf == p->historybuffer + HISTORY_SIZE) {
+ memmove(p->historybuffer, p->buf, PREDICTOR_SIZE * sizeof(int32_t));
+ p->buf = p->historybuffer;
+ }
+
+ p->filterA[0] = currentA + ((p->filterA[0] * 31) >> 5);
+ *(decoded0++) = p->filterA[0];
+ }
+
+ p->lastA[0] = currentA;
+}
+
+static void do_init_filter(APEFilter *f, int16_t * buf, int order)
+{
+ f->coeffs = buf;
+ f->historybuffer = buf + order;
+ f->delay = f->historybuffer + order * 2;
+ f->adaptcoeffs = f->historybuffer + order;
+
+ memset(f->historybuffer, 0, (order * 2) * sizeof(int16_t));
+ memset(f->coeffs, 0, order * sizeof(int16_t));
+ f->avg = 0;
+}
+
+static void init_filter(APEContext * ctx, APEFilter *f, int16_t * buf, int order)
+{
+ do_init_filter(&f[0], buf, order);
+ do_init_filter(&f[1], buf + order * 3 + HISTORY_SIZE, order);
+}
+
+static inline void do_apply_filter(int version, APEFilter *f, int32_t *data, int count, int order, int fracbits)
+{
+ int res;
+ int absres;
+
+ while (count--) {
+ /* round fixedpoint scalar product */
+ res = (scalarproduct(f->delay - order, f->coeffs, order) + (1 << (fracbits - 1))) >> fracbits;
+
+ if (*data < 0)
+ vector_add(f->coeffs, f->adaptcoeffs - order, order);
+ else if (*data > 0)
+ vector_sub(f->coeffs, f->adaptcoeffs - order, order);
+
+ res += *data;
+
+ *data++ = res;
+
+ /* Update the output history */
+ *f->delay++ = av_clip_int16(res);
+
+ if (version < 3980) {
+ /* Version ??? to < 3.98 files (untested) */
+ f->adaptcoeffs[0] = (res == 0) ? 0 : ((res >> 28) & 8) - 4;
+ f->adaptcoeffs[-4] >>= 1;
+ f->adaptcoeffs[-8] >>= 1;
+ } else {
+ /* Version 3.98 and later files */
+
+ /* Update the adaption coefficients */
+ absres = (res < 0 ? -res : res);
+
+ if (absres > (f->avg * 3))
+ *f->adaptcoeffs = ((res >> 25) & 64) - 32;
+ else if (absres > (f->avg * 4) / 3)
+ *f->adaptcoeffs = ((res >> 26) & 32) - 16;
+ else if (absres > 0)
+ *f->adaptcoeffs = ((res >> 27) & 16) - 8;
+ else
+ *f->adaptcoeffs = 0;
+
+ f->avg += (absres - f->avg) / 16;
+
+ f->adaptcoeffs[-1] >>= 1;
+ f->adaptcoeffs[-2] >>= 1;
+ f->adaptcoeffs[-8] >>= 1;
+ }
+
+ f->adaptcoeffs++;
+
+ /* Have we filled the history buffer? */
+ if (f->delay == f->historybuffer + HISTORY_SIZE + (order * 2)) {
+ memmove(f->historybuffer, f->delay - (order * 2),
+ (order * 2) * sizeof(int16_t));
+ f->delay = f->historybuffer + order * 2;
+ f->adaptcoeffs = f->historybuffer + order;
+ }
+ }
+}
+
+static void apply_filter(APEContext * ctx, APEFilter *f,
+ int32_t * data0, int32_t * data1,
+ int count, int order, int fracbits)
+{
+ do_apply_filter(ctx->fileversion, &f[0], data0, count, order, fracbits);
+ if (data1)
+ do_apply_filter(ctx->fileversion, &f[1], data1, count, order, fracbits);
+}
+
+static void ape_apply_filters(APEContext * ctx, int32_t * decoded0,
+ int32_t * decoded1, int count)
+{
+ int i;
+
+ for (i = 0; i < APE_FILTER_LEVELS; i++) {
+ if (!ape_filter_orders[ctx->fset][i])
+ break;
+ apply_filter(ctx, ctx->filters[i], decoded0, decoded1, count, ape_filter_orders[ctx->fset][i], ape_filter_fracbits[ctx->fset][i]);
+ }
+}
+
+static void init_frame_decoder(APEContext * ctx)
+{
+ int i;
+ init_entropy_decoder(ctx);
+ init_predictor_decoder(ctx);
+
+ for (i = 0; i < APE_FILTER_LEVELS; i++) {
+ if (!ape_filter_orders[ctx->fset][i])
+ break;
+ init_filter(ctx, ctx->filters[i], ctx->filterbuf[i], ape_filter_orders[ctx->fset][i]);
+ }
+}
+
+static void ape_unpack_mono(APEContext * ctx, int count)
+{
+ int32_t left;
+ int32_t *decoded0 = ctx->decoded0;
+ int32_t *decoded1 = ctx->decoded1;
+
+ if (ctx->frameflags & APE_FRAMECODE_STEREO_SILENCE) {
+ entropy_decode(ctx, count, 0);
+ /* We are pure silence, so we're done. */
+ av_log(ctx->avctx, AV_LOG_DEBUG, "pure silence mono\n");
+ return;
+ }
+
+ entropy_decode(ctx, count, 0);
+ ape_apply_filters(ctx, decoded0, NULL, count);
+
+ /* Now apply the predictor decoding */
+ predictor_decode_mono(ctx, count);
+
+ /* Pseudo-stereo - just copy left channel to right channel */
+ if (ctx->channels == 2) {
+ while (count--) {
+ left = *decoded0;
+ *(decoded1++) = *(decoded0++) = left;
+ }
+ }
+}
+
+static void ape_unpack_stereo(APEContext * ctx, int count)
+{
+ int32_t left, right;
+ int32_t *decoded0 = ctx->decoded0;
+ int32_t *decoded1 = ctx->decoded1;
+
+ if (ctx->frameflags & APE_FRAMECODE_STEREO_SILENCE) {
+ /* We are pure silence, so we're done. */
+ av_log(ctx->avctx, AV_LOG_DEBUG, "pure silence stereo\n");
+ return;
+ }
+
+ entropy_decode(ctx, count, 1);
+ ape_apply_filters(ctx, decoded0, decoded1, count);
+
+ /* Now apply the predictor decoding */
+ predictor_decode_stereo(ctx, count);
+
+ /* Decorrelate and scale to output depth */
+ while (count--) {
+ left = *decoded1 - (*decoded0 / 2);
+ right = left + *decoded0;
+
+ *(decoded0++) = left;
+ *(decoded1++) = right;
+ }
+}
+
+static int ape_decode_frame(AVCodecContext * avctx,
+ void *data, int *data_size,
+ uint8_t * buf, int buf_size)
+{
+ APEContext *s = avctx->priv_data;
+ int16_t *samples = data;
+ int nblocks;
+ int i, n;
+ int blockstodecode;
+ int bytes_used;
+
+ if (buf_size == 0 && !s->samples) {
+ *data_size = 0;
+ return 0;
+ }
+
+ /* should not happen but who knows */
+ if (BLOCKS_PER_LOOP * 2 * avctx->channels > *data_size) {
+ av_log (avctx, AV_LOG_ERROR, "Packet size is too big to be handled in lavc! (max is %d where you have %d)\n", *data_size, s->samples * 2 * avctx->channels);
+ return -1;
+ }
+
+ if(!s->samples){
+ s->data = av_realloc(s->data, (buf_size + 3) & ~3);
+ s->dsp.bswap_buf(s->data, buf, buf_size >> 2);
+ s->ptr = s->last_ptr = s->data;
+ s->data_end = s->data + buf_size;
+
+ nblocks = s->samples = bytestream_get_be32(&s->ptr);
+ n = bytestream_get_be32(&s->ptr);
+ if(n < 0 || n > 3){
+ av_log(avctx, AV_LOG_ERROR, "Incorrect offset passed\n");
+ s->data = NULL;
+ return -1;
+ }
+ s->ptr += n;
+
+ s->currentframeblocks = nblocks;
+ buf += 4;
+ if (s->samples <= 0) {
+ *data_size = 0;
+ return buf_size;
+ }
+
+ memset(s->decoded0, 0, sizeof(s->decoded0));
+ memset(s->decoded1, 0, sizeof(s->decoded1));
+
+ /* Initialize the frame decoder */
+ init_frame_decoder(s);
+ }
+
+ if (!s->data) {
+ *data_size = 0;
+ return buf_size;
+ }
+
+ nblocks = s->samples;
+ blockstodecode = FFMIN(BLOCKS_PER_LOOP, nblocks);
+
+ if ((s->channels == 1) || (s->frameflags & APE_FRAMECODE_PSEUDO_STEREO))
+ ape_unpack_mono(s, blockstodecode);
+ else
+ ape_unpack_stereo(s, blockstodecode);
+
+ for (i = 0; i < blockstodecode; i++) {
+ *samples++ = s->decoded0[i];
+ if(s->channels == 2)
+ *samples++ = s->decoded1[i];
+ }
+
+ s->samples -= blockstodecode;
+
+ *data_size = blockstodecode * 2 * s->channels;
+ bytes_used = s->samples ? s->ptr - s->last_ptr : buf_size;
+ s->last_ptr = s->ptr;
+ return bytes_used;
+}
+
+AVCodec ape_decoder = {
+ "ape",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_APE,
+ sizeof(APEContext),
+ ape_decode_init,
+ NULL,
+ ape_decode_close,
+ ape_decode_frame,
+};
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 53f24f25d0..5ec822a2cb 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -33,8 +33,8 @@
#define AV_STRINGIFY(s) AV_TOSTRING(s)
#define AV_TOSTRING(s) #s
-#define LIBAVCODEC_VERSION_INT ((51<<16)+(43<<8)+0)
-#define LIBAVCODEC_VERSION 51.43.0
+#define LIBAVCODEC_VERSION_INT ((51<<16)+(44<<8)+0)
+#define LIBAVCODEC_VERSION 51.44.0
#define LIBAVCODEC_BUILD LIBAVCODEC_VERSION_INT
#define LIBAVCODEC_IDENT "Lavc" AV_STRINGIFY(LIBAVCODEC_VERSION)
@@ -260,6 +260,7 @@ enum CodecID {
CODEC_ID_GSM_MS, /* as found in WAV */
CODEC_ID_ATRAC3,
CODEC_ID_VOXWARE,
+ CODEC_ID_APE,
/* subtitle codecs */
CODEC_ID_DVD_SUBTITLE= 0x17000,
diff --git a/libavformat/Makefile b/libavformat/Makefile
index c7d4fadd88..1023199413 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -20,6 +20,7 @@ OBJS-$(CONFIG_AIFF_MUXER) += aiff.o riff.o
OBJS-$(CONFIG_AMR_DEMUXER) += amr.o
OBJS-$(CONFIG_AMR_MUXER) += amr.o
OBJS-$(CONFIG_APC_DEMUXER) += apc.o
+OBJS-$(CONFIG_APE_DEMUXER) += ape.o
OBJS-$(CONFIG_ASF_DEMUXER) += asf.o riff.o
OBJS-$(CONFIG_ASF_MUXER) += asf-enc.o riff.o
OBJS-$(CONFIG_ASF_STREAM_MUXER) += asf-enc.o riff.o
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index 0a783ba6c4..1b7af2aca2 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -53,6 +53,7 @@ void av_register_all(void)
REGISTER_MUXDEMUX (AIFF, aiff);
REGISTER_MUXDEMUX (AMR, amr);
REGISTER_DEMUXER (APC, apc);
+ REGISTER_DEMUXER (APE, ape);
REGISTER_MUXDEMUX (ASF, asf);
REGISTER_MUXER (ASF_STREAM, asf_stream);
REGISTER_MUXDEMUX (AU, au);
diff --git a/libavformat/allformats.h b/libavformat/allformats.h
index eb51c9620c..ff66203816 100644
--- a/libavformat/allformats.h
+++ b/libavformat/allformats.h
@@ -29,6 +29,7 @@ extern AVInputFormat ac3_demuxer;
extern AVInputFormat aiff_demuxer;
extern AVInputFormat amr_demuxer;
extern AVInputFormat apc_demuxer;
+extern AVInputFormat ape_demuxer;
extern AVInputFormat asf_demuxer;
extern AVInputFormat au_demuxer;
extern AVInputFormat audio_beos_demuxer;
diff --git a/libavformat/ape.c b/libavformat/ape.c
new file mode 100644
index 0000000000..336871484e
--- /dev/null
+++ b/libavformat/ape.c
@@ -0,0 +1,392 @@
+/*
+ * Monkey's Audio APE demuxer
+ * Copyright (c) 2007 Benjamin Zores <ben@geexbox.org>
+ * based upon libdemac from Dave Chapman.
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stdio.h>
+
+#include "avformat.h"
+
+/* The earliest and latest file formats supported by this library */
+#define APE_MIN_VERSION 3970
+#define APE_MAX_VERSION 3990
+
+#define MAC_FORMAT_FLAG_8_BIT 1 // is 8-bit [OBSOLETE]
+#define MAC_FORMAT_FLAG_CRC 2 // uses the new CRC32 error detection [OBSOLETE]
+#define MAC_FORMAT_FLAG_HAS_PEAK_LEVEL 4 // uint32 nPeakLevel after the header [OBSOLETE]
+#define MAC_FORMAT_FLAG_24_BIT 8 // is 24-bit [OBSOLETE]
+#define MAC_FORMAT_FLAG_HAS_SEEK_ELEMENTS 16 // has the number of seek elements after the peak level
+#define MAC_FORMAT_FLAG_CREATE_WAV_HEADER 32 // create the wave header on decompression (not stored)
+
+#define MAC_SUBFRAME_SIZE 4608
+
+#define APE_EXTRADATA_SIZE 6
+
+typedef struct {
+ int64_t pos;
+ int nblocks;
+ int size;
+ int skip;
+ int64_t pts;
+} APEFrame;
+
+typedef struct {
+ /* Derived fields */
+ uint32_t junklength;
+ uint32_t firstframe;
+ uint32_t totalsamples;
+ int currentframe;
+ APEFrame *frames;
+
+ /* Info from Descriptor Block */
+ char magic[4];
+ int16_t fileversion;
+ int16_t padding1;
+ uint32_t descriptorlength;
+ uint32_t headerlength;
+ uint32_t seektablelength;
+ uint32_t wavheaderlength;
+ uint32_t audiodatalength;
+ uint32_t audiodatalength_high;
+ uint32_t wavtaillength;
+ uint8_t md5[16];
+
+ /* Info from Header Block */
+ uint16_t compressiontype;
+ uint16_t formatflags;
+ uint32_t blocksperframe;
+ uint32_t finalframeblocks;
+ uint32_t totalframes;
+ uint16_t bps;
+ uint16_t channels;
+ uint32_t samplerate;
+
+ /* Seektable */
+ uint32_t *seektable;
+} APEContext;
+
+static int ape_probe(AVProbeData * p)
+{
+ if (p->buf[0] == 'M' && p->buf[1] == 'A' && p->buf[2] == 'C' && p->buf[3] == ' ')
+ return AVPROBE_SCORE_MAX;
+
+ return 0;
+}
+
+static void ape_dumpinfo(APEContext * ape_ctx)
+{
+ int i;
+
+ av_log(NULL, AV_LOG_DEBUG, "Descriptor Block:\n\n");
+ av_log(NULL, AV_LOG_DEBUG, "magic = \"%c%c%c%c\"\n", ape_ctx->magic[0], ape_ctx->magic[1], ape_ctx->magic[2], ape_ctx->magic[3]);
+ av_log(NULL, AV_LOG_DEBUG, "fileversion = %d\n", ape_ctx->fileversion);
+ av_log(NULL, AV_LOG_DEBUG, "descriptorlength = %d\n", ape_ctx->descriptorlength);
+ av_log(NULL, AV_LOG_DEBUG, "headerlength = %d\n", ape_ctx->headerlength);
+ av_log(NULL, AV_LOG_DEBUG, "seektablelength = %d\n", ape_ctx->seektablelength);
+ av_log(NULL, AV_LOG_DEBUG, "wavheaderlength = %d\n", ape_ctx->wavheaderlength);
+ av_log(NULL, AV_LOG_DEBUG, "audiodatalength = %d\n", ape_ctx->audiodatalength);
+ av_log(NULL, AV_LOG_DEBUG, "audiodatalength_high = %d\n", ape_ctx->audiodatalength_high);
+ av_log(NULL, AV_LOG_DEBUG, "wavtaillength = %d\n", ape_ctx->wavtaillength);
+ av_log(NULL, AV_LOG_DEBUG, "md5 = ");
+ for (i = 0; i < 16; i++)
+ av_log(NULL, AV_LOG_DEBUG, "%02x", ape_ctx->md5[i]);
+ av_log(NULL, AV_LOG_DEBUG, "\n");
+
+ av_log(NULL, AV_LOG_DEBUG, "\nHeader Block:\n\n");
+
+ av_log(NULL, AV_LOG_DEBUG, "compressiontype = %d\n", ape_ctx->compressiontype);
+ av_log(NULL, AV_LOG_DEBUG, "formatflags = %d\n", ape_ctx->formatflags);
+ av_log(NULL, AV_LOG_DEBUG, "blocksperframe = %d\n", ape_ctx->blocksperframe);
+ av_log(NULL, AV_LOG_DEBUG, "finalframeblocks = %d\n", ape_ctx->finalframeblocks);
+ av_log(NULL, AV_LOG_DEBUG, "totalframes = %d\n", ape_ctx->totalframes);
+ av_log(NULL, AV_LOG_DEBUG, "bps = %d\n", ape_ctx->bps);
+ av_log(NULL, AV_LOG_DEBUG, "channels = %d\n", ape_ctx->channels);
+ av_log(NULL, AV_LOG_DEBUG, "samplerate = %d\n", ape_ctx->samplerate);
+
+ av_log(NULL, AV_LOG_DEBUG, "\nSeektable\n\n");
+ if ((ape_ctx->seektablelength / sizeof(uint32_t)) != ape_ctx->totalframes) {
+ av_log(NULL, AV_LOG_DEBUG, "No seektable\n");
+ } else {
+ for (i = 0; i < ape_ctx->seektablelength / sizeof(uint32_t); i++) {
+ if (i < ape_ctx->totalframes - 1) {
+ av_log(NULL, AV_LOG_DEBUG, "%8d %d (%d bytes)\n", i, ape_ctx->seektable[i], ape_ctx->seektable[i + 1] - ape_ctx->seektable[i]);
+ } else {
+ av_log(NULL, AV_LOG_DEBUG, "%8d %d\n", i, ape_ctx->seektable[i]);
+ }
+ }
+ }
+
+ av_log(NULL, AV_LOG_DEBUG, "\nFrames\n\n");
+ for (i = 0; i < ape_ctx->totalframes; i++)
+ av_log(NULL, AV_LOG_DEBUG, "%8d %8lld %8d (%d samples)\n", i, ape_ctx->frames[i].pos, ape_ctx->frames[i].size, ape_ctx->frames[i].nblocks);
+
+ av_log(NULL, AV_LOG_DEBUG, "\nCalculated information:\n\n");
+ av_log(NULL, AV_LOG_DEBUG, "junklength = %d\n", ape_ctx->junklength);
+ av_log(NULL, AV_LOG_DEBUG, "firstframe = %d\n", ape_ctx->firstframe);
+ av_log(NULL, AV_LOG_DEBUG, "totalsamples = %d\n", ape_ctx->totalsamples);
+}
+
+static int ape_read_header(AVFormatContext * s, AVFormatParameters * ap)
+{
+ ByteIOContext *pb = &s->pb;
+ APEContext *ape = s->priv_data;
+ AVStream *st;
+ uint32_t tag;
+ int i;
+ int total_blocks;
+ int64_t pts;
+
+ /* TODO: Skip any leading junk such as id3v2 tags */
+ ape->junklength = 0;
+
+ tag = get_le32(pb);
+ if (tag != MKTAG('M', 'A', 'C', ' '))
+ return -1;
+
+ ape->fileversion = get_le16(pb);
+
+ if (ape->fileversion < APE_MIN_VERSION || ape->fileversion > APE_MAX_VERSION) {
+ av_log(s, AV_LOG_ERROR, "Unsupported file version - %d.%02d\n", ape->fileversion / 1000, (ape->fileversion % 1000) / 10);
+ return -1;
+ }
+
+ if (ape->fileversion >= 3980) {
+ ape->padding1 = get_le16(pb);
+ ape->descriptorlength = get_le32(pb);
+ ape->headerlength = get_le32(pb);
+ ape->seektablelength = get_le32(pb);
+ ape->wavheaderlength = get_le32(pb);
+ ape->audiodatalength = get_le32(pb);
+ ape->audiodatalength_high = get_le32(pb);
+ ape->wavtaillength = get_le32(pb);
+ get_buffer(pb, ape->md5, 16);
+
+ /* Skip any unknown bytes at the end of the descriptor.
+ This is for future compatibility */
+ if (ape->descriptorlength > 52)
+ url_fseek(pb, ape->descriptorlength - 52, SEEK_CUR);
+
+ /* Read header data */
+ ape->compressiontype = get_le16(pb);
+ ape->formatflags = get_le16(pb);
+ ape->blocksperframe = get_le32(pb);
+ ape->finalframeblocks = get_le32(pb);
+ ape->totalframes = get_le32(pb);
+ ape->bps = get_le16(pb);
+ ape->channels = get_le16(pb);
+ ape->samplerate = get_le32(pb);
+ } else {
+ ape->descriptorlength = 0;
+ ape->headerlength = 32;
+
+ ape->compressiontype = get_le16(pb);
+ ape->formatflags = get_le16(pb);
+ ape->channels = get_le16(pb);
+ ape->samplerate = get_le32(pb);
+ ape->wavheaderlength = get_le32(pb);
+ ape->wavtaillength = get_le32(pb);
+ ape->totalframes = get_le32(pb);
+ ape->finalframeblocks = get_le32(pb);
+
+ if (ape->formatflags & MAC_FORMAT_FLAG_HAS_PEAK_LEVEL) {
+ url_fseek(pb, 4, SEEK_CUR); /* Skip the peak level */
+ ape->headerlength += 4;
+ }
+
+ if (ape->formatflags & MAC_FORMAT_FLAG_HAS_SEEK_ELEMENTS) {
+ ape->seektablelength = get_le32(pb);
+ ape->headerlength += 4;
+ ape->seektablelength *= sizeof(int32_t);
+ } else
+ ape->seektablelength = ape->totalframes * sizeof(int32_t);
+
+ if (ape->formatflags & MAC_FORMAT_FLAG_8_BIT)
+ ape->bps = 8;
+ else if (ape->formatflags & MAC_FORMAT_FLAG_24_BIT)
+ ape->bps = 24;
+ else
+ ape->bps = 16;
+
+ if (ape->fileversion >= 3950)
+ ape->blocksperframe = 73728 * 4;
+ else if (ape->fileversion >= 3900 || (ape->fileversion >= 3800 && ape->compressiontype >= 4000))
+ ape->blocksperframe = 73728;
+ else
+ ape->blocksperframe = 9216;
+
+ /* Skip any stored wav header */
+ if (!(ape->formatflags & MAC_FORMAT_FLAG_CREATE_WAV_HEADER))
+ url_fskip(pb, ape->wavheaderlength);
+ }
+
+ if(ape->totalframes > UINT_MAX / sizeof(APEFrame)){
+ av_log(s, AV_LOG_ERROR, "Too many frames: %d\n", ape->totalframes);
+ return -1;
+ }
+ ape->frames = av_malloc(ape->totalframes * sizeof(APEFrame));
+ if(!ape->frames)
+ return AVERROR_NOMEM;
+ ape->firstframe = ape->junklength + ape->descriptorlength + ape->headerlength + ape->seektablelength + ape->wavheaderlength;
+ ape->currentframe = 0;
+
+
+ ape->totalsamples = ape->finalframeblocks;
+ if (ape->totalframes > 1)
+ ape->totalsamples += ape->blocksperframe * (ape->totalframes - 1);
+
+ if (ape->seektablelength > 0) {
+ ape->seektable = av_malloc(ape->seektablelength);
+ for (i = 0; i < ape->seektablelength / sizeof(uint32_t); i++)
+ ape->seektable[i] = get_le32(pb);
+ }
+
+ ape->frames[0].pos = ape->firstframe;
+ ape->frames[0].nblocks = ape->blocksperframe;
+ ape->frames[0].skip = 0;
+ for (i = 1; i < ape->totalframes; i++) {
+ ape->frames[i].pos = ape->seektable[i]; //ape->frames[i-1].pos + ape->blocksperframe;
+ ape->frames[i].nblocks = ape->blocksperframe;
+ ape->frames[i - 1].size = ape->frames[i].pos - ape->frames[i - 1].pos;
+ ape->frames[i].skip = (ape->frames[i].pos - ape->frames[0].pos) & 3;
+ }
+ ape->frames[ape->totalframes - 1].size = ape->finalframeblocks * 4;
+ ape->frames[ape->totalframes - 1].nblocks = ape->finalframeblocks;
+
+ for (i = 0; i < ape->totalframes; i++) {
+ if(ape->frames[i].skip){
+ ape->frames[i].pos -= ape->frames[i].skip;
+ ape->frames[i].size += ape->frames[i].skip;
+ }
+ ape->frames[i].size = (ape->frames[i].size + 3) & ~3;
+ }
+
+
+ ape_dumpinfo(ape);
+
+ av_log(s, AV_LOG_DEBUG, "Decoding file - v%d.%02d, compression level %d\n", ape->fileversion / 1000, (ape->fileversion % 1000) / 10, ape->compressiontype);
+
+ /* now we are ready: build format streams */
+ st = av_new_stream(s, 0);
+ if (!st)
+ return -1;
+
+ total_blocks = (ape->totalframes == 0) ? 0 : ((ape->totalframes - 1) * ape->blocksperframe) + ape->finalframeblocks;
+
+ st->codec->codec_type = CODEC_TYPE_AUDIO;
+ st->codec->codec_id = CODEC_ID_APE;
+ st->codec->codec_tag = MKTAG('A', 'P', 'E', ' ');
+ st->codec->channels = ape->channels;
+ st->codec->sample_rate = ape->samplerate;
+ st->codec->bits_per_sample = ape->bps;
+ st->codec->frame_size = MAC_SUBFRAME_SIZE;
+
+ st->nb_frames = ape->totalframes;
+ s->start_time = 0;
+ s->duration = (int64_t) total_blocks * AV_TIME_BASE / ape->samplerate;
+ av_set_pts_info(st, 64, MAC_SUBFRAME_SIZE, ape->samplerate);
+
+ st->codec->extradata = av_malloc(APE_EXTRADATA_SIZE);
+ st->codec->extradata_size = APE_EXTRADATA_SIZE;
+ AV_WL16(st->codec->extradata + 0, ape->fileversion);
+ AV_WL16(st->codec->extradata + 2, ape->compressiontype);
+ AV_WL16(st->codec->extradata + 4, ape->formatflags);
+
+ pts = 0;
+ for (i = 0; i < ape->totalframes; i++) {
+ ape->frames[i].pts = pts;
+ av_add_index_entry(st, ape->frames[i].pos, ape->frames[i].pts, 0, 0, AVINDEX_KEYFRAME);
+ pts += ape->blocksperframe / MAC_SUBFRAME_SIZE;
+ }
+
+ return 0;
+}
+
+static int ape_read_packet(AVFormatContext * s, AVPacket * pkt)
+{
+ int ret;
+ int nblocks;
+ APEContext *ape = s->priv_data;
+ uint32_t extra_size = 8;
+
+ if (url_feof(&s->pb))
+ return AVERROR_IO;
+ if (ape->currentframe > ape->totalframes)
+ return AVERROR_IO;
+
+ url_fseek (&s->pb, ape->frames[ape->currentframe].pos, SEEK_SET);
+
+ /* Calculate how many blocks there are in this frame */
+ if (ape->currentframe == (ape->totalframes - 1))
+ nblocks = ape->finalframeblocks;
+ else
+ nblocks = ape->blocksperframe;
+
+ if (av_new_packet(pkt, ape->frames[ape->currentframe].size + extra_size) < 0)
+ return AVERROR_NOMEM;
+
+ AV_WL32(pkt->data , nblocks);
+ AV_WL32(pkt->data + 4, ape->frames[ape->currentframe].skip);
+ ret = get_buffer(&s->pb, pkt->data + extra_size, ape->frames[ape->currentframe].size);
+
+ pkt->pts = ape->frames[ape->currentframe].pts;
+ pkt->stream_index = 0;
+
+ /* note: we need to modify the packet size here to handle the last
+ packet */
+ pkt->size = ret + extra_size;
+
+ ape->currentframe++;
+
+ return 0;
+}
+
+static int ape_read_close(AVFormatContext * s)
+{
+ APEContext *ape = s->priv_data;
+
+ av_freep(&ape->frames);
+ av_freep(&ape->seektable);
+ return 0;
+}
+
+static int ape_read_seek(AVFormatContext *s, int stream_index, int64_t timestamp, int flags)
+{
+ AVStream *st = s->streams[stream_index];
+ APEContext *ape = s->priv_data;
+ int index = av_index_search_timestamp(st, timestamp, flags);
+
+ if (index < 0)
+ return -1;
+
+ ape->currentframe = index;
+ return 0;
+}
+
+AVInputFormat ape_demuxer = {
+ "ape",
+ "Monkey's Audio",
+ sizeof(APEContext),
+ ape_probe,
+ ape_read_header,
+ ape_read_packet,
+ ape_read_close,
+ ape_read_seek,
+ .extensions = "ape,apl,mac"
+};
diff --git a/libavformat/avformat.h b/libavformat/avformat.h
index 577c215a44..a58200c312 100644
--- a/libavformat/avformat.h
+++ b/libavformat/avformat.h
@@ -21,8 +21,8 @@
#ifndef AVFORMAT_H
#define AVFORMAT_H
-#define LIBAVFORMAT_VERSION_INT ((51<<16)+(12<<8)+3)
-#define LIBAVFORMAT_VERSION 51.12.3
+#define LIBAVFORMAT_VERSION_INT ((51<<16)+(13<<8)+3)
+#define LIBAVFORMAT_VERSION 51.13.3
#define LIBAVFORMAT_BUILD LIBAVFORMAT_VERSION_INT
#define LIBAVFORMAT_IDENT "Lavf" AV_STRINGIFY(LIBAVFORMAT_VERSION)