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Monkey Audio decoder

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Originally committed as revision 10484 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Kostya Shishkov 2007-09-13 03:22:47 +00:00
parent 48fe9238a0
commit bf4a1f17ee
12 changed files with 1328 additions and 4 deletions
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1
Changelog
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@ -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:

2
doc/general.texi
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@ -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.

1
libavcodec/Makefile
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@ -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

1
libavcodec/allcodecs.c
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@ -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);

1
libavcodec/allcodecs.h
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@ -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;

922
libavcodec/apedec.c Normal file
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@ -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,
};

5
libavcodec/avcodec.h
View File

@ -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,

1
libavformat/Makefile
View File

@ -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

1
libavformat/allformats.c
View File

@ -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);

1
libavformat/allformats.h
View File

@ -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;

392
libavformat/ape.c Normal file
View File

@ -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"
};

4
libavformat/avformat.h
View File

@ -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)