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FFmpeg/libavcodec/apedec.c
Michael Niedermayer 6faf0a21e1 Merge remote-tracking branch 'qatar/master' 
* qatar/master: (53 commits)
  probe: Restore identification of files with very large id3 tags and no extension.
  probe: Remove id3 tag presence as a criteria to do file extension checking.
  mpegts: MP4 SL support
  mpegts: MP4 OD support
  mpegts: Add support for Sections in PMT
  mpegts: Replace the MP4 descriptor parser with a recursive parser.
  mpegts: Add support for multiple mp4 descriptors
  mpegts: Parse mpeg2 SL descriptors.
  isom: Add MPEG4SYSTEMS dummy object type indication.
  aacdec: allow output reconfiguration on channel changes
  nellymoserenc: take float input samples instead of int16
  nellymoserdec: use dsp functions for overlap and windowing
  nellymoserdec: do not fail if there is extra data in the packet
  nellymoserdec: fail if output buffer is too small
  nellymoserdec: remove pointless buffer size check.
  lavf: add init_put_byte() to the list of visible symbols.
  seek-test: free options dictionary after use
  snow: do not draw_edge if emu_edge is set
  tools/pktdumper: update to recent avformat api
  seek-test: update to recent avformat api
  ...

Conflicts:
	doc/APIchanges
	libavcodec/mpegaudiodec.c
	libavcodec/nellymoserdec.c
	libavcodec/snow.c
	libavcodec/version.h
	libavcodec/wmadec.c
	libavformat/avformat.h
	libavformat/mpegts.c
	libavformat/mxfdec.c
	libavformat/utils.c
	libavformat/wtv.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-10-29 02:08:54 +02:00

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/*
* 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 "get_bits.h"
#include "bytestream.h"
#include "libavutil/audioconvert.h"
#include "libavutil/avassert.h"
/**
* @file
* 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 uint8_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
const uint8_t *ptr; ///< current position in frame data
const uint8_t *last_ptr; ///< position where last 4608-sample block ended
int error;
} APEContext;
// TODO: dsputilize
static av_cold 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]);
av_freep(&s->data);
return 0;
}
static av_cold 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 AVERROR(EINVAL);
}
if (avctx->bits_per_coded_sample != 16) {
av_log(avctx, AV_LOG_ERROR, "Only 16-bit samples are supported\n");
return AVERROR(EINVAL);
}
if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "Only mono and stereo is supported\n");
return AVERROR(EINVAL);
}
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 AVERROR_INVALIDDATA;
}
s->fset = s->compression_level / 1000 - 1;
for (i = 0; i < APE_FILTER_LEVELS; i++) {
if (!ape_filter_orders[s->fset][i])
break;
FF_ALLOC_OR_GOTO(avctx, s->filterbuf[i],
(ape_filter_orders[s->fset][i] * 3 + HISTORY_SIZE) * 4,
filter_alloc_fail);
}
dsputil_init(&s->dsp, avctx);
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
return 0;
filter_alloc_fail:
ape_decode_close(avctx);
return AVERROR(ENOMEM);
}
/**
* @name APE range decoding functions
* @{
*/
#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 <<= 8;
if(ctx->ptr < ctx->data_end) {
ctx->rc.buffer += *ctx->ptr;
ctx->ptr++;
} else {
ctx->error = 1;
}
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 ctx decoder context
* @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 ctx decoder context
* @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 ctx decoder context
* @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 uint16_t counts_3970[22] = {
0, 14824, 28224, 39348, 47855, 53994, 58171, 60926,
62682, 63786, 64463, 64878, 65126, 65276, 65365, 65419,
65450, 65469, 65480, 65487, 65491, 65493,
};
/**
* Probability ranges for symbols in Monkey Audio version 3.97
*/
static const uint16_t counts_diff_3970[21] = {
14824, 13400, 11124, 8507, 6139, 4177, 2755, 1756,
1104, 677, 415, 248, 150, 89, 54, 31,
19, 11, 7, 4, 2,
};
/**
* Fixed probabilities for symbols in Monkey Audio version 3.98
*/
static const uint16_t counts_3980[22] = {
0, 19578, 36160, 48417, 56323, 60899, 63265, 64435,
64971, 65232, 65351, 65416, 65447, 65466, 65476, 65482,
65485, 65488, 65490, 65491, 65492, 65493,
};
/**
* Probability ranges for symbols in Monkey Audio version 3.98
*/
static const uint16_t counts_diff_3980[21] = {
19578, 16582, 12257, 7906, 4576, 2366, 1170, 536,
261, 119, 65, 31, 19, 10, 6, 3,
3, 2, 1, 1, 1,
};
/**
* Decode symbol
* @param ctx decoder context
* @param counts probability range start position
* @param counts_diff probability range widths
*/
static inline int range_get_symbol(APEContext *ctx,
const uint16_t counts[],
const uint16_t counts_diff[])
{
int symbol, cf;
cf = range_decode_culshift(ctx, 16);
if(cf > 65492){
symbol= cf - 65535 + 63;
range_decode_update(ctx, 1, cf);
if(cf > 65535)
ctx->error=1;
return symbol;
}
/* 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)
{
int lim = rice->k ? (1 << (rice->k + 4)) : 0;
rice->ksum += ((x + 1) / 2) - ((rice->ksum + 16) >> 5);
if (rice->ksum < lim)
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 < 3990) {
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);
}
if (pivot < 0x10000) {
base = range_decode_culfreq(ctx, pivot);
range_decode_update(ctx, 1, base);
} else {
int base_hi = pivot, base_lo;
int bbits = 0;
while (base_hi & ~0xFFFF) {
base_hi >>= 1;
bbits++;
}
base_hi = range_decode_culfreq(ctx, base_hi + 1);
range_decode_update(ctx, 1, base_hi);
base_lo = range_decode_culfreq(ctx, 1 << bbits);
range_decode_update(ctx, 1, base_lo);
base = (base_hi << bbits) + base_lo;
}
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 int init_entropy_decoder(APEContext *ctx)
{
/* Read the CRC */
if (ctx->data_end - ctx->ptr < 6)
return AVERROR_INVALIDDATA;
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;
if (ctx->data_end - ctx->ptr < 6)
return AVERROR_INVALIDDATA;
ctx->frameflags = bytestream_get_be32(&ctx->ptr);
}
/* Keep a count of the blocks decoded in this frame */
ctx->blocksdecoded = 0;
/* Initialize 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);
return 0;
}
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;
/* Initialize and zero the coefficients */
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 av_always_inline 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, sign;
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);
sign = APESIGN(decoded);
p->coeffsA[filter][0] += p->buf[adaptA ] * sign;
p->coeffsA[filter][1] += p->buf[adaptA - 1] * sign;
p->coeffsA[filter][2] += p->buf[adaptA - 2] * sign;
p->coeffsA[filter][3] += p->buf[adaptA - 3] * sign;
p->coeffsB[filter][0] += p->buf[adaptB ] * sign;
p->coeffsB[filter][1] += p->buf[adaptB - 1] * sign;
p->coeffsB[filter][2] += p->buf[adaptB - 2] * sign;
p->coeffsB[filter][3] += p->buf[adaptB - 3] * sign;
p->coeffsB[filter][4] += p->buf[adaptB - 4] * sign;
return p->filterA[filter];
}
static void predictor_decode_stereo(APEContext *ctx, int count)
{
APEPredictor *p = &ctx->predictor;
int32_t *decoded0 = ctx->decoded0;
int32_t *decoded1 = ctx->decoded1;
while (count--) {
/* Predictor Y */
*decoded0 = predictor_update_filter(p, *decoded0, 0, YDELAYA, YDELAYB,
YADAPTCOEFFSA, YADAPTCOEFFSB);
decoded0++;
*decoded1 = predictor_update_filter(p, *decoded1, 1, XDELAYA, XDELAYB,
XADAPTCOEFFSA, XADAPTCOEFFSB);
decoded1++;
/* 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, sign;
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]);
sign = APESIGN(A);
p->coeffsA[0][0] += p->buf[YADAPTCOEFFSA ] * sign;
p->coeffsA[0][1] += p->buf[YADAPTCOEFFSA - 1] * sign;
p->coeffsA[0][2] += p->buf[YADAPTCOEFFSA - 2] * sign;
p->coeffsA[0][3] += p->buf[YADAPTCOEFFSA - 3] * sign;
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 void do_apply_filter(APEContext *ctx, int version, APEFilter *f,
int32_t *data, int count, int order, int fracbits)
{
int res;
int absres;
while (count--) {
/* round fixedpoint scalar product */
res = ctx->dsp.scalarproduct_and_madd_int16(f->coeffs, f->delay - order,
f->adaptcoeffs - order,
order, APESIGN(*data));
res = (res + (1 << (fracbits - 1))) >> fracbits;
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 = FFABS(res);
if (absres)
*f->adaptcoeffs = ((res & (1<<31)) - (1<<30)) >>
(25 + (absres <= f->avg*3) + (absres <= f->avg*4/3));
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, ctx->fileversion, &f[0], data0, count, order, fracbits);
if (data1)
do_apply_filter(ctx, 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 int init_frame_decoder(APEContext *ctx)
{
int i, ret;
if ((ret = init_entropy_decoder(ctx)) < 0)
return ret;
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]);
}
return 0;
}
static void ape_unpack_mono(APEContext *ctx, int count)
{
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) {
memcpy(decoded1, decoded0, count * sizeof(*decoded1));
}
}
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,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
APEContext *s = avctx->priv_data;
int16_t *samples = data;
uint32_t nblocks;
int i;
int blockstodecode;
int bytes_used;
/* should not happen but who knows */
if (BLOCKS_PER_LOOP * 2 * avctx->channels > *data_size) {
av_log (avctx, AV_LOG_ERROR, "Output buffer is too small.\n");
return AVERROR(EINVAL);
}
/* this should never be negative, but bad things will happen if it is, so
check it just to make sure. */
av_assert0(s->samples >= 0);
if(!s->samples){
uint32_t offset;
void *tmp_data;
if (buf_size < 8) {
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
return AVERROR_INVALIDDATA;
}
tmp_data = av_realloc(s->data, FFALIGN(buf_size, 4));
if (!tmp_data)
return AVERROR(ENOMEM);
s->data = tmp_data;
s->dsp.bswap_buf((uint32_t*)s->data, (const uint32_t*)buf, buf_size >> 2);
s->ptr = s->last_ptr = s->data;
s->data_end = s->data + buf_size;
nblocks = bytestream_get_be32(&s->ptr);
offset = bytestream_get_be32(&s->ptr);
if (offset > 3) {
av_log(avctx, AV_LOG_ERROR, "Incorrect offset passed\n");
s->data = NULL;
return AVERROR_INVALIDDATA;
}
if (s->data_end - s->ptr < offset) {
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
return AVERROR_INVALIDDATA;
}
s->ptr += offset;
if (!nblocks || nblocks > INT_MAX) {
av_log(avctx, AV_LOG_ERROR, "Invalid sample count: %u.\n", nblocks);
return AVERROR_INVALIDDATA;
}
s->currentframeblocks = s->samples = nblocks;
memset(s->decoded0, 0, sizeof(s->decoded0));
memset(s->decoded1, 0, sizeof(s->decoded1));
/* Initialize the frame decoder */
if (init_frame_decoder(s) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error reading frame header\n");
return AVERROR_INVALIDDATA;
}
}
if (!s->data) {
*data_size = 0;
return buf_size;
}
nblocks = s->samples;
blockstodecode = FFMIN(BLOCKS_PER_LOOP, nblocks);
s->error=0;
if ((s->channels == 1) || (s->frameflags & APE_FRAMECODE_PSEUDO_STEREO))
ape_unpack_mono(s, blockstodecode);
else
ape_unpack_stereo(s, blockstodecode);
emms_c();
if (s->error) {
s->samples=0;
av_log(avctx, AV_LOG_ERROR, "Error decoding frame\n");
return AVERROR_INVALIDDATA;
}
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;
}
static void ape_flush(AVCodecContext *avctx)
{
APEContext *s = avctx->priv_data;
s->samples= 0;
}
AVCodec ff_ape_decoder = {
.name = "ape",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_APE,
.priv_data_size = sizeof(APEContext),
.init = ape_decode_init,
.close = ape_decode_close,
.decode = ape_decode_frame,
.capabilities = CODEC_CAP_SUBFRAMES,
.flush = ape_flush,
.long_name = NULL_IF_CONFIG_SMALL("Monkey's Audio"),
};
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