41 #define MAX_CHANNELS 2
42 #define MAX_BYTESPERSAMPLE 3
44 #define APE_FRAMECODE_MONO_SILENCE 1
45 #define APE_FRAMECODE_STEREO_SILENCE 3
46 #define APE_FRAMECODE_PSEUDO_STEREO 4
48 #define HISTORY_SIZE 512
49 #define PREDICTOR_ORDER 8
51 #define PREDICTOR_SIZE 50
53 #define YDELAYA (18 + PREDICTOR_ORDER*4)
54 #define YDELAYB (18 + PREDICTOR_ORDER*3)
55 #define XDELAYA (18 + PREDICTOR_ORDER*2)
56 #define XDELAYB (18 + PREDICTOR_ORDER)
58 #define YADAPTCOEFFSA 18
59 #define XADAPTCOEFFSA 14
60 #define YADAPTCOEFFSB 10
61 #define XADAPTCOEFFSB 5
76 #define APE_FILTER_LEVELS 3
223 *v1++ += mul * *v3++;
254 "%d bits per coded sample", s->
bps);
332 #define TOP_VALUE ((unsigned int)1 << (CODE_BITS-1))
333 #define SHIFT_BITS (CODE_BITS - 9)
334 #define EXTRA_BITS ((CODE_BITS-2) % 8 + 1)
335 #define BOTTOM_VALUE (TOP_VALUE >> 8)
408 #define MODEL_ELEMENTS 64
414 0, 14824, 28224, 39348, 47855, 53994, 58171, 60926,
415 62682, 63786, 64463, 64878, 65126, 65276, 65365, 65419,
416 65450, 65469, 65480, 65487, 65491, 65493,
423 14824, 13400, 11124, 8507, 6139, 4177, 2755, 1756,
424 1104, 677, 415, 248, 150, 89, 54, 31,
432 0, 19578, 36160, 48417, 56323, 60899, 63265, 64435,
433 64971, 65232, 65351, 65416, 65447, 65466, 65476, 65482,
434 65485, 65488, 65490, 65491, 65492, 65493,
441 19578, 16582, 12257, 7906, 4576, 2366, 1170, 536,
442 261, 119, 65, 31, 19, 10, 6, 3,
453 const uint16_t counts[],
454 const uint16_t counts_diff[])
461 symbol= cf - 65535 + 63;
468 for (symbol = 0; counts[symbol + 1] <= cf; symbol++);
478 int lim = rice->
k ? (1 << (rice->
k + 4)) : 0;
479 rice->
ksum += ((x + 1) / 2) - ((rice->
ksum + 16) >> 5);
481 if (rice->
ksum < lim)
483 else if (rice->
ksum >= (1 << (rice->
k + 5)))
502 unsigned int x, overflow;
507 while (overflow >= 16) {
516 x = (overflow << rice->
k) +
get_bits(gb, rice->
k);
518 rice->
ksum += x - (rice->
ksum + 8 >> 4);
519 if (rice->
ksum < (rice->
k ? 1 << (rice->
k + 4) : 0))
521 else if (rice->
ksum >= (1 << (rice->
k + 5)) && rice->
k < 24)
533 unsigned int x, overflow;
542 tmpk = (rice->
k < 1) ? 0 : rice->
k - 1;
550 }
else if (tmpk <= 32) {
557 x += overflow << tmpk;
570 unsigned int x, overflow;
573 pivot = rice->
ksum >> 5;
584 if (pivot < 0x10000) {
588 int base_hi = pivot, base_lo;
591 while (base_hi & ~0xFFFF) {
600 base = (base_hi << bbits) + base_lo;
603 x = base + overflow * pivot;
618 int ksummax, ksummin;
621 for (i = 0; i <
FFMIN(blockstodecode, 5); i++) {
623 rice->
ksum += out[i];
626 for (; i <
FFMIN(blockstodecode, 64); i++) {
628 rice->
ksum += out[i];
631 ksummax = 1 << rice->
k + 7;
632 ksummin = rice->
k ? (1 << rice->
k + 6) : 0;
633 for (; i < blockstodecode; i++) {
635 rice->
ksum += out[i] - out[i - 64];
636 while (rice->
ksum < ksummin) {
638 ksummin = rice->
k ? ksummin >> 1 : 0;
641 while (rice->
ksum >= ksummax) {
646 ksummin = ksummin ? ksummin << 1 : 128;
650 for (i = 0; i < blockstodecode; i++) {
652 out[i] = (out[i] >> 1) + 1;
654 out[i] = -(out[i] >> 1);
676 while (blockstodecode--)
684 int blocks = blockstodecode;
686 while (blockstodecode--)
696 while (blockstodecode--)
704 int blocks = blockstodecode;
706 while (blockstodecode--)
721 while (blockstodecode--) {
731 while (blockstodecode--)
740 while (blockstodecode--) {
752 ctx->
CRC = bytestream_get_be32(&ctx->
ptr);
760 ctx->
CRC &= ~0x80000000;
841 return (x < 0) - (x > 0);
857 predictionA = p->
buf[delayA] * 2 - p->
buf[delayA - 1];
860 if ((decoded ^ predictionA) > 0)
872 const int delayA,
const int delayB,
873 const int start,
const int shift)
875 int32_t predictionA, predictionB, sign;
888 d1 = (p->
buf[delayA] - p->
buf[delayA - 1]) << 1;
889 d0 = p->
buf[delayA] + ((p->
buf[delayA - 2] - p->
buf[delayA - 1]) << 3);
890 d3 = p->
buf[delayB] * 2 - p->
buf[delayB - 1];
921 memset(coeffs, 0, order *
sizeof(*coeffs));
922 for (i = 0; i < order; i++)
923 delay[i] = buffer[i];
924 for (i = order; i < length; i++) {
927 for (j = 0; j < order; j++) {
928 dotprod += delay[j] * coeffs[j];
929 coeffs[j] -= (((delay[j] >> 30) & 2) - 1) * sign;
931 buffer[i] -= dotprod >> shift;
932 for (j = 0; j < order - 1; j++)
933 delay[j] = delay[j + 1];
934 delay[order - 1] = buffer[i];
942 int32_t coeffs[8] = { 0 }, delay[8] = { 0 };
944 for (i = 0; i < length; i++) {
947 for (j = 7; j >= 0; j--) {
948 dotprod += delay[j] * coeffs[j];
949 coeffs[j] -= (((delay[j] >> 30) & 2) - 1) * sign;
951 for (j = 7; j > 0; j--)
952 delay[j] = delay[j - 1];
953 delay[0] = buffer[i];
954 buffer[i] -= dotprod >> 9;
963 int32_t coeffs[256], delay[256];
964 int start = 4, shift = 10;
971 int order = 128,
shift2 = 11;
986 int X = *decoded0,
Y = *decoded1;
1018 int32_t coeffs[256], delay[256];
1019 int start = 4, shift = 10;
1025 int order = 128,
shift2 = 11;
1068 d0 = p->
buf[delayA ];
1069 d1 = p->
buf[delayA ] - p->
buf[delayA - 1];
1070 d2 = p->
buf[delayA - 1] - p->
buf[delayA - 2];
1071 d3 = p->
buf[delayA - 2] - p->
buf[delayA - 3];
1100 int Y = *decoded1, X = *decoded0;
1142 const int delayA,
const int delayB,
1143 const int adaptA,
const int adaptB)
1145 int32_t predictionA, predictionB, sign;
1149 p->
buf[delayA - 1] = p->
buf[delayA] - p->
buf[delayA - 1];
1160 p->
buf[delayB - 1] = p->
buf[delayB] - p->
buf[delayB - 1];
1170 p->
lastA[
filter] = decoded + ((predictionA + (predictionB >> 1)) >> 10);
1220 int32_t predictionA, currentA,
A, sign;
1224 currentA = p->
lastA[0];
1237 currentA = A + (predictionA >> 10);
1258 *(decoded0++) = p->
filterA[0];
1261 p->
lastA[0] = currentA;
1283 int32_t *
data,
int count,
int order,
int fracbits)
1294 res = (res + (1 << (fracbits - 1))) >> fracbits;
1299 *f->
delay++ = av_clip_int16(res);
1301 if (version < 3980) {
1303 f->
adaptcoeffs[0] = (res == 0) ? 0 : ((res >> 28) & 8) - 4;
1310 absres =
FFABS(res);
1312 *f->
adaptcoeffs = ((res & (-1<<31)) ^ (-1<<30)) >>
1313 (25 + (absres <= f->
avg*3) + (absres <= f->avg*4/3));
1317 f->
avg += (absres - f->
avg) / 16;
1338 int count,
int order,
int fracbits)
1413 left = *decoded1 - (*decoded0 / 2);
1414 right = left + *decoded0;
1416 *(decoded0++) = left;
1417 *(decoded1++) = right;
1422 int *got_frame_ptr,
AVPacket *avpkt)
1438 uint32_t nblocks, offset;
1445 if (avpkt->
size < 8) {
1449 buf_size = avpkt->
size & ~3;
1450 if (buf_size != avpkt->
size) {
1452 "extra bytes at the end will be skipped.\n");
1461 memset(s->
data + (buf_size & ~3), 0, buf_size & 3);
1465 nblocks = bytestream_get_be32(&s->
ptr);
1466 offset = bytestream_get_be32(&s->
ptr);
1486 if (!nblocks || nblocks > INT_MAX) {
1544 for (ch = 0; ch < s->
channels; ch++) {
1546 for (i = 0; i < blockstodecode; i++)
1547 *sample8++ = (s->
decoded[ch][i] + 0x80) & 0xff;
1551 for (ch = 0; ch < s->
channels; ch++) {
1552 sample16 = (int16_t *)frame->
data[ch];
1553 for (i = 0; i < blockstodecode; i++)
1554 *sample16++ = s->
decoded[ch][i];
1558 for (ch = 0; ch < s->
channels; ch++) {
1560 for (i = 0; i < blockstodecode; i++)
1561 *sample24++ = s->
decoded[ch][i] << 8;
1579 #define OFFSET(x) offsetof(APEContext, x)
1580 #define PAR (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM)
1583 {
"all",
"no maximum. decode all samples for each packet at once", 0,
AV_OPT_TYPE_CONST, { .i64 = INT_MAX }, INT_MIN, INT_MAX,
PAR,
"max_samples" },
static int init_frame_decoder(APEContext *ctx)
static const int32_t initial_coeffs_3930[4]
void(* bswap_buf)(uint32_t *dst, const uint32_t *src, int w)
static void decode_array_0000(APEContext *ctx, GetBitContext *gb, int32_t *out, APERice *rice, int blockstodecode)
int compression_level
compression levels
static av_always_inline int filter_3800(APEPredictor *p, const int decoded, const int filter, const int delayA, const int delayB, const int start, const int shift)
int32_t coeffsB[2][5]
adaption coefficients
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
This structure describes decoded (raw) audio or video data.
static void range_start_decoding(APEContext *ctx)
Start the decoder.
static void apply_filter(APEContext *ctx, APEFilter *f, int32_t *data0, int32_t *data1, int count, int order, int fracbits)
int fileversion
codec version, very important in decoding process
void ff_apedsp_init_arm(APEDSPContext *c)
static void entropy_decode_stereo_0000(APEContext *ctx, int blockstodecode)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
#define AV_LOG_WARNING
Something somehow does not look correct.
static void skip_bits_long(GetBitContext *s, int n)
static int APESIGN(int32_t x)
Get inverse sign of integer (-1 for positive, 1 for negative and 0 for zero)
static void update_rice(APERice *rice, unsigned int x)
static void entropy_decode_stereo_3900(APEContext *ctx, int blockstodecode)
static av_cold int ape_decode_init(AVCodecContext *avctx)
unsigned int buffer
buffer for input/output
static int init_entropy_decoder(APEContext *ctx)
static void ape_flush(AVCodecContext *avctx)
static void entropy_decode_stereo_3930(APEContext *ctx, int blockstodecode)
void ff_apedsp_init_ppc(APEDSPContext *c)
static av_always_inline int predictor_update_3930(APEPredictor *p, const int decoded, const int filter, const int delayA)
#define AV_CH_LAYOUT_STEREO
int16_t * filterbuf[APE_FILTER_LEVELS]
filter memory
static void predictor_decode_mono_3800(APEContext *ctx, int count)
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
#define av_assert0(cond)
assert() equivalent, that is always enabled.
static int ape_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
static int decode(MimicContext *ctx, int quality, int num_coeffs, int is_iframe)
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
enum AVSampleFormat sample_fmt
audio sample format
int16_t * delay
filtered values
static void do_init_filter(APEFilter *f, int16_t *buf, int order)
static const int32_t initial_coeffs_a_3800[3]
static void entropy_decode_stereo_3860(APEContext *ctx, int blockstodecode)
static void entropy_decode_mono_3990(APEContext *ctx, int blockstodecode)
static void ape_unpack_mono(APEContext *ctx, int count)
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
APERangecoder rc
rangecoder used to decode actual values
#define CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
static const uint8_t ape_filter_fracbits[5][APE_FILTER_LEVELS]
Filter fraction bits depending on compression level.
static void ape_apply_filters(APEContext *ctx, int32_t *decoded0, int32_t *decoded1, int count)
bitstream reader API header.
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
void(* entropy_decode_stereo)(struct APEContext *ctx, int blockstodecode)
static const uint16_t counts_3970[22]
Fixed probabilities for symbols in Monkey Audio version 3.97.
static void range_dec_normalize(APEContext *ctx)
Perform normalization.
static int get_bits_left(GetBitContext *gb)
static const uint16_t counts_diff_3980[21]
Probability ranges for symbols in Monkey Audio version 3.98.
int32_t(* scalarproduct_and_madd_int16)(int16_t *v1, const int16_t *v2, const int16_t *v3, int len, int mul)
Calculate scalar product of v1 and v2, and v1[i] += v3[i] * mul.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static av_cold int ape_decode_close(AVCodecContext *avctx)
#define CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
static int ape_decode_value_3900(APEContext *ctx, APERice *rice)
int32_t historybuffer[HISTORY_SIZE+PREDICTOR_SIZE]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
simple assert() macros that are a bit more flexible than ISO C assert().
void av_log(void *avcl, int level, const char *fmt,...)
const char * name
Name of the codec implementation.
static int range_decode_culshift(APEContext *ctx, int shift)
Decode value with given size in bits.
#define APE_FILTER_LEVELS
uint64_t channel_layout
Audio channel layout.
static int range_decode_bits(APEContext *ctx, int n)
Decode n bits (n <= 16) without modelling.
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
audio channel layout utility functions
static void predictor_decode_mono_3930(APEContext *ctx, int count)
uint8_t * data
current frame data
static const uint16_t ape_filter_orders[5][APE_FILTER_LEVELS]
Filter orders depending on compression level.
static int get_rice_ook(GetBitContext *gb, int k)
static void long_filter_high_3800(int32_t *buffer, int order, int shift, int32_t *coeffs, int32_t *delay, int length)
static av_always_inline int filter_fast_3320(APEPredictor *p, const int decoded, const int filter, const int delayA)
static void ape_unpack_stereo(APEContext *ctx, int count)
const uint8_t * ptr
current position in frame data
static int range_decode_culfreq(APEContext *ctx, int tot_f)
Calculate culmulative frequency for next symbol.
static void predictor_decode_stereo_3930(APEContext *ctx, int count)
uint32_t help
bytes_to_follow resp. intermediate value
static void entropy_decode_stereo_3990(APEContext *ctx, int blockstodecode)
#define APE_FRAMECODE_PSEUDO_STEREO
uint32_t range
length of interval
if(ac->has_optimized_func)
int samples
samples left to decode in current frame
#define AVERROR_PATCHWELCOME
Not yet implemented in Libav, patches welcome.
int fset
which filter set to use (calculated from compression level)
static int ape_decode_value_3860(APEContext *ctx, GetBitContext *gb, APERice *rice)
APERice riceX
rice code parameters for the second channel
Libavcodec external API header.
AVSampleFormat
Audio Sample Formats.
static void predictor_decode_stereo_3950(APEContext *ctx, int count)
static void predictor_decode_stereo_3800(APEContext *ctx, int count)
#define APE_FRAMECODE_STEREO_SILENCE
static void init_filter(APEContext *ctx, APEFilter *f, int16_t *buf, int order)
int frameflags
frame flags
main external API structure.
static void(WINAPI *cond_broadcast)(pthread_cond_t *cond)
static void close(AVCodecParserContext *s)
static int ape_decode_value_3990(APEContext *ctx, APERice *rice)
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static const uint16_t counts_3980[22]
Fixed probabilities for symbols in Monkey Audio version 3.98.
static int range_get_symbol(APEContext *ctx, const uint16_t counts[], const uint16_t counts_diff[])
Decode symbol.
Describe the class of an AVClass context structure.
uint32_t low
low end of interval
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
int flags
global decoder flags
void(* predictor_decode_mono)(struct APEContext *ctx, int count)
APECompressionLevel
Possible compression levels.
int32_t coeffsA[2][4]
adaption coefficients
static void range_decode_update(APEContext *ctx, int sy_f, int lt_f)
Update decoding state.
static void entropy_decode_mono_3900(APEContext *ctx, int blockstodecode)
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
static const int32_t initial_coeffs_fast_3320[1]
static void do_apply_filter(APEContext *ctx, int version, APEFilter *f, int32_t *data, int count, int order, int fracbits)
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
#define PREDICTOR_SIZE
Total size of all predictor histories.
void(* predictor_decode_stereo)(struct APEContext *ctx, int count)
static const uint16_t counts_diff_3970[21]
Probability ranges for symbols in Monkey Audio version 3.97.
int blocks_per_loop
maximum number of samples to decode for each call
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_dlog(ac->avr,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> out
#define CODEC_CAP_SUBFRAMES
Codec can output multiple frames per AVPacket Normally demuxers return one frame at a time...
void av_fast_malloc(void *ptr, unsigned int *size, size_t min_size)
Allocate a buffer, reusing the given one if large enough.
uint8_t * data_end
frame data end
common internal api header.
APERice riceY
rice code parameters for the first channel
static const int shift2[6]
static int get_unary(GetBitContext *gb, int stop, int len)
Get unary code of limited length.
#define FF_ALLOC_OR_GOTO(ctx, p, size, label)
static av_cold void flush(AVCodecContext *avctx)
Flush (reset) the frame ID after seeking.
APEFilter filters[APE_FILTER_LEVELS][2]
filters used for reconstruction
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)
int16_t * coeffs
actual coefficients used in filtering
static av_cold int init(AVCodecParserContext *s)
static void init_predictor_decoder(APEContext *ctx)
static const int32_t initial_coeffs_b_3800[2]
APEPredictor predictor
predictor used for final reconstruction
static const AVClass ape_decoder_class
void(* entropy_decode_mono)(struct APEContext *ctx, int blockstodecode)
int channels
number of audio channels
static void long_filter_ehigh_3830(int32_t *buffer, int length)
static void predictor_decode_mono_3950(APEContext *ctx, int count)
Filters applied to the decoded data.
static int32_t scalarproduct_and_madd_int16_c(int16_t *v1, const int16_t *v2, const int16_t *v3, int order, int mul)
int32_t * decoded[MAX_CHANNELS]
decoded data for each channel
void ff_apedsp_init_x86(APEDSPContext *c)
int data_size
frame data allocated size
static const AVOption options[]
#define AV_CH_LAYOUT_MONO
int16_t * adaptcoeffs
adaptive filter coefficients used for correcting of actual filter coefficients
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
static void entropy_decode_mono_0000(APEContext *ctx, int blockstodecode)
int16_t * historybuffer
filter memory
static void entropy_decode_mono_3860(APEContext *ctx, int blockstodecode)