38 #define MAX_CHANNELS 2
39 #define MAX_BYTESPERSAMPLE 3
41 #define APE_FRAMECODE_MONO_SILENCE 1
42 #define APE_FRAMECODE_STEREO_SILENCE 3
43 #define APE_FRAMECODE_PSEUDO_STEREO 4
45 #define HISTORY_SIZE 512
46 #define PREDICTOR_ORDER 8
48 #define PREDICTOR_SIZE 50
50 #define YDELAYA (18 + PREDICTOR_ORDER*4)
51 #define YDELAYB (18 + PREDICTOR_ORDER*3)
52 #define XDELAYA (18 + PREDICTOR_ORDER*2)
53 #define XDELAYB (18 + PREDICTOR_ORDER)
55 #define YADAPTCOEFFSA 18
56 #define XADAPTCOEFFSA 14
57 #define YADAPTCOEFFSB 10
58 #define XADAPTCOEFFSB 5
73 #define APE_FILTER_LEVELS 3
239 "%d bits per coded sample", s->
bps);
308 #define TOP_VALUE ((unsigned int)1 << (CODE_BITS-1))
309 #define SHIFT_BITS (CODE_BITS - 9)
310 #define EXTRA_BITS ((CODE_BITS-2) % 8 + 1)
311 #define BOTTOM_VALUE (TOP_VALUE >> 8)
384 #define MODEL_ELEMENTS 64
390 0, 14824, 28224, 39348, 47855, 53994, 58171, 60926,
391 62682, 63786, 64463, 64878, 65126, 65276, 65365, 65419,
392 65450, 65469, 65480, 65487, 65491, 65493,
399 14824, 13400, 11124, 8507, 6139, 4177, 2755, 1756,
400 1104, 677, 415, 248, 150, 89, 54, 31,
408 0, 19578, 36160, 48417, 56323, 60899, 63265, 64435,
409 64971, 65232, 65351, 65416, 65447, 65466, 65476, 65482,
410 65485, 65488, 65490, 65491, 65492, 65493,
417 19578, 16582, 12257, 7906, 4576, 2366, 1170, 536,
418 261, 119, 65, 31, 19, 10, 6, 3,
429 const uint16_t counts[],
430 const uint16_t counts_diff[])
437 symbol= cf - 65535 + 63;
444 for (symbol = 0; counts[symbol + 1] <= cf; symbol++);
454 int lim = rice->
k ? (1 << (rice->
k + 4)) : 0;
455 rice->
ksum += ((x + 1) / 2) - ((rice->
ksum + 16) >> 5);
457 if (rice->
ksum < lim)
459 else if (rice->
ksum >= (1 << (rice->
k + 5)))
478 unsigned int x, overflow;
483 while (overflow >= 16) {
492 x = (overflow << rice->
k) +
get_bits(gb, rice->
k);
494 rice->
ksum += x - (rice->
ksum + 8 >> 4);
495 if (rice->
ksum < (rice->
k ? 1 << (rice->
k + 4) : 0))
497 else if (rice->
ksum >= (1 << (rice->
k + 5)) && rice->
k < 24)
509 unsigned int x, overflow;
518 tmpk = (rice->
k < 1) ? 0 : rice->
k - 1;
522 else if (tmpk <= 32) {
529 x += overflow << tmpk;
542 unsigned int x, overflow;
545 pivot = rice->
ksum >> 5;
556 if (pivot < 0x10000) {
560 int base_hi = pivot, base_lo;
563 while (base_hi & ~0xFFFF) {
572 base = (base_hi << bbits) + base_lo;
575 x = base + overflow * pivot;
590 int ksummax, ksummin;
593 for (i = 0; i <
FFMIN(blockstodecode, 5); i++) {
595 rice->
ksum += out[i];
598 for (; i <
FFMIN(blockstodecode, 64); i++) {
600 rice->
ksum += out[i];
603 ksummax = 1 << rice->
k + 7;
604 ksummin = rice->
k ? (1 << rice->
k + 6) : 0;
605 for (; i < blockstodecode; i++) {
607 rice->
ksum += out[i] - out[i - 64];
608 while (rice->
ksum < ksummin) {
610 ksummin = rice->
k ? ksummin >> 1 : 0;
613 while (rice->
ksum >= ksummax) {
618 ksummin = ksummin ? ksummin << 1 : 128;
622 for (i = 0; i < blockstodecode; i++) {
624 out[i] = (out[i] >> 1) + 1;
626 out[i] = -(out[i] >> 1);
648 while (blockstodecode--)
656 int blocks = blockstodecode;
658 while (blockstodecode--)
668 while (blockstodecode--)
676 int blocks = blockstodecode;
678 while (blockstodecode--)
693 while (blockstodecode--) {
703 while (blockstodecode--)
712 while (blockstodecode--) {
724 ctx->
CRC = bytestream_get_be32(&ctx->
ptr);
732 ctx->
CRC &= ~0x80000000;
813 return (x < 0) - (x > 0);
829 predictionA = p->
buf[delayA] * 2 - p->
buf[delayA - 1];
832 if ((decoded ^ predictionA) > 0)
844 const int delayA,
const int delayB,
845 const int start,
const int shift)
847 int32_t predictionA, predictionB, sign;
860 d1 = (p->
buf[delayA] - p->
buf[delayA - 1]) << 1;
861 d0 = p->
buf[delayA] + ((p->
buf[delayA - 2] - p->
buf[delayA - 1]) << 3);
862 d3 = p->
buf[delayB] * 2 - p->
buf[delayB - 1];
893 memset(coeffs, 0, order *
sizeof(*coeffs));
894 for (i = 0; i < order; i++)
895 delay[i] = buffer[i];
896 for (i = order; i < length; i++) {
899 for (j = 0; j < order; j++) {
900 dotprod += delay[j] * coeffs[j];
901 coeffs[j] -= (((delay[j] >> 30) & 2) - 1) * sign;
903 buffer[i] -= dotprod >> shift;
904 for (j = 0; j < order - 1; j++)
905 delay[j] = delay[j + 1];
906 delay[order - 1] = buffer[i];
916 memset(coeffs, 0,
sizeof(coeffs));
917 memset(delay, 0,
sizeof(delay));
918 for (i = 0; i < length; i++) {
921 for (j = 7; j >= 0; j--) {
922 dotprod += delay[j] * coeffs[j];
923 coeffs[j] -= (((delay[j] >> 30) & 2) - 1) * sign;
925 for (j = 7; j > 0; j--)
926 delay[j] = delay[j - 1];
927 delay[0] = buffer[i];
928 buffer[i] -= dotprod >> 9;
938 int start = 4, shift = 10;
945 int order = 128,
shift2 = 11;
960 int X = *decoded0,
Y = *decoded1;
993 int start = 4, shift = 10;
999 int order = 128,
shift2 = 11;
1042 d0 = p->
buf[delayA ];
1043 d1 = p->
buf[delayA ] - p->
buf[delayA - 1];
1044 d2 = p->
buf[delayA - 1] - p->
buf[delayA - 2];
1045 d3 = p->
buf[delayA - 2] - p->
buf[delayA - 3];
1074 int Y = *decoded1, X = *decoded0;
1116 const int delayA,
const int delayB,
1117 const int adaptA,
const int adaptB)
1119 int32_t predictionA, predictionB, sign;
1123 p->
buf[delayA - 1] = p->
buf[delayA] - p->
buf[delayA - 1];
1134 p->
buf[delayB - 1] = p->
buf[delayB] - p->
buf[delayB - 1];
1144 p->
lastA[
filter] = decoded + ((predictionA + (predictionB >> 1)) >> 10);
1194 int32_t predictionA, currentA,
A, sign;
1198 currentA = p->
lastA[0];
1211 currentA = A + (predictionA >> 10);
1232 *(decoded0++) = p->
filterA[0];
1235 p->
lastA[0] = currentA;
1257 int32_t *
data,
int count,
int order,
int fracbits)
1267 res = (res + (1 << (fracbits - 1))) >> fracbits;
1272 *f->
delay++ = av_clip_int16(res);
1274 if (version < 3980) {
1276 f->
adaptcoeffs[0] = (res == 0) ? 0 : ((res >> 28) & 8) - 4;
1283 absres =
FFABS(res);
1285 *f->
adaptcoeffs = ((res & (-1<<31)) ^ (-1<<30)) >>
1286 (25 + (absres <= f->
avg*3) + (absres <= f->avg*4/3));
1290 f->
avg += (absres - f->
avg) / 16;
1311 int count,
int order,
int fracbits)
1386 left = *decoded1 - (*decoded0 / 2);
1387 right = left + *decoded0;
1389 *(decoded0++) = left;
1390 *(decoded1++) = right;
1395 int *got_frame_ptr,
AVPacket *avpkt)
1411 uint32_t nblocks, offset;
1418 if (avpkt->
size < 8) {
1422 buf_size = avpkt->
size & ~3;
1423 if (buf_size != avpkt->
size) {
1425 "extra bytes at the end will be skipped.\n");
1433 memset(s->
data + (buf_size & ~3), 0, buf_size & 3);
1437 nblocks = bytestream_get_be32(&s->
ptr);
1438 offset = bytestream_get_be32(&s->
ptr);
1458 if (!nblocks || nblocks > INT_MAX) {
1515 for (ch = 0; ch < s->
channels; ch++) {
1517 for (i = 0; i < blockstodecode; i++)
1518 *sample8++ = (s->
decoded[ch][i] + 0x80) & 0xff;
1522 for (ch = 0; ch < s->
channels; ch++) {
1523 sample16 = (int16_t *)frame->
data[ch];
1524 for (i = 0; i < blockstodecode; i++)
1525 *sample16++ = s->
decoded[ch][i];
1529 for (ch = 0; ch < s->
channels; ch++) {
1531 for (i = 0; i < blockstodecode; i++)
1532 *sample24++ = s->
decoded[ch][i] << 8;
1550 #define OFFSET(x) offsetof(APEContext, x)
1551 #define PAR (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM)
1554 {
"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]
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)
static const int16_t coeffs[28]
av_cold void ff_dsputil_init(DSPContext *c, AVCodecContext *avctx)
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
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)
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
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=av_sample_fmt_is_planar(in_fmt);out_planar=av_sample_fmt_is_planar(out_fmt);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_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_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.
#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...
void(* bswap_buf)(uint32_t *dst, const uint32_t *src, int w)
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.
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.
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
#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)
AVSampleFormat
Audio Sample Formats.
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.
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.
int32_t * decoded[MAX_CHANNELS]
decoded data for each channel
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)