23 #include "libavutil/avassert.h"
24 #include "libavutil/channel_layout.h"
25 #include "libavutil/opt.h"
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);
309 #define TOP_VALUE ((unsigned int)1 << (CODE_BITS-1))
310 #define SHIFT_BITS (CODE_BITS - 9)
311 #define EXTRA_BITS ((CODE_BITS-2) % 8 + 1)
312 #define BOTTOM_VALUE (TOP_VALUE >> 8)
385 #define MODEL_ELEMENTS 64
391 0, 14824, 28224, 39348, 47855, 53994, 58171, 60926,
392 62682, 63786, 64463, 64878, 65126, 65276, 65365, 65419,
393 65450, 65469, 65480, 65487, 65491, 65493,
400 14824, 13400, 11124, 8507, 6139, 4177, 2755, 1756,
401 1104, 677, 415, 248, 150, 89, 54, 31,
409 0, 19578, 36160, 48417, 56323, 60899, 63265, 64435,
410 64971, 65232, 65351, 65416, 65447, 65466, 65476, 65482,
411 65485, 65488, 65490, 65491, 65492, 65493,
418 19578, 16582, 12257, 7906, 4576, 2366, 1170, 536,
419 261, 119, 65, 31, 19, 10, 6, 3,
430 const uint16_t counts[],
431 const uint16_t counts_diff[])
438 symbol= cf - 65535 + 63;
445 for (symbol = 0; counts[symbol + 1] <= cf; symbol++);
455 int lim = rice->
k ? (1 << (rice->
k + 4)) : 0;
456 rice->
ksum += ((x + 1) / 2) - ((rice->
ksum + 16) >> 5);
458 if (rice->
ksum < lim)
460 else if (rice->
ksum >= (1 << (rice->
k + 5)))
479 unsigned int x, overflow;
484 while (overflow >= 16) {
493 x = (overflow << rice->
k) +
get_bits(gb, rice->
k);
498 rice->
ksum += x - (rice->
ksum + 8 >> 4);
499 if (rice->
ksum < (rice->
k ? 1 << (rice->
k + 4) : 0))
501 else if (rice->
ksum >= (1 << (rice->
k + 5)) && rice->
k < 24)
513 unsigned int x, overflow;
522 tmpk = (rice->
k < 1) ? 0 : rice->
k - 1;
530 }
else if (tmpk <= 32) {
537 x += overflow << tmpk;
550 unsigned int x, overflow;
553 pivot = rice->
ksum >> 5;
564 if (pivot < 0x10000) {
568 int base_hi = pivot, base_lo;
571 while (base_hi & ~0xFFFF) {
580 base = (base_hi << bbits) + base_lo;
583 x = base + overflow * pivot;
598 int ksummax, ksummin;
601 for (i = 0; i < 5; i++) {
603 rice->
ksum += out[i];
606 for (; i < 64; i++) {
608 rice->
ksum += out[i];
611 ksummax = 1 << rice->
k + 7;
612 ksummin = rice->
k ? (1 << rice->
k + 6) : 0;
613 for (; i < blockstodecode; i++) {
615 rice->
ksum += out[i] - out[i - 64];
616 while (rice->
ksum < ksummin) {
618 ksummin = rice->
k ? ksummin >> 1 : 0;
621 while (rice->
ksum >= ksummax) {
626 ksummin = ksummin ? ksummin << 1 : 128;
630 for (i = 0; i < blockstodecode; i++) {
632 out[i] = (out[i] >> 1) + 1;
634 out[i] = -(out[i] >> 1);
656 while (blockstodecode--)
664 int blocks = blockstodecode;
666 while (blockstodecode--)
676 while (blockstodecode--)
684 int blocks = blockstodecode;
686 while (blockstodecode--)
701 while (blockstodecode--) {
711 while (blockstodecode--)
720 while (blockstodecode--) {
732 ctx->
CRC = bytestream_get_be32(&ctx->
ptr);
740 ctx->
CRC &= ~0x80000000;
821 return (x < 0) - (x > 0);
837 predictionA = p->
buf[delayA] * 2 - p->
buf[delayA - 1];
840 if ((decoded ^ predictionA) > 0)
852 const int delayA,
const int delayB,
855 int32_t predictionA, predictionB, sign;
868 d1 = (p->
buf[delayA] - p->
buf[delayA - 1]) << 1;
869 d0 = p->
buf[delayA] + ((p->
buf[delayA - 2] - p->
buf[delayA - 1]) << 3);
870 d3 = p->
buf[delayB] * 2 - p->
buf[delayB - 1];
901 memset(coeffs, 0, order *
sizeof(*coeffs));
902 for (i = 0; i < order; i++)
903 delay[i] = buffer[i];
904 for (i = order; i <
length; i++) {
907 for (j = 0; j < order; j++) {
908 dotprod += delay[j] * coeffs[j];
909 coeffs[j] -= (((delay[j] >> 30) & 2) - 1) * sign;
911 buffer[i] -= dotprod >>
shift;
912 for (j = 0; j < order - 1; j++)
913 delay[j] = delay[j + 1];
914 delay[order - 1] = buffer[i];
924 memset(coeffs, 0,
sizeof(coeffs));
925 memset(delay, 0,
sizeof(delay));
926 for (i = 0; i <
length; i++) {
929 for (j = 7; j >= 0; j--) {
930 dotprod += delay[j] * coeffs[j];
931 coeffs[j] -= (((delay[j] >> 30) & 2) - 1) * sign;
933 for (j = 7; j > 0; j--)
934 delay[j] = delay[j - 1];
935 delay[0] = buffer[i];
936 buffer[i] -= dotprod >> 9;
953 int order = 128,
shift2 = 11;
968 int X = *decoded0,
Y = *decoded1;
1007 int order = 128,
shift2 = 11;
1050 d0 = p->
buf[delayA ];
1051 d1 = p->
buf[delayA ] - p->
buf[delayA - 1];
1052 d2 = p->
buf[delayA - 1] - p->
buf[delayA - 2];
1053 d3 = p->
buf[delayA - 2] - p->
buf[delayA - 3];
1082 int Y = *decoded1, X = *decoded0;
1124 const int delayA,
const int delayB,
1125 const int adaptA,
const int adaptB)
1127 int32_t predictionA, predictionB, sign;
1131 p->
buf[delayA - 1] = p->
buf[delayA] - p->
buf[delayA - 1];
1142 p->
buf[delayB - 1] = p->
buf[delayB] - p->
buf[delayB - 1];
1152 p->
lastA[
filter] = decoded + ((predictionA + (predictionB >> 1)) >> 10);
1202 int32_t predictionA, currentA,
A, sign;
1206 currentA = p->
lastA[0];
1219 currentA = A + (predictionA >> 10);
1240 *(decoded0++) = p->
filterA[0];
1243 p->
lastA[0] = currentA;
1275 res = (res + (1 << (fracbits - 1))) >> fracbits;
1280 *f->
delay++ = av_clip_int16(res);
1282 if (version < 3980) {
1284 f->
adaptcoeffs[0] = (res == 0) ? 0 : ((res >> 28) & 8) - 4;
1291 absres =
FFABS(res);
1293 *f->
adaptcoeffs = ((res & (-1<<31)) ^ (-1<<30)) >>
1294 (25 + (absres <= f->
avg*3) + (absres <= f->avg*4/3));
1298 f->
avg += (absres - f->
avg) / 16;
1319 int count,
int order,
int fracbits)
1394 left = *decoded1 - (*decoded0 / 2);
1395 right = left + *decoded0;
1397 *(decoded0++) = left;
1398 *(decoded1++) = right;
1403 int *got_frame_ptr,
AVPacket *avpkt)
1419 uint32_t nblocks,
offset;
1426 if (avpkt->
size < 8) {
1430 buf_size = avpkt->
size & ~3;
1431 if (buf_size != avpkt->
size) {
1433 "extra bytes at the end will be skipped.\n");
1441 memset(s->
data + (buf_size & ~3), 0, buf_size & 3);
1445 nblocks = bytestream_get_be32(&s->
ptr);
1446 offset = bytestream_get_be32(&s->
ptr);
1466 if (!nblocks || nblocks > INT_MAX) {
1520 for (ch = 0; ch < s->
channels; ch++) {
1522 for (i = 0; i < blockstodecode; i++)
1523 *sample8++ = (s->
decoded[ch][i] + 0x80) & 0xff;
1527 for (ch = 0; ch < s->
channels; ch++) {
1528 sample16 = (int16_t *)frame->
data[ch];
1529 for (i = 0; i < blockstodecode; i++)
1530 *sample16++ = s->
decoded[ch][i];
1534 for (ch = 0; ch < s->
channels; ch++) {
1536 for (i = 0; i < blockstodecode; i++)
1537 *sample24++ = s->
decoded[ch][i] << 8;
1555 #define OFFSET(x) offsetof(APEContext, x)
1556 #define PAR (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM)
1559 {
"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_PATCHWELCOME
static int shift(int a, int b)
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.
#define LIBAVUTIL_VERSION_INT
static void skip_bits_long(GetBitContext *s, int n)
static av_cold int init(AVCodecContext *avctx)
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)
void av_log(void *avcl, int level, const char *fmt,...) av_printf_format(3
Send the specified message to the log if the level is less than or equal to the current av_log_level...
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
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
static void predictor_decode_mono_3800(APEContext *ctx, int count)
void av_freep(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
if((e=av_dict_get(options,"", NULL, AV_DICT_IGNORE_SUFFIX)))
void av_fast_malloc(void *ptr, unsigned int *size, size_t min_size)
Allocate a buffer, reusing the given one if large enough.
static int ape_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
const char * av_default_item_name(void *ctx)
Return the context name.
enum AVSampleFormat sample_fmt
audio sample format
int16_t * delay
filtered values
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
static void do_init_filter(APEFilter *f, int16_t *buf, int order)
static const uint8_t offset[511][2]
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)
const char * name
Name of the codec implementation.
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)
const OptionDef options[]
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.
static int range_decode_culshift(APEContext *ctx, int shift)
Decode value with given size in bits.
#define APE_FILTER_LEVELS
Libavcodec external API header.
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)
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
static void flush(AVCodecContext *avctx)
uint32_t range
length of interval
int samples
samples left to decode in current frame
int fset
which filter set to use (calculated from compression level)
static int ape_decode_value_3860(APEContext *ctx, GetBitContext *gb, APERice *rice)
AVSampleFormat
Audio Sample Formats.
APERice riceX
rice code parameters for the second channel
static void predictor_decode_stereo_3950(APEContext *ctx, int count)
typedef void(RENAME(mix_any_func_type))
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 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)
#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...
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)
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)
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 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
#define AVERROR_INVALIDDATA
void(* entropy_decode_mono)(struct APEContext *ctx, int blockstodecode)
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){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));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);ff_audio_convert_init_arm(ac);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
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 enum AVSampleFormat sample_fmts[]
int32_t * decoded[MAX_CHANNELS]
decoded data for each channel
static int decode(AVCodecContext *avctx, void *data, int *got_frame, AVPacket *avpkt)
#define av_assert0(cond)
assert() equivalent, that is always enabled.
int data_size
frame data allocated size
#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
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
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)