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 
  241                               "%d bits per coded sample", s->
bps);
 
  312 #define TOP_VALUE    ((unsigned int)1 << (CODE_BITS-1)) 
  313 #define SHIFT_BITS   (CODE_BITS - 9) 
  314 #define EXTRA_BITS   ((CODE_BITS-2) % 8 + 1) 
  315 #define BOTTOM_VALUE (TOP_VALUE >> 8) 
  388 #define MODEL_ELEMENTS 64 
  394         0, 14824, 28224, 39348, 47855, 53994, 58171, 60926,
 
  395     62682, 63786, 64463, 64878, 65126, 65276, 65365, 65419,
 
  396     65450, 65469, 65480, 65487, 65491, 65493,
 
  403     14824, 13400, 11124, 8507, 6139, 4177, 2755, 1756,
 
  404     1104, 677, 415, 248, 150, 89, 54, 31,
 
  412         0, 19578, 36160, 48417, 56323, 60899, 63265, 64435,
 
  413     64971, 65232, 65351, 65416, 65447, 65466, 65476, 65482,
 
  414     65485, 65488, 65490, 65491, 65492, 65493,
 
  421     19578, 16582, 12257, 7906, 4576, 2366, 1170, 536,
 
  422     261, 119, 65, 31, 19, 10, 6, 3,
 
  433                                    const uint16_t counts[],
 
  434                                    const uint16_t counts_diff[])
 
  441         symbol= cf - 65535 + 63;
 
  448     for (symbol = 0; counts[symbol + 1] <= cf; symbol++);
 
  458     int lim = rice->
k ? (1 << (rice->
k + 4)) : 0;
 
  459     rice->
ksum += ((x + 1) / 2) - ((rice->
ksum + 16) >> 5);
 
  461     if (rice->
ksum < lim)
 
  463     else if (rice->
ksum >= (1 << (rice->
k + 5)))
 
  482     unsigned int x, overflow;
 
  487         while (overflow >= 16) {
 
  496         x = (overflow << rice->
k) + 
get_bits(gb, rice->
k);
 
  501     rice->
ksum += x - (rice->
ksum + 8 >> 4);
 
  502     if (rice->
ksum < (rice->
k ? 1 << (rice->
k + 4) : 0))
 
  504     else if (rice->
ksum >= (1 << (rice->
k + 5)) && rice->
k < 24)
 
  508     return ((x >> 1) ^ ((x & 1) - 1)) + 1;
 
  513     unsigned int x, overflow;
 
  522         tmpk = (rice->
k < 1) ? 0 : rice->
k - 1;
 
  530     } 
else if (tmpk <= 31) {
 
  537     x += overflow << tmpk;
 
  542     return ((x >> 1) ^ ((x & 1) - 1)) + 1;
 
  547     unsigned int x, overflow;
 
  550     pivot = rice->
ksum >> 5;
 
  561     if (pivot < 0x10000) {
 
  565         int base_hi = pivot, base_lo;
 
  568         while (base_hi & ~0xFFFF) {
 
  577         base = (base_hi << bbits) + base_lo;
 
  580     x = base + overflow * pivot;
 
  585     return ((x >> 1) ^ ((x & 1) - 1)) + 1;
 
  592     int ksummax, ksummin;
 
  595     for (i = 0; i < 
FFMIN(blockstodecode, 5); i++) {
 
  597         rice->
ksum += out[i];
 
  602     for (; i < 
FFMIN(blockstodecode, 64); i++) {
 
  604         rice->
ksum += out[i];
 
  609     ksummax = 1 << rice->
k + 7;
 
  610     ksummin = rice->
k ? (1 << rice->
k + 6) : 0;
 
  611     for (; i < blockstodecode; i++) {
 
  613         rice->
ksum += out[i] - out[i - 64];
 
  614         while (rice->
ksum < ksummin) {
 
  616             ksummin = rice->
k ? ksummin >> 1 : 0;
 
  619         while (rice->
ksum >= ksummax) {
 
  624             ksummin = ksummin ? ksummin << 1 : 128;
 
  628     for (i = 0; i < blockstodecode; i++)
 
  629         out[i] = ((out[i] >> 1) ^ ((out[i] & 1) - 1)) + 1;
 
  650     while (blockstodecode--)
 
  658     int blocks = blockstodecode;
 
  660     while (blockstodecode--)
 
  670     while (blockstodecode--)
 
  678     int blocks = blockstodecode;
 
  680     while (blockstodecode--)
 
  695     while (blockstodecode--) {
 
  705     while (blockstodecode--)
 
  714     while (blockstodecode--) {
 
  726         ctx->
CRC = bytestream_get_be32(&ctx->
ptr);
 
  734         ctx->
CRC &= ~0x80000000;
 
  815     return (x < 0) - (x > 0);
 
  831     predictionA = p->
buf[delayA] * 2 - p->
buf[delayA - 1];
 
  834     if ((decoded ^ predictionA) > 0)
 
  846                                         const int delayA,  
const int delayB,
 
  849     int32_t predictionA, predictionB, sign;
 
  862     d1 = (p->
buf[delayA] - p->
buf[delayA - 1]) << 1;
 
  863     d0 =  p->
buf[delayA] + ((p->
buf[delayA - 2] - p->
buf[delayA - 1]) << 3);
 
  864     d3 =  p->
buf[delayB] * 2 - p->
buf[delayB - 1];
 
  895     memset(coeffs, 0, order * 
sizeof(*coeffs));
 
  896     for (i = 0; i < order; i++)
 
  897         delay[i] = buffer[i];
 
  898     for (i = order; i < 
length; i++) {
 
  901         for (j = 0; j < order; j++) {
 
  902             dotprod += delay[j] * coeffs[j];
 
  903             coeffs[j] += ((delay[j] >> 31) | 1) * sign;
 
  905         buffer[i] -= dotprod >> 
shift;
 
  906         for (j = 0; j < order - 1; j++)
 
  907             delay[j] = delay[j + 1];
 
  908         delay[order - 1] = buffer[i];
 
  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] >> 31) | 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;
 
  944         int order = 128, 
shift2 = 11;
 
  959         int X = *decoded0, 
Y = *decoded1;
 
  997         int order = 128, 
shift2 = 11;
 
 1040     d0 = p->
buf[delayA    ];
 
 1041     d1 = p->
buf[delayA    ] - p->
buf[delayA - 1];
 
 1042     d2 = p->
buf[delayA - 1] - p->
buf[delayA - 2];
 
 1043     d3 = p->
buf[delayA - 2] - p->
buf[delayA - 3];
 
 1072         int Y = *decoded1, X = *decoded0;
 
 1114                                                     const int delayA,  
const int delayB,
 
 1115                                                     const int adaptA,  
const int adaptB)
 
 1117     int32_t predictionA, predictionB, sign;
 
 1121     p->
buf[delayA - 1] = p->
buf[delayA] - p->
buf[delayA - 1];
 
 1132     p->
buf[delayB - 1] = p->
buf[delayB] - p->
buf[delayB - 1];
 
 1142     p->
lastA[
filter] = decoded + ((predictionA + (predictionB >> 1)) >> 10);
 
 1192     int32_t predictionA, currentA, 
A, sign;
 
 1196     currentA = p->
lastA[0];
 
 1209         currentA = A + (predictionA >> 10);
 
 1230         *(decoded0++) = p->
filterA[0];
 
 1233     p->
lastA[0] = currentA;
 
 1266         res = (res + (1 << (fracbits - 1))) >> fracbits;
 
 1271         *f->
delay++ = av_clip_int16(res);
 
 1273         if (version < 3980) {
 
 1275             f->
adaptcoeffs[0]  = (res == 0) ? 0 : ((res >> 28) & 8) - 4;
 
 1282             absres = 
FFABS(res);
 
 1284                 *f->
adaptcoeffs = ((res & (-1<<31)) ^ (-1<<30)) >>
 
 1285                                   (25 + (absres <= f->
avg*3) + (absres <= f->avg*4/3));
 
 1289             f->
avg += (absres - f->
avg) / 16;
 
 1310                          int count, 
int order, 
int fracbits)
 
 1385         left = *decoded1 - (*decoded0 / 2);
 
 1386         right = left + *decoded0;
 
 1388         *(decoded0++) = left;
 
 1389         *(decoded1++) = right;
 
 1394                             int *got_frame_ptr, 
AVPacket *avpkt)
 
 1410         uint32_t nblocks, 
offset;
 
 1417         if (avpkt->
size < 8) {
 
 1421         buf_size = avpkt->
size & ~3;
 
 1422         if (buf_size != avpkt->
size) {
 
 1424                    "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);
 
 1459         if (!nblocks || nblocks > INT_MAX) {
 
 1514         for (ch = 0; ch < s->
channels; ch++) {
 
 1516             for (i = 0; i < blockstodecode; i++)
 
 1517                 *sample8++ = (s->
decoded[ch][i] + 0x80) & 0xff;
 
 1521         for (ch = 0; ch < s->
channels; ch++) {
 
 1522             sample16 = (int16_t *)frame->
data[ch];
 
 1523             for (i = 0; i < blockstodecode; i++)
 
 1524                 *sample16++ = s->
decoded[ch][i];
 
 1528         for (ch = 0; ch < s->
channels; ch++) {
 
 1530             for (i = 0; i < blockstodecode; i++)
 
 1531                 *sample24++ = s->
decoded[ch][i] << 8;
 
 1549 #define OFFSET(x) offsetof(APEContext, x) 
 1550 #define PAR (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM) 
 1553     { 
"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. 
 
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. 
 
ptrdiff_t const GLvoid * data
 
static void flush(AVCodecContext *avctx)
 
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 void long_filter_high_3800(int32_t *buffer, int order, int shift, int length)
 
static int init_entropy_decoder(APEContext *ctx)
 
static void ape_flush(AVCodecContext *avctx)
 
void av_fast_padded_malloc(void *ptr, unsigned int *size, size_t min_size)
Same behaviour av_fast_malloc but the buffer has additional AV_INPUT_BUFFER_PADDING_SIZE at the end w...
 
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
 
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_RL16
 
int16_t * filterbuf[APE_FILTER_LEVELS]
filter memory 
 
static void predictor_decode_mono_3800(APEContext *ctx, int count)
 
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_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
 
#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)
 
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 
 
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)
 
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(). 
 
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
 
static const uint8_t offset[127][2]
 
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. 
 
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)
 
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
 
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. 
 
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
 
static void predictor_decode_stereo_3930(APEContext *ctx, int count)
 
av_cold void ff_llauddsp_init(LLAudDSPContext *c)
 
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 
 
int samples
samples left to decode in current frame 
 
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, 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 
 
AVSampleFormat
Audio sample formats. 
 
static void predictor_decode_stereo_3950(APEContext *ctx, int count)
 
static void predictor_decode_stereo_3800(APEContext *ctx, int count)
 
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext. 
 
#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 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. 
 
#define AV_CODEC_CAP_SUBFRAMES
Codec can output multiple frames per AVPacket Normally demuxers return one frame at a time...
 
uint32_t low
low end of interval 
 
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. 
 
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 
 
static int decode(AVCodecContext *avctx, void *data, int *got_sub, AVPacket *avpkt)
 
void av_fast_malloc(void *ptr, unsigned int *size, size_t min_size)
Allocate a buffer, reusing the given one if large enough. 
 
static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
 
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)
 
int16_t * coeffs
actual coefficients used in filtering 
 
static void init_predictor_decoder(APEContext *ctx)
 
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
 
static const int32_t initial_coeffs_b_3800[2]
 
APEPredictor predictor
predictor used for final reconstruction 
 
static const AVClass ape_decoder_class
 
static const int16_t coeffs[]
 
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. 
 
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_log(ac->avr, AV_LOG_TRACE,"%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
 
static enum AVSampleFormat sample_fmts[]
 
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 
 
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators. 
 
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)