26 #define BITSTREAM_READER_LE 41 #define DSD_BYTE_READY(low,high) (!(((low) ^ (high)) & 0xff000000)) 44 #define PTABLE_BINS (1<<PTABLE_BITS) 45 #define PTABLE_MASK (PTABLE_BINS-1) 48 #define DOWN 0x00010000 52 #define VALUE_ONE (1 << PRECISION) 53 #define PRECISION_USE 12 57 #define MAX_HISTORY_BITS 5 58 #define MAX_HISTORY_BINS (1 << MAX_HISTORY_BITS) 59 #define MAX_BIN_BYTES 1280 // for value_lookup, per bin (2k - 512 - 256) 98 #define WV_MAX_FRAME_DECODERS 14 119 #define LEVEL_DECAY(a) (((a) + 0x80) >> 8) 128 e = (1 << (p + 1)) - k - 1;
147 int balance = (sl[1] - sl[0] + br[1] + 1) >> 1;
148 if (balance > br[0]) {
151 }
else if (-balance > br[0]) {
155 br[1] = br[0] + balance;
156 br[0] = br[0] - balance;
161 if (sl[i] - br[i] > -0x100)
240 if (ctx->
hybrid && !channel) {
268 if (add >= 0x2000000U) {
276 int mid = (base * 2
U + add + 1) >> 1;
281 add -= (mid - (unsigned)base);
284 add = mid - (unsigned)base - 1;
285 mid = (base * 2
U + add + 1) >> 1;
292 return sign ? ~ret :
ret;
314 *crc = *crc * 9 + (S & 0xffff) * 3 + ((
unsigned)S >> 16);
318 bit = (S & s->
and) | s->
or;
319 bit = ((S + bit) << s->
shift) - bit;
338 const int max_bits = 1 + 23 + 8 + 1;
350 if (S >= 0x1000000U) {
369 S |= (1 <<
shift) - 1;
395 *crc = *crc * 27 + S * 9 + exp * 3 + sign;
397 value.u = (sign << 31) | (exp << 23) |
S;
418 int value = 0x808000, rate = rate_i << 8;
420 for (
int c = (rate + 128) >> 8;
c--;)
427 if (value > 0x010000) {
428 rate += (rate * rate_s + 128) >> 8;
430 for (
int c = (rate + 64) >> 7;
c--;)
444 uint8_t *dst_l = dst_left, *dst_r = dst_right;
448 uint32_t low, high,
value;
453 rate_i = bytestream2_get_byte(&s->
gbyte);
454 rate_s = bytestream2_get_byte(&s->
gbyte);
470 sp->
factor = bytestream2_get_byte(&s->
gbyte) & 0xff;
471 sp->
factor |= (bytestream2_get_byte(&s->
gbyte) << 8) & 0xff00;
475 value = bytestream2_get_be32(&s->
gbyte);
479 while (total_samples--) {
489 uint32_t
split = low + ((high - low) >> 8) * (*pp >> 16);
491 if (value <= split) {
502 value = (value << 8) | bytestream2_get_byte(&s->
gbyte);
503 high = (high << 8) | 0xff;
508 sp[0].
byte = (sp[0].
byte << 1) | (sp[0].fltr0 & 1);
510 ((sp[0].value ^ (sp[0].value - (sp[0].fltr6 * 16))) >> 31);
524 split = low + ((high - low) >> 8) * (*pp >> 16);
526 if (value <= split) {
537 value = (value << 8) | bytestream2_get_byte(&s->
gbyte);
538 high = (high << 8) | 0xff;
543 sp[1].
byte = (sp[1].
byte << 1) | (sp[1].fltr0 & 1);
545 ((sp[1].value ^ (sp[1].value - (sp[1].fltr6 * 16))) >> 31);
556 checksum += (checksum << 1) + (*dst_l = sp[0].
byte & 0xff);
561 checksum += (checksum << 1) + (*dst_r = filters[1].
byte & 0xff);
571 memset(dst_left, 0x69, s->
samples * 4);
574 memset(dst_right, 0x69, s->
samples * 4);
582 uint8_t *dst_l = dst_left, *dst_r = dst_right;
583 uint8_t history_bits, max_probability;
584 int total_summed_probabilities = 0;
585 int total_samples = s->
samples;
587 int history_bins, p0, p1, chan;
589 uint32_t low, high,
value;
594 history_bits = bytestream2_get_byte(&s->
gbyte);
599 history_bins = 1 << history_bits;
600 max_probability = bytestream2_get_byte(&s->
gbyte);
602 if (max_probability < 0xff) {
607 int code = bytestream2_get_byte(&s->
gbyte);
609 if (code > max_probability) {
610 int zcount = code - max_probability;
612 while (outptr < outend && zcount--)
622 if (outptr < outend ||
632 for (p0 = 0; p0 < history_bins; p0++) {
635 for (
int i = 0;
i < 256;
i++)
639 total_summed_probabilities += sum_values;
641 if (total_summed_probabilities > history_bins *
MAX_BIN_BYTES)
646 for (
int i = 0;
i < 256;
i++) {
659 low = 0; high = 0xffffffff;
660 value = bytestream2_get_be32(&s->
gbyte);
665 while (total_samples--) {
675 value = bytestream2_get_be32(&s->
gbyte);
685 index = (value - low) / mult;
712 checksum += (checksum << 1) + code;
715 p0 = code & (history_bins-1);
718 p1 = code & (history_bins-1);
722 value = (value << 8) | bytestream2_get_byte(&s->
gbyte);
723 high = (high << 8) | 0xff;
732 memset(dst_left, 0x69, s->
samples * 4);
735 memset(dst_right, 0x69, s->
samples * 4);
743 uint8_t *dst_l = dst_left, *dst_r = dst_right;
744 int total_samples = s->
samples;
750 while (total_samples--) {
751 checksum += (checksum << 1) + (*dst_l = bytestream2_get_byte(&s->
gbyte));
755 checksum += (checksum << 1) + (*dst_r = bytestream2_get_byte(&s->
gbyte));
764 memset(dst_left, 0x69, s->
samples * 4);
767 memset(dst_right, 0x69, s->
samples * 4);
774 void *dst_l,
void *dst_r,
const int type)
780 uint32_t crc = 0xFFFFFFFF;
782 int16_t *dst16_l = dst_l;
783 int16_t *dst16_r = dst_r;
786 float *dstfl_l = dst_l;
787 float *dstfl_r = dst_r;
797 for (i = 0; i < s->
terms; i++) {
820 L2 = L + (unsigned)((
int)(s->
decorr[
i].
weightA * (unsigned)A + 512) >> 10);
821 R2 = R + (unsigned)((
int)(s->
decorr[
i].
weightB * (unsigned)B + 512) >> 10);
829 }
else if (t == -1) {
839 R2 = R + (unsigned)((
int)(s->
decorr[
i].
weightB * (unsigned)L2 + 512) >> 10);
859 L2 = L + (unsigned)((
int)(s->
decorr[
i].
weightA * (unsigned)R2 + 512) >> 10);
867 if (
FFABS((int64_t)L) +
FFABS((int64_t)R) > (1<<19)) {
875 L += (unsigned)(R -= (
unsigned)(L >> 1));
876 crc = (crc * 3 +
L) * 3 + R;
905 void *dst,
const int type)
911 uint32_t crc = 0xFFFFFFFF;
913 int16_t *dst16 = dst;
923 for (i = 0; i < s->
terms; i++) {
939 S = T + (unsigned)((
int)(s->
decorr[
i].
weightA * (unsigned)A + 512) >> 10);
996 if (channels > INT_MAX /
sizeof(*s->
dsdctx))
1080 const uint8_t *buf,
int buf_size)
1086 void *samples_l =
NULL, *samples_r =
NULL;
1088 int got_terms = 0, got_weights = 0, got_samples = 0,
1089 got_entropy = 0, got_pcm = 0, got_float = 0, got_hybrid = 0;
1092 int bpp, chan = 0, orig_bpp,
sample_rate = 0, rate_x = 1, dsd_mode = 0;
1094 uint64_t chmask = 0;
1101 s = wc->
fdec[block_no];
1109 memset(s->
ch, 0,
sizeof(s->
ch));
1116 s->
samples = bytestream2_get_le32(&gb);
1149 s->
CRC = bytestream2_get_le32(&gb);
1153 id = bytestream2_get_byte(&gb);
1154 size = bytestream2_get_byte(&gb);
1156 size |= (bytestream2_get_le16u(&gb)) << 8;
1163 "Got incorrect block %02X with size %i\n",
id, size);
1168 "Block size %i is out of bounds\n", size);
1180 for (i = 0; i < s->
terms; i++) {
1198 for (i = 0; i <
weights; i++) {
1199 t = (int8_t)bytestream2_get_byte(&gb);
1205 t = (int8_t)bytestream2_get_byte(&gb);
1220 for (i = s->
terms - 1; (i >= 0) && (t < size); i--) {
1223 wp_exp2(bytestream2_get_le16(&gb));
1225 wp_exp2(bytestream2_get_le16(&gb));
1229 wp_exp2(bytestream2_get_le16(&gb));
1231 wp_exp2(bytestream2_get_le16(&gb));
1237 wp_exp2(bytestream2_get_le16(&gb));
1239 wp_exp2(bytestream2_get_le16(&gb));
1244 wp_exp2(bytestream2_get_le16(&gb));
1247 wp_exp2(bytestream2_get_le16(&gb));
1258 "Entropy vars size should be %i, got %i.\n",
1264 for (i = 0; i < 3; i++) {
1276 for (i = 0; i < (s->
stereo_in + 1); i++) {
1281 for (i = 0; i < (s->
stereo_in + 1); i++) {
1283 wp_exp2((int16_t)bytestream2_get_le16(&gb));
1286 for (i = 0; i < (s->
stereo_in + 1); i++)
1295 "Invalid INT32INFO, size = %i\n",
1303 "Invalid INT32INFO, extra_bits = %d (> 30)\n", val[0]);
1305 }
else if (val[0]) {
1307 }
else if (val[1]) {
1309 }
else if (val[2]) {
1312 }
else if (val[3]) {
1316 if (s->
shift > 31) {
1318 "Invalid INT32INFO, shift = %d (> 31)\n", s->
shift);
1335 "Invalid FLOATINFO, size = %i\n", size);
1344 "Invalid FLOATINFO, shift = %d (> 31)\n", s->
float_shift);
1364 rate_x = bytestream2_get_byte(&gb);
1367 rate_x = 1 << rate_x;
1368 dsd_mode = bytestream2_get_byte(&gb);
1369 if (dsd_mode && dsd_mode != 1 && dsd_mode != 3) {
1394 "Insufficient channel information\n");
1397 chan = bytestream2_get_byte(&gb);
1400 chmask = bytestream2_get_byte(&gb);
1403 chmask = bytestream2_get_le16(&gb);
1406 chmask = bytestream2_get_le24(&gb);
1409 chmask = bytestream2_get_le32(&gb);
1412 size = bytestream2_get_byte(&gb);
1413 chan |= (bytestream2_get_byte(&gb) & 0xF) << 8;
1417 " instead of %i.\n", chan, avctx->
channels);
1418 chmask = bytestream2_get_le24(&gb);
1421 size = bytestream2_get_byte(&gb);
1422 chan |= (bytestream2_get_byte(&gb) & 0xF) << 8;
1426 " instead of %i.\n", chan, avctx->
channels);
1427 chmask = bytestream2_get_le32(&gb);
1441 sample_rate = bytestream2_get_le24(&gb);
1446 if (
id & WP_IDF_ODD)
1467 if (s->
hybrid && !got_hybrid) {
1478 if (size < wanted) {
1485 if (!got_pcm && !got_dsd) {
1497 int new_channels = avctx->
channels;
1510 if (new_samplerate * (uint64_t)rate_x > INT_MAX)
1512 new_samplerate *= rate_x;
1516 new_channels = chan;
1518 new_chmask = chmask;
1520 new_channels = s->
stereo ? 2 : 1;
1535 !!got_dsd != !!wc->
dsdctx) {
1574 if (dsd_mode == 3) {
1576 }
else if (dsd_mode == 1) {
1588 if (dsd_mode == 3) {
1590 }
else if (dsd_mode == 1) {
1602 memcpy(samples_r, samples_l, bpp * s->
samples);
1628 int *got_frame_ptr,
AVPacket *avpkt)
1632 int buf_size = avpkt->
size;
1644 frame_flags =
AV_RL32(buf + 24);
1654 frame_size =
AV_RL32(buf + 4) - 12;
1657 if (frame_size <= 0 || frame_size > buf_size) {
1659 "Block %d has invalid size (size %d vs. %d bytes left)\n",
1660 s->
block, frame_size, buf_size);
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
void av_buffer_unref(AVBufferRef **buf)
Free a given reference and automatically free the buffer if there are no more references to it...
This structure describes decoded (raw) audio or video data.
ptrdiff_t const GLvoid * data
static void flush(AVCodecContext *avctx)
static void wavpack_decode_flush(AVCodecContext *avctx)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
#define AV_LOG_WARNING
Something somehow does not look correct.
static int wv_dsd_reset(WavpackContext *s, int channels)
static av_cold int init(AVCodecContext *avctx)
static int wavpack_decode_block(AVCodecContext *avctx, int block_no, const uint8_t *buf, int buf_size)
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
#define WV_FLT_SHIFT_ONES
static int dsd_channel(AVCodecContext *avctx, void *frmptr, int jobnr, int threadnr)
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
#define WV_HYBRID_BITRATE
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
#define AV_CH_LAYOUT_STEREO
static void error(const char *err)
av_cold void ff_init_dsd_data(void)
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before ff_thread_await_progress() has been called on them.reget_buffer() and buffer age optimizations no longer work.*The contents of buffers must not be written to after ff_thread_report_progress() has been called on them.This includes draw_edges().Porting codecs to frame threading
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
static int get_unary_0_33(GetBitContext *gb)
Get unary code terminated by a 0 with a maximum length of 33.
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
enum AVSampleFormat sample_fmt
audio sample format
static int wv_unpack_dsd_high(WavpackFrameContext *s, uint8_t *dst_left, uint8_t *dst_right)
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
AVCodec ff_wavpack_decoder
void ff_dsd2pcm_translate(DSDContext *s, size_t samples, int lsbf, const uint8_t *src, ptrdiff_t src_stride, float *dst, ptrdiff_t dst_stride)
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Multithreading support functions.
GLsizei GLboolean const GLfloat * value
int av_frame_ref(AVFrame *dst, const AVFrame *src)
Set up a new reference to the data described by the source frame.
#define u(width, name, range_min, range_max)
bitstream reader API header.
static av_always_inline int wp_log2(uint32_t val)
static const uint16_t table[]
#define FF_CODEC_CAP_ALLOCATE_PROGRESS
WavpackFrameContext * fdec[WV_MAX_FRAME_DECODERS]
static av_always_inline int bytestream2_get_bytes_left(GetByteContext *g)
uint8_t probabilities[MAX_HISTORY_BINS][256]
static int wv_unpack_dsd_copy(WavpackFrameContext *s, uint8_t *dst_left, uint8_t *dst_right)
static int get_bits_left(GetBitContext *gb)
static int update_error_limit(WavpackFrameContext *ctx)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void ff_thread_release_buffer(AVCodecContext *avctx, ThreadFrame *f)
Wrapper around release_buffer() frame-for multithreaded codecs.
static const int weights[]
#define DSD_BYTE_READY(low, high)
static av_always_inline void bytestream2_skip(GetByteContext *g, unsigned int size)
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
static av_always_inline int wp_exp2(int16_t val)
static av_always_inline unsigned int bytestream2_get_buffer(GetByteContext *g, uint8_t *dst, unsigned int size)
uint8_t * value_lookup[MAX_HISTORY_BINS]
const char * name
Name of the codec implementation.
#define AV_CODEC_CAP_FRAME_THREADS
Codec supports frame-level multithreading.
uint64_t channel_layout
Audio channel layout.
static char * split(char *message, char delim)
#define ONLY_IF_THREADS_ENABLED(x)
Define a function with only the non-default version specified.
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before as well as code calling up to before the decode process starts Call have so the codec calls ff_thread_report set FF_CODEC_CAP_ALLOCATE_PROGRESS in AVCodec caps_internal and use ff_thread_get_buffer() to allocate frames.The frames must then be freed with ff_thread_release_buffer().Otherwise decode directly into the user-supplied frames.Call ff_thread_report_progress() after some part of the current picture has decoded.A good place to put this is where draw_horiz_band() is called-add this if it isn't called anywhere
audio channel layout utility functions
static int16_t mult(Float11 *f1, Float11 *f2)
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
static int wv_unpack_stereo(WavpackFrameContext *s, GetBitContext *gb, void *dst_l, void *dst_r, const int type)
static const int wv_rates[16]
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before as well as code calling up to before the decode process starts Call ff_thread_finish_setup() afterwards.If some code can't be moved
static int wv_unpack_mono(WavpackFrameContext *s, GetBitContext *gb, void *dst, const int type)
void ff_thread_report_progress(ThreadFrame *f, int n, int field)
Notify later decoding threads when part of their reference picture is ready.
#define WV_FLT_SHIFT_SAME
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
#define AV_EF_EXPLODE
abort decoding on minor error detection
#define WV_FLT_SHIFT_SENT
static volatile int checksum
int ff_thread_ref_frame(ThreadFrame *dst, const ThreadFrame *src)
#define AV_CODEC_CAP_SLICE_THREADS
Codec supports slice-based (or partition-based) multithreading.
int(* execute2)(struct AVCodecContext *c, int(*func)(struct AVCodecContext *c2, void *arg, int jobnr, int threadnr), void *arg2, int *ret, int count)
The codec may call this to execute several independent things.
static av_cold int wavpack_decode_end(AVCodecContext *avctx)
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_WB16 unsigned int_TMPL byte
int sample_rate
samples per second
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
main external API structure.
F H1 F F H1 F F F F H1<-F-------F-------F v v v H2 H3 H2^^^F-------F-------F-> H1<-F-------F-------F|||||||||F H1 F|||||||||F H1 Funavailable fullpel samples(outside the picture for example) shall be equalto the closest available fullpel sampleSmaller pel interpolation:--------------------------if diag_mc is set then points which lie on a line between 2 vertically, horizontally or diagonally adjacent halfpel points shall be interpolatedlinearly with rounding to nearest and halfway values rounded up.points which lie on 2 diagonals at the same time should only use the onediagonal not containing the fullpel point F--> O q O<--h1-> O q O<--F v\/v\/v O O O O O O O|/|\|q q q q q|/|\|O O O O O O O^/\^/\^h2--> O q O<--h3-> O q O<--h2 v\/v\/v O O O O O O O|\|/|q q q q q|\|/|O O O O O O O^/\^/\^F--> O q O<--h1-> O q O<--Fthe remaining points shall be bilinearly interpolated from theup to 4 surrounding halfpel and fullpel points, again rounding should be tonearest and halfway values rounded upcompliant Snow decoders MUST support 1-1/8 pel luma and 1/2-1/16 pel chromainterpolation at leastOverlapped block motion compensation:-------------------------------------FIXMELL band prediction:===================Each sample in the LL0 subband is predicted by the median of the left, top andleft+top-topleft samples, samples outside the subband shall be considered tobe 0.To reverse this prediction in the decoder apply the following.for(y=0;y< height;y++){for(x=0;x< width;x++){sample[y][x]+=median(sample[y-1][x], sample[y][x-1], sample[y-1][x]+sample[y][x-1]-sample[y-1][x-1]);}}sample[-1][*]=sample[*][-1]=0;width, height here are the width and height of the LL0 subband not of the finalvideoDequantization:===============FIXMEWavelet Transform:==================Snow supports 2 wavelet transforms, the symmetric biorthogonal 5/3 integertransform and an integer approximation of the symmetric biorthogonal 9/7daubechies wavelet.2D IDWT(inverse discrete wavelet transform)--------------------------------------------The 2D IDWT applies a 2D filter recursively, each time combining the4 lowest frequency subbands into a single subband until only 1 subbandremains.The 2D filter is done by first applying a 1D filter in the vertical directionand then applying it in the horizontal one.------------------------------------------------------------|LL0|HL0|||||||||||||---+---|HL1||L0|H0|HL1||LL1|HL1|||||LH0|HH0|||||||||||||-------+-------|-> L1 H1 LH1 HH1 LH1 HH1 LH1 HH1 L2
uint8_t * data
The data buffer.
static float wv_get_value_float(WavpackFrameContext *s, uint32_t *crc, int S)
static int wv_get_value_integer(WavpackFrameContext *s, uint32_t *crc, unsigned S)
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before as well as code calling up to before the decode process starts Call have update_thread_context() run it in the next thread.Add AV_CODEC_CAP_FRAME_THREADS to the codec capabilities.There will be very little speed gain at this point but it should work.If there are inter-frame dependencies
AVBufferRef * av_buffer_allocz(int size)
Same as av_buffer_alloc(), except the returned buffer will be initialized to zero.
static unsigned int get_bits1(GetBitContext *s)
static void init_ptable(int *table, int rate_i, int rate_s)
#define AV_EF_CRCCHECK
Verify checksums embedded in the bitstream (could be of either encoded or decoded data...
refcounted data buffer API
static const int factor[16]
static int wavpack_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
static int wv_check_crc(WavpackFrameContext *s, uint32_t crc, uint32_t crc_extra_bits)
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
static int wv_get_value(WavpackFrameContext *ctx, GetBitContext *gb, int channel, int *last)
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some it can consider them to be part of the FIFO and delay acknowledging a status change accordingly Example code
#define UPDATE_WEIGHT_CLIP(weight, delta, samples, in)
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
static av_always_inline unsigned get_tail(GetBitContext *gb, int k)
A reference to a data buffer.
static int wv_unpack_dsd_fast(WavpackFrameContext *s, uint8_t *dst_left, uint8_t *dst_right)
common internal api header.
#define bit(string, value)
channel
Use these values when setting the channel map with ebur128_set_channel().
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
GetBitContext gb_extra_bits
#define xf(width, name, var, range_min, range_max, subs,...)
int channels
number of audio channels
static const struct PPFilter filters[]
uint16_t summed_probabilities[MAX_HISTORY_BINS][256]
static float add(float src0, float src1)
static av_cold int wv_alloc_frame_context(WavpackContext *c)
uint8_t value_lookup_buffer[MAX_HISTORY_BINS *MAX_BIN_BYTES]
static av_cold int wavpack_decode_init(AVCodecContext *avctx)
#define FFSWAP(type, a, b)
#define WV_MAX_FRAME_DECODERS
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
uint8_t ** extended_data
pointers to the data planes/channels.
#define AV_CH_LAYOUT_MONO
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_RL32
static double val(void *priv, double ch)
This structure stores compressed data.
#define AV_GET_BUFFER_FLAG_REF
The decoder will keep a reference to the frame and may reuse it later.
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 av_always_inline int get_bitsz(GetBitContext *s, int n)
Read 0-25 bits.
int av_buffer_replace(AVBufferRef **pdst, AVBufferRef *src)
Ensure dst refers to the same data as src.