Go to the documentation of this file.
34 #define ATRAC9_SF_VLC_BITS 8
35 #define ATRAC9_COEFF_VLC_BITS 9
128 grad_range[1] =
get_bits(gb, 6) + 1;
134 if (grad_range[0] >= grad_range[1] || grad_range[1] > 31)
137 if (
b->grad_boundary >
b->q_unit_cnt)
140 values = grad_value[1] - grad_value[0];
141 sign = 1 - 2*(
values < 0);
142 base = grad_value[0] + sign;
144 curve =
s->alloc_curve[grad_range[1] - grad_range[0] - 1];
146 for (
int i = 0;
i <=
b->q_unit_cnt;
i++)
147 b->gradient[
i] = grad_value[
i >= grad_range[0]];
149 for (
int i = grad_range[0];
i < grad_range[1];
i++)
150 b->gradient[
i] =
base + sign*((
int)(
scale*curve[
i - grad_range[0]]));
158 memset(
c->precision_mask, 0,
sizeof(
c->precision_mask));
159 for (
int i = 1;
i <
b->q_unit_cnt;
i++) {
160 const int delta =
FFABS(
c->scalefactors[
i] -
c->scalefactors[
i - 1]) - 1;
162 const int neg =
c->scalefactors[
i - 1] >
c->scalefactors[
i];
168 for (
int i = 0;
i <
b->q_unit_cnt;
i++) {
169 c->precision_coarse[
i] =
c->scalefactors[
i];
170 c->precision_coarse[
i] +=
c->precision_mask[
i] -
b->gradient[
i];
171 if (
c->precision_coarse[
i] < 0)
173 switch (
b->grad_mode) {
175 c->precision_coarse[
i] >>= 1;
178 c->precision_coarse[
i] = (3 *
c->precision_coarse[
i]) >> 3;
181 c->precision_coarse[
i] >>= 2;
186 for (
int i = 0;
i <
b->q_unit_cnt;
i++)
187 c->precision_coarse[
i] =
c->scalefactors[
i] -
b->gradient[
i];
191 for (
int i = 0;
i <
b->q_unit_cnt;
i++)
192 c->precision_coarse[
i] =
FFMAX(
c->precision_coarse[
i], 1);
194 for (
int i = 0;
i <
b->grad_boundary;
i++)
195 c->precision_coarse[
i]++;
197 for (
int i = 0;
i <
b->q_unit_cnt;
i++) {
198 c->precision_fine[
i] = 0;
199 if (
c->precision_coarse[
i] > 15) {
200 c->precision_fine[
i] =
FFMIN(
c->precision_coarse[
i], 30) - 15;
201 c->precision_coarse[
i] = 15;
211 if (
b->has_band_ext) {
212 if (
b->q_unit_cnt < 13 ||
b->q_unit_cnt > 20)
216 b->channel[1].band_ext =
get_bits(gb, 2);
217 b->channel[1].band_ext = ext_band > 2 ?
b->channel[1].band_ext : 4;
224 if (!
b->has_band_ext_data)
227 if (!
b->has_band_ext) {
233 b->channel[0].band_ext =
get_bits(gb, 2);
234 b->channel[0].band_ext = ext_band > 2 ?
b->channel[0].band_ext : 4;
237 for (
int i = 0;
i <= stereo;
i++) {
240 for (
int j = 0; j < count; j++) {
249 for (
int i = 0;
i <= stereo;
i++) {
252 for (
int j = 0; j < count; j++) {
263 int channel_idx,
int first_in_pkt)
265 static const uint8_t mode_map[2][4] = { { 0, 1, 2, 3 }, { 0, 2, 3, 4 } };
266 const int mode = mode_map[channel_idx][
get_bits(gb, 2)];
268 memset(
c->scalefactors, 0,
sizeof(
c->scalefactors));
270 if (first_in_pkt && (
mode == 4 || ((
mode == 3) && !channel_idx))) {
284 for (
int i = 1;
i <
b->band_ext_q_unit;
i++) {
287 c->scalefactors[
i] =
val & ((1 <<
len) - 1);
290 for (
int i = 0;
i <
b->band_ext_q_unit;
i++)
291 c->scalefactors[
i] +=
base - sf_weights[
i];
298 for (
int i = 0;
i <
b->band_ext_q_unit;
i++)
304 const int *baseline =
mode == 4 ?
c->scalefactors_prev :
305 channel_idx ?
b->channel[0].scalefactors :
306 c->scalefactors_prev;
307 const int baseline_len =
mode == 4 ?
b->q_unit_cnt_prev :
308 channel_idx ?
b->band_ext_q_unit :
312 const int unit_cnt =
FFMIN(
b->band_ext_q_unit, baseline_len);
315 for (
int i = 0;
i < unit_cnt;
i++) {
317 c->scalefactors[
i] = baseline[
i] + dist;
320 for (
int i = unit_cnt;
i <
b->band_ext_q_unit;
i++)
326 const int *baseline = channel_idx ?
b->channel[0].scalefactors :
327 c->scalefactors_prev;
328 const int baseline_len = channel_idx ?
b->band_ext_q_unit :
333 const int unit_cnt =
FFMIN(
b->band_ext_q_unit, baseline_len);
338 for (
int i = 1;
i < unit_cnt;
i++) {
341 c->scalefactors[
i] =
val & ((1 <<
len) - 1);
344 for (
int i = 0;
i < unit_cnt;
i++)
345 c->scalefactors[
i] +=
base + baseline[
i];
347 for (
int i = unit_cnt;
i <
b->band_ext_q_unit;
i++)
353 for (
int i = 0;
i <
b->band_ext_q_unit;
i++)
354 if (
c->scalefactors[
i] < 0 ||
c->scalefactors[
i] > 31)
357 memcpy(
c->scalefactors_prev,
c->scalefactors,
sizeof(
c->scalefactors));
366 const int last_sf =
c->scalefactors[
c->q_unit_cnt];
368 memset(
c->codebookset, 0,
sizeof(
c->codebookset));
370 if (
c->q_unit_cnt <= 1)
372 if (
s->samplerate_idx > 7)
375 c->scalefactors[
c->q_unit_cnt] =
c->scalefactors[
c->q_unit_cnt - 1];
377 if (
c->q_unit_cnt > 12) {
378 for (
int i = 0;
i < 12;
i++)
379 avg +=
c->scalefactors[
i];
383 for (
int i = 8;
i <
c->q_unit_cnt;
i++) {
384 const int prev =
c->scalefactors[
i - 1];
385 const int cur =
c->scalefactors[
i ];
386 const int next =
c->scalefactors[
i + 1];
388 if ((cur -
min >= 3 || 2*cur - prev - next >= 3))
389 c->codebookset[
i] = 1;
393 for (
int i = 12;
i <
c->q_unit_cnt;
i++) {
394 const int cur =
c->scalefactors[
i];
396 const int min =
FFMIN(
c->scalefactors[
i + 1],
c->scalefactors[
i - 1]);
397 if (
c->codebookset[
i])
400 c->codebookset[
i] = (((cur -
min) >= 2) && (cur >= (
avg - cnd)));
403 c->scalefactors[
c->q_unit_cnt] = last_sf;
409 const int max_prec =
s->samplerate_idx > 7 ? 1 : 7;
411 memset(
c->q_coeffs_coarse, 0,
sizeof(
c->q_coeffs_coarse));
413 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
416 const int prec =
c->precision_coarse[
i] + 1;
418 if (prec <= max_prec) {
419 const int cb =
c->codebookset[
i];
425 for (
int j = 0; j < groups; j++) {
428 for (
int k = 0; k < huff->
value_cnt; k++) {
436 for (
int j = 0; j <
bands; j++)
445 memset(
c->q_coeffs_fine, 0,
sizeof(
c->q_coeffs_fine));
447 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
450 const int len =
c->precision_fine[
i] + 1;
452 if (
c->precision_fine[
i] <= 0)
455 for (
int j = start; j < end; j++)
463 memset(
c->coeffs, 0,
sizeof(
c->coeffs));
465 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
472 for (
int j = start; j < end; j++) {
473 const float vc =
c->q_coeffs_coarse[j] * coarse_c;
474 const float vf =
c->q_coeffs_fine[j] * fine_c;
475 c->coeffs[j] = vc + vf;
483 float *
src =
b->channel[
b->cpe_base_channel].coeffs;
484 float *dst =
b->channel[!
b->cpe_base_channel].coeffs;
489 if (
b->q_unit_cnt <=
b->stereo_q_unit)
492 for (
int i =
b->stereo_q_unit; i < b->q_unit_cnt;
i++) {
493 const int sign =
b->is_signs[
i];
496 for (
int j = start; j < end; j++)
497 dst[j] = sign*
src[j];
504 for (
int i = 0;
i <= stereo;
i++) {
505 float *coeffs =
b->channel[
i].coeffs;
506 for (
int j = 0; j <
b->q_unit_cnt; j++) {
509 const int scalefactor =
b->channel[
i].scalefactors[j];
511 for (
int k = start; k < end; k++)
518 int start,
int count)
521 for (
int i = 0;
i < count;
i += 2) {
524 c->coeffs[start +
i + 0] =
tmp[0];
525 c->coeffs[start +
i + 1] =
tmp[1];
529 for (
int i = 0;
i < count;
i++)
530 c->coeffs[start +
i] /= maxval;
534 const int s_unit,
const int e_unit)
536 for (
int i = s_unit;
i < e_unit;
i++) {
539 for (
int j = start; j < end; j++)
540 c->coeffs[j] *= sf[
i - s_unit];
547 const int g_units[4] = {
551 FFMAX(g_units[2], 22),
554 const int g_bins[4] = {
561 for (
int ch = 0; ch <= stereo; ch++) {
565 for (
int i = 0;
i < 3;
i++)
566 for (
int j = 0; j < (g_bins[
i + 1] - g_bins[
i + 0]); j++)
567 c->coeffs[g_bins[
i] + j] =
c->coeffs[g_bins[
i] - j - 1];
569 switch (
c->band_ext) {
571 float sf[6] = { 0.0f };
572 const int l = g_units[3] - g_units[0] - 1;
605 for (
int i = g_units[0];
i < g_units[3];
i++)
613 const float g_sf[2] = {
618 for (
int i = 0;
i < 2;
i++)
619 for (
int j = g_bins[
i + 0]; j < g_bins[
i + 1]; j++)
620 c->coeffs[j] *= g_sf[
i];
627 for (
int i = g_bins[0];
i < g_bins[3];
i++) {
635 const float g_sf[3] = { 0.7079468f*m, 0.5011902f*m, 0.3548279f*m };
637 for (
int i = 0;
i < 3;
i++)
638 for (
int j = g_bins[
i + 0]; j < g_bins[
i + 1]; j++)
639 c->coeffs[j] *= g_sf[
i];
648 int frame_idx,
int block_idx)
656 const int precision = reuse_params ? 8 : 4;
657 c->q_unit_cnt =
b->q_unit_cnt = 2;
659 memset(
c->scalefactors, 0,
sizeof(
c->scalefactors));
660 memset(
c->q_coeffs_fine, 0,
sizeof(
c->q_coeffs_fine));
661 memset(
c->q_coeffs_coarse, 0,
sizeof(
c->q_coeffs_coarse));
663 for (
int i = 0;
i <
b->q_unit_cnt;
i++) {
665 c->precision_coarse[
i] = precision;
666 c->precision_fine[
i] = 0;
669 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
672 for (
int j = start; j < end; j++)
673 c->q_coeffs_coarse[j] =
get_bits(gb,
c->precision_coarse[
i] + 1);
682 if (first_in_pkt && reuse_params) {
689 int stereo_band, ext_band;
690 const int min_band_count =
s->samplerate_idx > 7 ? 1 : 3;
692 b->band_count =
get_bits(gb, 4) + min_band_count;
695 b->band_ext_q_unit =
b->stereo_q_unit =
b->q_unit_cnt;
704 stereo_band =
get_bits(gb, 4) + min_band_count;
705 if (stereo_band >
b->band_count) {
714 if (
b->has_band_ext) {
715 ext_band =
get_bits(gb, 4) + min_band_count;
716 if (ext_band < b->band_count) {
735 b->cpe_base_channel = 0;
739 for (
int i =
b->stereo_q_unit; i < b->q_unit_cnt;
i++)
752 for (
int i = 0;
i <= stereo;
i++) {
754 c->q_unit_cnt =
i ==
b->cpe_base_channel ?
b->q_unit_cnt :
766 b->q_unit_cnt_prev =
b->has_band_ext ?
b->band_ext_q_unit :
b->q_unit_cnt;
771 if (
b->has_band_ext &&
b->has_band_ext_data)
775 for (
int i = 0;
i <= stereo;
i++) {
777 const int dst_idx =
s->block_config->plane_map[block_idx][
i];
778 const int wsize = 1 <<
s->frame_log2;
779 const ptrdiff_t
offset = wsize*frame_idx*
sizeof(
float);
782 s->tx_fn(
s->tx,
s->temp,
c->coeffs,
sizeof(
float));
783 s->fdsp->vector_fmul_window(dst,
c->prev_win,
s->temp,
784 s->imdct_win, wsize >> 1);
785 memcpy(
c->prev_win,
s->temp + (wsize >> 1),
sizeof(
float)*wsize >> 1);
792 int *got_frame_ptr,
AVPacket *avpkt)
809 for (
int j = 0; j <
s->block_config->count; j++) {
826 for (
int j = 0; j <
s->block_config->count; j++) {
829 for (
int i = 0;
i <= stereo;
i++) {
831 memset(
c->prev_win, 0,
sizeof(
c->prev_win));
847 int nb_bits,
int nb_codes,
862 const uint8_t (*
tab)[2];
866 for (
int i = 1;
i < 7;
i++) {
875 for (
int i = 2;
i < 6;
i++) {
887 for (
int i = 0;
i < 2;
i++) {
888 for (
int j = 2; j < 8; j++) {
889 for (
int k =
i; k < 4; k++) {
904 int err,
version, block_config_idx, superframe_idx, alloc_c_len;
938 block_config_idx =
get_bits(&gb, 3);
939 if (block_config_idx > 5) {
946 avctx->
ch_layout =
s->block_config->channel_layout;
955 s->avg_frame_size =
get_bits(&gb, 11) + 1;
958 if (superframe_idx & 1) {
963 s->frame_count = 1 << superframe_idx;
966 scale = 1.0f / 32768.0;
968 1 <<
s->frame_log2, &
scale, 0);
977 for (
int i = 0;
i < (1 <<
s->frame_log2);
i++) {
978 const int len = 1 <<
s->frame_log2;
979 const float sidx = (
i + 0.5f) /
len;
980 const float eidx = (
len -
i - 0.5f) /
len;
983 s->imdct_win[
i] = s_c / ((s_c * s_c) + (e_c * e_c));
988 for (
int i = 1;
i <= alloc_c_len;
i++)
989 for (
int j = 0; j <
i; j++)
1009 #if FF_API_SUBFRAMES
1010 AV_CODEC_CAP_SUBFRAMES |
static av_cold int atrac9_decode_close(AVCodecContext *avctx)
int32_t q_coeffs_coarse[256]
@ AV_SAMPLE_FMT_FLTP
float, planar
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
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
int sample_rate
samples per second
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
static double cb(void *priv, double x, double y)
static const float at9_band_ext_scales_m2[]
static int read_scalefactors(ATRAC9Context *s, ATRAC9BlockData *b, ATRAC9ChannelData *c, GetBitContext *gb, int channel_idx, int first_in_pkt)
This structure describes decoded (raw) audio or video data.
static const av_cold VLCElem * atrac9_init_vlc(VLCInitState *state, int nb_bits, int nb_codes, const uint8_t(**tab)[2], int offset)
static const int at9_tab_samplerates[]
static av_always_inline av_const unsigned av_clip_uintp2_c(int a, int p)
Clip a signed integer to an unsigned power of two range.
static void calc_codebook_idx(ATRAC9Context *s, ATRAC9BlockData *b, ATRAC9ChannelData *c)
static const uint16_t table[]
#define ATRAC9_COEFF_VLC_BITS
static const ATRAC9BlockConfig at9_block_layout[]
av_cold int av_tx_init(AVTXContext **ctx, av_tx_fn *tx, enum AVTXType type, int inv, int len, const void *scale, uint64_t flags)
Initialize a transform context with the given configuration (i)MDCTs with an odd length are currently...
static void read_coeffs_fine(ATRAC9Context *s, ATRAC9BlockData *b, ATRAC9ChannelData *c, GetBitContext *gb)
static void skip_bits(GetBitContext *s, int n)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
static const uint8_t at9_tab_band_ext_cnt[][6]
AVCodec p
The public AVCodec.
static void calc_precision(ATRAC9Context *s, ATRAC9BlockData *b, ATRAC9ChannelData *c)
AVChannelLayout ch_layout
Audio channel layout.
if it could not because there are no more frames
static const struct twinvq_data tab
int flags
AV_CODEC_FLAG_*.
static double val(void *priv, double ch)
uint8_t alloc_curve[48][48]
static void scale_band_ext_coeffs(ATRAC9ChannelData *c, float sf[6], const int s_unit, const int e_unit)
static int ff_thread_once(char *control, void(*routine)(void))
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define FF_ARRAY_ELEMS(a)
static int parse_band_ext(ATRAC9Context *s, ATRAC9BlockData *b, GetBitContext *gb, int stereo)
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
void(* av_tx_fn)(AVTXContext *s, void *out, void *in, ptrdiff_t stride)
Function pointer to a function to perform the transform.
For static VLCs, the number of bits can often be hardcoded at each get_vlc2() callsite.
static const uint8_t at9_tab_sri_max_bands[]
@ AV_TX_FLOAT_MDCT
Standard MDCT with a sample data type of float, double or int32_t, respecively.
#define FF_CODEC_DECODE_CB(func)
static const uint8_t at9_q_unit_to_codebookidx[]
static void fill_with_noise(ATRAC9Context *s, ATRAC9ChannelData *c, int start, int count)
void av_bmg_get(AVLFG *lfg, double out[2])
Get the next two numbers generated by a Box-Muller Gaussian generator using the random numbers issued...
static const float bands[]
static const float at9_band_ext_scales_m0[][5][32]
#define CODEC_LONG_NAME(str)
static const uint8_t at9_sfb_a_tab[][2]
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
#define ATRAC9_SF_VLC_BITS
static const HuffmanCodebook at9_huffman_sf_unsigned[]
static unsigned int get_bits1(GetBitContext *s)
static av_cold int atrac9_decode_init(AVCodecContext *avctx)
int32_t q_coeffs_fine[256]
static int parse_gradient(ATRAC9Context *s, ATRAC9BlockData *b, GetBitContext *gb)
static const VLCElem * sf_vlc[2][8]
static av_always_inline int get_vlc2(GetBitContext *s, const VLCElem *table, int bits, int max_depth)
Parse a vlc code.
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
static av_cold void atrac9_init_static(void)
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
Context structure for the Lagged Fibonacci PRNG.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
int(* init)(AVBSFContext *ctx)
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
static void scale(int *out, const int *in, const int w, const int h, const int shift)
#define DECLARE_ALIGNED(n, t, v)
static void apply_band_extension(ATRAC9Context *s, ATRAC9BlockData *b, const int stereo)
enum AVSampleFormat sample_fmt
audio sample format
static void dequantize(ATRAC9Context *s, ATRAC9BlockData *b, ATRAC9ChannelData *c)
static const uint8_t at9_tab_band_q_unit_map[]
static const HuffmanCodebook at9_huffman_sf_signed[]
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
static void skip_bits1(GetBitContext *s)
static const int at9_q_unit_to_coeff_idx[]
av_cold void av_tx_uninit(AVTXContext **ctx)
Frees a context and sets *ctx to NULL, does nothing when *ctx == NULL.
static const float at9_quant_step_coarse[]
int32_t scalefactors_prev[31]
const ATRAC9BlockConfig * block_config
int nb_samples
number of audio samples (per channel) described by this frame
#define i(width, name, range_min, range_max)
static const uint8_t at9_tab_band_ext_lengths[][6][4]
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
static int atrac9_decode_block(ATRAC9Context *s, GetBitContext *gb, ATRAC9BlockData *b, AVFrame *frame, int frame_idx, int block_idx)
uint8_t ** extended_data
pointers to the data planes/channels.
static const float at9_band_ext_scales_m3[][2]
static const float at9_scalefactor_c[]
static const float at9_band_ext_scales_m4[]
const char * name
Name of the codec implementation.
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
static const uint8_t * align_get_bits(GetBitContext *s)
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_RL32
static const uint8_t at9_coeffs_tab[][2]
static const VLCElem * coeff_vlc[2][8][4]
main external API structure.
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
static const float at9_quant_step_fine[]
static av_const int sign_extend(int val, unsigned bits)
static void atrac9_decode_flush(AVCodecContext *avctx)
static void apply_scalefactors(ATRAC9Context *s, ATRAC9BlockData *b, const int stereo)
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 values
static int atrac9_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
static void read_coeffs_coarse(ATRAC9Context *s, ATRAC9BlockData *b, ATRAC9ChannelData *c, GetBitContext *gb)
static const uint8_t at9_tab_sf_weights[][32]
const av_cold VLCElem * ff_vlc_init_tables_from_lengths(VLCInitState *state, int nb_bits, int nb_codes, const int8_t *lens, int lens_wrap, const void *symbols, int symbols_wrap, int symbols_size, int offset, int flags)
static const uint8_t at9_tab_band_ext_group[][3]
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
#define VLC_INIT_STATE(_table)
This structure stores compressed data.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
static const uint8_t at9_sfb_b_tab[][2]
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
static const uint8_t at9_tab_b_dist[]
const FFCodec ff_atrac9_decoder
static const HuffmanCodebook at9_huffman_coeffs[][8][4]
static const uint8_t at9_tab_sri_frame_log2[]
static void apply_intensity_stereo(ATRAC9Context *s, ATRAC9BlockData *b, const int stereo)
static const uint8_t at9_q_unit_to_coeff_cnt[]