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34 #define ATRAC9_SF_VLC_BITS 8
35 #define ATRAC9_COEFF_VLC_BITS 9
127 grad_range[1] =
get_bits(gb, 6) + 1;
133 if (grad_range[0] >= grad_range[1] || grad_range[1] > 31)
136 if (
b->grad_boundary >
b->q_unit_cnt)
139 values = grad_value[1] - grad_value[0];
140 sign = 1 - 2*(
values < 0);
141 base = grad_value[0] + sign;
143 curve =
s->alloc_curve[grad_range[1] - grad_range[0] - 1];
145 for (
int i = 0;
i <=
b->q_unit_cnt;
i++)
146 b->gradient[
i] = grad_value[
i >= grad_range[0]];
148 for (
int i = grad_range[0];
i < grad_range[1];
i++)
149 b->gradient[
i] =
base + sign*((
int)(
scale*curve[
i - grad_range[0]]));
157 memset(
c->precision_mask, 0,
sizeof(
c->precision_mask));
158 for (
int i = 1;
i <
b->q_unit_cnt;
i++) {
159 const int delta =
FFABS(
c->scalefactors[
i] -
c->scalefactors[
i - 1]) - 1;
161 const int neg =
c->scalefactors[
i - 1] >
c->scalefactors[
i];
167 for (
int i = 0;
i <
b->q_unit_cnt;
i++) {
168 c->precision_coarse[
i] =
c->scalefactors[
i];
169 c->precision_coarse[
i] +=
c->precision_mask[
i] -
b->gradient[
i];
170 if (
c->precision_coarse[
i] < 0)
172 switch (
b->grad_mode) {
174 c->precision_coarse[
i] >>= 1;
177 c->precision_coarse[
i] = (3 *
c->precision_coarse[
i]) >> 3;
180 c->precision_coarse[
i] >>= 2;
185 for (
int i = 0;
i <
b->q_unit_cnt;
i++)
186 c->precision_coarse[
i] =
c->scalefactors[
i] -
b->gradient[
i];
190 for (
int i = 0;
i <
b->q_unit_cnt;
i++)
191 c->precision_coarse[
i] =
FFMAX(
c->precision_coarse[
i], 1);
193 for (
int i = 0;
i <
b->grad_boundary;
i++)
194 c->precision_coarse[
i]++;
196 for (
int i = 0;
i <
b->q_unit_cnt;
i++) {
197 c->precision_fine[
i] = 0;
198 if (
c->precision_coarse[
i] > 15) {
199 c->precision_fine[
i] =
FFMIN(
c->precision_coarse[
i], 30) - 15;
200 c->precision_coarse[
i] = 15;
210 if (
b->has_band_ext) {
211 if (
b->q_unit_cnt < 13 ||
b->q_unit_cnt > 20)
215 b->channel[1].band_ext =
get_bits(gb, 2);
216 b->channel[1].band_ext = ext_band > 2 ?
b->channel[1].band_ext : 4;
223 if (!
b->has_band_ext_data)
226 if (!
b->has_band_ext) {
232 b->channel[0].band_ext =
get_bits(gb, 2);
233 b->channel[0].band_ext = ext_band > 2 ?
b->channel[0].band_ext : 4;
236 for (
int i = 0;
i <= stereo;
i++) {
239 for (
int j = 0; j < count; j++) {
248 for (
int i = 0;
i <= stereo;
i++) {
251 for (
int j = 0; j < count; j++) {
262 int channel_idx,
int first_in_pkt)
264 static const uint8_t mode_map[2][4] = { { 0, 1, 2, 3 }, { 0, 2, 3, 4 } };
265 const int mode = mode_map[channel_idx][
get_bits(gb, 2)];
267 memset(
c->scalefactors, 0,
sizeof(
c->scalefactors));
269 if (first_in_pkt && (
mode == 4 || ((
mode == 3) && !channel_idx))) {
283 for (
int i = 1;
i <
b->band_ext_q_unit;
i++) {
286 c->scalefactors[
i] =
val & ((1 <<
len) - 1);
289 for (
int i = 0;
i <
b->band_ext_q_unit;
i++)
290 c->scalefactors[
i] +=
base - sf_weights[
i];
297 for (
int i = 0;
i <
b->band_ext_q_unit;
i++)
303 const int *baseline =
mode == 4 ?
c->scalefactors_prev :
304 channel_idx ?
b->channel[0].scalefactors :
305 c->scalefactors_prev;
306 const int baseline_len =
mode == 4 ?
b->q_unit_cnt_prev :
307 channel_idx ?
b->band_ext_q_unit :
311 const int unit_cnt =
FFMIN(
b->band_ext_q_unit, baseline_len);
314 for (
int i = 0;
i < unit_cnt;
i++) {
316 c->scalefactors[
i] = baseline[
i] + dist;
319 for (
int i = unit_cnt;
i <
b->band_ext_q_unit;
i++)
325 const int *baseline = channel_idx ?
b->channel[0].scalefactors :
326 c->scalefactors_prev;
327 const int baseline_len = channel_idx ?
b->band_ext_q_unit :
332 const int unit_cnt =
FFMIN(
b->band_ext_q_unit, baseline_len);
337 for (
int i = 1;
i < unit_cnt;
i++) {
340 c->scalefactors[
i] =
val & ((1 <<
len) - 1);
343 for (
int i = 0;
i < unit_cnt;
i++)
344 c->scalefactors[
i] +=
base + baseline[
i];
346 for (
int i = unit_cnt;
i <
b->band_ext_q_unit;
i++)
352 for (
int i = 0;
i <
b->band_ext_q_unit;
i++)
353 if (
c->scalefactors[
i] < 0 ||
c->scalefactors[
i] > 31)
356 memcpy(
c->scalefactors_prev,
c->scalefactors,
sizeof(
c->scalefactors));
365 const int last_sf =
c->scalefactors[
c->q_unit_cnt];
367 memset(
c->codebookset, 0,
sizeof(
c->codebookset));
369 if (
c->q_unit_cnt <= 1)
371 if (
s->samplerate_idx > 7)
374 c->scalefactors[
c->q_unit_cnt] =
c->scalefactors[
c->q_unit_cnt - 1];
376 if (
c->q_unit_cnt > 12) {
377 for (
int i = 0;
i < 12;
i++)
378 avg +=
c->scalefactors[
i];
382 for (
int i = 8;
i <
c->q_unit_cnt;
i++) {
383 const int prev =
c->scalefactors[
i - 1];
384 const int cur =
c->scalefactors[
i ];
385 const int next =
c->scalefactors[
i + 1];
387 if ((cur -
min >= 3 || 2*cur - prev - next >= 3))
388 c->codebookset[
i] = 1;
392 for (
int i = 12;
i <
c->q_unit_cnt;
i++) {
393 const int cur =
c->scalefactors[
i];
395 const int min =
FFMIN(
c->scalefactors[
i + 1],
c->scalefactors[
i - 1]);
396 if (
c->codebookset[
i])
399 c->codebookset[
i] = (((cur -
min) >= 2) && (cur >= (
avg - cnd)));
402 c->scalefactors[
c->q_unit_cnt] = last_sf;
408 const int max_prec =
s->samplerate_idx > 7 ? 1 : 7;
410 memset(
c->q_coeffs_coarse, 0,
sizeof(
c->q_coeffs_coarse));
412 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
415 const int prec =
c->precision_coarse[
i] + 1;
417 if (prec <= max_prec) {
418 const int cb =
c->codebookset[
i];
424 for (
int j = 0; j < groups; j++) {
427 for (
int k = 0; k < huff->
value_cnt; k++) {
435 for (
int j = 0; j <
bands; j++)
444 memset(
c->q_coeffs_fine, 0,
sizeof(
c->q_coeffs_fine));
446 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
449 const int len =
c->precision_fine[
i] + 1;
451 if (
c->precision_fine[
i] <= 0)
454 for (
int j = start; j < end; j++)
462 memset(
c->coeffs, 0,
sizeof(
c->coeffs));
464 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
471 for (
int j = start; j < end; j++) {
472 const float vc =
c->q_coeffs_coarse[j] * coarse_c;
473 const float vf =
c->q_coeffs_fine[j] * fine_c;
474 c->coeffs[j] = vc + vf;
482 float *
src =
b->channel[
b->cpe_base_channel].coeffs;
483 float *dst =
b->channel[!
b->cpe_base_channel].coeffs;
488 if (
b->q_unit_cnt <=
b->stereo_q_unit)
491 for (
int i =
b->stereo_q_unit; i < b->q_unit_cnt;
i++) {
492 const int sign =
b->is_signs[
i];
495 for (
int j = start; j < end; j++)
496 dst[j] = sign*
src[j];
503 for (
int i = 0;
i <= stereo;
i++) {
504 float *coeffs =
b->channel[
i].coeffs;
505 for (
int j = 0; j <
b->q_unit_cnt; j++) {
508 const int scalefactor =
b->channel[
i].scalefactors[j];
510 for (
int k = start; k < end; k++)
517 int start,
int count)
520 for (
int i = 0;
i < count;
i += 2) {
523 c->coeffs[start +
i + 0] =
tmp[0];
524 c->coeffs[start +
i + 1] =
tmp[1];
528 for (
int i = 0;
i < count;
i++)
529 c->coeffs[start +
i] /= maxval;
533 const int s_unit,
const int e_unit)
535 for (
int i = s_unit;
i < e_unit;
i++) {
538 for (
int j = start; j < end; j++)
539 c->coeffs[j] *= sf[
i - s_unit];
546 const int g_units[4] = {
550 FFMAX(g_units[2], 22),
553 const int g_bins[4] = {
560 for (
int ch = 0; ch <= stereo; ch++) {
564 for (
int i = 0;
i < 3;
i++)
565 for (
int j = 0; j < (g_bins[
i + 1] - g_bins[
i + 0]); j++)
566 c->coeffs[g_bins[
i] + j] =
c->coeffs[g_bins[
i] - j - 1];
568 switch (
c->band_ext) {
570 float sf[6] = { 0.0f };
571 const int l = g_units[3] - g_units[0] - 1;
604 for (
int i = g_units[0];
i < g_units[3];
i++)
612 const float g_sf[2] = {
617 for (
int i = 0;
i < 2;
i++)
618 for (
int j = g_bins[
i + 0]; j < g_bins[
i + 1]; j++)
619 c->coeffs[j] *= g_sf[
i];
626 for (
int i = g_bins[0];
i < g_bins[3];
i++) {
634 const float g_sf[3] = { 0.7079468f*m, 0.5011902f*m, 0.3548279f*m };
636 for (
int i = 0;
i < 3;
i++)
637 for (
int j = g_bins[
i + 0]; j < g_bins[
i + 1]; j++)
638 c->coeffs[j] *= g_sf[
i];
647 int frame_idx,
int block_idx)
655 const int precision = reuse_params ? 8 : 4;
656 c->q_unit_cnt =
b->q_unit_cnt = 2;
658 memset(
c->scalefactors, 0,
sizeof(
c->scalefactors));
659 memset(
c->q_coeffs_fine, 0,
sizeof(
c->q_coeffs_fine));
660 memset(
c->q_coeffs_coarse, 0,
sizeof(
c->q_coeffs_coarse));
662 for (
int i = 0;
i <
b->q_unit_cnt;
i++) {
664 c->precision_coarse[
i] = precision;
665 c->precision_fine[
i] = 0;
668 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
671 for (
int j = start; j < end; j++)
672 c->q_coeffs_coarse[j] =
get_bits(gb,
c->precision_coarse[
i] + 1);
681 if (first_in_pkt && reuse_params) {
688 int stereo_band, ext_band;
689 const int min_band_count =
s->samplerate_idx > 7 ? 1 : 3;
691 b->band_count =
get_bits(gb, 4) + min_band_count;
694 b->band_ext_q_unit =
b->stereo_q_unit =
b->q_unit_cnt;
703 stereo_band =
get_bits(gb, 4) + min_band_count;
704 if (stereo_band >
b->band_count) {
713 if (
b->has_band_ext) {
714 ext_band =
get_bits(gb, 4) + min_band_count;
715 if (ext_band < b->band_count) {
734 b->cpe_base_channel = 0;
738 for (
int i =
b->stereo_q_unit; i < b->q_unit_cnt;
i++)
751 for (
int i = 0;
i <= stereo;
i++) {
753 c->q_unit_cnt =
i ==
b->cpe_base_channel ?
b->q_unit_cnt :
765 b->q_unit_cnt_prev =
b->has_band_ext ?
b->band_ext_q_unit :
b->q_unit_cnt;
770 if (
b->has_band_ext &&
b->has_band_ext_data)
774 for (
int i = 0;
i <= stereo;
i++) {
776 const int dst_idx =
s->block_config->plane_map[block_idx][
i];
777 const int wsize = 1 <<
s->frame_log2;
778 const ptrdiff_t
offset = wsize*frame_idx*
sizeof(
float);
779 float *dst = (
float *)(
frame->extended_data[dst_idx] +
offset);
781 s->imdct.imdct_half(&
s->imdct,
s->temp,
c->coeffs);
782 s->fdsp->vector_fmul_window(dst,
c->prev_win,
s->temp,
783 s->imdct_win, wsize >> 1);
784 memcpy(
c->prev_win,
s->temp + (wsize >> 1),
sizeof(
float)*wsize >> 1);
791 int *got_frame_ptr,
AVPacket *avpkt)
806 for (
int j = 0; j <
s->block_config->count; j++) {
823 for (
int j = 0; j <
s->block_config->count; j++) {
826 for (
int i = 0;
i <= stereo;
i++) {
828 memset(
c->prev_win, 0,
sizeof(
c->prev_win));
844 const uint8_t (**
tab)[2],
845 unsigned *buf_offset,
int offset)
852 &(*
tab)[0][1], 2, &(*
tab)[0][0], 2, 1,
860 const uint8_t (*
tab)[2];
865 for (
int i = 1;
i < 7;
i++) {
874 for (
int i = 2;
i < 6;
i++) {
886 for (
int i = 0;
i < 2;
i++) {
887 for (
int j = 2; j < 8; j++) {
888 for (
int k =
i; k < 4; k++) {
902 int version, block_config_idx, superframe_idx, alloc_c_len;
934 block_config_idx =
get_bits(&gb, 3);
935 if (block_config_idx > 5) {
942 avctx->
ch_layout =
s->block_config->channel_layout;
951 s->avg_frame_size =
get_bits(&gb, 11) + 1;
954 if (superframe_idx & 1) {
959 s->frame_count = 1 << superframe_idx;
962 if (
ff_mdct_init(&
s->imdct,
s->frame_log2 + 1, 1, 1.0f / 32768.0f))
970 for (
int i = 0;
i < (1 <<
s->frame_log2);
i++) {
971 const int len = 1 <<
s->frame_log2;
972 const float sidx = (
i + 0.5f) /
len;
973 const float eidx = (
len -
i - 0.5f) /
len;
976 s->imdct_win[
i] = s_c / ((s_c * s_c) + (e_c * e_c));
981 for (
int i = 1;
i <= alloc_c_len;
i++)
982 for (
int j = 0; j <
i; j++)
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.
static VLC coeff_vlc[2][8][4]
#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 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)
#define ATRAC9_COEFF_VLC_BITS
static const ATRAC9BlockConfig at9_block_layout[]
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 av_always_inline float scale(float x, float s)
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.
static const uint8_t at9_tab_sri_max_bands[]
#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)
static VLCElem vlc_buf[16716]
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]
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)
int ff_init_vlc_from_lengths(VLC *vlc, 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, void *logctx)
Build VLC decoding tables suitable for use with get_vlc2()
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 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
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
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.
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
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[]
static const float at9_quant_step_coarse[]
#define DECLARE_ALIGNED(n, t, v)
int32_t scalefactors_prev[31]
const ATRAC9BlockConfig * block_config
#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)
static const float at9_band_ext_scales_m3[][2]
static const float at9_scalefactor_c[]
static const float at9_band_ext_scales_m4[]
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
const char * name
Name of the codec implementation.
static av_cold void atrac9_init_vlc(VLC *vlc, int nb_bits, int nb_codes, const uint8_t(**tab)[2], unsigned *buf_offset, int offset)
#define INIT_VLC_STATIC_OVERLONG
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
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
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]
main external API structure.
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]
static const uint8_t at9_tab_band_ext_group[][3]
#define AV_CODEC_CAP_SUBFRAMES
Codec can output multiple frames per AVPacket Normally demuxers return one frame at a time,...
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
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[]