Go to the documentation of this file.
35 #define ATRAC9_SF_VLC_BITS 8
36 #define ATRAC9_COEFF_VLC_BITS 9
129 grad_range[1] =
get_bits(gb, 6) + 1;
135 if (grad_range[0] >= grad_range[1] || grad_range[1] > 31)
138 if (
b->grad_boundary >
b->q_unit_cnt)
141 values = grad_value[1] - grad_value[0];
142 sign = 1 - 2*(
values < 0);
143 base = grad_value[0] + sign;
145 curve =
s->alloc_curve[grad_range[1] - grad_range[0] - 1];
147 for (
int i = 0;
i <=
b->q_unit_cnt;
i++)
148 b->gradient[
i] = grad_value[
i >= grad_range[0]];
150 for (
int i = grad_range[0];
i < grad_range[1];
i++)
151 b->gradient[
i] =
base + sign*((
int)(
scale*curve[
i - grad_range[0]]));
159 memset(
c->precision_mask, 0,
sizeof(
c->precision_mask));
160 for (
int i = 1;
i <
b->q_unit_cnt;
i++) {
161 const int delta =
FFABS(
c->scalefactors[
i] -
c->scalefactors[
i - 1]) - 1;
163 const int neg =
c->scalefactors[
i - 1] >
c->scalefactors[
i];
169 for (
int i = 0;
i <
b->q_unit_cnt;
i++) {
170 c->precision_coarse[
i] =
c->scalefactors[
i];
171 c->precision_coarse[
i] +=
c->precision_mask[
i] -
b->gradient[
i];
172 if (
c->precision_coarse[
i] < 0)
174 switch (
b->grad_mode) {
176 c->precision_coarse[
i] >>= 1;
179 c->precision_coarse[
i] = (3 *
c->precision_coarse[
i]) >> 3;
182 c->precision_coarse[
i] >>= 2;
187 for (
int i = 0;
i <
b->q_unit_cnt;
i++)
188 c->precision_coarse[
i] =
c->scalefactors[
i] -
b->gradient[
i];
192 for (
int i = 0;
i <
b->q_unit_cnt;
i++)
193 c->precision_coarse[
i] =
FFMAX(
c->precision_coarse[
i], 1);
195 for (
int i = 0;
i <
b->grad_boundary;
i++)
196 c->precision_coarse[
i]++;
198 for (
int i = 0;
i <
b->q_unit_cnt;
i++) {
199 c->precision_fine[
i] = 0;
200 if (
c->precision_coarse[
i] > 15) {
201 c->precision_fine[
i] =
FFMIN(
c->precision_coarse[
i], 30) - 15;
202 c->precision_coarse[
i] = 15;
212 if (
b->has_band_ext) {
213 if (
b->q_unit_cnt < 13 ||
b->q_unit_cnt > 20)
217 b->channel[1].band_ext =
get_bits(gb, 2);
218 b->channel[1].band_ext = ext_band > 2 ?
b->channel[1].band_ext : 4;
225 if (!
b->has_band_ext_data)
228 if (!
b->has_band_ext) {
234 b->channel[0].band_ext =
get_bits(gb, 2);
235 b->channel[0].band_ext = ext_band > 2 ?
b->channel[0].band_ext : 4;
238 for (
int i = 0;
i <= stereo;
i++) {
241 for (
int j = 0; j < count; j++) {
250 for (
int i = 0;
i <= stereo;
i++) {
253 for (
int j = 0; j < count; j++) {
264 int channel_idx,
int first_in_pkt)
266 static const uint8_t mode_map[2][4] = { { 0, 1, 2, 3 }, { 0, 2, 3, 4 } };
267 const int mode = mode_map[channel_idx][
get_bits(gb, 2)];
269 memset(
c->scalefactors, 0,
sizeof(
c->scalefactors));
271 if (first_in_pkt && (
mode == 4 || ((
mode == 3) && !channel_idx))) {
285 for (
int i = 1;
i <
b->band_ext_q_unit;
i++) {
288 c->scalefactors[
i] =
val & ((1 <<
len) - 1);
291 for (
int i = 0;
i <
b->band_ext_q_unit;
i++)
292 c->scalefactors[
i] +=
base - sf_weights[
i];
299 for (
int i = 0;
i <
b->band_ext_q_unit;
i++)
305 const int *baseline =
mode == 4 ?
c->scalefactors_prev :
306 channel_idx ?
b->channel[0].scalefactors :
307 c->scalefactors_prev;
308 const int baseline_len =
mode == 4 ?
b->q_unit_cnt_prev :
309 channel_idx ?
b->band_ext_q_unit :
313 const int unit_cnt =
FFMIN(
b->band_ext_q_unit, baseline_len);
316 for (
int i = 0;
i < unit_cnt;
i++) {
318 c->scalefactors[
i] = baseline[
i] + dist;
321 for (
int i = unit_cnt;
i <
b->band_ext_q_unit;
i++)
327 const int *baseline = channel_idx ?
b->channel[0].scalefactors :
328 c->scalefactors_prev;
329 const int baseline_len = channel_idx ?
b->band_ext_q_unit :
334 const int unit_cnt =
FFMIN(
b->band_ext_q_unit, baseline_len);
339 for (
int i = 1;
i < unit_cnt;
i++) {
342 c->scalefactors[
i] =
val & ((1 <<
len) - 1);
345 for (
int i = 0;
i < unit_cnt;
i++)
346 c->scalefactors[
i] +=
base + baseline[
i];
348 for (
int i = unit_cnt;
i <
b->band_ext_q_unit;
i++)
354 for (
int i = 0;
i <
b->band_ext_q_unit;
i++)
355 if (
c->scalefactors[
i] < 0 ||
c->scalefactors[
i] > 31)
358 memcpy(
c->scalefactors_prev,
c->scalefactors,
sizeof(
c->scalefactors));
367 const int last_sf =
c->scalefactors[
c->q_unit_cnt];
369 memset(
c->codebookset, 0,
sizeof(
c->codebookset));
371 if (
c->q_unit_cnt <= 1)
373 if (
s->samplerate_idx > 7)
376 c->scalefactors[
c->q_unit_cnt] =
c->scalefactors[
c->q_unit_cnt - 1];
378 if (
c->q_unit_cnt > 12) {
379 for (
int i = 0;
i < 12;
i++)
380 avg +=
c->scalefactors[
i];
384 for (
int i = 8;
i <
c->q_unit_cnt;
i++) {
385 const int prev =
c->scalefactors[
i - 1];
386 const int cur =
c->scalefactors[
i ];
387 const int next =
c->scalefactors[
i + 1];
389 if ((cur -
min >= 3 || 2*cur - prev - next >= 3))
390 c->codebookset[
i] = 1;
394 for (
int i = 12;
i <
c->q_unit_cnt;
i++) {
395 const int cur =
c->scalefactors[
i];
397 const int min =
FFMIN(
c->scalefactors[
i + 1],
c->scalefactors[
i - 1]);
398 if (
c->codebookset[
i])
401 c->codebookset[
i] = (((cur -
min) >= 2) && (cur >= (
avg - cnd)));
404 c->scalefactors[
c->q_unit_cnt] = last_sf;
410 const int max_prec =
s->samplerate_idx > 7 ? 1 : 7;
412 memset(
c->q_coeffs_coarse, 0,
sizeof(
c->q_coeffs_coarse));
414 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
417 const int prec =
c->precision_coarse[
i] + 1;
419 if (prec <= max_prec) {
420 const int cb =
c->codebookset[
i];
426 for (
int j = 0; j < groups; j++) {
429 for (
int k = 0; k < huff->
value_cnt; k++) {
437 for (
int j = 0; j <
bands; j++)
446 memset(
c->q_coeffs_fine, 0,
sizeof(
c->q_coeffs_fine));
448 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
451 const int len =
c->precision_fine[
i] + 1;
453 if (
c->precision_fine[
i] <= 0)
456 for (
int j = start; j < end; j++)
464 memset(
c->coeffs, 0,
sizeof(
c->coeffs));
466 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
473 for (
int j = start; j < end; j++) {
474 const float vc =
c->q_coeffs_coarse[j] * coarse_c;
475 const float vf =
c->q_coeffs_fine[j] * fine_c;
476 c->coeffs[j] = vc +
vf;
484 float *
src =
b->channel[
b->cpe_base_channel].coeffs;
485 float *
dst =
b->channel[!
b->cpe_base_channel].coeffs;
490 if (
b->q_unit_cnt <=
b->stereo_q_unit)
493 for (
int i =
b->stereo_q_unit; i < b->q_unit_cnt;
i++) {
494 const int sign =
b->is_signs[
i];
497 for (
int j = start; j < end; j++)
505 for (
int i = 0;
i <= stereo;
i++) {
506 float *coeffs =
b->channel[
i].coeffs;
507 for (
int j = 0; j <
b->q_unit_cnt; j++) {
510 const int scalefactor =
b->channel[
i].scalefactors[j];
512 for (
int k = start; k < end; k++)
519 int start,
int count)
522 for (
int i = 0;
i < count;
i += 2) {
525 c->coeffs[start +
i + 0] =
tmp[0];
526 c->coeffs[start +
i + 1] =
tmp[1];
530 for (
int i = 0;
i < count;
i++)
531 c->coeffs[start +
i] /= maxval;
535 const int s_unit,
const int e_unit)
537 for (
int i = s_unit;
i < e_unit;
i++) {
540 for (
int j = start; j < end; j++)
541 c->coeffs[j] *= sf[
i - s_unit];
548 const int g_units[4] = {
552 FFMAX(g_units[2], 22),
555 const int g_bins[4] = {
562 for (
int ch = 0; ch <= stereo; ch++) {
566 for (
int i = 0;
i < 3;
i++)
567 for (
int j = 0; j < (g_bins[
i + 1] - g_bins[
i + 0]); j++)
568 c->coeffs[g_bins[
i] + j] =
c->coeffs[g_bins[
i] - j - 1];
570 switch (
c->band_ext) {
572 float sf[6] = { 0.0f };
573 const int l = g_units[3] - g_units[0] - 1;
606 for (
int i = g_units[0];
i < g_units[3];
i++)
614 const float g_sf[2] = {
619 for (
int i = 0;
i < 2;
i++)
620 for (
int j = g_bins[
i + 0]; j < g_bins[
i + 1]; j++)
621 c->coeffs[j] *= g_sf[
i];
628 for (
int i = g_bins[0];
i < g_bins[3];
i++) {
636 const float g_sf[3] = { 0.7079468f*m, 0.5011902f*m, 0.3548279f*m };
638 for (
int i = 0;
i < 3;
i++)
639 for (
int j = g_bins[
i + 0]; j < g_bins[
i + 1]; j++)
640 c->coeffs[j] *= g_sf[
i];
649 int frame_idx,
int block_idx)
657 const int precision = reuse_params ? 8 : 4;
658 c->q_unit_cnt =
b->q_unit_cnt = 2;
660 memset(
c->scalefactors, 0,
sizeof(
c->scalefactors));
661 memset(
c->q_coeffs_fine, 0,
sizeof(
c->q_coeffs_fine));
662 memset(
c->q_coeffs_coarse, 0,
sizeof(
c->q_coeffs_coarse));
664 for (
int i = 0;
i <
b->q_unit_cnt;
i++) {
666 c->precision_coarse[
i] = precision;
667 c->precision_fine[
i] = 0;
670 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
673 for (
int j = start; j < end; j++)
674 c->q_coeffs_coarse[j] =
get_bits(gb,
c->precision_coarse[
i] + 1);
683 if (first_in_pkt && reuse_params) {
690 int stereo_band, ext_band;
691 const int min_band_count =
s->samplerate_idx > 7 ? 1 : 3;
693 b->band_count =
get_bits(gb, 4) + min_band_count;
696 b->band_ext_q_unit =
b->stereo_q_unit =
b->q_unit_cnt;
705 stereo_band =
get_bits(gb, 4) + min_band_count;
706 if (stereo_band >
b->band_count) {
715 if (
b->has_band_ext) {
716 ext_band =
get_bits(gb, 4) + min_band_count;
717 if (ext_band < b->band_count) {
736 b->cpe_base_channel = 0;
740 for (
int i =
b->stereo_q_unit; i < b->q_unit_cnt;
i++)
753 for (
int i = 0;
i <= stereo;
i++) {
755 c->q_unit_cnt =
i ==
b->cpe_base_channel ?
b->q_unit_cnt :
767 b->q_unit_cnt_prev =
b->has_band_ext ?
b->band_ext_q_unit :
b->q_unit_cnt;
772 if (
b->has_band_ext &&
b->has_band_ext_data)
776 for (
int i = 0;
i <= stereo;
i++) {
778 const int dst_idx =
s->block_config->plane_map[block_idx][
i];
779 const int wsize = 1 <<
s->frame_log2;
780 const ptrdiff_t
offset = wsize*frame_idx*
sizeof(
float);
783 s->tx_fn(
s->tx,
s->temp,
c->coeffs,
sizeof(
float));
784 s->fdsp->vector_fmul_window(
dst,
c->prev_win,
s->temp,
785 s->imdct_win, wsize >> 1);
786 memcpy(
c->prev_win,
s->temp + (wsize >> 1),
sizeof(
float)*wsize >> 1);
793 int *got_frame_ptr,
AVPacket *avpkt)
810 for (
int j = 0; j <
s->block_config->count; j++) {
827 for (
int j = 0; j <
s->block_config->count; j++) {
830 for (
int i = 0;
i <= stereo;
i++) {
832 memset(
c->prev_win, 0,
sizeof(
c->prev_win));
848 int nb_bits,
int nb_codes,
863 const uint8_t (*
tab)[2];
867 for (
int i = 1;
i < 7;
i++) {
876 for (
int i = 2;
i < 6;
i++) {
883 hf->size, &
tab, -16);
888 for (
int i = 0;
i < 2;
i++) {
889 for (
int j = 2; j < 8; j++) {
890 for (
int k =
i; k < 4; k++) {
905 int err,
version, block_config_idx, superframe_idx, alloc_c_len;
939 block_config_idx =
get_bits(&gb, 3);
940 if (block_config_idx > 5) {
947 avctx->
ch_layout =
s->block_config->channel_layout;
956 s->avg_frame_size =
get_bits(&gb, 11) + 1;
959 if (superframe_idx & 1) {
964 s->frame_count = 1 << superframe_idx;
967 scale = 1.0f / 32768.0;
969 1 <<
s->frame_log2, &
scale, 0);
978 for (
int i = 0;
i < (1 <<
s->frame_log2);
i++) {
979 const int len = 1 <<
s->frame_log2;
980 const float sidx = (
i + 0.5f) /
len;
981 const float eidx = (
len -
i - 0.5f) /
len;
984 s->imdct_win[
i] = s_c / ((s_c * s_c) + (e_c * e_c));
989 for (
int i = 1;
i <= alloc_c_len;
i++)
990 for (
int j = 0; j <
i; j++)
1010 #if FF_API_SUBFRAMES
1011 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.
#define DECLARE_ALIGNED(n, t, v)
uint8_t ptrdiff_t const uint8_t ptrdiff_t int intptr_t intptr_t int int16_t * dst
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
#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[]
uint8_t ptrdiff_t const uint8_t ptrdiff_t int const int8_t * hf
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.
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]
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).
static void scale(int *out, const int *in, const int w, const int h, const int shift)
#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.
uint8_t ptrdiff_t const uint8_t ptrdiff_t int const int8_t const int8_t * vf
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[]