Go to the documentation of this file.
118 grad_range[1] =
get_bits(gb, 6) + 1;
124 if (grad_range[0] >= grad_range[1] || grad_range[1] > 31)
127 if (
b->grad_boundary >
b->q_unit_cnt)
130 values = grad_value[1] - grad_value[0];
131 sign = 1 - 2*(
values < 0);
132 base = grad_value[0] + sign;
134 curve =
s->alloc_curve[grad_range[1] - grad_range[0] - 1];
136 for (
int i = 0;
i <=
b->q_unit_cnt;
i++)
137 b->gradient[
i] = grad_value[
i >= grad_range[0]];
139 for (
int i = grad_range[0];
i < grad_range[1];
i++)
140 b->gradient[
i] =
base + sign*((
int)(scale*curve[
i - grad_range[0]]));
148 memset(
c->precision_mask, 0,
sizeof(
c->precision_mask));
149 for (
int i = 1;
i <
b->q_unit_cnt;
i++) {
150 const int delta =
FFABS(
c->scalefactors[
i] -
c->scalefactors[
i - 1]) - 1;
152 const int neg =
c->scalefactors[
i - 1] >
c->scalefactors[
i];
158 for (
int i = 0;
i <
b->q_unit_cnt;
i++) {
159 c->precision_coarse[
i] =
c->scalefactors[
i];
160 c->precision_coarse[
i] +=
c->precision_mask[
i] -
b->gradient[
i];
161 if (
c->precision_coarse[
i] < 0)
163 switch (
b->grad_mode) {
165 c->precision_coarse[
i] >>= 1;
168 c->precision_coarse[
i] = (3 *
c->precision_coarse[
i]) >> 3;
171 c->precision_coarse[
i] >>= 2;
176 for (
int i = 0;
i <
b->q_unit_cnt;
i++)
177 c->precision_coarse[
i] =
c->scalefactors[
i] -
b->gradient[
i];
181 for (
int i = 0;
i <
b->q_unit_cnt;
i++)
182 c->precision_coarse[
i] =
FFMAX(
c->precision_coarse[
i], 1);
184 for (
int i = 0;
i <
b->grad_boundary;
i++)
185 c->precision_coarse[
i]++;
187 for (
int i = 0;
i <
b->q_unit_cnt;
i++) {
188 c->precision_fine[
i] = 0;
189 if (
c->precision_coarse[
i] > 15) {
190 c->precision_fine[
i] =
FFMIN(
c->precision_coarse[
i], 30) - 15;
191 c->precision_coarse[
i] = 15;
201 if (
b->has_band_ext) {
202 if (
b->q_unit_cnt < 13 ||
b->q_unit_cnt > 20)
206 b->channel[1].band_ext =
get_bits(gb, 2);
207 b->channel[1].band_ext = ext_band > 2 ?
b->channel[1].band_ext : 4;
214 if (!
b->has_band_ext_data)
217 if (!
b->has_band_ext) {
223 b->channel[0].band_ext =
get_bits(gb, 2);
224 b->channel[0].band_ext = ext_band > 2 ?
b->channel[0].band_ext : 4;
227 for (
int i = 0;
i <= stereo;
i++) {
230 for (
int j = 0; j < count; j++) {
239 for (
int i = 0;
i <= stereo;
i++) {
242 for (
int j = 0; j < count; j++) {
253 int channel_idx,
int first_in_pkt)
255 static const uint8_t mode_map[2][4] = { { 0, 1, 2, 3 }, { 0, 2, 3, 4 } };
256 const int mode = mode_map[channel_idx][
get_bits(gb, 2)];
258 memset(
c->scalefactors, 0,
sizeof(
c->scalefactors));
260 if (first_in_pkt && (
mode == 4 || ((
mode == 3) && !channel_idx))) {
274 for (
int i = 1;
i <
b->band_ext_q_unit;
i++) {
276 c->scalefactors[
i] =
val & ((1 <<
len) - 1);
279 for (
int i = 0;
i <
b->band_ext_q_unit;
i++)
280 c->scalefactors[
i] +=
base - sf_weights[
i];
287 for (
int i = 0;
i <
b->band_ext_q_unit;
i++)
293 const int *baseline =
mode == 4 ?
c->scalefactors_prev :
294 channel_idx ?
b->channel[0].scalefactors :
295 c->scalefactors_prev;
296 const int baseline_len =
mode == 4 ?
b->q_unit_cnt_prev :
297 channel_idx ?
b->band_ext_q_unit :
301 const int unit_cnt =
FFMIN(
b->band_ext_q_unit, baseline_len);
304 for (
int i = 0;
i < unit_cnt;
i++) {
306 c->scalefactors[
i] = baseline[
i] + dist;
309 for (
int i = unit_cnt;
i <
b->band_ext_q_unit;
i++)
315 const int *baseline = channel_idx ?
b->channel[0].scalefactors :
316 c->scalefactors_prev;
317 const int baseline_len = channel_idx ?
b->band_ext_q_unit :
322 const int unit_cnt =
FFMIN(
b->band_ext_q_unit, baseline_len);
327 for (
int i = 1;
i < unit_cnt;
i++) {
329 c->scalefactors[
i] =
val & ((1 <<
len) - 1);
332 for (
int i = 0;
i < unit_cnt;
i++)
333 c->scalefactors[
i] +=
base + baseline[
i];
335 for (
int i = unit_cnt;
i <
b->band_ext_q_unit;
i++)
341 for (
int i = 0;
i <
b->band_ext_q_unit;
i++)
342 if (
c->scalefactors[
i] < 0 ||
c->scalefactors[
i] > 31)
345 memcpy(
c->scalefactors_prev,
c->scalefactors,
sizeof(
c->scalefactors));
354 const int last_sf =
c->scalefactors[
c->q_unit_cnt];
356 memset(
c->codebookset, 0,
sizeof(
c->codebookset));
358 if (
c->q_unit_cnt <= 1)
360 if (
s->samplerate_idx > 7)
363 c->scalefactors[
c->q_unit_cnt] =
c->scalefactors[
c->q_unit_cnt - 1];
365 if (
c->q_unit_cnt > 12) {
366 for (
int i = 0;
i < 12;
i++)
367 avg +=
c->scalefactors[
i];
371 for (
int i = 8;
i <
c->q_unit_cnt;
i++) {
372 const int prev =
c->scalefactors[
i - 1];
373 const int cur =
c->scalefactors[
i ];
374 const int next =
c->scalefactors[
i + 1];
376 if ((cur -
min >= 3 || 2*cur - prev - next >= 3))
377 c->codebookset[
i] = 1;
381 for (
int i = 12;
i <
c->q_unit_cnt;
i++) {
382 const int cur =
c->scalefactors[
i];
384 const int min =
FFMIN(
c->scalefactors[
i + 1],
c->scalefactors[
i - 1]);
385 if (
c->codebookset[
i])
388 c->codebookset[
i] = (((cur -
min) >= 2) && (cur >= (
avg - cnd)));
391 c->scalefactors[
c->q_unit_cnt] = last_sf;
397 const int max_prec =
s->samplerate_idx > 7 ? 1 : 7;
399 memset(
c->q_coeffs_coarse, 0,
sizeof(
c->q_coeffs_coarse));
401 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
404 const int prec =
c->precision_coarse[
i] + 1;
406 if (prec <= max_prec) {
407 const int cb =
c->codebookset[
i];
409 const VLC *
tab = &
s->coeff_vlc[
cb][prec][cbi];
413 for (
int j = 0; j < groups; j++) {
416 for (
int k = 0; k < huff->
value_cnt; k++) {
424 for (
int j = 0; j <
bands; j++)
433 memset(
c->q_coeffs_fine, 0,
sizeof(
c->q_coeffs_fine));
435 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
438 const int len =
c->precision_fine[
i] + 1;
440 if (
c->precision_fine[
i] <= 0)
443 for (
int j = start; j <
end; j++)
451 memset(
c->coeffs, 0,
sizeof(
c->coeffs));
453 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
460 for (
int j = start; j <
end; j++) {
461 const float vc =
c->q_coeffs_coarse[j] * coarse_c;
462 const float vf =
c->q_coeffs_fine[j] * fine_c;
463 c->coeffs[j] = vc + vf;
471 float *
src =
b->channel[
b->cpe_base_channel].coeffs;
472 float *dst =
b->channel[!
b->cpe_base_channel].coeffs;
477 if (
b->q_unit_cnt <=
b->stereo_q_unit)
480 for (
int i =
b->stereo_q_unit; i < b->q_unit_cnt;
i++) {
481 const int sign =
b->is_signs[
i];
484 for (
int j = start; j <
end; j++)
485 dst[j] = sign*
src[j];
492 for (
int i = 0;
i <= stereo;
i++) {
493 float *coeffs =
b->channel[
i].coeffs;
494 for (
int j = 0; j <
b->q_unit_cnt; j++) {
497 const int scalefactor =
b->channel[
i].scalefactors[j];
499 for (
int k = start; k <
end; k++)
506 int start,
int count)
509 for (
int i = 0;
i < count;
i += 2) {
512 c->coeffs[start +
i + 0] =
tmp[0];
513 c->coeffs[start +
i + 1] =
tmp[1];
517 for (
int i = 0;
i < count;
i++)
518 c->coeffs[start +
i] /= maxval;
522 const int s_unit,
const int e_unit)
524 for (
int i = s_unit;
i < e_unit;
i++) {
527 for (
int j = start; j <
end; j++)
528 c->coeffs[j] *= sf[
i - s_unit];
535 const int g_units[4] = {
539 FFMAX(g_units[2], 22),
542 const int g_bins[4] = {
549 for (
int ch = 0; ch <= stereo; ch++) {
553 for (
int i = 0;
i < 3;
i++)
554 for (
int j = 0; j < (g_bins[
i + 1] - g_bins[
i + 0]); j++)
555 c->coeffs[g_bins[
i] + j] =
c->coeffs[g_bins[
i] - j - 1];
557 switch (
c->band_ext) {
559 float sf[6] = { 0.0f };
560 const int l = g_units[3] - g_units[0] - 1;
593 for (
int i = g_units[0];
i < g_units[3];
i++)
601 const float g_sf[2] = {
606 for (
int i = 0;
i < 2;
i++)
607 for (
int j = g_bins[
i + 0]; j < g_bins[
i + 1]; j++)
608 c->coeffs[j] *= g_sf[
i];
615 for (
int i = g_bins[0];
i < g_bins[3];
i++) {
617 c->coeffs[
i] *= scale;
623 const float g_sf[3] = { 0.7079468f*m, 0.5011902f*m, 0.3548279f*m };
625 for (
int i = 0;
i < 3;
i++)
626 for (
int j = g_bins[
i + 0]; j < g_bins[
i + 1]; j++)
627 c->coeffs[j] *= g_sf[
i];
636 int frame_idx,
int block_idx)
644 const int precision = reuse_params ? 8 : 4;
645 c->q_unit_cnt =
b->q_unit_cnt = 2;
647 memset(
c->scalefactors, 0,
sizeof(
c->scalefactors));
648 memset(
c->q_coeffs_fine, 0,
sizeof(
c->q_coeffs_fine));
649 memset(
c->q_coeffs_coarse, 0,
sizeof(
c->q_coeffs_coarse));
651 for (
int i = 0;
i <
b->q_unit_cnt;
i++) {
653 c->precision_coarse[
i] = precision;
654 c->precision_fine[
i] = 0;
657 for (
int i = 0;
i <
c->q_unit_cnt;
i++) {
660 for (
int j = start; j <
end; j++)
661 c->q_coeffs_coarse[j] =
get_bits(gb,
c->precision_coarse[
i] + 1);
670 if (first_in_pkt && reuse_params) {
677 int stereo_band, ext_band;
678 const int min_band_count =
s->samplerate_idx > 7 ? 1 : 3;
680 b->band_count =
get_bits(gb, 4) + min_band_count;
683 b->band_ext_q_unit =
b->stereo_q_unit =
b->q_unit_cnt;
692 stereo_band =
get_bits(gb, 4) + min_band_count;
693 if (stereo_band >
b->band_count) {
702 if (
b->has_band_ext) {
703 ext_band =
get_bits(gb, 4) + min_band_count;
704 if (ext_band < b->band_count) {
723 b->cpe_base_channel = 0;
727 for (
int i =
b->stereo_q_unit; i < b->q_unit_cnt;
i++)
740 for (
int i = 0;
i <= stereo;
i++) {
742 c->q_unit_cnt =
i ==
b->cpe_base_channel ?
b->q_unit_cnt :
754 b->q_unit_cnt_prev =
b->has_band_ext ?
b->band_ext_q_unit :
b->q_unit_cnt;
759 if (
b->has_band_ext &&
b->has_band_ext_data)
763 for (
int i = 0;
i <= stereo;
i++) {
765 const int dst_idx =
s->block_config->plane_map[block_idx][
i];
766 const int wsize = 1 <<
s->frame_log2;
767 const ptrdiff_t
offset = wsize*frame_idx*
sizeof(float);
768 float *dst = (
float *)(
frame->extended_data[dst_idx] +
offset);
770 s->imdct.imdct_half(&
s->imdct,
s->temp,
c->coeffs);
771 s->fdsp->vector_fmul_window(dst,
c->prev_win,
s->temp,
772 s->imdct_win, wsize >> 1);
773 memcpy(
c->prev_win,
s->temp + (wsize >> 1),
sizeof(
float)*wsize >> 1);
780 int *got_frame_ptr,
AVPacket *avpkt)
796 for (
int j = 0; j <
s->block_config->count; j++) {
813 for (
int j = 0; j <
s->block_config->count; j++) {
816 for (
int i = 0;
i <= stereo;
i++) {
818 memset(
c->prev_win, 0,
sizeof(
c->prev_win));
827 for (
int i = 1;
i < 7;
i++)
829 for (
int i = 2;
i < 6;
i++)
831 for (
int i = 0;
i < 2;
i++)
832 for (
int j = 0; j < 8; j++)
833 for (
int k = 0; k < 4; k++)
846 int version, block_config_idx, superframe_idx, alloc_c_len;
878 block_config_idx =
get_bits(&gb, 3);
879 if (block_config_idx > 5) {
895 s->avg_frame_size =
get_bits(&gb, 11) + 1;
898 if (superframe_idx & 1) {
903 s->frame_count = 1 << superframe_idx;
906 if (
ff_mdct_init(&
s->imdct,
s->frame_log2 + 1, 1, 1.0f / 32768.0f))
914 for (
int i = 0;
i < (1 <<
s->frame_log2);
i++) {
915 const int len = 1 <<
s->frame_log2;
916 const float sidx = (
i + 0.5f) /
len;
917 const float eidx = (
len -
i - 0.5f) /
len;
920 s->imdct_win[
i] = s_c / ((s_c * s_c) + (e_c * e_c));
925 for (
int i = 1;
i <= alloc_c_len;
i++)
926 for (
int j = 0; j <
i; j++)
930 for (
int i = 1;
i < 7;
i++) {
938 for (
int i = 2;
i < 6;
i++) {
943 for (
int j = 0; j < nums; j++)
947 hf->
codes, 2, 2, sym,
sizeof(*sym),
sizeof(*sym), 0);
951 for (
int i = 0;
i < 2;
i++) {
952 for (
int j = 0; j < 8; j++) {
953 for (
int k = 0; k < 4; k++) {
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_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
static av_cold int init(AVCodecContext *avctx)
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
uint64_t channel_layout
Audio channel layout.
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)
static av_cold int end(AVCodecContext *avctx)
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.
#define init_vlc(vlc, nb_bits, nb_codes, bits, bits_wrap, bits_size, codes, codes_wrap, codes_size, flags)
static void calc_codebook_idx(ATRAC9Context *s, ATRAC9BlockData *b, ATRAC9ChannelData *c)
static const ATRAC9BlockConfig at9_block_layout[]
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
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]
static void calc_precision(ATRAC9Context *s, ATRAC9BlockData *b, ATRAC9ChannelData *c)
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)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
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[]
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
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...
void ff_free_vlc(VLC *vlc)
static const float bands[]
static const float at9_band_ext_scales_m0[][5][32]
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
AVCodec ff_atrac9_decoder
static void flush(AVCodecContext *avctx)
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)
int ff_init_vlc_sparse(VLC *vlc_arg, int nb_bits, int nb_codes, const void *bits, int bits_wrap, int bits_size, const void *codes, int codes_wrap, int codes_size, const void *symbols, int symbols_wrap, int symbols_size, int flags)
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
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
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() 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 int atrac9_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
static const int at9_q_unit_to_coeff_idx[]
int channels
number of audio channels
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.
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
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[]
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)
#define FF_ARRAY_ELEMS(a)
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_RL32
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 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.
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
static const uint8_t at9_tab_b_dist[]
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[]