Go to the documentation of this file.
99 enum OCStatus oc_type,
int get_new_frame);
101 #define overread_err "Input buffer exhausted before END element found\n"
106 for (
i = 0;
i < tags;
i++) {
109 sum += (1 + (syn_ele ==
TYPE_CPE)) *
203 uint8_t (*layout_map)[3],
int offset, uint64_t
left,
204 uint64_t right,
int pos, uint64_t *
layout)
208 .av_position =
left | right,
210 .elem_id = layout_map[
offset][1],
213 if (e2c_vec[
offset].av_position != UINT64_MAX)
221 .elem_id = layout_map[
offset][1],
225 .av_position = right,
227 .elem_id = layout_map[
offset + 1][1],
230 if (
left != UINT64_MAX)
233 if (right != UINT64_MAX)
243 int num_pos_channels = 0;
247 for (
i = current;
i < tags;
i++) {
248 if (layout_map[
i][2] !=
pos)
258 num_pos_channels += 2;
269 return num_pos_channels;
273 uint64_t *
layout,
int tags,
int layer,
int pos,
int *current)
275 int i = *current, j = 0;
278 if (nb_channels < 0 || nb_channels > 5)
282 while (nb_channels) {
287 .syn_ele = layout_map[
i][0],
288 .elem_id = layout_map[
i][1],
291 *
layout |= e2c_vec[
i].av_position;
301 while (nb_channels & 1) {
308 .syn_ele = layout_map[
i][0],
309 .elem_id = layout_map[
i][1],
312 *
layout |= e2c_vec[
i].av_position;
318 while (nb_channels >= 2) {
329 while (nb_channels & 1) {
334 .syn_ele = layout_map[
i][0],
335 .elem_id = layout_map[
i][1],
338 *
layout |= e2c_vec[
i].av_position;
352 int i, n, total_non_cc_elements;
359 for (n = 0,
i = 0; n < 3 &&
i < tags; n++) {
374 total_non_cc_elements = n =
i;
392 for (
i = 1;
i < n;
i++)
402 for (
i = 0;
i < total_non_cc_elements;
i++) {
418 ac->
oc[0] = ac->
oc[1];
431 ac->
oc[1] = ac->
oc[0];
446 enum OCStatus oc_type,
int get_new_frame)
452 uint8_t type_counts[
TYPE_END] = { 0 };
455 memcpy(ac->
oc[1].
layout_map, layout_map, tags *
sizeof(layout_map[0]));
458 for (
i = 0;
i < tags;
i++) {
459 int type = layout_map[
i][0];
460 int id = layout_map[
i][1];
469 #if FF_API_OLD_CHANNEL_LAYOUT
478 for (
i = 0;
i < tags;
i++) {
479 int type = layout_map[
i][0];
480 int id = layout_map[
i][1];
481 int iid = id_map[
type][
id];
482 int position = layout_map[
i][2];
526 for (j = 0; j <= 1; j++) {
541 uint8_t (*layout_map)[3],
545 if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
546 channel_config > 14) {
548 "invalid default channel configuration (%d)\n",
554 *tags *
sizeof(*layout_map));
572 " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
597 &layout_map_tags, 2) < 0)
616 layout_map[0][0] = layout_map[1][0] =
TYPE_SCE;
618 layout_map[0][1] = 0;
619 layout_map[1][1] = 1;
664 "This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n",
686 "This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
751 layout_map[0][2] =
type;
757 int reference_position) {
769 uint8_t (*layout_map)[3],
772 int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
782 "Sample rate index in program config element does not "
783 "match the sample rate index configured by the container.\n");
800 if (
get_bits_left(gb) < 5 * (num_front + num_side + num_back + num_cc) + 4 *(num_lfe + num_assoc_data + num_cc)) {
840 int get_bit_alignment,
844 int extension_flag,
ret, ep_config, res_flags;
864 if (channel_config == 0) {
866 tags =
decode_pce(avctx, m4ac, layout_map, gb, get_bit_alignment);
871 &tags, channel_config)))
877 }
else if (m4ac->
sbr == 1 && m4ac->
ps == -1)
883 if (extension_flag) {
896 "AAC data resilience (flags %x)",
912 "epConfig %d", ep_config);
924 int ret, ep_config, res_flags;
927 const int ELDEXT_TERM = 0;
936 "AAC data resilience (flags %x)",
947 while (
get_bits(gb, 4) != ELDEXT_TERM) {
961 &tags, channel_config)))
970 "epConfig %d", ep_config);
992 int get_bit_alignment,
1006 "invalid sampling rate index %d\n",
1014 "invalid low delay sampling rate index %d\n",
1040 "Audio object type %s%d",
1041 m4ac->
sbr == 1 ?
"SBR+" :
"",
1047 "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
1058 const uint8_t *
data, int64_t bit_size,
1064 if (bit_size < 0 || bit_size > INT_MAX) {
1069 ff_dlog(avctx,
"audio specific config size %d\n", (
int)bit_size >> 3);
1070 for (
i = 0; i < bit_size >> 3;
i++)
1090 union {
unsigned u;
int s; } v = { previous_val * 1664525
u + 1013904223 };
1103 if (92017 <= rate)
return 0;
1104 else if (75132 <= rate)
return 1;
1105 else if (55426 <= rate)
return 2;
1106 else if (46009 <= rate)
return 3;
1107 else if (37566 <= rate)
return 4;
1108 else if (27713 <= rate)
return 5;
1109 else if (23004 <= rate)
return 6;
1110 else if (18783 <= rate)
return 7;
1111 else if (13856 <= rate)
return 8;
1112 else if (11502 <= rate)
return 9;
1113 else if (9391 <= rate)
return 10;
1184 int layout_map_tags;
1227 #define MDCT_INIT(s, fn, len, sval) \
1229 ret = av_tx_init(&s, &fn, TX_TYPE, 1, len, &scale, 0); \
1279 "Invalid Predictor Reset Group.\n");
1325 "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
1338 for (
i = 0;
i < 7;
i++) {
1395 "Prediction is not allowed in AAC-LC.\n");
1400 "LTP in ER AAC LD not yet implemented.\n");
1412 "Number of scalefactor bands in group (%d) "
1413 "exceeds limit (%d).\n",
1440 while (k < ics->max_sfb) {
1441 uint8_t sect_end = k;
1443 int sect_band_type =
get_bits(gb, 4);
1444 if (sect_band_type == 12) {
1450 sect_end += sect_len_incr;
1455 if (sect_end > ics->
max_sfb) {
1457 "Number of bands (%d) exceeds limit (%d).\n",
1461 }
while (sect_len_incr == (1 <<
bits) - 1);
1462 for (; k < sect_end; k++) {
1463 band_type [idx] = sect_band_type;
1464 band_type_run_end[idx++] = sect_end;
1482 unsigned int global_gain,
1485 int band_type_run_end[120])
1493 int run_end = band_type_run_end[idx];
1494 if (band_type[idx] ==
ZERO_BT) {
1495 for (;
i < run_end;
i++, idx++)
1499 for (;
i < run_end;
i++, idx++) {
1502 if (
offset[2] != clipped_offset) {
1504 "If you heard an audible artifact, there may be a bug in the decoder. "
1505 "Clipped intensity stereo position (%d -> %d)",
1506 offset[2], clipped_offset);
1509 sf[idx] = 100 - clipped_offset;
1514 }
else if (band_type[idx] ==
NOISE_BT) {
1515 for (;
i < run_end;
i++, idx++) {
1516 if (noise_flag-- > 0)
1521 if (
offset[1] != clipped_offset) {
1523 "If you heard an audible artifact, there may be a bug in the decoder. "
1524 "Clipped noise gain (%d -> %d)",
1525 offset[1], clipped_offset);
1528 sf[idx] = -(100 + clipped_offset);
1534 for (;
i < run_end;
i++, idx++) {
1538 "Scalefactor (%d) out of range.\n",
offset[0]);
1557 const uint16_t *swb_offset,
int num_swb)
1562 if (pulse_swb >= num_swb)
1564 pulse->
pos[0] = swb_offset[pulse_swb];
1566 if (pulse->
pos[0] >= swb_offset[num_swb])
1571 if (pulse->
pos[
i] >= swb_offset[num_swb])
1586 int w,
filt,
i, coef_len, coef_res, coef_compress;
1599 "TNS filter order %d is greater than maximum %d.\n",
1607 coef_len = coef_res + 3 - coef_compress;
1608 tmp2_idx = 2 * coef_compress + coef_res;
1631 if (ms_present == 1) {
1632 for (idx = 0; idx < max_idx; idx++)
1634 }
else if (ms_present == 2) {
1653 int pulse_present,
const Pulse *pulse,
1657 int i, k,
g, idx = 0;
1670 const unsigned cbt_m1 = band_type[idx] - 1;
1676 for (group = 0; group < (
AAC_SIGNE)g_len; group++, cfo+=128) {
1677 memset(cfo, 0, off_len *
sizeof(*cfo));
1679 }
else if (cbt_m1 ==
NOISE_BT - 1) {
1680 for (group = 0; group < (
AAC_SIGNE)g_len; group++, cfo+=128) {
1683 for (k = 0; k < off_len; k++) {
1688 band_energy = ac->
fdsp->scalarproduct_fixed(cfo, cfo, off_len);
1694 for (k = 0; k < off_len; k++) {
1711 switch (cbt_m1 >> 1) {
1713 for (group = 0; group < (
AAC_SIGNE)g_len; group++, cfo+=128) {
1727 cf =
VMUL4(cf, vq, cb_idx, sf + idx);
1734 for (group = 0; group < (
AAC_SIGNE)g_len; group++, cfo+=128) {
1747 nnz = cb_idx >> 8 & 15;
1760 for (group = 0; group < (
AAC_SIGNE)g_len; group++, cfo+=128) {
1774 cf =
VMUL2(cf, vq, cb_idx, sf + idx);
1782 for (group = 0; group < (
AAC_SIGNE)g_len; group++, cfo+=128) {
1795 nnz = cb_idx >> 8 & 15;
1796 sign = nnz ?
SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
1801 cf =
VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1808 for (group = 0; group < (
AAC_SIGNE)g_len; group++, cfo+=128) {
1814 uint32_t *icf = (uint32_t *) cf;
1829 if (cb_idx == 0x0000) {
1840 for (j = 0; j < 2; j++) {
1875 unsigned v = ((
const uint32_t*)vq)[cb_idx & 15];
1876 *icf++ = (
bits & 1
U<<31) | v;
1895 if (pulse_present) {
1901 if (band_type[idx] !=
NOISE_BT && sf[idx]) {
1905 ico = co + (co > 0 ? -ico : ico);
1907 coef_base[ pulse->
pos[
i] ] = ico;
1925 const unsigned cbt_m1 = band_type[idx] - 1;
1931 for (group = 0; group < (
int)g_len; group++, cfo+=128) {
1960 k < sce->ics.swb_offset[sfb + 1];
1977 static const uint8_t gain_mode[4][3] = {
1988 uint8_t max_band =
get_bits(gb, 2);
1989 for (bd = 0; bd < max_band; bd++) {
1990 for (wd = 0; wd < gain_mode[
mode][0]; wd++) {
1991 uint8_t adjust_num =
get_bits(gb, 3);
1992 for (ad = 0; ad < adjust_num; ad++) {
1995 : gain_mode[
mode][2]));
2016 int global_gain, eld_syntax, er_syntax, pulse_present = 0;
2032 if (!common_window && !scale_flag) {
2047 if (!eld_syntax && (pulse_present =
get_bits1(gb))) {
2050 "Pulse tool not allowed in eight short sequence.\n");
2056 "Pulse data corrupt or invalid.\n");
2062 if (tns->
present && !er_syntax) {
2076 if (tns->
present && er_syntax) {
2105 int g,
i, group, idx = 0;
2113 for (group = 0; group < ics->
group_len[
g]; group++) {
2114 ac->
fdsp->butterflies_fixed(ch0 + group * 128 +
offsets[
i],
2118 for (group = 0; group < ics->
group_len[
g]; group++) {
2145 int g, group,
i, idx = 0;
2153 for (;
i < bt_run_end;
i++, idx++) {
2158 for (group = 0; group < ics->
group_len[
g]; group++)
2174 idx += bt_run_end -
i;
2190 int i,
ret, common_window, ms_present = 0;
2193 common_window = eld_syntax ||
get_bits1(gb);
2194 if (common_window) {
2205 if (ms_present == 3) {
2208 }
else if (ms_present)
2216 if (common_window) {
2230 1.09050773266525765921,
2231 1.18920711500272106672,
2275 for (
c = 0;
c < num_gain;
c++) {
2285 if ((
abs(gain_cache)-1024) >> 3 > 30)
2290 coup->
gain[
c][0] = gain_cache;
2293 for (sfb = 0; sfb < sce->
ics.
max_sfb; sfb++, idx++) {
2306 if ((
abs(gain_cache)-1024) >> 3 > 30)
2311 coup->
gain[
c][idx] = gain_cache;
2329 int num_excl_chan = 0;
2332 for (
i = 0;
i < 7;
i++)
2336 return num_excl_chan / 7;
2348 int drc_num_bands = 1;
2369 for (
i = 0;
i < drc_num_bands;
i++) {
2382 for (
i = 0;
i < drc_num_bands;
i++) {
2393 int i, major, minor;
2400 for(
i=0;
i+1<
sizeof(buf) &&
len>=8;
i++,
len-=8)
2407 if (sscanf(buf,
"libfaac %d.%d", &major, &minor) == 2){
2444 "SBR with 960 frame length");
2499 int bottom, top, order, start, end,
size, inc;
2521 if ((
size = end - start) <= 0)
2533 for (m = 0; m <
size; m++, start += inc)
2534 for (
i = 1;
i <=
FFMIN(m, order);
i++)
2538 for (m = 0; m <
size; m++, start += inc) {
2539 tmp[0] = coef[start];
2540 for (
i = 1;
i <=
FFMIN(m, order);
i++)
2542 for (
i = order;
i > 0;
i--)
2565 memset(in, 0, 448 *
sizeof(*in));
2572 memset(in + 1024 + 576, 0, 448 *
sizeof(*in));
2589 int16_t num_samples = 2048;
2591 if (ltp->
lag < 1024)
2592 num_samples = ltp->
lag + 1024;
2593 for (
i = 0;
i < num_samples;
i++)
2595 memset(&predTime[
i], 0, (2048 -
i) *
sizeof(*predTime));
2622 memcpy(saved_ltp, saved, 512 *
sizeof(*saved_ltp));
2623 memset(saved_ltp + 576, 0, 448 *
sizeof(*saved_ltp));
2626 for (
i = 0;
i < 64;
i++)
2629 memcpy(saved_ltp, ac->
buf_mdct + 512, 448 *
sizeof(*saved_ltp));
2630 memset(saved_ltp + 576, 0, 448 *
sizeof(*saved_ltp));
2633 for (
i = 0;
i < 64;
i++)
2638 for (
i = 0;
i < 512;
i++)
2665 for (
i = 0;
i < 1024;
i += 128)
2681 memcpy(
out, saved, 448 *
sizeof(*
out));
2689 memcpy(
out + 448 + 4*128,
temp, 64 *
sizeof(*
out));
2692 memcpy(
out + 576, buf + 64, 448 *
sizeof(*
out));
2698 memcpy( saved,
temp + 64, 64 *
sizeof(*saved));
2702 memcpy( saved + 448, buf + 7*128 + 64, 64 *
sizeof(*saved));
2704 memcpy( saved, buf + 512, 448 *
sizeof(*saved));
2705 memcpy( saved + 448, buf + 7*128 + 64, 64 *
sizeof(*saved));
2707 memcpy( saved, buf + 512, 512 *
sizeof(*saved));
2729 for (
i = 0;
i < 8;
i++)
2746 memcpy(
out, saved, 420 *
sizeof(*
out));
2754 memcpy(
out + 420 + 4*120,
temp, 60 *
sizeof(*
out));
2757 memcpy(
out + 540, buf + 60, 420 *
sizeof(*
out));
2763 memcpy( saved,
temp + 60, 60 *
sizeof(*saved));
2767 memcpy( saved + 420, buf + 7*120 + 60, 60 *
sizeof(*saved));
2769 memcpy( saved, buf + 480, 420 *
sizeof(*saved));
2770 memcpy( saved + 420, buf + 7*120 + 60, 60 *
sizeof(*saved));
2772 memcpy( saved, buf + 480, 480 *
sizeof(*saved));
2789 memcpy(
out, saved, 192 *
sizeof(*
out));
2791 memcpy(
out + 320, buf + 64, 192 *
sizeof(*
out));
2797 memcpy(saved, buf + 256, 256 *
sizeof(*saved));
2808 const int n2 = n >> 1;
2809 const int n4 = n >> 2;
2818 for (
i = 0;
i < n2;
i+=2) {
2820 temp = in[
i ]; in[
i ] = -in[n - 1 -
i]; in[n - 1 -
i] =
temp;
2821 temp = -in[
i + 1]; in[
i + 1] = in[n - 2 -
i]; in[n - 2 -
i] =
temp;
2829 for (
i = 0;
i < n;
i+=2) {
2840 for (
i = n4;
i < n2;
i ++) {
2846 for (
i = 0;
i < n2;
i ++) {
2852 for (
i = 0;
i < n4;
i ++) {
2859 memmove(saved + n, saved, 2 * n *
sizeof(*saved));
2860 memcpy( saved, buf, n *
sizeof(*saved));
2885 apply_coupling_method(ac, &cc->
ch[0], cce,
index);
2890 apply_coupling_method(ac, &cc->
ch[1], cce,
index++);
2978 int layout_map_tags,
ret;
2986 "More than one AAC RDB per ADTS frame");
3009 layout_map_tags = 2;
3010 layout_map[0][0] = layout_map[1][0] =
TYPE_SCE;
3012 layout_map[0][1] = 0;
3013 layout_map[1][1] = 1;
3060 if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
3068 if (!(che=
get_che(ac, elem_type, elem_id))) {
3070 "channel element %d.%d is not allocated\n",
3071 elem_type, elem_id);
3077 switch (elem_type) {
3115 int samples = 0, multiplier, audio_found = 0, pce_found = 0;
3116 int is_dmono, sce_count = 0;
3117 int payload_alignment;
3156 if (che_presence[elem_type][elem_id]) {
3157 int error = che_presence[elem_type][elem_id] > 1;
3159 elem_type, elem_id);
3165 che_presence[elem_type][elem_id]++;
3167 if (!(che=
get_che(ac, elem_type, elem_id))) {
3169 elem_type, elem_id);
3177 switch (elem_type) {
3208 if (pce_found && !pushed) {
3221 "Not evaluating a further program_config_element as this construct is dubious at best.\n");
3241 while (elem_id > 0) {
3258 che_prev_type = elem_type;
3281 if (ac->
oc[1].
status && audio_found) {
3301 is_dmono = ac->
dmono_mode && sce_count == 2 &&
3318 int *got_frame_ptr,
AVPacket *avpkt)
3321 const uint8_t *buf = avpkt->
data;
3322 int buf_size = avpkt->
size;
3327 size_t new_extradata_size;
3330 &new_extradata_size);
3331 size_t jp_dualmono_size;
3336 if (new_extradata) {
3341 new_extradata_size * 8LL, 1);
3348 if (jp_dualmono && jp_dualmono_size > 0)
3353 if (INT_MAX / 8 <= buf_size)
3373 for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
3374 if (buf[buf_offset])
3377 return buf_size > buf_offset ? buf_consumed : buf_size;
3426 #define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
3428 {
"dual_mono_mode",
"Select the channel to decode for dual mono",
3437 {
"channel_order",
"Order in which the channels are to be exported",
3442 {
"coded",
"order in which the channels are coded in the bitstream",
static void error(const char *err)
enum RawDataBlockType type[8]
Type of channel element to be coupled - SCE or CPE.
static void vector_pow43(int *coefs, int len)
int frame_size
Number of samples per channel in an audio frame.
av_cold int avpriv_kbd_window_init(float *window, float alpha, int n)
Generate a Kaiser-Bessel Derived Window.
CouplingPoint
The point during decoding at which channel coupling is applied.
#define FF_ENABLE_DEPRECATION_WARNINGS
@ AV_SAMPLE_FMT_FLTP
float, planar
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
#define AV_LOG_WARNING
Something somehow does not look correct.
static av_always_inline int lcg_random(unsigned previous_val)
linear congruential pseudorandom number generator
const uint8_t ff_tns_max_bands_128[]
#define AV_EF_EXPLODE
abort decoding on minor error detection
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
Mid/Side stereo decoding; reference: 4.6.8.1.3.
static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx, GetBitContext *gb, MPEG4AudioConfig *m4ac, int channel_config)
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
Update the LTP buffer for next frame.
static int get_bits_left(GetBitContext *gb)
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
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
Conduct IMDCT and windowing.
#define AV_CHANNEL_LAYOUT_STEREO
int sample_rate
samples per second
#define u(width, name, range_min, range_max)
const float *const ff_aac_codebook_vector_vals[]
@ AV_PKT_DATA_NEW_EXTRADATA
The AV_PKT_DATA_NEW_EXTRADATA is used to notify the codec or the format that the extradata buffer was...
static int decode_fill(AACContext *ac, GetBitContext *gb, int len)
static av_cold int aac_decode_init(AVCodecContext *avctx)
static INTFLOAT aac_kbd_short_120[120]
int skip_samples
Number of audio samples to skip at the start of the next decoded frame.
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
#define GET_VLC(code, name, gb, table, bits, max_depth)
If the vlc code is invalid and max_depth=1, then no bits will be removed.
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics, GetBitContext *gb)
void ff_cbrt_tableinit(void)
void(* vector_fmul_reverse)(float *dst, const float *src0, const float *src1, int len)
Calculate the entry wise product of two vectors of floats, and store the result in a vector of floats...
static int get_bits_count(const GetBitContext *s)
void(* subband_scale)(int *dst, int *src, int scale, int offset, int len, void *log_context)
This structure describes decoded (raw) audio or video data.
static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
Skip data_stream_element; reference: table 4.10.
static void apply_dependent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply dependent channel coupling (applied before IMDCT).
static int AAC_RENAME() compute_lpc_coefs(const LPC_TYPE *autoc, int max_order, LPC_TYPE *lpc, int lpc_stride, int fail, int normalize)
Levinson-Durbin recursion.
const uint8_t ff_aac_num_swb_960[]
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb, int ms_present)
Decode Mid/Side data; reference: table 4.54.
@ AOT_ER_AAC_LTP
N Error Resilient Long Term Prediction.
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
ChannelElement * tag_che_map[4][MAX_ELEM_ID]
#define FF_COMPLIANCE_STRICT
Strictly conform to all the things in the spec no matter what consequences.
const uint8_t ff_aac_num_swb_120[]
#define AV_LOG_VERBOSE
Detailed information.
int8_t used[MAX_LTP_LONG_SFB]
static int decode_band_types(AACContext *ac, enum BandType band_type[120], int band_type_run_end[120], GetBitContext *gb, IndividualChannelStream *ics)
Decode band types (section_data payload); reference: table 4.46.
static int * DEC_SQUAD(int *dst, unsigned idx)
static AVOnce aac_table_init
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
Apply AAC-Main style frequency domain prediction.
#define UPDATE_CACHE(name, gb)
enum AVChannelOrder order
Channel order used in this layout.
const uint8_t ff_aac_num_swb_480[]
static void pop_output_configuration(AACContext *ac)
Restore the previous output configuration if and only if the current configuration is unlocked.
static int assign_channels(struct elem_to_channel e2c_vec[MAX_ELEM_ID], uint8_t(*layout_map)[3], uint64_t *layout, int tags, int layer, int pos, int *current)
INTFLOAT * ret
PCM output.
const VLCElem * ff_vlc_spectral[11]
#define AVERROR_UNKNOWN
Unknown error, typically from an external library.
int nb_channels
Number of channels in this layout.
static int decode_spectrum_and_dequant(AACContext *ac, INTFLOAT coef[1024], GetBitContext *gb, const INTFLOAT sf[120], int pulse_present, const Pulse *pulse, const IndividualChannelStream *ics, enum BandType band_type[120])
Decode spectral data; reference: table 4.50.
const uint16_t *const ff_swb_offset_128[]
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
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 int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac, uint8_t(*layout_map)[3], GetBitContext *gb, int byte_align_ref)
Decode program configuration element; reference: table 4.2.
#define FF_DEBUG_PICT_INFO
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
const uint8_t ff_tns_max_bands_1024[]
void(* butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len)
Calculate the sum and difference of two vectors of floats.
static void reset_all_predictors(PredictorState *ps)
#define GET_CACHE(name, gb)
static int set_default_channel_config(AACContext *ac, AVCodecContext *avctx, uint8_t(*layout_map)[3], int *tags, int channel_config)
Set up channel positions based on a default channel configuration as specified in table 1....
static void skip_bits(GetBitContext *s, int n)
Dynamic Range Control - decoded from the bitstream but not processed further.
int num_swb
number of scalefactor window bands
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
enum CouplingPoint coupling_point
The point during decoding at which coupling is applied.
static SDL_Window * window
const uint8_t ff_aac_num_swb_512[]
@ OC_LOCKED
Output configuration locked in place.
void(* apply_ltp)(AACContext *ac, SingleChannelElement *sce)
INTFLOAT saved[1536]
overlap
static const INTFLOAT ltp_coef[8]
INTFLOAT ret_buf[2048]
PCM output buffer.
AVChannelLayout ch_layout
Audio channel layout.
int id_select[8]
element id
#define AV_EF_BITSTREAM
detect bitstream specification deviations
static int count_paired_channels(uint8_t(*layout_map)[3], int tags, int pos, int current)
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext *gb)
Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4....
int flags
AV_CODEC_FLAG_*.
#define POW_SF2_ZERO
ff_aac_pow2sf_tab index corresponding to pow(2, 0);
static void decode_gain_control(SingleChannelElement *sce, GetBitContext *gb)
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 type
static float * VMUL2(float *dst, const float *v, unsigned idx, const float *scale)
static av_always_inline float scale(float x, float s)
static void decode_ltp(LongTermPrediction *ltp, GetBitContext *gb, uint8_t max_sfb)
Decode Long Term Prediction data; reference: table 4.xx.
AVFixedDSPContext * avpriv_alloc_fixed_dsp(int bit_exact)
Allocate and initialize a fixed DSP context.
static __device__ float fabsf(float a)
uint8_t prediction_used[41]
IndividualChannelStream ics
static int aac_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
static void spectral_to_sample(AACContext *ac, int samples)
Convert spectral data to samples, applying all supported tools as appropriate.
@ AOT_ER_AAC_LC
N Error Resilient Low Complexity.
static int ff_thread_once(char *control, void(*routine)(void))
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
@ ZERO_BT
Scalefactors and spectral data are all zero.
#define FF_ARRAY_ELEMS(a)
static av_cold int che_configure(AACContext *ac, enum ChannelPosition che_pos, int type, int id, int *channels)
Check for the channel element in the current channel position configuration.
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
Decode a channel_pair_element; reference: table 4.4.
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
#define MDCT_INIT(s, fn, len, sval)
int exclude_mask[MAX_CHANNELS]
Channels to be excluded from DRC processing.
void(* vector_pow43)(int *coefs, int len)
#define AV_CH_LAYOUT_22POINT2
#define CLOSE_READER(name, gb)
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
@ NOISE_BT
Spectral data are scaled white noise not coded in the bitstream.
float(* scalarproduct_float)(const float *v1, const float *v2, int len)
Calculate the scalar product of two vectors of floats.
@ AOT_ER_AAC_LD
N Error Resilient Low Delay.
static const AVClass aac_decoder_class
@ OC_TRIAL_FRAME
Output configuration under trial specified by a frame header.
INTFLOAT coeffs[1024]
coefficients for IMDCT, maybe processed
static void windowing_and_mdct_ltp(AACContext *ac, INTFLOAT *out, INTFLOAT *in, IndividualChannelStream *ics)
Apply windowing and MDCT to obtain the spectral coefficient from the predicted sample by LTP.
const uint16_t *const ff_swb_offset_960[]
static int sample_rate_idx(int rate)
static av_always_inline void reset_predict_state(PredictorState *ps)
static const int offsets[]
int num_coupled
number of target elements
static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext *gb)
Decode dynamic range information; reference: table 4.52.
@ AV_CHANNEL_ORDER_UNSPEC
Only the channel count is specified, without any further information about the channel order.
FF_ENABLE_DEPRECATION_WARNINGS int av_channel_layout_from_mask(AVChannelLayout *channel_layout, uint64_t mask)
Initialize a native channel layout from a bitmask indicating which channels are present.
@ OC_NONE
Output unconfigured.
@ INTENSITY_BT2
Scalefactor data are intensity stereo positions (out of phase).
static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present)
intensity stereo decoding; reference: 4.6.8.2.3
#define av_assert0(cond)
assert() equivalent, that is always enabled.
static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
#define SKIP_BITS(name, gb, num)
static int aac_decode_er_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, GetBitContext *gb)
static int decode_scalefactors(AACContext *ac, INTFLOAT sf[120], GetBitContext *gb, unsigned int global_gain, IndividualChannelStream *ics, enum BandType band_type[120], int band_type_run_end[120])
Decode scalefactors; reference: table 4.47.
int force_dmono_mode
0->not dmono, 1->use first channel, 2->use second channel
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
Individual Channel Stream.
#define SCALE_DIFF_ZERO
codebook index corresponding to zero scalefactor indices difference
INTFLOAT coef[8][4][TNS_MAX_ORDER]
#define NOISE_PRE
preamble for NOISE_BT, put in bitstream with the first noise band
const uint16_t * swb_offset
table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular wind...
static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx, GetBitContext *gb, int get_bit_alignment, MPEG4AudioConfig *m4ac, int channel_config)
Decode GA "General Audio" specific configuration; reference: table 4.1.
int warned_num_aac_frames
@ INTENSITY_BT
Scalefactor data are intensity stereo positions (in phase).
#define LIBAVUTIL_VERSION_INT
Describe the class of an AVClass context structure.
static int decode_pulses(Pulse *pulse, GetBitContext *gb, const uint16_t *swb_offset, int num_swb)
Decode pulse data; reference: table 4.7.
static void flush(AVCodecContext *avctx)
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
enum AACOutputChannelOrder output_channel_order
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
const float ff_aac_eld_window_480[1800]
static void imdct_and_windowing_960(AACContext *ac, SingleChannelElement *sce)
Conduct IMDCT and windowing.
const uint8_t ff_aac_num_swb_128[]
static void decode_channel_map(uint8_t layout_map[][3], enum ChannelPosition type, GetBitContext *gb, int n)
Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
struct AVCodecInternal * internal
Private context used for internal data.
const int8_t ff_tags_per_config[16]
int AAC_RENAME() ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr, int id_aac)
Initialize one SBR context.
const char * av_default_item_name(void *ptr)
Return the context name.
static unsigned int get_bits1(GetBitContext *s)
static av_cold int aac_decode_close(AVCodecContext *avctx)
#define LAST_SKIP_BITS(name, gb, num)
static __device__ float sqrtf(float a)
int band_top[17]
Indicates the top of the i-th DRC band in units of 4 spectral lines.
ChannelElement * che[4][MAX_ELEM_ID]
static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
Apply the long term prediction.
const uint16_t *const ff_swb_offset_480[]
#define AV_CH_FRONT_CENTER
PredictorState predictor_state[MAX_PREDICTORS]
static av_always_inline int get_vlc2(GetBitContext *s, const VLCElem *table, int bits, int max_depth)
Parse a vlc code.
static int decode_audio_specific_config(AACContext *ac, AVCodecContext *avctx, MPEG4AudioConfig *m4ac, const uint8_t *data, int64_t bit_size, int sync_extension)
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
Decode coupling_channel_element; reference: table 4.8.
static void apply_channel_coupling(AACContext *ac, ChannelElement *cc, enum RawDataBlockType type, int elem_id, enum CouplingPoint coupling_point, void(*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
channel coupling transformation interface
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 band_type_run_end[120]
band type run end points
const uint8_t ff_tns_max_bands_512[]
static float * VMUL2S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)
uint8_t layout_map[MAX_ELEM_ID *4][3]
void(* apply_tns)(INTFLOAT coef[1024], TemporalNoiseShaping *tns, IndividualChannelStream *ics, int decode)
static float * VMUL4(float *dst, const float *v, unsigned idx, const float *scale)
static float * VMUL4S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)
const uint8_t ff_aac_pred_sfb_max[]
static int decode_audio_specific_config_gb(AACContext *ac, AVCodecContext *avctx, MPEG4AudioConfig *m4ac, GetBitContext *gb, int get_bit_alignment, int sync_extension)
Decode audio specific configuration; reference: table 1.13.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
@ AOT_ER_AAC_SCALABLE
N Error Resilient Scalable.
SingleChannelElement ch[2]
const uint16_t *const ff_swb_offset_1024[]
static void relative_align_get_bits(GetBitContext *gb, int reference_position)
@ AOT_AAC_SCALABLE
N Scalable.
An AVChannelLayout holds information about the channel layout of audio data.
void(* vector_fmul)(float *dst, const float *src0, const float *src1, int len)
Calculate the entry wise product of two vectors of floats and store the result in a vector of floats.
const int16_t ff_aac_channel_map[3][4][6]
void ff_aac_float_common_init(void)
static INTFLOAT sine_120[120]
int warned_remapping_once
static int decode_ics(AACContext *ac, SingleChannelElement *sce, GetBitContext *gb, int common_window, int scale_flag)
Decode an individual_channel_stream payload; reference: table 4.44.
int sample_rate
Sample rate of the audio data.
static void noise_scale(int *coefs, int scale, int band_energy, int len)
static INTFLOAT sine_960[960]
enum AVSampleFormat sample_fmt
audio sample format
uint32_t ff_cbrt_tab[1<< 13]
int prog_ref_level
A reference level for the long-term program audio level for all channels combined.
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
OCStatus
Output configuration status.
int ff_mpeg4audio_get_config_gb(MPEG4AudioConfig *c, GetBitContext *gb, int sync_extension, void *logctx)
Parse MPEG-4 systems extradata from a potentially unaligned GetBitContext to retrieve audio configura...
static int push_output_configuration(AACContext *ac)
Save current output configuration if and only if it has been locked.
const uint8_t ff_tns_max_bands_480[]
#define OPEN_READER(name, gb)
static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID], uint8_t(*layout_map)[3], int offset, uint64_t left, uint64_t right, int pos, uint64_t *layout)
const uint16_t *const ff_swb_offset_512[]
VLCElem ff_vlc_scalefactors[352]
void AAC_RENAME() ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac, INTFLOAT *L, INTFLOAT *R)
Apply one SBR element to one AAC element.
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 output_configure(AACContext *ac, uint8_t layout_map[MAX_ELEM_ID *4][3], int tags, enum OCStatus oc_type, int get_new_frame)
Configure output channel order based on the current program configuration element.
@ AV_CHAN_UNUSED
Channel is empty can be safely skipped.
uint8_t ms_mask[128]
Set if mid/side stereo is used for each scalefactor window band.
int dyn_rng_ctl[17]
DRC magnitude information.
av_cold void av_tx_uninit(AVTXContext **ctx)
Frees a context and sets *ctx to NULL, does nothing when *ctx == NULL.
int av_channel_layout_compare(const AVChannelLayout *chl, const AVChannelLayout *chl1)
Check whether two channel layouts are semantically the same, i.e.
#define AV_LOG_INFO
Standard information.
void ff_sine_window_init(float *window, int n)
Generate a sine window.
@ OC_GLOBAL_HDR
Output configuration set in a global header but not yet locked.
@ AOT_AAC_SSR
N (code in SoC repo) Scalable Sample Rate.
static INTFLOAT aac_kbd_long_960[960]
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 layout
static av_cold void aac_static_table_init(void)
int nb_samples
number of audio samples (per channel) described by this frame
static void aacdec_init(AACContext *ac)
Single Channel Element - used for both SCE and LFE elements.
#define i(width, name, range_min, range_max)
static av_cold void init_sine_windows_fixed(void)
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
static const AVOption options[]
SpectralBandReplication sbr
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
static unsigned int show_bits(GetBitContext *s, int n)
Show 1-25 bits.
uint8_t * av_packet_get_side_data(const AVPacket *pkt, enum AVPacketSideDataType type, size_t *size)
Get side information from packet.
const float ff_aac_eld_window_512[1920]
uint8_t ** extended_data
pointers to the data planes/channels.
channel element - generic struct for SCE/CPE/CCE/LFE
@ AOT_ER_AAC_ELD
N Error Resilient Enhanced Low Delay.
static void apply_independent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply independent channel coupling (applied after IMDCT).
static av_always_inline void predict(PredictorState *ps, float *coef, int output_enable)
#define AV_CH_LAYOUT_NATIVE
Channel mask value used for AVCodecContext.request_channel_layout to indicate that the user requests ...
#define NOISE_PRE_BITS
length of preamble
#define FF_DEBUG_STARTCODE
static av_always_inline float cbrtf(float x)
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics, GetBitContext *gb)
Decode Individual Channel Stream info; reference: table 4.6.
void av_frame_unref(AVFrame *frame)
Unreference all the buffers referenced by frame and reset the frame fields.
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
const uint8_t ff_aac_channel_layout_map[16][16][3]
OutputConfiguration oc[2]
static const int8_t filt[NUMTAPS *2]
static void apply_tns(INTFLOAT coef_param[1024], TemporalNoiseShaping *tns, IndividualChannelStream *ics, int decode)
Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4....
static void reset_predictor_group(PredictorState *ps, int group_num)
@ OC_TRIAL_PCE
Output configuration under trial specified by an inband PCE.
static void subband_scale(int *dst, int *src, int scale, int offset, int len, void *log_context)
DynamicRangeControl che_drc
const uint16_t *const ff_swb_offset_120[]
@ AOT_ER_BSAC
N Error Resilient Bit-Sliced Arithmetic Coding.
int pce_instance_tag
Indicates with which program the DRC info is associated.
@ AV_PKT_DATA_JP_DUALMONO
An AV_PKT_DATA_JP_DUALMONO side data packet indicates that the packet may contain "dual mono" audio s...
const uint8_t ff_aac_num_swb_1024[]
#define FFSWAP(type, a, b)
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
INTFLOAT sf[120]
scalefactors
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
SingleChannelElement * output_element[MAX_CHANNELS]
Points to each SingleChannelElement.
#define AACDEC_FLAGS
AVOptions for Japanese DTV specific extensions (ADTS only)
static const uint8_t * align_get_bits(GetBitContext *s)
void(* update_ltp)(AACContext *ac, SingleChannelElement *sce)
int ch_select[8]
[0] shared list of gains; [1] list of gains for right channel; [2] list of gains for left channel; [3...
Tag MUST be and< 10hcoeff half pel interpolation filter coefficients, hcoeff[0] are the 2 middle coefficients[1] are the next outer ones and so on, resulting in a filter like:...eff[2], hcoeff[1], hcoeff[0], hcoeff[0], hcoeff[1], hcoeff[2] ... the sign of the coefficients is not explicitly stored but alternates after each coeff and coeff[0] is positive, so ...,+,-,+,-,+,+,-,+,-,+,... hcoeff[0] is not explicitly stored but found by subtracting the sum of all stored coefficients with signs from 32 hcoeff[0]=32 - hcoeff[1] - hcoeff[2] - ... a good choice for hcoeff and htaps is htaps=6 hcoeff={40,-10, 2} an alternative which requires more computations at both encoder and decoder side and may or may not be better is htaps=8 hcoeff={42,-14, 6,-2}ref_frames minimum of the number of available reference frames and max_ref_frames for example the first frame after a key frame always has ref_frames=1spatial_decomposition_type wavelet type 0 is a 9/7 symmetric compact integer wavelet 1 is a 5/3 symmetric compact integer wavelet others are reserved stored as delta from last, last is reset to 0 if always_reset||keyframeqlog quality(logarithmic quantizer scale) stored as delta from last, last is reset to 0 if always_reset||keyframemv_scale stored as delta from last, last is reset to 0 if always_reset||keyframe FIXME check that everything works fine if this changes between framesqbias dequantization bias stored as delta from last, last is reset to 0 if always_reset||keyframeblock_max_depth maximum depth of the block tree stored as delta from last, last is reset to 0 if always_reset||keyframequant_table quantization tableHighlevel bitstream structure:==============================--------------------------------------------|Header|--------------------------------------------|------------------------------------|||Block0||||split?||||yes no||||......... intra?||||:Block01 :yes no||||:Block02 :....... ..........||||:Block03 ::y DC ::ref index:||||:Block04 ::cb DC ::motion x :||||......... :cr DC ::motion y :||||....... ..........|||------------------------------------||------------------------------------|||Block1|||...|--------------------------------------------|------------ ------------ ------------|||Y subbands||Cb subbands||Cr subbands||||--- ---||--- ---||--- ---|||||LL0||HL0||||LL0||HL0||||LL0||HL0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||LH0||HH0||||LH0||HH0||||LH0||HH0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HL1||LH1||||HL1||LH1||||HL1||LH1|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HH1||HL2||||HH1||HL2||||HH1||HL2|||||...||...||...|||------------ ------------ ------------|--------------------------------------------Decoding process:=================------------|||Subbands|------------||||------------|Intra DC||||LL0 subband prediction ------------|\ Dequantization ------------------- \||Reference frames|\ IDWT|------- -------|Motion \|||Frame 0||Frame 1||Compensation . OBMC v -------|------- -------|--------------. \------> Frame n output Frame Frame<----------------------------------/|...|------------------- Range Coder:============Binary Range Coder:------------------- The implemented range coder is an adapted version based upon "Range encoding: an algorithm for removing redundancy from a digitised message." by G. N. N. Martin. The symbols encoded by the Snow range coder are bits(0|1). The associated probabilities are not fix but change depending on the symbol mix seen so far. bit seen|new state ---------+----------------------------------------------- 0|256 - state_transition_table[256 - old_state];1|state_transition_table[old_state];state_transition_table={ 0, 0, 0, 0, 0, 0, 0, 0, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 37, 38, 39, 40, 41, 42, 43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 56, 57, 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, 70, 71, 72, 73, 74, 75, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, 85, 86, 87, 88, 89, 90, 91, 92, 93, 94, 94, 95, 96, 97, 98, 99, 100, 101, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 114, 115, 116, 117, 118, 119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 133, 134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164, 165, 166, 167, 168, 169, 170, 171, 171, 172, 173, 174, 175, 176, 177, 178, 179, 180, 181, 182, 183, 184, 185, 186, 187, 188, 189, 190, 190, 191, 192, 194, 194, 195, 196, 197, 198, 199, 200, 201, 202, 202, 204, 205, 206, 207, 208, 209, 209, 210, 211, 212, 213, 215, 215, 216, 217, 218, 219, 220, 220, 222, 223, 224, 225, 226, 227, 227, 229, 229, 230, 231, 232, 234, 234, 235, 236, 237, 238, 239, 240, 241, 242, 243, 244, 245, 246, 247, 248, 248, 0, 0, 0, 0, 0, 0, 0};FIXME Range Coding of integers:------------------------- FIXME Neighboring Blocks:===================left and top are set to the respective blocks unless they are outside of the image in which case they are set to the Null block top-left is set to the top left block unless it is outside of the image in which case it is set to the left block if this block has no larger parent block or it is at the left side of its parent block and the top right block is not outside of the image then the top right block is used for top-right else the top-left block is used Null block y, cb, cr are 128 level, ref, mx and my are 0 Motion Vector Prediction:=========================1. the motion vectors of all the neighboring blocks are scaled to compensate for the difference of reference frames scaled_mv=(mv *(256 *(current_reference+1)/(mv.reference+1))+128)> the median of the scaled left
static ChannelElement * get_che(AACContext *ac, int type, int elem_id)
static void imdct_and_window(TwinVQContext *tctx, enum TwinVQFrameType ftype, int wtype, float *in, float *prev, int ch)
@ AV_CHAN_NONE
Invalid channel index.
main external API structure.
int AAC_RENAME() ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr, GetBitContext *gb, int crc, int cnt, int id_aac)
Decode one SBR element.
#define AV_PROFILE_AAC_HE_V2
#define SHOW_UBITS(name, gb, num)
int ps
-1 implicit, 1 presence
void ff_aacdec_init_mips(AACContext *c)
#define NOISE_OFFSET
subtracted from global gain, used as offset for the preamble
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns, GetBitContext *gb, const IndividualChannelStream *ics)
Decode Temporal Noise Shaping data; reference: table 4.48.
enum WindowSequence window_sequence[2]
const uint8_t ff_mpeg4audio_channels[15]
void ff_init_ff_sine_windows(int index)
initialize the specified entry of ff_sine_windows
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
void AAC_RENAME() ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
Close one SBR context.
av_cold void ff_aacdec_common_init_once(void)
int sbr
-1 implicit, 1 presence
Filter the word “frame” indicates either a video frame or a group of audio samples
int band_incr
Number of DRC bands greater than 1 having DRC info.
#define AV_CH_FRONT_RIGHT
int av_channel_layout_copy(AVChannelLayout *dst, const AVChannelLayout *src)
Make a copy of a channel layout.
static int * DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
#define FF_DISABLE_DEPRECATION_WARNINGS
AVChannelLayout ch_layout
void(* windowing_and_mdct_ltp)(AACContext *ac, INTFLOAT *out, INTFLOAT *in, IndividualChannelStream *ics)
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
static int frame_configure_elements(AVCodecContext *avctx)
static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt, ChannelElement *che, enum RawDataBlockType elem_type)
Decode extension data (incomplete); reference: table 4.51.
#define avpriv_request_sample(...)
void(* vector_fmul_window)(float *dst, const float *src0, const float *src1, const float *win, int len)
Overlap/add with window function.
#define AV_PROFILE_AAC_HE
static int count_channels(uint8_t(*layout)[3], int tags)
int dyn_rng_sgn[17]
DRC sign information; 0 - positive, 1 - negative.
This structure stores compressed data.
static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
uint8_t max_sfb
number of scalefactor bands per group
INTFLOAT ltp_state[3072]
time signal for LTP
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
int interpolation_scheme
Indicates the interpolation scheme used in the SBR QMF domain.
static av_always_inline int fixed_sqrt(int x, int bits)
Calculate the square root.
void AAC_RENAME() ff_aac_sbr_init(void)
Initialize SBR.
static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, GetBitContext *gb, const AVPacket *avpkt)
static int * DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
static int * DEC_SPAIR(int *dst, unsigned idx)
static const INTFLOAT *const tns_tmp2_map[4]
float ff_aac_pow2sf_tab[428]
void(* imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce)
enum BandType band_type[128]
band types
uint8_t use_kb_window[2]
If set, use Kaiser-Bessel window, otherwise use a sine window.
@ AOT_AAC_LC
Y Low Complexity.
@ AOT_AAC_LTP
Y Long Term Prediction.
static const float cce_scale[]
int dmono_mode
0->not dmono, 1->use first channel, 2->use second channel
int predictor_reset_group
static uint64_t sniff_channel_order(uint8_t(*layout_map)[3], int tags)
int predictor_initialized