116 static const char overread_err[] =
"Input buffer exhausted before END element found\n";
121 for (i = 0; i < tags; i++) {
122 int syn_ele =
layout[i][0];
124 sum += (1 + (syn_ele ==
TYPE_CPE)) *
144 int type,
int id,
int *channels)
147 if (!ac->
che[type][
id]) {
162 if (ac->
che[type][
id])
172 int type,
id, ch, ret;
175 for (type = 0; type < 4; type++) {
194 for (ch = 0; ch < avctx->
channels; ch++) {
210 uint8_t (*layout_map)[3],
int offset, uint64_t left,
211 uint64_t right,
int pos)
213 if (layout_map[offset][0] ==
TYPE_CPE) {
215 .av_position = left | right,
217 .elem_id = layout_map[offset][1],
225 .elem_id = layout_map[offset][1],
229 .av_position = right,
231 .elem_id = layout_map[offset + 1][1],
241 int num_pos_channels = 0;
245 for (i = *current; i < tags; i++) {
246 if (layout_map[i][2] != pos)
256 num_pos_channels += 2;
267 return num_pos_channels;
272 int i, n, total_non_cc_elements;
274 int num_front_channels, num_side_channels, num_back_channels;
283 if (num_front_channels < 0)
287 if (num_side_channels < 0)
291 if (num_back_channels < 0)
295 if (num_front_channels & 1) {
299 .elem_id = layout_map[i][1],
303 num_front_channels--;
305 if (num_front_channels >= 4) {
310 num_front_channels -= 2;
312 if (num_front_channels >= 2) {
317 num_front_channels -= 2;
319 while (num_front_channels >= 2) {
324 num_front_channels -= 2;
327 if (num_side_channels >= 2) {
332 num_side_channels -= 2;
334 while (num_side_channels >= 2) {
339 num_side_channels -= 2;
342 while (num_back_channels >= 4) {
347 num_back_channels -= 2;
349 if (num_back_channels >= 2) {
354 num_back_channels -= 2;
356 if (num_back_channels) {
360 .elem_id = layout_map[i][1],
371 .elem_id = layout_map[i][1],
380 .elem_id = layout_map[i][1],
387 total_non_cc_elements = n = i;
390 for (i = 1; i < n; i++)
391 if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
399 for (i = 0; i < total_non_cc_elements; i++) {
400 layout_map[i][0] = e2c_vec[i].
syn_ele;
401 layout_map[i][1] = e2c_vec[i].
elem_id;
403 if (e2c_vec[i].av_position != UINT64_MAX) {
416 ac->
oc[0] = ac->
oc[1];
427 ac->
oc[1] = ac->
oc[0];
440 uint8_t layout_map[MAX_ELEM_ID * 4][3],
int tags,
441 enum OCStatus oc_type,
int get_new_frame)
444 int i, channels = 0, ret;
448 memcpy(ac->
oc[1].
layout_map, layout_map, tags *
sizeof(layout_map[0]));
456 for (i = 0; i < tags; i++) {
457 int type = layout_map[i][0];
458 int id = layout_map[i][1];
459 int position = layout_map[i][2];
466 if (ac->
oc[1].
m4ac.
ps == 1 && channels == 2) {
498 if (channel_config < 1 || channel_config > 7) {
500 "invalid default channel configuration (%d)\n",
506 *tags *
sizeof(*layout_map));
520 uint8_t layout_map[MAX_ELEM_ID*4][3];
525 &layout_map_tags, 2) < 0)
537 uint8_t layout_map[MAX_ELEM_ID * 4][3];
542 &layout_map_tags, 1) < 0)
628 layout_map[0][0] = syn_ele;
630 layout_map[0][2] = type;
644 int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
654 "Sample rate index in program config element does not "
655 "match the sample rate index configured by the container.\n");
711 int extension_flag, ret, ep_config, res_flags;
712 uint8_t layout_map[MAX_ELEM_ID*4][3];
728 if (channel_config == 0) {
730 tags =
decode_pce(avctx, m4ac, layout_map, gb);
735 &tags, channel_config)))
741 }
else if (m4ac->
sbr == 1 && m4ac->
ps == -1)
747 if (extension_flag) {
760 "AAC data resilience (flags %x)",
776 "epConfig %d", ep_config);
788 int ret, ep_config, res_flags;
789 uint8_t layout_map[MAX_ELEM_ID*4][3];
791 const int ELDEXT_TERM = 0;
804 "AAC data resilience (flags %x)",
815 while (
get_bits(gb, 4) != ELDEXT_TERM) {
829 &tags, channel_config)))
838 "epConfig %d", ep_config);
874 sync_extension)) < 0)
878 "invalid sampling rate index %d\n",
885 "invalid low delay sampling rate index %d\n",
909 "Audio object type %s%d",
910 m4ac->
sbr == 1 ?
"SBR+" :
"",
916 "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
933 union {
unsigned u;
int s; } v = { previous_val * 1664525u + 1013904223 };
956 if (92017 <= rate)
return 0;
957 else if (75132 <= rate)
return 1;
958 else if (55426 <= rate)
return 2;
959 else if (46009 <= rate)
return 3;
960 else if (37566 <= rate)
return 4;
961 else if (27713 <= rate)
return 5;
962 else if (23004 <= rate)
return 6;
963 else if (18783 <= rate)
return 7;
964 else if (13856 <= rate)
return 8;
965 else if (11502 <= rate)
return 9;
966 else if (9391 <= rate)
return 10;
977 #define AAC_INIT_VLC_STATIC(num, size) \
978 INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
979 ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]), \
980 sizeof(ff_aac_spectral_bits[num][0]), \
981 ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
982 sizeof(ff_aac_spectral_codes[num][0]), \
1003 uint8_t layout_map[MAX_ELEM_ID*4][3];
1004 int layout_map_tags;
1107 "Invalid Predictor Reset Group.\n");
1148 "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
1161 for (i = 0; i < 7; i++) {
1200 "Prediction is not allowed in AAC-LC.\n");
1205 "LTP in ER AAC LD not yet implemented.\n");
1216 "Number of scalefactor bands in group (%d) "
1217 "exceeds limit (%d).\n",
1241 while (k < ics->max_sfb) {
1244 int sect_band_type =
get_bits(gb, 4);
1245 if (sect_band_type == 12) {
1250 sect_len_incr =
get_bits(gb, bits);
1251 sect_end += sect_len_incr;
1256 if (sect_end > ics->
max_sfb) {
1258 "Number of bands (%d) exceeds limit (%d).\n",
1262 }
while (sect_len_incr == (1 << bits) - 1);
1263 for (; k < sect_end; k++) {
1264 band_type [idx] = sect_band_type;
1265 band_type_run_end[idx++] = sect_end;
1283 unsigned int global_gain,
1286 int band_type_run_end[120])
1289 int offset[3] = { global_gain, global_gain - 90, 0 };
1293 for (i = 0; i < ics->
max_sfb;) {
1294 int run_end = band_type_run_end[idx];
1295 if (band_type[idx] ==
ZERO_BT) {
1296 for (; i < run_end; i++, idx++)
1300 for (; i < run_end; i++, idx++) {
1301 offset[2] +=
get_vlc2(gb, vlc_scalefactors.
table, 7, 3) - 60;
1302 clipped_offset = av_clip(offset[2], -155, 100);
1303 if (offset[2] != clipped_offset) {
1305 "If you heard an audible artifact, there may be a bug in the decoder. "
1306 "Clipped intensity stereo position (%d -> %d)",
1307 offset[2], clipped_offset);
1311 }
else if (band_type[idx] ==
NOISE_BT) {
1312 for (; i < run_end; i++, idx++) {
1313 if (noise_flag-- > 0)
1314 offset[1] +=
get_bits(gb, 9) - 256;
1316 offset[1] +=
get_vlc2(gb, vlc_scalefactors.
table, 7, 3) - 60;
1317 clipped_offset = av_clip(offset[1], -100, 155);
1318 if (offset[1] != clipped_offset) {
1320 "If you heard an audible artifact, there may be a bug in the decoder. "
1321 "Clipped noise gain (%d -> %d)",
1322 offset[1], clipped_offset);
1327 for (; i < run_end; i++, idx++) {
1328 offset[0] +=
get_vlc2(gb, vlc_scalefactors.
table, 7, 3) - 60;
1329 if (offset[0] > 255
U) {
1331 "Scalefactor (%d) out of range.\n", offset[0]);
1346 const uint16_t *swb_offset,
int num_swb)
1351 if (pulse_swb >= num_swb)
1353 pulse->
pos[0] = swb_offset[pulse_swb];
1355 if (pulse->
pos[0] > 1023)
1358 for (i = 1; i < pulse->
num_pulse; i++) {
1360 if (pulse->
pos[i] > 1023)
1375 int w, filt, i, coef_len, coef_res, coef_compress;
1382 for (filt = 0; filt < tns->
n_filt[w]; filt++) {
1386 if ((tns->
order[w][filt] =
get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
1388 "TNS filter order %d is greater than maximum %d.\n",
1389 tns->
order[w][filt], tns_max_order);
1390 tns->
order[w][filt] = 0;
1393 if (tns->
order[w][filt]) {
1396 coef_len = coef_res + 3 - coef_compress;
1397 tmp2_idx = 2 * coef_compress + coef_res;
1399 for (i = 0; i < tns->
order[w][filt]; i++)
1419 if (ms_present == 1) {
1424 }
else if (ms_present == 2) {
1430 static inline float *
VMUL2(
float *dst,
const float *v,
unsigned idx,
1434 *dst++ = v[idx & 15] * s;
1435 *dst++ = v[idx>>4 & 15] * s;
1441 static inline float *
VMUL4(
float *dst,
const float *v,
unsigned idx,
1445 *dst++ = v[idx & 3] * s;
1446 *dst++ = v[idx>>2 & 3] * s;
1447 *dst++ = v[idx>>4 & 3] * s;
1448 *dst++ = v[idx>>6 & 3] * s;
1454 static inline float *
VMUL2S(
float *dst,
const float *v,
unsigned idx,
1455 unsigned sign,
const float *scale)
1459 s0.
f = s1.
f = *scale;
1460 s0.
i ^= sign >> 1 << 31;
1463 *dst++ = v[idx & 15] * s0.
f;
1464 *dst++ = v[idx>>4 & 15] * s1.
f;
1471 static inline float *
VMUL4S(
float *dst,
const float *v,
unsigned idx,
1472 unsigned sign,
const float *scale)
1474 unsigned nz = idx >> 12;
1478 t.
i = s.
i ^ (sign & 1
U<<31);
1479 *dst++ = v[idx & 3] * t.
f;
1481 sign <<= nz & 1; nz >>= 1;
1482 t.
i = s.
i ^ (sign & 1
U<<31);
1483 *dst++ = v[idx>>2 & 3] * t.
f;
1485 sign <<= nz & 1; nz >>= 1;
1486 t.
i = s.
i ^ (sign & 1
U<<31);
1487 *dst++ = v[idx>>4 & 3] * t.
f;
1490 t.
i = s.
i ^ (sign & 1
U<<31);
1491 *dst++ = v[idx>>6 & 3] * t.
f;
1511 int pulse_present,
const Pulse *pulse,
1515 int i, k,
g, idx = 0;
1518 float *coef_base = coef;
1521 memset(coef + g * 128 + offsets[ics->
max_sfb], 0,
1522 sizeof(
float) * (c - offsets[ics->
max_sfb]));
1527 for (i = 0; i < ics->
max_sfb; i++, idx++) {
1528 const unsigned cbt_m1 = band_type[idx] - 1;
1529 float *cfo = coef + offsets[
i];
1530 int off_len = offsets[i + 1] - offsets[
i];
1534 for (group = 0; group < g_len; group++, cfo+=128) {
1535 memset(cfo, 0, off_len *
sizeof(
float));
1537 }
else if (cbt_m1 ==
NOISE_BT - 1) {
1538 for (group = 0; group < g_len; group++, cfo+=128) {
1542 for (k = 0; k < off_len; k++) {
1548 scale = sf[idx] / sqrtf(band_energy);
1557 switch (cbt_m1 >> 1) {
1559 for (group = 0; group < g_len; group++, cfo+=128) {
1569 cb_idx = cb_vector_idx[code];
1570 cf =
VMUL4(cf, vq, cb_idx, sf + idx);
1576 for (group = 0; group < g_len; group++, cfo+=128) {
1588 cb_idx = cb_vector_idx[code];
1589 nnz = cb_idx >> 8 & 15;
1592 cf =
VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1598 for (group = 0; group < g_len; group++, cfo+=128) {
1608 cb_idx = cb_vector_idx[code];
1609 cf =
VMUL2(cf, vq, cb_idx, sf + idx);
1616 for (group = 0; group < g_len; group++, cfo+=128) {
1628 cb_idx = cb_vector_idx[code];
1629 nnz = cb_idx >> 8 & 15;
1630 sign = nnz ?
SHOW_UBITS(
re, gb, nnz) << (cb_idx >> 12) : 0;
1632 cf =
VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1638 for (group = 0; group < g_len; group++, cfo+=128) {
1640 uint32_t *icf = (uint32_t *) cf;
1659 cb_idx = cb_vector_idx[code];
1665 for (j = 0; j < 2; j++) {
1687 unsigned v = ((
const uint32_t*)vq)[cb_idx & 15];
1688 *icf++ = (bits & 1
U<<31) | v;
1705 if (pulse_present) {
1707 for (i = 0; i < pulse->
num_pulse; i++) {
1708 float co = coef_base[ pulse->
pos[
i] ];
1709 while (offsets[idx + 1] <= pulse->
pos[i])
1711 if (band_type[idx] !=
NOISE_BT && sf[idx]) {
1712 float ico = -pulse->
amp[
i];
1715 ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1717 coef_base[ pulse->
pos[
i] ] =
cbrtf(fabsf(ico)) * ico * sf[idx];
1728 tmp.
i = (tmp.
i + 0x00008000
U) & 0xFFFF0000U;
1736 tmp.
i = (tmp.
i + 0x00007FFF
U + (tmp.
i & 0x00010000
U >> 16)) & 0xFFFF0000
U;
1744 pun.
i &= 0xFFFF0000
U;
1751 const float a = 0.953125;
1752 const float alpha = 0.90625;
1756 float r0 = ps->
r0, r1 = ps->
r1;
1757 float cor0 = ps->
cor0, cor1 = ps->
cor1;
1758 float var0 = ps->
var0, var1 = ps->
var1;
1760 k1 = var0 > 1 ? cor0 *
flt16_even(a / var0) : 0;
1761 k2 = var1 > 1 ? cor1 *
flt16_even(a / var1) : 0;
1796 k < sce->ics.swb_offset[sfb + 1];
1825 int global_gain, eld_syntax, er_syntax, pulse_present = 0;
1841 if (!common_window && !scale_flag) {
1855 if (!eld_syntax && (pulse_present =
get_bits1(gb))) {
1858 "Pulse tool not allowed in eight short sequence.\n");
1863 "Pulse data corrupt or invalid.\n");
1868 if (tns->
present && !er_syntax)
1877 if (tns->
present && er_syntax)
1900 int g,
i, group, idx = 0;
1903 for (i = 0; i < ics->
max_sfb; i++, idx++) {
1907 for (group = 0; group < ics->
group_len[
g]; group++) {
1909 ch1 + group * 128 + offsets[i],
1910 offsets[i+1] - offsets[i]);
1933 int g, group,
i, idx = 0;
1937 for (i = 0; i < ics->
max_sfb;) {
1941 for (; i < bt_run_end; i++, idx++) {
1942 c = -1 + 2 * (sce1->
band_type[idx] - 14);
1944 c *= 1 - 2 * cpe->
ms_mask[idx];
1945 scale = c * sce1->
sf[idx];
1946 for (group = 0; group < ics->
group_len[
g]; group++)
1948 coef0 + group * 128 + offsets[i],
1950 offsets[i + 1] - offsets[i]);
1954 idx += bt_run_end -
i;
1970 int i, ret, common_window, ms_present = 0;
1973 common_window = eld_syntax ||
get_bits1(gb);
1974 if (common_window) {
1985 if (ms_present == 3) {
1988 }
else if (ms_present)
1991 if ((ret =
decode_ics(ac, &cpe->
ch[0], gb, common_window, 0)))
1993 if ((ret =
decode_ics(ac, &cpe->
ch[1], gb, common_window, 0)))
1996 if (common_window) {
2010 1.09050773266525765921,
2011 1.18920711500272106672,
2046 scale = cce_scale[
get_bits(gb, 2)];
2051 for (c = 0; c < num_gain; c++) {
2055 float gain_cache = 1.0;
2058 gain = cge ?
get_vlc2(gb, vlc_scalefactors.
table, 7, 3) - 60: 0;
2059 gain_cache =
powf(scale, -gain);
2062 coup->
gain[c][0] = gain_cache;
2065 for (sfb = 0; sfb < sce->
ics.
max_sfb; sfb++, idx++) {
2076 gain_cache =
powf(scale, -t) * s;
2079 coup->
gain[c][idx] = gain_cache;
2097 int num_excl_chan = 0;
2100 for (i = 0; i < 7; i++)
2104 return num_excl_chan / 7;
2116 int drc_num_bands = 1;
2137 for (i = 0; i < drc_num_bands; i++) {
2150 for (i = 0; i < drc_num_bands; i++) {
2222 int bottom, top, order, start, end,
size, inc;
2228 for (filt = 0; filt < tns->
n_filt[w]; filt++) {
2231 order = tns->
order[w][filt];
2240 if ((size = end - start) <= 0)
2252 for (m = 0; m <
size; m++, start += inc)
2253 for (i = 1; i <=
FFMIN(m, order); i++)
2254 coef[start] -= coef[start - i * inc] * lpc[i - 1];
2257 for (m = 0; m <
size; m++, start += inc) {
2258 tmp[0] = coef[start];
2259 for (i = 1; i <=
FFMIN(m, order); i++)
2260 coef[start] += tmp[i] * lpc[i - 1];
2261 for (i = order; i > 0; i--)
2262 tmp[i] = tmp[i - 1];
2284 memset(in, 0, 448 *
sizeof(
float));
2291 memset(in + 1024 + 576, 0, 448 *
sizeof(
float));
2306 float *predTime = sce->
ret;
2308 int16_t num_samples = 2048;
2310 if (ltp->
lag < 1024)
2311 num_samples = ltp->
lag + 1024;
2312 for (i = 0; i < num_samples; i++)
2314 memset(&predTime[i], 0, (2048 - i) *
sizeof(
float));
2323 for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
2324 sce->
coeffs[i] += predFreq[i];
2334 float *saved = sce->
saved;
2335 float *saved_ltp = sce->
coeffs;
2341 memcpy(saved_ltp, saved, 512 *
sizeof(
float));
2342 memset(saved_ltp + 576, 0, 448 *
sizeof(
float));
2344 for (i = 0; i < 64; i++)
2345 saved_ltp[i + 512] = ac->
buf_mdct[1023 - i] * swindow[63 - i];
2347 memcpy(saved_ltp, ac->
buf_mdct + 512, 448 *
sizeof(
float));
2348 memset(saved_ltp + 576, 0, 448 *
sizeof(
float));
2350 for (i = 0; i < 64; i++)
2351 saved_ltp[i + 512] = ac->
buf_mdct[1023 - i] * swindow[63 - i];
2354 for (i = 0; i < 512; i++)
2355 saved_ltp[i + 512] = ac->
buf_mdct[1023 - i] * lwindow[511 - i];
2371 float *saved = sce->
saved;
2376 float *temp = ac->
temp;
2381 for (i = 0; i < 1024; i += 128)
2396 memcpy( out, saved, 448 *
sizeof(
float));
2404 memcpy( out + 448 + 4*128, temp, 64 *
sizeof(
float));
2407 memcpy( out + 576, buf + 64, 448 *
sizeof(
float));
2413 memcpy( saved, temp + 64, 64 *
sizeof(
float));
2417 memcpy( saved + 448, buf + 7*128 + 64, 64 *
sizeof(
float));
2419 memcpy( saved, buf + 512, 448 *
sizeof(
float));
2420 memcpy( saved + 448, buf + 7*128 + 64, 64 *
sizeof(
float));
2422 memcpy( saved, buf + 512, 512 *
sizeof(
float));
2431 float *saved = sce->
saved;
2440 memcpy(out, saved, 192 *
sizeof(
float));
2442 memcpy( out + 320, buf + 64, 192 *
sizeof(
float));
2448 memcpy(saved, buf + 256, 256 *
sizeof(
float));
2455 float *saved = sce->
saved;
2460 const int n2 = n >> 1;
2461 const int n4 = n >> 2;
2468 for (i = 0; i < n2; i+=2) {
2470 temp = in[
i ]; in[
i ] = -in[n - 1 -
i]; in[n - 1 -
i] = temp;
2471 temp = -in[i + 1]; in[i + 1] = in[n - 2 -
i]; in[n - 2 -
i] = temp;
2474 for (i = 0; i < n; i+=2) {
2484 for (i = n4; i < n2; i ++) {
2485 out[i - n4] = buf[n2 - 1 -
i] * window[i - n4] +
2486 saved[ i + n2] * window[i + n - n4] +
2487 -saved[ n + n2 - 1 -
i] * window[i + 2*n - n4] +
2488 -saved[2*n + n2 +
i] * window[i + 3*n - n4];
2490 for (i = 0; i < n2; i ++) {
2491 out[n4 +
i] = buf[
i] * window[i + n2 - n4] +
2492 -saved[ n - 1 -
i] * window[i + n2 + n - n4] +
2493 -saved[ n +
i] * window[i + n2 + 2*n - n4] +
2494 saved[2*n + n - 1 -
i] * window[i + n2 + 3*n - n4];
2496 for (i = 0; i < n4; i ++) {
2497 out[n2 + n4 +
i] = buf[ i + n2] * window[i + n - n4] +
2498 -saved[ n2 - 1 -
i] * window[i + 2*n - n4] +
2499 -saved[ n + n2 +
i] * window[i + 3*n - n4];
2503 memmove(saved + n, saved, 2 * n *
sizeof(
float));
2504 memcpy( saved, buf, n *
sizeof(
float));
2518 float *dest = target->
coeffs;
2519 const float *src = cce->
ch[0].
coeffs;
2520 int g,
i, group, k, idx = 0;
2523 "Dependent coupling is not supported together with LTP\n");
2527 for (i = 0; i < ics->
max_sfb; i++, idx++) {
2530 for (group = 0; group < ics->
group_len[
g]; group++) {
2531 for (k = offsets[i]; k < offsets[i + 1]; k++) {
2533 dest[group * 128 + k] += gain * src[group * 128 + k];
2554 const float *src = cce->
ch[0].
ret;
2555 float *dest = target->
ret;
2558 for (i = 0; i <
len; i++)
2559 dest[i] += gain * src[i];
2582 if (coup->
type[c] == type && coup->
id_select[c] == elem_id) {
2584 apply_coupling_method(ac, &cc->
ch[0], cce, index);
2589 apply_coupling_method(ac, &cc->
ch[1], cce, index++);
2614 for (type = 3; type >= 0; type--) {
2658 uint8_t layout_map[MAX_ELEM_ID*4][3];
2659 int layout_map_tags, ret;
2665 "More than one AAC RDB per ADTS frame");
2722 if (chan_config < 0 || chan_config >= 8) {
2730 if (!(che=
get_che(ac, elem_type, elem_id))) {
2732 "channel element %d.%d is not allocated\n",
2733 elem_type, elem_id);
2738 switch (elem_type) {
2770 int samples = 0, multiplier, audio_found = 0, pce_found = 0;
2799 if (!(che=
get_che(ac, elem_type, elem_id))) {
2801 elem_type, elem_id);
2808 switch (elem_type) {
2834 uint8_t layout_map[MAX_ELEM_ID*4][3];
2844 "Not evaluating a further program_config_element as this construct is dubious at best.\n");
2872 elem_type_prev = elem_type;
2887 samples <<= multiplier;
2899 *got_frame_ptr = !!samples;
2908 int *got_frame_ptr,
AVPacket *avpkt)
2912 int buf_size = avpkt->
size;
2917 int new_extradata_size;
2920 &new_extradata_size);
2922 if (new_extradata) {
2929 memcpy(avctx->
extradata, new_extradata, new_extradata_size);
2956 for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
2957 if (buf[buf_offset])
2960 return buf_size > buf_offset ? buf_consumed : buf_size;
2969 for (type = 0; type < 4; type++) {
2970 if (ac->
che[type][i])
2984 #define LOAS_SYNC_WORD 0x2b7
3010 int sync_extension = 0;
3011 int bits_consumed, esize;
3019 if (config_start_bit % 8) {
3021 "Non-byte-aligned audio-specific config");
3027 gb->
buffer + (config_start_bit / 8),
3028 asclen, sync_extension);
3030 if (bits_consumed < 0)
3040 esize = (bits_consumed+7) / 8;
3055 return bits_consumed;
3061 int ret, audio_mux_version =
get_bits(gb, 1);
3064 if (audio_mux_version)
3069 if (audio_mux_version)
3089 if (!audio_mux_version) {
3120 if (audio_mux_version) {
3143 int mux_slot_length = 0;
3146 mux_slot_length += tmp;
3147 }
while (tmp == 255);
3148 return mux_slot_length;
3164 if (!use_same_mux) {
3169 "no decoder config found\n");
3177 }
else if (mux_slot_length_bytes * 8 + 256 <
get_bits_left(gb)) {
3179 "frame length mismatch %d << %d\n",
3189 int *got_frame_ptr,
AVPacket *avpkt)
3204 if (muxlength > avpkt->
size)
3228 "ADTS header detected, probably as result of configuration "
int predictor_initialized
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.
static float * VMUL4S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)
float(* scalarproduct_float)(const float *v1, const float *v2, int len)
Calculate the scalar product of two vectors of floats.
void * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
Conduct IMDCT and windowing.
uint8_t use_kb_window[2]
If set, use Kaiser-Bessel window, otherwise use a sine window.
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
static int decode_audio_specific_config(AACContext *ac, AVCodecContext *avctx, MPEG4AudioConfig *m4ac, const uint8_t *data, int bit_size, int sync_extension)
Decode audio specific configuration; reference: table 1.13.
static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
static const int8_t tags_per_config[16]
static void push_output_configuration(AACContext *ac)
Save current output configuration if and only if it has been locked.
static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present)
intensity stereo decoding; reference: 4.6.8.2.3
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
#define AV_LOG_WARNING
Something somehow does not look correct.
av_cold void ff_kbd_window_init(float *window, float alpha, int n)
Generate a Kaiser-Bessel Derived Window.
static void skip_bits_long(GetBitContext *s, int n)
static float * VMUL2S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)
ChannelElement * che[4][MAX_ELEM_ID]
static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
av_cold void ff_aac_sbr_init(void)
Initialize SBR.
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics, GetBitContext *gb)
Decode Individual Channel Stream info; reference: table 4.6.
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_dlog(ac->avr,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
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)
#define FF_ARRAY_ELEMS(a)
uint8_t ms_mask[128]
Set if mid/side stereo is used for each scalefactor window band.
av_dlog(ac->avr,"%d samples - audio_convert: %s to %s (%s)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt), use_generic?ac->func_descr_generic:ac->func_descr)
static void apply_independent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply independent channel coupling (applied after IMDCT).
static int frame_configure_elements(AVCodecContext *avctx)
static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac, uint8_t(*layout_map)[3], GetBitContext *gb)
Decode program configuration element; reference: table 4.2.
Dynamic Range Control - decoded from the bitstream but not processed further.
float coef[8][4][TNS_MAX_ORDER]
#define FF_PROFILE_AAC_HE_V2
static av_always_inline void predict(PredictorState *ps, float *coef, int output_enable)
enum RawDataBlockType type[8]
Type of channel element to be coupled - SCE or CPE.
Spectral data are scaled white noise not coded in the bitstream.
static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024], GetBitContext *gb, const float sf[120], int pulse_present, const Pulse *pulse, const IndividualChannelStream *ics, enum BandType band_type[120])
Decode spectral data; reference: table 4.50.
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
int band_incr
Number of DRC bands greater than 1 having DRC info.
const uint8_t ff_aac_num_swb_128[]
av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
Close one SBR context.
const uint16_t * swb_offset
table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular wind...
N Error Resilient Long Term Prediction.
static int decode(MimicContext *ctx, int quality, int num_coeffs, int is_iframe)
static av_always_inline int lcg_random(int previous_val)
linear congruential pseudorandom number generator
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
enum AVSampleFormat sample_fmt
audio sample format
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
Apply AAC-Main style frequency domain prediction.
static const uint8_t aac_channel_layout_map[7][5][3]
uint8_t layout_map[MAX_ELEM_ID *4][3]
Output configuration under trial specified by an inband PCE.
#define INIT_VLC_STATIC(vlc, bits, a, b, c, d, e, f, g, static_size)
SingleChannelElement ch[2]
const uint16_t *const ff_swb_offset_512[]
void(* vector_fmul)(float *dst, const float *src0, const float *src1, int len)
Calculate the product of two vectors of floats and store the result in a vector of floats...
N Error Resilient Low Delay.
static VLC vlc_scalefactors
const uint8_t ff_aac_scalefactor_bits[121]
CouplingPoint
The point during decoding at which channel coupling is applied.
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
int num_coupled
number of target elements
#define AV_CH_LOW_FREQUENCY
int exclude_mask[MAX_CHANNELS]
Channels to be excluded from DRC processing.
static int aac_decode_frame_int(AVCodecContext *avctx, void *data, int *got_frame_ptr, GetBitContext *gb)
#define CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr, GetBitContext *gb_host, int crc, int cnt, int id_aac)
Decode Spectral Band Replication extension data; reference: table 4.55.
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.
SingleChannelElement * output_element[MAX_CHANNELS]
Points to each SingleChannelElement.
static int decode_pulses(Pulse *pulse, GetBitContext *gb, const uint16_t *swb_offset, int num_swb)
Decode pulse data; reference: table 4.7.
static int count_paired_channels(uint8_t(*layout_map)[3], int tags, int pos, int *current)
static int get_bits_count(const GetBitContext *s)
Scalefactor data are intensity stereo positions.
bitstream reader API header.
static int read_stream_mux_config(struct LATMContext *latmctx, GetBitContext *gb)
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.
#define CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
int id_select[8]
element id
const float *const ff_aac_codebook_vector_vals[]
static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext *gb)
Decode dynamic range information; reference: table 4.52.
N Error Resilient Low Complexity.
ChannelElement * tag_che_map[4][MAX_ELEM_ID]
Output configuration set in a global header but not yet locked.
AACContext aac_ctx
containing AACContext
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
static uint32_t latm_get_value(GetBitContext *b)
static av_cold int aac_decode_close(AVCodecContext *avctx)
static int get_bits_left(GetBitContext *gb)
void(* vector_fmul_window)(float *dst, const float *src0, const float *src1, const float *win, int len)
Overlap/add with window function.
int dyn_rng_sgn[17]
DRC sign information; 0 - positive, 1 - negative.
float coeffs[1024]
coefficients for IMDCT
#define UPDATE_CACHE(name, gb)
PredictorState predictor_state[MAX_PREDICTORS]
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void av_free(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc(). ...
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.
SpectralBandReplication sbr
enum CouplingPoint coupling_point
The point during decoding at which coupling is applied.
int frame_length_type
0/1 variable/fixed frame length
const uint8_t ff_aac_num_swb_1024[]
FmtConvertContext fmt_conv
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
float ff_aac_kbd_long_1024[1024]
Spectral Band Replication definitions and structures.
void(* mdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
void av_log(void *avcl, int level, const char *fmt,...)
const char * name
Name of the codec implementation.
uint8_t max_sfb
number of scalefactor bands per group
static int latm_decode_frame(AVCodecContext *avctx, void *out, int *got_frame_ptr, AVPacket *avpkt)
void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac, float *L, float *R)
Apply one SBR element to one AAC element.
#define LOAS_SYNC_WORD
11 bits LOAS sync word
AVCodec ff_aac_latm_decoder
#define CLOSE_READER(name, gb)
int num_swb
number of scalefactor window bands
#define AAC_INIT_VLC_STATIC(num, size)
int prog_ref_level
A reference level for the long-term program audio level for all channels combined.
Output configuration locked in place.
uint64_t channel_layout
Audio channel layout.
#define SKIP_BITS(name, gb, num)
#define POW_SF2_ZERO
ff_aac_pow2sf_tab index corresponding to pow(2, 0);
static const float cce_scale[]
#define FF_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
N Error Resilient Scalable.
static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
AAC Spectral Band Replication function declarations.
enum WindowSequence window_sequence[2]
const uint8_t ff_aac_num_swb_512[]
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
static void pop_output_configuration(AACContext *ac)
Restore the previous output configuration if and only if the current configuration is unlocked...
int predictor_reset_group
static void reset_predictor_group(PredictorState *ps, int group_num)
int dyn_rng_ctl[17]
DRC magnitude information.
int initialized
initilized after a valid extradata was seen
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb, int ms_present)
Decode Mid/Side data; reference: table 4.54.
static unsigned int show_bits(GetBitContext *s, int n)
Show 1-25 bits.
const float ff_aac_eld_window[1920]
#define LAST_SKIP_BITS(name, gb, num)
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
AAC definitions and structures.
#define AV_CH_FRONT_LEFT_OF_CENTER
const uint8_t ff_tns_max_bands_1024[]
#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 void cbrt_tableinit(void)
#define AV_CH_FRONT_CENTER
static void apply_dependent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply dependent channel coupling (applied before IMDCT).
int pce_instance_tag
Indicates with which program the DRC info is associated.
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...
#define SHOW_UBITS(name, gb, num)
#define AV_CH_FRONT_RIGHT_OF_CENTER
int interpolation_scheme
Indicates the interpolation scheme used in the SBR QMF domain.
if(ac->has_optimized_func)
#define AVERROR_PATCHWELCOME
Not yet implemented in Libav, patches welcome.
static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx, GetBitContext *gb, MPEG4AudioConfig *m4ac, int channel_config)
Decode GA "General Audio" specific configuration; reference: table 4.1.
int ch_select[8]
[0] shared list of gains; [1] list of gains for right channel; [2] list of gains for left channel; [3...
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
Mid/Side stereo decoding; reference: 4.6.8.1.3.
int frame_size
Number of samples per channel in an audio frame.
int frame_length
frame length for fixed frame length
#define AV_LOG_INFO
Standard information.
Libavcodec external API header.
AVSampleFormat
Audio Sample Formats.
int audio_mux_version_A
LATM syntax version.
int sample_rate
samples per second
float ff_aac_kbd_short_128[128]
static int compute_lpc_coefs(const LPC_TYPE *autoc, int max_order, LPC_TYPE *lpc, int lpc_stride, int fail, int normalize)
Levinson-Durbin recursion.
static uint32_t cbrt_tab[1<< 13]
#define AV_CH_LAYOUT_NATIVE
Channel mask value used for AVCodecContext.request_channel_layout to indicate that the user requests ...
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
Update the LTP buffer for next frame.
main external API structure.
static void(WINAPI *cond_broadcast)(pthread_cond_t *cond)
static void close(AVCodecParserContext *s)
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 ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
#define OPEN_READER(name, gb)
IndividualChannelStream ics
int avpriv_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
Parse AAC frame header.
static av_always_inline float cbrtf(float x)
#define AVERROR_BUG
Bug detected, please report the issue.
static unsigned int get_bits1(GetBitContext *s)
static void skip_bits1(GetBitContext *s)
int sample_rate
Sample rate of the audio data.
static void skip_bits(GetBitContext *s, int n)
static av_cold int latm_decode_init(AVCodecContext *avctx)
static void spectral_to_sample(AACContext *ac)
Convert spectral data to float samples, applying all supported tools as appropriate.
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
#define GET_CACHE(name, gb)
static int read_audio_mux_element(struct LATMContext *latmctx, GetBitContext *gb)
static int latm_decode_audio_specific_config(struct LATMContext *latmctx, GetBitContext *gb, int asclen)
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
Apply the long term prediction.
static float * VMUL2(float *dst, const float *v, unsigned idx, const float *scale)
OCStatus
Output configuration status.
N Error Resilient Bit-Sliced Arithmetic Coding.
av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
Initialize one SBR context.
static void reset_all_predictors(PredictorState *ps)
const uint32_t ff_aac_scalefactor_code[121]
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
void(* imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
static const char overread_err[]
Output configuration under trial specified by a frame header.
const uint8_t ff_tns_max_bands_128[]
static const uint64_t aac_channel_layout[8]
static void imdct_and_window(TwinVQContext *tctx, enum TwinVQFrameType ftype, int wtype, float *in, float *prev, int ch)
float ltp_state[3072]
time signal for LTP
void av_frame_unref(AVFrame *frame)
Unreference all the buffers referenced by frame and reset the frame fields.
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
int band_type_run_end[120]
band type run end points
float sf[120]
scalefactors
#define AV_CH_BACK_CENTER
av_cold void avpriv_float_dsp_init(AVFloatDSPContext *fdsp, int bit_exact)
Initialize a float DSP context.
int band_top[17]
Indicates the top of the i-th DRC band in units of 4 spectral lines.
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns, GetBitContext *gb, const IndividualChannelStream *ics)
Decode Temporal Noise Shaping data; reference: table 4.48.
Scalefactor data are intensity stereo positions.
N Error Resilient Enhanced Low Delay.
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
Decode a channel_pair_element; reference: table 4.4.
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_dlog(ac->avr,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> out
DynamicRangeControl che_drc
static av_always_inline void reset_predict_state(PredictorState *ps)
OutputConfiguration oc[2]
const uint8_t ff_aac_pred_sfb_max[]
uint8_t prediction_used[41]
common internal api header.
Single Channel Element - used for both SCE and LFE elements.
#define CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
const uint16_t *const ff_swb_offset_1024[]
static av_cold int aac_decode_init(AVCodecContext *avctx)
Individual Channel Stream.
float ff_aac_pow2sf_tab[428]
static ChannelElement * get_che(AACContext *ac, int type, int elem_id)
static av_cold int init(AVCodecParserContext *s)
int avpriv_mpeg4audio_get_config(MPEG4AudioConfig *c, const uint8_t *buf, int bit_size, int sync_extension)
Parse MPEG-4 systems extradata to retrieve audio configuration.
static const float ltp_coef[8]
const uint16_t *const ff_aac_codebook_vector_idx[]
static void windowing_and_mdct_ltp(AACContext *ac, float *out, float *in, IndividualChannelStream *ics)
Apply windowing and MDCT to obtain the spectral coefficient from the predicted sample by LTP...
channel element - generic struct for SCE/CPE/CCE/LFE
static int aac_decode_er_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, GetBitContext *gb)
static uint64_t sniff_channel_order(uint8_t(*layout_map)[3], int tags)
const uint8_t ff_tns_max_bands_512[]
av_cold void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx)
Scalefactors and spectral data are all zero.
int channels
number of audio channels
const uint8_t ff_mpeg4audio_channels[8]
VLC_TYPE(* table)[2]
code, bits
static int decode_scalefactors(AACContext *ac, float 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.
static const uint8_t * align_get_bits(GetBitContext *s)
#define FF_PROFILE_AAC_HE
enum BandType band_type[128]
band types
static int set_default_channel_config(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 int sample_rate_idx(int rate)
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...
#define AV_CH_FRONT_RIGHT
static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
Skip data_stream_element; reference: table 4.10.
static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx, GetBitContext *gb, MPEG4AudioConfig *m4ac, int channel_config)
float ret_buf[2048]
PCM output buffer.
void ff_aac_tableinit(void)
void(* butterflies_float)(float *restrict v1, float *restrict v2, int len)
Calculate the sum and difference of two vectors of floats.
int sbr
-1 implicit, 1 presence
uint8_t * av_packet_get_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int *size)
Get side information from packet.
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics, GetBitContext *gb)
static void apply_tns(float coef[1024], TemporalNoiseShaping *tns, IndividualChannelStream *ics, int decode)
Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4...
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
Decode coupling_channel_element; reference: table 4.8.
static int aac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
#define FFSWAP(type, a, b)
int ps
-1 implicit, 1 presence
int8_t used[MAX_LTP_LONG_SFB]
static av_always_inline float flt16_trunc(float pf)
const uint16_t *const ff_swb_offset_128[]
static av_always_inline float flt16_even(float pf)
static const float *const tns_tmp2_map[4]
uint8_t ** extended_data
pointers to the data planes/channels.
uint64_t request_channel_layout
Request decoder to use this channel layout if it can (0 for default)
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
static VLC vlc_spectral[11]
static int count_channels(uint8_t(*layout)[3], int tags)
void ff_init_ff_sine_windows(int index)
initialize the specified entry of ff_sine_windows
static av_always_inline float flt16_round(float pf)
static void decode_ltp(LongTermPrediction *ltp, GetBitContext *gb, uint8_t max_sfb)
Decode Long Term Prediction data; reference: table 4.xx.
static float * VMUL4(float *dst, const float *v, unsigned idx, const float *scale)
void(* vector_fmul_reverse)(float *dst, const float *src0, const float *src1, int len)
Calculate the product of two vectors of floats, and store the result in a vector of floats...