48 #define MAX_LSPS_ALIGN16 16
51 #define MAX_FRAMESIZE 160
52 #define MAX_SIGNAL_HISTORY 416
53 #define MAX_SFRAMESIZE (MAX_FRAMESIZE * MAX_FRAMES)
55 #define SFRAME_CACHE_MAXSIZE 256 175 uint16_t block_conv_table[4];
238 int aw_first_pulse_off[2];
249 float gain_pred_err[6];
268 float sin[511], cos[511];
301 int cntr[8] = { 0 },
n, res;
303 memset(vbm_tree, 0xff,
sizeof(vbm_tree[0]) * 25);
304 for (
n = 0;
n < 17;
n++) {
308 vbm_tree[res * 3 + cntr[res]++] =
n;
318 10, 10, 10, 12, 12, 12,
321 static const uint16_t codes[] = {
322 0x0000, 0x0001, 0x0002,
323 0x000c, 0x000d, 0x000e,
324 0x003c, 0x003d, 0x003e,
325 0x00fc, 0x00fd, 0x00fe,
326 0x03fc, 0x03fd, 0x03fe,
327 0x0ffc, 0x0ffd, 0x0ffe,
328 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff
332 bits, 1, 1, codes, 2, 2, 132);
343 for (n = 0; n < s->
lsps; n++)
368 int n,
flags, pitch_range, lsp16_flag;
381 "Invalid extradata size %d (should be 46)\n",
400 memcpy(&s->
sin[255], s->
cos, 256 *
sizeof(s->
cos[0]));
401 for (n = 0; n < 255; n++) {
409 "Invalid denoise filter strength %d (max=11)\n",
417 lsp16_flag = flags & 0x1000;
423 for (n = 0; n < s->
lsps; n++)
438 if (pitch_range <= 0) {
448 int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8,
452 "Unsupported samplerate %d (min=%d, max=%d)\n",
502 const float *speech_synth,
506 float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor;
507 float mem = *gain_mem;
509 for (i = 0; i <
size; i++) {
510 speech_energy += fabsf(speech_synth[i]);
511 postfilter_energy += fabsf(in[i]);
513 gain_scale_factor = postfilter_energy == 0.0 ? 0.0 :
514 (1.0 -
alpha) * speech_energy / postfilter_energy;
516 for (i = 0; i <
size; i++) {
517 mem = alpha * mem + gain_scale_factor;
518 out[i] = in[i] *
mem;
546 float optimal_gain = 0, dot;
549 *best_hist_ptr =
NULL;
554 if (dot > optimal_gain) {
558 }
while (--ptr >= end);
560 if (optimal_gain <= 0)
566 if (optimal_gain <= dot) {
567 dot = dot / (dot + 0.6 * optimal_gain);
572 for (n = 0; n <
size; n++)
573 out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]);
605 float irange, angle_mul, gain_mul, range, sq;
610 #define log_range(var, assign) do { \ 611 float tmp = log10f(assign); var = tmp; \ 612 max = FFMAX(max, tmp); min = FFMIN(min, tmp); \ 614 log_range(last_coeff, lpcs[1] * lpcs[1]);
615 for (n = 1; n < 64; n++)
616 log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] +
617 lpcs[n * 2 + 1] * lpcs[n * 2 + 1]);
628 irange = 64.0 / range;
632 for (n = 0; n <= 64; n++) {
635 idx =
lrint((max - lpcs[n]) * irange - 1);
638 lpcs[
n] = angle_mul * pwr;
641 idx = av_clipf((pwr * gain_mul - 0.0295) * 70.570526123, 0, INT_MAX / 2);
645 powf(1.0331663, idx - 127);
658 idx = 255 + av_clip(lpcs[64], -255, 255);
659 coeffs[0] = coeffs[0] * s->
cos[idx];
660 idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255);
661 last_coeff = coeffs[64] * s->
cos[idx];
663 idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255);
664 coeffs[n * 2 + 1] = coeffs[
n] * s->
sin[idx];
665 coeffs[n * 2] = coeffs[
n] * s->
cos[idx];
669 idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255);
670 coeffs[n * 2 + 1] = coeffs[
n] * s->
sin[idx];
671 coeffs[n * 2] = coeffs[
n] * s->
cos[idx];
679 memset(&coeffs[remainder], 0,
sizeof(coeffs[0]) * (128 - remainder));
683 coeffs[remainder - 1] = 0;
690 for (n = 0; n < remainder; n++)
721 float *synth_pf,
int size,
724 int remainder, lim,
n;
730 tilted_lpcs[0] = 1.0;
731 memcpy(&tilted_lpcs[1], lpcs,
sizeof(lpcs[0]) * s->
lsps);
732 memset(&tilted_lpcs[s->
lsps + 1], 0,
733 sizeof(tilted_lpcs[0]) * (128 - s->
lsps - 1));
735 tilted_lpcs, s->
lsps + 2);
741 remainder =
FFMIN(127 - size, size - 1);
746 memset(&synth_pf[size], 0,
sizeof(synth_pf[0]) * (128 - size));
751 for (n = 1; n < 64; n++) {
752 float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1];
753 synth_pf[n * 2] = v1 *
coeffs[n * 2] - v2 *
coeffs[n * 2 + 1];
754 synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1];
762 for (n = 0; n < lim; n++)
772 for (n = 0; n < lim; n++)
774 if (lim < remainder) {
803 float *samples,
int size,
804 const float *lpcs,
float *zero_exc_pf,
809 *synth_filter_in = zero_exc_pf;
818 synth_filter_in = synth_filter_in_buf;
822 synth_filter_in, size, s->
lsps);
823 memcpy(&synth_pf[-s->
lsps], &synth_pf[size - s->
lsps],
824 sizeof(synth_pf[0]) * s->
lsps);
836 (
const float[2]) { -1.99997, 1.0 },
837 (
const float[2]) { -1.9330735188, 0.93589198496 },
857 const uint16_t *values,
858 const uint16_t *
sizes,
861 const double *base_q)
865 memset(lsps, 0, num *
sizeof(*lsps));
866 for (n = 0; n < n_stages; n++) {
867 const uint8_t *t_off = &table[values[
n] * num];
868 double base = base_q[
n], mul = mul_q[
n];
870 for (m = 0; m < num; m++)
871 lsps[m] += base + mul * t_off[m];
873 table += sizes[
n] * num;
889 static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 };
890 static const double mul_lsf[4] = {
891 5.2187144800e-3, 1.4626986422e-3,
892 9.6179549166e-4, 1.1325736225e-3
894 static const double base_lsf[4] = {
895 M_PI * -2.15522e-1,
M_PI * -6.1646e-2,
896 M_PI * -3.3486e-2,
M_PI * -5.7408e-2
914 double *i_lsps,
const double *old,
915 double *
a1,
double *
a2,
int q_mode)
917 static const uint16_t vec_sizes[3] = { 128, 64, 64 };
918 static const double mul_lsf[3] = {
919 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3
921 static const double base_lsf[3] = {
922 M_PI * -1.07448e-1,
M_PI * -5.2706e-2,
M_PI * -5.1634e-2
924 const float (*ipol_tab)[2][10] = q_mode ?
936 for (n = 0; n < 10; n++) {
937 double delta = old[
n] - i_lsps[
n];
938 a1[
n] = ipol_tab[
interpol][0][
n] * delta + i_lsps[
n];
939 a1[10 +
n] = ipol_tab[
interpol][1][
n] * delta + i_lsps[
n];
951 static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 };
952 static const double mul_lsf[5] = {
953 3.3439586280e-3, 6.9908173703e-4,
954 3.3216608306e-3, 1.0334960326e-3,
957 static const double base_lsf[5] = {
958 M_PI * -1.27576e-1,
M_PI * -2.4292e-2,
959 M_PI * -1.28094e-1,
M_PI * -3.2128e-2,
983 double *i_lsps,
const double *old,
984 double *
a1,
double *
a2,
int q_mode)
986 static const uint16_t vec_sizes[3] = { 128, 128, 128 };
987 static const double mul_lsf[3] = {
988 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3
990 static const double base_lsf[3] = {
993 const float (*ipol_tab)[2][16] = q_mode ?
1005 for (n = 0; n < 16; n++) {
1006 double delta = old[
n] - i_lsps[
n];
1007 a1[
n] = ipol_tab[
interpol][0][
n] * delta + i_lsps[
n];
1008 a1[16 +
n] = ipol_tab[
interpol][1][
n] * delta + i_lsps[
n];
1035 static const int16_t start_offset[94] = {
1036 -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11,
1037 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26,
1038 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43,
1039 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67,
1040 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91,
1041 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115,
1042 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139,
1043 141, 143, 145, 147, 149, 151, 153, 155, 157, 159
1049 if ((bits =
get_bits(gb, 6)) >= 54) {
1051 bits += (bits - 54) * 3 +
get_bits(gb, 2);
1057 for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ;
1070 if (start_offset[bits] < 0)
1087 uint16_t use_mask_mem[9];
1088 uint16_t *use_mask = use_mask_mem + 2;
1097 pulse_start,
n, idx, range, aidx, start_off = 0;
1106 if (block_idx == 0) {
1115 pulse_start = s->
aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0;
1120 memset(&use_mask[-2], 0, 2 *
sizeof(use_mask[0]));
1121 memset( use_mask, -1, 5 *
sizeof(use_mask[0]));
1122 memset(&use_mask[5], 0, 2 *
sizeof(use_mask[0]));
1126 uint16_t *use_mask_ptr = &use_mask[idx >> 4];
1127 int first_sh = 16 - (idx & 15);
1128 *use_mask_ptr++ &= 0xFFFF
u << first_sh;
1129 excl_range -= first_sh;
1130 if (excl_range >= 16) {
1131 *use_mask_ptr++ = 0;
1132 *use_mask_ptr &= 0xFFFF >> (excl_range - 16);
1134 *use_mask_ptr &= 0xFFFF >> excl_range;
1139 for (n = 0; n <= aidx; pulse_start++) {
1140 for (idx = pulse_start; idx < 0; idx += fcb->
pitch_lag) ;
1142 if (use_mask[0]) idx = 0x0F;
1143 else if (use_mask[1]) idx = 0x1F;
1144 else if (use_mask[2]) idx = 0x2F;
1145 else if (use_mask[3]) idx = 0x3F;
1146 else if (use_mask[4]) idx = 0x4F;
1150 if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) {
1151 use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15));
1157 fcb->
x[fcb->
n] = start_off;
1181 int n, v_mask, i_mask, sh, n_pulses;
1195 for (n = n_pulses - 1; n >= 0; n--, val >>= sh) {
1196 fcb->
y[fcb->
n] = (val & v_mask) ? -1.0 : 1.0;
1197 fcb->
x[fcb->
n] = (val & i_mask) * n_pulses + n +
1199 while (fcb->
x[fcb->
n] < 0)
1205 int num2 = (val & 0x1FF) >> 1,
delta, idx;
1207 if (num2 < 1 * 79) {
delta = 1; idx = num2 + 1; }
1208 else if (num2 < 2 * 78) {
delta = 3; idx = num2 + 1 - 1 * 77; }
1209 else if (num2 < 3 * 77) {
delta = 5; idx = num2 + 1 - 2 * 76; }
1210 else {
delta = 7; idx = num2 + 1 - 3 * 75; }
1211 v = (val & 0x200) ? -1.0 : 1.0;
1216 fcb->
x[fcb->
n + 1] = idx;
1217 fcb->
y[fcb->
n + 1] = (val & 1) ? -v : v;
1235 static int pRNG(
int frame_cntr,
int block_num,
int block_size)
1247 static const unsigned int div_tbl[9][2] = {
1248 { 8332, 3 * 715827883
U },
1249 { 4545, 0 * 390451573
U },
1250 { 3124, 11 * 268435456
U },
1251 { 2380, 15 * 204522253
U },
1252 { 1922, 23 * 165191050
U },
1253 { 1612, 23 * 138547333
U },
1254 { 1388, 27 * 119304648
U },
1255 { 1219, 16 * 104755300
U },
1256 { 1086, 39 * 93368855
U }
1258 unsigned int z, y, x =
MUL16(block_num, 1877) + frame_cntr;
1259 if (x >= 0xFFFF) x -= 0xFFFF;
1261 y = x - 9 *
MULH(477218589, x);
1262 z = (uint16_t) (x * div_tbl[y][0] +
UMULH(x, div_tbl[y][1]));
1264 return z % (1000 - block_size);
1272 int block_idx,
int size,
1294 for (n = 0; n <
size; n++)
1303 int block_idx,
int size,
1304 int block_pitch_sh2,
1308 static const float gain_coeff[6] = {
1309 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458
1312 int n, idx, gain_weight;
1316 memset(pulses, 0,
sizeof(*pulses) * size);
1333 for (n = 0; n <
size; n++)
1345 for (n = 0; n < 5; n++) {
1351 fcb.
x[fcb.
n] = n + 5 * pos1;
1352 fcb.
y[fcb.
n++] = sign;
1355 fcb.
x[fcb.
n] = n + 5 * pos2;
1356 fcb.
y[fcb.
n++] = (pos1 < pos2) ? -sign : sign;
1376 for (n = 0; n < gain_weight; n++)
1382 for (n = 0; n <
size; n +=
len) {
1384 int abs_idx = block_idx * size +
n;
1387 int pitch = (pitch_sh16 + 0x6FFF) >> 16;
1388 int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000;
1389 idx = idx_sh16 >> 16;
1392 next_idx_sh16 = (idx_sh16) &~ 0xFFFF;
1394 next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF;
1405 int block_pitch = block_pitch_sh2 >> 2;
1406 idx = block_pitch_sh2 & 3;
1413 sizeof(
float) * size);
1418 acb_gain, fcb_gain, size);
1437 int block_idx,
int size,
1438 int block_pitch_sh2,
1439 const double *lsps,
const double *prev_lsps,
1441 float *excitation,
float *synth)
1452 frame_desc, excitation);
1455 fac = (block_idx + 0.5) / frame_desc->
n_blocks;
1456 for (n = 0; n < s->
lsps; n++)
1457 i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n]));
1480 const double *lsps,
const double *prev_lsps,
1481 float *excitation,
float *synth)
1484 int n, n_blocks_x2, log_n_blocks_x2,
av_uninit(cur_pitch_val);
1492 "Invalid frame type VLC code, skipping\n");
1515 int fac = n * 2 + 1;
1517 pitch[
n] = (
MUL16(fac, cur_pitch_val) +
1559 last_block_pitch = av_clip(block_pitch,
1565 if (block_pitch < t1) {
1569 if (block_pitch <
t2) {
1574 if (block_pitch <
t3) {
1581 pitch[
n] = bl_pitch_sh2 >> 2;
1586 bl_pitch_sh2 = pitch[
n] << 2;
1595 synth_block(s, gb, n, block_nsamples, bl_pitch_sh2,
1597 &excitation[n * block_nsamples],
1598 &synth[n * block_nsamples]);
1607 for (n = 0; n < s->
lsps; n++)
1608 i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n]));
1614 for (n = 0; n < s->
lsps; n++)
1615 i_lsps[n] = cos(lsps[n]);
1617 postfilter(s, &synth[80], &samples[80], 80, lpcs,
1621 memcpy(samples, synth, 160 *
sizeof(synth[0]));
1661 lsps[0] =
FFMAX(lsps[0], 0.0015 *
M_PI);
1662 for (n = 1; n < num; n++)
1663 lsps[n] =
FFMAX(lsps[n], lsps[n - 1] + 0.0125 *
M_PI);
1664 lsps[num - 1] =
FFMIN(lsps[num - 1], 0.9985 *
M_PI);
1668 for (n = 1; n < num; n++) {
1669 if (lsps[n] < lsps[n - 1]) {
1670 for (m = 1; m < num; m++) {
1671 double tmp = lsps[m];
1672 for (l = m - 1; l >= 0; l--) {
1673 if (lsps[l] <= tmp)
break;
1674 lsps[l + 1] = lsps[l];
1714 s->
lsps *
sizeof(*synth));
1737 "Superframe encodes > %d samples (%d), not allowed\n",
1747 for (n = 0; n < s->
lsps; n++)
1748 prev_lsps[n] = s->
prev_lsps[n] - mean_lsf[n];
1755 for (n = 0; n < s->
lsps; n++) {
1756 lsps[0][
n] = mean_lsf[
n] + (a1[
n] - a2[n * 2]);
1757 lsps[1][
n] = mean_lsf[
n] + (a1[s->
lsps +
n] - a2[n * 2 + 1]);
1758 lsps[2][
n] += mean_lsf[
n];
1760 for (n = 0; n < 3; n++)
1769 samples = (
float *)frame->
data[0];
1772 for (n = 0; n < 3; n++) {
1776 if (s->
lsps == 10) {
1781 for (m = 0; m < s->
lsps; m++)
1782 lsps[n][m] += mean_lsf[m];
1788 lsps[n], n == 0 ? s->
prev_lsps : lsps[n - 1],
1790 &synth[s->
lsps + n * MAX_FRAMESIZE]))) {
1815 s->
lsps *
sizeof(*synth));
1835 unsigned int res, n_superframes = 0;
1845 n_superframes += res;
1846 }
while (res == 0x3F);
1871 int rmn_bytes, rmn_bits;
1874 if (rmn_bits < nbits)
1878 rmn_bits &= 7; rmn_bytes >>= 3;
1879 if ((rmn_bits =
FFMIN(rmn_bits, nbits)) > 0)
1882 FFMIN(nbits - rmn_bits, rmn_bytes << 3));
1897 int *got_frame_ptr,
AVPacket *avpkt)
1960 }
else if (*got_frame_ptr) {
Description of frame types.
static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb, int block_idx, AMRFixed *fcb)
Apply first set of pitch-adaptive window pulses.
av_cold void ff_rdft_end(RDFTContext *s)
static const uint8_t wmavoice_dq_lsp16r2[0x500]
const char const char void * val
int do_apf
whether to apply the averaged projection filter (APF)
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
static int pRNG(int frame_cntr, int block_num, int block_size)
Generate a random number from frame_cntr and block_idx, which will live in the range [0...
static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25])
Set up the variable bit mode (VBM) tree from container extradata.
void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP synthesis filter.
float gain_pred_err[6]
cache for gain prediction
static float alpha(float a)
This structure describes decoded (raw) audio or video data.
int aw_next_pulse_off_cache
the position (relative to start of the second block) at which pulses should start to be positioned...
int nb_superframes
number of superframes in current packet
static void flush(AVCodecContext *avctx)
float postfilter_agc
gain control memory, used in adaptive_gain_control()
void ff_acelp_apply_order_2_transfer_function(float *out, const float *in, const float zero_coeffs[2], const float pole_coeffs[2], float gain, float mem[2], int n)
Apply an order 2 rational transfer function in-place.
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
static void postfilter(WMAVoiceContext *s, const float *synth, float *samples, int size, const float *lpcs, float *zero_exc_pf, int fcb_type, int pitch)
Averaging projection filter, the postfilter used in WMAVoice.
Memory handling functions.
void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b, float weight_coeff_a, float weight_coeff_b, int length)
float implementation of weighted sum of two vectors.
static void skip_bits_long(GetBitContext *s, int n)
static av_cold int init(AVCodecContext *avctx)
#define INIT_VLC_STATIC(vlc, bits, a, b, c, d, e, f, g, static_size)
#define avpriv_request_sample(...)
float synth_filter_out_buf[0x80+MAX_LSPS_ALIGN16]
aligned buffer for postfilter speech synthesis
static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, const int *pitch)
Parse the offset of the first pitch-adaptive window pulses, and the distribution of pulses between th...
static const int8_t pulses[4]
Number of non-zero pulses in the MP-MLQ excitation.
int aw_n_pulses[2]
number of AW-pulses in each block; note that this number can be negative (in which case it basically ...
static int interpol(MBContext *s, uint32_t *color, int x, int y, int linesize)
void avpriv_copy_bits(PutBitContext *pb, const uint8_t *src, int length)
Copy the content of src to the bitstream.
static void stabilize_lsps(double *lsps, int num)
Ensure minimum value for first item, maximum value for last value, proper spacing between each value ...
static const float wmavoice_gain_codebook_fcb[128]
static const uint8_t wmavoice_dq_lsp16i1[0x640]
static const uint8_t wmavoice_dq_lsp16r1[0x500]
int spillover_nbits
number of bits of the previous packet's last superframe preceding this packet's first full superframe...
void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size)
Add fixed vector to an array from a sparse representation.
hardcoded (fixed) codebook with per-block gain values
int block_pitch_nbits
number of bits used to specify the first block's pitch value
static const uint8_t wmavoice_dq_lsp16i3[0x300]
static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, float *samples, const double *lsps, const double *prev_lsps, float *excitation, float *synth)
Synthesize output samples for a single frame.
adaptive codebook with per-frame pitch, which we interpolate to get a per-sample pitch.
static void calc_input_response(WMAVoiceContext *s, float *lpcs, int fcb_type, float *coeffs, int remainder)
Derive denoise filter coefficients (in real domain) from the LPCs.
static void dequant_lsp10i(GetBitContext *gb, double *lsps)
Parse 10 independently-coded LSPs.
#define MAX_LSPS_ALIGN16
same as MAX_LSPS; needs to be multiple
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, int block_idx, AMRFixed *fcb)
Apply second set of pitch-adaptive window pulses.
Innovation (fixed) codebook pulse sets in combinations of either single pulses or pulse pairs...
static const float wmavoice_ipol1_coeffs[17 *9]
static const uint8_t wmavoice_dq_lsp16i2[0x3c0]
comfort noise during silence generated from a hardcoded (fixed) codebook with per-frame (low) gain va...
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
#define av_assert0(cond)
assert() equivalent, that is always enabled.
int spillover_bitsize
number of bits used to specify spillover_nbits in the packet header = ceil(log2(ctx->block_align << 3...
int block_delta_pitch_nbits
number of bits used to specify the delta pitch between this and the last block's pitch value...
enum AVSampleFormat sample_fmt
audio sample format
Sparse representation for the algebraic codebook (fixed) vector.
static const uint8_t wmavoice_dq_lsp16r3[0x600]
static const float wmavoice_gain_codebook_acb[128]
uint8_t log_n_blocks
log2(n_blocks)
int aw_first_pulse_off[2]
index of first sample to which to apply AW-pulses, or -0xff if unset
static av_cold int end(AVCodecContext *avctx)
int has_residual_lsps
if set, superframes contain one set of LSPs that cover all frames, encoded as independent and residua...
float tilted_lpcs_pf[0x80]
aligned buffer for LPC tilting
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
static float tilt_factor(const float *lpcs, int n_lpcs)
Get the tilt factor of a formant filter from its transfer function.
static const uint8_t wmavoice_dq_lsp10r[0x1400]
static void dequant_lsps(double *lsps, int num, const uint16_t *values, const uint16_t *sizes, int n_stages, const uint8_t *table, const double *mul_q, const double *base_q)
Dequantize LSPs.
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
static const float wmavoice_ipol2_coeffs[32]
Hamming-window sinc function (num = 32, x = [ 0, 31 ]): (0.54 + 0.46 * cos(2 * M_PI * x / (num - 1)))...
static int get_bits_count(const GetBitContext *s)
float dcf_mem[2]
DC filter history.
void av_memcpy_backptr(uint8_t *dst, int back, int cnt)
Overlapping memcpy() implementation.
bitstream reader API header.
static av_cold void wmavoice_flush(AVCodecContext *ctx)
float synth_history[MAX_LSPS]
see excitation_history
double prev_lsps[MAX_LSPS]
LSPs of the last frame of the previous superframe.
static void copy_bits(PutBitContext *pb, const uint8_t *data, int size, GetBitContext *gb, int nbits)
Copy (unaligned) bits from gb/data/size to pb.
static int get_bits_left(GetBitContext *gb)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static const double wmavoice_mean_lsf16[2][16]
int sframe_cache_size
set to >0 if we have data from an (incomplete) superframe from a previous packet that spilled over in...
static const float wmavoice_lsp10_intercoeff_b[32][2][10]
int block_pitch_range
range of the block pitch
static const float wmavoice_std_codebook[1000]
static const int sizes[][2]
int last_acb_type
frame type [0-2] of the previous frame
static const struct endianess table[]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
static const float wmavoice_gain_silence[256]
int denoise_filter_cache_size
samples in denoise_filter_cache
int history_nsamples
number of samples in history for signal prediction (through ACB)
static const uint8_t wmavoice_dq_lsp10i[0xf00]
static const float wmavoice_lsp10_intercoeff_a[32][2][10]
static const float wmavoice_energy_table[128]
LUT for 1.071575641632 * pow(1.0331663, n - 127)
Windows Media Voice (WMAVoice) tables.
const char * name
Name of the codec implementation.
int denoise_tilt_corr
Whether to apply tilt correction to the Wiener filter coefficients (postfilter)
int aw_idx_is_ext
whether the AW index was encoded in 8 bits (instead of 6)
no adaptive codebook (only hardcoded fixed)
static const uint8_t offset[127][2]
uint16_t block_conv_table[4]
boundaries for block pitch unit/scale conversion
DCTContext dst
contexts for phase shift (in Hilbert transform, part of postfilter)
int lsp_def_mode
defines different sets of LSP defaults [0, 1]
uint64_t channel_layout
Audio channel layout.
static int put_bits_count(PutBitContext *s)
int skip_bits_next
number of bits to skip at the next call to wmavoice_decode_packet() (since they're part of the previo...
static void dequant_lsp16r(GetBitContext *gb, double *i_lsps, const double *old, double *a1, double *a2, int q_mode)
Parse 16 independently-coded LSPs, and then derive the tables to generate LSPs for the other frames f...
int min_pitch_val
base value for pitch parsing code
WMA Voice decoding context.
static void wiener_denoise(WMAVoiceContext *s, int fcb_type, float *synth_pf, int size, const float *lpcs)
This function applies a Wiener filter on the (noisy) speech signal as a means to denoise it...
int denoise_strength
strength of denoising in Wiener filter [0-11]
uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE+AV_INPUT_BUFFER_PADDING_SIZE]
cache for superframe data split over multiple packets
audio channel layout utility functions
#define log_range(var, assign)
#define MAX_LSPS
maximum filter order
static VLC frame_type_vlc
Frame type VLC coding.
int pitch_nbits
number of bits used to specify the pitch value in the frame header
#define MAX_BLOCKS
maximum number of blocks per frame
float denoise_coeffs_pf[0x80]
aligned buffer for denoise coefficients
void(* dct_calc)(struct DCTContext *s, FFTSample *data)
static void dequant_lsp10r(GetBitContext *gb, double *i_lsps, const double *old, double *a1, double *a2, int q_mode)
Parse 10 independently-coded LSPs, and then derive the tables to generate LSPs for the other frames f...
static av_always_inline unsigned UMULH(unsigned a, unsigned b)
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
void(* rdft_calc)(struct RDFTContext *s, FFTSample *z)
static int kalman_smoothen(WMAVoiceContext *s, int pitch, const float *in, float *out, int size)
Kalman smoothing function.
void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
Apply tilt compensation filter, 1 - tilt * z-1.
static const float wmavoice_gain_universal[64]
void ff_acelp_lspd2lpc(const double *lsp, float *lpc, int lp_half_order)
Reconstruct LPC coefficients from the line spectral pair frequencies.
static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
Set up decoder with parameters from demuxer (extradata etc.).
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
static const uint8_t last_coeff[3]
static const struct frame_type_desc frame_descs[17]
float denoise_filter_cache[MAX_FRAMESIZE]
Libavcodec external API header.
int sample_rate
samples per second
void AAC_RENAME() ff_sine_window_init(INTFLOAT *window, int n)
Generate a sine window.
static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Packet decoding: a packet is anything that the (ASF) demuxer contains, and we expect that the demuxer...
main external API structure.
static int parse_packet_header(WMAVoiceContext *s)
Parse the packet header at the start of each packet (input data to this decoder). ...
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
AVCodec ff_wmavoice_decoder
int8_t vbm_tree[25]
converts VLC codes to frame type
static unsigned int get_bits1(GetBitContext *s)
static void synth_block(WMAVoiceContext *s, GetBitContext *gb, int block_idx, int size, int block_pitch_sh2, const double *lsps, const double *prev_lsps, const struct frame_type_desc *frame_desc, float *excitation, float *synth)
Parse data in a single block.
static av_cold int wmavoice_decode_end(AVCodecContext *ctx)
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;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);return NULL;} return ac;} 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;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->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);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
static void skip_bits(GetBitContext *s, int n)
av_cold int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType inverse)
Set up DCT.
#define AV_CODEC_CAP_SUBFRAMES
Codec can output multiple frames per AVPacket Normally demuxers return one frame at a time...
int pitch_diff_sh16
((cur_pitch_val - last_pitch_val) << 16) / MAX_FRAMESIZE
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
#define MAX_SFRAMESIZE
maximum number of samples per superframe
int lsp_q_mode
defines quantizer defaults [0, 1]
int frame_cntr
current frame index [0 - 0xFFFE]; is only used for comfort noise in pRNG()
void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP zero synthesis filter.
float avpriv_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
static void adaptive_gain_control(float *out, const float *in, const float *speech_synth, int size, float alpha, float *gain_mem)
Adaptive gain control (as used in postfilter).
Pitch-adaptive window (AW) pulse signals, used in particular for low-bitrate streams.
static const float mean_lsf[10]
#define SFRAME_CACHE_MAXSIZE
maximum cache size for frame data that
uint8_t fcb_type
Fixed codebook type (FCB_TYPE_*)
static void dequant_lsp16i(GetBitContext *gb, double *lsps)
Parse 16 independently-coded LSPs.
RDFTContext irdft
contexts for FFT-calculation in the postfilter (for denoise filter)
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, int *got_frame_ptr)
Synthesize output samples for a single superframe.
Per-block pitch with signal generation using a Hamming sinc window function.
static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, int block_idx, int size, const struct frame_type_desc *frame_desc, float *excitation)
Parse hardcoded signal for a single block.
uint8_t n_blocks
amount of blocks per frame (each block (contains 160/n_blocks samples)
common internal api header.
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
static av_cold void wmavoice_init_static_data(AVCodec *codec)
float excitation_history[MAX_SIGNAL_HISTORY]
cache of the signal of previous superframes, used as a history for signal generation ...
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
int last_pitch_val
pitch value of the previous frame
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
#define MAX_FRAMESIZE
maximum number of samples per frame
float silence_gain
set for use in blocks if ACB_TYPE_NONE
static const double wmavoice_mean_lsf10[2][10]
static const int16_t coeffs[]
int channels
number of audio channels
VLC_TYPE(* table)[2]
code, bits
av_cold void ff_dct_end(DCTContext *s)
void ff_acelp_interpolatef(float *out, const float *in, const float *filter_coeffs, int precision, int frac_pos, int filter_length, int length)
Floating point version of ff_acelp_interpolate()
int block_delta_pitch_hrange
1/2 range of the delta (full range is from -this to +this-1)
int max_pitch_val
max value + 1 for pitch parsing
int lsps
number of LSPs per frame [10 or 16]
#define MAX_FRAMES
maximum number of frames per superframe
static const float wmavoice_lsp16_intercoeff_b[32][2][16]
PutBitContext pb
bitstream writer for sframe_cache
uint8_t acb_type
Adaptive codebook type (ACB_TYPE_*)
static const float wmavoice_denoise_power_table[12][64]
LUT for f(x,y) = pow((y + 6.9) / 64, 0.025 * (x + 1)).
int dc_level
Predicted amount of DC noise, based on which a DC removal filter is used.
#define VLC_NBITS
number of bits to read per VLC iteration
static const float wmavoice_lsp16_intercoeff_a[32][2][16]
float cos[511]
8-bit cosine/sine windows over [-pi,pi] range
#define AV_CH_LAYOUT_MONO
av_cold int ff_rdft_init(RDFTContext *s, int nbits, enum RDFTransformType trans)
Set up a real FFT.
int aw_pulse_range
the range over which aw_pulse_set1() can apply the pulse, relative to the value in aw_first_pulse_off...
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
float zero_exc_pf[MAX_SIGNAL_HISTORY+MAX_SFRAMESIZE]
zero filter output (i.e.
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, int block_idx, int size, int block_pitch_sh2, const struct frame_type_desc *frame_desc, float *excitation)
Parse FCB/ACB signal for a single block.
uint8_t dbl_pulses
how many pulse vectors have pulse pairs (rather than just one single pulse) only if fcb_type == FCB_T...
#define MAX_SIGNAL_HISTORY
maximum excitation signal history
GetBitContext gb
packet bitreader.