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65 #define SPEEX_NB_MODES 3
66 #define SPEEX_INBAND_STEREO 9
70 #define NB_FRAME_SIZE 160
72 #define NB_SUBMODE_BITS 4
73 #define SB_SUBMODE_BITS 3
75 #define NB_SUBFRAME_SIZE 40
76 #define NB_NB_SUBFRAMES 4
77 #define NB_PITCH_START 17
78 #define NB_PITCH_END 144
80 #define NB_DEC_BUFFER (NB_FRAME_SIZE + 2 * NB_PITCH_END + NB_SUBFRAME_SIZE + 12)
82 #define SPEEX_MEMSET(dst, c, n) (memset((dst), (c), (n) * sizeof(*(dst))))
83 #define SPEEX_COPY(dst, src, n) (memcpy((dst), (src), (n) * sizeof(*(dst))))
85 #define LSP_LINEAR(i) (.25f * (i) + .25f)
86 #define LSP_LINEAR_HIGH(i) (.3125f * (i) + .75f)
87 #define LSP_DIV_256(x) (0.00390625f * (x))
88 #define LSP_DIV_512(x) (0.001953125f * (x))
89 #define LSP_DIV_1024(x) (0.0009765625f * (x))
127 float *,
float *,
float *,
128 const void *, int, int,
float, int, int,
130 float *, int, int, int,
float *);
134 float,
const void *, int,
int *,
140 float *,
float *,
const void *,
141 int, int,
float *,
float *,
228 const int req_size =
get_bits(gb, 4);
267 for (
int i = 0;
i < order;
i++)
271 for (
int i = 0;
i < 10;
i++)
275 for (
int i = 0;
i < 5;
i++)
279 for (
int i = 0;
i < 5;
i++)
284 float pitch_coef,
const void *par,
int nsf,
285 int *pitch_val,
float *gain_val,
GetBitContext *gb,
int count_lost,
286 int subframe_offset,
float last_pitch_gain,
int cdbk_offset)
289 pitch_coef =
fminf(pitch_coef, .99
f);
290 for (
int i = 0;
i < nsf;
i++) {
291 exc_out[
i] = exc[
i - start] * pitch_coef;
294 pitch_val[0] = start;
295 gain_val[0] = gain_val[2] = 0.f;
296 gain_val[1] = pitch_coef;
301 const uint32_t jflone = 0x3f800000;
302 const uint32_t jflmsk = 0x007fffff;
305 seed[0] = 1664525 *
seed[0] + 1013904223;
306 ran = jflone | (jflmsk &
seed[0]);
316 for (
int i = 0;
i < nsf;
i++)
323 int subvect_size, nb_subvect, have_sign, shape_bits;
325 const signed char *shape_cb;
326 int signs[10], ind[10];
337 for (
int i = 0;
i < nb_subvect;
i++) {
342 for (
int i = 0;
i < nb_subvect;
i++) {
343 const float s = signs[
i] ? -1.f : 1.f;
345 for (
int j = 0; j < subvect_size; j++)
346 exc[subvect_size *
i + j] +=
s * 0.03125
f * shape_cb[ind[
i] * subvect_size + j];
350 #define SUBMODE(x) st->submodes[st->submodeID]->x
352 #define gain_3tap_to_1tap(g) (FFABS(g[1]) + (g[0] > 0.f ? g[0] : -.5f * g[0]) + (g[2] > 0.f ? g[2] : -.5f * g[2]))
356 const void *par,
int nsf,
int *pitch_val,
float *gain_val,
GetBitContext *gb,
357 int count_lost,
int subframe_offset,
float last_pitch_gain,
int cdbk_offset)
359 int pitch, gain_index, gain_cdbk_size;
360 const int8_t *gain_cdbk;
361 const LtpParam *params;
364 params = (
const LtpParam *)par;
365 gain_cdbk_size = 1 << params->gain_bits;
366 gain_cdbk = params->gain_cdbk + 4 * gain_cdbk_size * cdbk_offset;
368 pitch =
get_bitsz(gb, params->pitch_bits);
370 gain_index =
get_bitsz(gb, params->gain_bits);
371 gain[0] = 0.015625f * gain_cdbk[gain_index * 4] + .5f;
372 gain[1] = 0.015625f * gain_cdbk[gain_index * 4 + 1] + .5f;
373 gain[2] = 0.015625f * gain_cdbk[gain_index * 4 + 2] + .5f;
375 if (count_lost && pitch > subframe_offset) {
376 float tmp = count_lost < 4 ? last_pitch_gain : 0.5f * last_pitch_gain;
382 if (gain_sum >
tmp && gain_sum > 0.
f) {
384 for (
int i = 0;
i < 3;
i++)
389 pitch_val[0] = pitch;
390 gain_val[0] = gain[0];
391 gain_val[1] = gain[1];
392 gain_val[2] = gain[2];
395 for (
int i = 0;
i < 3;
i++) {
397 int pp = pitch + 1 -
i;
401 for (
int j = 0; j < tmp1; j++)
402 exc_out[j] += gain[2 -
i] * exc[j - pp];
404 if (tmp3 > pp + pitch)
406 for (
int j = tmp1; j < tmp3; j++)
407 exc_out[j] += gain[2 -
i] * exc[j - pp - pitch];
415 for (
int i = 0;
i < order;
i++)
419 for (
int i = 0;
i < 10;
i++)
423 for (
int i = 0;
i < 5;
i++)
427 for (
int i = 0;
i < 5;
i++)
431 for (
int i = 0;
i < 5;
i++)
435 for (
int i = 0;
i < 5;
i++)
443 for (
int i = 0;
i < order;
i++)
447 for (
int i = 0;
i < order;
i++)
451 for (
int i = 0;
i < order;
i++)
537 .default_submode = 5,
545 .folding_gain = 0.9f,
549 .default_submode = 3,
557 .folding_gain = 0.7f,
561 .default_submode = 1,
569 for (
int i = 0;
i <
len;
i++)
576 static void bw_lpc(
float gamma,
const float *lpc_in,
577 float *lpc_out,
int order)
581 for (
int i = 0;
i < order;
i++) {
582 lpc_out[
i] =
tmp * lpc_in[
i];
587 static void iir_mem(
const float *x,
const float *den,
588 float *y,
int N,
int ord,
float *mem)
590 for (
int i = 0;
i <
N;
i++) {
591 float yi = x[
i] + mem[0];
593 for (
int j = 0; j < ord - 1; j++)
594 mem[j] = mem[j + 1] + den[j] * nyi;
595 mem[ord - 1] = den[ord - 1] * nyi;
600 static void highpass(
const float *x,
float *y,
int len,
float *mem,
int wide)
602 static const float Pcoef[2][3] = {{ 1.00000f, -1.92683f, 0.93071f }, { 1.00000f, -1.97226f, 0.97332f } };
603 static const float Zcoef[2][3] = {{ 0.96446f, -1.92879f, 0.96446f }, { 0.98645f, -1.97277f, 0.98645f } };
604 const float *den, *num;
608 for (
int i = 0;
i <
len;
i++) {
609 float yi = num[0] * x[
i] + mem[0];
610 mem[0] = mem[1] + num[1] * x[
i] + -den[1] * yi;
611 mem[1] = num[2] * x[
i] + -den[2] * yi;
616 #define median3(a, b, c) \
617 ((a) < (b) ? ((b) < (c) ? (b) : ((a) < (c) ? (c) : (a))) \
618 : ((c) < (b) ? (b) : ((c) < (a) ? (c) : (a))))
659 for (
int i = 0;
i <
len;
i++) {
660 if (!isnormal(vec[
i]) ||
fabsf(vec[
i]) < 1e-8
f)
669 for (
int i = 0;
i <
len;
i++)
677 for (
int i = 0;
i <
len;
i += 8) {
679 part += x[
i + 0] * y[
i + 0];
680 part += x[
i + 1] * y[
i + 1];
681 part += x[
i + 2] * y[
i + 2];
682 part += x[
i + 3] * y[
i + 3];
683 part += x[
i + 4] * y[
i + 4];
684 part += x[
i + 5] * y[
i + 5];
685 part += x[
i + 6] * y[
i + 6];
686 part += x[
i + 7] * y[
i + 7];
695 float corr[4][7], maxcorr;
698 for (
int i = 0;
i < 7;
i++)
700 for (
int i = 0;
i < 3;
i++) {
701 for (
int j = 0; j < 7; j++) {
711 for (
int k = i1; k < i2; k++)
713 corr[
i + 1][j] =
tmp;
717 maxcorr = corr[0][0];
718 for (
int i = 0;
i < 4;
i++) {
719 for (
int j = 0; j < 7; j++) {
720 if (corr[
i][j] > maxcorr) {
721 maxcorr = corr[
i][j];
727 for (
int i = 0;
i <
len;
i++) {
730 for (
int k = 0; k < 7; k++)
731 tmp += exc[
i - (pitch - maxj + 3) + k - 3] *
shift_filt[maxi - 1][k];
733 tmp = exc[
i - (pitch - maxj + 3)];
737 return pitch - maxj + 3;
740 static void multicomb(
const float *exc,
float *new_exc,
float *ak,
int p,
int nsf,
741 int pitch,
int max_pitch,
float comb_gain)
743 float old_ener, new_ener;
744 float iexc0_mag, iexc1_mag, exc_mag;
746 float corr0, corr1, gain0, gain1;
747 float pgain1, pgain2;
748 float c1,
c2, g1, g2;
749 float ngain, gg1, gg2;
750 int corr_pitch = pitch;
753 if (corr_pitch > max_pitch)
763 if (corr0 > iexc0_mag * exc_mag)
766 pgain1 = (corr0 / exc_mag) / iexc0_mag;
767 if (corr1 > iexc1_mag * exc_mag)
770 pgain2 = (corr1 / exc_mag) / iexc1_mag;
771 gg1 = exc_mag / iexc0_mag;
772 gg2 = exc_mag / iexc1_mag;
773 if (comb_gain > 0.
f) {
774 c1 = .4f * comb_gain + .07f;
775 c2 = .5f + 1.72f * (
c1 - .07f);
779 g1 = 1.f -
c2 * pgain1 * pgain1;
780 g2 = 1.f -
c2 * pgain2 * pgain2;
786 if (corr_pitch > max_pitch) {
787 gain0 = .7f * g1 * gg1;
788 gain1 = .3f * g2 * gg2;
790 gain0 = .6f * g1 * gg1;
791 gain1 = .6f * g2 * gg2;
793 for (
int i = 0;
i < nsf;
i++)
794 new_exc[
i] = exc[
i] + (gain0 * iexc[
i]) + (gain1 * iexc[
i + nsf]);
798 old_ener =
fmaxf(old_ener, 1.
f);
799 new_ener =
fmaxf(new_ener, 1.
f);
800 old_ener =
fminf(old_ener, new_ener);
801 ngain = old_ener / new_ener;
803 for (
int i = 0;
i < nsf;
i++)
808 float *lsp,
int len,
int subframe,
809 int nb_subframes,
float margin)
811 const float tmp = (1.f + subframe) / nb_subframes;
813 for (
int i = 0;
i <
len;
i++) {
814 lsp[
i] = (1.f -
tmp) * old_lsp[
i] +
tmp * new_lsp[
i];
817 for (
int i = 1;
i <
len - 1;
i++) {
818 lsp[
i] =
fmaxf(lsp[
i], lsp[
i - 1] + margin);
819 if (lsp[
i] > lsp[
i + 1] - margin)
820 lsp[
i] = .5f * (lsp[
i] + lsp[
i + 1] - margin);
824 static void lsp_to_lpc(
const float *freq,
float *ak,
int lpcrdr)
826 float xout1, xout2, xin1, xin2;
830 const int m = lpcrdr >> 1;
836 for (
int i = 0;
i < lpcrdr;
i++)
837 x_freq[
i] = -
cosf(freq[
i]);
843 for (
int j = 0; j <= lpcrdr; j++) {
845 for (
int i = 0;
i < m;
i++, i2 += 2) {
847 xout1 = xin1 + 2.f * x_freq[i2 ] * n0[0] + n0[1];
848 xout2 = xin2 + 2.f * x_freq[i2 + 1] * n0[2] + n0[3];
856 xout1 = xin1 + n0[4];
857 xout2 = xin2 - n0[5];
859 ak[j - 1] = (xout1 + xout2) * 0.5
f;
872 float ol_gain = 0, ol_pitch_coef = 0, best_pitch_gain = 0, pitch_average = 0;
873 int m, pitch, wideband, ol_pitch = 0, best_pitch = 40;
880 float pitch_gain[3] = { 0 };
890 int submode, advance;
921 }
else if (m == 14) {
925 }
else if (m == 13) {
943 float innov_gain = 0.f;
962 float fact, lsp_dist = 0;
979 if (
SUBMODE(forced_pitch_gain))
980 ol_pitch_coef = 0.066667f *
get_bits(gb, 4);
992 float *exc, *innov_save =
NULL,
tmp, ener;
993 int pit_min, pit_max,
offset, q_energy;
1005 if (
SUBMODE(lbr_pitch) != -1) {
1006 int margin =
SUBMODE(lbr_pitch);
1009 pit_min = ol_pitch - margin + 1;
1011 pit_max = ol_pitch + margin;
1014 pit_min = pit_max = ol_pitch;
1021 SUBMODE(ltp_unquant)(exc, exc32, pit_min, pit_max, ol_pitch_coef,
SUBMODE(LtpParam),
1029 pitch_average +=
tmp;
1030 if ((
tmp > best_pitch_gain &&
1031 FFABS(2 * best_pitch - pitch) >= 3 &&
1032 FFABS(3 * best_pitch - pitch) >= 4 &&
1033 FFABS(4 * best_pitch - pitch) >= 5) ||
1034 (
tmp > .6
f * best_pitch_gain &&
1035 (
FFABS(best_pitch - 2 * pitch) < 3 ||
1036 FFABS(best_pitch - 3 * pitch) < 4 ||
1037 FFABS(best_pitch - 4 * pitch) < 5)) ||
1038 ((.67
f *
tmp) > best_pitch_gain &&
1039 (
FFABS(2 * best_pitch - pitch) < 3 ||
1040 FFABS(3 * best_pitch - pitch) < 4 ||
1041 FFABS(4 * best_pitch - pitch) < 5))) {
1043 if (
tmp > best_pitch_gain)
1044 best_pitch_gain =
tmp;
1047 memset(innov, 0,
sizeof(innov));
1050 if (
SUBMODE(have_subframe_gain) == 3) {
1053 }
else if (
SUBMODE(have_subframe_gain) == 1) {
1068 if (
SUBMODE(double_codebook)) {
1074 innov[
i] += innov2[
i];
1077 exc[
i] = exc32[
i] + innov[
i];
1079 memcpy(innov_save, innov,
sizeof(innov));
1083 float g = ol_pitch_coef;
1096 float exci = exc[
i];
1097 exc[
i] = (.7f * exc[
i] + .3f * st->
voc_m1) + ((1.
f - .85
f *
g) * innov[
i]) - .15
f *
g * st->
voc_m2;
1119 float exc_ener, gain;
1123 gain =
fminf(ol_gain / (exc_ener + 1.
f), 2.
f);
1138 pi_g += ak[
i + 1] - ak[
i];
1162 static void qmf_synth(
const float *x1,
const float *x2,
const float *
a,
float *y,
int N,
int M,
float *mem1,
float *mem2)
1164 const int M2 =
M >> 1,
N2 =
N >> 1;
1165 float xx1[352], xx2[352];
1167 for (
int i = 0;
i <
N2;
i++)
1168 xx1[
i] = x1[
N2-1-
i];
1169 for (
int i = 0;
i < M2;
i++)
1170 xx1[
N2+
i] = mem1[2*
i+1];
1171 for (
int i = 0;
i <
N2;
i++)
1172 xx2[
i] = x2[
N2-1-
i];
1173 for (
int i = 0;
i < M2;
i++)
1174 xx2[
N2+
i] = mem2[2*
i+1];
1176 for (
int i = 0;
i <
N2;
i += 2) {
1177 float y0, y1, y2, y3;
1180 y0 = y1 = y2 = y3 = 0.f;
1184 for (
int j = 0; j < M2; j += 2) {
1190 x11 = xx1[
N2-1+j-
i];
1191 x21 = xx2[
N2-1+j-
i];
1193 y0 +=
a0 * (x11-x21);
1194 y1 +=
a1 * (x11+x21);
1195 y2 +=
a0 * (x10-x20);
1196 y3 +=
a1 * (x10+x20);
1202 y0 +=
a0 * (x10-x20);
1203 y1 +=
a1 * (x10+x20);
1204 y2 +=
a0 * (x11-x21);
1205 y3 +=
a1 * (x11+x21);
1207 y[2 *
i ] = 2.f * y0;
1208 y[2 *
i+1] = 2.f * y1;
1209 y[2 *
i+2] = 2.f * y2;
1210 y[2 *
i+3] = 2.f * y3;
1213 for (
int i = 0;
i < M2;
i++)
1214 mem1[2*
i+1] = xx1[
i];
1215 for (
int i = 0;
i < M2;
i++)
1216 mem2[2*
i+1] = xx2[
i];
1228 float *low_innov_alias;
1237 s->st[st->
modeID - 1].innov_save = low_innov_alias;
1273 memcpy(low_pi_gain,
s->st[st->
modeID - 1].pi_gain,
sizeof(low_pi_gain));
1274 memcpy(low_exc_rms,
s->st[st->
modeID - 1].exc_rms,
sizeof(low_exc_rms));
1282 float filter_ratio, el, rl, rh;
1283 float *innov_save =
NULL, *sp;
1304 rh += ak[
i + 1] - ak[
i];
1308 rl = low_pi_gain[sub];
1309 filter_ratio = (rl + .01f) / (rh + .01
f);
1312 if (!
SUBMODE(innovation_unquant)) {
1314 const float g =
expf(.125
f * (x - 10)) / filter_ratio;
1317 exc[
i ] =
mode->folding_gain * low_innov_alias[
offset +
i ] *
g;
1318 exc[
i + 1] = -
mode->folding_gain * low_innov_alias[
offset +
i + 1] *
g;
1323 el = low_exc_rms[sub];
1329 scale = (gc * el) / filter_ratio;
1335 if (
SUBMODE(double_codebook)) {
1342 exc[
i] += innov2[
i];
1348 innov_save[2 *
i] = exc[
i];
1352 memcpy(st->
exc_buf, exc,
sizeof(exc));
1399 const uint8_t *extradata,
int extradata_size)
1402 const uint8_t *buf =
av_strnstr(extradata,
"Speex ", extradata_size);
1409 s->version_id = bytestream_get_le32(&buf);
1411 s->rate = bytestream_get_le32(&buf);
1414 s->mode = bytestream_get_le32(&buf);
1417 s->bitstream_version = bytestream_get_le32(&buf);
1418 if (
s->bitstream_version != 4)
1420 s->nb_channels = bytestream_get_le32(&buf);
1421 if (
s->nb_channels <= 0 ||
s->nb_channels > 2)
1423 s->bitrate = bytestream_get_le32(&buf);
1424 s->frame_size = bytestream_get_le32(&buf);
1426 s->frame_size > INT32_MAX >> (
s->mode > 0))
1428 s->frame_size <<= (
s->mode > 0);
1429 s->vbr = bytestream_get_le32(&buf);
1430 s->frames_per_packet = bytestream_get_le32(&buf);
1431 if (
s->frames_per_packet <= 0 ||
1432 s->frames_per_packet > 64 ||
1433 s->frames_per_packet >= INT32_MAX /
s->nb_channels /
s->frame_size)
1435 s->extra_headers = bytestream_get_le32(&buf);
1459 if (
s->nb_channels <= 0 ||
s->nb_channels > 2)
1463 case 8000:
s->mode = 0;
break;
1464 case 16000:
s->mode = 1;
break;
1465 case 32000:
s->mode = 2;
break;
1466 default:
s->mode = 2;
1469 s->frames_per_packet = 64;
1487 s->pkt_size = ((
const uint8_t[]){ 5, 10, 15, 20, 20, 28, 28, 38, 38, 46, 62 })[
quality];
1494 s->frames_per_packet = 1;
1506 for (
int m = 0; m <=
s->mode; m++) {
1512 s->stereo.balance = 1.f;
1513 s->stereo.e_ratio = .5f;
1514 s->stereo.smooth_left = 1.f;
1515 s->stereo.smooth_right = 1.f;
1522 float balance, e_left, e_right, e_ratio;
1528 e_right = 1.f /
sqrtf(e_ratio * (1.
f + balance));
1529 e_left =
sqrtf(balance) * e_right;
1541 int *got_frame_ptr,
AVPacket *avpkt)
1544 int frames_per_packet =
s->frames_per_packet;
1545 const float scale = 1.f / 32768.f;
1546 int buf_size = avpkt->
size;
1550 if (
s->pkt_size && avpkt->
size == 62)
1551 buf_size =
s->pkt_size;
1555 frame->nb_samples =
FFALIGN(
s->frame_size * frames_per_packet, 4);
1559 dst = (
float *)
frame->extended_data[0];
1560 for (
int i = 0;
i < frames_per_packet;
i++) {
1568 frames_per_packet =
i + 1;
1573 dst = (
float *)
frame->extended_data[0];
1575 frame->nb_samples =
s->frame_size * frames_per_packet;
int submodeID
Activated sub-mode.
static const SplitCodebookParams split_cb_high
static const SpeexSubmode nb_submode4
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
uint32_t seed
Seed used for random number generation.
static const float h0[64]
int have_subframe_gain
Number of bits to use as sub-frame innovation gain.
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
static unsigned int show_bits1(GetBitContext *s)
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 const SpeexSubmode wb_submode2
static const int8_t hexc_10_32_table[320]
static const SpeexSubmode nb_submode3
int count_lost
Was the last frame lost?
static const float exc_gain_quant_scal1[2]
int32_t vbr
1 for a VBR decoding, 0 otherwise
int sample_rate
samples per second
float exc_buf[NB_DEC_BUFFER]
Excitation buffer.
int highpass_enabled
Is the input filter enabled.
static const int8_t hexc_table[1024]
int(* ltp_quant_func)(float *, float *, float *, float *, float *, float *, const void *, int, int, float, int, int, GetBitContext *, char *, float *, float *, int, int, int, float *)
Long-term predictor quantization.
float mem_hp[2]
High-pass filter memory.
static int get_bits_count(const GetBitContext *s)
static const int8_t exc_8_128_table[1024]
int32_t version_id
Version for Speex (for checking compatibility)
static const int8_t cdbk_nb_high1[320]
int modeID
ID of the mode.
int lpc_enh_enabled
1 when LPC enhancer is on, 0 otherwise
This structure describes decoded (raw) audio or video data.
float * exc
Start of excitation frame.
enum AVChannelOrder order
Channel order used in this layout.
int lpc_size
Order of LPC filter.
static const SpeexSubmode nb_submode8
int nb_channels
Number of channels in this layout.
int double_codebook
Apply innovation quantization twice for higher quality (and higher bit-rate)
static int speex_inband_handler(GetBitContext *gb, void *state, StereoState *stereo)
#define gain_3tap_to_1tap(g)
trying all byte sequences megabyte in length and selecting the best looking sequence will yield cases to try But a word about quality
static const SpeexSubmode wb_submode4
int subframe_size
Size of sub-frames used for decoding.
const void * LtpParam
Pitch parameters (options)
int32_t nb_channels
Number of channels decoded.
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
AVCodec p
The public AVCodec.
static const int8_t exc_5_256_table[1280]
#define LSP_LINEAR_HIGH(i)
AVChannelLayout ch_layout
Audio channel layout.
static int speex_default_user_handler(GetBitContext *gb, void *state, void *data)
static av_always_inline float av_int2float(uint32_t i)
Reinterpret a 32-bit integer as a float.
ltp_unquant_func ltp_unquant
Long-term predictor (pitch) un-quantizer.
void(* innovation_quant_func)(float *, float *, float *, float *, const void *, int, int, float *, float *, GetBitContext *, char *, int, int)
Innovation quantization function.
static const SplitCodebookParams split_cb_nb_lbr
int nb_subframes
Number of high-band sub-frames.
static __device__ float fabsf(float a)
static const SpeexSubmode wb_submode3
int32_t bitrate
Bit-rate used.
static const float e_ratio_quant[4]
const FFCodec ff_speex_decoder
static const SplitCodebookParams split_cb_nb_ulbr
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
float balance
Left/right balance info.
static void lsp_interpolate(const float *old_lsp, const float *new_lsp, float *lsp, int len, int subframe, int nb_subframes, float margin)
#define FF_CODEC_DECODE_CB(func)
static const SplitCodebookParams split_cb_sb
static int nb_decode(AVCodecContext *, void *, GetBitContext *, float *)
static int speex_std_stereo(GetBitContext *gb, void *state, void *data)
@ AV_CHANNEL_ORDER_UNSPEC
Only the channel count is specified, without any further information about the channel order.
static const int8_t gain_cdbk_lbr[128]
float fminf(float, float)
#define av_assert0(cond)
assert() equivalent, that is always enabled.
static const SpeexSubmode nb_submode7
static float speex_rand(float std, uint32_t *seed)
static const int8_t cdbk_nb_low2[320]
static const SpeexMode speex_modes[SPEEX_NB_MODES]
int modeID
ID of the decoder mode.
#define CODEC_LONG_NAME(str)
static const SpeexSubmode nb_submode6
innovation_unquant_func innovation_unquant
Innovation un-quantization.
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
float mem_sp[NB_ORDER]
Filter memory for synthesis signal.
#define SPEEX_MEMSET(dst, c, n)
static void lsp_to_lpc(const float *freq, float *ak, int lpcrdr)
static int speex_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
static const SpeexSubmode nb_submode1
Describe the class of an AVClass context structure.
int32_t frames_per_packet
Number of frames stored per Ogg packet.
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
int lpc_size
Order of high-band LPC analysis.
int default_submode
Default sub-mode to use when decoding.
int64_t bit_rate
the average bitrate
static unsigned int get_bits1(GetBitContext *s)
float exc_rms[NB_NB_SUBFRAMES]
RMS of excitation per subframe.
static const SplitCodebookParams split_cb_nb
static __device__ float sqrtf(float a)
int32_t bitstream_version
Version ID of the bit-stream.
static const int8_t exc_10_32_table[320]
int32_t extra_headers
Number of additional headers after the comments.
static const LtpParam ltp_params_nb
static const uint16_t wb_skip_table[8]
float comb_gain
Gain of enhancer comb filter.
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
static void lsp_unquant_nb(float *lsp, int order, GetBitContext *gb)
static const int8_t exc_5_64_table[320]
static const LtpParam ltp_params_lbr
uint8_t ptrdiff_t const uint8_t ptrdiff_t int intptr_t intptr_t int int16_t * dst
static const LtpParam ltp_params_med
static void sanitize_values(float *vec, float min_val, float max_val, int len)
float folding_gain
Folding gain.
void(* ltp_unquant_func)(float *, float *, int, int, float, const void *, int, int *, float *, GetBitContext *, int, int, float, int)
Long-term un-quantize.
float fmaxf(float, float)
enum AVSampleFormat sample_fmt
audio sample format
const SpeexSubmode *const * submodes
Sub-mode data.
static void signal_mul(const float *x, float *y, float scale, int len)
float old_qlsp[NB_ORDER]
Quantized LSPs for previous frame.
int frame_size
Length of high-band frames.
static void noise_codebook_unquant(float *exc, const void *par, int nsf, GetBitContext *gb, uint32_t *seed)
static void pitch_unquant_3tap(float *exc, float *exc_out, int start, int end, float pitch_coef, const void *par, int nsf, int *pitch_val, float *gain_val, GetBitContext *gb, int count_lost, int subframe_offset, float last_pitch_gain, int cdbk_offset)
static const int8_t gain_cdbk_nb[512]
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
static double a0(void *priv, double x, double y)
int frame_size
Size of frames used for decoding.
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 const int8_t cdbk_nb_high2[320]
static double fact(double i)
#define SPEEX_COPY(dst, src, n)
int subframe_size
Length of high-band sub-frames.
const void * innovation_params
Innovation quantization parameters.
static uint32_t ran(void)
static const int8_t exc_20_32_table[640]
static const float shift_filt[3][7]
static void multicomb(const float *exc, float *new_exc, float *ak, int p, int nsf, int pitch, int max_pitch, float comb_gain)
const signed char * shape_cb
void(* lsp_quant_func)(float *, float *, int, GetBitContext *)
Quantizes LSPs.
static void lsp_unquant_lbr(float *lsp, int order, GetBitContext *gb)
static const SplitCodebookParams split_cb_nb_med
static void forced_pitch_unquant(float *exc, float *exc_out, int start, int end, float pitch_coef, const void *par, int nsf, int *pitch_val, float *gain_val, GetBitContext *gb, int count_lost, int subframe_offset, float last_pitch_gain, int cdbk_offset)
static const SpeexSubmode nb_submode5
#define i(width, name, range_min, range_max)
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.
float interp_qlpc[NB_ORDER]
Interpolated quantized LPCs.
void(* lsp_unquant_func)(float *, int, GetBitContext *)
Decodes quantized LSPs.
static void iir_mem(const float *x, const float *den, float *y, int N, int ord, float *mem)
int full_frame_size
Length of full-band frames.
const char * name
Name of the codec implementation.
static float inner_prod(const float *x, const float *y, int len)
static const int8_t cdbk_nb[640]
static int decoder_init(SpeexContext *s, DecoderState *st, const SpeexMode *mode)
#define SPEEX_INBAND_STEREO
static int parse_speex_extradata(AVCodecContext *avctx, const uint8_t *extradata, int extradata_size)
lsp_unquant_func lsp_unquant
LSP unquantization function.
static const SplitCodebookParams split_cb_nb_vlbr
int(* decode)(AVCodecContext *avctx, void *dec, GetBitContext *gb, float *out)
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
char * av_strnstr(const char *haystack, const char *needle, size_t hay_length)
Locate the first occurrence of the string needle in the string haystack where not more than hay_lengt...
float smooth_right
Smoothed right channel gain.
static const float gc_quant_bound[16]
int last_pitch
Pitch of last correctly decoded frame.
float smooth_left
Smoothed left channel gain.
main external API structure.
const SpeexSubmode * submodes[NB_SUBMODES]
Sub-mode data for the mode.
float last_ol_gain
Open-loop gain for previous frame.
static const int8_t cdbk_nb_low1[320]
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
int is_wideband
If wideband is present.
static av_cold int speex_decode_close(AVCodecContext *avctx)
static av_always_inline int get_bitsz(GetBitContext *s, int n)
Read 0-25 bits.
static const int8_t high_lsp_cdbk2[512]
int32_t mode
Mode used (0 for narrowband, 1 for wideband)
DecoderState st[SPEEX_NB_MODES]
static const SpeexSubmode nb_submode2
static const LtpParam ltp_params_vlbr
int forced_pitch_gain
Use the same (forced) pitch gain for all sub-frames.
unsigned int codec_tag
fourcc (LSB first, so "ABCD" -> ('D'<<24) + ('C'<<16) + ('B'<<8) + 'A').
static void scale(int *out, const int *in, const int w, const int h, const int shift)
static const SpeexSubmode wb_submode1
static void highpass(const float *x, float *y, int len, float *mem, int wide)
This structure stores compressed data.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
static void speex_decode_stereo(float *data, int frame_size, StereoState *stereo)
void(* innovation_unquant_func)(float *, const void *, int, GetBitContext *, uint32_t *)
Innovation unquantization function.
int lbr_pitch
Set to -1 for "normal" modes, otherwise encode pitch using a global pitch and allowing a +- lbr_pitch...
static av_cold int speex_decode_init(AVCodecContext *avctx)
int32_t frame_size
Size of frames.
static void lsp_unquant_high(float *lsp, int order, GetBitContext *gb)
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
static const int8_t exc_10_16_table[160]
static void qmf_synth(const float *x1, const float *x2, const float *a, float *y, int N, int M, float *mem1, float *mem2)
static const float exc_gain_quant_scal3[8]
static double a1(void *priv, double x, double y)
#define MKTAG(a, b, c, d)
float last_pitch_gain
Pitch gain of last correctly decoded frame.
static const SplitCodebookParams split_cb_high_lbr
float pi_gain[NB_NB_SUBFRAMES]
Gain of LPC filter at theta=pi (fe/2)
int32_t rate
Sampling rate used.
static void bw_lpc(float gamma, const float *lpc_in, float *lpc_out, int order)
static int interp_pitch(const float *exc, float *interp, int pitch, int len)
static float compute_rms(const float *x, int len)
static int sb_decode(AVCodecContext *, void *, GetBitContext *, float *)
float * innov_save
If non-NULL, innovation is copied here.
float e_ratio
Ratio of energies: E(left+right)/[E(left)+E(right)]
static void split_cb_shape_sign_unquant(float *exc, const void *par, int nsf, GetBitContext *gb, uint32_t *seed)
static const int8_t high_lsp_cdbk[512]