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62 #define SPEEX_NB_MODES 3
63 #define SPEEX_INBAND_STEREO 9
67 #define NB_FRAME_SIZE 160
69 #define NB_SUBMODE_BITS 4
70 #define SB_SUBMODE_BITS 3
72 #define NB_SUBFRAME_SIZE 40
73 #define NB_NB_SUBFRAMES 4
74 #define NB_PITCH_START 17
75 #define NB_PITCH_END 144
77 #define NB_DEC_BUFFER (NB_FRAME_SIZE + 2 * NB_PITCH_END + NB_SUBFRAME_SIZE + 12)
79 #define SPEEX_MEMSET(dst, c, n) (memset((dst), (c), (n) * sizeof(*(dst))))
80 #define SPEEX_COPY(dst, src, n) (memcpy((dst), (src), (n) * sizeof(*(dst))))
82 #define LSP_LINEAR(i) (.25f * (i) + .25f)
83 #define LSP_LINEAR_HIGH(i) (.3125f * (i) + .75f)
84 #define LSP_DIV_256(x) (0.00390625f * (x))
85 #define LSP_DIV_512(x) (0.001953125f * (x))
86 #define LSP_DIV_1024(x) (0.0009765625f * (x))
124 float *,
float *,
float *,
131 float,
const void *,
int,
int *,
137 float *,
float *,
const void *,
138 int,
int,
float *,
float *,
225 const int req_size =
get_bits(gb, 4);
264 for (
int i = 0;
i < order;
i++)
268 for (
int i = 0;
i < 10;
i++)
272 for (
int i = 0;
i < 5;
i++)
276 for (
int i = 0;
i < 5;
i++)
281 float pitch_coef,
const void *par,
int nsf,
282 int *pitch_val,
float *gain_val,
GetBitContext *gb,
int count_lost,
283 int subframe_offset,
float last_pitch_gain,
int cdbk_offset)
286 pitch_coef =
fminf(pitch_coef, .99
f);
287 for (
int i = 0;
i < nsf;
i++) {
288 exc_out[
i] = exc[
i - start] * pitch_coef;
291 pitch_val[0] = start;
292 gain_val[0] = gain_val[2] = 0.f;
293 gain_val[1] = pitch_coef;
298 const uint32_t jflone = 0x3f800000;
299 const uint32_t jflmsk = 0x007fffff;
302 seed[0] = 1664525 *
seed[0] + 1013904223;
303 ran = jflone | (jflmsk &
seed[0]);
313 for (
int i = 0;
i < nsf;
i++)
320 int subvect_size, nb_subvect, have_sign, shape_bits;
322 const signed char *shape_cb;
323 int signs[10], ind[10];
334 for (
int i = 0;
i < nb_subvect;
i++) {
339 for (
int i = 0;
i < nb_subvect;
i++) {
340 const float s = signs[
i] ? -1.f : 1.f;
342 for (
int j = 0; j < subvect_size; j++)
343 exc[subvect_size *
i + j] +=
s * 0.03125
f * shape_cb[ind[
i] * subvect_size + j];
347 #define SUBMODE(x) st->submodes[st->submodeID]->x
349 #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]))
353 const void *par,
int nsf,
int *pitch_val,
float *gain_val,
GetBitContext *gb,
354 int count_lost,
int subframe_offset,
float last_pitch_gain,
int cdbk_offset)
356 int pitch, gain_index, gain_cdbk_size;
357 const int8_t *gain_cdbk;
358 const LtpParam *params;
361 params = (
const LtpParam *)par;
362 gain_cdbk_size = 1 << params->gain_bits;
363 gain_cdbk = params->gain_cdbk + 4 * gain_cdbk_size * cdbk_offset;
365 pitch =
get_bitsz(gb, params->pitch_bits);
367 gain_index =
get_bitsz(gb, params->gain_bits);
368 gain[0] = 0.015625f * gain_cdbk[gain_index * 4] + .5f;
369 gain[1] = 0.015625f * gain_cdbk[gain_index * 4 + 1] + .5f;
370 gain[2] = 0.015625f * gain_cdbk[gain_index * 4 + 2] + .5f;
372 if (count_lost && pitch > subframe_offset) {
373 float tmp = count_lost < 4 ? last_pitch_gain : 0.5f * last_pitch_gain;
379 if (gain_sum >
tmp && gain_sum > 0.
f) {
381 for (
int i = 0;
i < 3;
i++)
386 pitch_val[0] = pitch;
387 gain_val[0] = gain[0];
388 gain_val[1] = gain[1];
389 gain_val[2] = gain[2];
392 for (
int i = 0;
i < 3;
i++) {
394 int pp = pitch + 1 -
i;
398 for (
int j = 0; j < tmp1; j++)
399 exc_out[j] += gain[2 -
i] * exc[j - pp];
401 if (tmp3 > pp + pitch)
403 for (
int j = tmp1; j < tmp3; j++)
404 exc_out[j] += gain[2 -
i] * exc[j - pp - pitch];
412 for (
int i = 0;
i < order;
i++)
416 for (
int i = 0;
i < 10;
i++)
420 for (
int i = 0;
i < 5;
i++)
424 for (
int i = 0;
i < 5;
i++)
428 for (
int i = 0;
i < 5;
i++)
432 for (
int i = 0;
i < 5;
i++)
440 for (
int i = 0;
i < order;
i++)
444 for (
int i = 0;
i < order;
i++)
448 for (
int i = 0;
i < order;
i++)
534 .default_submode = 5,
542 .folding_gain = 0.9f,
546 .default_submode = 3,
554 .folding_gain = 0.7f,
558 .default_submode = 1,
566 for (
int i = 0;
i <
len;
i++)
570 return sqrtf(.1
f + sum /
len);
573 static void bw_lpc(
float gamma,
const float *lpc_in,
574 float *lpc_out,
int order)
578 for (
int i = 0;
i < order;
i++) {
579 lpc_out[
i] =
tmp * lpc_in[
i];
584 static void iir_mem(
const float *x,
const float *den,
585 float *y,
int N,
int ord,
float *mem)
587 for (
int i = 0;
i <
N;
i++) {
588 float yi = x[
i] + mem[0];
590 for (
int j = 0; j < ord - 1; j++)
591 mem[j] = mem[j + 1] + den[j] * nyi;
592 mem[ord - 1] = den[ord - 1] * nyi;
597 static void highpass(
const float *x,
float *y,
int len,
float *mem,
int wide)
599 static const float Pcoef[2][3] = {{ 1.00000f, -1.92683f, 0.93071f }, { 1.00000f, -1.97226f, 0.97332f } };
600 static const float Zcoef[2][3] = {{ 0.96446f, -1.92879f, 0.96446f }, { 0.98645f, -1.97277f, 0.98645f } };
601 const float *den, *num;
605 for (
int i = 0;
i <
len;
i++) {
606 float yi = num[0] * x[
i] + mem[0];
607 mem[0] = mem[1] + num[1] * x[
i] + -den[1] * yi;
608 mem[1] = num[2] * x[
i] + -den[2] * yi;
613 #define median3(a, b, c) \
614 ((a) < (b) ? ((b) < (c) ? (b) : ((a) < (c) ? (c) : (a))) \
615 : ((c) < (b) ? (b) : ((c) < (a) ? (c) : (a))))
656 for (
int i = 0;
i <
len;
i++) {
657 if (!isnormal(vec[
i]) ||
fabsf(vec[
i]) < 1e-8
f)
666 for (
int i = 0;
i <
len;
i++)
674 for (
int i = 0;
i <
len;
i += 8) {
676 part += x[
i + 0] * y[
i + 0];
677 part += x[
i + 1] * y[
i + 1];
678 part += x[
i + 2] * y[
i + 2];
679 part += x[
i + 3] * y[
i + 3];
680 part += x[
i + 4] * y[
i + 4];
681 part += x[
i + 5] * y[
i + 5];
682 part += x[
i + 6] * y[
i + 6];
683 part += x[
i + 7] * y[
i + 7];
692 float corr[4][7], maxcorr;
695 for (
int i = 0;
i < 7;
i++)
697 for (
int i = 0;
i < 3;
i++) {
698 for (
int j = 0; j < 7; j++) {
708 for (
int k = i1; k < i2; k++)
710 corr[
i + 1][j] =
tmp;
714 maxcorr = corr[0][0];
715 for (
int i = 0;
i < 4;
i++) {
716 for (
int j = 0; j < 7; j++) {
717 if (corr[
i][j] > maxcorr) {
718 maxcorr = corr[
i][j];
724 for (
int i = 0;
i <
len;
i++) {
727 for (
int k = 0; k < 7; k++)
728 tmp += exc[
i - (pitch - maxj + 3) + k - 3] *
shift_filt[maxi - 1][k];
730 tmp = exc[
i - (pitch - maxj + 3)];
734 return pitch - maxj + 3;
737 static void multicomb(
const float *exc,
float *new_exc,
float *ak,
int p,
int nsf,
738 int pitch,
int max_pitch,
float comb_gain)
740 float old_ener, new_ener;
741 float iexc0_mag, iexc1_mag, exc_mag;
743 float corr0, corr1, gain0, gain1;
744 float pgain1, pgain2;
745 float c1,
c2, g1, g2;
746 float ngain, gg1, gg2;
747 int corr_pitch = pitch;
750 if (corr_pitch > max_pitch)
755 iexc0_mag = sqrtf(1000.
f +
inner_prod(iexc, iexc, nsf));
756 iexc1_mag = sqrtf(1000.
f +
inner_prod(iexc + nsf, iexc + nsf, nsf));
760 if (corr0 > iexc0_mag * exc_mag)
763 pgain1 = (corr0 / exc_mag) / iexc0_mag;
764 if (corr1 > iexc1_mag * exc_mag)
767 pgain2 = (corr1 / exc_mag) / iexc1_mag;
768 gg1 = exc_mag / iexc0_mag;
769 gg2 = exc_mag / iexc1_mag;
770 if (comb_gain > 0.
f) {
771 c1 = .4f * comb_gain + .07f;
772 c2 = .5f + 1.72f * (
c1 - .07f);
776 g1 = 1.f -
c2 * pgain1 * pgain1;
777 g2 = 1.f -
c2 * pgain2 * pgain2;
783 if (corr_pitch > max_pitch) {
784 gain0 = .7f * g1 * gg1;
785 gain1 = .3f * g2 * gg2;
787 gain0 = .6f * g1 * gg1;
788 gain1 = .6f * g2 * gg2;
790 for (
int i = 0;
i < nsf;
i++)
791 new_exc[
i] = exc[
i] + (gain0 * iexc[
i]) + (gain1 * iexc[
i + nsf]);
795 old_ener =
fmaxf(old_ener, 1.
f);
796 new_ener =
fmaxf(new_ener, 1.
f);
797 old_ener =
fminf(old_ener, new_ener);
798 ngain = old_ener / new_ener;
800 for (
int i = 0;
i < nsf;
i++)
805 float *lsp,
int len,
int subframe,
806 int nb_subframes,
float margin)
808 const float tmp = (1.f + subframe) / nb_subframes;
810 for (
int i = 0;
i <
len;
i++) {
811 lsp[
i] = (1.f -
tmp) * old_lsp[
i] +
tmp * new_lsp[
i];
814 for (
int i = 1;
i <
len - 1;
i++) {
815 lsp[
i] =
fmaxf(lsp[
i], lsp[
i - 1] + margin);
816 if (lsp[
i] > lsp[
i + 1] - margin)
817 lsp[
i] = .5f * (lsp[
i] + lsp[
i + 1] - margin);
821 static void lsp_to_lpc(
const float *freq,
float *ak,
int lpcrdr)
823 float xout1, xout2, xin1, xin2;
827 const int m = lpcrdr >> 1;
833 for (
int i = 0;
i < lpcrdr;
i++)
834 x_freq[
i] = -
cosf(freq[
i]);
840 for (
int j = 0; j <= lpcrdr; j++) {
842 for (
int i = 0;
i < m;
i++, i2 += 2) {
844 xout1 = xin1 + 2.f * x_freq[i2 ] * n0[0] + n0[1];
845 xout2 = xin2 + 2.f * x_freq[i2 + 1] * n0[2] + n0[3];
853 xout1 = xin1 + n0[4];
854 xout2 = xin2 - n0[5];
856 ak[j - 1] = (xout1 + xout2) * 0.5
f;
869 float ol_gain = 0, ol_pitch_coef = 0, best_pitch_gain = 0, pitch_average = 0;
870 int m, pitch, wideband, ol_pitch = 0, best_pitch = 40;
877 float pitch_gain[3] = { 0 };
887 int submode, advance;
918 }
else if (m == 14) {
922 }
else if (m == 13) {
940 float innov_gain = 0.f;
959 float fact, lsp_dist = 0;
976 if (
SUBMODE(forced_pitch_gain))
977 ol_pitch_coef = 0.066667f *
get_bits(gb, 4);
989 float *exc, *innov_save =
NULL,
tmp, ener;
990 int pit_min, pit_max,
offset, q_energy;
1002 if (
SUBMODE(lbr_pitch) != -1) {
1003 int margin =
SUBMODE(lbr_pitch);
1006 pit_min = ol_pitch - margin + 1;
1008 pit_max = ol_pitch + margin;
1011 pit_min = pit_max = ol_pitch;
1018 SUBMODE(ltp_unquant)(exc, exc32, pit_min, pit_max, ol_pitch_coef,
SUBMODE(LtpParam),
1026 pitch_average +=
tmp;
1027 if ((
tmp > best_pitch_gain &&
1028 FFABS(2 * best_pitch - pitch) >= 3 &&
1029 FFABS(3 * best_pitch - pitch) >= 4 &&
1030 FFABS(4 * best_pitch - pitch) >= 5) ||
1031 (
tmp > .6
f * best_pitch_gain &&
1032 (
FFABS(best_pitch - 2 * pitch) < 3 ||
1033 FFABS(best_pitch - 3 * pitch) < 4 ||
1034 FFABS(best_pitch - 4 * pitch) < 5)) ||
1035 ((.67
f *
tmp) > best_pitch_gain &&
1036 (
FFABS(2 * best_pitch - pitch) < 3 ||
1037 FFABS(3 * best_pitch - pitch) < 4 ||
1038 FFABS(4 * best_pitch - pitch) < 5))) {
1040 if (
tmp > best_pitch_gain)
1041 best_pitch_gain =
tmp;
1044 memset(innov, 0,
sizeof(innov));
1047 if (
SUBMODE(have_subframe_gain) == 3) {
1050 }
else if (
SUBMODE(have_subframe_gain) == 1) {
1065 if (
SUBMODE(double_codebook)) {
1071 innov[
i] += innov2[
i];
1074 exc[
i] = exc32[
i] + innov[
i];
1076 memcpy(innov_save, innov,
sizeof(innov));
1080 float g = ol_pitch_coef;
1087 exc[st->
voc_offset] = sqrtf(2.
f * ol_pitch) * (
g * ol_gain);
1093 float exci = exc[
i];
1094 exc[
i] = (.7f * exc[
i] + .3f * st->
voc_m1) + ((1.
f - .85
f *
g) * innov[
i]) - .15
f *
g * st->
voc_m2;
1116 float exc_ener, gain;
1120 gain =
fminf(ol_gain / (exc_ener + 1.
f), 2.
f);
1135 pi_g += ak[
i + 1] - ak[
i];
1159 static void qmf_synth(
const float *x1,
const float *x2,
const float *
a,
float *y,
int N,
int M,
float *mem1,
float *mem2)
1161 const int M2 =
M >> 1,
N2 =
N >> 1;
1162 float xx1[352], xx2[352];
1164 for (
int i = 0;
i <
N2;
i++)
1165 xx1[
i] = x1[
N2-1-
i];
1166 for (
int i = 0;
i < M2;
i++)
1167 xx1[
N2+
i] = mem1[2*
i+1];
1168 for (
int i = 0;
i <
N2;
i++)
1169 xx2[
i] = x2[
N2-1-
i];
1170 for (
int i = 0;
i < M2;
i++)
1171 xx2[
N2+
i] = mem2[2*
i+1];
1173 for (
int i = 0;
i <
N2;
i += 2) {
1174 float y0, y1, y2, y3;
1177 y0 = y1 = y2 = y3 = 0.f;
1181 for (
int j = 0; j < M2; j += 2) {
1187 x11 = xx1[
N2-1+j-
i];
1188 x21 = xx2[
N2-1+j-
i];
1190 y0 +=
a0 * (x11-x21);
1191 y1 +=
a1 * (x11+x21);
1192 y2 +=
a0 * (x10-x20);
1193 y3 +=
a1 * (x10+x20);
1199 y0 +=
a0 * (x10-x20);
1200 y1 +=
a1 * (x10+x20);
1201 y2 +=
a0 * (x11-x21);
1202 y3 +=
a1 * (x11+x21);
1204 y[2 *
i ] = 2.f * y0;
1205 y[2 *
i+1] = 2.f * y1;
1206 y[2 *
i+2] = 2.f * y2;
1207 y[2 *
i+3] = 2.f * y3;
1210 for (
int i = 0;
i < M2;
i++)
1211 mem1[2*
i+1] = xx1[
i];
1212 for (
int i = 0;
i < M2;
i++)
1213 mem2[2*
i+1] = xx2[
i];
1225 float *low_innov_alias;
1234 s->st[st->
modeID - 1].innov_save = low_innov_alias;
1270 memcpy(low_pi_gain,
s->st[st->
modeID - 1].pi_gain,
sizeof(low_pi_gain));
1271 memcpy(low_exc_rms,
s->st[st->
modeID - 1].exc_rms,
sizeof(low_exc_rms));
1279 float filter_ratio, el, rl, rh;
1280 float *innov_save =
NULL, *
sp;
1301 rh += ak[
i + 1] - ak[
i];
1305 rl = low_pi_gain[
sub];
1306 filter_ratio = (rl + .01f) / (rh + .01
f);
1309 if (!
SUBMODE(innovation_unquant)) {
1311 const float g =
expf(.125
f * (x - 10)) / filter_ratio;
1314 exc[
i ] =
mode->folding_gain * low_innov_alias[
offset +
i ] *
g;
1315 exc[
i + 1] = -
mode->folding_gain * low_innov_alias[
offset +
i + 1] *
g;
1320 el = low_exc_rms[
sub];
1326 scale = (gc * el) / filter_ratio;
1332 if (
SUBMODE(double_codebook)) {
1339 exc[
i] += innov2[
i];
1345 innov_save[2 *
i] = exc[
i];
1349 memcpy(st->
exc_buf, exc,
sizeof(exc));
1396 const uint8_t *extradata,
int extradata_size)
1399 const uint8_t *buf = extradata;
1401 if (memcmp(buf,
"Speex ", 8))
1406 s->version_id = bytestream_get_le32(&buf);
1408 s->rate = bytestream_get_le32(&buf);
1411 s->mode = bytestream_get_le32(&buf);
1414 s->bitstream_version = bytestream_get_le32(&buf);
1415 if (
s->bitstream_version != 4)
1417 s->nb_channels = bytestream_get_le32(&buf);
1418 if (
s->nb_channels <= 0 ||
s->nb_channels > 2)
1420 s->bitrate = bytestream_get_le32(&buf);
1421 s->frame_size = bytestream_get_le32(&buf);
1424 s->vbr = bytestream_get_le32(&buf);
1425 s->frames_per_packet = bytestream_get_le32(&buf);
1426 if (
s->frames_per_packet <= 0 ||
1427 s->frames_per_packet > 64 ||
1428 s->frames_per_packet >= INT32_MAX /
s->nb_channels /
s->frame_size)
1430 s->extra_headers = bytestream_get_le32(&buf);
1454 if (
s->nb_channels <= 0)
1458 case 8000:
s->mode = 0;
break;
1459 case 16000:
s->mode = 1;
break;
1460 case 32000:
s->mode = 2;
break;
1461 default:
s->mode = 2;
1464 s->frames_per_packet = 1;
1482 s->pkt_size = ((
const uint8_t[]){ 5, 10, 15, 20, 20, 28, 28, 38, 38, 46, 62 })[
quality];
1489 s->frames_per_packet = 1;
1499 for (
int m = 0; m <=
s->mode; m++) {
1505 s->stereo.balance = 1.f;
1506 s->stereo.e_ratio = .5f;
1507 s->stereo.smooth_left = 1.f;
1508 s->stereo.smooth_right = 1.f;
1515 float balance, e_left, e_right, e_ratio;
1521 e_right = 1.f / sqrtf(e_ratio * (1.
f + balance));
1522 e_left = sqrtf(balance) * e_right;
1534 int *got_frame_ptr,
AVPacket *avpkt)
1538 const float scale = 1.f / 32768.f;
1539 int buf_size = avpkt->
size;
1543 if (
s->pkt_size && avpkt->
size == 62)
1544 buf_size =
s->pkt_size;
1548 frame->nb_samples =
FFALIGN(
s->frame_size *
s->frames_per_packet, 4);
1552 dst = (
float *)
frame->extended_data[0];
1553 for (
int i = 0;
i <
s->frames_per_packet;
i++) {
1561 dst = (
float *)
frame->extended_data[0];
1562 s->fdsp->vector_fmul_scalar(dst, dst,
scale,
frame->nb_samples *
frame->channels);
1563 frame->nb_samples =
s->frame_size *
s->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]
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
int have_subframe_gain
Number of bits to use as sub-frame innovation gain.
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]
const AVCodec ff_speex_decoder
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]
static float sub(float src0, float src1)
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 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.
int lpc_size
Order of LPC filter.
static const SpeexSubmode nb_submode8
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.
static const int8_t exc_5_256_table[1280]
#define LSP_LINEAR_HIGH(i)
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 av_always_inline float scale(float x, float s)
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]
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)
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)
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.
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 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
int32_t bitstream_version
Version ID of the bit-stream.
static const int8_t exc_10_32_table[320]
static int speex_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
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.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
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
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
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.
int channels
number of audio channels
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.
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
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
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]
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 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]
#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]