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FFmpeg/libavcodec/dss_sp.c
Andreas Rheinhardt a247ac640d avcodec: Constify AVCodecs 
Given that the AVCodec.next pointer has now been removed, most of the
AVCodecs are not modified at all any more and can therefore be made
const (as this patch does); the only exceptions are the very few codecs
for external libraries that have a init_static_data callback.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
2021-04-27 10:43:15 -03:00

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/*
* Digital Speech Standard - Standard Play mode (DSS SP) audio decoder.
* Copyright (C) 2014 Oleksij Rempel <linux@rempel-privat.de>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/mem.h"
#include "libavutil/mem_internal.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "get_bits.h"
#include "internal.h"
#define SUBFRAMES 4
#define PULSE_MAX 8
#define DSS_SP_FRAME_SIZE 42
#define DSS_SP_SAMPLE_COUNT (66 * SUBFRAMES)
#define DSS_SP_FORMULA(a, b, c) ((int)((((a) * (1 << 15)) + (b) * (unsigned)(c)) + 0x4000) >> 15)
typedef struct DssSpSubframe {
int16_t gain;
int32_t combined_pulse_pos;
int16_t pulse_pos[7];
int16_t pulse_val[7];
} DssSpSubframe;
typedef struct DssSpFrame {
int16_t filter_idx[14];
int16_t sf_adaptive_gain[SUBFRAMES];
int16_t pitch_lag[SUBFRAMES];
struct DssSpSubframe sf[SUBFRAMES];
} DssSpFrame;
typedef struct DssSpContext {
AVCodecContext *avctx;
int32_t excitation[288 + 6];
int32_t history[187];
DssSpFrame fparam;
int32_t working_buffer[SUBFRAMES][72];
int32_t audio_buf[15];
int32_t err_buf1[15];
int32_t lpc_filter[14];
int32_t filter[15];
int32_t vector_buf[72];
int noise_state;
int32_t err_buf2[15];
int pulse_dec_mode;
DECLARE_ALIGNED(16, uint8_t, bits)[DSS_SP_FRAME_SIZE +
AV_INPUT_BUFFER_PADDING_SIZE];
} DssSpContext;
/*
* Used for the coding/decoding of the pulse positions for the MP-MLQ codebook.
*/
static const uint32_t dss_sp_combinatorial_table[PULSE_MAX][72] = {
{ 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0 },
{ 0, 1, 2, 3, 4, 5,
6, 7, 8, 9, 10, 11,
12, 13, 14, 15, 16, 17,
18, 19, 20, 21, 22, 23,
24, 25, 26, 27, 28, 29,
30, 31, 32, 33, 34, 35,
36, 37, 38, 39, 40, 41,
42, 43, 44, 45, 46, 47,
48, 49, 50, 51, 52, 53,
54, 55, 56, 57, 58, 59,
60, 61, 62, 63, 64, 65,
66, 67, 68, 69, 70, 71 },
{ 0, 0, 1, 3, 6, 10,
15, 21, 28, 36, 45, 55,
66, 78, 91, 105, 120, 136,
153, 171, 190, 210, 231, 253,
276, 300, 325, 351, 378, 406,
435, 465, 496, 528, 561, 595,
630, 666, 703, 741, 780, 820,
861, 903, 946, 990, 1035, 1081,
1128, 1176, 1225, 1275, 1326, 1378,
1431, 1485, 1540, 1596, 1653, 1711,
1770, 1830, 1891, 1953, 2016, 2080,
2145, 2211, 2278, 2346, 2415, 2485 },
{ 0, 0, 0, 1, 4, 10,
20, 35, 56, 84, 120, 165,
220, 286, 364, 455, 560, 680,
816, 969, 1140, 1330, 1540, 1771,
2024, 2300, 2600, 2925, 3276, 3654,
4060, 4495, 4960, 5456, 5984, 6545,
7140, 7770, 8436, 9139, 9880, 10660,
11480, 12341, 13244, 14190, 15180, 16215,
17296, 18424, 19600, 20825, 22100, 23426,
24804, 26235, 27720, 29260, 30856, 32509,
34220, 35990, 37820, 39711, 41664, 43680,
45760, 47905, 50116, 52394, 54740, 57155 },
{ 0, 0, 0, 0, 1, 5,
15, 35, 70, 126, 210, 330,
495, 715, 1001, 1365, 1820, 2380,
3060, 3876, 4845, 5985, 7315, 8855,
10626, 12650, 14950, 17550, 20475, 23751,
27405, 31465, 35960, 40920, 46376, 52360,
58905, 66045, 73815, 82251, 91390, 101270,
111930, 123410, 135751, 148995, 163185, 178365,
194580, 211876, 230300, 249900, 270725, 292825,
316251, 341055, 367290, 395010, 424270, 455126,
487635, 521855, 557845, 595665, 635376, 677040,
720720, 766480, 814385, 864501, 916895, 971635 },
{ 0, 0, 0, 0, 0, 1,
6, 21, 56, 126, 252, 462,
792, 1287, 2002, 3003, 4368, 6188,
8568, 11628, 15504, 20349, 26334, 33649,
42504, 53130, 65780, 80730, 98280, 118755,
142506, 169911, 201376, 237336, 278256, 324632,
376992, 435897, 501942, 575757, 658008, 749398,
850668, 962598, 1086008, 1221759, 1370754, 1533939,
1712304, 1906884, 2118760, 2349060, 2598960, 2869685,
3162510, 3478761, 3819816, 4187106, 4582116, 5006386,
5461512, 5949147, 6471002, 7028847, 7624512, 8259888,
8936928, 9657648, 10424128, 11238513, 12103014, 13019909 },
{ 0, 0, 0, 0, 0, 0,
1, 7, 28, 84, 210, 462,
924, 1716, 3003, 5005, 8008, 12376,
18564, 27132, 38760, 54264, 74613, 100947,
134596, 177100, 230230, 296010, 376740, 475020,
593775, 736281, 906192, 1107568, 1344904, 1623160,
1947792, 2324784, 2760681, 3262623, 3838380, 4496388,
5245786, 6096454, 7059052, 8145060, 9366819, 10737573,
12271512, 13983816, 15890700, 18009460, 20358520, 22957480,
25827165, 28989675, 32468436, 36288252, 40475358, 45057474,
50063860, 55525372, 61474519, 67945521, 74974368, 82598880,
90858768, 99795696, 109453344, 119877472, 131115985, 143218999 },
{ 0, 0, 0, 0, 0, 0,
0, 1, 8, 36, 120, 330,
792, 1716, 3432, 6435, 11440, 19448,
31824, 50388, 77520, 116280, 170544, 245157,
346104, 480700, 657800, 888030, 1184040, 1560780,
2035800, 2629575, 3365856, 4272048, 5379616, 6724520,
8347680, 10295472, 12620256, 15380937, 18643560, 22481940,
26978328, 32224114, 38320568, 45379620, 53524680, 62891499,
73629072, 85900584, 99884400, 115775100, 133784560, 154143080,
177100560, 202927725, 231917400, 264385836, 300674088, 341149446,
386206920, 436270780, 491796152, 553270671, 621216192, 696190560,
778789440, 869648208, 969443904, 1078897248, 1198774720, 1329890705 },
};
static const int16_t dss_sp_filter_cb[14][32] = {
{ -32653, -32587, -32515, -32438, -32341, -32216, -32062, -31881,
-31665, -31398, -31080, -30724, -30299, -29813, -29248, -28572,
-27674, -26439, -24666, -22466, -19433, -16133, -12218, -7783,
-2834, 1819, 6544, 11260, 16050, 20220, 24774, 28120 },
{ -27503, -24509, -20644, -17496, -14187, -11277, -8420, -5595,
-3013, -624, 1711, 3880, 5844, 7774, 9739, 11592,
13364, 14903, 16426, 17900, 19250, 20586, 21803, 23006,
24142, 25249, 26275, 27300, 28359, 29249, 30118, 31183 },
{ -27827, -24208, -20943, -17781, -14843, -11848, -9066, -6297,
-3660, -910, 1918, 5025, 8223, 11649, 15086, 18423,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0 },
{ -17128, -11975, -8270, -5123, -2296, 183, 2503, 4707,
6798, 8945, 11045, 13239, 15528, 18248, 21115, 24785,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0 },
{ -21557, -17280, -14286, -11644, -9268, -7087, -4939, -2831,
-691, 1407, 3536, 5721, 8125, 10677, 13721, 17731,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0 },
{ -15030, -10377, -7034, -4327, -1900, 364, 2458, 4450,
6422, 8374, 10374, 12486, 14714, 16997, 19626, 22954,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0 },
{ -16155, -12362, -9698, -7460, -5258, -3359, -1547, 219,
1916, 3599, 5299, 6994, 8963, 11226, 13716, 16982,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0 },
{ -14742, -9848, -6921, -4648, -2769, -1065, 499, 2083,
3633, 5219, 6857, 8580, 10410, 12672, 15561, 20101,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0 },
{ -11099, -7014, -3855, -1025, 1680, 4544, 7807, 11932,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0 },
{ -9060, -4570, -1381, 1419, 4034, 6728, 9865, 14149,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0 },
{ -12450, -7985, -4596, -1734, 961, 3629, 6865, 11142,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0 },
{ -11831, -7404, -4010, -1096, 1606, 4291, 7386, 11482,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0 },
{ -13404, -9250, -5995, -3312, -890, 1594, 4464, 8198,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0 },
{ -11239, -7220, -4040, -1406, 971, 3321, 6006, 9697,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0 },
};
static const uint16_t dss_sp_fixed_cb_gain[64] = {
0, 4, 8, 13, 17, 22, 26, 31,
35, 40, 44, 48, 53, 58, 63, 69,
76, 83, 91, 99, 109, 119, 130, 142,
155, 170, 185, 203, 222, 242, 265, 290,
317, 346, 378, 414, 452, 494, 540, 591,
646, 706, 771, 843, 922, 1007, 1101, 1204,
1316, 1438, 1572, 1719, 1879, 2053, 2244, 2453,
2682, 2931, 3204, 3502, 3828, 4184, 4574, 5000,
};
static const int16_t dss_sp_pulse_val[8] = {
-31182, -22273, -13364, -4455, 4455, 13364, 22273, 31182
};
static const uint16_t binary_decreasing_array[] = {
32767, 16384, 8192, 4096, 2048, 1024, 512, 256,
128, 64, 32, 16, 8, 4, 2,
};
static const uint16_t dss_sp_unc_decreasing_array[] = {
32767, 26214, 20972, 16777, 13422, 10737, 8590, 6872,
5498, 4398, 3518, 2815, 2252, 1801, 1441,
};
static const uint16_t dss_sp_adaptive_gain[] = {
102, 231, 360, 488, 617, 746, 875, 1004,
1133, 1261, 1390, 1519, 1648, 1777, 1905, 2034,
2163, 2292, 2421, 2550, 2678, 2807, 2936, 3065,
3194, 3323, 3451, 3580, 3709, 3838, 3967, 4096,
};
static const int32_t dss_sp_sinc[67] = {
262, 293, 323, 348, 356, 336, 269, 139,
-67, -358, -733, -1178, -1668, -2162, -2607, -2940,
-3090, -2986, -2562, -1760, -541, 1110, 3187, 5651,
8435, 11446, 14568, 17670, 20611, 23251, 25460, 27125,
28160, 28512, 28160,
27125, 25460, 23251, 20611, 17670, 14568, 11446, 8435,
5651, 3187, 1110, -541, -1760, -2562, -2986, -3090,
-2940, -2607, -2162, -1668, -1178, -733, -358, -67,
139, 269, 336, 356, 348, 323, 293, 262,
};
static av_cold int dss_sp_decode_init(AVCodecContext *avctx)
{
DssSpContext *p = avctx->priv_data;
avctx->channel_layout = AV_CH_LAYOUT_MONO;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->channels = 1;
avctx->sample_rate = 11025;
memset(p->history, 0, sizeof(p->history));
p->pulse_dec_mode = 1;
p->avctx = avctx;
return 0;
}
static void dss_sp_unpack_coeffs(DssSpContext *p, const uint8_t *src)
{
GetBitContext gb;
DssSpFrame *fparam = &p->fparam;
int i;
int subframe_idx;
uint32_t combined_pitch;
uint32_t tmp;
uint32_t pitch_lag;
for (i = 0; i < DSS_SP_FRAME_SIZE; i += 2) {
p->bits[i] = src[i + 1];
p->bits[i + 1] = src[i];
}
init_get_bits(&gb, p->bits, DSS_SP_FRAME_SIZE * 8);
for (i = 0; i < 2; i++)
fparam->filter_idx[i] = get_bits(&gb, 5);
for (; i < 8; i++)
fparam->filter_idx[i] = get_bits(&gb, 4);
for (; i < 14; i++)
fparam->filter_idx[i] = get_bits(&gb, 3);
for (subframe_idx = 0; subframe_idx < 4; subframe_idx++) {
fparam->sf_adaptive_gain[subframe_idx] = get_bits(&gb, 5);
fparam->sf[subframe_idx].combined_pulse_pos = get_bits_long(&gb, 31);
fparam->sf[subframe_idx].gain = get_bits(&gb, 6);
for (i = 0; i < 7; i++)
fparam->sf[subframe_idx].pulse_val[i] = get_bits(&gb, 3);
}
for (subframe_idx = 0; subframe_idx < 4; subframe_idx++) {
unsigned int C72_binomials[PULSE_MAX] = {
72, 2556, 59640, 1028790, 13991544, 156238908, 1473109704,
3379081753
};
unsigned int combined_pulse_pos =
fparam->sf[subframe_idx].combined_pulse_pos;
int index = 6;
if (combined_pulse_pos < C72_binomials[PULSE_MAX - 1]) {
if (p->pulse_dec_mode) {
int pulse, pulse_idx;
pulse = PULSE_MAX - 1;
pulse_idx = 71;
combined_pulse_pos =
fparam->sf[subframe_idx].combined_pulse_pos;
/* this part seems to be close to g723.1 gen_fcb_excitation()
* RATE_6300 */
/* TODO: what is 7? size of subframe? */
for (i = 0; i < 7; i++) {
for (;
combined_pulse_pos <
dss_sp_combinatorial_table[pulse][pulse_idx];
--pulse_idx)
;
combined_pulse_pos -=
dss_sp_combinatorial_table[pulse][pulse_idx];
pulse--;
fparam->sf[subframe_idx].pulse_pos[i] = pulse_idx;
}
}
} else {
p->pulse_dec_mode = 0;
/* why do we need this? */
fparam->sf[subframe_idx].pulse_pos[6] = 0;
for (i = 71; i >= 0; i--) {
if (C72_binomials[index] <= combined_pulse_pos) {
combined_pulse_pos -= C72_binomials[index];
fparam->sf[subframe_idx].pulse_pos[6 - index] = i;
if (!index)
break;
--index;
}
--C72_binomials[0];
if (index) {
int a;
for (a = 0; a < index; a++)
C72_binomials[a + 1] -= C72_binomials[a];
}
}
}
}
combined_pitch = get_bits(&gb, 24);
fparam->pitch_lag[0] = (combined_pitch % 151) + 36;
combined_pitch /= 151;
for (i = 1; i < SUBFRAMES - 1; i++) {
fparam->pitch_lag[i] = combined_pitch % 48;
combined_pitch /= 48;
}
if (combined_pitch > 47) {
av_log (p->avctx, AV_LOG_WARNING, "combined_pitch was too large\n");
combined_pitch = 0;
}
fparam->pitch_lag[i] = combined_pitch;
pitch_lag = fparam->pitch_lag[0];
for (i = 1; i < SUBFRAMES; i++) {
if (pitch_lag > 162) {
fparam->pitch_lag[i] += 162 - 23;
} else {
tmp = pitch_lag - 23;
if (tmp < 36)
tmp = 36;
fparam->pitch_lag[i] += tmp;
}
pitch_lag = fparam->pitch_lag[i];
}
}
static void dss_sp_unpack_filter(DssSpContext *p)
{
int i;
for (i = 0; i < 14; i++)
p->lpc_filter[i] = dss_sp_filter_cb[i][p->fparam.filter_idx[i]];
}
static void dss_sp_convert_coeffs(int32_t *lpc_filter, int32_t *coeffs)
{
int a, a_plus, i;
coeffs[0] = 0x2000;
for (a = 0; a < 14; a++) {
a_plus = a + 1;
coeffs[a_plus] = lpc_filter[a] >> 2;
if (a_plus / 2 >= 1) {
for (i = 1; i <= a_plus / 2; i++) {
int coeff_1, coeff_2, tmp;
coeff_1 = coeffs[i];
coeff_2 = coeffs[a_plus - i];
tmp = DSS_SP_FORMULA(coeff_1, lpc_filter[a], coeff_2);
coeffs[i] = av_clip_int16(tmp);
tmp = DSS_SP_FORMULA(coeff_2, lpc_filter[a], coeff_1);
coeffs[a_plus - i] = av_clip_int16(tmp);
}
}
}
}
static void dss_sp_add_pulses(int32_t *vector_buf,
const struct DssSpSubframe *sf)
{
int i;
for (i = 0; i < 7; i++)
vector_buf[sf->pulse_pos[i]] += (dss_sp_fixed_cb_gain[sf->gain] *
dss_sp_pulse_val[sf->pulse_val[i]] +
0x4000) >> 15;
}
static void dss_sp_gen_exc(int32_t *vector, int32_t *prev_exc,
int pitch_lag, int gain)
{
int i;
/* do we actually need this check? we can use just [a3 - i % a3]
* for both cases */
if (pitch_lag < 72)
for (i = 0; i < 72; i++)
vector[i] = prev_exc[pitch_lag - i % pitch_lag];
else
for (i = 0; i < 72; i++)
vector[i] = prev_exc[pitch_lag - i];
for (i = 0; i < 72; i++) {
int tmp = gain * vector[i] >> 11;
vector[i] = av_clip_int16(tmp);
}
}
static void dss_sp_scale_vector(int32_t *vec, int bits, int size)
{
int i;
if (bits < 0)
for (i = 0; i < size; i++)
vec[i] = vec[i] >> -bits;
else
for (i = 0; i < size; i++)
vec[i] = vec[i] * (1 << bits);
}
static void dss_sp_update_buf(int32_t *hist, int32_t *vector)
{
int i;
for (i = 114; i > 0; i--)
vector[i + 72] = vector[i];
for (i = 0; i < 72; i++)
vector[72 - i] = hist[i];
}
static void dss_sp_shift_sq_sub(const int32_t *filter_buf,
int32_t *error_buf, int32_t *dst)
{
int a;
for (a = 0; a < 72; a++) {
int i, tmp;
tmp = dst[a] * filter_buf[0];
for (i = 14; i > 0; i--)
tmp -= error_buf[i] * (unsigned)filter_buf[i];
for (i = 14; i > 0; i--)
error_buf[i] = error_buf[i - 1];
tmp = (int)(tmp + 4096U) >> 13;
error_buf[1] = tmp;
dst[a] = av_clip_int16(tmp);
}
}
static void dss_sp_shift_sq_add(const int32_t *filter_buf, int32_t *audio_buf,
int32_t *dst)
{
int a;
for (a = 0; a < 72; a++) {
int i, tmp = 0;
audio_buf[0] = dst[a];
for (i = 14; i >= 0; i--)
tmp += audio_buf[i] * filter_buf[i];
for (i = 14; i > 0; i--)
audio_buf[i] = audio_buf[i - 1];
tmp = (tmp + 4096) >> 13;
dst[a] = av_clip_int16(tmp);
}
}
static void dss_sp_vec_mult(const int32_t *src, int32_t *dst,
const int16_t *mult)
{
int i;
dst[0] = src[0];
for (i = 1; i < 15; i++)
dst[i] = (src[i] * mult[i] + 0x4000) >> 15;
}
static int dss_sp_get_normalize_bits(int32_t *vector_buf, int16_t size)
{
unsigned int val;
int max_val;
int i;
val = 1;
for (i = 0; i < size; i++)
val |= FFABS(vector_buf[i]);
for (max_val = 0; val <= 0x4000; ++max_val)
val *= 2;
return max_val;
}
static int dss_sp_vector_sum(DssSpContext *p, int size)
{
int i, sum = 0;
for (i = 0; i < size; i++)
sum += FFABS(p->vector_buf[i]);
return sum;
}
static void dss_sp_sf_synthesis(DssSpContext *p, int32_t lpc_filter,
int32_t *dst, int size)
{
int32_t tmp_buf[15];
int32_t noise[72];
int bias, vsum_2 = 0, vsum_1 = 0, v36, normalize_bits;
int i, tmp;
if (size > 0) {
vsum_1 = dss_sp_vector_sum(p, size);
if (vsum_1 > 0xFFFFF)
vsum_1 = 0xFFFFF;
}
normalize_bits = dss_sp_get_normalize_bits(p->vector_buf, size);
dss_sp_scale_vector(p->vector_buf, normalize_bits - 3, size);
dss_sp_scale_vector(p->audio_buf, normalize_bits, 15);
dss_sp_scale_vector(p->err_buf1, normalize_bits, 15);
v36 = p->err_buf1[1];
dss_sp_vec_mult(p->filter, tmp_buf, binary_decreasing_array);
dss_sp_shift_sq_add(tmp_buf, p->audio_buf, p->vector_buf);
dss_sp_vec_mult(p->filter, tmp_buf, dss_sp_unc_decreasing_array);
dss_sp_shift_sq_sub(tmp_buf, p->err_buf1, p->vector_buf);
/* lpc_filter can be negative */
lpc_filter = lpc_filter >> 1;
if (lpc_filter >= 0)
lpc_filter = 0;
if (size > 1) {
for (i = size - 1; i > 0; i--) {
tmp = DSS_SP_FORMULA(p->vector_buf[i], lpc_filter,
p->vector_buf[i - 1]);
p->vector_buf[i] = av_clip_int16(tmp);
}
}
tmp = DSS_SP_FORMULA(p->vector_buf[0], lpc_filter, v36);
p->vector_buf[0] = av_clip_int16(tmp);
dss_sp_scale_vector(p->vector_buf, -normalize_bits, size);
dss_sp_scale_vector(p->audio_buf, -normalize_bits, 15);
dss_sp_scale_vector(p->err_buf1, -normalize_bits, 15);
if (size > 0)
vsum_2 = dss_sp_vector_sum(p, size);
if (vsum_2 >= 0x40)
tmp = (vsum_1 << 11) / vsum_2;
else
tmp = 1;
bias = 409 * tmp >> 15 << 15;
tmp = (bias + 32358 * p->noise_state) >> 15;
noise[0] = av_clip_int16(tmp);
for (i = 1; i < size; i++) {
tmp = (bias + 32358 * noise[i - 1]) >> 15;
noise[i] = av_clip_int16(tmp);
}
p->noise_state = noise[size - 1];
for (i = 0; i < size; i++) {
tmp = (p->vector_buf[i] * noise[i]) >> 11;
dst[i] = av_clip_int16(tmp);
}
}
static void dss_sp_update_state(DssSpContext *p, int32_t *dst)
{
int i, offset = 6, counter = 0, a = 0;
for (i = 0; i < 6; i++)
p->excitation[i] = p->excitation[288 + i];
for (i = 0; i < 72 * SUBFRAMES; i++)
p->excitation[6 + i] = dst[i];
do {
int tmp = 0;
for (i = 0; i < 6; i++)
tmp += p->excitation[offset--] * dss_sp_sinc[a + i * 11];
offset += 7;
tmp >>= 15;
dst[counter] = av_clip_int16(tmp);
counter++;
a = (a + 1) % 11;
if (!a)
offset++;
} while (offset < FF_ARRAY_ELEMS(p->excitation));
}
static void dss_sp_32to16bit(int16_t *dst, int32_t *src, int size)
{
int i;
for (i = 0; i < size; i++)
dst[i] = av_clip_int16(src[i]);
}
static int dss_sp_decode_one_frame(DssSpContext *p,
int16_t *abuf_dst, const uint8_t *abuf_src)
{
int i, j;
dss_sp_unpack_coeffs(p, abuf_src);
dss_sp_unpack_filter(p);
dss_sp_convert_coeffs(p->lpc_filter, p->filter);
for (j = 0; j < SUBFRAMES; j++) {
dss_sp_gen_exc(p->vector_buf, p->history,
p->fparam.pitch_lag[j],
dss_sp_adaptive_gain[p->fparam.sf_adaptive_gain[j]]);
dss_sp_add_pulses(p->vector_buf, &p->fparam.sf[j]);
dss_sp_update_buf(p->vector_buf, p->history);
for (i = 0; i < 72; i++)
p->vector_buf[i] = p->history[72 - i];
dss_sp_shift_sq_sub(p->filter,
p->err_buf2, p->vector_buf);
dss_sp_sf_synthesis(p, p->lpc_filter[0],
&p->working_buffer[j][0], 72);
}
dss_sp_update_state(p, &p->working_buffer[0][0]);
dss_sp_32to16bit(abuf_dst,
&p->working_buffer[0][0], 264);
return 0;
}
static int dss_sp_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
DssSpContext *p = avctx->priv_data;
AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int16_t *out;
int ret;
if (buf_size < DSS_SP_FRAME_SIZE) {
if (buf_size)
av_log(avctx, AV_LOG_WARNING,
"Expected %d bytes, got %d - skipping packet.\n",
DSS_SP_FRAME_SIZE, buf_size);
*got_frame_ptr = 0;
return AVERROR_INVALIDDATA;
}
frame->nb_samples = DSS_SP_SAMPLE_COUNT;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
out = (int16_t *)frame->data[0];
dss_sp_decode_one_frame(p, out, buf);
*got_frame_ptr = 1;
return DSS_SP_FRAME_SIZE;
}
const AVCodec ff_dss_sp_decoder = {
.name = "dss_sp",
.long_name = NULL_IF_CONFIG_SMALL("Digital Speech Standard - Standard Play mode (DSS SP)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_DSS_SP,
.priv_data_size = sizeof(DssSpContext),
.init = dss_sp_decode_init,
.decode = dss_sp_decode_frame,
.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
};
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