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FFmpeg/libavcodec/dss_sp.c

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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/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) (((((a) << 15) + (b) * (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 +
FF_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] << 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] * filter_buf[i];
for (i = 14; i > 0; i--)
error_buf[i] = error_buf[i - 1];
tmp = (tmp + 4096) >> 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) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed.\n");
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;
}
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 = CODEC_CAP_DR1,
};