/* * Digital Speech Standard - Standard Play mode (DSS SP) audio decoder. * Copyright (C) 2014 Oleksij Rempel * * 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, };