From c56b9b1eb278c5ef89d3f0832a56dfe4732cb68b Mon Sep 17 00:00:00 2001 From: Oleksij Rempel Date: Fri, 13 Feb 2015 08:36:16 +0100 Subject: lavc: Add DSS SP decoder Signed-off-by: Oleksij Rempel Signed-off-by: Luca Barbato Signed-off-by: Vittorio Giovara --- libavcodec/dss_sp.c | 780 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 780 insertions(+) create mode 100644 libavcodec/dss_sp.c (limited to 'libavcodec/dss_sp.c') diff --git a/libavcodec/dss_sp.c b/libavcodec/dss_sp.c new file mode 100644 index 0000000000..6fadcc685d --- /dev/null +++ b/libavcodec/dss_sp.c @@ -0,0 +1,780 @@ +/* + * Digital Speech Standard - Standard Play mode (DSS SP) audio decoder. + * Copyright (C) 2014 Oleksij Rempel + * + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; 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 { + 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; + + 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[(index ^ 7) - 1] = 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; i++) { + fparam->pitch_lag[i] = combined_pitch % 48; + combined_pitch /= 48; + } + + 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, +}; -- cgit v1.2.3