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