mirror of
https://github.com/xenia-project/FFmpeg.git
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503 lines
15 KiB
C
503 lines
15 KiB
C
/*
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* G.726 ADPCM audio codec
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* Copyright (c) 2004 Roman Shaposhnik
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*
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* This is a very straightforward rendition of the G.726
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* Section 4 "Computational Details".
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*
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* This file is part of Libav.
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*
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* Libav 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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* Libav 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 Libav; 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 <limits.h>
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#include "libavutil/avassert.h"
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "internal.h"
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#include "get_bits.h"
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#include "put_bits.h"
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/**
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* G.726 11bit float.
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* G.726 Standard uses rather odd 11bit floating point arithmentic for
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* numerous occasions. It's a mistery to me why they did it this way
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* instead of simply using 32bit integer arithmetic.
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*/
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typedef struct Float11 {
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uint8_t sign; /**< 1bit sign */
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uint8_t exp; /**< 4bit exponent */
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uint8_t mant; /**< 6bit mantissa */
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} Float11;
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static inline Float11* i2f(int i, Float11* f)
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{
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f->sign = (i < 0);
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if (f->sign)
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i = -i;
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f->exp = av_log2_16bit(i) + !!i;
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f->mant = i? (i<<6) >> f->exp : 1<<5;
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return f;
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}
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static inline int16_t mult(Float11* f1, Float11* f2)
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{
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int res, exp;
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exp = f1->exp + f2->exp;
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res = (((f1->mant * f2->mant) + 0x30) >> 4);
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res = exp > 19 ? res << (exp - 19) : res >> (19 - exp);
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return (f1->sign ^ f2->sign) ? -res : res;
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}
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static inline int sgn(int value)
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{
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return (value < 0) ? -1 : 1;
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}
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typedef struct G726Tables {
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const int* quant; /**< quantization table */
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const int16_t* iquant; /**< inverse quantization table */
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const int16_t* W; /**< special table #1 ;-) */
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const uint8_t* F; /**< special table #2 */
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} G726Tables;
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typedef struct G726Context {
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AVClass *class;
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AVFrame frame;
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G726Tables tbls; /**< static tables needed for computation */
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Float11 sr[2]; /**< prev. reconstructed samples */
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Float11 dq[6]; /**< prev. difference */
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int a[2]; /**< second order predictor coeffs */
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int b[6]; /**< sixth order predictor coeffs */
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int pk[2]; /**< signs of prev. 2 sez + dq */
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int ap; /**< scale factor control */
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int yu; /**< fast scale factor */
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int yl; /**< slow scale factor */
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int dms; /**< short average magnitude of F[i] */
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int dml; /**< long average magnitude of F[i] */
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int td; /**< tone detect */
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int se; /**< estimated signal for the next iteration */
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int sez; /**< estimated second order prediction */
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int y; /**< quantizer scaling factor for the next iteration */
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int code_size;
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} G726Context;
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static const int quant_tbl16[] = /**< 16kbit/s 2bits per sample */
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{ 260, INT_MAX };
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static const int16_t iquant_tbl16[] =
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{ 116, 365, 365, 116 };
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static const int16_t W_tbl16[] =
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{ -22, 439, 439, -22 };
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static const uint8_t F_tbl16[] =
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{ 0, 7, 7, 0 };
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static const int quant_tbl24[] = /**< 24kbit/s 3bits per sample */
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{ 7, 217, 330, INT_MAX };
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static const int16_t iquant_tbl24[] =
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{ INT16_MIN, 135, 273, 373, 373, 273, 135, INT16_MIN };
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static const int16_t W_tbl24[] =
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{ -4, 30, 137, 582, 582, 137, 30, -4 };
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static const uint8_t F_tbl24[] =
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{ 0, 1, 2, 7, 7, 2, 1, 0 };
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static const int quant_tbl32[] = /**< 32kbit/s 4bits per sample */
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{ -125, 79, 177, 245, 299, 348, 399, INT_MAX };
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static const int16_t iquant_tbl32[] =
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{ INT16_MIN, 4, 135, 213, 273, 323, 373, 425,
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425, 373, 323, 273, 213, 135, 4, INT16_MIN };
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static const int16_t W_tbl32[] =
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{ -12, 18, 41, 64, 112, 198, 355, 1122,
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1122, 355, 198, 112, 64, 41, 18, -12};
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static const uint8_t F_tbl32[] =
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{ 0, 0, 0, 1, 1, 1, 3, 7, 7, 3, 1, 1, 1, 0, 0, 0 };
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static const int quant_tbl40[] = /**< 40kbit/s 5bits per sample */
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{ -122, -16, 67, 138, 197, 249, 297, 338,
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377, 412, 444, 474, 501, 527, 552, INT_MAX };
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static const int16_t iquant_tbl40[] =
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{ INT16_MIN, -66, 28, 104, 169, 224, 274, 318,
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358, 395, 429, 459, 488, 514, 539, 566,
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566, 539, 514, 488, 459, 429, 395, 358,
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318, 274, 224, 169, 104, 28, -66, INT16_MIN };
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static const int16_t W_tbl40[] =
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{ 14, 14, 24, 39, 40, 41, 58, 100,
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141, 179, 219, 280, 358, 440, 529, 696,
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696, 529, 440, 358, 280, 219, 179, 141,
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100, 58, 41, 40, 39, 24, 14, 14 };
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static const uint8_t F_tbl40[] =
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{ 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 2, 3, 4, 5, 6, 6,
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6, 6, 5, 4, 3, 2, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0 };
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static const G726Tables G726Tables_pool[] =
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{{ quant_tbl16, iquant_tbl16, W_tbl16, F_tbl16 },
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{ quant_tbl24, iquant_tbl24, W_tbl24, F_tbl24 },
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{ quant_tbl32, iquant_tbl32, W_tbl32, F_tbl32 },
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{ quant_tbl40, iquant_tbl40, W_tbl40, F_tbl40 }};
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/**
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* Para 4.2.2 page 18: Adaptive quantizer.
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*/
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static inline uint8_t quant(G726Context* c, int d)
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{
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int sign, exp, i, dln;
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sign = i = 0;
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if (d < 0) {
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sign = 1;
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d = -d;
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}
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exp = av_log2_16bit(d);
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dln = ((exp<<7) + (((d<<7)>>exp)&0x7f)) - (c->y>>2);
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while (c->tbls.quant[i] < INT_MAX && c->tbls.quant[i] < dln)
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++i;
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if (sign)
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i = ~i;
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if (c->code_size != 2 && i == 0) /* I'm not sure this is a good idea */
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i = 0xff;
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return i;
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}
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/**
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* Para 4.2.3 page 22: Inverse adaptive quantizer.
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*/
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static inline int16_t inverse_quant(G726Context* c, int i)
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{
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int dql, dex, dqt;
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dql = c->tbls.iquant[i] + (c->y >> 2);
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dex = (dql>>7) & 0xf; /* 4bit exponent */
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dqt = (1<<7) + (dql & 0x7f); /* log2 -> linear */
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return (dql < 0) ? 0 : ((dqt<<dex) >> 7);
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}
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static int16_t g726_decode(G726Context* c, int I)
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{
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int dq, re_signal, pk0, fa1, i, tr, ylint, ylfrac, thr2, al, dq0;
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Float11 f;
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int I_sig= I >> (c->code_size - 1);
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dq = inverse_quant(c, I);
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/* Transition detect */
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ylint = (c->yl >> 15);
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ylfrac = (c->yl >> 10) & 0x1f;
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thr2 = (ylint > 9) ? 0x1f << 10 : (0x20 + ylfrac) << ylint;
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tr= (c->td == 1 && dq > ((3*thr2)>>2));
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if (I_sig) /* get the sign */
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dq = -dq;
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re_signal = c->se + dq;
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/* Update second order predictor coefficient A2 and A1 */
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pk0 = (c->sez + dq) ? sgn(c->sez + dq) : 0;
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dq0 = dq ? sgn(dq) : 0;
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if (tr) {
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c->a[0] = 0;
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c->a[1] = 0;
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for (i=0; i<6; i++)
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c->b[i] = 0;
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} else {
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/* This is a bit crazy, but it really is +255 not +256 */
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fa1 = av_clip((-c->a[0]*c->pk[0]*pk0)>>5, -256, 255);
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c->a[1] += 128*pk0*c->pk[1] + fa1 - (c->a[1]>>7);
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c->a[1] = av_clip(c->a[1], -12288, 12288);
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c->a[0] += 64*3*pk0*c->pk[0] - (c->a[0] >> 8);
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c->a[0] = av_clip(c->a[0], -(15360 - c->a[1]), 15360 - c->a[1]);
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for (i=0; i<6; i++)
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c->b[i] += 128*dq0*sgn(-c->dq[i].sign) - (c->b[i]>>8);
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}
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/* Update Dq and Sr and Pk */
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c->pk[1] = c->pk[0];
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c->pk[0] = pk0 ? pk0 : 1;
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c->sr[1] = c->sr[0];
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i2f(re_signal, &c->sr[0]);
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for (i=5; i>0; i--)
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c->dq[i] = c->dq[i-1];
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i2f(dq, &c->dq[0]);
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c->dq[0].sign = I_sig; /* Isn't it crazy ?!?! */
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c->td = c->a[1] < -11776;
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/* Update Ap */
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c->dms += (c->tbls.F[I]<<4) + ((- c->dms) >> 5);
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c->dml += (c->tbls.F[I]<<4) + ((- c->dml) >> 7);
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if (tr)
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c->ap = 256;
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else {
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c->ap += (-c->ap) >> 4;
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if (c->y <= 1535 || c->td || abs((c->dms << 2) - c->dml) >= (c->dml >> 3))
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c->ap += 0x20;
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}
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/* Update Yu and Yl */
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c->yu = av_clip(c->y + c->tbls.W[I] + ((-c->y)>>5), 544, 5120);
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c->yl += c->yu + ((-c->yl)>>6);
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/* Next iteration for Y */
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al = (c->ap >= 256) ? 1<<6 : c->ap >> 2;
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c->y = (c->yl + (c->yu - (c->yl>>6))*al) >> 6;
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/* Next iteration for SE and SEZ */
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c->se = 0;
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for (i=0; i<6; i++)
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c->se += mult(i2f(c->b[i] >> 2, &f), &c->dq[i]);
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c->sez = c->se >> 1;
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for (i=0; i<2; i++)
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c->se += mult(i2f(c->a[i] >> 2, &f), &c->sr[i]);
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c->se >>= 1;
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return av_clip(re_signal << 2, -0xffff, 0xffff);
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}
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static av_cold int g726_reset(G726Context *c)
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{
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int i;
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c->tbls = G726Tables_pool[c->code_size - 2];
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for (i=0; i<2; i++) {
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c->sr[i].mant = 1<<5;
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c->pk[i] = 1;
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}
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for (i=0; i<6; i++) {
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c->dq[i].mant = 1<<5;
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}
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c->yu = 544;
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c->yl = 34816;
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c->y = 544;
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return 0;
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}
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#if CONFIG_ADPCM_G726_ENCODER
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static int16_t g726_encode(G726Context* c, int16_t sig)
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{
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uint8_t i;
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i = quant(c, sig/4 - c->se) & ((1<<c->code_size) - 1);
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g726_decode(c, i);
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return i;
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}
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/* Interfacing to the libavcodec */
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static av_cold int g726_encode_init(AVCodecContext *avctx)
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{
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G726Context* c = avctx->priv_data;
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if (avctx->strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL &&
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avctx->sample_rate != 8000) {
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av_log(avctx, AV_LOG_ERROR, "Sample rates other than 8kHz are not "
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"allowed when the compliance level is higher than unofficial. "
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"Resample or reduce the compliance level.\n");
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return AVERROR(EINVAL);
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}
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av_assert0(avctx->sample_rate > 0);
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if(avctx->channels != 1){
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av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
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return AVERROR(EINVAL);
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}
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if (avctx->bit_rate)
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c->code_size = (avctx->bit_rate + avctx->sample_rate/2) / avctx->sample_rate;
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c->code_size = av_clip(c->code_size, 2, 5);
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avctx->bit_rate = c->code_size * avctx->sample_rate;
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avctx->bits_per_coded_sample = c->code_size;
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g726_reset(c);
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#if FF_API_OLD_ENCODE_AUDIO
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avctx->coded_frame = avcodec_alloc_frame();
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if (!avctx->coded_frame)
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return AVERROR(ENOMEM);
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avctx->coded_frame->key_frame = 1;
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#endif
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/* select a frame size that will end on a byte boundary and have a size of
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approximately 1024 bytes */
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avctx->frame_size = ((int[]){ 4096, 2736, 2048, 1640 })[c->code_size - 2];
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return 0;
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}
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#if FF_API_OLD_ENCODE_AUDIO
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static av_cold int g726_encode_close(AVCodecContext *avctx)
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{
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av_freep(&avctx->coded_frame);
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return 0;
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}
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#endif
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static int g726_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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const AVFrame *frame, int *got_packet_ptr)
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{
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G726Context *c = avctx->priv_data;
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const int16_t *samples = (const int16_t *)frame->data[0];
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PutBitContext pb;
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int i, ret, out_size;
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out_size = (frame->nb_samples * c->code_size + 7) / 8;
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if ((ret = ff_alloc_packet(avpkt, out_size))) {
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av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
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return ret;
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}
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init_put_bits(&pb, avpkt->data, avpkt->size);
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for (i = 0; i < frame->nb_samples; i++)
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put_bits(&pb, c->code_size, g726_encode(c, *samples++));
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flush_put_bits(&pb);
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avpkt->size = out_size;
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*got_packet_ptr = 1;
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return 0;
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}
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#define OFFSET(x) offsetof(G726Context, x)
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#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
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static const AVOption options[] = {
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{ "code_size", "Bits per code", OFFSET(code_size), AV_OPT_TYPE_INT, { .i64 = 4 }, 2, 5, AE },
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{ NULL },
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};
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static const AVClass class = {
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.class_name = "g726",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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static const AVCodecDefault defaults[] = {
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{ "b", "0" },
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{ NULL },
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};
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AVCodec ff_adpcm_g726_encoder = {
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.name = "g726",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_ADPCM_G726,
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.priv_data_size = sizeof(G726Context),
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.init = g726_encode_init,
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.encode2 = g726_encode_frame,
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#if FF_API_OLD_ENCODE_AUDIO
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.close = g726_encode_close,
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#endif
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.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
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.priv_class = &class,
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.defaults = defaults,
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};
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#endif
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#if CONFIG_ADPCM_G726_DECODER
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static av_cold int g726_decode_init(AVCodecContext *avctx)
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{
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G726Context* c = avctx->priv_data;
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if (avctx->strict_std_compliance >= FF_COMPLIANCE_STRICT &&
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avctx->sample_rate != 8000) {
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av_log(avctx, AV_LOG_ERROR, "Only 8kHz sample rate is allowed when "
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"the compliance level is strict. Reduce the compliance level "
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"if you wish to decode the stream anyway.\n");
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return AVERROR(EINVAL);
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}
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if(avctx->channels != 1){
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av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
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return AVERROR(EINVAL);
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}
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c->code_size = avctx->bits_per_coded_sample;
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if (c->code_size < 2 || c->code_size > 5) {
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av_log(avctx, AV_LOG_ERROR, "Invalid number of bits %d\n", c->code_size);
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return AVERROR(EINVAL);
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}
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g726_reset(c);
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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avcodec_get_frame_defaults(&c->frame);
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avctx->coded_frame = &c->frame;
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return 0;
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}
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static int g726_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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G726Context *c = avctx->priv_data;
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int16_t *samples;
|
|
GetBitContext gb;
|
|
int out_samples, ret;
|
|
|
|
out_samples = buf_size * 8 / c->code_size;
|
|
|
|
/* get output buffer */
|
|
c->frame.nb_samples = out_samples;
|
|
if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) {
|
|
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
|
|
return ret;
|
|
}
|
|
samples = (int16_t *)c->frame.data[0];
|
|
|
|
init_get_bits(&gb, buf, buf_size * 8);
|
|
|
|
while (out_samples--)
|
|
*samples++ = g726_decode(c, get_bits(&gb, c->code_size));
|
|
|
|
if (get_bits_left(&gb) > 0)
|
|
av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n");
|
|
|
|
*got_frame_ptr = 1;
|
|
*(AVFrame *)data = c->frame;
|
|
|
|
return buf_size;
|
|
}
|
|
|
|
static void g726_decode_flush(AVCodecContext *avctx)
|
|
{
|
|
G726Context *c = avctx->priv_data;
|
|
g726_reset(c);
|
|
}
|
|
|
|
AVCodec ff_adpcm_g726_decoder = {
|
|
.name = "g726",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_ADPCM_G726,
|
|
.priv_data_size = sizeof(G726Context),
|
|
.init = g726_decode_init,
|
|
.decode = g726_decode_frame,
|
|
.flush = g726_decode_flush,
|
|
.capabilities = CODEC_CAP_DR1,
|
|
.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
|
|
};
|
|
#endif
|