Use FFABS instead of abs.
[FFMpeg-mirror/ordered_chapters.git] / libavcodec / g726.c
blob01d1cec6f516359f7c24cee553283dded509782a
1 /*
2 * G.726 ADPCM audio codec
3 * Copyright (c) 2004 Roman Shaposhnik.
5 * This is a very straightforward rendition of the G.726
6 * Section 4 "Computational Details".
8 * This file is part of FFmpeg.
10 * FFmpeg is free software; you can redistribute it and/or
11 * modify it under the terms of the GNU Lesser General Public
12 * License as published by the Free Software Foundation; either
13 * version 2.1 of the License, or (at your option) any later version.
15 * FFmpeg is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18 * Lesser General Public License for more details.
20 * You should have received a copy of the GNU Lesser General Public
21 * License along with FFmpeg; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24 #include <limits.h>
25 #include "avcodec.h"
26 #include "bitstream.h"
28 /**
29 * G.726 11bit float.
30 * G.726 Standard uses rather odd 11bit floating point arithmentic for
31 * numerous occasions. It's a mistery to me why they did it this way
32 * instead of simply using 32bit integer arithmetic.
34 typedef struct Float11 {
35 int sign; /**< 1bit sign */
36 int exp; /**< 4bit exponent */
37 int mant; /**< 6bit mantissa */
38 } Float11;
40 static inline Float11* i2f(int16_t i, Float11* f)
42 f->sign = (i < 0);
43 if (f->sign)
44 i = -i;
45 f->exp = av_log2_16bit(i) + !!i;
46 f->mant = i? (i<<6) >> f->exp : 1<<5;
47 return f;
50 static inline int16_t mult(Float11* f1, Float11* f2)
52 int res, exp;
54 exp = f1->exp + f2->exp;
55 res = (((f1->mant * f2->mant) + 0x30) >> 4) << 7;
56 res = exp > 26 ? res << (exp - 26) : res >> (26 - exp);
57 return (f1->sign ^ f2->sign) ? -res : res;
60 static inline int sgn(int value)
62 return (value < 0) ? -1 : 1;
65 typedef struct G726Tables {
66 int bits; /**< bits per sample */
67 int* quant; /**< quantization table */
68 int* iquant; /**< inverse quantization table */
69 int* W; /**< special table #1 ;-) */
70 int* F; /**< special table #2 */
71 } G726Tables;
73 typedef struct G726Context {
74 G726Tables* tbls; /**< static tables needed for computation */
76 Float11 sr[2]; /**< prev. reconstructed samples */
77 Float11 dq[6]; /**< prev. difference */
78 int a[2]; /**< second order predictor coeffs */
79 int b[6]; /**< sixth order predictor coeffs */
80 int pk[2]; /**< signs of prev. 2 sez + dq */
82 int ap; /**< scale factor control */
83 int yu; /**< fast scale factor */
84 int yl; /**< slow scale factor */
85 int dms; /**< short average magnitude of F[i] */
86 int dml; /**< long average magnitude of F[i] */
87 int td; /**< tone detect */
89 int se; /**< estimated signal for the next iteration */
90 int sez; /**< estimated second order prediction */
91 int y; /**< quantizer scaling factor for the next iteration */
92 } G726Context;
94 static int quant_tbl16[] = /**< 16kbit/s 2bits per sample */
95 { 260, INT_MAX };
96 static int iquant_tbl16[] =
97 { 116, 365, 365, 116 };
98 static int W_tbl16[] =
99 { -22, 439, 439, -22 };
100 static int F_tbl16[] =
101 { 0, 7, 7, 0 };
103 static int quant_tbl24[] = /**< 24kbit/s 3bits per sample */
104 { 7, 217, 330, INT_MAX };
105 static int iquant_tbl24[] =
106 { INT_MIN, 135, 273, 373, 373, 273, 135, INT_MIN };
107 static int W_tbl24[] =
108 { -4, 30, 137, 582, 582, 137, 30, -4 };
109 static int F_tbl24[] =
110 { 0, 1, 2, 7, 7, 2, 1, 0 };
112 static int quant_tbl32[] = /**< 32kbit/s 4bits per sample */
113 { -125, 79, 177, 245, 299, 348, 399, INT_MAX };
114 static int iquant_tbl32[] =
115 { INT_MIN, 4, 135, 213, 273, 323, 373, 425,
116 425, 373, 323, 273, 213, 135, 4, INT_MIN };
117 static int W_tbl32[] =
118 { -12, 18, 41, 64, 112, 198, 355, 1122,
119 1122, 355, 198, 112, 64, 41, 18, -12};
120 static int F_tbl32[] =
121 { 0, 0, 0, 1, 1, 1, 3, 7, 7, 3, 1, 1, 1, 0, 0, 0 };
123 static int quant_tbl40[] = /**< 40kbit/s 5bits per sample */
124 { -122, -16, 67, 138, 197, 249, 297, 338,
125 377, 412, 444, 474, 501, 527, 552, INT_MAX };
126 static int iquant_tbl40[] =
127 { INT_MIN, -66, 28, 104, 169, 224, 274, 318,
128 358, 395, 429, 459, 488, 514, 539, 566,
129 566, 539, 514, 488, 459, 429, 395, 358,
130 318, 274, 224, 169, 104, 28, -66, INT_MIN };
131 static int W_tbl40[] =
132 { 14, 14, 24, 39, 40, 41, 58, 100,
133 141, 179, 219, 280, 358, 440, 529, 696,
134 696, 529, 440, 358, 280, 219, 179, 141,
135 100, 58, 41, 40, 39, 24, 14, 14 };
136 static int F_tbl40[] =
137 { 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 2, 3, 4, 5, 6, 6,
138 6, 6, 5, 4, 3, 2, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0 };
140 static G726Tables G726Tables_pool[] =
141 {{ 2, quant_tbl16, iquant_tbl16, W_tbl16, F_tbl16 },
142 { 3, quant_tbl24, iquant_tbl24, W_tbl24, F_tbl24 },
143 { 4, quant_tbl32, iquant_tbl32, W_tbl32, F_tbl32 },
144 { 5, quant_tbl40, iquant_tbl40, W_tbl40, F_tbl40 }};
148 * Para 4.2.2 page 18: Adaptive quantizer.
150 static inline uint8_t quant(G726Context* c, int d)
152 int sign, exp, i, dln;
154 sign = i = 0;
155 if (d < 0) {
156 sign = 1;
157 d = -d;
159 exp = av_log2_16bit(d);
160 dln = ((exp<<7) + (((d<<7)>>exp)&0x7f)) - (c->y>>2);
162 while (c->tbls->quant[i] < INT_MAX && c->tbls->quant[i] < dln)
163 ++i;
165 if (sign)
166 i = ~i;
167 if (c->tbls->bits != 2 && i == 0) /* I'm not sure this is a good idea */
168 i = 0xff;
170 return i;
174 * Para 4.2.3 page 22: Inverse adaptive quantizer.
176 static inline int16_t inverse_quant(G726Context* c, int i)
178 int dql, dex, dqt;
180 dql = c->tbls->iquant[i] + (c->y >> 2);
181 dex = (dql>>7) & 0xf; /* 4bit exponent */
182 dqt = (1<<7) + (dql & 0x7f); /* log2 -> linear */
183 return (dql < 0) ? 0 : ((dqt<<7) >> (14-dex));
186 static inline int16_t g726_iterate(G726Context* c, int16_t I)
188 int dq, re_signal, pk0, fa1, i, tr, ylint, ylfrac, thr2, al, dq0;
189 Float11 f;
191 dq = inverse_quant(c, I);
192 if (I >> (c->tbls->bits - 1)) /* get the sign */
193 dq = -dq;
194 re_signal = c->se + dq;
196 /* Transition detect */
197 ylint = (c->yl >> 15);
198 ylfrac = (c->yl >> 10) & 0x1f;
199 thr2 = (ylint > 9) ? 0x1f << 10 : (0x20 + ylfrac) << ylint;
200 if (c->td == 1 && abs(dq) > ((thr2+(thr2>>1))>>1))
201 tr = 1;
202 else
203 tr = 0;
205 /* Update second order predictor coefficient A2 and A1 */
206 pk0 = (c->sez + dq) ? sgn(c->sez + dq) : 0;
207 dq0 = dq ? sgn(dq) : 0;
208 if (tr) {
209 c->a[0] = 0;
210 c->a[1] = 0;
211 for (i=0; i<6; i++)
212 c->b[i] = 0;
213 } else {
214 /* This is a bit crazy, but it really is +255 not +256 */
215 fa1 = av_clip((-c->a[0]*c->pk[0]*pk0)>>5, -256, 255);
217 c->a[1] += 128*pk0*c->pk[1] + fa1 - (c->a[1]>>7);
218 c->a[1] = av_clip(c->a[1], -12288, 12288);
219 c->a[0] += 64*3*pk0*c->pk[0] - (c->a[0] >> 8);
220 c->a[0] = av_clip(c->a[0], -(15360 - c->a[1]), 15360 - c->a[1]);
222 for (i=0; i<6; i++)
223 c->b[i] += 128*dq0*sgn(-c->dq[i].sign) - (c->b[i]>>8);
226 /* Update Dq and Sr and Pk */
227 c->pk[1] = c->pk[0];
228 c->pk[0] = pk0 ? pk0 : 1;
229 c->sr[1] = c->sr[0];
230 i2f(re_signal, &c->sr[0]);
231 for (i=5; i>0; i--)
232 c->dq[i] = c->dq[i-1];
233 i2f(dq, &c->dq[0]);
234 c->dq[0].sign = I >> (c->tbls->bits - 1); /* Isn't it crazy ?!?! */
236 /* Update tone detect [I'm not sure 'tr == 0' is really needed] */
237 c->td = (tr == 0 && c->a[1] < -11776);
239 /* Update Ap */
240 c->dms += ((c->tbls->F[I]<<9) - c->dms) >> 5;
241 c->dml += ((c->tbls->F[I]<<11) - c->dml) >> 7;
242 if (tr)
243 c->ap = 256;
244 else if (c->y > 1535 && !c->td && (abs((c->dms << 2) - c->dml) < (c->dml >> 3)))
245 c->ap += (-c->ap) >> 4;
246 else
247 c->ap += (0x200 - c->ap) >> 4;
249 /* Update Yu and Yl */
250 c->yu = av_clip(c->y + (((c->tbls->W[I] << 5) - c->y) >> 5), 544, 5120);
251 c->yl += c->yu + ((-c->yl)>>6);
253 /* Next iteration for Y */
254 al = (c->ap >= 256) ? 1<<6 : c->ap >> 2;
255 c->y = (c->yl + (c->yu - (c->yl>>6))*al) >> 6;
257 /* Next iteration for SE and SEZ */
258 c->se = 0;
259 for (i=0; i<6; i++)
260 c->se += mult(i2f(c->b[i] >> 2, &f), &c->dq[i]);
261 c->sez = c->se >> 1;
262 for (i=0; i<2; i++)
263 c->se += mult(i2f(c->a[i] >> 2, &f), &c->sr[i]);
264 c->se >>= 1;
266 return av_clip(re_signal << 2, -0xffff, 0xffff);
269 static int g726_reset(G726Context* c, int bit_rate)
271 int i;
273 c->tbls = &G726Tables_pool[bit_rate/8000 - 2];
274 for (i=0; i<2; i++) {
275 i2f(0, &c->sr[i]);
276 c->a[i] = 0;
277 c->pk[i] = 1;
279 for (i=0; i<6; i++) {
280 i2f(0, &c->dq[i]);
281 c->b[i] = 0;
283 c->ap = 0;
284 c->dms = 0;
285 c->dml = 0;
286 c->yu = 544;
287 c->yl = 34816;
288 c->td = 0;
290 c->se = 0;
291 c->sez = 0;
292 c->y = 544;
294 return 0;
297 static int16_t g726_decode(G726Context* c, int16_t i)
299 return g726_iterate(c, i);
302 #ifdef CONFIG_ENCODERS
303 static int16_t g726_encode(G726Context* c, int16_t sig)
305 uint8_t i;
307 i = quant(c, sig/4 - c->se) & ((1<<c->tbls->bits) - 1);
308 g726_iterate(c, i);
309 return i;
311 #endif
313 /* Interfacing to the libavcodec */
315 typedef struct AVG726Context {
316 G726Context c;
317 int bits_left;
318 int bit_buffer;
319 int code_size;
320 } AVG726Context;
322 static int g726_init(AVCodecContext * avctx)
324 AVG726Context* c = (AVG726Context*)avctx->priv_data;
326 if (avctx->channels != 1 ||
327 (avctx->bit_rate != 16000 && avctx->bit_rate != 24000 &&
328 avctx->bit_rate != 32000 && avctx->bit_rate != 40000)) {
329 av_log(avctx, AV_LOG_ERROR, "G726: unsupported audio format\n");
330 return -1;
332 if (avctx->sample_rate != 8000 && avctx->strict_std_compliance>FF_COMPLIANCE_INOFFICIAL) {
333 av_log(avctx, AV_LOG_ERROR, "G726: unsupported audio format\n");
334 return -1;
336 g726_reset(&c->c, avctx->bit_rate);
337 c->code_size = c->c.tbls->bits;
338 c->bit_buffer = 0;
339 c->bits_left = 0;
341 avctx->coded_frame = avcodec_alloc_frame();
342 if (!avctx->coded_frame)
343 return AVERROR(ENOMEM);
344 avctx->coded_frame->key_frame = 1;
346 return 0;
349 static int g726_close(AVCodecContext *avctx)
351 av_freep(&avctx->coded_frame);
352 return 0;
355 #ifdef CONFIG_ENCODERS
356 static int g726_encode_frame(AVCodecContext *avctx,
357 uint8_t *dst, int buf_size, void *data)
359 AVG726Context *c = avctx->priv_data;
360 short *samples = data;
361 PutBitContext pb;
363 init_put_bits(&pb, dst, 1024*1024);
365 for (; buf_size; buf_size--)
366 put_bits(&pb, c->code_size, g726_encode(&c->c, *samples++));
368 flush_put_bits(&pb);
370 return put_bits_count(&pb)>>3;
372 #endif
374 static int g726_decode_frame(AVCodecContext *avctx,
375 void *data, int *data_size,
376 uint8_t *buf, int buf_size)
378 AVG726Context *c = avctx->priv_data;
379 short *samples = data;
380 uint8_t code;
381 uint8_t mask;
382 GetBitContext gb;
384 if (!buf_size)
385 goto out;
387 mask = (1<<c->code_size) - 1;
388 init_get_bits(&gb, buf, buf_size * 8);
389 if (c->bits_left) {
390 int s = c->code_size - c->bits_left;;
391 code = (c->bit_buffer << s) | get_bits(&gb, s);
392 *samples++ = g726_decode(&c->c, code & mask);
395 while (get_bits_count(&gb) + c->code_size <= buf_size*8)
396 *samples++ = g726_decode(&c->c, get_bits(&gb, c->code_size) & mask);
398 c->bits_left = buf_size*8 - get_bits_count(&gb);
399 c->bit_buffer = get_bits(&gb, c->bits_left);
401 out:
402 *data_size = (uint8_t*)samples - (uint8_t*)data;
403 return buf_size;
406 #ifdef CONFIG_ENCODERS
407 AVCodec adpcm_g726_encoder = {
408 "g726",
409 CODEC_TYPE_AUDIO,
410 CODEC_ID_ADPCM_G726,
411 sizeof(AVG726Context),
412 g726_init,
413 g726_encode_frame,
414 g726_close,
415 NULL,
417 #endif //CONFIG_ENCODERS
419 AVCodec adpcm_g726_decoder = {
420 "g726",
421 CODEC_TYPE_AUDIO,
422 CODEC_ID_ADPCM_G726,
423 sizeof(AVG726Context),
424 g726_init,
425 NULL,
426 g726_close,
427 g726_decode_frame,