1 /*
2 * G.726 ADPCM audio codec
3 * Copyright (c) 2004 Roman Shaposhnik
4 *
5 * This is a very straightforward rendition of the G.726
6 * Section 4 "Computational Details".
7 *
8 * This file is part of FFmpeg.
9 *
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.
14 *
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.
19 *
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
23 */
24
25 #include "config_components.h"
26
27 #include <limits.h>
28
29 #include "libavutil/channel_layout.h"
30 #include "libavutil/opt.h"
31 #include "avcodec.h"
32 #include "codec_internal.h"
33 #include "encode.h"
34 #include "internal.h"
35 #include "get_bits.h"
36 #include "put_bits.h"
37
38 /**
39 * G.726 11-bit float.
40 * G.726 Standard uses rather odd 11-bit floating point arithmetic for
41 * numerous occasions. It's a mystery to me why they did it this way
42 * instead of simply using 32-bit integer arithmetic.
43 */
44 typedef struct Float11 {
45 uint8_t sign; /**< 1 bit sign */
46 uint8_t exp; /**< 4 bits exponent */
47 uint8_t mant; /**< 6 bits mantissa */
48 } Float11;
49
i2f(int i,Float11 * f)50 static inline Float11* i2f(int i, Float11* f)
51 {
52 f->sign = (i < 0);
53 if (f->sign)
54 i = -i;
55 f->exp = av_log2_16bit(i) + !!i;
56 f->mant = i? (i<<6) >> f->exp : 1<<5;
57 return f;
58 }
59
mult(Float11 * f1,Float11 * f2)60 static inline int16_t mult(Float11* f1, Float11* f2)
61 {
62 int res, exp;
63
64 exp = f1->exp + f2->exp;
65 res = (((f1->mant * f2->mant) + 0x30) >> 4);
66 res = exp > 19 ? res << (exp - 19) : res >> (19 - exp);
67 return (f1->sign ^ f2->sign) ? -res : res;
68 }
69
sgn(int value)70 static inline int sgn(int value)
71 {
72 return (value < 0) ? -1 : 1;
73 }
74
75 typedef struct G726Tables {
76 const int* quant; /**< quantization table */
77 const int16_t* iquant; /**< inverse quantization table */
78 const int16_t* W; /**< special table #1 ;-) */
79 const uint8_t* F; /**< special table #2 */
80 } G726Tables;
81
82 typedef struct G726Context {
83 AVClass *class;
84 G726Tables tbls; /**< static tables needed for computation */
85
86 Float11 sr[2]; /**< prev. reconstructed samples */
87 Float11 dq[6]; /**< prev. difference */
88 int a[2]; /**< second order predictor coeffs */
89 int b[6]; /**< sixth order predictor coeffs */
90 int pk[2]; /**< signs of prev. 2 sez + dq */
91
92 int ap; /**< scale factor control */
93 int yu; /**< fast scale factor */
94 int yl; /**< slow scale factor */
95 int dms; /**< short average magnitude of F[i] */
96 int dml; /**< long average magnitude of F[i] */
97 int td; /**< tone detect */
98
99 int se; /**< estimated signal for the next iteration */
100 int sez; /**< estimated second order prediction */
101 int y; /**< quantizer scaling factor for the next iteration */
102 int code_size;
103 int little_endian; /**< little-endian bitstream as used in aiff and Sun AU */
104 } G726Context;
105
106 static const int quant_tbl16[] = /**< 16kbit/s 2 bits per sample */
107 { 260, INT_MAX };
108 static const int16_t iquant_tbl16[] =
109 { 116, 365, 365, 116 };
110 static const int16_t W_tbl16[] =
111 { -22, 439, 439, -22 };
112 static const uint8_t F_tbl16[] =
113 { 0, 7, 7, 0 };
114
115 static const int quant_tbl24[] = /**< 24kbit/s 3 bits per sample */
116 { 7, 217, 330, INT_MAX };
117 static const int16_t iquant_tbl24[] =
118 { INT16_MIN, 135, 273, 373, 373, 273, 135, INT16_MIN };
119 static const int16_t W_tbl24[] =
120 { -4, 30, 137, 582, 582, 137, 30, -4 };
121 static const uint8_t F_tbl24[] =
122 { 0, 1, 2, 7, 7, 2, 1, 0 };
123
124 static const int quant_tbl32[] = /**< 32kbit/s 4 bits per sample */
125 { -125, 79, 177, 245, 299, 348, 399, INT_MAX };
126 static const int16_t iquant_tbl32[] =
127 { INT16_MIN, 4, 135, 213, 273, 323, 373, 425,
128 425, 373, 323, 273, 213, 135, 4, INT16_MIN };
129 static const int16_t W_tbl32[] =
130 { -12, 18, 41, 64, 112, 198, 355, 1122,
131 1122, 355, 198, 112, 64, 41, 18, -12};
132 static const uint8_t F_tbl32[] =
133 { 0, 0, 0, 1, 1, 1, 3, 7, 7, 3, 1, 1, 1, 0, 0, 0 };
134
135 static const int quant_tbl40[] = /**< 40kbit/s 5 bits per sample */
136 { -122, -16, 67, 138, 197, 249, 297, 338,
137 377, 412, 444, 474, 501, 527, 552, INT_MAX };
138 static const int16_t iquant_tbl40[] =
139 { INT16_MIN, -66, 28, 104, 169, 224, 274, 318,
140 358, 395, 429, 459, 488, 514, 539, 566,
141 566, 539, 514, 488, 459, 429, 395, 358,
142 318, 274, 224, 169, 104, 28, -66, INT16_MIN };
143 static const int16_t W_tbl40[] =
144 { 14, 14, 24, 39, 40, 41, 58, 100,
145 141, 179, 219, 280, 358, 440, 529, 696,
146 696, 529, 440, 358, 280, 219, 179, 141,
147 100, 58, 41, 40, 39, 24, 14, 14 };
148 static const uint8_t F_tbl40[] =
149 { 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 2, 3, 4, 5, 6, 6,
150 6, 6, 5, 4, 3, 2, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0 };
151
152 static const G726Tables G726Tables_pool[] =
153 {{ quant_tbl16, iquant_tbl16, W_tbl16, F_tbl16 },
154 { quant_tbl24, iquant_tbl24, W_tbl24, F_tbl24 },
155 { quant_tbl32, iquant_tbl32, W_tbl32, F_tbl32 },
156 { quant_tbl40, iquant_tbl40, W_tbl40, F_tbl40 }};
157
158
159 /**
160 * Paragraph 4.2.2 page 18: Adaptive quantizer.
161 */
quant(G726Context * c,int d)162 static inline uint8_t quant(G726Context* c, int d)
163 {
164 int sign, exp, i, dln;
165
166 sign = i = 0;
167 if (d < 0) {
168 sign = 1;
169 d = -d;
170 }
171 exp = av_log2_16bit(d);
172 dln = ((exp<<7) + (((d<<7)>>exp)&0x7f)) - (c->y>>2);
173
174 while (c->tbls.quant[i] < INT_MAX && c->tbls.quant[i] < dln)
175 ++i;
176
177 if (sign)
178 i = ~i;
179 if (c->code_size != 2 && i == 0) /* I'm not sure this is a good idea */
180 i = 0xff;
181
182 return i;
183 }
184
185 /**
186 * Paragraph 4.2.3 page 22: Inverse adaptive quantizer.
187 */
inverse_quant(G726Context * c,int i)188 static inline int16_t inverse_quant(G726Context* c, int i)
189 {
190 int dql, dex, dqt;
191
192 dql = c->tbls.iquant[i] + (c->y >> 2);
193 dex = (dql>>7) & 0xf; /* 4-bit exponent */
194 dqt = (1<<7) + (dql & 0x7f); /* log2 -> linear */
195 return (dql < 0) ? 0 : ((dqt<<dex) >> 7);
196 }
197
g726_decode(G726Context * c,int I)198 static int16_t g726_decode(G726Context* c, int I)
199 {
200 int dq, re_signal, pk0, fa1, i, tr, ylint, ylfrac, thr2, al, dq0;
201 Float11 f;
202 int I_sig= I >> (c->code_size - 1);
203
204 dq = inverse_quant(c, I);
205
206 /* Transition detect */
207 ylint = (c->yl >> 15);
208 ylfrac = (c->yl >> 10) & 0x1f;
209 thr2 = (ylint > 9) ? 0x1f << 10 : (0x20 + ylfrac) << ylint;
210 tr= (c->td == 1 && dq > ((3*thr2)>>2));
211
212 if (I_sig) /* get the sign */
213 dq = -dq;
214 re_signal = (int16_t)(c->se + dq);
215
216 /* Update second order predictor coefficient A2 and A1 */
217 pk0 = (c->sez + dq) ? sgn(c->sez + dq) : 0;
218 dq0 = dq ? sgn(dq) : 0;
219 if (tr) {
220 c->a[0] = 0;
221 c->a[1] = 0;
222 for (i=0; i<6; i++)
223 c->b[i] = 0;
224 } else {
225 /* This is a bit crazy, but it really is +255 not +256 */
226 fa1 = av_clip_intp2((-c->a[0]*c->pk[0]*pk0)>>5, 8);
227
228 c->a[1] += 128*pk0*c->pk[1] + fa1 - (c->a[1]>>7);
229 c->a[1] = av_clip(c->a[1], -12288, 12288);
230 c->a[0] += 64*3*pk0*c->pk[0] - (c->a[0] >> 8);
231 c->a[0] = av_clip(c->a[0], -(15360 - c->a[1]), 15360 - c->a[1]);
232
233 for (i=0; i<6; i++)
234 c->b[i] += 128*dq0*sgn(-c->dq[i].sign) - (c->b[i]>>8);
235 }
236
237 /* Update Dq and Sr and Pk */
238 c->pk[1] = c->pk[0];
239 c->pk[0] = pk0 ? pk0 : 1;
240 c->sr[1] = c->sr[0];
241 i2f(re_signal, &c->sr[0]);
242 for (i=5; i>0; i--)
243 c->dq[i] = c->dq[i-1];
244 i2f(dq, &c->dq[0]);
245 c->dq[0].sign = I_sig; /* Isn't it crazy ?!?! */
246
247 c->td = c->a[1] < -11776;
248
249 /* Update Ap */
250 c->dms += (c->tbls.F[I]<<4) + ((- c->dms) >> 5);
251 c->dml += (c->tbls.F[I]<<4) + ((- c->dml) >> 7);
252 if (tr)
253 c->ap = 256;
254 else {
255 c->ap += (-c->ap) >> 4;
256 if (c->y <= 1535 || c->td || abs((c->dms << 2) - c->dml) >= (c->dml >> 3))
257 c->ap += 0x20;
258 }
259
260 /* Update Yu and Yl */
261 c->yu = av_clip(c->y + c->tbls.W[I] + ((-c->y)>>5), 544, 5120);
262 c->yl += c->yu + ((-c->yl)>>6);
263
264 /* Next iteration for Y */
265 al = (c->ap >= 256) ? 1<<6 : c->ap >> 2;
266 c->y = (c->yl + (c->yu - (c->yl>>6))*al) >> 6;
267
268 /* Next iteration for SE and SEZ */
269 c->se = 0;
270 for (i=0; i<6; i++)
271 c->se += mult(i2f(c->b[i] >> 2, &f), &c->dq[i]);
272 c->sez = c->se >> 1;
273 for (i=0; i<2; i++)
274 c->se += mult(i2f(c->a[i] >> 2, &f), &c->sr[i]);
275 c->se >>= 1;
276
277 return av_clip(re_signal * 4, -0xffff, 0xffff);
278 }
279
g726_reset(G726Context * c)280 static av_cold int g726_reset(G726Context *c)
281 {
282 int i;
283
284 c->tbls = G726Tables_pool[c->code_size - 2];
285 for (i=0; i<2; i++) {
286 c->sr[i].mant = 1<<5;
287 c->pk[i] = 1;
288 }
289 for (i=0; i<6; i++) {
290 c->dq[i].mant = 1<<5;
291 }
292 c->yu = 544;
293 c->yl = 34816;
294
295 c->y = 544;
296
297 return 0;
298 }
299
300 #if CONFIG_ADPCM_G726_ENCODER || CONFIG_ADPCM_G726LE_ENCODER
g726_encode(G726Context * c,int16_t sig)301 static int16_t g726_encode(G726Context* c, int16_t sig)
302 {
303 uint8_t i;
304
305 i = av_mod_uintp2(quant(c, sig/4 - c->se), c->code_size);
306 g726_decode(c, i);
307 return i;
308 }
309
310 /* Interfacing to the libavcodec */
311
g726_encode_init(AVCodecContext * avctx)312 static av_cold int g726_encode_init(AVCodecContext *avctx)
313 {
314 G726Context* c = avctx->priv_data;
315
316 c->little_endian = !strcmp(avctx->codec->name, "g726le");
317
318 if (avctx->strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL &&
319 avctx->sample_rate != 8000) {
320 av_log(avctx, AV_LOG_ERROR, "Sample rates other than 8kHz are not "
321 "allowed when the compliance level is higher than unofficial. "
322 "Resample or reduce the compliance level.\n");
323 return AVERROR(EINVAL);
324 }
325 if (avctx->sample_rate <= 0) {
326 av_log(avctx, AV_LOG_ERROR, "Invalid sample rate %d\n",
327 avctx->sample_rate);
328 return AVERROR(EINVAL);
329 }
330
331 if (avctx->ch_layout.nb_channels != 1) {
332 av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
333 return AVERROR(EINVAL);
334 }
335
336 if (avctx->bit_rate)
337 c->code_size = (avctx->bit_rate + avctx->sample_rate/2) / avctx->sample_rate;
338
339 c->code_size = av_clip(c->code_size, 2, 5);
340 avctx->bit_rate = c->code_size * avctx->sample_rate;
341 avctx->bits_per_coded_sample = c->code_size;
342
343 g726_reset(c);
344
345 /* select a frame size that will end on a byte boundary and have a size of
346 approximately 1024 bytes */
347 avctx->frame_size = ((int[]){ 4096, 2736, 2048, 1640 })[c->code_size - 2];
348
349 return 0;
350 }
351
g726_encode_frame(AVCodecContext * avctx,AVPacket * avpkt,const AVFrame * frame,int * got_packet_ptr)352 static int g726_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
353 const AVFrame *frame, int *got_packet_ptr)
354 {
355 G726Context *c = avctx->priv_data;
356 const int16_t *samples = (const int16_t *)frame->data[0];
357 PutBitContext pb;
358 int i, ret, out_size;
359
360 out_size = (frame->nb_samples * c->code_size + 7) / 8;
361 if ((ret = ff_get_encode_buffer(avctx, avpkt, out_size, 0)) < 0)
362 return ret;
363 init_put_bits(&pb, avpkt->data, avpkt->size);
364
365 for (i = 0; i < frame->nb_samples; i++)
366 if (c->little_endian) {
367 put_bits_le(&pb, c->code_size, g726_encode(c, *samples++));
368 } else {
369 put_bits(&pb, c->code_size, g726_encode(c, *samples++));
370 }
371
372 if (c->little_endian) {
373 flush_put_bits_le(&pb);
374 } else {
375 flush_put_bits(&pb);
376 }
377
378 *got_packet_ptr = 1;
379 return 0;
380 }
381
382 #define OFFSET(x) offsetof(G726Context, x)
383 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
384 static const AVOption options[] = {
385 { "code_size", "Bits per code", OFFSET(code_size), AV_OPT_TYPE_INT, { .i64 = 4 }, 2, 5, AE },
386 { NULL },
387 };
388
389 static const AVClass g726_class = {
390 .class_name = "g726",
391 .item_name = av_default_item_name,
392 .option = options,
393 .version = LIBAVUTIL_VERSION_INT,
394 };
395
396 static const FFCodecDefault defaults[] = {
397 { "b", "0" },
398 { NULL },
399 };
400 #endif
401
402 #if CONFIG_ADPCM_G726_ENCODER
403 const FFCodec ff_adpcm_g726_encoder = {
404 .p.name = "g726",
405 .p.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
406 .p.type = AVMEDIA_TYPE_AUDIO,
407 .p.id = AV_CODEC_ID_ADPCM_G726,
408 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_SMALL_LAST_FRAME,
409 .priv_data_size = sizeof(G726Context),
410 .init = g726_encode_init,
411 FF_CODEC_ENCODE_CB(g726_encode_frame),
412 .p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
413 AV_SAMPLE_FMT_NONE },
414 .p.priv_class = &g726_class,
415 .defaults = defaults,
416 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
417 };
418 #endif
419
420 #if CONFIG_ADPCM_G726LE_ENCODER
421 const FFCodec ff_adpcm_g726le_encoder = {
422 .p.name = "g726le",
423 .p.long_name = NULL_IF_CONFIG_SMALL("G.726 little endian ADPCM (\"right-justified\")"),
424 .p.type = AVMEDIA_TYPE_AUDIO,
425 .p.id = AV_CODEC_ID_ADPCM_G726LE,
426 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_SMALL_LAST_FRAME,
427 .priv_data_size = sizeof(G726Context),
428 .init = g726_encode_init,
429 FF_CODEC_ENCODE_CB(g726_encode_frame),
430 .p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
431 AV_SAMPLE_FMT_NONE },
432 .p.priv_class = &g726_class,
433 .defaults = defaults,
434 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
435 };
436 #endif
437
438 #if CONFIG_ADPCM_G726_DECODER || CONFIG_ADPCM_G726LE_DECODER
g726_decode_init(AVCodecContext * avctx)439 static av_cold int g726_decode_init(AVCodecContext *avctx)
440 {
441 G726Context* c = avctx->priv_data;
442
443 if (avctx->ch_layout.nb_channels > 1){
444 avpriv_request_sample(avctx, "Decoding more than one channel");
445 return AVERROR_PATCHWELCOME;
446 }
447 av_channel_layout_uninit(&avctx->ch_layout);
448 avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
449
450 c->little_endian = !strcmp(avctx->codec->name, "g726le");
451
452 c->code_size = avctx->bits_per_coded_sample;
453 if (c->code_size < 2 || c->code_size > 5) {
454 av_log(avctx, AV_LOG_ERROR, "Invalid number of bits %d\n", c->code_size);
455 return AVERROR(EINVAL);
456 }
457 g726_reset(c);
458
459 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
460
461 return 0;
462 }
463
g726_decode_frame(AVCodecContext * avctx,AVFrame * frame,int * got_frame_ptr,AVPacket * avpkt)464 static int g726_decode_frame(AVCodecContext *avctx, AVFrame *frame,
465 int *got_frame_ptr, AVPacket *avpkt)
466 {
467 const uint8_t *buf = avpkt->data;
468 int buf_size = avpkt->size;
469 G726Context *c = avctx->priv_data;
470 int16_t *samples;
471 GetBitContext gb;
472 int out_samples, ret;
473
474 out_samples = buf_size * 8 / c->code_size;
475
476 /* get output buffer */
477 frame->nb_samples = out_samples;
478 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
479 return ret;
480 samples = (int16_t *)frame->data[0];
481
482 init_get_bits(&gb, buf, buf_size * 8);
483
484 while (out_samples--)
485 *samples++ = g726_decode(c, c->little_endian ?
486 get_bits_le(&gb, c->code_size) :
487 get_bits(&gb, c->code_size));
488
489 if (get_bits_left(&gb) > 0)
490 av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n");
491
492 *got_frame_ptr = 1;
493
494 return buf_size;
495 }
496
g726_decode_flush(AVCodecContext * avctx)497 static void g726_decode_flush(AVCodecContext *avctx)
498 {
499 G726Context *c = avctx->priv_data;
500 g726_reset(c);
501 }
502 #endif
503
504 #if CONFIG_ADPCM_G726_DECODER
505 const FFCodec ff_adpcm_g726_decoder = {
506 .p.name = "g726",
507 .p.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
508 .p.type = AVMEDIA_TYPE_AUDIO,
509 .p.id = AV_CODEC_ID_ADPCM_G726,
510 .priv_data_size = sizeof(G726Context),
511 .init = g726_decode_init,
512 FF_CODEC_DECODE_CB(g726_decode_frame),
513 .flush = g726_decode_flush,
514 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
515 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
516 };
517 #endif
518
519 #if CONFIG_ADPCM_G726LE_DECODER
520 const FFCodec ff_adpcm_g726le_decoder = {
521 .p.name = "g726le",
522 .p.type = AVMEDIA_TYPE_AUDIO,
523 .p.id = AV_CODEC_ID_ADPCM_G726LE,
524 .priv_data_size = sizeof(G726Context),
525 .init = g726_decode_init,
526 FF_CODEC_DECODE_CB(g726_decode_frame),
527 .flush = g726_decode_flush,
528 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
529 .p.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM little-endian"),
530 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
531 };
532 #endif
533