FFmpeg: libavcodec/ralf.c Source File

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libavcodec

ralf.c

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1 /*

2 * RealAudio Lossless decoder

3 *

5 *

6 * This file is part of FFmpeg.

7 *

8 * FFmpeg is free software; you can redistribute it and/or

9 * modify it under the terms of the GNU Lesser General Public

10 * License as published by the Free Software Foundation; either

11 * version 2.1 of the License, or (at your option) any later version.

12 *

13 * FFmpeg is distributed in the hope that it will be useful,

14 * but WITHOUT ANY WARRANTY; without even the implied warranty of

15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU

16 * Lesser General Public License for more details.

17 *

18 * You should have received a copy of the GNU Lesser General Public

19 * License along with FFmpeg; if not, write to the Free Software

20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

21 */

23 /**

24 * @file

25 * This is a decoder for Real Audio Lossless format.

26 * Dedicated to the mastermind behind it, Ralph Wiggum.

27 */

29 #include "libavutil/attributes.h"

30 #include "libavutil/channel_layout.h"

31 #include "avcodec.h"

32 #include "get_bits.h"

33 #include "golomb.h"

34 #include "internal.h"

35 #include "unary.h"

36 #include "ralfdata.h"

38 #define FILTER_NONE 0

39 #define FILTER_RAW 642

41 typedef struct VLCSet {

42 VLC filter_params;

43 VLC bias;

44 VLC coding_mode;

45 VLC filter_coeffs[10][11];

46 VLC short_codes[15];

47 VLC long_codes[125];

48 } VLCSet;

50 #define RALF_MAX_PKT_SIZE 8192

52 typedef struct RALFContext {

53 int version;

54 int max_frame_size;

55 VLCSet sets[3];

56 int32_t channel_data[2][4096];

58 int filter_params; ///< combined filter parameters for the current channel data

59 int filter_length; ///< length of the filter for the current channel data

60 int filter_bits; ///< filter precision for the current channel data

61 int32_t filter[64];

63 int bias[2]; ///< a constant value added to channel data after filtering

65 int num_blocks; ///< number of blocks inside the frame

66 int sample_offset;

67 int block_size[1 << 12]; ///< size of the blocks

68 int block_pts[1 << 12]; ///< block start time (in milliseconds)

70 uint8_t pkt[16384];

71 int has_pkt;

72 } RALFContext;

74 #define MAX_ELEMS 644 // no RALF table uses more than that

76 static av_cold int init_ralf_vlc(VLC *vlc, const uint8_t *data, int elems)

77 {

78 uint8_t lens[MAX_ELEMS];

79 uint16_t codes[MAX_ELEMS];

80 int counts[17], prefixes[18];

81 int i, cur_len;

82 int max_bits = 0;

83 int nb = 0;

85 for (i = 0; i <= 16; i++)

86 counts[i] = 0;

87 for (i = 0; i < elems; i++) {

88 cur_len = (nb ? *data & 0xF : *data >> 4) + 1;

89 counts[cur_len]++;

90 max_bits = FFMAX(max_bits, cur_len);

91 lens[i] = cur_len;

92 data += nb;

93 nb ^= 1;

94 }

95 prefixes[1] = 0;

96 for (i = 1; i <= 16; i++)

97 prefixes[i + 1] = (prefixes[i] + counts[i]) << 1;

99 for (i = 0; i < elems; i++)

100 codes[i] = prefixes[lens[i]]++;

101

102 return ff_init_vlc_sparse(vlc, FFMIN(max_bits, 9), elems,

103 lens, 1, 1, codes, 2, 2, NULL, 0, 0, 0);

104 }

105

106 static av_cold int decode_close(AVCodecContext *avctx)

107 {

108 RALFContext *ctx = avctx->priv_data;

109 int i, j, k;

110

111 for (i = 0; i < 3; i++) {

112 ff_free_vlc(&ctx->sets[i].filter_params);

113 ff_free_vlc(&ctx->sets[i].bias);

114 ff_free_vlc(&ctx->sets[i].coding_mode);

115 for (j = 0; j < 10; j++)

116 for (k = 0; k < 11; k++)

117 ff_free_vlc(&ctx->sets[i].filter_coeffs[j][k]);

118 for (j = 0; j < 15; j++)

119 ff_free_vlc(&ctx->sets[i].short_codes[j]);

120 for (j = 0; j < 125; j++)

121 ff_free_vlc(&ctx->sets[i].long_codes[j]);

122 }

123

124 return 0;

125 }

126

127 static av_cold int decode_init(AVCodecContext *avctx)

128 {

129 RALFContext *ctx = avctx->priv_data;

130 int i, j, k;

131 int ret;

132

133 if (avctx->extradata_size < 24 || memcmp(avctx->extradata, "LSD:", 4)) {

134 av_log(avctx, AV_LOG_ERROR, "Extradata is not groovy, dude\n");

135 return AVERROR_INVALIDDATA;

136 }

137

138 ctx->version = AV_RB16(avctx->extradata + 4);

139 if (ctx->version != 0x103) {

140 avpriv_request_sample(avctx, "Unknown version %X", ctx->version);

141 return AVERROR_PATCHWELCOME;

142 }

143

144 avctx->channels = AV_RB16(avctx->extradata + 8);

145 avctx->sample_rate = AV_RB32(avctx->extradata + 12);

146 if (avctx->channels < 1 || avctx->channels > 2

147 || avctx->sample_rate < 8000 || avctx->sample_rate > 96000) {

148 av_log(avctx, AV_LOG_ERROR, "Invalid coding parameters %d Hz %d ch\n",

149 avctx->sample_rate, avctx->channels);

150 return AVERROR_INVALIDDATA;

151 }

152 avctx->sample_fmt = AV_SAMPLE_FMT_S16P;

153 avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO

154 : AV_CH_LAYOUT_MONO;

155

156 ctx->max_frame_size = AV_RB32(avctx->extradata + 16);

157 if (ctx->max_frame_size > (1 << 20) || !ctx->max_frame_size) {

158 av_log(avctx, AV_LOG_ERROR, "invalid frame size %d\n",

159 ctx->max_frame_size);

160 }

161 ctx->max_frame_size = FFMAX(ctx->max_frame_size, avctx->sample_rate);

162

163 for (i = 0; i < 3; i++) {

164 ret = init_ralf_vlc(&ctx->sets[i].filter_params, filter_param_def[i],

165 FILTERPARAM_ELEMENTS);

166 if (ret < 0) {

167 decode_close(avctx);

168 return ret;

169 }

170 ret = init_ralf_vlc(&ctx->sets[i].bias, bias_def[i], BIAS_ELEMENTS);

171 if (ret < 0) {

172 decode_close(avctx);

173 return ret;

174 }

175 ret = init_ralf_vlc(&ctx->sets[i].coding_mode, coding_mode_def[i],

176 CODING_MODE_ELEMENTS);

177 if (ret < 0) {

178 decode_close(avctx);

179 return ret;

180 }

181 for (j = 0; j < 10; j++) {

182 for (k = 0; k < 11; k++) {

183 ret = init_ralf_vlc(&ctx->sets[i].filter_coeffs[j][k],

184 filter_coeffs_def[i][j][k],

185 FILTER_COEFFS_ELEMENTS);

186 if (ret < 0) {

187 decode_close(avctx);

188 return ret;

189 }

190 }

191 }

192 for (j = 0; j < 15; j++) {

193 ret = init_ralf_vlc(&ctx->sets[i].short_codes[j],

194 short_codes_def[i][j], SHORT_CODES_ELEMENTS);

195 if (ret < 0) {

196 decode_close(avctx);

197 return ret;

198 }

199 }

200 for (j = 0; j < 125; j++) {

201 ret = init_ralf_vlc(&ctx->sets[i].long_codes[j],

202 long_codes_def[i][j], LONG_CODES_ELEMENTS);

203 if (ret < 0) {

204 decode_close(avctx);

205 return ret;

206 }

207 }

208 }

209

210 return 0;

211 }

212

213 static inline int extend_code(GetBitContext *gb, int val, int range, int bits)

214 {

215 if (val == 0) {

216 val = -range - get_ue_golomb(gb);

217 } else if (val == range * 2) {

218 val = range + get_ue_golomb(gb);

219 } else {

220 val -= range;

221 }

222 if (bits)

223 val = (val << bits) | get_bits(gb, bits);

224 return val;

225 }

226

227 static int decode_channel(RALFContext *ctx, GetBitContext *gb, int ch,

228 int length, int mode, int bits)

229 {

230 int i, t;

231 int code_params;

232 VLCSet *set = ctx->sets + mode;

233 VLC *code_vlc; int range, range2, add_bits;

234 int *dst = ctx->channel_data[ch];

235

236 ctx->filter_params = get_vlc2(gb, set->filter_params.table, 9, 2);

237 ctx->filter_bits = (ctx->filter_params - 2) >> 6;

238 ctx->filter_length = ctx->filter_params - (ctx->filter_bits << 6) - 1;

239

240 if (ctx->filter_params == FILTER_RAW) {

241 for (i = 0; i < length; i++)

242 dst[i] = get_bits(gb, bits);

243 ctx->bias[ch] = 0;

244 return 0;

245 }

246

247 ctx->bias[ch] = get_vlc2(gb, set->bias.table, 9, 2);

248 ctx->bias[ch] = extend_code(gb, ctx->bias[ch], 127, 4);

249

250 if (ctx->filter_params == FILTER_NONE) {

251 memset(dst, 0, sizeof(*dst) * length);

252 return 0;

253 }

254

255 if (ctx->filter_params > 1) {

256 int cmode = 0, coeff = 0;

257 VLC *vlc = set->filter_coeffs[ctx->filter_bits] + 5;

258

259 add_bits = ctx->filter_bits;

260

261 for (i = 0; i < ctx->filter_length; i++) {

262 t = get_vlc2(gb, vlc[cmode].table, vlc[cmode].bits, 2);

263 t = extend_code(gb, t, 21, add_bits);

264 if (!cmode)

265 coeff -= 12 << add_bits;

266 coeff = t - coeff;

267 ctx->filter[i] = coeff;

268

269 cmode = coeff >> add_bits;

270 if (cmode < 0) {

271 cmode = -1 - av_log2(-cmode);

272 if (cmode < -5)

273 cmode = -5;

274 } else if (cmode > 0) {

275 cmode = 1 + av_log2(cmode);

276 if (cmode > 5)

277 cmode = 5;

278 }

279 }

280 }

281

282 code_params = get_vlc2(gb, set->coding_mode.table, set->coding_mode.bits, 2);

283 if (code_params >= 15) {

284 add_bits = av_clip((code_params / 5 - 3) / 2, 0, 10);

285 if (add_bits > 9 && (code_params % 5) != 2)

286 add_bits--;

287 range = 10;

288 range2 = 21;

289 code_vlc = set->long_codes + code_params - 15;

290 } else {

291 add_bits = 0;

292 range = 6;

293 range2 = 13;

294 code_vlc = set->short_codes + code_params;

295 }

296

297 for (i = 0; i < length; i += 2) {

298 int code1, code2;

299

300 t = get_vlc2(gb, code_vlc->table, code_vlc->bits, 2);

301 code1 = t / range2;

302 code2 = t % range2;

303 dst[i] = extend_code(gb, code1, range, 0) << add_bits;

304 dst[i + 1] = extend_code(gb, code2, range, 0) << add_bits;

305 if (add_bits) {

306 dst[i] |= get_bits(gb, add_bits);

307 dst[i + 1] |= get_bits(gb, add_bits);

308 }

309 }

310

311 return 0;

312 }

313

314 static void apply_lpc(RALFContext *ctx, int ch, int length, int bits)

315 {

316 int i, j, acc;

317 int *audio = ctx->channel_data[ch];

318 int bias = 1 << (ctx->filter_bits - 1);

319 int max_clip = (1 << bits) - 1, min_clip = -max_clip - 1;

320

321 for (i = 1; i < length; i++) {

322 int flen = FFMIN(ctx->filter_length, i);

323

324 acc = 0;

325 for (j = 0; j < flen; j++)

326 acc += ctx->filter[j] * audio[i - j - 1];

327 if (acc < 0) {

328 acc = (acc + bias - 1) >> ctx->filter_bits;

329 acc = FFMAX(acc, min_clip);

330 } else {

331 acc = (acc + bias) >> ctx->filter_bits;

332 acc = FFMIN(acc, max_clip);

333 }

334 audio[i] += acc;

335 }

336 }

337

338 static int decode_block(AVCodecContext *avctx, GetBitContext *gb,

339 int16_t *dst0, int16_t *dst1)

340 {

341 RALFContext *ctx = avctx->priv_data;

342 int len, ch, ret;

343 int dmode, mode[2], bits[2];

344 int *ch0, *ch1;

345 int i, t, t2;

346

347 len = 12 - get_unary(gb, 0, 6);

348

349 if (len <= 7) len ^= 1; // codes for length = 6 and 7 are swapped

350 len = 1 << len;

351

352 if (ctx->sample_offset + len > ctx->max_frame_size) {

353 av_log(avctx, AV_LOG_ERROR,

354 "Decoder's stomach is crying, it ate too many samples\n");

355 return AVERROR_INVALIDDATA;

356 }

357

358 if (avctx->channels > 1)

359 dmode = get_bits(gb, 2) + 1;

360 else

361 dmode = 0;

362

363 mode[0] = (dmode == 4) ? 1 : 0;

364 mode[1] = (dmode >= 2) ? 2 : 0;

365 bits[0] = 16;

366 bits[1] = (mode[1] == 2) ? 17 : 16;

367

368 for (ch = 0; ch < avctx->channels; ch++) {

369 if ((ret = decode_channel(ctx, gb, ch, len, mode[ch], bits[ch])) < 0)

370 return ret;

371 if (ctx->filter_params > 1 && ctx->filter_params != FILTER_RAW) {

372 ctx->filter_bits += 3;

373 apply_lpc(ctx, ch, len, bits[ch]);

374 }

375 if (get_bits_left(gb) < 0)

376 return AVERROR_INVALIDDATA;

377 }

378 ch0 = ctx->channel_data[0];

379 ch1 = ctx->channel_data[1];

380 switch (dmode) {

381 case 0:

382 for (i = 0; i < len; i++)

383 dst0[i] = ch0[i] + ctx->bias[0];

384 break;

385 case 1:

386 for (i = 0; i < len; i++) {

387 dst0[i] = ch0[i] + ctx->bias[0];

388 dst1[i] = ch1[i] + ctx->bias[1];

389 }

390 break;

391 case 2:

392 for (i = 0; i < len; i++) {

393 ch0[i] += ctx->bias[0];

394 dst0[i] = ch0[i];

395 dst1[i] = ch0[i] - (ch1[i] + ctx->bias[1]);

396 }

397 break;

398 case 3:

399 for (i = 0; i < len; i++) {

400 t = ch0[i] + ctx->bias[0];

401 t2 = ch1[i] + ctx->bias[1];

402 dst0[i] = t + t2;

403 dst1[i] = t;

404 }

405 break;

406 case 4:

407 for (i = 0; i < len; i++) {

408 t = ch1[i] + ctx->bias[1];

409 t2 = ((ch0[i] + ctx->bias[0]) << 1) | (t & 1);

410 dst0[i] = (t2 + t) / 2;

411 dst1[i] = (t2 - t) / 2;

412 }

413 break;

414 }

415

416 ctx->sample_offset += len;

417

418 return 0;

419 }

420

421 static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,

422 AVPacket *avpkt)

423 {

424 RALFContext *ctx = avctx->priv_data;

425 AVFrame *frame = data;

426 int16_t *samples0;

427 int16_t *samples1;

428 int ret;

429 GetBitContext gb;

430 int table_size, table_bytes, i;

431 const uint8_t *src, *block_pointer;

432 int src_size;

433 int bytes_left;

434

435 if (ctx->has_pkt) {

436 ctx->has_pkt = 0;

437 table_bytes = (AV_RB16(avpkt->data) + 7) >> 3;

438 if (table_bytes + 3 > avpkt->size || avpkt->size > RALF_MAX_PKT_SIZE) {

439 av_log(avctx, AV_LOG_ERROR, "Wrong packet's breath smells of wrong data!\n");

440 return AVERROR_INVALIDDATA;

441 }

442 if (memcmp(ctx->pkt, avpkt->data, 2 + table_bytes)) {

443 av_log(avctx, AV_LOG_ERROR, "Wrong packet tails are wrong!\n");

444 return AVERROR_INVALIDDATA;

445 }

446

447 src = ctx->pkt;

448 src_size = RALF_MAX_PKT_SIZE + avpkt->size;

449 memcpy(ctx->pkt + RALF_MAX_PKT_SIZE, avpkt->data + 2 + table_bytes,

450 avpkt->size - 2 - table_bytes);

451 } else {

452 if (avpkt->size == RALF_MAX_PKT_SIZE) {

453 memcpy(ctx->pkt, avpkt->data, avpkt->size);

454 ctx->has_pkt = 1;

455 *got_frame_ptr = 0;

456

457 return avpkt->size;

458 }

459 src = avpkt->data;

460 src_size = avpkt->size;

461 }

462

463 frame->nb_samples = ctx->max_frame_size;

464 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)

465 return ret;

466 samples0 = (int16_t *)frame->data[0];

467 samples1 = (int16_t *)frame->data[1];

468

469 if (src_size < 5) {

470 av_log(avctx, AV_LOG_ERROR, "too short packets are too short!\n");

471 return AVERROR_INVALIDDATA;

472 }

473 table_size = AV_RB16(src);

474 table_bytes = (table_size + 7) >> 3;

475 if (src_size < table_bytes + 3) {

476 av_log(avctx, AV_LOG_ERROR, "short packets are short!\n");

477 return AVERROR_INVALIDDATA;

478 }

479 init_get_bits(&gb, src + 2, table_size);

480 ctx->num_blocks = 0;

481 while (get_bits_left(&gb) > 0) {

482 ctx->block_size[ctx->num_blocks] = get_bits(&gb, 13 + avctx->channels);

483 if (get_bits1(&gb)) {

484 ctx->block_pts[ctx->num_blocks] = get_bits(&gb, 9);

485 } else {

486 ctx->block_pts[ctx->num_blocks] = 0;

487 }

488 ctx->num_blocks++;

489 }

490

491 block_pointer = src + table_bytes + 2;

492 bytes_left = src_size - table_bytes - 2;

493 ctx->sample_offset = 0;

494 for (i = 0; i < ctx->num_blocks; i++) {

495 if (bytes_left < ctx->block_size[i]) {

496 av_log(avctx, AV_LOG_ERROR, "I'm pedaling backwards\n");

497 break;

498 }

499 init_get_bits(&gb, block_pointer, ctx->block_size[i] * 8);

500 if (decode_block(avctx, &gb, samples0 + ctx->sample_offset,

501 samples1 + ctx->sample_offset) < 0) {

502 av_log(avctx, AV_LOG_ERROR, "Sir, I got carsick in your office. Not decoding the rest of packet.\n");

503 break;

504 }

505 block_pointer += ctx->block_size[i];

506 bytes_left -= ctx->block_size[i];

507 }

508

509 frame->nb_samples = ctx->sample_offset;

510 *got_frame_ptr = ctx->sample_offset > 0;

511

512 return avpkt->size;

513 }

514

515 static void decode_flush(AVCodecContext *avctx)

516 {

517 RALFContext *ctx = avctx->priv_data;

518

519 ctx->has_pkt = 0;

520 }

521

522

523 AVCodec ff_ralf_decoder = {

524 .name = "ralf",

525 .long_name = NULL_IF_CONFIG_SMALL("RealAudio Lossless"),

526 .type = AVMEDIA_TYPE_AUDIO,

527 .id = AV_CODEC_ID_RALF,

528 .priv_data_size = sizeof(RALFContext),

529 .init = decode_init,

530 .close = decode_close,

531 .decode = decode_frame,

532 .flush = decode_flush,

533 .capabilities = AV_CODEC_CAP_DR1,

534 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

535 AV_SAMPLE_FMT_NONE },

536 };

NULL

#define NULL

Definition: coverity.c:32

RALFContext::max_frame_size

int max_frame_size

Definition: ralf.c:54

val

const char const char void * val

Definition: avisynth_c.h:771

AVERROR_INVALIDDATA

#define AVERROR_INVALIDDATA

Invalid data found when processing input.

Definition: error.h:59

AVFrame

This structure describes decoded (raw) audio or video data.

Definition: frame.h:201

data

ptrdiff_t const GLvoid * data

Definition: opengl_enc.c:101

flush

static void flush(AVCodecContext *avctx)

Definition: aacdec_template.c:500

AV_CODEC_ID_RALF

Definition: avcodec.h:616

RALFContext::num_blocks

int num_blocks

number of blocks inside the frame

Definition: ralf.c:65

get_bits

static unsigned int get_bits(GetBitContext *s, int n)

Read 1-25 bits.

Definition: get_bits.h:261

init

static av_cold int init(AVCodecContext *avctx)

Definition: avrndec.c:35

acc

int acc

Definition: yuv2rgb.c:546

ff_init_vlc_sparse

int ff_init_vlc_sparse(VLC *vlc_arg, int nb_bits, int nb_codes, const void *bits, int bits_wrap, int bits_size, const void *codes, int codes_wrap, int codes_size, const void *symbols, int symbols_wrap, int symbols_size, int flags)

Definition: bitstream.c:268

AVPacket::size

int size

Definition: avcodec.h:1680

AV_SAMPLE_FMT_NONE

Definition: samplefmt.h:59

VLCSet::filter_params

VLC filter_params

Definition: ralf.c:42

av_log2

int av_log2(unsigned v)

Definition: intmath.c:26

AV_CH_LAYOUT_STEREO

#define AV_CH_LAYOUT_STEREO

Definition: channel_layout.h:86

AV_RB16

uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_RB16

Definition: bytestream.h:87

src

#define src

Definition: vp8dsp.c:254

AVCodec

AVCodec.

Definition: avcodec.h:3739

set

static void set(uint8_t *a[], int ch, int index, int ch_count, enum AVSampleFormat f, double v)

Definition: swresample.c:59

decode

static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)

Definition: decode_audio.c:42

attributes.h

Macro definitions for various function/variable attributes.

bias_def

static const uint8_t bias_def[3][128]

Definition: ralfdata.h:123

MAX_ELEMS

#define MAX_ELEMS

Definition: ralf.c:74

FILTER_NONE

#define FILTER_NONE

Definition: ralf.c:38

filter_coeffs_def

static const uint8_t filter_coeffs_def[3][10][11][24]

Definition: ralfdata.h:188

avpriv_request_sample

void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2

Log a generic warning message about a missing feature.

bits

uint8_t bits

Definition: crc.c:296

AVCodecContext::sample_fmt

enum AVSampleFormat sample_fmt

audio sample format

Definition: avcodec.h:2531

uint8_t

Definition: audio_convert.c:194

av_cold

#define av_cold

Definition: attributes.h:82

uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch

Definition: audioconvert.c:56

RALFContext::sample_offset

int sample_offset

Definition: ralf.c:66

decode_block

static int decode_block(AVCodecContext *avctx, GetBitContext *gb, int16_t *dst0, int16_t *dst1)

Definition: ralf.c:338

AVCodecContext::extradata

uint8_t * extradata

some codecs need / can use extradata like Huffman tables.

Definition: avcodec.h:1876

AV_RB32

uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32

Definition: bytestream.h:87

frame

static AVFrame * frame

Definition: demuxing_decoding.c:53

AVPacket::data

uint8_t * data

Definition: avcodec.h:1679

get_bits.h

bitstream reader API header.

RALFContext::version

int version

Definition: ralf.c:53

av_log

#define av_log(a,...)

Definition: tableprint_vlc.h:28

ralfdata.h

get_bits_left

static int get_bits_left(GetBitContext *gb)

Definition: get_bits.h:587

AV_LOG_ERROR

#define AV_LOG_ERROR

Something went wrong and cannot losslessly be recovered.

Definition: log.h:176

get_ue_golomb

static int get_ue_golomb(GetBitContext *gb)

Read an unsigned Exp-Golomb code in the range 0 to 8190.

Definition: golomb.h:53

RALFContext::filter_bits

int filter_bits

filter precision for the current channel data

Definition: ralf.c:60

decode_frame

static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)

Definition: ralf.c:421

RALFContext::has_pkt

int has_pkt

Definition: ralf.c:71

decode_flush

static void decode_flush(AVCodecContext *avctx)

Definition: ralf.c:515

table

static const struct endianess table[]

NULL_IF_CONFIG_SMALL

#define NULL_IF_CONFIG_SMALL(x)

Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...

Definition: internal.h:179

decode_close

static av_cold int decode_close(AVCodecContext *avctx)

Definition: ralf.c:106

decode_channel

static int decode_channel(RALFContext *ctx, GetBitContext *gb, int ch, int length, int mode, int bits)

Definition: ralf.c:227

long_codes_def

static const uint8_t long_codes_def[3][125][224]

Definition: ralfdata.h:2036

AVMEDIA_TYPE_AUDIO

Definition: avutil.h:202

RALF_MAX_PKT_SIZE

#define RALF_MAX_PKT_SIZE

Definition: ralf.c:50

VLCSet::coding_mode

VLC coding_mode

Definition: ralf.c:44

length

GLsizei GLsizei * length

Definition: opengl_enc.c:115

AVCodec::name

const char * name

Name of the codec implementation.

Definition: avcodec.h:3746

FFMAX

#define FFMAX(a, b)

Definition: common.h:94

VLC

Definition: vlc.h:26

AVCodecContext::channel_layout

uint64_t channel_layout

Audio channel layout.

Definition: avcodec.h:2574

RALFContext::channel_data

int32_t channel_data[2][4096]

Definition: ralf.c:56

RALFContext

Definition: ralf.c:52

RALFContext::sets

VLCSet sets[3]

Definition: ralf.c:55

channel_layout.h

audio channel layout utility functions

FILTER_COEFFS_ELEMENTS

#define FILTER_COEFFS_ELEMENTS

Definition: ralfdata.h:31

FFMIN

#define FFMIN(a, b)

Definition: common.h:96

FILTER_RAW

#define FILTER_RAW

Definition: ralf.c:39

SHORT_CODES_ELEMENTS

#define SHORT_CODES_ELEMENTS

Definition: ralfdata.h:32

int32_t

Definition: audio_convert.c:194

ctx

AVFormatContext * ctx

Definition: movenc.c:48

coding_mode_def

static const uint8_t coding_mode_def[3][72]

Definition: ralfdata.h:163

get_vlc2

static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)

Parse a vlc code.

Definition: get_bits.h:554

VLC::bits

int bits

Definition: vlc.h:27

AVERROR_PATCHWELCOME

#define AVERROR_PATCHWELCOME

Not yet implemented in FFmpeg, patches welcome.

Definition: error.h:62

RALFContext::filter

int32_t filter[64]

Definition: ralf.c:61

RALFContext::filter_params

int filter_params

combined filter parameters for the current channel data

Definition: ralf.c:58

RALFContext::pkt

uint8_t pkt[16384]

Definition: ralf.c:70

avcodec.h

Libavcodec external API header.

AVSampleFormat

Audio sample formats.

Definition: samplefmt.h:58

AVCodecContext::sample_rate

int sample_rate

samples per second

Definition: avcodec.h:2523

ff_ralf_decoder

AVCodec ff_ralf_decoder

Definition: ralf.c:523

FILTERPARAM_ELEMENTS

#define FILTERPARAM_ELEMENTS

Definition: ralfdata.h:28

AVCodecContext

main external API structure.

Definition: avcodec.h:1761

code_vlc

static VLC code_vlc

Definition: wnv1.c:45

ff_get_buffer

int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)

Get a buffer for a frame.

Definition: decode.c:1669

VLCSet::long_codes

VLC long_codes[125]

Definition: ralf.c:47

AVCodecContext::extradata_size

int extradata_size

Definition: avcodec.h:1877

get_bits1

static unsigned int get_bits1(GetBitContext *s)

Definition: get_bits.h:313

unary.h

VLCSet

Definition: ralf.c:41

init_ralf_vlc

static av_cold int init_ralf_vlc(VLC *vlc, const uint8_t *data, int elems)

Definition: ralf.c:76

init_get_bits

static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)

Initialize GetBitContext.

Definition: get_bits.h:425

VLCSet::short_codes

VLC short_codes[15]

Definition: ralf.c:46

VLCSet::filter_coeffs

VLC filter_coeffs[10][11]

Definition: ralf.c:45

AVFrame::data

uint8_t * data[AV_NUM_DATA_POINTERS]

pointer to the picture/channel planes.

Definition: frame.h:215

RALFContext::block_size

int block_size[1<< 12]

size of the blocks

Definition: ralf.c:67

RALFContext::block_pts

int block_pts[1<< 12]

block start time (in milliseconds)

Definition: ralf.c:68

CODING_MODE_ELEMENTS

#define CODING_MODE_ELEMENTS

Definition: ralfdata.h:30

GetBitContext

Definition: get_bits.h:55

internal.h

common internal api header.

if(ret< 0)

Definition: vf_mcdeint.c:279

get_unary

static int get_unary(GetBitContext *gb, int stop, int len)

Get unary code of limited length.

Definition: unary.h:33

short_codes_def

static const uint8_t short_codes_def[3][15][88]

Definition: ralfdata.h:1577

VLCSet::bias

VLC bias

Definition: ralf.c:43

RALFContext::filter_length

int filter_length

length of the filter for the current channel data

Definition: ralf.c:59

AVCodecContext::priv_data

void * priv_data

Definition: avcodec.h:1803

decode_init

static av_cold int decode_init(AVCodecContext *avctx)

Definition: ralf.c:127

extend_code

static int extend_code(GetBitContext *gb, int val, int range, int bits)

Definition: ralf.c:213

len

int len

Definition: vorbis_enc_data.h:452

AVCodecContext::channels

int channels

number of audio channels

Definition: avcodec.h:2524

VLC::table

VLC_TYPE(* table)[2]

code, bits

Definition: vlc.h:28

filter_param_def

static const uint8_t filter_param_def[3][324]

Definition: ralfdata.h:35

coeff

static const double coeff[2][5]

Definition: vf_owdenoise.c:72

RALFContext::bias

int bias[2]

a constant value added to channel data after filtering

Definition: ralf.c:63

sample_fmts

static enum AVSampleFormat sample_fmts[]

Definition: adpcmenc.c:701

BIAS_ELEMENTS

#define BIAS_ELEMENTS

Definition: ralfdata.h:29

AV_SAMPLE_FMT_S16P

signed 16 bits, planar

Definition: samplefmt.h:67

LONG_CODES_ELEMENTS

#define LONG_CODES_ELEMENTS

Definition: ralfdata.h:33

AV_CH_LAYOUT_MONO

#define AV_CH_LAYOUT_MONO

Definition: channel_layout.h:85

apply_lpc

static void apply_lpc(RALFContext *ctx, int ch, int length, int bits)

Definition: ralf.c:314

golomb.h

exp golomb vlc stuff

AVPacket

This structure stores compressed data.

Definition: avcodec.h:1656

ff_free_vlc

void ff_free_vlc(VLC *vlc)

Definition: bitstream.c:354

mode

Use these values in ebur128_init (or'ed).

Definition: ebur128.h:83

AVFrame::nb_samples

int nb_samples

number of audio samples (per channel) described by this frame

Definition: frame.h:267

AV_CODEC_CAP_DR1

#define AV_CODEC_CAP_DR1

Codec uses get_buffer() for allocating buffers and supports custom allocators.

Definition: avcodec.h:1002

for

for(j=16;j >0;--j)

Definition: h264pred_template.c:469

#define t2

Definition: regdef.h:30

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