Typo in softfloat_reciprocal comment.
[FFMpeg-mirror/lagarith.git] / libavcodec / flacdec.c
blob781b4fadf1df3d8dcc3b9170276ed82469aa8695
1 /*
2 * FLAC (Free Lossless Audio Codec) decoder
3 * Copyright (c) 2003 Alex Beregszaszi
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 /**
23 * @file libavcodec/flacdec.c
24 * FLAC (Free Lossless Audio Codec) decoder
25 * @author Alex Beregszaszi
27 * For more information on the FLAC format, visit:
28 * http://flac.sourceforge.net/
30 * This decoder can be used in 1 of 2 ways: Either raw FLAC data can be fed
31 * through, starting from the initial 'fLaC' signature; or by passing the
32 * 34-byte streaminfo structure through avctx->extradata[_size] followed
33 * by data starting with the 0xFFF8 marker.
36 #include <limits.h>
38 #include "libavutil/crc.h"
39 #include "avcodec.h"
40 #include "internal.h"
41 #include "get_bits.h"
42 #include "bytestream.h"
43 #include "golomb.h"
44 #include "flac.h"
45 #include "flacdata.h"
47 #undef NDEBUG
48 #include <assert.h>
50 typedef struct FLACContext {
51 FLACSTREAMINFO
53 AVCodecContext *avctx; ///< parent AVCodecContext
54 GetBitContext gb; ///< GetBitContext initialized to start at the current frame
56 int blocksize; ///< number of samples in the current frame
57 int curr_bps; ///< bps for current subframe, adjusted for channel correlation and wasted bits
58 int sample_shift; ///< shift required to make output samples 16-bit or 32-bit
59 int is32; ///< flag to indicate if output should be 32-bit instead of 16-bit
60 int ch_mode; ///< channel decorrelation type in the current frame
61 int got_streaminfo; ///< indicates if the STREAMINFO has been read
63 int32_t *decoded[FLAC_MAX_CHANNELS]; ///< decoded samples
64 uint8_t *bitstream;
65 unsigned int bitstream_size;
66 unsigned int bitstream_index;
67 unsigned int allocated_bitstream_size;
68 } FLACContext;
70 static const int sample_size_table[] =
71 { 0, 8, 12, 0, 16, 20, 24, 0 };
73 static int64_t get_utf8(GetBitContext *gb)
75 int64_t val;
76 GET_UTF8(val, get_bits(gb, 8), return -1;)
77 return val;
80 static void allocate_buffers(FLACContext *s);
82 int ff_flac_is_extradata_valid(AVCodecContext *avctx,
83 enum FLACExtradataFormat *format,
84 uint8_t **streaminfo_start)
86 if (!avctx->extradata || avctx->extradata_size < FLAC_STREAMINFO_SIZE) {
87 av_log(avctx, AV_LOG_ERROR, "extradata NULL or too small.\n");
88 return 0;
90 if (AV_RL32(avctx->extradata) != MKTAG('f','L','a','C')) {
91 /* extradata contains STREAMINFO only */
92 if (avctx->extradata_size != FLAC_STREAMINFO_SIZE) {
93 av_log(avctx, AV_LOG_WARNING, "extradata contains %d bytes too many.\n",
94 FLAC_STREAMINFO_SIZE-avctx->extradata_size);
96 *format = FLAC_EXTRADATA_FORMAT_STREAMINFO;
97 *streaminfo_start = avctx->extradata;
98 } else {
99 if (avctx->extradata_size < 8+FLAC_STREAMINFO_SIZE) {
100 av_log(avctx, AV_LOG_ERROR, "extradata too small.\n");
101 return 0;
103 *format = FLAC_EXTRADATA_FORMAT_FULL_HEADER;
104 *streaminfo_start = &avctx->extradata[8];
106 return 1;
109 static av_cold int flac_decode_init(AVCodecContext *avctx)
111 enum FLACExtradataFormat format;
112 uint8_t *streaminfo;
113 FLACContext *s = avctx->priv_data;
114 s->avctx = avctx;
116 avctx->sample_fmt = SAMPLE_FMT_S16;
118 /* for now, the raw FLAC header is allowed to be passed to the decoder as
119 frame data instead of extradata. */
120 if (!avctx->extradata)
121 return 0;
123 if (!ff_flac_is_extradata_valid(avctx, &format, &streaminfo))
124 return -1;
126 /* initialize based on the demuxer-supplied streamdata header */
127 ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);
128 allocate_buffers(s);
129 s->got_streaminfo = 1;
131 return 0;
134 static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s)
136 av_log(avctx, AV_LOG_DEBUG, " Max Blocksize: %d\n", s->max_blocksize);
137 av_log(avctx, AV_LOG_DEBUG, " Max Framesize: %d\n", s->max_framesize);
138 av_log(avctx, AV_LOG_DEBUG, " Samplerate: %d\n", s->samplerate);
139 av_log(avctx, AV_LOG_DEBUG, " Channels: %d\n", s->channels);
140 av_log(avctx, AV_LOG_DEBUG, " Bits: %d\n", s->bps);
143 static void allocate_buffers(FLACContext *s)
145 int i;
147 assert(s->max_blocksize);
149 if (s->max_framesize == 0 && s->max_blocksize) {
150 s->max_framesize = ff_flac_get_max_frame_size(s->max_blocksize,
151 s->channels, s->bps);
154 for (i = 0; i < s->channels; i++) {
155 s->decoded[i] = av_realloc(s->decoded[i],
156 sizeof(int32_t)*s->max_blocksize);
159 if (s->allocated_bitstream_size < s->max_framesize)
160 s->bitstream= av_fast_realloc(s->bitstream,
161 &s->allocated_bitstream_size,
162 s->max_framesize);
165 void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
166 const uint8_t *buffer)
168 GetBitContext gb;
169 init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8);
171 skip_bits(&gb, 16); /* skip min blocksize */
172 s->max_blocksize = get_bits(&gb, 16);
173 if (s->max_blocksize < FLAC_MIN_BLOCKSIZE) {
174 av_log(avctx, AV_LOG_WARNING, "invalid max blocksize: %d\n",
175 s->max_blocksize);
176 s->max_blocksize = 16;
179 skip_bits(&gb, 24); /* skip min frame size */
180 s->max_framesize = get_bits_long(&gb, 24);
182 s->samplerate = get_bits_long(&gb, 20);
183 s->channels = get_bits(&gb, 3) + 1;
184 s->bps = get_bits(&gb, 5) + 1;
186 avctx->channels = s->channels;
187 avctx->sample_rate = s->samplerate;
188 avctx->bits_per_raw_sample = s->bps;
189 if (s->bps > 16)
190 avctx->sample_fmt = SAMPLE_FMT_S32;
191 else
192 avctx->sample_fmt = SAMPLE_FMT_S16;
194 s->samples = get_bits_long(&gb, 32) << 4;
195 s->samples |= get_bits(&gb, 4);
197 skip_bits_long(&gb, 64); /* md5 sum */
198 skip_bits_long(&gb, 64); /* md5 sum */
200 dump_headers(avctx, s);
203 void ff_flac_parse_block_header(const uint8_t *block_header,
204 int *last, int *type, int *size)
206 int tmp = bytestream_get_byte(&block_header);
207 if (last)
208 *last = tmp & 0x80;
209 if (type)
210 *type = tmp & 0x7F;
211 if (size)
212 *size = bytestream_get_be24(&block_header);
216 * Parse the STREAMINFO from an inline header.
217 * @param s the flac decoding context
218 * @param buf input buffer, starting with the "fLaC" marker
219 * @param buf_size buffer size
220 * @return non-zero if metadata is invalid
222 static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size)
224 int metadata_type, metadata_size;
226 if (buf_size < FLAC_STREAMINFO_SIZE+8) {
227 /* need more data */
228 return 0;
230 ff_flac_parse_block_header(&buf[4], NULL, &metadata_type, &metadata_size);
231 if (metadata_type != FLAC_METADATA_TYPE_STREAMINFO ||
232 metadata_size != FLAC_STREAMINFO_SIZE) {
233 return AVERROR_INVALIDDATA;
235 ff_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, &buf[8]);
236 allocate_buffers(s);
237 s->got_streaminfo = 1;
239 return 0;
243 * Determine the size of an inline header.
244 * @param buf input buffer, starting with the "fLaC" marker
245 * @param buf_size buffer size
246 * @return number of bytes in the header, or 0 if more data is needed
248 static int get_metadata_size(const uint8_t *buf, int buf_size)
250 int metadata_last, metadata_size;
251 const uint8_t *buf_end = buf + buf_size;
253 buf += 4;
254 do {
255 ff_flac_parse_block_header(buf, &metadata_last, NULL, &metadata_size);
256 buf += 4;
257 if (buf + metadata_size > buf_end) {
258 /* need more data in order to read the complete header */
259 return 0;
261 buf += metadata_size;
262 } while (!metadata_last);
264 return buf_size - (buf_end - buf);
267 static int decode_residuals(FLACContext *s, int channel, int pred_order)
269 int i, tmp, partition, method_type, rice_order;
270 int sample = 0, samples;
272 method_type = get_bits(&s->gb, 2);
273 if (method_type > 1) {
274 av_log(s->avctx, AV_LOG_ERROR, "illegal residual coding method %d\n",
275 method_type);
276 return -1;
279 rice_order = get_bits(&s->gb, 4);
281 samples= s->blocksize >> rice_order;
282 if (pred_order > samples) {
283 av_log(s->avctx, AV_LOG_ERROR, "invalid predictor order: %i > %i\n",
284 pred_order, samples);
285 return -1;
288 sample=
289 i= pred_order;
290 for (partition = 0; partition < (1 << rice_order); partition++) {
291 tmp = get_bits(&s->gb, method_type == 0 ? 4 : 5);
292 if (tmp == (method_type == 0 ? 15 : 31)) {
293 tmp = get_bits(&s->gb, 5);
294 for (; i < samples; i++, sample++)
295 s->decoded[channel][sample] = get_sbits_long(&s->gb, tmp);
296 } else {
297 for (; i < samples; i++, sample++) {
298 s->decoded[channel][sample] = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0);
301 i= 0;
304 return 0;
307 static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order)
309 const int blocksize = s->blocksize;
310 int32_t *decoded = s->decoded[channel];
311 int av_uninit(a), av_uninit(b), av_uninit(c), av_uninit(d), i;
313 /* warm up samples */
314 for (i = 0; i < pred_order; i++) {
315 decoded[i] = get_sbits_long(&s->gb, s->curr_bps);
318 if (decode_residuals(s, channel, pred_order) < 0)
319 return -1;
321 if (pred_order > 0)
322 a = decoded[pred_order-1];
323 if (pred_order > 1)
324 b = a - decoded[pred_order-2];
325 if (pred_order > 2)
326 c = b - decoded[pred_order-2] + decoded[pred_order-3];
327 if (pred_order > 3)
328 d = c - decoded[pred_order-2] + 2*decoded[pred_order-3] - decoded[pred_order-4];
330 switch (pred_order) {
331 case 0:
332 break;
333 case 1:
334 for (i = pred_order; i < blocksize; i++)
335 decoded[i] = a += decoded[i];
336 break;
337 case 2:
338 for (i = pred_order; i < blocksize; i++)
339 decoded[i] = a += b += decoded[i];
340 break;
341 case 3:
342 for (i = pred_order; i < blocksize; i++)
343 decoded[i] = a += b += c += decoded[i];
344 break;
345 case 4:
346 for (i = pred_order; i < blocksize; i++)
347 decoded[i] = a += b += c += d += decoded[i];
348 break;
349 default:
350 av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order);
351 return -1;
354 return 0;
357 static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order)
359 int i, j;
360 int coeff_prec, qlevel;
361 int coeffs[32];
362 int32_t *decoded = s->decoded[channel];
364 /* warm up samples */
365 for (i = 0; i < pred_order; i++) {
366 decoded[i] = get_sbits_long(&s->gb, s->curr_bps);
369 coeff_prec = get_bits(&s->gb, 4) + 1;
370 if (coeff_prec == 16) {
371 av_log(s->avctx, AV_LOG_ERROR, "invalid coeff precision\n");
372 return -1;
374 qlevel = get_sbits(&s->gb, 5);
375 if (qlevel < 0) {
376 av_log(s->avctx, AV_LOG_ERROR, "qlevel %d not supported, maybe buggy stream\n",
377 qlevel);
378 return -1;
381 for (i = 0; i < pred_order; i++) {
382 coeffs[i] = get_sbits(&s->gb, coeff_prec);
385 if (decode_residuals(s, channel, pred_order) < 0)
386 return -1;
388 if (s->bps > 16) {
389 int64_t sum;
390 for (i = pred_order; i < s->blocksize; i++) {
391 sum = 0;
392 for (j = 0; j < pred_order; j++)
393 sum += (int64_t)coeffs[j] * decoded[i-j-1];
394 decoded[i] += sum >> qlevel;
396 } else {
397 for (i = pred_order; i < s->blocksize-1; i += 2) {
398 int c;
399 int d = decoded[i-pred_order];
400 int s0 = 0, s1 = 0;
401 for (j = pred_order-1; j > 0; j--) {
402 c = coeffs[j];
403 s0 += c*d;
404 d = decoded[i-j];
405 s1 += c*d;
407 c = coeffs[0];
408 s0 += c*d;
409 d = decoded[i] += s0 >> qlevel;
410 s1 += c*d;
411 decoded[i+1] += s1 >> qlevel;
413 if (i < s->blocksize) {
414 int sum = 0;
415 for (j = 0; j < pred_order; j++)
416 sum += coeffs[j] * decoded[i-j-1];
417 decoded[i] += sum >> qlevel;
421 return 0;
424 static inline int decode_subframe(FLACContext *s, int channel)
426 int type, wasted = 0;
427 int i, tmp;
429 s->curr_bps = s->bps;
430 if (channel == 0) {
431 if (s->ch_mode == FLAC_CHMODE_RIGHT_SIDE)
432 s->curr_bps++;
433 } else {
434 if (s->ch_mode == FLAC_CHMODE_LEFT_SIDE || s->ch_mode == FLAC_CHMODE_MID_SIDE)
435 s->curr_bps++;
438 if (get_bits1(&s->gb)) {
439 av_log(s->avctx, AV_LOG_ERROR, "invalid subframe padding\n");
440 return -1;
442 type = get_bits(&s->gb, 6);
444 if (get_bits1(&s->gb)) {
445 wasted = 1;
446 while (!get_bits1(&s->gb))
447 wasted++;
448 s->curr_bps -= wasted;
450 if (s->curr_bps > 32) {
451 av_log_missing_feature(s->avctx, "decorrelated bit depth > 32", 0);
452 return -1;
455 //FIXME use av_log2 for types
456 if (type == 0) {
457 tmp = get_sbits_long(&s->gb, s->curr_bps);
458 for (i = 0; i < s->blocksize; i++)
459 s->decoded[channel][i] = tmp;
460 } else if (type == 1) {
461 for (i = 0; i < s->blocksize; i++)
462 s->decoded[channel][i] = get_sbits_long(&s->gb, s->curr_bps);
463 } else if ((type >= 8) && (type <= 12)) {
464 if (decode_subframe_fixed(s, channel, type & ~0x8) < 0)
465 return -1;
466 } else if (type >= 32) {
467 if (decode_subframe_lpc(s, channel, (type & ~0x20)+1) < 0)
468 return -1;
469 } else {
470 av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n");
471 return -1;
474 if (wasted) {
475 int i;
476 for (i = 0; i < s->blocksize; i++)
477 s->decoded[channel][i] <<= wasted;
480 return 0;
484 * Validate and decode a frame header.
485 * @param avctx AVCodecContext to use as av_log() context
486 * @param gb GetBitContext from which to read frame header
487 * @param[out] fi frame information
488 * @return non-zero on error, 0 if ok
490 static int decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
491 FLACFrameInfo *fi)
493 int bs_code, sr_code, bps_code;
495 /* frame sync code */
496 skip_bits(gb, 16);
498 /* block size and sample rate codes */
499 bs_code = get_bits(gb, 4);
500 sr_code = get_bits(gb, 4);
502 /* channels and decorrelation */
503 fi->ch_mode = get_bits(gb, 4);
504 if (fi->ch_mode < FLAC_MAX_CHANNELS) {
505 fi->channels = fi->ch_mode + 1;
506 fi->ch_mode = FLAC_CHMODE_INDEPENDENT;
507 } else if (fi->ch_mode <= FLAC_CHMODE_MID_SIDE) {
508 fi->channels = 2;
509 } else {
510 av_log(avctx, AV_LOG_ERROR, "invalid channel mode: %d\n", fi->ch_mode);
511 return -1;
514 /* bits per sample */
515 bps_code = get_bits(gb, 3);
516 if (bps_code == 3 || bps_code == 7) {
517 av_log(avctx, AV_LOG_ERROR, "invalid sample size code (%d)\n",
518 bps_code);
519 return -1;
521 fi->bps = sample_size_table[bps_code];
523 /* reserved bit */
524 if (get_bits1(gb)) {
525 av_log(avctx, AV_LOG_ERROR, "broken stream, invalid padding\n");
526 return -1;
529 /* sample or frame count */
530 if (get_utf8(gb) < 0) {
531 av_log(avctx, AV_LOG_ERROR, "utf8 fscked\n");
532 return -1;
535 /* blocksize */
536 if (bs_code == 0) {
537 av_log(avctx, AV_LOG_ERROR, "reserved blocksize code: 0\n");
538 return -1;
539 } else if (bs_code == 6) {
540 fi->blocksize = get_bits(gb, 8) + 1;
541 } else if (bs_code == 7) {
542 fi->blocksize = get_bits(gb, 16) + 1;
543 } else {
544 fi->blocksize = ff_flac_blocksize_table[bs_code];
547 /* sample rate */
548 if (sr_code < 12) {
549 fi->samplerate = ff_flac_sample_rate_table[sr_code];
550 } else if (sr_code == 12) {
551 fi->samplerate = get_bits(gb, 8) * 1000;
552 } else if (sr_code == 13) {
553 fi->samplerate = get_bits(gb, 16);
554 } else if (sr_code == 14) {
555 fi->samplerate = get_bits(gb, 16) * 10;
556 } else {
557 av_log(avctx, AV_LOG_ERROR, "illegal sample rate code %d\n",
558 sr_code);
559 return -1;
562 /* header CRC-8 check */
563 skip_bits(gb, 8);
564 if (av_crc(av_crc_get_table(AV_CRC_8_ATM), 0, gb->buffer,
565 get_bits_count(gb)/8)) {
566 av_log(avctx, AV_LOG_ERROR, "header crc mismatch\n");
567 return -1;
570 return 0;
573 static int decode_frame(FLACContext *s)
575 int i;
576 GetBitContext *gb = &s->gb;
577 FLACFrameInfo fi;
579 if (decode_frame_header(s->avctx, gb, &fi)) {
580 av_log(s->avctx, AV_LOG_ERROR, "invalid frame header\n");
581 return -1;
584 if (fi.channels != s->channels) {
585 av_log(s->avctx, AV_LOG_ERROR, "switching channel layout mid-stream "
586 "is not supported\n");
587 return -1;
589 s->ch_mode = fi.ch_mode;
591 if (fi.bps && fi.bps != s->bps) {
592 av_log(s->avctx, AV_LOG_ERROR, "switching bps mid-stream is not "
593 "supported\n");
594 return -1;
596 if (s->bps > 16) {
597 s->avctx->sample_fmt = SAMPLE_FMT_S32;
598 s->sample_shift = 32 - s->bps;
599 s->is32 = 1;
600 } else {
601 s->avctx->sample_fmt = SAMPLE_FMT_S16;
602 s->sample_shift = 16 - s->bps;
603 s->is32 = 0;
606 if (fi.blocksize > s->max_blocksize) {
607 av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", fi.blocksize,
608 s->max_blocksize);
609 return -1;
611 s->blocksize = fi.blocksize;
613 if (fi.samplerate == 0) {
614 fi.samplerate = s->samplerate;
615 } else if (fi.samplerate != s->samplerate) {
616 av_log(s->avctx, AV_LOG_WARNING, "sample rate changed from %d to %d\n",
617 s->samplerate, fi.samplerate);
619 s->samplerate = s->avctx->sample_rate = fi.samplerate;
621 // dump_headers(s->avctx, (FLACStreaminfo *)s);
623 /* subframes */
624 for (i = 0; i < s->channels; i++) {
625 if (decode_subframe(s, i) < 0)
626 return -1;
629 align_get_bits(gb);
631 /* frame footer */
632 skip_bits(gb, 16); /* data crc */
634 return 0;
637 static int flac_decode_frame(AVCodecContext *avctx,
638 void *data, int *data_size,
639 AVPacket *avpkt)
641 const uint8_t *buf = avpkt->data;
642 int buf_size = avpkt->size;
643 FLACContext *s = avctx->priv_data;
644 int i, j = 0, input_buf_size = 0, bytes_read = 0;
645 int16_t *samples_16 = data;
646 int32_t *samples_32 = data;
647 int alloc_data_size= *data_size;
648 int output_size;
650 *data_size=0;
652 if (s->max_framesize == 0) {
653 s->max_framesize= FFMAX(4, buf_size); // should hopefully be enough for the first header
654 s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
657 if (1 && s->max_framesize) { //FIXME truncated
658 if (s->bitstream_size < 4 || AV_RL32(s->bitstream) != MKTAG('f','L','a','C'))
659 buf_size= FFMIN(buf_size, s->max_framesize - FFMIN(s->bitstream_size, s->max_framesize));
660 input_buf_size= buf_size;
662 if (s->bitstream_size + buf_size < buf_size || s->bitstream_index + s->bitstream_size + buf_size < s->bitstream_index)
663 return -1;
665 if (s->allocated_bitstream_size < s->bitstream_size + buf_size)
666 s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->bitstream_size + buf_size);
668 if (s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size) {
669 memmove(s->bitstream, &s->bitstream[s->bitstream_index],
670 s->bitstream_size);
671 s->bitstream_index=0;
673 memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size],
674 buf, buf_size);
675 buf= &s->bitstream[s->bitstream_index];
676 buf_size += s->bitstream_size;
677 s->bitstream_size= buf_size;
679 if (buf_size < s->max_framesize && input_buf_size) {
680 return input_buf_size;
684 /* check that there is at least the smallest decodable amount of data.
685 this amount corresponds to the smallest valid FLAC frame possible.
686 FF F8 69 02 00 00 9A 00 00 34 46 */
687 if (buf_size < 11)
688 goto end;
690 /* check for inline header */
691 if (AV_RB32(buf) == MKBETAG('f','L','a','C')) {
692 if (!s->got_streaminfo && parse_streaminfo(s, buf, buf_size)) {
693 av_log(s->avctx, AV_LOG_ERROR, "invalid header\n");
694 return -1;
696 bytes_read = get_metadata_size(buf, buf_size);
697 goto end;
700 /* check for frame sync code and resync stream if necessary */
701 if ((AV_RB16(buf) & 0xFFFE) != 0xFFF8) {
702 const uint8_t *buf_end = buf + buf_size;
703 av_log(s->avctx, AV_LOG_ERROR, "FRAME HEADER not here\n");
704 while (buf+2 < buf_end && (AV_RB16(buf) & 0xFFFE) != 0xFFF8)
705 buf++;
706 bytes_read = buf_size - (buf_end - buf);
707 goto end; // we may not have enough bits left to decode a frame, so try next time
710 /* decode frame */
711 init_get_bits(&s->gb, buf, buf_size*8);
712 if (decode_frame(s) < 0) {
713 av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
714 s->bitstream_size=0;
715 s->bitstream_index=0;
716 return -1;
718 bytes_read = (get_bits_count(&s->gb)+7)/8;
720 /* check if allocated data size is large enough for output */
721 output_size = s->blocksize * s->channels * (s->is32 ? 4 : 2);
722 if (output_size > alloc_data_size) {
723 av_log(s->avctx, AV_LOG_ERROR, "output data size is larger than "
724 "allocated data size\n");
725 goto end;
727 *data_size = output_size;
729 #define DECORRELATE(left, right)\
730 assert(s->channels == 2);\
731 for (i = 0; i < s->blocksize; i++) {\
732 int a= s->decoded[0][i];\
733 int b= s->decoded[1][i];\
734 if (s->is32) {\
735 *samples_32++ = (left) << s->sample_shift;\
736 *samples_32++ = (right) << s->sample_shift;\
737 } else {\
738 *samples_16++ = (left) << s->sample_shift;\
739 *samples_16++ = (right) << s->sample_shift;\
742 break;
744 switch (s->ch_mode) {
745 case FLAC_CHMODE_INDEPENDENT:
746 for (j = 0; j < s->blocksize; j++) {
747 for (i = 0; i < s->channels; i++) {
748 if (s->is32)
749 *samples_32++ = s->decoded[i][j] << s->sample_shift;
750 else
751 *samples_16++ = s->decoded[i][j] << s->sample_shift;
754 break;
755 case FLAC_CHMODE_LEFT_SIDE:
756 DECORRELATE(a,a-b)
757 case FLAC_CHMODE_RIGHT_SIDE:
758 DECORRELATE(a+b,b)
759 case FLAC_CHMODE_MID_SIDE:
760 DECORRELATE( (a-=b>>1) + b, a)
763 end:
764 if (bytes_read > buf_size) {
765 av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size);
766 s->bitstream_size=0;
767 s->bitstream_index=0;
768 return -1;
771 if (s->bitstream_size) {
772 s->bitstream_index += bytes_read;
773 s->bitstream_size -= bytes_read;
774 return input_buf_size;
775 } else
776 return bytes_read;
779 static av_cold int flac_decode_close(AVCodecContext *avctx)
781 FLACContext *s = avctx->priv_data;
782 int i;
784 for (i = 0; i < s->channels; i++) {
785 av_freep(&s->decoded[i]);
787 av_freep(&s->bitstream);
789 return 0;
792 static void flac_flush(AVCodecContext *avctx)
794 FLACContext *s = avctx->priv_data;
796 s->bitstream_size=
797 s->bitstream_index= 0;
800 AVCodec flac_decoder = {
801 "flac",
802 CODEC_TYPE_AUDIO,
803 CODEC_ID_FLAC,
804 sizeof(FLACContext),
805 flac_decode_init,
806 NULL,
807 flac_decode_close,
808 flac_decode_frame,
809 CODEC_CAP_DELAY | CODEC_CAP_SUBFRAMES, /* FIXME: add a FLAC parser so that
810 we will not need to use either
811 of these capabilities */
812 .flush= flac_flush,
813 .long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),