2 * RealAudio Lossless decoder
4 * Copyright (c) 2012 Konstantin Shishkov
6 * This file is part of Libav.
8 * Libav 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.
13 * Libav 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.
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with Libav; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
25 * This is a decoder for Real Audio Lossless format.
26 * Dedicated to the mastermind behind it, Ralph Wiggum.
33 #include "libavutil/audioconvert.h"
37 #define FILTER_RAW 642
39 typedef struct VLCSet {
43 VLC filter_coeffs[10][11];
48 #define RALF_MAX_PKT_SIZE 8192
50 typedef struct RALFContext {
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
63 int bias[2]; ///< a constant value added to channel data after filtering
65 int num_blocks; ///< number of blocks inside the frame
67 int block_size[1 << 12]; ///< size of the blocks
68 int block_pts[1 << 12]; ///< block start time (in milliseconds)
74 #define MAX_ELEMS 644 // no RALF table uses more than that
76 static int init_ralf_vlc(VLC *vlc, const uint8_t *data, int elems)
78 uint8_t lens[MAX_ELEMS];
79 uint16_t codes[MAX_ELEMS];
80 int counts[17], prefixes[18];
85 for (i = 0; i <= 16; i++)
87 for (i = 0; i < elems; i++) {
88 cur_len = (nb ? *data & 0xF : *data >> 4) + 1;
90 max_bits = FFMAX(max_bits, cur_len);
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]]++;
102 return ff_init_vlc_sparse(vlc, FFMIN(max_bits, 9), elems,
103 lens, 1, 1, codes, 2, 2, NULL, 0, 0, 0);
106 static av_cold int decode_close(AVCodecContext *avctx)
108 RALFContext *ctx = avctx->priv_data;
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]);
127 static av_cold int decode_init(AVCodecContext *avctx)
129 RALFContext *ctx = avctx->priv_data;
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;
138 ctx->version = AV_RB16(avctx->extradata + 4);
139 if (ctx->version != 0x103) {
140 av_log_ask_for_sample(avctx, "unknown version %X\n", ctx->version);
141 return AVERROR_PATCHWELCOME;
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;
152 avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
153 avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO
156 avcodec_get_frame_defaults(&ctx->frame);
157 avctx->coded_frame = &ctx->frame;
159 ctx->max_frame_size = AV_RB32(avctx->extradata + 16);
160 if (ctx->max_frame_size > (1 << 20) || !ctx->max_frame_size) {
161 av_log(avctx, AV_LOG_ERROR, "invalid frame size %d\n",
162 ctx->max_frame_size);
164 ctx->max_frame_size = FFMAX(ctx->max_frame_size, avctx->sample_rate);
166 for (i = 0; i < 3; i++) {
167 ret = init_ralf_vlc(&ctx->sets[i].filter_params, filter_param_def[i],
168 FILTERPARAM_ELEMENTS);
173 ret = init_ralf_vlc(&ctx->sets[i].bias, bias_def[i], BIAS_ELEMENTS);
178 ret = init_ralf_vlc(&ctx->sets[i].coding_mode, coding_mode_def[i],
179 CODING_MODE_ELEMENTS);
184 for (j = 0; j < 10; j++) {
185 for (k = 0; k < 11; k++) {
186 ret = init_ralf_vlc(&ctx->sets[i].filter_coeffs[j][k],
187 filter_coeffs_def[i][j][k],
188 FILTER_COEFFS_ELEMENTS);
195 for (j = 0; j < 15; j++) {
196 ret = init_ralf_vlc(&ctx->sets[i].short_codes[j],
197 short_codes_def[i][j], SHORT_CODES_ELEMENTS);
203 for (j = 0; j < 125; j++) {
204 ret = init_ralf_vlc(&ctx->sets[i].long_codes[j],
205 long_codes_def[i][j], LONG_CODES_ELEMENTS);
216 static inline int extend_code(GetBitContext *gb, int val, int range, int bits)
219 val = -range - get_ue_golomb(gb);
220 } else if (val == range * 2) {
221 val = range + get_ue_golomb(gb);
226 val = (val << bits) | get_bits(gb, bits);
230 static int decode_channel(RALFContext *ctx, GetBitContext *gb, int ch,
231 int length, int mode, int bits)
235 VLCSet *set = ctx->sets + mode;
236 VLC *code_vlc; int range, range2, add_bits;
237 int *dst = ctx->channel_data[ch];
239 ctx->filter_params = get_vlc2(gb, set->filter_params.table, 9, 2);
240 ctx->filter_bits = (ctx->filter_params - 2) >> 6;
241 ctx->filter_length = ctx->filter_params - (ctx->filter_bits << 6) - 1;
243 if (ctx->filter_params == FILTER_RAW) {
244 for (i = 0; i < length; i++)
245 dst[i] = get_bits(gb, bits);
250 ctx->bias[ch] = get_vlc2(gb, set->bias.table, 9, 2);
251 ctx->bias[ch] = extend_code(gb, ctx->bias[ch], 127, 4);
253 if (ctx->filter_params == FILTER_NONE) {
254 memset(dst, 0, sizeof(*dst) * length);
258 if (ctx->filter_params > 1) {
259 int cmode = 0, coeff = 0;
260 VLC *vlc = set->filter_coeffs[ctx->filter_bits] + 5;
262 add_bits = ctx->filter_bits;
264 for (i = 0; i < ctx->filter_length; i++) {
265 t = get_vlc2(gb, vlc[cmode].table, vlc[cmode].bits, 2);
266 t = extend_code(gb, t, 21, add_bits);
268 coeff -= 12 << add_bits;
270 ctx->filter[i] = coeff;
272 cmode = coeff >> add_bits;
274 cmode = -1 - av_log2(-cmode);
277 } else if (cmode > 0) {
278 cmode = 1 + av_log2(cmode);
285 code_params = get_vlc2(gb, set->coding_mode.table, set->coding_mode.bits, 2);
286 if (code_params >= 15) {
287 add_bits = av_clip((code_params / 5 - 3) / 2, 0, 10);
288 if (add_bits > 9 && (code_params % 5) != 2)
292 code_vlc = set->long_codes + code_params - 15;
297 code_vlc = set->short_codes + code_params;
300 for (i = 0; i < length; i += 2) {
303 t = get_vlc2(gb, code_vlc->table, code_vlc->bits, 2);
306 dst[i] = extend_code(gb, code1, range, 0) << add_bits;
307 dst[i + 1] = extend_code(gb, code2, range, 0) << add_bits;
309 dst[i] |= get_bits(gb, add_bits);
310 dst[i + 1] |= get_bits(gb, add_bits);
317 static void apply_lpc(RALFContext *ctx, int ch, int length, int bits)
320 int *audio = ctx->channel_data[ch];
321 int bias = 1 << (ctx->filter_bits - 1);
322 int max_clip = (1 << bits) - 1, min_clip = -max_clip - 1;
324 for (i = 1; i < length; i++) {
325 int flen = FFMIN(ctx->filter_length, i);
328 for (j = 0; j < flen; j++)
329 acc += ctx->filter[j] * audio[i - j - 1];
331 acc = (acc + bias - 1) >> ctx->filter_bits;
332 acc = FFMAX(acc, min_clip);
334 acc = (acc + bias) >> ctx->filter_bits;
335 acc = FFMIN(acc, max_clip);
341 static int decode_block(AVCodecContext *avctx, GetBitContext *gb,
342 int16_t *dst0, int16_t *dst1)
344 RALFContext *ctx = avctx->priv_data;
346 int dmode, mode[2], bits[2];
350 len = 12 - get_unary(gb, 0, 6);
352 if (len <= 7) len ^= 1; // codes for length = 6 and 7 are swapped
355 if (ctx->sample_offset + len > ctx->max_frame_size) {
356 av_log(avctx, AV_LOG_ERROR,
357 "Decoder's stomach is crying, it ate too many samples\n");
358 return AVERROR_INVALIDDATA;
361 if (avctx->channels > 1)
362 dmode = get_bits(gb, 2) + 1;
366 mode[0] = (dmode == 4) ? 1 : 0;
367 mode[1] = (dmode >= 2) ? 2 : 0;
369 bits[1] = (mode[1] == 2) ? 17 : 16;
371 for (ch = 0; ch < avctx->channels; ch++) {
372 if ((ret = decode_channel(ctx, gb, ch, len, mode[ch], bits[ch])) < 0)
374 if (ctx->filter_params > 1 && ctx->filter_params != FILTER_RAW) {
375 ctx->filter_bits += 3;
376 apply_lpc(ctx, ch, len, bits[ch]);
378 if (get_bits_left(gb) < 0)
379 return AVERROR_INVALIDDATA;
381 ch0 = ctx->channel_data[0];
382 ch1 = ctx->channel_data[1];
385 for (i = 0; i < len; i++)
386 dst0[i] = ch0[i] + ctx->bias[0];
389 for (i = 0; i < len; i++) {
390 dst0[i] = ch0[i] + ctx->bias[0];
391 dst1[i] = ch1[i] + ctx->bias[1];
395 for (i = 0; i < len; i++) {
396 ch0[i] += ctx->bias[0];
398 dst1[i] = ch0[i] - (ch1[i] + ctx->bias[1]);
402 for (i = 0; i < len; i++) {
403 t = ch0[i] + ctx->bias[0];
404 t2 = ch1[i] + ctx->bias[1];
410 for (i = 0; i < len; i++) {
411 t = ch1[i] + ctx->bias[1];
412 t2 = ((ch0[i] + ctx->bias[0]) << 1) | (t & 1);
413 dst0[i] = (t2 + t) / 2;
414 dst1[i] = (t2 - t) / 2;
419 ctx->sample_offset += len;
424 static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
427 RALFContext *ctx = avctx->priv_data;
432 int table_size, table_bytes, i;
433 const uint8_t *src, *block_pointer;
439 table_bytes = (AV_RB16(avpkt->data) + 7) >> 3;
440 if (table_bytes + 3 > avpkt->size || avpkt->size > RALF_MAX_PKT_SIZE) {
441 av_log(avctx, AV_LOG_ERROR, "Wrong packet's breath smells of wrong data!\n");
442 return AVERROR_INVALIDDATA;
444 if (memcmp(ctx->pkt, avpkt->data, 2 + table_bytes)) {
445 av_log(avctx, AV_LOG_ERROR, "Wrong packet tails are wrong!\n");
446 return AVERROR_INVALIDDATA;
450 src_size = RALF_MAX_PKT_SIZE + avpkt->size;
451 memcpy(ctx->pkt + RALF_MAX_PKT_SIZE, avpkt->data + 2 + table_bytes,
452 avpkt->size - 2 - table_bytes);
454 if (avpkt->size == RALF_MAX_PKT_SIZE) {
455 memcpy(ctx->pkt, avpkt->data, avpkt->size);
462 src_size = avpkt->size;
465 ctx->frame.nb_samples = ctx->max_frame_size;
466 if ((ret = avctx->get_buffer(avctx, &ctx->frame)) < 0) {
467 av_log(avctx, AV_LOG_ERROR, "Me fail get_buffer()? That's unpossible!\n");
470 samples0 = (int16_t *)ctx->frame.data[0];
471 samples1 = (int16_t *)ctx->frame.data[1];
474 av_log(avctx, AV_LOG_ERROR, "too short packets are too short!\n");
475 return AVERROR_INVALIDDATA;
477 table_size = AV_RB16(src);
478 table_bytes = (table_size + 7) >> 3;
479 if (src_size < table_bytes + 3) {
480 av_log(avctx, AV_LOG_ERROR, "short packets are short!\n");
481 return AVERROR_INVALIDDATA;
483 init_get_bits(&gb, src + 2, table_size);
485 while (get_bits_left(&gb) > 0) {
486 ctx->block_size[ctx->num_blocks] = get_bits(&gb, 15);
487 if (get_bits1(&gb)) {
488 ctx->block_pts[ctx->num_blocks] = get_bits(&gb, 9);
490 ctx->block_pts[ctx->num_blocks] = 0;
495 block_pointer = src + table_bytes + 2;
496 bytes_left = src_size - table_bytes - 2;
497 ctx->sample_offset = 0;
498 for (i = 0; i < ctx->num_blocks; i++) {
499 if (bytes_left < ctx->block_size[i]) {
500 av_log(avctx, AV_LOG_ERROR, "I'm pedaling backwards\n");
503 init_get_bits(&gb, block_pointer, ctx->block_size[i] * 8);
504 if (decode_block(avctx, &gb, samples0 + ctx->sample_offset,
505 samples1 + ctx->sample_offset) < 0) {
506 av_log(avctx, AV_LOG_ERROR, "Sir, I got carsick in your office. Not decoding the rest of packet.\n");
509 block_pointer += ctx->block_size[i];
510 bytes_left -= ctx->block_size[i];
513 ctx->frame.nb_samples = ctx->sample_offset;
514 *got_frame_ptr = ctx->sample_offset > 0;
515 *(AVFrame*)data = ctx->frame;
520 static void decode_flush(AVCodecContext *avctx)
522 RALFContext *ctx = avctx->priv_data;
528 AVCodec ff_ralf_decoder = {
530 .type = AVMEDIA_TYPE_AUDIO,
531 .id = AV_CODEC_ID_RALF,
532 .priv_data_size = sizeof(RALFContext),
534 .close = decode_close,
535 .decode = decode_frame,
536 .flush = decode_flush,
537 .capabilities = CODEC_CAP_DR1,
538 .long_name = NULL_IF_CONFIG_SMALL("RealAudio Lossless"),
539 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
540 AV_SAMPLE_FMT_NONE },