2 * Real Audio 1.0 (14.4K)
3 * Copyright (c) 2003 the ffmpeg project
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
23 #include "bitstream.h"
26 #define NBLOCKS 4 /* number of segments within a block */
27 #define BLOCKSIZE 40 /* (quarter) block size in 16-bit words (80 bytes) */
28 #define HALFBLOCK 20 /* BLOCKSIZE/2 */
29 #define BUFFERSIZE 146 /* for do_output */
32 /* internal globals */
34 unsigned int old_energy; ///< previous frame energy
36 /* the swapped buffers */
37 unsigned int lpc_tables[2][10];
38 unsigned int *lpc_coef; ///< LPC coefficients
39 unsigned int *lpc_coef_old; ///< previous frame LPC coefficients
40 unsigned int lpc_refl_rms;
41 unsigned int lpc_refl_rms_old;
43 unsigned int buffer[5];
44 uint16_t adapt_cb[148]; ///< adaptive codebook
47 static int ra144_decode_init(AVCodecContext * avctx)
49 RA144Context *ractx = avctx->priv_data;
51 ractx->lpc_coef = ractx->lpc_tables[0];
52 ractx->lpc_coef_old = ractx->lpc_tables[1];
58 * Evaluate sqrt(x << 24). x must fit in 20 bits. This value is evaluated in an
59 * odd way to make the output identical to the binary decoder.
61 static int t_sqrt(unsigned int x)
69 return (ff_sqrt(x << 20) << s) << 2;
73 * Evaluate the LPC filter coefficients from the reflection coefficients.
74 * Does the inverse of the eval_refl() function.
76 static void eval_coefs(const int *refl, int *coefs)
83 for (x=0; x < 10; x++) {
87 b1[y] = ((refl[x] * b2[x-y-1]) >> 12) + b2[y];
89 FFSWAP(int *, b1, b2);
92 for (x=0; x < 10; x++)
97 static void rotate_block(const int16_t *source, int16_t *target, int offset)
100 source += BUFFERSIZE - offset;
102 while (i<BLOCKSIZE) {
103 target[i++] = source[k++];
110 /* inverse root mean square */
111 static int irms(const int16_t *data, int factor)
113 unsigned int i, sum = 0;
115 for (i=0; i < BLOCKSIZE; i++)
116 sum += data[i] * data[i];
119 return 0; /* OOPS - division by zero */
121 return (0x20000000 / (t_sqrt(sum) >> 8)) * factor;
124 /* multiply/add wavetable */
125 static void add_wav(int n, int skip_first, int *m, const int16_t *s1,
126 const int8_t *s2, const int8_t *s3, int16_t *dest)
132 for (i=!skip_first; i<3; i++)
133 v[i] = (wavtable1[n][i] * m[i]) >> (wavtable2[n][i] + 1);
135 for (i=0; i < BLOCKSIZE; i++)
136 dest[i] = ((*(s1++))*v[0] + (*(s2++))*v[1] + (*(s3++))*v[2]) >> 12;
140 static void lpc_filter(const int16_t *lpc_coefs, const int16_t *adapt_coef,
141 void *out, int *statbuf, int len)
147 memcpy(work, statbuf,20);
148 memcpy(work + 10, adapt_coef, len * 2);
150 for (i=0; i<len; i++) {
155 sum += lpc_coefs[9-x] * ptr[x];
159 new_val = ptr[10] - sum;
161 if (new_val < -32768 || new_val > 32767) {
162 memset(out, 0, len * 2);
163 memset(statbuf, 0, 20);
171 memcpy(out, work+10, len * 2);
172 memcpy(statbuf, work + 40, 20);
175 static unsigned int rescale_rms(int rms, int energy)
177 return (rms * energy) >> 10;
180 static unsigned int rms(const int *data)
183 unsigned int res = 0x10000;
186 for (x=0; x<10; x++) {
187 res = (((0x1000000 - (*data) * (*data)) >> 12) * res) >> 12;
192 while (res <= 0x3fff) {
206 /* do quarter-block output */
207 static void do_output_subblock(RA144Context *ractx,
208 const uint16_t *lpc_coefs, unsigned int gval,
209 int16_t *output_buffer, GetBitContext *gb)
211 uint16_t buffer_a[40];
213 int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none
214 int gain = get_bits(gb, 8);
215 int cb1_idx = get_bits(gb, 7);
216 int cb2_idx = get_bits(gb, 7);
220 cba_idx += HALFBLOCK - 1;
221 rotate_block(ractx->adapt_cb, buffer_a, cba_idx);
222 m[0] = irms(buffer_a, gval) >> 12;
227 m[1] = ((ftable1[cb1_idx] >> 4) * gval) >> 8;
228 m[2] = ((ftable2[cb2_idx] >> 4) * gval) >> 8;
230 memmove(ractx->adapt_cb, ractx->adapt_cb + BLOCKSIZE,
231 (BUFFERSIZE - BLOCKSIZE) * 2);
233 block = ractx->adapt_cb + BUFFERSIZE - BLOCKSIZE;
235 add_wav(gain, cba_idx, m, buffer_a, etable1[cb1_idx], etable2[cb2_idx],
238 lpc_filter(lpc_coefs, block, output_buffer, ractx->buffer, BLOCKSIZE);
241 static void int_to_int16(int16_t *decsp, const int *inp)
246 *(decsp++) = *(inp++);
250 * Evaluate the reflection coefficients from the filter coefficients.
251 * Does the inverse of the eval_coefs() function.
253 * @return 1 if one of the reflection coefficients is of magnitude greater than
256 static int eval_refl(const int16_t *coefs, int *refl, RA144Context *ractx)
266 for (i=0; i < 10; i++)
267 buffer2[i] = coefs[i];
269 u = refl[9] = bp2[9];
271 if (u + 0x1000 > 0x1fff) {
272 av_log(ractx, AV_LOG_ERROR, "Overflow. Broken sample?\n");
276 for (c=8; c >= 0; c--) {
283 b = 0x1000-((u * u) >> 12);
289 bp1[u] = ((bp2[u] - ((refl[c+1] * bp2[c-u]) >> 12)) * (0x1000000 / b)) >> 12;
291 refl[c] = u = bp1[c];
293 if ((u + 0x1000) > 0x1fff)
296 FFSWAP(int *, bp1, bp2);
301 static int interp(RA144Context *ractx, int16_t *decsp, int block_num,
302 int copynew, int energy)
305 int a = block_num + 1;
309 // Interpolate block coefficients from the this frame forth block and
310 // last frame forth block
312 decsp[x] = (a * ractx->lpc_coef[x] + b * ractx->lpc_coef_old[x])>> 2;
314 if (eval_refl(decsp, work, ractx)) {
315 // The interpolated coefficients are unstable, copy either new or old
318 int_to_int16(decsp, ractx->lpc_coef);
319 return rescale_rms(ractx->lpc_refl_rms, energy);
321 int_to_int16(decsp, ractx->lpc_coef_old);
322 return rescale_rms(ractx->lpc_refl_rms_old, energy);
325 return rescale_rms(rms(work), energy);
329 /* Uncompress one block (20 bytes -> 160*2 bytes) */
330 static int ra144_decode_frame(AVCodecContext * avctx,
331 void *vdata, int *data_size,
332 const uint8_t * buf, int buf_size)
334 static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
335 unsigned int refl_rms[4]; // RMS of the reflection coefficients
336 uint16_t block_coefs[4][30]; // LPC coefficients of each sub-block
337 unsigned int lpc_refl[10]; // LPC reflection coefficients of the frame
339 int16_t *data = vdata;
342 RA144Context *ractx = avctx->priv_data;
346 av_log(avctx, AV_LOG_ERROR,
347 "Frame too small (%d bytes). Truncated file?\n", buf_size);
350 init_get_bits(&gb, buf, 20 * 8);
353 // "<< 1"? Doesn't this make one value out of two of the table useless?
354 lpc_refl[i] = decodetable[i][get_bits(&gb, sizes[i]) << 1];
356 eval_coefs(lpc_refl, ractx->lpc_coef);
357 ractx->lpc_refl_rms = rms(lpc_refl);
359 energy = decodeval[get_bits(&gb, 5) << 1]; // Useless table entries?
361 refl_rms[0] = interp(ractx, block_coefs[0], 0, 0, ractx->old_energy);
362 refl_rms[1] = interp(ractx, block_coefs[1], 1, energy > ractx->old_energy,
363 t_sqrt(energy*ractx->old_energy) >> 12);
364 refl_rms[2] = interp(ractx, block_coefs[2], 2, 1, energy);
365 refl_rms[3] = rescale_rms(ractx->lpc_refl_rms, energy);
367 int_to_int16(block_coefs[3], ractx->lpc_coef);
370 for (c=0; c<4; c++) {
371 do_output_subblock(ractx, block_coefs[c], refl_rms[c], data, &gb);
373 for (i=0; i<BLOCKSIZE; i++) {
374 *data = av_clip_int16(*data << 2);
379 ractx->old_energy = energy;
380 ractx->lpc_refl_rms_old = ractx->lpc_refl_rms;
382 FFSWAP(unsigned int *, ractx->lpc_coef_old, ractx->lpc_coef);
389 AVCodec ra_144_decoder =
394 sizeof(RA144Context),
399 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),