2 * Real Audio 1.0 (14.4K)
4 * Copyright (c) 2008 Vitor Sessak
5 * Copyright (c) 2003 Nick Kurshev
6 * Based on public domain decoder at http://www.honeypot.net/audio
8 * This file is part of FFmpeg.
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.
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.
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
28 #include "celp_filters.h"
30 #define NBLOCKS 4 ///< number of subblocks within a block
31 #define BLOCKSIZE 40 ///< subblock size in 16-bit words
32 #define BUFFERSIZE 146 ///< the size of the adaptive codebook
36 AVCodecContext *avctx;
38 unsigned int old_energy; ///< previous frame energy
40 unsigned int lpc_tables[2][10];
42 /** LPC coefficients: lpc_coef[0] is the coefficients of the current frame
43 * and lpc_coef[1] of the previous one. */
44 unsigned int *lpc_coef[2];
46 unsigned int lpc_refl_rms[2];
48 /** The current subblock padded by the last 10 values of the previous one. */
49 int16_t curr_sblock[50];
51 /** Adaptive codebook, its size is two units bigger to avoid a
53 uint16_t adapt_cb[146+2];
56 static av_cold int ra144_decode_init(AVCodecContext * avctx)
58 RA144Context *ractx = avctx->priv_data;
62 ractx->lpc_coef[0] = ractx->lpc_tables[0];
63 ractx->lpc_coef[1] = ractx->lpc_tables[1];
65 avctx->sample_fmt = SAMPLE_FMT_S16;
70 * Evaluate sqrt(x << 24). x must fit in 20 bits. This value is evaluated in an
71 * odd way to make the output identical to the binary decoder.
73 static int t_sqrt(unsigned int x)
81 return ff_sqrt(x << 20) << s;
85 * Evaluate the LPC filter coefficients from the reflection coefficients.
86 * Does the inverse of the eval_refl() function.
88 static void eval_coefs(int *coefs, const int *refl)
95 for (i=0; i < 10; i++) {
99 b1[j] = ((refl[i] * b2[i-j-1]) >> 12) + b2[j];
101 FFSWAP(int *, b1, b2);
104 for (i=0; i < 10; i++)
109 * Copy the last offset values of *source to *target. If those values are not
110 * enough to fill the target buffer, fill it with another copy of those values.
112 static void copy_and_dup(int16_t *target, const int16_t *source, int offset)
114 source += BUFFERSIZE - offset;
116 memcpy(target, source, FFMIN(BLOCKSIZE, offset)*sizeof(*target));
117 if (offset < BLOCKSIZE)
118 memcpy(target + offset, source, (BLOCKSIZE - offset)*sizeof(*target));
121 /** inverse root mean square */
122 static int irms(const int16_t *data)
124 unsigned int i, sum = 0;
126 for (i=0; i < BLOCKSIZE; i++)
127 sum += data[i] * data[i];
130 return 0; /* OOPS - division by zero */
132 return 0x20000000 / (t_sqrt(sum) >> 8);
135 static void add_wav(int16_t *dest, int n, int skip_first, int *m,
136 const int16_t *s1, const int8_t *s2, const int8_t *s3)
142 for (i=!skip_first; i<3; i++)
143 v[i] = (gain_val_tab[n][i] * m[i]) >> gain_exp_tab[n];
146 for (i=0; i < BLOCKSIZE; i++)
147 dest[i] = (s1[i]*v[0] + s2[i]*v[1] + s3[i]*v[2]) >> 12;
149 for (i=0; i < BLOCKSIZE; i++)
150 dest[i] = ( s2[i]*v[1] + s3[i]*v[2]) >> 12;
154 static unsigned int rescale_rms(unsigned int rms, unsigned int energy)
156 return (rms * energy) >> 10;
159 static unsigned int rms(const int *data)
162 unsigned int res = 0x10000;
165 for (i=0; i < 10; i++) {
166 res = (((0x1000000 - data[i]*data[i]) >> 12) * res) >> 12;
171 while (res <= 0x3fff) {
177 return t_sqrt(res) >> b;
180 static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs,
181 int gval, GetBitContext *gb)
183 uint16_t buffer_a[40];
185 int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none
186 int gain = get_bits(gb, 8);
187 int cb1_idx = get_bits(gb, 7);
188 int cb2_idx = get_bits(gb, 7);
192 cba_idx += BLOCKSIZE/2 - 1;
193 copy_and_dup(buffer_a, ractx->adapt_cb, cba_idx);
194 m[0] = (irms(buffer_a) * gval) >> 12;
199 m[1] = (cb1_base[cb1_idx] * gval) >> 8;
200 m[2] = (cb2_base[cb2_idx] * gval) >> 8;
202 memmove(ractx->adapt_cb, ractx->adapt_cb + BLOCKSIZE,
203 (BUFFERSIZE - BLOCKSIZE) * sizeof(*ractx->adapt_cb));
205 block = ractx->adapt_cb + BUFFERSIZE - BLOCKSIZE;
207 add_wav(block, gain, cba_idx, m, cba_idx? buffer_a: NULL,
208 cb1_vects[cb1_idx], cb2_vects[cb2_idx]);
210 memcpy(ractx->curr_sblock, ractx->curr_sblock + 40,
211 10*sizeof(*ractx->curr_sblock));
213 if (ff_celp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs,
214 block, BLOCKSIZE, 10, 1, 0xfff))
215 memset(ractx->curr_sblock, 0, 50*sizeof(*ractx->curr_sblock));
218 static void int_to_int16(int16_t *out, const int *inp)
222 for (i=0; i < 10; i++)
227 * Evaluate the reflection coefficients from the filter coefficients.
228 * Does the inverse of the eval_coefs() function.
230 * @return 1 if one of the reflection coefficients is greater than
233 static int eval_refl(int *refl, const int16_t *coefs, AVCodecContext *avctx)
241 for (i=0; i < 10; i++)
242 buffer2[i] = coefs[i];
246 if ((unsigned) bp2[9] + 0x1000 > 0x1fff) {
247 av_log(avctx, AV_LOG_ERROR, "Overflow. Broken sample?\n");
251 for (i=8; i >= 0; i--) {
252 b = 0x1000-((bp2[i+1] * bp2[i+1]) >> 12);
257 for (j=0; j <= i; j++)
258 bp1[j] = ((bp2[j] - ((refl[i+1] * bp2[i-j]) >> 12)) * (0x1000000 / b)) >> 12;
260 if ((unsigned) bp1[i] + 0x1000 > 0x1fff)
265 FFSWAP(int *, bp1, bp2);
270 static int interp(RA144Context *ractx, int16_t *out, int a,
271 int copyold, int energy)
277 // Interpolate block coefficients from the this frame's forth block and
278 // last frame's forth block.
280 out[i] = (a * ractx->lpc_coef[0][i] + b * ractx->lpc_coef[1][i])>> 2;
282 if (eval_refl(work, out, ractx->avctx)) {
283 // The interpolated coefficients are unstable, copy either new or old
285 int_to_int16(out, ractx->lpc_coef[copyold]);
286 return rescale_rms(ractx->lpc_refl_rms[copyold], energy);
288 return rescale_rms(rms(work), energy);
292 /** Uncompress one block (20 bytes -> 160*2 bytes). */
293 static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
294 int *data_size, AVPacket *avpkt)
296 const uint8_t *buf = avpkt->data;
297 int buf_size = avpkt->size;
298 static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
299 unsigned int refl_rms[4]; // RMS of the reflection coefficients
300 uint16_t block_coefs[4][10]; // LPC coefficients of each sub-block
301 unsigned int lpc_refl[10]; // LPC reflection coefficients of the frame
303 int16_t *data = vdata;
306 RA144Context *ractx = avctx->priv_data;
309 if (*data_size < 2*160)
313 av_log(avctx, AV_LOG_ERROR,
314 "Frame too small (%d bytes). Truncated file?\n", buf_size);
318 init_get_bits(&gb, buf, 20 * 8);
321 lpc_refl[i] = lpc_refl_cb[i][get_bits(&gb, sizes[i])];
323 eval_coefs(ractx->lpc_coef[0], lpc_refl);
324 ractx->lpc_refl_rms[0] = rms(lpc_refl);
326 energy = energy_tab[get_bits(&gb, 5)];
328 refl_rms[0] = interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
329 refl_rms[1] = interp(ractx, block_coefs[1], 2, energy <= ractx->old_energy,
330 t_sqrt(energy*ractx->old_energy) >> 12);
331 refl_rms[2] = interp(ractx, block_coefs[2], 3, 0, energy);
332 refl_rms[3] = rescale_rms(ractx->lpc_refl_rms[0], energy);
334 int_to_int16(block_coefs[3], ractx->lpc_coef[0]);
336 for (i=0; i < 4; i++) {
337 do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb);
339 for (j=0; j < BLOCKSIZE; j++)
340 *data++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2);
343 ractx->old_energy = energy;
344 ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
346 FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
352 AVCodec ra_144_decoder =
357 sizeof(RA144Context),
362 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),