mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-26 19:01:44 +02:00
381 lines
9.5 KiB
C
381 lines
9.5 KiB
C
/*
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* Real Audio 1.0 (14.4K)
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* Copyright (c) 2003 the ffmpeg project
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avcodec.h"
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#include "bitstream.h"
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#include "ra144.h"
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#define NBLOCKS 4 /* number of segments within a block */
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#define BLOCKSIZE 40 /* (quarter) block size in 16-bit words (80 bytes) */
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#define HALFBLOCK 20 /* BLOCKSIZE/2 */
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#define BUFFERSIZE 146 /* for do_output */
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/* internal globals */
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typedef struct {
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unsigned int old_energy;
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/* the swapped buffers */
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unsigned int lpc_tables[4][10];
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unsigned int *lpc_refl; ///< LPC reflection coefficients
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unsigned int *lpc_coef; ///< LPC coefficients
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unsigned int *lpc_refl_old; ///< previous frame LPC reflection coefs
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unsigned int *lpc_coef_old; ///< previous frame LPC coefficients
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unsigned int buffer[5];
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uint16_t adapt_cb[148]; ///< adaptive codebook
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} RA144Context;
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static int ra144_decode_init(AVCodecContext * avctx)
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{
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RA144Context *ractx = avctx->priv_data;
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ractx->lpc_refl = ractx->lpc_tables[0];
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ractx->lpc_coef = ractx->lpc_tables[1];
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ractx->lpc_refl_old = ractx->lpc_tables[2];
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ractx->lpc_coef_old = ractx->lpc_tables[3];
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return 0;
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}
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/**
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* Evaluate sqrt(x << 24). x must fit in 20 bits. This value is evaluated in an
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* odd way to make the output identical to the binary decoder.
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*/
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static int t_sqrt(unsigned int x)
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{
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int s = 0;
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while (x > 0xfff) {
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s++;
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x = x >> 2;
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}
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return (ff_sqrt(x << 20) << s) << 2;
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}
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/* do 'voice' */
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static void do_voice(const int *a1, int *a2)
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{
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int buffer[10];
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int *b1 = buffer;
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int *b2 = a2;
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int x, y;
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for (x=0; x < 10; x++) {
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b1[x] = a1[x] << 4;
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for (y=0; y < x; y++)
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b1[y] = ((a1[x] * b2[x-y-1]) >> 12) + b2[y];
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FFSWAP(int *, b1, b2);
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}
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for (x=0; x < 10; x++)
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a2[x] >>= 4;
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}
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/* rotate block */
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static void rotate_block(const int16_t *source, int16_t *target, int offset)
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{
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int i=0, k=0;
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source += BUFFERSIZE - offset;
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while (i<BLOCKSIZE) {
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target[i++] = source[k++];
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if (k == offset)
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k = 0;
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}
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}
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/* inverse root mean square */
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static int irms(const int16_t *data, int factor)
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{
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unsigned int i, sum = 0;
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for (i=0; i < BLOCKSIZE; i++)
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sum += data[i] * data[i];
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if (sum == 0)
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return 0; /* OOPS - division by zero */
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return (0x20000000 / (t_sqrt(sum) >> 8)) * factor;
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}
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/* multiply/add wavetable */
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static void add_wav(int n, int skip_first, int *m, const int16_t *s1,
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const int8_t *s2, const int8_t *s3, int16_t *dest)
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{
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int i;
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int v[3];
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v[0] = 0;
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for (i=!skip_first; i<3; i++)
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v[i] = (wavtable1[n][i] * m[i]) >> (wavtable2[n][i] + 1);
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for (i=0; i < BLOCKSIZE; i++)
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dest[i] = ((*(s1++))*v[0] + (*(s2++))*v[1] + (*(s3++))*v[2]) >> 12;
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}
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static void final(const int16_t *i1, const int16_t *i2,
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void *out, int *statbuf, int len)
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{
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int x, i;
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uint16_t work[50];
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int16_t *ptr = work;
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memcpy(work, statbuf,20);
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memcpy(work + 10, i2, len * 2);
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for (i=0; i<len; i++) {
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int sum = 0;
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int new_val;
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for(x=0; x<10; x++)
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sum += i1[9-x] * ptr[x];
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sum >>= 12;
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new_val = ptr[10] - sum;
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if (new_val < -32768 || new_val > 32767) {
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memset(out, 0, len * 2);
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memset(statbuf, 0, 20);
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return;
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}
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ptr[10] = new_val;
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ptr++;
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}
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memcpy(out, work+10, len * 2);
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memcpy(statbuf, work + 40, 20);
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}
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static unsigned int rms(const int *data, int f)
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{
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int x;
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unsigned int res = 0x10000;
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int b = 0;
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for (x=0; x<10; x++) {
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res = (((0x1000000 - (*data) * (*data)) >> 12) * res) >> 12;
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if (res == 0)
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return 0;
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if (res > 0x10000)
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return 0; /* We're screwed, might as well go out with a bang. :P */
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while (res <= 0x3fff) {
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b++;
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res <<= 2;
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}
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data++;
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}
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if (res > 0)
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res = t_sqrt(res);
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res >>= (b + 10);
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res = (res * f) >> 10;
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return res;
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}
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/* do quarter-block output */
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static void do_output_subblock(RA144Context *ractx,
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const uint16_t *gsp, unsigned int gval,
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int16_t *output_buffer, GetBitContext *gb)
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{
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uint16_t buffer_a[40];
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uint16_t *block;
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int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none
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int gain = get_bits(gb, 8);
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int cb1_idx = get_bits(gb, 7);
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int cb2_idx = get_bits(gb, 7);
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int m[3];
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if (cba_idx) {
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cba_idx += HALFBLOCK - 1;
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rotate_block(ractx->adapt_cb, buffer_a, cba_idx);
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m[0] = irms(buffer_a, gval) >> 12;
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} else {
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m[0] = 0;
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}
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m[1] = ((ftable1[cb1_idx] >> 4) * gval) >> 8;
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m[2] = ((ftable2[cb2_idx] >> 4) * gval) >> 8;
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memmove(ractx->adapt_cb, ractx->adapt_cb + BLOCKSIZE,
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(BUFFERSIZE - BLOCKSIZE) * 2);
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block = ractx->adapt_cb + BUFFERSIZE - BLOCKSIZE;
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add_wav(gain, cba_idx, m, buffer_a, etable1[cb1_idx], etable2[cb2_idx],
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block);
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final(gsp, block, output_buffer, ractx->buffer, BLOCKSIZE);
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}
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static int dec1(int16_t *decsp, const int *data, const int *inp, int f)
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{
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int i;
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for (i=0; i<30; i++)
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*(decsp++) = *(inp++);
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return rms(data, f);
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}
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static int eq(const int16_t *in, int *target)
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{
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int retval = 0;
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int b, c, i;
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unsigned int u;
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int buffer1[10];
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int buffer2[10];
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int *bp1 = buffer1;
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int *bp2 = buffer2;
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for (i=0; i < 10; i++)
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buffer2[i] = in[i];
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u = target[9] = bp2[9];
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if (u + 0x1000 > 0x1fff)
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return 0; /* We're screwed, might as well go out with a bang. :P */
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for (c=8; c >= 0; c--) {
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if (u == 0x1000)
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u++;
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if (u == 0xfffff000)
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u--;
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b = 0x1000-((u * u) >> 12);
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if (b == 0)
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b++;
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for (u=0; u<=c; u++)
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bp1[u] = ((bp2[u] - ((target[c+1] * bp2[c-u]) >> 12)) * (0x1000000 / b)) >> 12;
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target[c] = u = bp1[c];
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if ((u + 0x1000) > 0x1fff)
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retval = 1;
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FFSWAP(int *, bp1, bp2);
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}
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return retval;
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}
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static int dec2(int16_t *decsp, const int *data, const int *inp,
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int f, const int *inp2, int a)
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{
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int work[10];
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int b = NBLOCKS - a;
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int x;
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for (x=0; x<30; x++)
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decsp[x] = (a * inp[x] + b * inp2[x]) >> 2;
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if (eq(decsp, work))
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return dec1(decsp, data, inp, f);
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else
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return rms(work, f);
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}
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/* Uncompress one block (20 bytes -> 160*2 bytes) */
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static int ra144_decode_frame(AVCodecContext * avctx,
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void *vdata, int *data_size,
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const uint8_t * buf, int buf_size)
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{
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static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
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unsigned int gbuf1[4];
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uint16_t gbuf2[4][30];
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unsigned int a, c;
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int i;
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int16_t *data = vdata;
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unsigned int energy;
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RA144Context *ractx = avctx->priv_data;
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GetBitContext gb;
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if(buf_size < 20) {
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av_log(avctx, AV_LOG_ERROR,
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"Frame too small (%d bytes). Truncated file?\n", buf_size);
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return buf_size;
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}
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init_get_bits(&gb, buf, 20 * 8);
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for (i=0; i<10; i++)
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// "<< 1"? Doesn't this make one value out of two of the table useless?
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ractx->lpc_refl[i] = decodetable[i][get_bits(&gb, sizes[i]) << 1];
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do_voice(ractx->lpc_refl, ractx->lpc_coef);
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energy = decodeval[get_bits(&gb, 5) << 1]; // Useless table entries?
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a = t_sqrt(energy*ractx->old_energy) >> 12;
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gbuf1[0] = dec2(gbuf2[0], ractx->lpc_refl_old, ractx->lpc_coef_old, ractx->old_energy, ractx->lpc_coef, 3);
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if (ractx->old_energy < energy) {
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gbuf1[1] = dec2(gbuf2[1], ractx->lpc_refl, ractx->lpc_coef, a, ractx->lpc_coef_old, 2);
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} else {
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gbuf1[1] = dec2(gbuf2[1], ractx->lpc_refl_old, ractx->lpc_coef_old, a, ractx->lpc_coef, 2);
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}
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gbuf1[2] = dec2(gbuf2[2], ractx->lpc_refl, ractx->lpc_coef, energy, ractx->lpc_coef_old, 3);
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gbuf1[3] = dec1(gbuf2[3], ractx->lpc_refl, ractx->lpc_coef, energy);
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/* do output */
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for (c=0; c<4; c++) {
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do_output_subblock(ractx, gbuf2[c], gbuf1[c], data, &gb);
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for (i=0; i<BLOCKSIZE; i++) {
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*data = av_clip_int16(*data << 2);
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data++;
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}
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}
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ractx->old_energy = energy;
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FFSWAP(unsigned int *, ractx->lpc_refl_old, ractx->lpc_refl);
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FFSWAP(unsigned int *, ractx->lpc_coef_old, ractx->lpc_coef);
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*data_size = 2*160;
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return 20;
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}
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AVCodec ra_144_decoder =
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{
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"real_144",
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CODEC_TYPE_AUDIO,
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CODEC_ID_RA_144,
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sizeof(RA144Context),
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ra144_decode_init,
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NULL,
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NULL,
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ra144_decode_frame,
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.long_name = "RealAudio 1.0 (14.4K)",
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};
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