mirror of
https://github.com/xenia-project/FFmpeg.git
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400 lines
10 KiB
C
400 lines
10 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 subblocks within a block
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#define BLOCKSIZE 40 ///< subblock size in 16-bit words
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#define BUFFERSIZE 146 ///< the size of the adaptive codebook
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typedef struct {
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unsigned int old_energy; ///< previous frame energy
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unsigned int lpc_tables[2][10];
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/** LPC coefficients: lpc_coef[0] is the coefficients of the current frame
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* and lpc_coef[1] of the previous one */
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unsigned int *lpc_coef[2];
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unsigned int lpc_refl_rms[2];
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/** the current subblock padded by the last 10 values of the previous one*/
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int16_t curr_sblock[50];
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/** adaptive codebook. Its size is two units bigger to avoid a
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* buffer overflow */
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uint16_t adapt_cb[148];
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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_coef[0] = ractx->lpc_tables[0];
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ractx->lpc_coef[1] = ractx->lpc_tables[1];
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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 = 2;
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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;
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}
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/**
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* Evaluate the LPC filter coefficients from the reflection coefficients.
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* Does the inverse of the eval_refl() function.
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*/
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static void eval_coefs(int *coefs, const int *refl)
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{
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int buffer[10];
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int *b1 = buffer;
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int *b2 = coefs;
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int x, y;
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for (x=0; x < 10; x++) {
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b1[x] = refl[x] << 4;
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for (y=0; y < x; y++)
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b1[y] = ((refl[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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coefs[x] >>= 4;
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}
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/**
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* Copy the last offset values of *source to *target. If those values are not
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* enough to fill the target buffer, fill it with another copy of those values.
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*/
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static void copy_and_dup(int16_t *target, const int16_t *source, int offset)
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{
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source += BUFFERSIZE - offset;
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if (offset > BLOCKSIZE) {
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memcpy(target, source, BLOCKSIZE*sizeof(*target));
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} else {
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memcpy(target, source, offset*sizeof(*target));
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memcpy(target + offset, source, (BLOCKSIZE - offset)*sizeof(*target));
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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)
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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);
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}
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static void add_wav(int16_t *dest, int n, int skip_first, int *m,
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const int16_t *s1, const int8_t *s2, const int8_t *s3)
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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] = (gain_val_tab[n][i] * m[i]) >> (gain_exp_tab[n][i] + 1);
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for (i=0; i < BLOCKSIZE; i++)
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dest[i] = (s1[i]*v[0] + s2[i]*v[1] + s3[i]*v[2]) >> 12;
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}
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/**
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* LPC Filter. Each output value is predicted from the 10 previous computed
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* ones. It overwrites the input with the output.
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*
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* @param in the input of the filter. It should be an array of size len + 10.
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* The 10 first input values are used to evaluate the first filtered one.
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*/
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static void lpc_filter(uint16_t *in, const int16_t *lpc_coefs, int len)
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{
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int x, i;
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int16_t *ptr = in;
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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 += lpc_coefs[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(in, 0, 50*sizeof(*in));
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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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}
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static unsigned int rescale_rms(unsigned int rms, unsigned int energy)
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{
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return (rms * energy) >> 10;
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}
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static unsigned int rms(const int *data)
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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[x]*data[x]) >> 12) * res) >> 12;
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if (res == 0)
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return 0;
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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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}
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res = t_sqrt(res);
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res >>= (b + 10);
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return res;
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}
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static void do_output_subblock(RA144Context *ractx,
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const uint16_t *lpc_coefs, int gval,
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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 += BLOCKSIZE/2 - 1;
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copy_and_dup(buffer_a, ractx->adapt_cb, 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] = (cb1_base[cb1_idx] * gval) >> 8;
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m[2] = (cb2_base[cb2_idx] * gval) >> 8;
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memmove(ractx->adapt_cb, ractx->adapt_cb + BLOCKSIZE,
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(BUFFERSIZE - BLOCKSIZE) * sizeof(*ractx->adapt_cb));
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block = ractx->adapt_cb + BUFFERSIZE - BLOCKSIZE;
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add_wav(block, gain, cba_idx, m, buffer_a,
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cb1_vects[cb1_idx], cb2_vects[cb2_idx]);
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memcpy(ractx->curr_sblock, ractx->curr_sblock + 40,
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10*sizeof(*ractx->curr_sblock));
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memcpy(ractx->curr_sblock + 10, block,
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BLOCKSIZE*sizeof(*ractx->curr_sblock));
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lpc_filter(ractx->curr_sblock, lpc_coefs, BLOCKSIZE);
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}
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static void int_to_int16(int16_t *out, const int *inp)
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{
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int i;
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for (i=0; i<30; i++)
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*(out++) = *(inp++);
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}
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/**
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* Evaluate the reflection coefficients from the filter coefficients.
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* Does the inverse of the eval_coefs() function.
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*
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* @return 1 if one of the reflection coefficients is of magnitude greater than
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* 4095, 0 if not.
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*/
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static int eval_refl(int *refl, const int16_t *coefs, RA144Context *ractx)
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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] = coefs[i];
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u = refl[9] = bp2[9];
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if (u + 0x1000 > 0x1fff) {
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av_log(ractx, AV_LOG_ERROR, "Overflow. Broken sample?\n");
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return 1;
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}
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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] - ((refl[c+1] * bp2[c-u]) >> 12)) * (0x1000000 / b)) >> 12;
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refl[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 interp(RA144Context *ractx, int16_t *out, int block_num,
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int copynew, int energy)
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{
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int work[10];
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int a = block_num + 1;
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int b = NBLOCKS - a;
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int x;
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// Interpolate block coefficients from the this frame forth block and
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// last frame forth block
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for (x=0; x<30; x++)
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out[x] = (a * ractx->lpc_coef[0][x] + b * ractx->lpc_coef[1][x])>> 2;
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if (eval_refl(work, out, ractx)) {
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// The interpolated coefficients are unstable, copy either new or old
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// coefficients
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if (copynew) {
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int_to_int16(out, ractx->lpc_coef[0]);
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return rescale_rms(ractx->lpc_refl_rms[0], energy);
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} else {
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int_to_int16(out, ractx->lpc_coef[1]);
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return rescale_rms(ractx->lpc_refl_rms[1], energy);
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}
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} else {
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return rescale_rms(rms(work), energy);
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}
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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 refl_rms[4]; // RMS of the reflection coefficients
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uint16_t block_coefs[4][30]; // LPC coefficients of each sub-block
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unsigned int lpc_refl[10]; // LPC reflection coefficients of the frame
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int i, c;
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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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*data_size = 0;
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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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lpc_refl[i] = lpc_refl_cb[i][get_bits(&gb, sizes[i])];
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eval_coefs(ractx->lpc_coef[0], lpc_refl);
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ractx->lpc_refl_rms[0] = rms(lpc_refl);
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energy = energy_tab[get_bits(&gb, 5)];
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refl_rms[0] = interp(ractx, block_coefs[0], 0, 0, ractx->old_energy);
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refl_rms[1] = interp(ractx, block_coefs[1], 1, energy > ractx->old_energy,
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t_sqrt(energy*ractx->old_energy) >> 12);
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refl_rms[2] = interp(ractx, block_coefs[2], 2, 1, energy);
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refl_rms[3] = rescale_rms(ractx->lpc_refl_rms[0], energy);
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int_to_int16(block_coefs[3], ractx->lpc_coef[0]);
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for (c=0; c<4; c++) {
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do_output_subblock(ractx, block_coefs[c], refl_rms[c], &gb);
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for (i=0; i<BLOCKSIZE; i++)
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*data++ = av_clip_int16(ractx->curr_sblock[i + 10] << 2);
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}
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ractx->old_energy = energy;
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ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
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FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
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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 = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
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};
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