ffmpeg/libavcodec/ra144.c
Vitor Sessak d981252c27 Rename variables
Originally committed as revision 13596 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-06-01 13:16:13 +00:00

395 lines
10 KiB
C

/*
* Real Audio 1.0 (14.4K)
* Copyright (c) 2003 the ffmpeg project
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
#include "bitstream.h"
#include "ra144.h"
#define NBLOCKS 4 /* number of segments within a block */
#define BLOCKSIZE 40 /* (quarter) block size in 16-bit words (80 bytes) */
#define HALFBLOCK 20 /* BLOCKSIZE/2 */
#define BUFFERSIZE 146 /* for do_output */
/* internal globals */
typedef struct {
unsigned int old_energy; ///< previous frame energy
/* the swapped buffers */
unsigned int lpc_tables[4][10];
unsigned int *lpc_refl; ///< LPC reflection coefficients
unsigned int *lpc_coef; ///< LPC coefficients
unsigned int *lpc_refl_old; ///< previous frame LPC reflection coefs
unsigned int *lpc_coef_old; ///< previous frame LPC coefficients
unsigned int buffer[5];
uint16_t adapt_cb[148]; ///< adaptive codebook
} RA144Context;
static int ra144_decode_init(AVCodecContext * avctx)
{
RA144Context *ractx = avctx->priv_data;
ractx->lpc_refl = ractx->lpc_tables[0];
ractx->lpc_coef = ractx->lpc_tables[1];
ractx->lpc_refl_old = ractx->lpc_tables[2];
ractx->lpc_coef_old = ractx->lpc_tables[3];
return 0;
}
/**
* Evaluate sqrt(x << 24). x must fit in 20 bits. This value is evaluated in an
* odd way to make the output identical to the binary decoder.
*/
static int t_sqrt(unsigned int x)
{
int s = 0;
while (x > 0xfff) {
s++;
x = x >> 2;
}
return (ff_sqrt(x << 20) << s) << 2;
}
/**
* Evaluate the LPC filter coefficients from the reflection coefficients.
* Does the inverse of the eq() function.
*/
static void do_voice(const int *refl, int *coefs)
{
int buffer[10];
int *b1 = buffer;
int *b2 = coefs;
int x, y;
for (x=0; x < 10; x++) {
b1[x] = refl[x] << 4;
for (y=0; y < x; y++)
b1[y] = ((refl[x] * b2[x-y-1]) >> 12) + b2[y];
FFSWAP(int *, b1, b2);
}
for (x=0; x < 10; x++)
coefs[x] >>= 4;
}
/* rotate block */
static void rotate_block(const int16_t *source, int16_t *target, int offset)
{
int i=0, k=0;
source += BUFFERSIZE - offset;
while (i<BLOCKSIZE) {
target[i++] = source[k++];
if (k == offset)
k = 0;
}
}
/* inverse root mean square */
static int irms(const int16_t *data, int factor)
{
unsigned int i, sum = 0;
for (i=0; i < BLOCKSIZE; i++)
sum += data[i] * data[i];
if (sum == 0)
return 0; /* OOPS - division by zero */
return (0x20000000 / (t_sqrt(sum) >> 8)) * factor;
}
/* multiply/add wavetable */
static void add_wav(int n, int skip_first, int *m, const int16_t *s1,
const int8_t *s2, const int8_t *s3, int16_t *dest)
{
int i;
int v[3];
v[0] = 0;
for (i=!skip_first; i<3; i++)
v[i] = (wavtable1[n][i] * m[i]) >> (wavtable2[n][i] + 1);
for (i=0; i < BLOCKSIZE; i++)
dest[i] = ((*(s1++))*v[0] + (*(s2++))*v[1] + (*(s3++))*v[2]) >> 12;
}
static void final(const int16_t *i1, const int16_t *i2,
void *out, int *statbuf, int len)
{
int x, i;
uint16_t work[50];
int16_t *ptr = work;
memcpy(work, statbuf,20);
memcpy(work + 10, i2, len * 2);
for (i=0; i<len; i++) {
int sum = 0;
int new_val;
for(x=0; x<10; x++)
sum += i1[9-x] * ptr[x];
sum >>= 12;
new_val = ptr[10] - sum;
if (new_val < -32768 || new_val > 32767) {
memset(out, 0, len * 2);
memset(statbuf, 0, 20);
return;
}
ptr[10] = new_val;
ptr++;
}
memcpy(out, work+10, len * 2);
memcpy(statbuf, work + 40, 20);
}
static unsigned int rms(const int *data, int f)
{
int x;
unsigned int res = 0x10000;
int b = 0;
for (x=0; x<10; x++) {
res = (((0x1000000 - (*data) * (*data)) >> 12) * res) >> 12;
if (res == 0)
return 0;
if (res > 0x10000)
return 0; /* We're screwed, might as well go out with a bang. :P */
while (res <= 0x3fff) {
b++;
res <<= 2;
}
data++;
}
if (res > 0)
res = t_sqrt(res);
res >>= (b + 10);
res = (res * f) >> 10;
return res;
}
/* do quarter-block output */
static void do_output_subblock(RA144Context *ractx,
const uint16_t *gsp, unsigned int gval,
int16_t *output_buffer, GetBitContext *gb)
{
uint16_t buffer_a[40];
uint16_t *block;
int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none
int gain = get_bits(gb, 8);
int cb1_idx = get_bits(gb, 7);
int cb2_idx = get_bits(gb, 7);
int m[3];
if (cba_idx) {
cba_idx += HALFBLOCK - 1;
rotate_block(ractx->adapt_cb, buffer_a, cba_idx);
m[0] = irms(buffer_a, gval) >> 12;
} else {
m[0] = 0;
}
m[1] = ((ftable1[cb1_idx] >> 4) * gval) >> 8;
m[2] = ((ftable2[cb2_idx] >> 4) * gval) >> 8;
memmove(ractx->adapt_cb, ractx->adapt_cb + BLOCKSIZE,
(BUFFERSIZE - BLOCKSIZE) * 2);
block = ractx->adapt_cb + BUFFERSIZE - BLOCKSIZE;
add_wav(gain, cba_idx, m, buffer_a, etable1[cb1_idx], etable2[cb2_idx],
block);
final(gsp, block, output_buffer, ractx->buffer, BLOCKSIZE);
}
static int dec1(int16_t *decsp, const int *data, const int *inp, int f)
{
int i;
for (i=0; i<30; i++)
*(decsp++) = *(inp++);
return rms(data, f);
}
/**
* Evaluate the reflection coefficients from the filter coefficients.
* Does the inverse of the do_voice() function.
*
* @return 1 if one of the reflection coefficients is of magnitude greater than
* 4095, 0 if not.
*/
static int eq(const int16_t *coefs, int *refl)
{
int retval = 0;
int b, c, i;
unsigned int u;
int buffer1[10];
int buffer2[10];
int *bp1 = buffer1;
int *bp2 = buffer2;
for (i=0; i < 10; i++)
buffer2[i] = coefs[i];
u = refl[9] = bp2[9];
if (u + 0x1000 > 0x1fff)
return 0; /* We're screwed, might as well go out with a bang. :P */
for (c=8; c >= 0; c--) {
if (u == 0x1000)
u++;
if (u == 0xfffff000)
u--;
b = 0x1000-((u * u) >> 12);
if (b == 0)
b++;
for (u=0; u<=c; u++)
bp1[u] = ((bp2[u] - ((refl[c+1] * bp2[c-u]) >> 12)) * (0x1000000 / b)) >> 12;
refl[c] = u = bp1[c];
if ((u + 0x1000) > 0x1fff)
retval = 1;
FFSWAP(int *, bp1, bp2);
}
return retval;
}
static int dec2(RA144Context *ractx, int16_t *decsp, int block_num,
int copynew, int f)
{
int work[10];
int a = block_num + 1;
int b = NBLOCKS - a;
int x;
// Interpolate block coefficients from the this frame forth block and
// last frame forth block
for (x=0; x<30; x++)
decsp[x] = (a * ractx->lpc_coef[x] + b * ractx->lpc_coef_old[x])>> 2;
if (eq(decsp, work)) {
// The interpolated coefficients are unstable, copy either new or old
// coefficients
if (copynew)
return dec1(decsp, ractx->lpc_refl, ractx->lpc_coef, f);
else
return dec1(decsp, ractx->lpc_refl_old, ractx->lpc_coef_old, f);
} else {
return rms(work, f);
}
}
/* Uncompress one block (20 bytes -> 160*2 bytes) */
static int ra144_decode_frame(AVCodecContext * avctx,
void *vdata, int *data_size,
const uint8_t * buf, int buf_size)
{
static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
unsigned int refl_rms[4]; // RMS of the reflection coefficients
uint16_t gbuf2[4][30];
int i, c;
int16_t *data = vdata;
unsigned int energy;
RA144Context *ractx = avctx->priv_data;
GetBitContext gb;
if(buf_size < 20) {
av_log(avctx, AV_LOG_ERROR,
"Frame too small (%d bytes). Truncated file?\n", buf_size);
return buf_size;
}
init_get_bits(&gb, buf, 20 * 8);
for (i=0; i<10; i++)
// "<< 1"? Doesn't this make one value out of two of the table useless?
ractx->lpc_refl[i] = decodetable[i][get_bits(&gb, sizes[i]) << 1];
do_voice(ractx->lpc_refl, ractx->lpc_coef);
energy = decodeval[get_bits(&gb, 5) << 1]; // Useless table entries?
refl_rms[0] = dec2(ractx, gbuf2[0], 0, 0, ractx->old_energy);
refl_rms[1] = dec2(ractx, gbuf2[1], 1, energy > ractx->old_energy,
t_sqrt(energy*ractx->old_energy) >> 12);
refl_rms[2] = dec2(ractx, gbuf2[2], 2, 1, energy);
refl_rms[3] = dec1(gbuf2[3], ractx->lpc_refl, ractx->lpc_coef, energy);
/* do output */
for (c=0; c<4; c++) {
do_output_subblock(ractx, gbuf2[c], refl_rms[c], data, &gb);
for (i=0; i<BLOCKSIZE; i++) {
*data = av_clip_int16(*data << 2);
data++;
}
}
ractx->old_energy = energy;
FFSWAP(unsigned int *, ractx->lpc_refl_old, ractx->lpc_refl);
FFSWAP(unsigned int *, ractx->lpc_coef_old, ractx->lpc_coef);
*data_size = 2*160;
return 20;
}
AVCodec ra_144_decoder =
{
"real_144",
CODEC_TYPE_AUDIO,
CODEC_ID_RA_144,
sizeof(RA144Context),
ra144_decode_init,
NULL,
NULL,
ra144_decode_frame,
.long_name = "RealAudio 1.0 (14.4K)",
};