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/*
* 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 subblocks within a block
#define BLOCKSIZE 40 ///< subblock size in 16-bit words
#define BUFFERSIZE 146 ///< the size of the adaptive codebook
typedef struct {
unsigned int old_energy; ///< previous frame energy
unsigned int lpc_tables[2][10];
/** LPC coefficients: lpc_coef[0] is the coefficients of the current frame
* and lpc_coef[1] of the previous one */
unsigned int *lpc_coef[2];
unsigned int lpc_refl_rms[2];
/** the current subblock padded by the last 10 values of the previous one*/
int16_t curr_sblock[50];
/** adaptive codebook. Its size is two units bigger to avoid a
* buffer overflow */
uint16_t adapt_cb[148];
} RA144Context;
static int ra144_decode_init(AVCodecContext * avctx)
{
RA144Context *ractx = avctx->priv_data;
ractx->lpc_coef[0] = ractx->lpc_tables[0];
ractx->lpc_coef[1] = ractx->lpc_tables[1];
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 = 2;
while (x > 0xfff) {
s++;
x = x >> 2;
}
return ff_sqrt(x << 20) << s;
}
/**
* Evaluate the LPC filter coefficients from the reflection coefficients.
* Does the inverse of the eval_refl() function.
*/
static void eval_coefs(int *coefs, const int *refl)
{
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;
}
/**
* Copy the last offset values of *source to *target. If those values are not
* enough to fill the target buffer, fill it with another copy of those values.
*/
static void copy_and_dup(int16_t *target, const int16_t *source, int offset)
{
source += BUFFERSIZE - offset;
if (offset > BLOCKSIZE) {
memcpy(target, source, BLOCKSIZE*sizeof(*target));
} else {
memcpy(target, source, offset*sizeof(*target));
memcpy(target + offset, source, (BLOCKSIZE - offset)*sizeof(*target));
}
}
/** inverse root mean square */
static int irms(const int16_t *data)
{
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);
}
static void add_wav(int16_t *dest, int n, int skip_first, int *m,
const int16_t *s1, const int8_t *s2, const int8_t *s3)
{
int i;
int v[3];
v[0] = 0;
for (i=!skip_first; i<3; i++)
v[i] = (gain_val_tab[n][i] * m[i]) >> (gain_exp_tab[n][i] + 1);
for (i=0; i < BLOCKSIZE; i++)
dest[i] = (s1[i]*v[0] + s2[i]*v[1] + s3[i]*v[2]) >> 12;
}
/**
* LPC Filter. Each output value is predicted from the 10 previous computed
* ones. It overwrites the input with the output.
*
* @param in the input of the filter. It should be an array of size len + 10.
* The 10 first input values are used to evaluate the first filtered one.
*/
static void lpc_filter(uint16_t *in, const int16_t *lpc_coefs, int len)
{
int x, i;
int16_t *ptr = in;
for (i=0; i<len; i++) {
int sum = 0;
int new_val;
for(x=0; x<10; x++)
sum += lpc_coefs[9-x] * ptr[x];
sum >>= 12;
new_val = ptr[10] - sum;
if (new_val < -32768 || new_val > 32767) {
memset(in, 0, 50*sizeof(*in));
return;
}
ptr[10] = new_val;
ptr++;
}
}
static unsigned int rescale_rms(unsigned int rms, unsigned int energy)
{
return (rms * energy) >> 10;
}
static unsigned int rms(const int *data)
{
int x;
unsigned int res = 0x10000;
int b = 0;
for (x=0; x<10; x++) {
res = (((0x1000000 - data[x]*data[x]) >> 12) * res) >> 12;
if (res == 0)
return 0;
while (res <= 0x3fff) {
b++;
res <<= 2;
}
}
res = t_sqrt(res);
res >>= (b + 10);
return res;
}
static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs,
int gval, 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 += BLOCKSIZE/2 - 1;
copy_and_dup(buffer_a, ractx->adapt_cb, cba_idx);
m[0] = (irms(buffer_a) * gval) >> 12;
} else {
m[0] = 0;
}
m[1] = (cb1_base[cb1_idx] * gval) >> 8;
m[2] = (cb2_base[cb2_idx] * gval) >> 8;
memmove(ractx->adapt_cb, ractx->adapt_cb + BLOCKSIZE,
(BUFFERSIZE - BLOCKSIZE) * sizeof(*ractx->adapt_cb));
block = ractx->adapt_cb + BUFFERSIZE - BLOCKSIZE;
add_wav(block, gain, cba_idx, m, buffer_a,
cb1_vects[cb1_idx], cb2_vects[cb2_idx]);
memcpy(ractx->curr_sblock, ractx->curr_sblock + 40,
10*sizeof(*ractx->curr_sblock));
memcpy(ractx->curr_sblock + 10, block,
BLOCKSIZE*sizeof(*ractx->curr_sblock));
lpc_filter(ractx->curr_sblock, lpc_coefs, BLOCKSIZE);
}
static void int_to_int16(int16_t *out, const int *inp)
{
int i;
for (i=0; i<30; i++)
*(out++) = *(inp++);
}
/**
* Evaluate the reflection coefficients from the filter coefficients.
* Does the inverse of the eval_coefs() function.
*
* @return 1 if one of the reflection coefficients is of magnitude greater than
* 4095, 0 if not.
*/
static int eval_refl(int *refl, const int16_t *coefs, RA144Context *ractx)
{
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) {
av_log(ractx, AV_LOG_ERROR, "Overflow. Broken sample?\n");
return 1;
}
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 interp(RA144Context *ractx, int16_t *out, int block_num,
int copyold, int energy)
{
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++)
out[x] = (a * ractx->lpc_coef[0][x] + b * ractx->lpc_coef[1][x])>> 2;
if (eval_refl(work, out, ractx)) {
// The interpolated coefficients are unstable, copy either new or old
// coefficients
int_to_int16(out, ractx->lpc_coef[copyold]);
return rescale_rms(ractx->lpc_refl_rms[copyold], energy);
} else {
return rescale_rms(rms(work), energy);
}
}
/** 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 block_coefs[4][30]; // LPC coefficients of each sub-block
unsigned int lpc_refl[10]; // LPC reflection coefficients of the frame
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);
*data_size = 0;
return buf_size;
}
init_get_bits(&gb, buf, 20 * 8);
for (i=0; i<10; i++)
lpc_refl[i] = lpc_refl_cb[i][get_bits(&gb, sizes[i])];
eval_coefs(ractx->lpc_coef[0], lpc_refl);
ractx->lpc_refl_rms[0] = rms(lpc_refl);
energy = energy_tab[get_bits(&gb, 5)];
refl_rms[0] = interp(ractx, block_coefs[0], 0, 1, ractx->old_energy);
refl_rms[1] = interp(ractx, block_coefs[1], 1, energy <= ractx->old_energy,
t_sqrt(energy*ractx->old_energy) >> 12);
refl_rms[2] = interp(ractx, block_coefs[2], 2, 0, energy);
refl_rms[3] = rescale_rms(ractx->lpc_refl_rms[0], energy);
int_to_int16(block_coefs[3], ractx->lpc_coef[0]);
for (c=0; c<4; c++) {
do_output_subblock(ractx, block_coefs[c], refl_rms[c], &gb);
for (i=0; i<BLOCKSIZE; i++)
*data++ = av_clip_int16(ractx->curr_sblock[i + 10] << 2);
}
ractx->old_energy = energy;
ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
*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 = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
};
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