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/*
* RealAudio 2.0 (28.8K)
* 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"
#define ALT_BITSTREAM_READER_LE
#include "bitstream.h"
#include "ra288.h"
#include "lpc.h"
#include "celp_math.h"
typedef struct {
float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A)
float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB)
/** speech data history (spec: SB).
* Its first 70 coefficients are updated only at backward filtering.
*/
float sp_hist[111];
/// speech part of the gain autocorrelation (spec: REXP)
float sp_rec[37];
/** log-gain history (spec: SBLG).
* Its first 28 coefficients are updated only at backward filtering.
*/
float gain_hist[38];
/// recursive part of the gain autocorrelation (spec: REXPLG)
float gain_rec[11];
} RA288Context;
static av_cold int ra288_decode_init(AVCodecContext *avctx)
{
avctx->sample_fmt = SAMPLE_FMT_FLT;
return 0;
}
static void apply_window(float *tgt, const float *m1, const float *m2, int n)
{
while (n--)
*tgt++ = *m1++ * *m2++;
}
static void convolve(float *tgt, const float *src, int len, int n)
{
for (; n >= 0; n--)
tgt[n] = ff_dot_productf(src, src - n, len);
}
static void decode(RA288Context *ractx, float gain, int cb_coef)
{
int i, j;
double sumsum;
float sum, buffer[5];
float *block = ractx->sp_hist + 70 + 36; // current block
float *gain_block = ractx->gain_hist + 28;
memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
/* block 46 of G.728 spec */
sum = 32.;
for (i=0; i < 10; i++)
sum -= gain_block[9-i] * ractx->gain_lpc[i];
/* block 47 of G.728 spec */
sum = av_clipf(sum, 0, 60);
/* block 48 of G.728 spec */
/* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
for (i=0; i < 5; i++)
buffer[i] = codetable[cb_coef][i] * sumsum;
sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.);
sum = FFMAX(sum, 1);
/* shift and store */
memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
gain_block[9] = 10 * log10(sum) - 32;
for (i=0; i < 5; i++) {
block[i] = buffer[i];
for (j=0; j < 36; j++)
block[i] -= block[i-1-j]*ractx->sp_lpc[j];
}
/* output */
for (i=0; i < 5; i++)
block[i] = av_clipf(block[i], -4095./4096., 4095./4096.);
}
/**
* Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
*
* @param order filter order
* @param n input length
* @param non_rec number of non-recursive samples
* @param out filter output
* @param hist pointer to the input history of the filter
* @param out pointer to the non-recursive part of the output
* @param out2 pointer to the recursive part of the output
* @param window pointer to the windowing function table
*/
static void do_hybrid_window(int order, int n, int non_rec, float *out,
float *hist, float *out2, const float *window)
{
int i;
float buffer1[order + 1];
float buffer2[order + 1];
float work[order + n + non_rec];
apply_window(work, window, hist, order + n + non_rec);
convolve(buffer1, work + order , n , order);
convolve(buffer2, work + order + n, non_rec, order);
for (i=0; i <= order; i++) {
out2[i] = out2[i] * 0.5625 + buffer1[i];
out [i] = out2[i] + buffer2[i];
}
/* Multiply by the white noise correcting factor (WNCF). */
*out *= 257./256.;
}
/**
* Backward synthesis filter, find the LPC coefficients from past speech data.
*/
static void backward_filter(float *hist, float *rec, const float *window,
float *lpc, const float *tab,
int order, int n, int non_rec, int move_size)
{
float temp[order+1];
do_hybrid_window(order, n, non_rec, temp, hist, rec, window);
if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
apply_window(lpc, lpc, tab, order);
memmove(hist, hist + n, move_size*sizeof(*hist));
}
static int ra288_decode_frame(AVCodecContext * avctx, void *data,
int *data_size, const uint8_t * buf,
int buf_size)
{
float *out = data;
int i, j;
RA288Context *ractx = avctx->priv_data;
GetBitContext gb;
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR,
"Error! Input buffer is too small [%d<%d]\n",
buf_size, avctx->block_align);
return 0;
}
if (*data_size < 32*5*4)
return -1;
init_get_bits(&gb, buf, avctx->block_align * 8);
for (i=0; i < 32; i++) {
float gain = amptable[get_bits(&gb, 3)];
int cb_coef = get_bits(&gb, 6 + (i&1));
decode(ractx, gain, cb_coef);
for (j=0; j < 5; j++)
*(out++) = ractx->sp_hist[70 + 36 + j];
if ((i & 7) == 3) {
backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window,
ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window,
ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
}
}
*data_size = (char *)out - (char *)data;
return avctx->block_align;
}
AVCodec ra_288_decoder =
{
"real_288",
CODEC_TYPE_AUDIO,
CODEC_ID_RA_288,
sizeof(RA288Context),
ra288_decode_init,
NULL,
NULL,
ra288_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
};
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