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/*
* Real Audio 1.0 (14.4K)
*
* Copyright (c) 2008 Vitor Sessak
* Copyright (c) 2003 Nick Kurshev
* Based on public domain decoder at http://www.honeypot.net/audio
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/channel_layout.h"
#include "avcodec.h"
#include "bitstream.h"
#include "internal.h"
#include "ra144.h"
static av_cold int ra144_decode_init(AVCodecContext * avctx)
{
RA144Context *ractx = avctx->priv_data;
ractx->avctx = avctx;
ractx->lpc_coef[0] = ractx->lpc_tables[0];
ractx->lpc_coef[1] = ractx->lpc_tables[1];
avctx->channels = 1;
avctx->channel_layout = AV_CH_LAYOUT_MONO;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}
static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs,
int gval, BitstreamContext *bc)
{
int cba_idx = bitstream_read(bc, 7); // index of the adaptive CB, 0 if none
int gain = bitstream_read(bc, 8);
int cb1_idx = bitstream_read(bc, 7);
int cb2_idx = bitstream_read(bc, 7);
ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, gval,
gain);
}
/** Uncompress one block (20 bytes -> 160*2 bytes). */
static int ra144_decode_frame(AVCodecContext * avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
static const uint8_t sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
unsigned int refl_rms[NBLOCKS]; // RMS of the reflection coefficients
uint16_t block_coefs[NBLOCKS][LPC_ORDER]; // LPC coefficients of each sub-block
unsigned int lpc_refl[LPC_ORDER]; // LPC reflection coefficients of the frame
int i, j;
int ret;
int16_t *samples;
unsigned int energy;
RA144Context *ractx = avctx->priv_data;
BitstreamContext bc;
if (buf_size < FRAMESIZE) {
av_log(avctx, AV_LOG_ERROR,
"Frame too small (%d bytes). Truncated file?\n", buf_size);
*got_frame_ptr = 0;
return AVERROR_INVALIDDATA;
}
/* get output buffer */
frame->nb_samples = NBLOCKS * BLOCKSIZE;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)frame->data[0];
bitstream_init(&bc, buf, FRAMESIZE * 8);
for (i = 0; i < LPC_ORDER; i++)
lpc_refl[i] = ff_lpc_refl_cb[i][bitstream_read(&bc, sizes[i])];
ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);
energy = ff_energy_tab[bitstream_read(&bc, 5)];
refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
energy <= ractx->old_energy,
ff_t_sqrt(energy*ractx->old_energy) >> 12);
refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);
ff_int_to_int16(block_coefs[3], ractx->lpc_coef[0]);
for (i=0; i < NBLOCKS; i++) {
do_output_subblock(ractx, block_coefs[i], refl_rms[i], &bc);
for (j=0; j < BLOCKSIZE; j++)
*samples++ = av_clip_int16(ractx->curr_sblock[j + 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]);
*got_frame_ptr = 1;
return FRAMESIZE;
}
AVCodec ff_ra_144_decoder = {
.name = "real_144",
.long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_RA_144,
.priv_data_size = sizeof(RA144Context),
.init = ra144_decode_init,
.decode = ra144_decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
};
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