mirror of
https://github.com/xenia-project/FFmpeg.git
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138 lines
4.5 KiB
C
138 lines
4.5 KiB
C
/*
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* Real Audio 1.0 (14.4K)
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*
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* Copyright (c) 2008 Vitor Sessak
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* Copyright (c) 2003 Nick Kurshev
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* Based on public domain decoder at http://www.honeypot.net/audio
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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 "libavutil/channel_layout.h"
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#include "avcodec.h"
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#include "get_bits.h"
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#include "internal.h"
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#include "ra144.h"
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static av_cold int ra144_decode_init(AVCodecContext * avctx)
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{
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RA144Context *ractx = avctx->priv_data;
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ractx->avctx = avctx;
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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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avctx->channels = 1;
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avctx->channel_layout = AV_CH_LAYOUT_MONO;
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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return 0;
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}
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static void do_output_subblock(RA144Context *ractx, const int16_t *lpc_coefs,
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int gval, GetBitContext *gb)
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{
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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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ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, gval,
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gain);
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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, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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AVFrame *frame = data;
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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static const uint8_t sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
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unsigned int refl_rms[NBLOCKS]; // RMS of the reflection coefficients
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int16_t block_coefs[NBLOCKS][LPC_ORDER]; // LPC coefficients of each sub-block
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unsigned int lpc_refl[LPC_ORDER]; // LPC reflection coefficients of the frame
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int i, j;
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int ret;
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int16_t *samples;
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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 < FRAME_SIZE) {
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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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*got_frame_ptr = 0;
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return AVERROR_INVALIDDATA;
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}
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/* get output buffer */
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frame->nb_samples = NBLOCKS * BLOCKSIZE;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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samples = (int16_t *)frame->data[0];
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init_get_bits8(&gb, buf, FRAME_SIZE);
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for (i = 0; i < LPC_ORDER; i++)
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lpc_refl[i] = ff_lpc_refl_cb[i][get_bits(&gb, sizes[i])];
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ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
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ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);
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energy = ff_energy_tab[get_bits(&gb, 5)];
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refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
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refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
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energy <= ractx->old_energy,
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ff_t_sqrt(energy*ractx->old_energy) >> 12);
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refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
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refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);
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ff_int_to_int16(block_coefs[3], ractx->lpc_coef[0]);
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for (i=0; i < NBLOCKS; i++) {
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do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb);
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for (j=0; j < BLOCKSIZE; j++)
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*samples++ = av_clip_int16(ractx->curr_sblock[j + 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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*got_frame_ptr = 1;
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return FRAME_SIZE;
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}
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AVCodec ff_ra_144_decoder = {
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.name = "real_144",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_RA_144,
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.priv_data_size = sizeof(RA144Context),
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.init = ra144_decode_init,
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.decode = ra144_decode_frame,
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.capabilities = CODEC_CAP_DR1,
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.long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
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};
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