2 * Real Audio 1.0 (14.4K)
4 * Copyright (c) 2008 Vitor Sessak
5 * Copyright (c) 2003 Nick Kurshev
6 * Based on public domain decoder at http://www.honeypot.net/audio
8 * This file is part of Libav.
10 * Libav is free software; you can redistribute it and/or
11 * modify it under the terms of the GNU Lesser General Public
12 * License as published by the Free Software Foundation; either
13 * version 2.1 of the License, or (at your option) any later version.
15 * Libav is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18 * Lesser General Public License for more details.
20 * You should have received a copy of the GNU Lesser General Public
21 * License along with Libav; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
25 #include "libavutil/channel_layout.h"
28 #include "bitstream.h"
33 static av_cold int ra144_decode_init(AVCodecContext * avctx)
35 RA144Context *ractx = avctx->priv_data;
39 ractx->lpc_coef[0] = ractx->lpc_tables[0];
40 ractx->lpc_coef[1] = ractx->lpc_tables[1];
43 avctx->channel_layout = AV_CH_LAYOUT_MONO;
44 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
49 static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs,
50 int gval, BitstreamContext *bc)
52 int cba_idx = bitstream_read(bc, 7); // index of the adaptive CB, 0 if none
53 int gain = bitstream_read(bc, 8);
54 int cb1_idx = bitstream_read(bc, 7);
55 int cb2_idx = bitstream_read(bc, 7);
57 ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, gval,
61 /** Uncompress one block (20 bytes -> 160*2 bytes). */
62 static int ra144_decode_frame(AVCodecContext * avctx, void *data,
63 int *got_frame_ptr, AVPacket *avpkt)
65 AVFrame *frame = data;
66 const uint8_t *buf = avpkt->data;
67 int buf_size = avpkt->size;
68 static const uint8_t sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
69 unsigned int refl_rms[NBLOCKS]; // RMS of the reflection coefficients
70 uint16_t block_coefs[NBLOCKS][LPC_ORDER]; // LPC coefficients of each sub-block
71 unsigned int lpc_refl[LPC_ORDER]; // LPC reflection coefficients of the frame
77 RA144Context *ractx = avctx->priv_data;
80 if (buf_size < FRAMESIZE) {
81 av_log(avctx, AV_LOG_ERROR,
82 "Frame too small (%d bytes). Truncated file?\n", buf_size);
84 return AVERROR_INVALIDDATA;
87 /* get output buffer */
88 frame->nb_samples = NBLOCKS * BLOCKSIZE;
89 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
90 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
93 samples = (int16_t *)frame->data[0];
95 bitstream_init8(&bc, buf, FRAMESIZE);
97 for (i = 0; i < LPC_ORDER; i++)
98 lpc_refl[i] = ff_lpc_refl_cb[i][bitstream_read(&bc, sizes[i])];
100 ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
101 ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);
103 energy = ff_energy_tab[bitstream_read(&bc, 5)];
105 refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
106 refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
107 energy <= ractx->old_energy,
108 ff_t_sqrt(energy*ractx->old_energy) >> 12);
109 refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
110 refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);
112 ff_int_to_int16(block_coefs[3], ractx->lpc_coef[0]);
114 for (i=0; i < NBLOCKS; i++) {
115 do_output_subblock(ractx, block_coefs[i], refl_rms[i], &bc);
117 for (j=0; j < BLOCKSIZE; j++)
118 *samples++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2);
121 ractx->old_energy = energy;
122 ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
124 FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
131 AVCodec ff_ra_144_decoder = {
133 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
134 .type = AVMEDIA_TYPE_AUDIO,
135 .id = AV_CODEC_ID_RA_144,
136 .priv_data_size = sizeof(RA144Context),
137 .init = ra144_decode_init,
138 .decode = ra144_decode_frame,
139 .capabilities = AV_CODEC_CAP_DR1,