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"
32 static av_cold int ra144_decode_init(AVCodecContext * avctx)
34 RA144Context *ractx = avctx->priv_data;
38 ractx->lpc_coef[0] = ractx->lpc_tables[0];
39 ractx->lpc_coef[1] = ractx->lpc_tables[1];
42 avctx->channel_layout = AV_CH_LAYOUT_MONO;
43 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
48 static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs,
49 int gval, GetBitContext *gb)
51 int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none
52 int gain = get_bits(gb, 8);
53 int cb1_idx = get_bits(gb, 7);
54 int cb2_idx = get_bits(gb, 7);
56 ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, gval,
60 /** Uncompress one block (20 bytes -> 160*2 bytes). */
61 static int ra144_decode_frame(AVCodecContext * avctx, void *data,
62 int *got_frame_ptr, AVPacket *avpkt)
64 AVFrame *frame = data;
65 const uint8_t *buf = avpkt->data;
66 int buf_size = avpkt->size;
67 static const uint8_t sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
68 unsigned int refl_rms[NBLOCKS]; // RMS of the reflection coefficients
69 uint16_t block_coefs[NBLOCKS][LPC_ORDER]; // LPC coefficients of each sub-block
70 unsigned int lpc_refl[LPC_ORDER]; // LPC reflection coefficients of the frame
76 RA144Context *ractx = avctx->priv_data;
79 if (buf_size < FRAMESIZE) {
80 av_log(avctx, AV_LOG_ERROR,
81 "Frame too small (%d bytes). Truncated file?\n", buf_size);
83 return AVERROR_INVALIDDATA;
86 /* get output buffer */
87 frame->nb_samples = NBLOCKS * BLOCKSIZE;
88 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
89 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
92 samples = (int16_t *)frame->data[0];
94 init_get_bits(&gb, buf, FRAMESIZE * 8);
96 for (i = 0; i < LPC_ORDER; i++)
97 lpc_refl[i] = ff_lpc_refl_cb[i][get_bits(&gb, sizes[i])];
99 ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
100 ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);
102 energy = ff_energy_tab[get_bits(&gb, 5)];
104 refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
105 refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
106 energy <= ractx->old_energy,
107 ff_t_sqrt(energy*ractx->old_energy) >> 12);
108 refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
109 refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);
111 ff_int_to_int16(block_coefs[3], ractx->lpc_coef[0]);
113 for (i=0; i < NBLOCKS; i++) {
114 do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb);
116 for (j=0; j < BLOCKSIZE; j++)
117 *samples++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2);
120 ractx->old_energy = energy;
121 ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
123 FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
130 AVCodec ff_ra_144_decoder = {
132 .type = AVMEDIA_TYPE_AUDIO,
133 .id = AV_CODEC_ID_RA_144,
134 .priv_data_size = sizeof(RA144Context),
135 .init = ra144_decode_init,
136 .decode = ra144_decode_frame,
137 .capabilities = CODEC_CAP_DR1,
138 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),