/* * 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 "get_bits.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, GetBitContext *gb) { int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none int gain = get_bits(gb, 8); int cb1_idx = get_bits(gb, 7); int cb2_idx = get_bits(gb, 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; GetBitContext gb; 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]; init_get_bits(&gb, buf, FRAMESIZE * 8); for (i = 0; i < LPC_ORDER; i++) lpc_refl[i] = ff_lpc_refl_cb[i][get_bits(&gb, sizes[i])]; ff_eval_coefs(ractx->lpc_coef[0], lpc_refl); ractx->lpc_refl_rms[0] = ff_rms(lpc_refl); energy = ff_energy_tab[get_bits(&gb, 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], &gb); 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, };