Mercurial > libavcodec.hg
view ra144dec.c @ 12340:2d15f62f4f8a libavcodec
VP8: move zeroing of luma DC block into the WHT
Lets us do the zeroing in asm instead of C.
Also makes it consistent with the way the regular iDCT code does it.
author | darkshikari |
---|---|
date | Mon, 02 Aug 2010 20:18:09 +0000 |
parents | 159554445343 |
children |
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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 FFmpeg. * * FFmpeg 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. * * FFmpeg 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 FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "libavutil/intmath.h" #include "avcodec.h" #include "get_bits.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->sample_fmt = 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 *vdata, int *data_size, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2}; unsigned int refl_rms[4]; // RMS of the reflection coefficients uint16_t block_coefs[4][10]; // LPC coefficients of each sub-block unsigned int lpc_refl[10]; // LPC reflection coefficients of the frame int i, j; int16_t *data = vdata; unsigned int energy; RA144Context *ractx = avctx->priv_data; GetBitContext gb; if (*data_size < 2*160) return -1; if(buf_size < 20) { av_log(avctx, AV_LOG_ERROR, "Frame too small (%d bytes). Truncated file?\n", buf_size); *data_size = 0; return buf_size; } init_get_bits(&gb, buf, 20 * 8); for (i=0; i<10; 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 < 4; i++) { do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb); for (j=0; j < BLOCKSIZE; j++) *data++ = 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]); *data_size = 2*160; return 20; } AVCodec ra_144_decoder = { "real_144", AVMEDIA_TYPE_AUDIO, CODEC_ID_RA_144, sizeof(RA144Context), ra144_decode_init, NULL, NULL, ra144_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"), };