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author | Francesco Lavra <francescolavra@interfree.it> | 2010-06-11 08:01:51 +0000 |
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committer | Vitor Sessak <vitor1001@gmail.com> | 2010-06-11 08:01:51 +0000 |
commit | 0acaf8b0f219470d843cb2e8f1943c6e88ad76bc (patch) | |
tree | f20c66748fa379f9db0d335438d01ddc91895409 /libavcodec/ra144dec.c | |
parent | ede13f55f00dd6c56897fd4bc88c202b2f087455 (diff) | |
download | ffmpeg-0acaf8b0f219470d843cb2e8f1943c6e88ad76bc.tar.gz |
Split ra144.c in common code (to be shared with the future encoder) and
decoder.
Patch by Francesco Lavra (firstnamelastname@interfree.it)
Originally committed as revision 23573 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/ra144dec.c')
-rw-r--r-- | libavcodec/ra144dec.c | 154 |
1 files changed, 154 insertions, 0 deletions
diff --git a/libavcodec/ra144dec.c b/libavcodec/ra144dec.c new file mode 100644 index 0000000000..b47201ad17 --- /dev/null +++ b/libavcodec/ra144dec.c @@ -0,0 +1,154 @@ +/* + * 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" +#include "celp_filters.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) +{ + uint16_t buffer_a[40]; + uint16_t *block; + 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); + int m[3]; + + if (cba_idx) { + cba_idx += BLOCKSIZE/2 - 1; + copy_and_dup(buffer_a, ractx->adapt_cb, cba_idx); + m[0] = (irms(buffer_a) * gval) >> 12; + } else { + m[0] = 0; + } + + m[1] = (cb1_base[cb1_idx] * gval) >> 8; + m[2] = (cb2_base[cb2_idx] * gval) >> 8; + + memmove(ractx->adapt_cb, ractx->adapt_cb + BLOCKSIZE, + (BUFFERSIZE - BLOCKSIZE) * sizeof(*ractx->adapt_cb)); + + block = ractx->adapt_cb + BUFFERSIZE - BLOCKSIZE; + + add_wav(block, gain, cba_idx, m, cba_idx? buffer_a: NULL, + cb1_vects[cb1_idx], cb2_vects[cb2_idx]); + + memcpy(ractx->curr_sblock, ractx->curr_sblock + 40, + 10*sizeof(*ractx->curr_sblock)); + + if (ff_celp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs, + block, BLOCKSIZE, 10, 1, 0xfff)) + memset(ractx->curr_sblock, 0, 50*sizeof(*ractx->curr_sblock)); +} + +/** 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] = lpc_refl_cb[i][get_bits(&gb, sizes[i])]; + + eval_coefs(ractx->lpc_coef[0], lpc_refl); + ractx->lpc_refl_rms[0] = rms(lpc_refl); + + energy = energy_tab[get_bits(&gb, 5)]; + + refl_rms[0] = interp(ractx, block_coefs[0], 1, 1, ractx->old_energy); + refl_rms[1] = interp(ractx, block_coefs[1], 2, energy <= ractx->old_energy, + t_sqrt(energy*ractx->old_energy) >> 12); + refl_rms[2] = interp(ractx, block_coefs[2], 3, 0, energy); + refl_rms[3] = rescale_rms(ractx->lpc_refl_rms[0], energy); + + 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)"), +}; |