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41f55277fa
* qatar/master: (34 commits) h264: reset h->ref_count in case of errors in ff_h264_decode_ref_pic_list_reordering() error_resilience: fix the check for missing references in ff_er_frame_end() for H264 4xm: prevent NULL dereference with invalid huffman table 4xmdemux: prevent use of uninitialized memory 4xm: clear FF_INPUT_BUFFER_PADDING_SIZE bytes in temporary buffers ptx: check for out of bound reads tiffdec: fix out of bound reads/writes eacmv: check for out of bound reads eacmv: fix potential pointer arithmetic overflows adpcm: fix out of bound reads due to integer overflow anm: prevent infinite loop avsdemux: check for out of bound writes avs: check for out of bound reads avsdemux: check for corrupted data AVOptions: refactor set_number/write_number AVOptions: cosmetics, rename static av_set_number2() to write_number(). AVOptions: cosmetics, move and rename static av_set_number(). AVOptions: split av_set_string3 into opt type-specific functions avidec: fix signed overflow in avi_sync() mxfdec: Fix some buffer overreads caused by the misuse of AVPacket related functions. ... Conflicts: Changelog configure libavcodec/ptx.c libavcodec/ra144.c libavcodec/vaapi_vc1.c libavcodec/vc1.c libavcodec/version.h libavformat/4xm.c libavformat/avidec.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
130 lines
4.3 KiB
C
130 lines
4.3 KiB
C
/*
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* Real Audio 1.0 (14.4K)
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*
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* Copyright (c) 2008 Vitor Sessak
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* Copyright (c) 2003 Nick Kurshev
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* Based on public domain decoder at http://www.honeypot.net/audio
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/intmath.h"
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#include "avcodec.h"
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#include "get_bits.h"
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#include "ra144.h"
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static av_cold int ra144_decode_init(AVCodecContext * avctx)
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{
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RA144Context *ractx = avctx->priv_data;
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ractx->avctx = avctx;
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ractx->lpc_coef[0] = ractx->lpc_tables[0];
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ractx->lpc_coef[1] = ractx->lpc_tables[1];
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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return 0;
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}
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static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs,
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int gval, GetBitContext *gb)
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{
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int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none
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int gain = get_bits(gb, 8);
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int cb1_idx = get_bits(gb, 7);
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int cb2_idx = get_bits(gb, 7);
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ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, gval,
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gain);
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}
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/** Uncompress one block (20 bytes -> 160*2 bytes). */
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static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
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int *data_size, AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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static const uint8_t sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
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unsigned int refl_rms[NBLOCKS]; // RMS of the reflection coefficients
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uint16_t block_coefs[NBLOCKS][LPC_ORDER]; // LPC coefficients of each sub-block
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unsigned int lpc_refl[LPC_ORDER]; // LPC reflection coefficients of the frame
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int i, j;
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int out_size;
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int16_t *data = vdata;
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unsigned int energy;
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RA144Context *ractx = avctx->priv_data;
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GetBitContext gb;
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out_size = NBLOCKS * BLOCKSIZE * av_get_bytes_per_sample(avctx->sample_fmt);
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if (*data_size < out_size) {
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av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
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return AVERROR(EINVAL);
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}
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if(buf_size < FRAMESIZE) {
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av_log(avctx, AV_LOG_ERROR,
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"Frame too small (%d bytes). Truncated file?\n", buf_size);
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*data_size = 0;
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return buf_size;
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}
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init_get_bits(&gb, buf, FRAMESIZE * 8);
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for (i = 0; i < LPC_ORDER; i++)
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lpc_refl[i] = ff_lpc_refl_cb[i][get_bits(&gb, sizes[i])];
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ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
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ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);
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energy = ff_energy_tab[get_bits(&gb, 5)];
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refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
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refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
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energy <= ractx->old_energy,
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ff_t_sqrt(energy*ractx->old_energy) >> 12);
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refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
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refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);
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ff_int_to_int16(block_coefs[3], ractx->lpc_coef[0]);
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for (i=0; i < NBLOCKS; i++) {
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do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb);
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for (j=0; j < BLOCKSIZE; j++)
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*data++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2);
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}
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ractx->old_energy = energy;
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ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
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FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
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*data_size = out_size;
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return FRAMESIZE;
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}
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AVCodec ff_ra_144_decoder = {
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.name = "real_144",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_RA_144,
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.priv_data_size = sizeof(RA144Context),
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.init = ra144_decode_init,
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.decode = ra144_decode_frame,
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.long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
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};
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