Mercurial > libavcodec.hg
view dtsdec.c @ 2497:69adfbbdcdeb libavcodec
- samples from mplayer ftp in the "adv" profile seem to have profile=2,
which isn't the advanced one; and indeed, using adv. profile parser fails.
Using normal parser works, and that's what is done
- attempt at taking care of stride for NORM2 bitplane decoding
- duplication of much code from msmpeg4.c; this code isn't yet used, but
goes down as far as the block layer (mainly Transform Type stuff, the
remains are wild editing without checking). Unusable yet, and lacks the AC
decoding (but a step further in bitstream parsing)
patch by anonymous
author | michael |
---|---|
date | Fri, 04 Feb 2005 02:20:38 +0000 |
parents | ef568cc0972c |
children | ef2149182f1c |
line wrap: on
line source
/* * dtsdec.c : free DTS Coherent Acoustics stream decoder. * Copyright (C) 2004 Benjamin Zores <ben@geexbox.org> * * This file is part of libavcodec. * * This library is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This library 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ #ifdef HAVE_AV_CONFIG_H #undef HAVE_AV_CONFIG_H #endif #include "avcodec.h" #include <dts.h> #include <stdlib.h> #include <string.h> #ifdef HAVE_MALLOC_H #include <malloc.h> #endif #define INBUF_SIZE 4096 #define BUFFER_SIZE 4096 #define HEADER_SIZE 14 #ifdef LIBDTS_FIXED #define CONVERT_LEVEL (1 << 26) #define CONVERT_BIAS 0 #else #define CONVERT_LEVEL 1 #define CONVERT_BIAS 384 #endif static inline int16_t convert (int32_t i) { #ifdef LIBDTS_FIXED i >>= 15; #else i -= 0x43c00000; #endif return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i); } void convert2s16_2 (sample_t * _f, int16_t * s16) { int i; int32_t * f = (int32_t *) _f; for (i = 0; i < 256; i++) { s16[2*i] = convert (f[i]); s16[2*i+1] = convert (f[i+256]); } } void convert2s16_4 (sample_t * _f, int16_t * s16) { int i; int32_t * f = (int32_t *) _f; for (i = 0; i < 256; i++) { s16[4*i] = convert (f[i]); s16[4*i+1] = convert (f[i+256]); s16[4*i+2] = convert (f[i+512]); s16[4*i+3] = convert (f[i+768]); } } void convert2s16_5 (sample_t * _f, int16_t * s16) { int i; int32_t * f = (int32_t *) _f; for (i = 0; i < 256; i++) { s16[5*i] = convert (f[i]); s16[5*i+1] = convert (f[i+256]); s16[5*i+2] = convert (f[i+512]); s16[5*i+3] = convert (f[i+768]); s16[5*i+4] = convert (f[i+1024]); } } static void convert2s16_multi (sample_t * _f, int16_t * s16, int flags) { int i; int32_t * f = (int32_t *) _f; switch (flags) { case DTS_MONO: for (i = 0; i < 256; i++) { s16[5*i] = s16[5*i+1] = s16[5*i+2] = s16[5*i+3] = 0; s16[5*i+4] = convert (f[i]); } break; case DTS_CHANNEL: case DTS_STEREO: case DTS_DOLBY: convert2s16_2 (_f, s16); break; case DTS_3F: for (i = 0; i < 256; i++) { s16[5*i] = convert (f[i]); s16[5*i+1] = convert (f[i+512]); s16[5*i+2] = s16[5*i+3] = 0; s16[5*i+4] = convert (f[i+256]); } break; case DTS_2F2R: convert2s16_4 (_f, s16); break; case DTS_3F2R: convert2s16_5 (_f, s16); break; case DTS_MONO | DTS_LFE: for (i = 0; i < 256; i++) { s16[6*i] = s16[6*i+1] = s16[6*i+2] = s16[6*i+3] = 0; s16[6*i+4] = convert (f[i+256]); s16[6*i+5] = convert (f[i]); } break; case DTS_CHANNEL | DTS_LFE: case DTS_STEREO | DTS_LFE: case DTS_DOLBY | DTS_LFE: for (i = 0; i < 256; i++) { s16[6*i] = convert (f[i+256]); s16[6*i+1] = convert (f[i+512]); s16[6*i+2] = s16[6*i+3] = s16[6*i+4] = 0; s16[6*i+5] = convert (f[i]); } break; case DTS_3F | DTS_LFE: for (i = 0; i < 256; i++) { s16[6*i] = convert (f[i+256]); s16[6*i+1] = convert (f[i+768]); s16[6*i+2] = s16[6*i+3] = 0; s16[6*i+4] = convert (f[i+512]); s16[6*i+5] = convert (f[i]); } break; case DTS_2F2R | DTS_LFE: for (i = 0; i < 256; i++) { s16[6*i] = convert (f[i+256]); s16[6*i+1] = convert (f[i+512]); s16[6*i+2] = convert (f[i+768]); s16[6*i+3] = convert (f[i+1024]); s16[6*i+4] = 0; s16[6*i+5] = convert (f[i]); } break; case DTS_3F2R | DTS_LFE: for (i = 0; i < 256; i++) { s16[6*i] = convert (f[i+256]); s16[6*i+1] = convert (f[i+768]); s16[6*i+2] = convert (f[i+1024]); s16[6*i+3] = convert (f[i+1280]); s16[6*i+4] = convert (f[i+512]); s16[6*i+5] = convert (f[i]); } break; } } static int channels_multi (int flags) { if (flags & DTS_LFE) return 6; else if (flags & 1) /* center channel */ return 5; else if ((flags & DTS_CHANNEL_MASK) == DTS_2F2R) return 4; else return 2; } static int dts_decode_frame (AVCodecContext *avctx, void *data, int *data_size, uint8_t *buff, int buff_size) { uint8_t * start = buff; uint8_t * end = buff + buff_size; static uint8_t buf[BUFFER_SIZE]; static uint8_t * bufptr = buf; static uint8_t * bufpos = buf + HEADER_SIZE; static int sample_rate; static int frame_length; static int flags; int bit_rate; int len; dts_state_t *state = avctx->priv_data; *data_size = 0; while (1) { len = end - start; if (!len) break; if (len > bufpos - bufptr) len = bufpos - bufptr; memcpy (bufptr, start, len); bufptr += len; start += len; if (bufptr == bufpos) { if (bufpos == buf + HEADER_SIZE) { int length; length = dts_syncinfo (state, buf, &flags, &sample_rate, &bit_rate, &frame_length); if (!length) { av_log (NULL, AV_LOG_INFO, "skip\n"); for (bufptr = buf; bufptr < buf + HEADER_SIZE-1; bufptr++) bufptr[0] = bufptr[1]; continue; } bufpos = buf + length; } else { level_t level; sample_t bias; int i; flags = 2; /* ???????????? */ level = CONVERT_LEVEL; bias = CONVERT_BIAS; flags |= DTS_ADJUST_LEVEL; if (dts_frame (state, buf, &flags, &level, bias)) goto error; avctx->sample_rate = sample_rate; avctx->channels = channels_multi (flags); avctx->bit_rate = bit_rate; for (i = 0; i < dts_blocks_num (state); i++) { if (dts_block (state)) goto error; { int chans; chans = channels_multi (flags); convert2s16_multi (dts_samples (state), data, flags & (DTS_CHANNEL_MASK | DTS_LFE)); data += 256 * sizeof (int16_t) * chans; *data_size += 256 * sizeof (int16_t) * chans; } } bufptr = buf; bufpos = buf + HEADER_SIZE; continue; error: av_log (NULL, AV_LOG_ERROR, "error\n"); bufptr = buf; bufpos = buf + HEADER_SIZE; } } } return buff_size; } static int dts_decode_init (AVCodecContext *avctx) { avctx->priv_data = dts_init (0); if (avctx->priv_data == NULL) return 1; return 0; } static int dts_decode_end (AVCodecContext *s) { return 0; } AVCodec dts_decoder = { "dts", CODEC_TYPE_AUDIO, CODEC_ID_DTS, sizeof (dts_state_t *), dts_decode_init, NULL, dts_decode_end, dts_decode_frame, };