Mercurial > libavcodec.hg
view dtsdec.c @ 4166:eced83504436 libavcodec
mp3 header (de)compression bitstream filter
this will make mp3 frames 4 bytes smaller, it will not give you binary identical mp3 files, but it will give you mp3 files which decode to binary identical output
this will only work in containers providing at least packet size, sample_rate and number of channels
bugreports about mp3 files for which this fails are welcome
and this is experimental (dont expect compatibility and dont even expect to be able to decompress what you compressed, hell dont even expect this to work without editing the source a little)
author | michael |
---|---|
date | Fri, 10 Nov 2006 01:41:53 +0000 |
parents | 418b123f1b74 |
children | f97a2081b5b1 |
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/* * dtsdec.c : free DTS Coherent Acoustics stream decoder. * Copyright (C) 2004 Benjamin Zores <ben@geexbox.org> * * This file is part of FFmpeg. * * FFmpeg 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. * * 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with FFmpeg; 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 BUFFER_SIZE 18726 #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) return start - buff; if (bufpos != buf + HEADER_SIZE) break; { 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; } } { 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; return start-buff; error: av_log (NULL, AV_LOG_ERROR, "error\n"); bufptr = buf; bufpos = buf + HEADER_SIZE; } return start-buff; } 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, };