echo " --enable-avisynth allow reading AVISynth script files [default=no]"
echo " --enable-libamr-nb enable libamr-nb floating point audio codec"
echo " --enable-libamr-wb enable libamr-wb floating point audio codec"
- echo " --enable-libdts enable GPLed libdts support [default=no]"
echo " --enable-libfaac enable FAAC support via libfaac [default=no]"
echo " --enable-libfaad enable FAAD support via libfaad [default=no]"
echo " --enable-libfaadbin build FAAD support with runtime linking [default=no]"
libamr
libamr_nb
libamr_wb
- libdts
libfaac
libfaad
libfaadbin
libamr_nb_encoder_deps="libamr_nb"
libamr_wb_decoder_deps="libamr_wb"
libamr_wb_encoder_deps="libamr_wb"
-libdts_decoder_deps="libdts"
libgsm_decoder_deps="libgsm"
libgsm_encoder_deps="libgsm"
libgsm_ms_decoder_deps="libgsm"
liba52bin="no"
libamr_nb="no"
libamr_wb="no"
-libdts="no"
libfaac="no"
libfaad2="no"
libfaad="no"
die_gpl_disabled "liba52" liba52
die_gpl_disabled "libxvidcore" xvid
die_gpl_disabled "x264" x264
- die_gpl_disabled "libdts" libdts
die_gpl_disabled "FAAD2" libfaad2
die_gpl_disabled "The X11 grabber" x11grab
die_gpl_disabled "The software scaler" swscaler
enabled liba52 && require liba52 a52dec/a52.h a52_init -la52
enabled libamr_nb && require libamrnb amrnb/interf_dec.h Speech_Decode_Frame_init -lamrnb -lm
enabled libamr_wb && require libamrwb amrwb/dec_if.h D_IF_init -lamrwb -lm
-enabled libdts && require libdts dts.h dts_init -ldts -lm
enabled libgsm && require libgsm gsm.h gsm_create -lgsm
enabled libmp3lame && require LAME lame/lame.h lame_init -lmp3lame -lm
enabled libtheora && require libtheora theora/theora.h theora_info_init -ltheora -logg
echo "liba52 dlopened $liba52bin"
echo "libamr-nb support $libamr_nb"
echo "libamr-wb support $libamr_wb"
-echo "libdts support $libdts"
echo "libfaac enabled $libfaac"
echo "libfaad enabled $libfaad"
echo "faadbin enabled $libfaadbin"
+++ /dev/null
-/*
- * 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "avcodec.h"
-#include <dts.h>
-
-#include <stdlib.h>
-#include <string.h>
-
-#define BUFFER_SIZE 18726
-#define HEADER_SIZE 14
-
-#define CONVERT_LEVEL 1
-#define CONVERT_BIAS 0
-
-typedef struct DTSContext {
- dts_state_t *state;
- uint8_t buf[BUFFER_SIZE];
- uint8_t *bufptr;
- uint8_t *bufpos;
-} DTSContext;
-
-static inline int16_t
-convert(sample_t s)
-{
- return s * 0x7fff;
-}
-
-static void
-convert2s16_multi(sample_t *f, int16_t *s16, int flags)
-{
- int i;
-
- switch(flags & (DTS_CHANNEL_MASK | DTS_LFE)){
- 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:
- for(i = 0; i < 256; i++){
- s16[2*i] = convert(f[i]);
- s16[2*i+1] = convert(f[i+256]);
- }
- break;
- case DTS_3F:
- for(i = 0; i < 256; i++){
- s16[5*i] = convert(f[i+256]);
- s16[5*i+1] = convert(f[i+512]);
- s16[5*i+2] = s16[5*i+3] = 0;
- s16[5*i+4] = convert(f[i]);
- }
- break;
- case DTS_2F2R:
- 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]);
- }
- break;
- case DTS_3F2R:
- for(i = 0; i < 256; i++){
- s16[5*i] = convert(f[i+256]);
- s16[5*i+1] = convert(f[i+512]);
- s16[5*i+2] = convert(f[i+768]);
- s16[5*i+3] = convert(f[i+1024]);
- s16[5*i+4] = convert(f[i]);
- }
- 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]);
- s16[6*i+5] = convert(f[i+256]);
- }
- 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]);
- s16[6*i+1] = convert(f[i+256]);
- s16[6*i+2] = s16[6*i+3] = s16[6*i+4] = 0;
- s16[6*i+5] = convert(f[i+512]);
- }
- 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+512]);
- s16[6*i+2] = s16[6*i+3] = 0;
- s16[6*i+4] = convert(f[i]);
- s16[6*i+5] = convert(f[i+768]);
- }
- break;
- case DTS_2F2R | DTS_LFE:
- for(i = 0; i < 256; i++){
- s16[6*i] = convert(f[i]);
- s16[6*i+1] = convert(f[i+256]);
- s16[6*i+2] = convert(f[i+512]);
- s16[6*i+3] = convert(f[i+768]);
- s16[6*i+4] = 0;
- s16[6*i+5] = convert(f[i+1024]);
- }
- 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+512]);
- s16[6*i+2] = convert(f[i+768]);
- s16[6*i+3] = convert(f[i+1024]);
- s16[6*i+4] = convert(f[i]);
- s16[6*i+5] = convert(f[i+1280]);
- }
- break;
- }
-}
-
-static int
-channels_multi(int flags)
-{
- switch(flags & (DTS_CHANNEL_MASK | DTS_LFE)){
- case DTS_CHANNEL:
- case DTS_STEREO:
- case DTS_DOLBY:
- return 2;
- case DTS_2F2R:
- return 4;
- case DTS_MONO:
- case DTS_3F:
- case DTS_3F2R:
- return 5;
- case DTS_MONO | DTS_LFE:
- case DTS_CHANNEL | DTS_LFE:
- case DTS_STEREO | DTS_LFE:
- case DTS_DOLBY | DTS_LFE:
- case DTS_3F | DTS_LFE:
- case DTS_2F2R | DTS_LFE:
- case DTS_3F2R | DTS_LFE:
- return 6;
- }
-
- return -1;
-}
-
-static int
-dts_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
- uint8_t * buff, int buff_size)
-{
- DTSContext *s = avctx->priv_data;
- uint8_t *start = buff;
- uint8_t *end = buff + buff_size;
- int16_t *out_samples = data;
- int sample_rate;
- int frame_length;
- int flags;
- int bit_rate;
- int len;
- level_t level;
- sample_t bias;
- int nblocks;
- int i;
-
- *data_size = 0;
-
- while(1) {
- int length;
-
- len = end - start;
- if(!len)
- break;
- if(len > s->bufpos - s->bufptr)
- len = s->bufpos - s->bufptr;
- memcpy(s->bufptr, start, len);
- s->bufptr += len;
- start += len;
- if(s->bufptr != s->bufpos)
- return start - buff;
- if(s->bufpos != s->buf + HEADER_SIZE)
- break;
-
- length = dts_syncinfo(s->state, s->buf, &flags, &sample_rate,
- &bit_rate, &frame_length);
- if(!length) {
- av_log(NULL, AV_LOG_INFO, "skip\n");
- for(s->bufptr = s->buf; s->bufptr < s->buf + HEADER_SIZE - 1; s->bufptr++)
- s->bufptr[0] = s->bufptr[1];
- continue;
- }
- s->bufpos = s->buf + length;
- }
-
- level = CONVERT_LEVEL;
- bias = CONVERT_BIAS;
-
- flags |= DTS_ADJUST_LEVEL;
- if(dts_frame(s->state, s->buf, &flags, &level, bias)) {
- av_log(avctx, AV_LOG_ERROR, "dts_frame() failed\n");
- goto end;
- }
-
- avctx->sample_rate = sample_rate;
- avctx->channels = channels_multi(flags);
- avctx->bit_rate = bit_rate;
-
- nblocks = dts_blocks_num(s->state);
-
- for(i = 0; i < nblocks; i++) {
- if(dts_block(s->state)) {
- av_log(avctx, AV_LOG_ERROR, "dts_block() failed\n");
- goto end;
- }
-
- convert2s16_multi(dts_samples(s->state), out_samples, flags);
-
- out_samples += 256 * avctx->channels;
- *data_size += 256 * sizeof(int16_t) * avctx->channels;
- }
-
-end:
- s->bufptr = s->buf;
- s->bufpos = s->buf + HEADER_SIZE;
- return start - buff;
-}
-
-static int
-dts_decode_init(AVCodecContext * avctx)
-{
- DTSContext *s = avctx->priv_data;
- s->bufptr = s->buf;
- s->bufpos = s->buf + HEADER_SIZE;
- s->state = dts_init(0);
- if(s->state == NULL)
- return -1;
-
- return 0;
-}
-
-static int
-dts_decode_end(AVCodecContext * avctx)
-{
- DTSContext *s = avctx->priv_data;
- dts_free(s->state);
- return 0;
-}
-
-AVCodec libdts_decoder = {
- "libdts",
- CODEC_TYPE_AUDIO,
- CODEC_ID_DTS,
- sizeof(DTSContext),
- dts_decode_init,
- NULL,
- dts_decode_end,
- dts_decode_frame,
-};
projects / frescor / ffmpeg.git / commitdiff