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remove libdts decoder, we have a native dts decoder

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Originally committed as revision 9051 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Aurelien Jacobs 2007-05-17 17:04:00 +00:00
parent facbea9596
commit 670a6b133b
5 changed files with 0 additions and 288 deletions
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7
configure vendored
View File

@ -85,7 +85,6 @@ show_help(){
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]"
@ -578,7 +577,6 @@ CONFIG_LIST='
libamr
libamr_nb
libamr_wb
libdts
libfaac
libfaad
libfaadbin
@ -710,7 +708,6 @@ libamr_nb_decoder_deps="libamr_nb"
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"
@ -826,7 +823,6 @@ liba52="no"
liba52bin="no"
libamr_nb="no"
libamr_wb="no"
libdts="no"
libfaac="no"
libfaad2="no"
libfaad="no"
@ -1282,7 +1278,6 @@ if disabled gpl ; then
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
@ -1619,7 +1614,6 @@ enabled_any libamr_nb libamr_wb && enable libamr
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
@ -1881,7 +1875,6 @@ echo "liba52 support $liba52"
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"

1
libavcodec/Makefile
View File

@ -278,7 +278,6 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER) += adpcm.o
# external codec libraries
OBJS-$(CONFIG_LIBAMR) += amr.o
OBJS-$(CONFIG_LIBA52) += a52dec.o
OBJS-$(CONFIG_LIBDTS) += dtsdec.o
OBJS-$(CONFIG_LIBFAAC) += faac.o
OBJS-$(CONFIG_LIBFAAD) += faad.o
OBJS-$(CONFIG_LIBGSM) += libgsm.o

1
libavcodec/allcodecs.c
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@ -178,7 +178,6 @@ void avcodec_register_all(void)
REGISTER_ENCDEC (LIBAMR_NB, libamr_nb);
REGISTER_ENCDEC (LIBAMR_WB, libamr_wb);
REGISTER_DECODER(LIBA52, liba52);
REGISTER_DECODER(LIBDTS, libdts);
REGISTER_ENCDEC (LIBGSM, libgsm);
REGISTER_ENCDEC (LIBGSM_MS, libgsm_ms);
REGISTER_ENCODER(LIBTHEORA, libtheora);

1
libavcodec/avcodec.h
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@ -2428,7 +2428,6 @@ extern AVCodec libamr_nb_decoder;
extern AVCodec libamr_nb_encoder;
extern AVCodec libamr_wb_decoder;
extern AVCodec libamr_wb_encoder;
extern AVCodec libdts_decoder;
extern AVCodec libgsm_decoder;
extern AVCodec libgsm_encoder;
extern AVCodec libgsm_ms_decoder;

278
libavcodec/dtsdec.c
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@ -1,278 +0,0 @@
/*
* 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,
};