remove libdts decoder, we have a native dts decoder

Originally committed as revision 9051 to svn://svn.ffmpeg.org/ffmpeg/trunk

5 files changed, 0 insertions, 288 deletions

diff --git a/configure b/configure
index e541dc887f..db70fdac2e 100755
--- a/configure
+++ b/configure
@@ -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           $libfaadbin"
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 6857fa4b28..a63eec7516 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -278,7 +278,6 @@ OBJS-$(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
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 4c18f5742a..be4239618b 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -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);
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 95b26a7a3d..a86011d16b 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -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;
diff --git a/libavcodec/dtsdec.c b/libavcodec/dtsdec.c
deleted file mode 100644
index cf77a57294..0000000000
--- a/libavcodec/dtsdec.c
+++ /dev/null
@@ -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,
-};