remove dts_internal.h

avoiding code redundance
license copy paste fix
patch by (Benjamin Zores <ben at geexbox dot org>)

Originally committed as revision 3315 to svn://svn.ffmpeg.org/ffmpeg/trunk

libavcodec/dts_internal.h

-/*
- * dts_internal.h
- * Copyright (C) 2004 Gildas Bazin <gbazin@videolan.org>
- * Copyright (C) 2000-2003 Michel Lespinasse <walken@zoy.org>
- * Copyright (C) 1999-2000 Aaron Holtzman <aholtzma@ess.engr.uvic.ca>
- *
- * This file is part of dtsdec, a free DTS Coherent Acoustics stream decoder.
- * See http://www.videolan.org/dtsdec.html for updates.
- *
- * dtsdec 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.
- *
- * dtsdec 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#define DTS_SUBFRAMES_MAX (16)
-#define DTS_PRIM_CHANNELS_MAX (5)
-#define DTS_SUBBANDS (32)
-#define DTS_ABITS_MAX (32) /* Should be 28 */
-#define DTS_SUBSUBFAMES_MAX (4)
-#define DTS_LFE_MAX (3)
-
-struct dts_state_s {
-
-    /* Frame header */
-    int frame_type;             /* type of the current frame */
-    int samples_deficit;        /* deficit sample count */
-    int crc_present;            /* crc is present in the bitstream */
-    int sample_blocks;          /* number of PCM sample blocks */
-    int frame_size;             /* primary frame byte size */
-    int amode;                  /* audio channels arrangement */
-    int sample_rate;            /* audio sampling rate */
-    int bit_rate;               /* transmission bit rate */
-
-    int downmix;                /* embedded downmix enabled */
-    int dynrange;               /* embedded dynamic range flag */
-    int timestamp;              /* embedded time stamp flag */
-    int aux_data;               /* auxiliary data flag */
-    int hdcd;                   /* source material is mastered in HDCD */
-    int ext_descr;              /* extension audio descriptor flag */
-    int ext_coding;             /* extended coding flag */
-    int aspf;                   /* audio sync word insertion flag */
-    int lfe;                    /* low frequency effects flag */
-    int predictor_history;      /* predictor history flag */
-    int header_crc;             /* header crc check bytes */
-    int multirate_inter;        /* multirate interpolator switch */
-    int version;                /* encoder software revision */
-    int copy_history;           /* copy history */
-    int source_pcm_res;         /* source pcm resolution */
-    int front_sum;              /* front sum/difference flag */
-    int surround_sum;           /* surround sum/difference flag */
-    int dialog_norm;            /* dialog normalisation parameter */
-
-    /* Primary audio coding header */
-    int subframes;              /* number of subframes */
-    int prim_channels;          /* number of primary audio channels */
-    /* subband activity count */
-    int subband_activity[DTS_PRIM_CHANNELS_MAX];
-    /* high frequency vq start subband */
-    int vq_start_subband[DTS_PRIM_CHANNELS_MAX];
-    /* joint intensity coding index */
-    int joint_intensity[DTS_PRIM_CHANNELS_MAX];
-    /* transient mode code book */
-    int transient_huffman[DTS_PRIM_CHANNELS_MAX];
-    /* scale factor code book */
-    int scalefactor_huffman[DTS_PRIM_CHANNELS_MAX];
-    /* bit allocation quantizer select */
-    int bitalloc_huffman[DTS_PRIM_CHANNELS_MAX];
-    /* quantization index codebook select */
-    int quant_index_huffman[DTS_PRIM_CHANNELS_MAX][DTS_ABITS_MAX];
-    /* scale factor adjustment */
-    float scalefactor_adj[DTS_PRIM_CHANNELS_MAX][DTS_ABITS_MAX];
-
-    /* Primary audio coding side information */
-    int subsubframes;           /* number of subsubframes */
-    int partial_samples;        /* partial subsubframe samples count */
-    /* prediction mode (ADPCM used or not) */
-    int prediction_mode[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
-    /* prediction VQ coefs */
-    int prediction_vq[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
-    /* bit allocation index */
-    int bitalloc[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
-    /* transition mode (transients) */
-    int transition_mode[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
-    /* scale factors (2 if transient)*/
-    int scale_factor[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS][2];
-    /* joint subband scale factors codebook */
-    int joint_huff[DTS_PRIM_CHANNELS_MAX];
-    /* joint subband scale factors */
-    int joint_scale_factor[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
-    /* stereo downmix coefficients */
-    int downmix_coef[DTS_PRIM_CHANNELS_MAX][2];
-    /* dynamic range coefficient */
-    int dynrange_coef;
-
-    /* VQ encoded high frequency subbands */
-    int high_freq_vq[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
-
-    /* Low frequency effect data */
-    double lfe_data[2*DTS_SUBSUBFAMES_MAX*DTS_LFE_MAX * 2 /*history*/];
-    int lfe_scale_factor;
-
-    /* Subband samples history (for ADPCM) */
-    double subband_samples_hist[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS][4];
-    double subband_fir_hist[DTS_PRIM_CHANNELS_MAX][512];
-    double subband_fir_noidea[DTS_PRIM_CHANNELS_MAX][64];
-
-    /* Audio output */
-    level_t clev;            /* centre channel mix level */
-    level_t slev;            /* surround channels mix level */
-
-    int output;              /* type of output */
-    level_t level;           /* output level */
-    sample_t bias;           /* output bias */
-
-    sample_t * samples;      /* pointer to the internal audio samples buffer */
-    int downmixed;
-
-    int dynrnge;             /* apply dynamic range */
-    level_t dynrng;          /* dynamic range */
-    void * dynrngdata;       /* dynamic range callback funtion and data */
-    level_t (* dynrngcall) (level_t range, void * dynrngdata);
-
-    /* Bitstream handling */
-    uint32_t * buffer_start;
-    uint32_t bits_left;
-    uint32_t current_word;
-    int      word_mode;         /* 16/14 bits word format (1 -> 16, 0 -> 14) */
-    int      bigendian_mode;    /* endianness (1 -> be, 0 -> le) */
-
-    /* Current position in DTS frame */
-    int current_subframe;
-    int current_subsubframe;
-
-    /* Pre-calculated cosine modulation coefs for the QMF */
-    double cos_mod[544];
-
-    /* Debug flag */
-    int debug_flag;
-};
-
-#define LEVEL_PLUS6DB 2.0
-#define LEVEL_PLUS3DB 1.4142135623730951
-#define LEVEL_3DB 0.7071067811865476
-#define LEVEL_45DB 0.5946035575013605
-#define LEVEL_6DB 0.5
-
-int dts_downmix_init (int input, int flags, level_t * level,
-                      level_t clev, level_t slev);
-int dts_downmix_coeff (level_t * coeff, int acmod, int output, level_t level,
-                       level_t clev, level_t slev);
-void dts_downmix (sample_t * samples, int acmod, int output, sample_t bias,
-                  level_t clev, level_t slev);
-void dts_upmix (sample_t * samples, int acmod, int output);
-
-#define ROUND(x) ((int)((x) + ((x) > 0 ? 0.5 : -0.5)))
-
-#ifndef LIBDTS_FIXED
-
-typedef sample_t quantizer_t;
-#define SAMPLE(x) (x)
-#define LEVEL(x) (x)
-#define MUL(a,b) ((a) * (b))
-#define MUL_L(a,b) ((a) * (b))
-#define MUL_C(a,b) ((a) * (b))
-#define DIV(a,b) ((a) / (b))
-#define BIAS(x) ((x) + bias)
-
-#else /* LIBDTS_FIXED */
-
-typedef int16_t quantizer_t;
-#define SAMPLE(x) (sample_t)((x) * (1 << 30))
-#define LEVEL(x) (level_t)((x) * (1 << 26))
-
-#if 0
-#define MUL(a,b) ((int)(((int64_t)(a) * (b) + (1 << 29)) >> 30))
-#define MUL_L(a,b) ((int)(((int64_t)(a) * (b) + (1 << 25)) >> 26))
-#elif 1
-#define MUL(a,b) \
-({ int32_t _ta=(a), _tb=(b), _tc; \
-   _tc=(_ta & 0xffff)*(_tb >> 16)+(_ta >> 16)*(_tb & 0xffff); (int32_t)(((_tc >> 14))+ (((_ta >> 16)*(_tb >> 16)) << 2 )); })
-#define MUL_L(a,b) \
-({ int32_t _ta=(a), _tb=(b), _tc; \
-   _tc=(_ta & 0xffff)*(_tb >> 16)+(_ta >> 16)*(_tb & 0xffff); (int32_t)((_tc >> 10) + (((_ta >> 16)*(_tb >> 16)) << 6)); })
-#else
-#define MUL(a,b) (((a) >> 15) * ((b) >> 15))
-#define MUL_L(a,b) (((a) >> 13) * ((b) >> 13))
-#endif
-
-#define MUL_C(a,b) MUL_L (a, LEVEL (b))
-#define DIV(a,b) ((((int64_t)LEVEL (a)) << 26) / (b))
-#define BIAS(x) (x)
-
-#endif

libavcodec/dtsdec.c

@@ -4,19 +4,19 @@
  *
  * 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 Lesser 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 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
- * Lesser General Public License for more details.
+ * 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 Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ * 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
@@ -25,12 +25,10 @@
 
 #include "avcodec.h"
 #include <dts.h>
-#include "dts_internal.h"
 
 #include <stdlib.h>
 #include <string.h>
 #include <malloc.h>
-#include <math.h>
 
 #define INBUF_SIZE 4096
 #define BUFFER_SIZE 4096
@@ -44,26 +42,6 @@
 #define CONVERT_BIAS 384
 #endif
 
-static void
-pre_calc_cosmod (dts_state_t * state)
-{
-  int i, j, k;
-
-  for (j=0,k=0;k<16;k++)
-    for (i=0;i<16;i++)
-      state->cos_mod[j++] = cos((2*i+1)*(2*k+1)*M_PI/64);
-
-  for (k=0;k<16;k++)
-    for (i=0;i<16;i++)
-      state->cos_mod[j++] = cos((i)*(2*k+1)*M_PI/32);
-
-  for (k=0;k<16;k++)
-    state->cos_mod[j++] = 0.25/(2*cos((2*k+1)*M_PI/128));
-
-  for (k=0;k<16;k++)
-    state->cos_mod[j++] = -0.25/(2.0*sin((2*k+1)*M_PI/128));
-}
-
 static inline
 int16_t convert (int32_t i)
 {
@@ -311,23 +289,10 @@ dts_decode_frame (AVCodecContext *avctx, void *data, int *data_size,
 static int
 dts_decode_init (AVCodecContext *avctx)
 {
-  dts_state_t * state;
-  int i;
-
-  state = avctx->priv_data;
-  memset (state, 0, sizeof (dts_state_t));
-
-  state->samples = (sample_t *) memalign (16, 256 * 12 * sizeof (sample_t));
-  if (state->samples == NULL)
+  avctx->priv_data = dts_init (0);
+  if (avctx->priv_data == NULL)
     return 1;
 
-  for (i = 0; i < 256 * 12; i++)
-    state->samples[i] = 0;
-
-  /* Pre-calculate cosine modulation coefficients */
-  pre_calc_cosmod (state);
-  state->downmixed = 1;
-
   return 0;
 }
 
@@ -341,7 +306,7 @@ AVCodec dts_decoder = {
   "dts", 
   CODEC_TYPE_AUDIO,
   CODEC_ID_DTS,
-  sizeof (dts_state_t),
+  sizeof (dts_state_t *),
   dts_decode_init,
   NULL,
   dts_decode_end,