libmpcodecs/ad_liba52.c

   1 /*
   2  * This file is part of MPlayer.
   3  *
   4  * MPlayer is free software; you can redistribute it and/or modify
   5  * it under the terms of the GNU General Public License as published by
   6  * the Free Software Foundation; either version 2 of the License, or
   7  * (at your option) any later version.
   8  *
   9  * MPlayer is distributed in the hope that it will be useful,
  10  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  11  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  12  * GNU General Public License for more details.
  13  *
  14  * You should have received a copy of the GNU General Public License along
  15  * with MPlayer; if not, write to the Free Software Foundation, Inc.,
  16  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  17  */
  18
  19 #define _XOPEN_SOURCE 600
  20 #include <stdio.h>
  21 #include <stdlib.h>
  22 #include <unistd.h>
  23 #include <math.h>
  24 #include <assert.h>
  25
  26 #include "config.h"
  27 #include "options.h"
  28 #include "mp_msg.h"
  29 #include "mpbswap.h"
  30
  31 #include "ad_internal.h"
  32
  33 #include "cpudetect.h"
  34
  35 #include "libaf/af_format.h"
  36
  37 #include <a52dec/a52.h>
  38 #include <a52dec/mm_accel.h>
  39 int (* a52_resample) (float * _f, int16_t * s16);
  40
  41 static a52_state_t *a52_state;
  42 static uint32_t a52_flags=0;
  43 /** Used by a52_resample_float, it defines the mapping between liba52
  44  * channels and output channels.  The ith nibble from the right in the
  45  * hex representation of channel_map is the index of the source
  46  * channel corresponding to the ith output channel.  Source channels are
  47  * indexed 1-6.  Silent output channels are marked by 0xf. */
  48 static uint32_t channel_map;
  49
  50 #define DRC_NO_ACTION      0
  51 #define DRC_NO_COMPRESSION 1
  52 #define DRC_CALLBACK       2
  53
  54 /** The output is multiplied by this var.  Used for volume control */
  55 static sample_t a52_level = 1;
  56 static int a52_drc_action = DRC_NO_ACTION;
  57
  58 static const ad_info_t info =
  59 {
  60         "AC3 decoding with liba52",
  61         "liba52",
  62         "Nick Kurshev",
  63         "Michel LESPINASSE",
  64         ""
  65 };
  66
  67 LIBAD_EXTERN(liba52)
  68
  69 static int a52_fillbuff(sh_audio_t *sh_audio)
  70 {
  71 int length=0;
  72 int flags=0;
  73 int sample_rate=0;
  74 int bit_rate=0;
  75
  76     sh_audio->a_in_buffer_len=0;
  77     /* sync frame:*/
  78 while(1){
  79     while(sh_audio->a_in_buffer_len<8){
  80         int c=demux_getc(sh_audio->ds);
  81         if(c<0) return -1; /* EOF*/
  82         sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++]=c;
  83     }
  84     if(sh_audio->format==MKTAG('d','n','e','t')) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8);
  85     length = a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
  86     if(length>=7 && length<=3840) break; /* we're done.*/
  87     /* bad file => resync*/
  88     if(sh_audio->format==MKTAG('d','n','e','t')) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8);
  89     memmove(sh_audio->a_in_buffer,sh_audio->a_in_buffer+1,7);
  90     --sh_audio->a_in_buffer_len;
  91 }
  92     mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"a52: len=%d  flags=0x%X  %d Hz %d bit/s\n",length,flags,sample_rate,bit_rate);
  93     sh_audio->samplerate=sample_rate;
  94     sh_audio->i_bps=bit_rate/8;
  95     sh_audio->samplesize=sh_audio->sample_format==AF_FORMAT_FLOAT_NE ? 4 : 2;
  96     demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+8,length-8);
  97     if(sh_audio->format==MKTAG('d','n','e','t'))
  98         swab(sh_audio->a_in_buffer+8,sh_audio->a_in_buffer+8,length-8);
  99
 100 #ifdef CONFIG_LIBA52_INTERNAL
 101     if(crc16_block(sh_audio->a_in_buffer+2,length-2)!=0)
 102         mp_msg(MSGT_DECAUDIO,MSGL_STATUS,"a52: CRC check failed!  \n");
 103 #endif
 104
 105     return length;
 106 }
 107
 108 /* returns: number of available channels*/
 109 static int a52_printinfo(sh_audio_t *sh_audio){
 110 int flags, sample_rate, bit_rate;
 111 char* mode="unknown";
 112 int channels=0;
 113   a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
 114   switch(flags&A52_CHANNEL_MASK){
 115     case A52_CHANNEL: mode="channel"; channels=2; break;
 116     case A52_MONO: mode="mono"; channels=1; break;
 117     case A52_STEREO: mode="stereo"; channels=2; break;
 118     case A52_3F: mode="3f";channels=3;break;
 119     case A52_2F1R: mode="2f+1r";channels=3;break;
 120     case A52_3F1R: mode="3f+1r";channels=4;break;
 121     case A52_2F2R: mode="2f+2r";channels=4;break;
 122     case A52_3F2R: mode="3f+2r";channels=5;break;
 123     case A52_CHANNEL1: mode="channel1"; channels=2; break;
 124     case A52_CHANNEL2: mode="channel2"; channels=2; break;
 125     case A52_DOLBY: mode="dolby"; channels=2; break;
 126   }
 127   mp_msg(MSGT_DECAUDIO,MSGL_V,"AC3: %d.%d (%s%s)  %d Hz  %3.1f kbit/s\n",
 128         channels, (flags&A52_LFE)?1:0,
 129         mode, (flags&A52_LFE)?"+lfe":"",
 130         sample_rate, bit_rate*0.001f);
 131   return (flags&A52_LFE) ? (channels+1) : channels;
 132 }
 133
 134 static sample_t dynrng_call (sample_t c, void *data)
 135 {
 136     struct MPOpts *opts = data;
 137     //fprintf(stderr, "(%f, %f): %f\n", (double)c, (double)drc_level, (double)pow((double)c, drc_level));
 138     //fprintf(stderr, "(%lf, %lf): %lf\n", (double)c, opts->drc_level, pow(c, opts->drc_level));
 139     return pow(c, opts->drc_level);
 140 }
 141
 142
 143 static int preinit(sh_audio_t *sh)
 144 {
 145   struct MPOpts *opts = sh->opts;
 146   /* Dolby AC3 audio: */
 147   /* however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame */
 148   if (sh->samplesize < 4) sh->samplesize = 4;
 149   sh->audio_out_minsize=opts->audio_output_channels*sh->samplesize*256*6;
 150   sh->audio_in_minsize=3840;
 151   a52_level = 1.0;
 152   return 1;
 153 }
 154
 155 /**
 156  * \brief Function to convert the "planar" float format used by liba52
 157  * into the interleaved float format used by libaf/libao2.
 158  * \param in the input buffer containing the planar samples.
 159  * \param out the output buffer where the interleaved result is stored.
 160  */
 161 static int a52_resample_float(float *in, int16_t *out)
 162 {
 163     unsigned long i;
 164     float *p = (float*) out;
 165     for (i = 0; i != 256; i++) {
 166         unsigned long map = channel_map;
 167         do {
 168             unsigned long ch = map & 15;
 169             if (ch == 15)
 170                 *p = 0;
 171             else
 172                 *p = in[i + ((ch-1)<<8)];
 173             p++;
 174         } while ((map >>= 4));
 175     }
 176     return (int16_t*) p - out;
 177 }
 178
 179 static int init(sh_audio_t *sh_audio)
 180 {
 181     struct MPOpts *opts = sh_audio->opts;
 182   uint32_t a52_accel=0;
 183   sample_t level=a52_level, bias=384;
 184   int flags=0;
 185   /* Dolby AC3 audio:*/
 186 #ifdef MM_ACCEL_X86_SSE
 187   if(gCpuCaps.hasSSE) a52_accel|=MM_ACCEL_X86_SSE;
 188 #endif
 189   if(gCpuCaps.hasMMX) a52_accel|=MM_ACCEL_X86_MMX;
 190   if(gCpuCaps.hasMMX2) a52_accel|=MM_ACCEL_X86_MMXEXT;
 191   if(gCpuCaps.has3DNow) a52_accel|=MM_ACCEL_X86_3DNOW;
 192 #ifdef MM_ACCEL_X86_3DNOWEXT
 193   if(gCpuCaps.has3DNowExt) a52_accel|=MM_ACCEL_X86_3DNOWEXT;
 194 #endif
 195 #ifdef MM_ACCEL_PPC_ALTIVEC
 196   if(gCpuCaps.hasAltiVec) a52_accel|=MM_ACCEL_PPC_ALTIVEC;
 197 #endif
 198   a52_state=a52_init (a52_accel);
 199   if (a52_state == NULL) {
 200         mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n");
 201         return 0;
 202   }
 203   sh_audio->sample_format = AF_FORMAT_FLOAT_NE;
 204   if(a52_fillbuff(sh_audio)<0){
 205         mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n");
 206         return 0;
 207   }
 208
 209   /* Init a52 dynrng */
 210   if (opts->drc_level < 0.001) {
 211           /* level == 0 --> no compression, init library without callback */
 212           a52_drc_action = DRC_NO_COMPRESSION;
 213   } else if (opts->drc_level > 0.999 && opts->drc_level < 1.001) {
 214           /* level == 1 --> full compression, do nothing at all (library default = full compression) */
 215           a52_drc_action = DRC_NO_ACTION;
 216   } else {
 217           a52_drc_action = DRC_CALLBACK;
 218   }
 219   /* Library init for dynrng has to be done for each frame, see decode_audio() */
 220
 221
 222   /* 'a52 cannot upmix' hotfix:*/
 223   a52_printinfo(sh_audio);
 224   sh_audio->channels=opts->audio_output_channels;
 225 while(sh_audio->channels>0){
 226   switch(sh_audio->channels){
 227             case 1: a52_flags=A52_MONO; break;
 228 /*          case 2: a52_flags=A52_STEREO; break;*/
 229             case 2: a52_flags=A52_DOLBY; break;
 230 /*          case 3: a52_flags=A52_3F; break;*/
 231             case 3: a52_flags=A52_2F1R; break;
 232             case 4: a52_flags=A52_2F2R; break; /* 2+2*/
 233             case 5: a52_flags=A52_3F2R; break;
 234             case 6: a52_flags=A52_3F2R|A52_LFE; break; /* 5.1*/
 235   }
 236   /* test:*/
 237   flags=a52_flags|A52_ADJUST_LEVEL;
 238   mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags before a52_frame: 0x%X\n",flags);
 239   if (a52_frame (a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
 240     mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: error decoding frame -> nosound\n");
 241     return 0;
 242   }
 243   mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags);
 244   /* frame decoded, let's init resampler:*/
 245   channel_map = 0;
 246   if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE) {
 247       if (!(flags & A52_LFE)) {
 248           switch ((flags<<3) | sh_audio->channels) {
 249             case (A52_MONO    << 3) | 1: channel_map = 0x1; break;
 250             case (A52_CHANNEL << 3) | 2:
 251             case (A52_STEREO  << 3) | 2:
 252             case (A52_DOLBY   << 3) | 2: channel_map =    0x21; break;
 253             case (A52_2F1R    << 3) | 3: channel_map =   0x321; break;
 254             case (A52_2F2R    << 3) | 4: channel_map =  0x4321; break;
 255             case (A52_3F      << 3) | 5: channel_map = 0x2ff31; break;
 256             case (A52_3F2R    << 3) | 5: channel_map = 0x25431; break;
 257           }
 258       } else if (sh_audio->channels == 6) {
 259           switch (flags & ~A52_LFE) {
 260             case A52_MONO   : channel_map = 0x12ffff; break;
 261             case A52_CHANNEL:
 262             case A52_STEREO :
 263             case A52_DOLBY  : channel_map = 0x1fff32; break;
 264             case A52_3F     : channel_map = 0x13ff42; break;
 265             case A52_2F1R   : channel_map = 0x1f4432; break;
 266             case A52_2F2R   : channel_map = 0x1f5432; break;
 267             case A52_3F2R   : channel_map = 0x136542; break;
 268           }
 269       }
 270       if (channel_map) {
 271           a52_resample = a52_resample_float;
 272           break;
 273       }
 274   } else
 275   break;
 276 }
 277   if(sh_audio->channels<=0){
 278     mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: no resampler. try different channel setup!\n");
 279     return 0;
 280   }
 281   return 1;
 282 }
 283
 284 static void uninit(sh_audio_t *sh)
 285 {
 286   a52_free(a52_state);
 287 }
 288
 289 static int control(sh_audio_t *sh,int cmd,void* arg, ...)
 290 {
 291     switch(cmd)
 292     {
 293       case ADCTRL_RESYNC_STREAM:
 294       case ADCTRL_SKIP_FRAME:
 295           a52_fillbuff(sh);
 296           return CONTROL_TRUE;
 297       case ADCTRL_SET_VOLUME: {
 298           float vol = *(float*)arg;
 299           if (vol > 60.0) vol = 60.0;
 300           a52_level = vol <= -200.0 ? 0 : pow(10.0,vol/20.0);
 301           return CONTROL_TRUE;
 302       }
 303       case ADCTRL_QUERY_FORMAT:
 304           if (*(int*)arg == AF_FORMAT_S16_NE ||
 305               *(int*)arg == AF_FORMAT_FLOAT_NE)
 306               return CONTROL_TRUE;
 307           return CONTROL_FALSE;
 308     }
 309   return CONTROL_UNKNOWN;
 310 }
 311
 312 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
 313 {
 314     sample_t level=a52_level, bias=384;
 315     int flags=a52_flags|A52_ADJUST_LEVEL;
 316     int i,len=-1;
 317         if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE)
 318             bias = 0;
 319         if(!sh_audio->a_in_buffer_len)
 320             if(a52_fillbuff(sh_audio)<0) return len; /* EOF */
 321         sh_audio->a_in_buffer_len=0;
 322         if (a52_frame (a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
 323             mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error decoding frame\n");
 324             return len;
 325         }
 326
 327         /* handle dynrng */
 328         if (a52_drc_action != DRC_NO_ACTION) {
 329             if (a52_drc_action == DRC_NO_COMPRESSION)
 330                 a52_dynrng(a52_state, NULL, NULL);
 331             else
 332                 a52_dynrng(a52_state, dynrng_call, sh_audio->opts);
 333         }
 334
 335         len=0;
 336         for (i = 0; i < 6; i++) {
 337             if (a52_block (a52_state)){
 338                 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error at resampling\n");
 339                 break;
 340             }
 341             len+=2*a52_resample(a52_samples(a52_state),(int16_t *)&buf[len]);
 342         }
 343         assert(len <= maxlen);
 344   return len;
 345 }