libaf/af.h

   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 #ifndef MPLAYER_AF_H
  20 #define MPLAYER_AF_H
  21
  22 #include <stdio.h>
  23
  24 #include "config.h"
  25
  26 #include "options.h"
  27 #include "af_format.h"
  28 #include "control.h"
  29 #include "cpudetect.h"
  30 #include "mp_msg.h"
  31
  32 struct af_instance_s;
  33
  34 // Number of channels
  35 #ifndef AF_NCH
  36 #define AF_NCH 8
  37 #endif
  38
  39 // Audio data chunk
  40 typedef struct af_data_s
  41 {
  42   void* audio;  // data buffer
  43   int len;      // buffer length
  44   int rate;     // sample rate
  45   int nch;      // number of channels
  46   int format;   // format
  47   int bps;      // bytes per sample
  48 } af_data_t;
  49
  50
  51 // Flags used for defining the behavior of an audio filter
  52 #define AF_FLAGS_REENTRANT      0x00000000
  53 #define AF_FLAGS_NOT_REENTRANT  0x00000001
  54
  55 /* Audio filter information not specific for current instance, but for
  56    a specific filter */
  57 typedef struct af_info_s
  58 {
  59   const char *info;
  60   const char *name;
  61   const char *author;
  62   const char *comment;
  63   const int flags;
  64   int (*open)(struct af_instance_s* vf);
  65 } af_info_t;
  66
  67 // Linked list of audio filters
  68 typedef struct af_instance_s
  69 {
  70   af_info_t* info;
  71   int (*control)(struct af_instance_s* af, int cmd, void* arg);
  72   void (*uninit)(struct af_instance_s* af);
  73   af_data_t* (*play)(struct af_instance_s* af, af_data_t* data);
  74   void* setup;    // setup data for this specific instance and filter
  75   af_data_t* data; // configuration for outgoing data stream
  76   struct af_instance_s* next;
  77   struct af_instance_s* prev;
  78   double delay; /* Delay caused by the filter, in units of bytes read without
  79                  * corresponding output */
  80   double mul; /* length multiplier: how much does this instance change
  81                  the length of the buffer. */
  82 }af_instance_t;
  83
  84 // Initialization flags
  85 extern int* af_cpu_speed;
  86
  87 #define AF_INIT_AUTO            0x00000000
  88 #define AF_INIT_SLOW            0x00000001
  89 #define AF_INIT_FAST            0x00000002
  90 #define AF_INIT_FORCE           0x00000003
  91 #define AF_INIT_TYPE_MASK       0x00000003
  92
  93 #define AF_INIT_INT             0x00000000
  94 #define AF_INIT_FLOAT           0x00000004
  95 #define AF_INIT_FORMAT_MASK     0x00000004
  96
  97 // Default init type
  98 #ifndef AF_INIT_TYPE
  99 #define AF_INIT_TYPE (af_cpu_speed?*af_cpu_speed:AF_INIT_SLOW)
 100 #endif
 101
 102 // Configuration switches
 103 typedef struct af_cfg_s{
 104   int force;    // Initialization type
 105   char** list;  /* list of names of filters that are added to filter
 106                    list during first initialization of stream */
 107 }af_cfg_t;
 108
 109 // Current audio stream
 110 typedef struct af_stream
 111 {
 112   // The first and last filter in the list
 113   af_instance_t* first;
 114   af_instance_t* last;
 115   // Storage for input and output data formats
 116   af_data_t input;
 117   af_data_t output;
 118   // Configuration for this stream
 119   af_cfg_t cfg;
 120   struct MPOpts *opts;
 121 }af_stream_t;
 122
 123 /*********************************************
 124 // Return values
 125 */
 126
 127 #define AF_DETACH   2
 128 #define AF_OK       1
 129 #define AF_TRUE     1
 130 #define AF_FALSE    0
 131 #define AF_UNKNOWN -1
 132 #define AF_ERROR   -2
 133 #define AF_FATAL   -3
 134
 135
 136
 137 /*********************************************
 138 // Export functions
 139 */
 140
 141 /**
 142  * \defgroup af_chain Audio filter chain functions
 143  * \{
 144  * \param s filter chain
 145  */
 146
 147 /**
 148  * \brief Initialize the stream "s".
 149  * \return 0 on success, -1 on failure
 150  *
 151  * This function creates a new filter list if necessary, according
 152  * to the values set in input and output. Input and output should contain
 153  * the format of the current movie and the format of the preferred output
 154  * respectively.
 155  * Filters to convert to the preferred output format are inserted
 156  * automatically, except when they are set to 0.
 157  * The function is reentrant i.e. if called with an already initialized
 158  * stream the stream will be reinitialized.
 159  */
 160 int af_init(af_stream_t* s);
 161
 162 /**
 163  * \brief Uninit and remove all filters from audio filter chain
 164  */
 165 void af_uninit(af_stream_t* s);
 166
 167 /**
 168  * \brief  Reinit the filter list from the given filter on downwards
 169  * \param  Filter instance to begin the reinit from
 170  * \return AF_OK on success or AF_ERROR on failure
 171  */
 172 int af_reinit(af_stream_t* s, af_instance_t* af);
 173
 174 /**
 175  * \brief This function adds the filter "name" to the stream s.
 176  * \param name name of filter to add
 177  * \return pointer to the new filter, NULL if insert failed
 178  *
 179  * The filter will be inserted somewhere nice in the
 180  * list of filters (i.e. at the beginning unless the
 181  * first filter is the format filter (why??).
 182  */
 183 af_instance_t* af_add(af_stream_t* s, char* name);
 184
 185 /**
 186  * \brief Uninit and remove the filter "af"
 187  * \param af filter to remove
 188  */
 189 void af_remove(af_stream_t* s, af_instance_t* af);
 190
 191 /**
 192  * \brief find filter in chain by name
 193  * \param name name of the filter to find
 194  * \return first filter with right name or NULL if not found
 195  *
 196  * This function is used for finding already initialized filters
 197  */
 198 af_instance_t* af_get(af_stream_t* s, char* name);
 199
 200 /**
 201  * \brief filter data chunk through the filters in the list
 202  * \param data data to play
 203  * \return resulting data
 204  * \ingroup af_chain
 205  */
 206 af_data_t* af_play(af_stream_t* s, af_data_t* data);
 207
 208 /**
 209  * \brief send control to all filters, starting with the last until
 210  *        one accepts the command with AF_OK.
 211  * \param cmd filter control command
 212  * \param arg argument for filter command
 213  * \return the accepting filter or NULL if none was found
 214  */
 215 af_instance_t *af_control_any_rev (af_stream_t* s, int cmd, void* arg);
 216
 217 /**
 218  * \brief calculate average ratio of filter output lenth to input length
 219  * \return the ratio
 220  */
 221 double af_calc_filter_multiplier(af_stream_t* s);
 222
 223 /**
 224  * \brief Calculate the total delay caused by the filters
 225  * \return delay in bytes of "missing" output
 226  */
 227 double af_calc_delay(af_stream_t* s);
 228
 229 /** \} */ // end of af_chain group
 230
 231 // Helper functions and macros used inside the audio filters
 232
 233 /**
 234  * \defgroup af_filter Audio filter helper functions
 235  * \{
 236  */
 237
 238 /* Helper function called by the macro with the same name only to be
 239    called from inside filters */
 240 int af_resize_local_buffer(af_instance_t* af, af_data_t* data);
 241
 242 /* Helper function used to calculate the exact buffer length needed
 243    when buffers are resized. The returned length is >= than what is
 244    needed */
 245 int af_lencalc(double mul, af_data_t* data);
 246
 247 /**
 248  * \brief convert dB to gain value
 249  * \param n number of values to convert
 250  * \param in [in] values in dB, <= -200 will become 0 gain
 251  * \param out [out] gain values
 252  * \param k input values are divided by this
 253  * \param mi minimum dB value, input will be clamped to this
 254  * \param ma maximum dB value, input will be clamped to this
 255  * \return AF_ERROR on error, AF_OK otherwise
 256  */
 257 int af_from_dB(int n, float* in, float* out, float k, float mi, float ma);
 258
 259 /**
 260  * \brief convert gain value to dB
 261  * \param n number of values to convert
 262  * \param in [in] gain values, 0 wil become -200 dB
 263  * \param out [out] values in dB
 264  * \param k output values will be multiplied by this
 265  * \return AF_ERROR on error, AF_OK otherwise
 266  */
 267 int af_to_dB(int n, float* in, float* out, float k);
 268
 269 /**
 270  * \brief convert milliseconds to sample time
 271  * \param n number of values to convert
 272  * \param in [in] values in milliseconds
 273  * \param out [out] sample time values
 274  * \param rate sample rate
 275  * \param mi minimum ms value, input will be clamped to this
 276  * \param ma maximum ms value, input will be clamped to this
 277  * \return AF_ERROR on error, AF_OK otherwise
 278  */
 279 int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma);
 280
 281 /**
 282  * \brief convert sample time to milliseconds
 283  * \param n number of values to convert
 284  * \param in [in] sample time values
 285  * \param out [out] values in milliseconds
 286  * \param rate sample rate
 287  * \return AF_ERROR on error, AF_OK otherwise
 288  */
 289 int af_to_ms(int n, int* in, float* out, int rate);
 290
 291 /**
 292  * \brief test if output format matches
 293  * \param af audio filter
 294  * \param out needed format, will be overwritten by available
 295  *            format if they do not match
 296  * \return AF_FALSE if formats do not match, AF_OK if they match
 297  *
 298  * compares the format, bps, rate and nch values of af->data with out
 299  */
 300 int af_test_output(struct af_instance_s* af, af_data_t* out);
 301
 302 /**
 303  * \brief soft clipping function using sin()
 304  * \param a input value
 305  * \return clipped value
 306  */
 307 float af_softclip(float a);
 308
 309 /** \} */ // end of af_filter group, but more functions of this group below
 310
 311 /** Print a list of all available audio filters */
 312 void af_help(void);
 313
 314 /**
 315  * \brief fill the missing parameters in the af_data_t structure
 316  * \param data structure to fill
 317  * \ingroup af_filter
 318  *
 319  * Currently only sets bps based on format
 320  */
 321 void af_fix_parameters(af_data_t *data);
 322
 323 /** Memory reallocation macro: if a local buffer is used (i.e. if the
 324    filter doesn't operate on the incoming buffer this macro must be
 325    called to ensure the buffer is big enough.
 326  * \ingroup af_filter
 327  */
 328 #define RESIZE_LOCAL_BUFFER(a,d)\
 329 ((a->data->len < af_lencalc(a->mul,d))?af_resize_local_buffer(a,d):AF_OK)
 330
 331 /* Some other useful macro definitions*/
 332 #ifndef min
 333 #define min(a,b)(((a)>(b))?(b):(a))
 334 #endif
 335
 336 #ifndef max
 337 #define max(a,b)(((a)>(b))?(a):(b))
 338 #endif
 339
 340 #ifndef clamp
 341 #define clamp(a,min,max) (((a)>(max))?(max):(((a)<(min))?(min):(a)))
 342 #endif
 343
 344 #ifndef sign
 345 #define sign(a) (((a)>0)?(1):(-1))
 346 #endif
 347
 348 #ifndef lrnd
 349 #define lrnd(a,b) ((b)((a)>=0.0?(a)+0.5:(a)-0.5))
 350 #endif
 351
 352 #endif /* MPLAYER_AF_H */