libaf: add opts pointer to struct af_stream
[mplayer/glamo.git] / libaf / af.h
blob98ae163bd1fa86c0998ef878c647e3c15e388572
1 /*
2 * This file is part of MPlayer.
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.
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.
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.
19 #ifndef MPLAYER_AF_H
20 #define MPLAYER_AF_H
22 #include <stdio.h>
24 #include "config.h"
26 #include "options.h"
27 #include "af_format.h"
28 #include "control.h"
29 #include "cpudetect.h"
30 #include "mp_msg.h"
32 struct af_instance_s;
34 // Number of channels
35 #ifndef AF_NCH
36 #define AF_NCH 8
37 #endif
39 // Audio data chunk
40 typedef struct af_data_s
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;
51 // Flags used for defining the behavior of an audio filter
52 #define AF_FLAGS_REENTRANT 0x00000000
53 #define AF_FLAGS_NOT_REENTRANT 0x00000001
55 /* Audio filter information not specific for current instance, but for
56 a specific filter */
57 typedef struct af_info_s
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;
67 // Linked list of audio filters
68 typedef struct af_instance_s
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;
84 // Initialization flags
85 extern int* af_cpu_speed;
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
93 #define AF_INIT_INT 0x00000000
94 #define AF_INIT_FLOAT 0x00000004
95 #define AF_INIT_FORMAT_MASK 0x00000004
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
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;
109 // Current audio stream
110 typedef struct af_stream
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;
123 /*********************************************
124 // Return values
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
137 /*********************************************
138 // Export functions
142 * \defgroup af_chain Audio filter chain functions
143 * \{
144 * \param s filter chain
148 * \brief Initialize the stream "s".
149 * \return 0 on success, -1 on failure
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.
160 int af_init(af_stream_t* s);
163 * \brief Uninit and remove all filters from audio filter chain
165 void af_uninit(af_stream_t* s);
168 * \brief This function adds the filter "name" to the stream s.
169 * \param name name of filter to add
170 * \return pointer to the new filter, NULL if insert failed
172 * The filter will be inserted somewhere nice in the
173 * list of filters (i.e. at the beginning unless the
174 * first filter is the format filter (why??).
176 af_instance_t* af_add(af_stream_t* s, char* name);
179 * \brief Uninit and remove the filter "af"
180 * \param af filter to remove
182 void af_remove(af_stream_t* s, af_instance_t* af);
185 * \brief find filter in chain by name
186 * \param name name of the filter to find
187 * \return first filter with right name or NULL if not found
189 * This function is used for finding already initialized filters
191 af_instance_t* af_get(af_stream_t* s, char* name);
194 * \brief filter data chunk through the filters in the list
195 * \param data data to play
196 * \return resulting data
197 * \ingroup af_chain
199 af_data_t* af_play(af_stream_t* s, af_data_t* data);
202 * \brief send control to all filters, starting with the last until
203 * one accepts the command with AF_OK.
204 * \param cmd filter control command
205 * \param arg argument for filter command
206 * \return the accepting filter or NULL if none was found
208 af_instance_t *af_control_any_rev (af_stream_t* s, int cmd, void* arg);
211 * \brief calculate average ratio of filter output lenth to input length
212 * \return the ratio
214 double af_calc_filter_multiplier(af_stream_t* s);
217 * \brief Calculate the total delay caused by the filters
218 * \return delay in bytes of "missing" output
220 double af_calc_delay(af_stream_t* s);
222 /** \} */ // end of af_chain group
224 // Helper functions and macros used inside the audio filters
227 * \defgroup af_filter Audio filter helper functions
228 * \{
231 /* Helper function called by the macro with the same name only to be
232 called from inside filters */
233 int af_resize_local_buffer(af_instance_t* af, af_data_t* data);
235 /* Helper function used to calculate the exact buffer length needed
236 when buffers are resized. The returned length is >= than what is
237 needed */
238 int af_lencalc(double mul, af_data_t* data);
241 * \brief convert dB to gain value
242 * \param n number of values to convert
243 * \param in [in] values in dB, <= -200 will become 0 gain
244 * \param out [out] gain values
245 * \param k input values are divided by this
246 * \param mi minimum dB value, input will be clamped to this
247 * \param ma maximum dB value, input will be clamped to this
248 * \return AF_ERROR on error, AF_OK otherwise
250 int af_from_dB(int n, float* in, float* out, float k, float mi, float ma);
253 * \brief convert gain value to dB
254 * \param n number of values to convert
255 * \param in [in] gain values, 0 wil become -200 dB
256 * \param out [out] values in dB
257 * \param k output values will be multiplied by this
258 * \return AF_ERROR on error, AF_OK otherwise
260 int af_to_dB(int n, float* in, float* out, float k);
263 * \brief convert milliseconds to sample time
264 * \param n number of values to convert
265 * \param in [in] values in milliseconds
266 * \param out [out] sample time values
267 * \param rate sample rate
268 * \param mi minimum ms value, input will be clamped to this
269 * \param ma maximum ms value, input will be clamped to this
270 * \return AF_ERROR on error, AF_OK otherwise
272 int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma);
275 * \brief convert sample time to milliseconds
276 * \param n number of values to convert
277 * \param in [in] sample time values
278 * \param out [out] values in milliseconds
279 * \param rate sample rate
280 * \return AF_ERROR on error, AF_OK otherwise
282 int af_to_ms(int n, int* in, float* out, int rate);
285 * \brief test if output format matches
286 * \param af audio filter
287 * \param out needed format, will be overwritten by available
288 * format if they do not match
289 * \return AF_FALSE if formats do not match, AF_OK if they match
291 * compares the format, bps, rate and nch values of af->data with out
293 int af_test_output(struct af_instance_s* af, af_data_t* out);
296 * \brief soft clipping function using sin()
297 * \param a input value
298 * \return clipped value
300 float af_softclip(float a);
302 /** \} */ // end of af_filter group, but more functions of this group below
304 /** Print a list of all available audio filters */
305 void af_help(void);
308 * \brief fill the missing parameters in the af_data_t structure
309 * \param data structure to fill
310 * \ingroup af_filter
312 * Currently only sets bps based on format
314 void af_fix_parameters(af_data_t *data);
316 /** Memory reallocation macro: if a local buffer is used (i.e. if the
317 filter doesn't operate on the incoming buffer this macro must be
318 called to ensure the buffer is big enough.
319 * \ingroup af_filter
321 #define RESIZE_LOCAL_BUFFER(a,d)\
322 ((a->data->len < af_lencalc(a->mul,d))?af_resize_local_buffer(a,d):AF_OK)
324 /* Some other useful macro definitions*/
325 #ifndef min
326 #define min(a,b)(((a)>(b))?(b):(a))
327 #endif
329 #ifndef max
330 #define max(a,b)(((a)>(b))?(a):(b))
331 #endif
333 #ifndef clamp
334 #define clamp(a,min,max) (((a)>(max))?(max):(((a)<(min))?(min):(a)))
335 #endif
337 #ifndef sign
338 #define sign(a) (((a)>0)?(1):(-1))
339 #endif
341 #ifndef lrnd
342 #define lrnd(a,b) ((b)((a)>=0.0?(a)+0.5:(a)-0.5))
343 #endif
345 #endif /* MPLAYER_AF_H */