Drop pointless _s suffix from 'struct af_stream'.
[mplayer.git] / libaf / af.h
blobfe146906e9cf4608138a3d5089560a8c36865cbf
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 "af_format.h"
27 #include "control.h"
28 #include "cpudetect.h"
29 #include "mp_msg.h"
31 struct af_instance_s;
33 // Number of channels
34 #ifndef AF_NCH
35 #define AF_NCH 8
36 #endif
38 // Audio data chunk
39 typedef struct af_data_s
41 void* audio; // data buffer
42 int len; // buffer length
43 int rate; // sample rate
44 int nch; // number of channels
45 int format; // format
46 int bps; // bytes per sample
47 } af_data_t;
50 // Flags used for defining the behavior of an audio filter
51 #define AF_FLAGS_REENTRANT 0x00000000
52 #define AF_FLAGS_NOT_REENTRANT 0x00000001
54 /* Audio filter information not specific for current instance, but for
55 a specific filter */
56 typedef struct af_info_s
58 const char *info;
59 const char *name;
60 const char *author;
61 const char *comment;
62 const int flags;
63 int (*open)(struct af_instance_s* vf);
64 } af_info_t;
66 // Linked list of audio filters
67 typedef struct af_instance_s
69 af_info_t* info;
70 int (*control)(struct af_instance_s* af, int cmd, void* arg);
71 void (*uninit)(struct af_instance_s* af);
72 af_data_t* (*play)(struct af_instance_s* af, af_data_t* data);
73 void* setup; // setup data for this specific instance and filter
74 af_data_t* data; // configuration for outgoing data stream
75 struct af_instance_s* next;
76 struct af_instance_s* prev;
77 double delay; /* Delay caused by the filter, in units of bytes read without
78 * corresponding output */
79 double mul; /* length multiplier: how much does this instance change
80 the length of the buffer. */
81 }af_instance_t;
83 // Initialization flags
84 extern int* af_cpu_speed;
86 #define AF_INIT_AUTO 0x00000000
87 #define AF_INIT_SLOW 0x00000001
88 #define AF_INIT_FAST 0x00000002
89 #define AF_INIT_FORCE 0x00000003
90 #define AF_INIT_TYPE_MASK 0x00000003
92 #define AF_INIT_INT 0x00000000
93 #define AF_INIT_FLOAT 0x00000004
94 #define AF_INIT_FORMAT_MASK 0x00000004
96 // Default init type
97 #ifndef AF_INIT_TYPE
98 #define AF_INIT_TYPE (af_cpu_speed?*af_cpu_speed:AF_INIT_SLOW)
99 #endif
101 // Configuration switches
102 typedef struct af_cfg_s{
103 int force; // Initialization type
104 char** list; /* list of names of filters that are added to filter
105 list during first initialization of stream */
106 }af_cfg_t;
108 // Current audio stream
109 typedef struct af_stream
111 // The first and last filter in the list
112 af_instance_t* first;
113 af_instance_t* last;
114 // Storage for input and output data formats
115 af_data_t input;
116 af_data_t output;
117 // Configuration for this stream
118 af_cfg_t cfg;
119 }af_stream_t;
121 /*********************************************
122 // Return values
125 #define AF_DETACH 2
126 #define AF_OK 1
127 #define AF_TRUE 1
128 #define AF_FALSE 0
129 #define AF_UNKNOWN -1
130 #define AF_ERROR -2
131 #define AF_FATAL -3
135 /*********************************************
136 // Export functions
140 * \defgroup af_chain Audio filter chain functions
141 * \{
142 * \param s filter chain
146 * \brief Initialize the stream "s".
147 * \return 0 on success, -1 on failure
149 * This function creates a new filter list if necessary, according
150 * to the values set in input and output. Input and output should contain
151 * the format of the current movie and the format of the preferred output
152 * respectively.
153 * Filters to convert to the preferred output format are inserted
154 * automatically, except when they are set to 0.
155 * The function is reentrant i.e. if called with an already initialized
156 * stream the stream will be reinitialized.
158 int af_init(af_stream_t* s);
161 * \brief Uninit and remove all filters from audio filter chain
163 void af_uninit(af_stream_t* s);
166 * \brief This function adds the filter "name" to the stream s.
167 * \param name name of filter to add
168 * \return pointer to the new filter, NULL if insert failed
170 * The filter will be inserted somewhere nice in the
171 * list of filters (i.e. at the beginning unless the
172 * first filter is the format filter (why??).
174 af_instance_t* af_add(af_stream_t* s, char* name);
177 * \brief Uninit and remove the filter "af"
178 * \param af filter to remove
180 void af_remove(af_stream_t* s, af_instance_t* af);
183 * \brief find filter in chain by name
184 * \param name name of the filter to find
185 * \return first filter with right name or NULL if not found
187 * This function is used for finding already initialized filters
189 af_instance_t* af_get(af_stream_t* s, char* name);
192 * \brief filter data chunk through the filters in the list
193 * \param data data to play
194 * \return resulting data
195 * \ingroup af_chain
197 af_data_t* af_play(af_stream_t* s, af_data_t* data);
200 * \brief send control to all filters, starting with the last until
201 * one accepts the command with AF_OK.
202 * \param cmd filter control command
203 * \param arg argument for filter command
204 * \return the accepting filter or NULL if none was found
206 af_instance_t *af_control_any_rev (af_stream_t* s, int cmd, void* arg);
209 * \brief calculate average ratio of filter output lenth to input length
210 * \return the ratio
212 double af_calc_filter_multiplier(af_stream_t* s);
215 * \brief Calculate the total delay caused by the filters
216 * \return delay in bytes of "missing" output
218 double af_calc_delay(af_stream_t* s);
220 /** \} */ // end of af_chain group
222 // Helper functions and macros used inside the audio filters
225 * \defgroup af_filter Audio filter helper functions
226 * \{
229 /* Helper function called by the macro with the same name only to be
230 called from inside filters */
231 int af_resize_local_buffer(af_instance_t* af, af_data_t* data);
233 /* Helper function used to calculate the exact buffer length needed
234 when buffers are resized. The returned length is >= than what is
235 needed */
236 int af_lencalc(double mul, af_data_t* data);
239 * \brief convert dB to gain value
240 * \param n number of values to convert
241 * \param in [in] values in dB, <= -200 will become 0 gain
242 * \param out [out] gain values
243 * \param k input values are divided by this
244 * \param mi minimum dB value, input will be clamped to this
245 * \param ma maximum dB value, input will be clamped to this
246 * \return AF_ERROR on error, AF_OK otherwise
248 int af_from_dB(int n, float* in, float* out, float k, float mi, float ma);
251 * \brief convert gain value to dB
252 * \param n number of values to convert
253 * \param in [in] gain values, 0 wil become -200 dB
254 * \param out [out] values in dB
255 * \param k output values will be multiplied by this
256 * \return AF_ERROR on error, AF_OK otherwise
258 int af_to_dB(int n, float* in, float* out, float k);
261 * \brief convert milliseconds to sample time
262 * \param n number of values to convert
263 * \param in [in] values in milliseconds
264 * \param out [out] sample time values
265 * \param rate sample rate
266 * \param mi minimum ms value, input will be clamped to this
267 * \param ma maximum ms value, input will be clamped to this
268 * \return AF_ERROR on error, AF_OK otherwise
270 int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma);
273 * \brief convert sample time to milliseconds
274 * \param n number of values to convert
275 * \param in [in] sample time values
276 * \param out [out] values in milliseconds
277 * \param rate sample rate
278 * \return AF_ERROR on error, AF_OK otherwise
280 int af_to_ms(int n, int* in, float* out, int rate);
283 * \brief test if output format matches
284 * \param af audio filter
285 * \param out needed format, will be overwritten by available
286 * format if they do not match
287 * \return AF_FALSE if formats do not match, AF_OK if they match
289 * compares the format, bps, rate and nch values of af->data with out
291 int af_test_output(struct af_instance_s* af, af_data_t* out);
294 * \brief soft clipping function using sin()
295 * \param a input value
296 * \return clipped value
298 float af_softclip(float a);
300 /** \} */ // end of af_filter group, but more functions of this group below
302 /** Print a list of all available audio filters */
303 void af_help(void);
306 * \brief fill the missing parameters in the af_data_t structure
307 * \param data structure to fill
308 * \ingroup af_filter
310 * Currently only sets bps based on format
312 void af_fix_parameters(af_data_t *data);
314 /** Memory reallocation macro: if a local buffer is used (i.e. if the
315 filter doesn't operate on the incoming buffer this macro must be
316 called to ensure the buffer is big enough.
317 * \ingroup af_filter
319 #define RESIZE_LOCAL_BUFFER(a,d)\
320 ((a->data->len < af_lencalc(a->mul,d))?af_resize_local_buffer(a,d):AF_OK)
322 /* Some other useful macro definitions*/
323 #ifndef min
324 #define min(a,b)(((a)>(b))?(b):(a))
325 #endif
327 #ifndef max
328 #define max(a,b)(((a)>(b))?(a):(b))
329 #endif
331 #ifndef clamp
332 #define clamp(a,min,max) (((a)>(max))?(max):(((a)<(min))?(min):(a)))
333 #endif
335 #ifndef sign
336 #define sign(a) (((a)>0)?(1):(-1))
337 #endif
339 #ifndef lrnd
340 #define lrnd(a,b) ((b)((a)>=0.0?(a)+0.5:(a)-0.5))
341 #endif
343 #endif /* MPLAYER_AF_H */