Mercurial > mplayer.hg
annotate libaf/af.c @ 36425:e89214f37501
Remove unused definitions.
author | ib |
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date | Sun, 24 Nov 2013 00:44:44 +0000 |
parents | 9b4ba0fb999b |
children | 3bc23ae2a154 |
rev | line source |
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1 /* |
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2 * This file is part of MPlayer. |
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3 * |
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4 * MPlayer is free software; you can redistribute it and/or modify |
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5 * it under the terms of the GNU General Public License as published by |
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6 * the Free Software Foundation; either version 2 of the License, or |
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7 * (at your option) any later version. |
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8 * |
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9 * MPlayer is distributed in the hope that it will be useful, |
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10 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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12 * GNU General Public License for more details. |
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13 * |
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14 * You should have received a copy of the GNU General Public License along |
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15 * with MPlayer; if not, write to the Free Software Foundation, Inc., |
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16 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. |
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17 */ |
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18 |
35951 | 19 #define _BSD_SOURCE |
20 | |
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21 #include "config.h" |
7568 | 22 #include <stdio.h> |
23 #include <stdlib.h> | |
24 #include <string.h> | |
34174
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Add missing mp_msg.h #includes, remove some unnecessary ones.
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25 |
30313
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Add a proper header for our strsep implementation so strsep will
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26 #include "osdep/strsep.h" |
32272
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Automatic downmix using the pan filter when the requested channel count is 2
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27 #include "libmpcodecs/dec_audio.h" |
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28 #include "mp_msg.h" |
7568 | 29 #include "af.h" |
30 | |
31 // Static list of filters | |
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32 extern const af_info_t af_info_dummy; |
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33 extern const af_info_t af_info_delay; |
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34 extern const af_info_t af_info_channels; |
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35 extern const af_info_t af_info_format; |
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36 extern const af_info_t af_info_resample; |
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37 extern const af_info_t af_info_volume; |
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38 extern const af_info_t af_info_equalizer; |
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39 extern const af_info_t af_info_gate; |
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40 extern const af_info_t af_info_comp; |
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41 extern const af_info_t af_info_pan; |
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42 extern const af_info_t af_info_surround; |
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43 extern const af_info_t af_info_sub; |
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44 extern const af_info_t af_info_export; |
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45 extern const af_info_t af_info_volnorm; |
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46 extern const af_info_t af_info_extrastereo; |
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47 extern const af_info_t af_info_lavcac3enc; |
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48 extern const af_info_t af_info_lavcresample; |
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49 extern const af_info_t af_info_sweep; |
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50 extern const af_info_t af_info_hrtf; |
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51 extern const af_info_t af_info_ladspa; |
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52 extern const af_info_t af_info_center; |
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53 extern const af_info_t af_info_sinesuppress; |
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54 extern const af_info_t af_info_karaoke; |
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55 extern const af_info_t af_info_scaletempo; |
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56 extern const af_info_t af_info_stats; |
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57 extern const af_info_t af_info_bs2b; |
7568 | 58 |
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59 static const af_info_t * const filter_list[] = { |
10908 | 60 &af_info_dummy, |
61 &af_info_delay, | |
62 &af_info_channels, | |
63 &af_info_format, | |
64 &af_info_resample, | |
65 &af_info_volume, | |
66 &af_info_equalizer, | |
67 &af_info_gate, | |
68 &af_info_comp, | |
69 &af_info_pan, | |
70 &af_info_surround, | |
71 &af_info_sub, | |
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72 #if HAVE_SYS_MMAN_H |
10908 | 73 &af_info_export, |
74 #endif | |
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75 &af_info_volnorm, |
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76 &af_info_extrastereo, |
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77 #ifdef CONFIG_FFMPEG |
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78 &af_info_lavcac3enc, |
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79 &af_info_lavcresample, |
13713 | 80 #endif |
13721 | 81 &af_info_sweep, |
13996 | 82 &af_info_hrtf, |
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83 #ifdef CONFIG_LADSPA |
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84 &af_info_ladspa, |
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85 #endif |
14750
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filter for adding a center channel, adding a high pass filter would be nice
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86 &af_info_center, |
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87 &af_info_sinesuppress, |
18470 | 88 &af_info_karaoke, |
24896 | 89 &af_info_scaletempo, |
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90 &af_info_stats, |
29093 | 91 #ifdef CONFIG_LIBBS2B |
92 &af_info_bs2b, | |
93 #endif | |
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94 NULL |
7568 | 95 }; |
96 | |
8167 | 97 // CPU speed |
98 int* af_cpu_speed = NULL; | |
99 | |
7568 | 100 /* Find a filter in the static list of filters using it's name. This |
101 function is used internally */ | |
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102 static const af_info_t* af_find(char*name) |
7568 | 103 { |
104 int i=0; | |
105 while(filter_list[i]){ | |
106 if(!strcmp(filter_list[i]->name,name)) | |
107 return filter_list[i]; | |
108 i++; | |
109 } | |
29049 | 110 mp_msg(MSGT_AFILTER, MSGL_ERR, "Couldn't find audio filter '%s'\n",name); |
7568 | 111 return NULL; |
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112 } |
7568 | 113 |
7615 | 114 /* Find filter in the dynamic filter list using it's name This |
115 function is used for finding already initialized filters */ | |
116 af_instance_t* af_get(af_stream_t* s, char* name) | |
117 { | |
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118 af_instance_t* af=s->first; |
7615 | 119 // Find the filter |
120 while(af != NULL){ | |
121 if(!strcmp(af->info->name,name)) | |
122 return af; | |
123 af=af->next; | |
124 } | |
125 return NULL; | |
126 } | |
127 | |
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128 /*/ Function for creating a new filter of type name. The name may |
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129 contain the commandline parameters for the filter */ |
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130 static af_instance_t* af_create(af_stream_t* s, const char* name_with_cmd) |
7568 | 131 { |
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132 char* name = strdup(name_with_cmd); |
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133 char* cmdline = name; |
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134 |
7568 | 135 // Allocate space for the new filter and reset all pointers |
136 af_instance_t* new=malloc(sizeof(af_instance_t)); | |
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137 if (!name || !new) { |
29049 | 138 mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Could not allocate memory\n"); |
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139 goto err_out; |
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140 } |
7568 | 141 memset(new,0,sizeof(af_instance_t)); |
142 | |
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143 // Check for commandline parameters |
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144 strsep(&cmdline, "="); |
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145 |
7568 | 146 // Find filter from name |
7615 | 147 if(NULL == (new->info=af_find(name))) |
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148 goto err_out; |
7615 | 149 |
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150 /* Make sure that the filter is not already in the list if it is |
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151 non-reentrant */ |
7615 | 152 if(new->info->flags & AF_FLAGS_NOT_REENTRANT){ |
153 if(af_get(s,name)){ | |
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154 mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] There can only be one instance of" |
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155 " the filter '%s' in each stream\n",name); |
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156 goto err_out; |
7615 | 157 } |
158 } | |
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159 |
29049 | 160 mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Adding filter %s \n",name); |
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161 |
7568 | 162 // Initialize the new filter |
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163 if(AF_OK == new->info->open(new) && |
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164 AF_ERROR < new->control(new,AF_CONTROL_POST_CREATE,&s->cfg)){ |
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165 if(cmdline){ |
25306 | 166 if(AF_ERROR>=new->control(new,AF_CONTROL_COMMAND_LINE,cmdline)) |
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167 goto err_out; |
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168 } |
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169 free(name); |
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170 return new; |
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171 } |
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172 |
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173 err_out: |
7568 | 174 free(new); |
29049 | 175 mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Couldn't create or open audio filter '%s'\n", |
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176 name); |
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177 free(name); |
7568 | 178 return NULL; |
179 } | |
180 | |
181 /* Create and insert a new filter of type name before the filter in the | |
182 argument. This function can be called during runtime, the return | |
183 value is the new filter */ | |
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184 static af_instance_t* af_prepend(af_stream_t* s, af_instance_t* af, const char* name) |
7568 | 185 { |
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186 // Create the new filter and make sure it is OK |
7615 | 187 af_instance_t* new=af_create(s,name); |
7568 | 188 if(!new) |
189 return NULL; | |
190 // Update pointers | |
191 new->next=af; | |
192 if(af){ | |
193 new->prev=af->prev; | |
194 af->prev=new; | |
195 } | |
196 else | |
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197 s->last=new; |
7568 | 198 if(new->prev) |
199 new->prev->next=new; | |
200 else | |
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201 s->first=new; |
7568 | 202 return new; |
203 } | |
204 | |
205 /* Create and insert a new filter of type name after the filter in the | |
206 argument. This function can be called during runtime, the return | |
207 value is the new filter */ | |
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208 static af_instance_t* af_append(af_stream_t* s, af_instance_t* af, const char* name) |
7568 | 209 { |
210 // Create the new filter and make sure it is OK | |
7615 | 211 af_instance_t* new=af_create(s,name); |
7568 | 212 if(!new) |
213 return NULL; | |
214 // Update pointers | |
215 new->prev=af; | |
216 if(af){ | |
217 new->next=af->next; | |
218 af->next=new; | |
219 } | |
220 else | |
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221 s->first=new; |
7568 | 222 if(new->next) |
223 new->next->prev=new; | |
224 else | |
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225 s->last=new; |
7568 | 226 return new; |
227 } | |
228 | |
229 // Uninit and remove the filter "af" | |
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230 void af_remove(af_stream_t* s, af_instance_t* af) |
7568 | 231 { |
232 if(!af) return; | |
233 | |
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234 // Print friendly message |
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235 mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Removing filter %s \n",af->info->name); |
8607 | 236 |
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237 // Notify filter before changing anything |
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238 af->control(af,AF_CONTROL_PRE_DESTROY,0); |
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239 |
7568 | 240 // Detach pointers |
241 if(af->prev) | |
242 af->prev->next=af->next; | |
243 else | |
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244 s->first=af->next; |
7568 | 245 if(af->next) |
246 af->next->prev=af->prev; | |
247 else | |
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248 s->last=af->prev; |
7568 | 249 |
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250 // Uninitialize af and free memory |
7568 | 251 af->uninit(af); |
252 free(af); | |
253 } | |
254 | |
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255 int af_reinit(af_stream_t* s, af_instance_t* af) |
7568 | 256 { |
257 do{ | |
258 af_data_t in; // Format of the input to current filter | |
259 int rv=0; // Return value | |
260 | |
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261 // Check if there are any filters left in the list |
35248 | 262 if(!af) |
263 return af_append(s,s->first,"dummy") ? AF_ERROR : AF_UNKNOWN; | |
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264 |
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265 // Check if this is the first filter |
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266 if(!af->prev) |
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267 memcpy(&in,&(s->input),sizeof(af_data_t)); |
7568 | 268 else |
269 memcpy(&in,af->prev->data,sizeof(af_data_t)); | |
270 // Reset just in case... | |
271 in.audio=NULL; | |
272 in.len=0; | |
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273 |
7568 | 274 rv = af->control(af,AF_CONTROL_REINIT,&in); |
275 switch(rv){ | |
276 case AF_OK: | |
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277 af = af->next; |
7568 | 278 break; |
279 case AF_FALSE:{ // Configuration filter is needed | |
8167 | 280 // Do auto insertion only if force is not specified |
281 if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){ | |
282 af_instance_t* new = NULL; | |
283 // Insert channels filter | |
284 if((af->prev?af->prev->data->nch:s->input.nch) != in.nch){ | |
285 // Create channels filter | |
286 if(NULL == (new = af_prepend(s,af,"channels"))) | |
287 return AF_ERROR; | |
288 // Set number of output channels | |
289 if(AF_OK != (rv = new->control(new,AF_CONTROL_CHANNELS,&in.nch))) | |
290 return rv; | |
291 // Initialize channels filter | |
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292 if(!new->prev) |
8167 | 293 memcpy(&in,&(s->input),sizeof(af_data_t)); |
294 else | |
295 memcpy(&in,new->prev->data,sizeof(af_data_t)); | |
296 if(AF_OK != (rv = new->control(new,AF_CONTROL_REINIT,&in))) | |
297 return rv; | |
298 } | |
299 // Insert format filter | |
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300 if((af->prev?af->prev->data->format:s->input.format) != in.format){ |
8167 | 301 // Create format filter |
302 if(NULL == (new = af_prepend(s,af,"format"))) | |
303 return AF_ERROR; | |
8607 | 304 // Set output bits per sample |
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305 in.format |= af_bits2fmt(in.bps*8); |
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306 if(AF_OK != (rv = new->control(new,AF_CONTROL_FORMAT_FMT,&in.format))) |
8167 | 307 return rv; |
308 // Initialize format filter | |
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309 if(!new->prev) |
8167 | 310 memcpy(&in,&(s->input),sizeof(af_data_t)); |
311 else | |
312 memcpy(&in,new->prev->data,sizeof(af_data_t)); | |
313 if(AF_OK != (rv = new->control(new,AF_CONTROL_REINIT,&in))) | |
314 return rv; | |
315 } | |
8607 | 316 if(!new){ // Should _never_ happen |
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317 mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Unable to correct audio format. " |
34261 | 318 "This error should never occur, please send a bug report.\n"); |
7568 | 319 return AF_ERROR; |
8607 | 320 } |
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321 af=new->next; |
7568 | 322 } |
16072 | 323 else { |
29049 | 324 mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Automatic filter insertion disabled " |
16072 | 325 "but formats do not match. Giving up.\n"); |
326 return AF_ERROR; | |
327 } | |
7568 | 328 break; |
329 } | |
330 case AF_DETACH:{ // Filter is redundant and wants to be unloaded | |
8167 | 331 // Do auto remove only if force is not specified |
332 if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){ | |
333 af_instance_t* aft=af->prev; | |
334 af_remove(s,af); | |
335 if(aft) | |
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336 af=aft->next; |
8167 | 337 else |
338 af=s->first; // Restart configuration | |
339 } | |
7568 | 340 break; |
341 } | |
342 default: | |
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343 mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Reinitialization did not work, audio" |
8607 | 344 " filter '%s' returned error code %i\n",af->info->name,rv); |
7568 | 345 return AF_ERROR; |
346 } | |
347 }while(af); | |
348 return AF_OK; | |
349 } | |
350 | |
351 // Uninit and remove all filters | |
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352 void af_uninit(af_stream_t* s) |
7568 | 353 { |
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354 while(s->first) |
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355 af_remove(s,s->first); |
7568 | 356 } |
357 | |
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358 /** |
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359 * Extend the filter chain so we get the required output format at the end. |
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360 * \return AF_ERROR on error, AF_OK if successful. |
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361 */ |
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362 static int fixup_output_format(af_stream_t* s) |
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363 { |
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364 af_instance_t* af = NULL; |
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365 // Check number of output channels fix if not OK |
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366 // If needed always inserted last -> easy to screw up other filters |
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367 if(s->output.nch && s->last->data->nch!=s->output.nch){ |
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368 if(!strcmp(s->last->info->name,"format")) |
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369 af = af_prepend(s,s->last,"channels"); |
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370 else |
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371 af = af_append(s,s->last,"channels"); |
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372 // Init the new filter |
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373 if(!af || (AF_OK != af->control(af,AF_CONTROL_CHANNELS,&(s->output.nch)))) |
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374 return AF_ERROR; |
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375 if(AF_OK != af_reinit(s,af)) |
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376 return AF_ERROR; |
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377 } |
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378 |
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379 // Check output format fix if not OK |
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380 if(s->output.format != AF_FORMAT_UNKNOWN && |
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381 s->last->data->format != s->output.format){ |
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382 if(strcmp(s->last->info->name,"format")) |
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383 af = af_append(s,s->last,"format"); |
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384 else |
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385 af = s->last; |
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386 // Init the new filter |
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387 s->output.format |= af_bits2fmt(s->output.bps*8); |
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388 if(!af || (AF_OK != af->control(af,AF_CONTROL_FORMAT_FMT,&(s->output.format)))) |
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389 return AF_ERROR; |
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390 if(AF_OK != af_reinit(s,af)) |
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391 return AF_ERROR; |
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392 } |
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393 |
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394 // Re init again just in case |
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395 if(AF_OK != af_reinit(s,s->first)) |
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396 return AF_ERROR; |
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397 |
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398 if (s->output.format == AF_FORMAT_UNKNOWN) |
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399 s->output.format = s->last->data->format; |
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400 if (!s->output.nch) s->output.nch = s->last->data->nch; |
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401 if (!s->output.rate) s->output.rate = s->last->data->rate; |
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402 if((s->last->data->format != s->output.format) || |
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403 (s->last->data->nch != s->output.nch) || |
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404 (s->last->data->rate != s->output.rate)) { |
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405 return AF_ERROR; |
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406 } |
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407 return AF_OK; |
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408 } |
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409 |
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410 /** |
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411 * Automatic downmix to stereo in case the codec does not implement it. |
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412 */ |
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413 static void af_downmix(af_stream_t* s) |
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414 { |
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415 static const char * const downmix_strs[AF_NCH + 1] = { |
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416 /* FL FR RL RR FC LF AL AR */ |
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417 [3] = "pan=2:" "0.6:0:" "0:0.6:" "0.4:0.4", |
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418 [4] = "pan=2:" "0.6:0:" "0:0.6:" "0.4:0:" "0:0.4", |
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419 [5] = "pan=2:" "0.5:0:" "0:0.5:" "0.2:0:" "0:0.2:" "0.3:0.3", |
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420 [6] = "pan=2:" "0.4:0:" "0:0.4:" "0.2:0:" "0:0.2:" "0.3:0.3:" "0.1:0.1", |
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421 [7] = "pan=2:" "0.4:0:" "0:0.4:" "0.2:0:" "0:0.2:" "0.3:0.3:" "0.1:0:" "0:0.1", |
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422 [8] = "pan=2:" "0.4:0:" "0:0.4:" "0.15:0:" "0:0.15:" "0.25:0.25:" "0.1:0.1:" "0.1:0:" "0:0.1", |
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423 }; |
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424 const char *af_pan_str = downmix_strs[s->input.nch]; |
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425 |
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426 if (af_pan_str) |
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|
427 af_append(s, s->first, af_pan_str); |
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Automatic downmix using the pan filter when the requested channel count is 2
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428 } |
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Automatic downmix using the pan filter when the requested channel count is 2
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429 |
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Adding support for multiple audio streams and removing annoying message from resample and format
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430 /* Initialize the stream "s". This function creates a new filter list |
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Adding support for multiple audio streams and removing annoying message from resample and format
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431 if necessary according to the values set in input and output. Input |
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Adding support for multiple audio streams and removing annoying message from resample and format
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432 and output should contain the format of the current movie and the |
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Adding support for multiple audio streams and removing annoying message from resample and format
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|
433 formate of the preferred output respectively. The function is |
8819fdf88b5d
Adding support for multiple audio streams and removing annoying message from resample and format
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|
434 reentrant i.e. if called with an already initialized stream the |
15811
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make -srate work again, unify audio filter init and preinit.
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|
435 stream will be reinitialized. |
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|
436 If one of the prefered output parameters is 0 the one that needs |
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437 no conversion is used (i.e. the output format in the last filter). |
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438 The return value is 0 if success and -1 if failure */ |
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make -srate work again, unify audio filter init and preinit.
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439 int af_init(af_stream_t* s) |
7568 | 440 { |
441 int i=0; | |
442 | |
7571
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Adding support for multiple audio streams and removing annoying message from resample and format
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443 // Sanity check |
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Adding support for multiple audio streams and removing annoying message from resample and format
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444 if(!s) return -1; |
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Adding support for multiple audio streams and removing annoying message from resample and format
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445 |
7568 | 446 // Precaution in case caller is misbehaving |
7571
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Adding support for multiple audio streams and removing annoying message from resample and format
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447 s->input.audio = s->output.audio = NULL; |
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Adding support for multiple audio streams and removing annoying message from resample and format
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448 s->input.len = s->output.len = 0; |
7568 | 449 |
450 // Figure out how fast the machine is | |
8167 | 451 if(AF_INIT_AUTO == (AF_INIT_TYPE_MASK & s->cfg.force)) |
452 s->cfg.force = (s->cfg.force & ~AF_INIT_TYPE_MASK) | AF_INIT_TYPE; | |
7568 | 453 |
454 // Check if this is the first call | |
7571
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Adding support for multiple audio streams and removing annoying message from resample and format
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455 if(!s->first){ |
32272
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Automatic downmix using the pan filter when the requested channel count is 2
cigaes
parents:
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|
456 // Append a downmix pan filter at the beginning of the chain if needed |
fd4cd29b6121
Automatic downmix using the pan filter when the requested channel count is 2
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457 if (s->input.nch != audio_output_channels && audio_output_channels == 2) |
fd4cd29b6121
Automatic downmix using the pan filter when the requested channel count is 2
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458 af_downmix(s); |
7568 | 459 // Add all filters in the list (if there are any) |
32263 | 460 if (s->cfg.list) { |
7571
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Adding support for multiple audio streams and removing annoying message from resample and format
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461 while(s->cfg.list[i]){ |
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Adding support for multiple audio streams and removing annoying message from resample and format
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462 if(!af_append(s,s->last,s->cfg.list[i++])) |
7568 | 463 return -1; |
464 } | |
465 } | |
466 } | |
8607 | 467 |
32263 | 468 // If we do not have any filters otherwise |
469 // add dummy to make automatic format conversion work | |
470 if (!s->first && !af_append(s, s->first, "dummy")) | |
471 return -1; | |
472 | |
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whitespace cosmetics: Remove all trailing whitespace.
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473 // Init filters |
7571
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Adding support for multiple audio streams and removing annoying message from resample and format
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474 if(AF_OK != af_reinit(s,s->first)) |
7568 | 475 return -1; |
476 | |
15312
0313ef8b0730
Prevent segfault when filter chain is empty (e.g. because all
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|
477 // make sure the chain is not empty and valid (e.g. because of AF_DETACH) |
0313ef8b0730
Prevent segfault when filter chain is empty (e.g. because all
reimar
parents:
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diff
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|
478 if (!s->first) |
0313ef8b0730
Prevent segfault when filter chain is empty (e.g. because all
reimar
parents:
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|
479 if (!af_append(s,s->first,"dummy") || AF_OK != af_reinit(s,s->first)) |
0313ef8b0730
Prevent segfault when filter chain is empty (e.g. because all
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|
480 return -1; |
0313ef8b0730
Prevent segfault when filter chain is empty (e.g. because all
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|
481 |
7568 | 482 // Check output format |
7745
1d3a3dc1f488
Adding volume control and moving control() call parameters to a seperate file
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483 if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){ |
7568 | 484 af_instance_t* af = NULL; // New filter |
485 // Check output frequency if not OK fix with resample | |
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make -srate work again, unify audio filter init and preinit.
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486 if(s->output.rate && s->last->data->rate!=s->output.rate){ |
14326
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Use lavcresample only when libavcodec is compiled in.
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|
487 // try to find a filter that can change samplrate |
9261d7dcf5e7
Use lavcresample only when libavcodec is compiled in.
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|
488 af = af_control_any_rev(s, AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET, |
9261d7dcf5e7
Use lavcresample only when libavcodec is compiled in.
reimar
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|
489 &(s->output.rate)); |
9261d7dcf5e7
Use lavcresample only when libavcodec is compiled in.
reimar
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diff
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|
490 if (!af) { |
33157
8073274a9ff2
Add const to type for pointer to a string constant.
reimar
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diff
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|
491 const char *resampler = "resample"; |
32142
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build system: Merge all FFmpeg library checks into a single FFmpeg check.
diego
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diff
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|
492 #ifdef CONFIG_FFMPEG |
14326
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Use lavcresample only when libavcodec is compiled in.
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493 if ((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_SLOW) |
9261d7dcf5e7
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|
494 resampler = "lavcresample"; |
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Use lavcresample only when libavcodec is compiled in.
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|
495 #endif |
7745
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Adding volume control and moving control() call parameters to a seperate file
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|
496 if((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_SLOW){ |
7571
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Adding support for multiple audio streams and removing annoying message from resample and format
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497 if(!strcmp(s->first->info->name,"format")) |
14326
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498 af = af_append(s,s->first,resampler); |
7568 | 499 else |
14326
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Use lavcresample only when libavcodec is compiled in.
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500 af = af_prepend(s,s->first,resampler); |
29263
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whitespace cosmetics: Remove all trailing whitespace.
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501 } |
7568 | 502 else{ |
7571
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Adding support for multiple audio streams and removing annoying message from resample and format
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503 if(!strcmp(s->last->info->name,"format")) |
14326
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Use lavcresample only when libavcodec is compiled in.
reimar
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changeset
|
504 af = af_prepend(s,s->last,resampler); |
7568 | 505 else |
14326
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Use lavcresample only when libavcodec is compiled in.
reimar
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|
506 af = af_append(s,s->last,resampler); |
7568 | 507 } |
508 // Init the new filter | |
15191
7eab9c86ae19
change list traversal so the loop begins at the first filter after removing
henry
parents:
14818
diff
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|
509 if(!af || (AF_OK != af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET, |
8607 | 510 &(s->output.rate)))) |
7568 | 511 return -1; |
11859
b8bee4f4b8bb
if the user wants fast, use fast code! otherwise the user has to put
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parents:
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|
512 // Use lin int if the user wants fast |
b8bee4f4b8bb
if the user wants fast, use fast code! otherwise the user has to put
rfelker
parents:
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changeset
|
513 if ((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_FAST) { |
b8bee4f4b8bb
if the user wants fast, use fast code! otherwise the user has to put
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diff
changeset
|
514 char args[32]; |
14326
9261d7dcf5e7
Use lavcresample only when libavcodec is compiled in.
reimar
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|
515 sprintf(args, "%d", s->output.rate); |
32142
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build system: Merge all FFmpeg library checks into a single FFmpeg check.
diego
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|
516 #ifdef CONFIG_FFMPEG |
14326
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Use lavcresample only when libavcodec is compiled in.
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|
517 if (strcmp(resampler, "lavcresample") == 0) |
9261d7dcf5e7
Use lavcresample only when libavcodec is compiled in.
reimar
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|
518 strcat(args, ":1"); |
9261d7dcf5e7
Use lavcresample only when libavcodec is compiled in.
reimar
parents:
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diff
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|
519 else |
9261d7dcf5e7
Use lavcresample only when libavcodec is compiled in.
reimar
parents:
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diff
changeset
|
520 #endif |
9261d7dcf5e7
Use lavcresample only when libavcodec is compiled in.
reimar
parents:
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diff
changeset
|
521 strcat(args, ":0:0"); |
11859
b8bee4f4b8bb
if the user wants fast, use fast code! otherwise the user has to put
rfelker
parents:
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diff
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|
522 af->control(af, AF_CONTROL_COMMAND_LINE, args); |
b8bee4f4b8bb
if the user wants fast, use fast code! otherwise the user has to put
rfelker
parents:
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diff
changeset
|
523 } |
14326
9261d7dcf5e7
Use lavcresample only when libavcodec is compiled in.
reimar
parents:
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diff
changeset
|
524 } |
7571
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Adding support for multiple audio streams and removing annoying message from resample and format
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|
525 if(AF_OK != af_reinit(s,af)) |
7568 | 526 return -1; |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29093
diff
changeset
|
527 } |
30617
22aef54d8833
Move code that makes the filter chain match the desired output format into
reimar
parents:
30313
diff
changeset
|
528 if (AF_OK != fixup_output_format(s)) { |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29093
diff
changeset
|
529 // Something is stuffed audio out will not work |
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
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diff
changeset
|
530 mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Unable to setup filter system can not" |
34261 | 531 " meet sound-card demands, please send a bug report. \n"); |
7571
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Adding support for multiple audio streams and removing annoying message from resample and format
anders
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diff
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|
532 af_uninit(s); |
7568 | 533 return -1; |
534 } | |
535 } | |
536 return 0; | |
537 } | |
538 | |
7649
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
7617
diff
changeset
|
539 /* Add filter during execution. This function adds the filter "name" |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
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diff
changeset
|
540 to the stream s. The filter will be inserted somewhere nice in the |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
7617
diff
changeset
|
541 list of filters. The return value is a pointer to the new filter, |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
7617
diff
changeset
|
542 If the filter couldn't be added the return value is NULL. */ |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
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diff
changeset
|
543 af_instance_t* af_add(af_stream_t* s, char* name){ |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
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diff
changeset
|
544 af_instance_t* new; |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
7617
diff
changeset
|
545 // Sanity check |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
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diff
changeset
|
546 if(!s || !s->first || !name) |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
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diff
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|
547 return NULL; |
34261 | 548 // Insert the filter somewhere nice |
7649
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
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diff
changeset
|
549 if(!strcmp(s->first->info->name,"format")) |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
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diff
changeset
|
550 new = af_append(s, s->first, name); |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
7617
diff
changeset
|
551 else |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
7617
diff
changeset
|
552 new = af_prepend(s, s->first, name); |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
7617
diff
changeset
|
553 if(!new) |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
7617
diff
changeset
|
554 return NULL; |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
7617
diff
changeset
|
555 |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
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diff
changeset
|
556 // Reinitalize the filter list |
30617
22aef54d8833
Move code that makes the filter chain match the desired output format into
reimar
parents:
30313
diff
changeset
|
557 if(AF_OK != af_reinit(s, s->first) || |
22aef54d8833
Move code that makes the filter chain match the desired output format into
reimar
parents:
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diff
changeset
|
558 AF_OK != fixup_output_format(s)){ |
7649
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
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diff
changeset
|
559 free(new); |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
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diff
changeset
|
560 return NULL; |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
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diff
changeset
|
561 } |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
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diff
changeset
|
562 return new; |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
parents:
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diff
changeset
|
563 } |
90e16aa8ae5f
Adding functionality for adding filters during execution
anders
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7617
diff
changeset
|
564 |
7568 | 565 // Filter data chunk through the filters in the list |
7571
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Adding support for multiple audio streams and removing annoying message from resample and format
anders
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diff
changeset
|
566 af_data_t* af_play(af_stream_t* s, af_data_t* data) |
7568 | 567 { |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
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diff
changeset
|
568 af_instance_t* af=s->first; |
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29093
diff
changeset
|
569 // Iterate through all filters |
7568 | 570 do{ |
16815
78c84594247b
semi-hack: avoid passing 0-length blocks to audio filters.
reimar
parents:
16627
diff
changeset
|
571 if (data->len <= 0) break; |
7568 | 572 data=af->play(af,data); |
573 af=af->next; | |
25115
5a0da5dcadd3
Prevent from using data->len when data is NULL (when play() return NULL).
ulion
parents:
24900
diff
changeset
|
574 }while(af && data); |
7568 | 575 return data; |
576 } | |
577 | |
24888 | 578 /* Calculate the minimum output buffer size for given input data d |
579 * when using the RESIZE_LOCAL_BUFFER macro. The +t+1 part ensures the | |
580 * value is >= len*mul rounded upwards to whole samples even if the | |
581 * double 'mul' is inexact. */ | |
582 int af_lencalc(double mul, af_data_t* d) | |
583 { | |
584 int t = d->bps * d->nch; | |
585 return d->len * mul + t + 1; | |
7568 | 586 } |
587 | |
24892 | 588 // Calculate average ratio of filter output size to input size |
589 double af_calc_filter_multiplier(af_stream_t* s) | |
7598
48f8c731efb5
Adding function for estimating required buffer length
anders
parents:
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diff
changeset
|
590 { |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
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diff
changeset
|
591 af_instance_t* af=s->first; |
24888 | 592 double mul = 1; |
7598
48f8c731efb5
Adding function for estimating required buffer length
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diff
changeset
|
593 // Iterate through all filters and calculate total multiplication factor |
48f8c731efb5
Adding function for estimating required buffer length
anders
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diff
changeset
|
594 do{ |
24888 | 595 mul *= af->mul; |
596 af=af->next; | |
7598
48f8c731efb5
Adding function for estimating required buffer length
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|
597 }while(af); |
8607 | 598 |
24892 | 599 return mul; |
7598
48f8c731efb5
Adding function for estimating required buffer length
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diff
changeset
|
600 } |
48f8c731efb5
Adding function for estimating required buffer length
anders
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diff
changeset
|
601 |
24900 | 602 /* Calculate the total delay [bytes output] caused by the filters */ |
7665
fbd5445cc853
Adding function for calculating the delay caused by the filters
anders
parents:
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diff
changeset
|
603 double af_calc_delay(af_stream_t* s) |
fbd5445cc853
Adding function for calculating the delay caused by the filters
anders
parents:
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diff
changeset
|
604 { |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29093
diff
changeset
|
605 af_instance_t* af=s->first; |
7665
fbd5445cc853
Adding function for calculating the delay caused by the filters
anders
parents:
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diff
changeset
|
606 register double delay = 0.0; |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29093
diff
changeset
|
607 // Iterate through all filters |
7665
fbd5445cc853
Adding function for calculating the delay caused by the filters
anders
parents:
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diff
changeset
|
608 while(af){ |
fbd5445cc853
Adding function for calculating the delay caused by the filters
anders
parents:
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diff
changeset
|
609 delay += af->delay; |
24900 | 610 delay *= af->mul; |
7665
fbd5445cc853
Adding function for calculating the delay caused by the filters
anders
parents:
7649
diff
changeset
|
611 af=af->next; |
fbd5445cc853
Adding function for calculating the delay caused by the filters
anders
parents:
7649
diff
changeset
|
612 } |
fbd5445cc853
Adding function for calculating the delay caused by the filters
anders
parents:
7649
diff
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613 return delay; |
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614 } |
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615 |
7568 | 616 /* Helper function called by the macro with the same name this |
617 function should not be called directly */ | |
24890 | 618 int af_resize_local_buffer(af_instance_t* af, af_data_t* data) |
7568 | 619 { |
620 // Calculate new length | |
7589 | 621 register int len = af_lencalc(af->mul,data); |
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622 mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Reallocating memory in module %s, " |
8607 | 623 "old len = %i, new len = %i\n",af->info->name,af->data->len,len); |
7568 | 624 // If there is a buffer free it |
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625 free(af->data->audio); |
7568 | 626 // Create new buffer and check that it is OK |
627 af->data->audio = malloc(len); | |
628 if(!af->data->audio){ | |
29049 | 629 mp_msg(MSGT_AFILTER, MSGL_FATAL, "[libaf] Could not allocate memory \n"); |
7568 | 630 return AF_ERROR; |
631 } | |
632 af->data->len=len; | |
633 return AF_OK; | |
634 } | |
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635 |
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636 // documentation in af.h |
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637 af_instance_t *af_control_any_rev (af_stream_t* s, int cmd, void* arg) { |
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638 int res = AF_UNKNOWN; |
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639 af_instance_t* filt = s->last; |
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640 while (filt) { |
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641 res = filt->control(filt, cmd, arg); |
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642 if (res == AF_OK) |
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643 return filt; |
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644 filt = filt->prev; |
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645 } |
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646 return NULL; |
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647 } |
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648 |
13269
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ivo
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649 void af_help (void) { |
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650 int i = 0; |
29049 | 651 mp_msg(MSGT_AFILTER, MSGL_INFO, "Available audio filters:\n"); |
13269
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652 while (filter_list[i]) { |
13566
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653 if (filter_list[i]->comment && filter_list[i]->comment[0]) |
29049 | 654 mp_msg(MSGT_AFILTER, MSGL_INFO, " %-15s: %s (%s)\n", filter_list[i]->name, filter_list[i]->info, filter_list[i]->comment); |
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655 else |
29049 | 656 mp_msg(MSGT_AFILTER, MSGL_INFO, " %-15s: %s\n", filter_list[i]->name, filter_list[i]->info); |
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657 i++; |
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658 } |
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659 } |
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660 |
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661 void af_fix_parameters(af_data_t *data) |
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662 { |
34254
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reimar
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663 if (data->nch < 0 || data->nch > AF_NCH) { |
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664 mp_msg(MSGT_AFILTER, MSGL_ERR, "Invalid number of channels %i, assuming 2.\n", data->nch); |
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reimar
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665 data->nch = 2; |
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666 } |
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667 data->bps = af_fmt2bits(data->format)/8; |
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668 } |