Mercurial > mplayer.hg
annotate libao2/ao_sdl.c @ 10040:901cb1391c13
Fixed a lot of typos.
author | lumag |
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date | Thu, 01 May 2003 12:30:44 +0000 |
parents | 12b1790038b0 |
children | 6a96649b235f |
rev | line source |
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972 | 1 /* |
2 * ao_sdl.c - libao2 SDLlib Audio Output Driver for MPlayer | |
3 * | |
4 * This driver is under the same license as MPlayer. | |
5 * (http://mplayer.sf.net) | |
6 * | |
7 * Copyleft 2001 by Felix Bünemann (atmosfear@users.sf.net) | |
8 * | |
9 * Thanks to Arpi for nice ringbuffer-code! | |
10 * | |
11 */ | |
12 | |
966 | 13 #include <stdio.h> |
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14 #include <stdlib.h> |
966 | 15 |
8027 | 16 #include "../config.h" |
17 #include "../mp_msg.h" | |
18 | |
966 | 19 #include "audio_out.h" |
20 #include "audio_out_internal.h" | |
1066 | 21 #include "afmt.h" |
6184 | 22 #include <SDL.h> |
966 | 23 |
972 | 24 #include "../libvo/fastmemcpy.h" |
25 | |
966 | 26 static ao_info_t info = |
27 { | |
28 "SDLlib audio output", | |
29 "sdl", | |
30 "Felix Buenemann <atmosfear@users.sourceforge.net>", | |
31 "" | |
32 }; | |
33 | |
34 LIBAO_EXTERN(sdl) | |
35 | |
972 | 36 // Samplesize used by the SDLlib AudioSpec struct |
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37 #define SAMPLESIZE 1024 |
972 | 38 |
39 // General purpose Ring-buffering routines | |
40 | |
974 | 41 #define BUFFSIZE 4096 |
1066 | 42 #define NUM_BUFS 16 |
972 | 43 |
44 static unsigned char *buffer[NUM_BUFS]; | |
45 | |
46 static unsigned int buf_read=0; | |
47 static unsigned int buf_write=0; | |
48 static unsigned int buf_read_pos=0; | |
49 static unsigned int buf_write_pos=0; | |
6184 | 50 static unsigned int volume=127; |
972 | 51 static int full_buffers=0; |
52 static int buffered_bytes=0; | |
53 | |
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54 |
972 | 55 static int write_buffer(unsigned char* data,int len){ |
56 int len2=0; | |
57 int x; | |
58 while(len>0){ | |
59 if(full_buffers==NUM_BUFS) break; | |
60 x=BUFFSIZE-buf_write_pos; | |
61 if(x>len) x=len; | |
62 memcpy(buffer[buf_write]+buf_write_pos,data+len2,x); | |
63 len2+=x; len-=x; | |
64 buffered_bytes+=x; buf_write_pos+=x; | |
65 if(buf_write_pos>=BUFFSIZE){ | |
66 // block is full, find next! | |
67 buf_write=(buf_write+1)%NUM_BUFS; | |
68 ++full_buffers; | |
69 buf_write_pos=0; | |
70 } | |
71 } | |
72 return len2; | |
73 } | |
74 | |
75 static int read_buffer(unsigned char* data,int len){ | |
76 int len2=0; | |
77 int x; | |
78 while(len>0){ | |
79 if(full_buffers==0) break; // no more data buffered! | |
80 x=BUFFSIZE-buf_read_pos; | |
81 if(x>len) x=len; | |
82 memcpy(data+len2,buffer[buf_read]+buf_read_pos,x); | |
6184 | 83 SDL_MixAudio(data+len2, data+len2, x, volume); |
972 | 84 len2+=x; len-=x; |
85 buffered_bytes-=x; buf_read_pos+=x; | |
86 if(buf_read_pos>=BUFFSIZE){ | |
87 // block is empty, find next! | |
88 buf_read=(buf_read+1)%NUM_BUFS; | |
89 --full_buffers; | |
90 buf_read_pos=0; | |
91 } | |
92 } | |
93 return len2; | |
94 } | |
95 | |
96 // end ring buffer stuff | |
966 | 97 |
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98 #if defined(HPUX) || defined(sun) && defined(__svr4__) |
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99 /* setenv is missing on solaris and HPUX */ |
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100 static void setenv(const char *name, const char *val, int _xx) |
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101 { |
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102 int len = strlen(name) + strlen(val) + 2; |
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103 char *env = malloc(len); |
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104 |
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105 if (env != NULL) { |
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106 strcpy(env, name); |
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107 strcat(env, "="); |
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108 strcat(env, val); |
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109 putenv(env); |
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110 } |
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111 } |
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112 #endif |
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113 |
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114 |
966 | 115 // to set/get/query special features/parameters |
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116 static int control(int cmd,void *arg){ |
6184 | 117 switch (cmd) { |
118 case AOCONTROL_GET_VOLUME: | |
119 { | |
120 ao_control_vol_t* vol = (ao_control_vol_t*)arg; | |
121 vol->left = vol->right = (float)((volume + 127)/2.55); | |
122 return CONTROL_OK; | |
123 } | |
124 case AOCONTROL_SET_VOLUME: | |
125 { | |
126 float diff; | |
127 ao_control_vol_t* vol = (ao_control_vol_t*)arg; | |
128 diff = (vol->left+vol->right) / 2; | |
129 volume = (int)(diff * 2.55) - 127; | |
130 return CONTROL_OK; | |
131 } | |
132 } | |
133 return -1; | |
966 | 134 } |
135 | |
972 | 136 // SDL Callback function |
137 void outputaudio(void *unused, Uint8 *stream, int len) { | |
138 //SDL_MixAudio(stream, read_buffer(buffers, len), len, SDL_MIX_MAXVOLUME); | |
1066 | 139 //if(!full_buffers) printf("SDL: Buffer underrun!\n"); |
140 | |
972 | 141 read_buffer(stream, len); |
1066 | 142 //printf("SDL: Full Buffers: %i\n", full_buffers); |
966 | 143 } |
144 | |
145 // open & setup audio device | |
146 // return: 1=success 0=fail | |
147 static int init(int rate,int channels,int format,int flags){ | |
148 | |
972 | 149 /* SDL Audio Specifications */ |
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150 SDL_AudioSpec aspec, obtained; |
972 | 151 |
152 int i; | |
153 /* Allocate ring-buffer memory */ | |
983 | 154 for(i=0;i<NUM_BUFS;i++) buffer[i]=(unsigned char *) malloc(BUFFSIZE); |
972 | 155 |
8027 | 156 mp_msg(MSGT_AO,MSGL_INFO,"SDL: Samplerate: %iHz Channels: %s Format %s\n", rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format)); |
983 | 157 |
1189 | 158 if(ao_subdevice) { |
159 setenv("SDL_AUDIODRIVER", ao_subdevice, 1); | |
8027 | 160 mp_msg(MSGT_AO,MSGL_INFO,"SDL: using %s audio driver\n", ao_subdevice); |
1189 | 161 } |
3095 | 162 |
7660 | 163 ao_data.channels=channels; |
164 ao_data.samplerate=rate; | |
165 ao_data.format=format; | |
166 | |
3137 | 167 ao_data.bps=channels*rate; |
3095 | 168 if(format != AFMT_U8 && format != AFMT_S8) |
169 ao_data.bps*=2; | |
966 | 170 |
1066 | 171 /* The desired audio format (see SDL_AudioSpec) */ |
172 switch(format) { | |
173 case AFMT_U8: | |
174 aspec.format = AUDIO_U8; | |
175 break; | |
176 case AFMT_S16_LE: | |
177 aspec.format = AUDIO_S16LSB; | |
178 break; | |
179 case AFMT_S16_BE: | |
180 aspec.format = AUDIO_S16MSB; | |
181 break; | |
182 case AFMT_S8: | |
183 aspec.format = AUDIO_S8; | |
184 break; | |
185 case AFMT_U16_LE: | |
186 aspec.format = AUDIO_U16LSB; | |
187 break; | |
188 case AFMT_U16_BE: | |
189 aspec.format = AUDIO_U16MSB; | |
190 break; | |
191 default: | |
8027 | 192 mp_msg(MSGT_AO,MSGL_WARN,"SDL: Unsupported audio format: 0x%x.\n", format); |
1066 | 193 return 0; |
194 } | |
195 | |
966 | 196 /* The desired audio frequency in samples-per-second. */ |
197 aspec.freq = rate; | |
198 | |
199 /* Number of channels (mono/stereo) */ | |
200 aspec.channels = channels; | |
201 | |
202 /* The desired size of the audio buffer in samples. This number should be a power of two, and may be adjusted by the audio driver to a value more suitable for the hardware. Good values seem to range between 512 and 8192 inclusive, depending on the application and CPU speed. Smaller values yield faster response time, but can lead to underflow if the application is doing heavy processing and cannot fill the audio buffer in time. A stereo sample consists of both right and left channels in LR ordering. Note that the number of samples is directly related to time by the following formula: ms = (samples*1000)/freq */ | |
972 | 203 aspec.samples = SAMPLESIZE; |
966 | 204 |
205 /* This should be set to a function that will be called when the audio device is ready for more data. It is passed a pointer to the audio buffer, and the length in bytes of the audio buffer. This function usually runs in a separate thread, and so you should protect data structures that it accesses by calling SDL_LockAudio and SDL_UnlockAudio in your code. The callback prototype is: | |
206 void callback(void *userdata, Uint8 *stream, int len); userdata is the pointer stored in userdata field of the SDL_AudioSpec. stream is a pointer to the audio buffer you want to fill with information and len is the length of the audio buffer in bytes. */ | |
972 | 207 aspec.callback = outputaudio; |
966 | 208 |
209 /* This pointer is passed as the first parameter to the callback function. */ | |
210 aspec.userdata = NULL; | |
211 | |
972 | 212 /* initialize the SDL Audio system */ |
213 if (SDL_Init (SDL_INIT_AUDIO/*|SDL_INIT_NOPARACHUTE*/)) { | |
8027 | 214 mp_msg(MSGT_AO,MSGL_ERR,"SDL: Initializing of SDL Audio failed: %s.\n", SDL_GetError()); |
972 | 215 return 0; |
216 } | |
217 | |
966 | 218 /* Open the audio device and start playing sound! */ |
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219 if(SDL_OpenAudio(&aspec, &obtained) < 0) { |
8027 | 220 mp_msg(MSGT_AO,MSGL_ERR,"SDL: Unable to open audio: %s\n", SDL_GetError()); |
966 | 221 return(0); |
222 } | |
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223 |
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224 /* did we got what we wanted ? */ |
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225 ao_data.channels=obtained.channels; |
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226 ao_data.samplerate=obtained.freq; |
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227 |
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228 switch(obtained.format) { |
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229 case AUDIO_U8 : |
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230 ao_data.format = AFMT_U8; |
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231 break; |
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232 case AUDIO_S16LSB : |
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233 ao_data.format = AFMT_S16_LE; |
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234 break; |
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235 case AUDIO_S16MSB : |
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236 ao_data.format = AFMT_S16_BE; |
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237 break; |
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238 case AUDIO_S8 : |
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239 ao_data.format = AFMT_S8; |
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240 break; |
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241 case AUDIO_U16LSB : |
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242 ao_data.format = AFMT_U16_LE; |
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243 break; |
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244 case AUDIO_U16MSB : |
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245 ao_data.format = AFMT_U16_BE; |
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246 break; |
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247 default: |
8027 | 248 mp_msg(MSGT_AO,MSGL_WARN,"SDL: Unsupported SDL audio format: 0x%x.\n", obtained.format); |
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249 return 0; |
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250 } |
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251 |
8027 | 252 mp_msg(MSGT_AO,MSGL_V,"SDL: buf size = %d\n",obtained.size); |
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253 ao_data.buffersize=obtained.size; |
974 | 254 |
966 | 255 /* unsilence audio, if callback is ready */ |
256 SDL_PauseAudio(0); | |
257 | |
258 return 1; | |
259 } | |
260 | |
261 // close audio device | |
262 static void uninit(){ | |
8027 | 263 mp_msg(MSGT_AO,MSGL_V,"SDL: Audio Subsystem shutting down!\n"); |
966 | 264 SDL_CloseAudio(); |
972 | 265 SDL_QuitSubSystem(SDL_INIT_AUDIO); |
966 | 266 } |
267 | |
268 // stop playing and empty buffers (for seeking/pause) | |
269 static void reset(){ | |
1066 | 270 |
1091 | 271 //printf("SDL: reset called!\n"); |
1066 | 272 |
972 | 273 /* Reset ring-buffer state */ |
274 buf_read=0; | |
275 buf_write=0; | |
276 buf_read_pos=0; | |
277 buf_write_pos=0; | |
278 | |
279 full_buffers=0; | |
280 buffered_bytes=0; | |
966 | 281 |
282 } | |
283 | |
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284 // stop playing, keep buffers (for pause) |
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285 static void audio_pause() |
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286 { |
1070 | 287 |
1091 | 288 //printf("SDL: audio_pause called!\n"); |
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289 SDL_PauseAudio(1); |
1070 | 290 |
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291 } |
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292 |
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293 // resume playing, after audio_pause() |
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294 static void audio_resume() |
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295 { |
1091 | 296 //printf("SDL: audio_resume called!\n"); |
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297 SDL_PauseAudio(0); |
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298 } |
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299 |
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300 |
966 | 301 // return: how many bytes can be played without blocking |
302 static int get_space(){ | |
972 | 303 return (NUM_BUFS-full_buffers)*BUFFSIZE - buf_write_pos; |
966 | 304 } |
305 | |
306 // plays 'len' bytes of 'data' | |
307 // it should round it down to outburst*n | |
308 // return: number of bytes played | |
309 static int play(void* data,int len,int flags){ | |
972 | 310 |
985 | 311 #if 0 |
972 | 312 int ret; |
966 | 313 |
972 | 314 /* Audio locking prohibits call of outputaudio */ |
966 | 315 SDL_LockAudio(); |
972 | 316 // copy audio stream into ring-buffer |
317 ret = write_buffer(data, len); | |
966 | 318 SDL_UnlockAudio(); |
319 | |
972 | 320 return ret; |
321 #else | |
322 return write_buffer(data, len); | |
323 #endif | |
966 | 324 } |
325 | |
3095 | 326 // return: delay in seconds between first and last sample in buffer |
327 static float get_delay(){ | |
328 return (float)(buffered_bytes + ao_data.buffersize)/(float)ao_data.bps; | |
966 | 329 } |
330 | |
331 | |
332 | |
333 | |
334 | |
335 |