Mercurial > mplayer.hg
annotate libao2/ao_sdl.c @ 13325:f26d2c9aec04
2pass encoding support for x264(r46).
patch by Loren Merritt and Jeff Clagg
author | iive |
---|---|
date | Sun, 12 Sep 2004 15:36:43 +0000 |
parents | 7b9b4f07d2c4 |
children | c1955840883d |
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. | |
11750 | 5 * (http://www.mplayerhq.hu) |
972 | 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> |
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15 #include <string.h> |
966 | 16 |
8027 | 17 #include "../config.h" |
18 #include "../mp_msg.h" | |
19 | |
966 | 20 #include "audio_out.h" |
21 #include "audio_out_internal.h" | |
1066 | 22 #include "afmt.h" |
6184 | 23 #include <SDL.h> |
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24 #include "osdep/timer.h" |
966 | 25 |
972 | 26 #include "../libvo/fastmemcpy.h" |
27 | |
966 | 28 static ao_info_t info = |
29 { | |
30 "SDLlib audio output", | |
31 "sdl", | |
32 "Felix Buenemann <atmosfear@users.sourceforge.net>", | |
33 "" | |
34 }; | |
35 | |
36 LIBAO_EXTERN(sdl) | |
37 | |
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38 // turn this on if you want to use the slower SDL_MixAudio |
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39 #undef USE_SDL_INTERNAL_MIXER |
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40 |
972 | 41 // Samplesize used by the SDLlib AudioSpec struct |
12113 | 42 #ifdef WIN32 |
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43 #define SAMPLESIZE 2048 |
12113 | 44 #else |
45 #define SAMPLESIZE 1024 | |
46 #endif | |
972 | 47 |
48 // General purpose Ring-buffering routines | |
49 | |
974 | 50 #define BUFFSIZE 4096 |
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51 #define NUM_BUFS 8 |
972 | 52 |
53 static unsigned char *buffer[NUM_BUFS]; | |
54 | |
55 static unsigned int buf_read=0; | |
56 static unsigned int buf_write=0; | |
57 static unsigned int buf_read_pos=0; | |
58 static unsigned int buf_write_pos=0; | |
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59 #ifdef USE_SDL_INTERNAL_MIXER |
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60 static unsigned char volume=SDL_MIX_MAXVOLUME; |
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61 #endif |
972 | 62 static int full_buffers=0; |
63 static int buffered_bytes=0; | |
64 | |
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65 |
972 | 66 static int write_buffer(unsigned char* data,int len){ |
67 int len2=0; | |
68 int x; | |
69 while(len>0){ | |
70 if(full_buffers==NUM_BUFS) break; | |
71 x=BUFFSIZE-buf_write_pos; | |
72 if(x>len) x=len; | |
73 memcpy(buffer[buf_write]+buf_write_pos,data+len2,x); | |
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74 if (buf_write_pos==0) |
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75 ++full_buffers; |
972 | 76 len2+=x; len-=x; |
77 buffered_bytes+=x; buf_write_pos+=x; | |
78 if(buf_write_pos>=BUFFSIZE){ | |
79 // block is full, find next! | |
80 buf_write=(buf_write+1)%NUM_BUFS; | |
81 buf_write_pos=0; | |
82 } | |
83 } | |
84 return len2; | |
85 } | |
86 | |
87 static int read_buffer(unsigned char* data,int len){ | |
88 int len2=0; | |
89 int x; | |
90 while(len>0){ | |
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91 if(buffered_bytes==0) break; // no more data buffered! |
972 | 92 x=BUFFSIZE-buf_read_pos; |
93 if(x>len) x=len; | |
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94 if (x>buffered_bytes) x=buffered_bytes; |
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95 #ifdef USE_SDL_INTERNAL_MIXER |
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96 SDL_MixAudio(data+len2,buffer[buf_read]+buf_read_pos,x,volume); |
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97 #else |
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98 memcpy(data+len2,buffer[buf_read]+buf_read_pos,x); |
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99 #endif |
972 | 100 len2+=x; len-=x; |
101 buffered_bytes-=x; buf_read_pos+=x; | |
102 if(buf_read_pos>=BUFFSIZE){ | |
103 // block is empty, find next! | |
104 buf_read=(buf_read+1)%NUM_BUFS; | |
105 --full_buffers; | |
106 buf_read_pos=0; | |
107 } | |
108 } | |
109 return len2; | |
110 } | |
111 | |
112 // end ring buffer stuff | |
966 | 113 |
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114 #if defined(__MINGW32__) || defined(HPUX) || defined(sgi) || (defined(sun) && defined(__svr4__)) |
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115 /* setenv is missing on win32, solaris, IRIX and HPUX */ |
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116 static void setenv(const char *name, const char *val, int _xx) |
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117 { |
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118 int len = strlen(name) + strlen(val) + 2; |
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119 char *env = malloc(len); |
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120 |
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121 if (env != NULL) { |
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122 strcpy(env, name); |
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123 strcat(env, "="); |
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124 strcat(env, val); |
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125 putenv(env); |
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126 } |
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127 } |
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128 #endif |
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129 |
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130 |
966 | 131 // to set/get/query special features/parameters |
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132 static int control(int cmd,void *arg){ |
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133 #ifdef USE_SDL_INTERNAL_MIXER |
6184 | 134 switch (cmd) { |
135 case AOCONTROL_GET_VOLUME: | |
136 { | |
137 ao_control_vol_t* vol = (ao_control_vol_t*)arg; | |
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138 vol->left = vol->right = volume * 100 / SDL_MIX_MAXVOLUME; |
6184 | 139 return CONTROL_OK; |
140 } | |
141 case AOCONTROL_SET_VOLUME: | |
142 { | |
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143 int diff; |
6184 | 144 ao_control_vol_t* vol = (ao_control_vol_t*)arg; |
145 diff = (vol->left+vol->right) / 2; | |
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146 volume = diff * SDL_MIX_MAXVOLUME / 100; |
6184 | 147 return CONTROL_OK; |
148 } | |
149 } | |
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150 #endif |
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151 return CONTROL_UNKNOWN; |
966 | 152 } |
153 | |
972 | 154 // SDL Callback function |
155 void outputaudio(void *unused, Uint8 *stream, int len) { | |
156 //SDL_MixAudio(stream, read_buffer(buffers, len), len, SDL_MIX_MAXVOLUME); | |
1066 | 157 //if(!full_buffers) printf("SDL: Buffer underrun!\n"); |
158 | |
972 | 159 read_buffer(stream, len); |
1066 | 160 //printf("SDL: Full Buffers: %i\n", full_buffers); |
966 | 161 } |
162 | |
163 // open & setup audio device | |
164 // return: 1=success 0=fail | |
165 static int init(int rate,int channels,int format,int flags){ | |
166 | |
972 | 167 /* SDL Audio Specifications */ |
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168 SDL_AudioSpec aspec, obtained; |
972 | 169 |
170 int i; | |
171 /* Allocate ring-buffer memory */ | |
983 | 172 for(i=0;i<NUM_BUFS;i++) buffer[i]=(unsigned char *) malloc(BUFFSIZE); |
972 | 173 |
8027 | 174 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 | 175 |
1189 | 176 if(ao_subdevice) { |
177 setenv("SDL_AUDIODRIVER", ao_subdevice, 1); | |
8027 | 178 mp_msg(MSGT_AO,MSGL_INFO,"SDL: using %s audio driver\n", ao_subdevice); |
1189 | 179 } |
3095 | 180 |
7660 | 181 ao_data.channels=channels; |
182 ao_data.samplerate=rate; | |
183 ao_data.format=format; | |
184 | |
3137 | 185 ao_data.bps=channels*rate; |
3095 | 186 if(format != AFMT_U8 && format != AFMT_S8) |
187 ao_data.bps*=2; | |
966 | 188 |
1066 | 189 /* The desired audio format (see SDL_AudioSpec) */ |
190 switch(format) { | |
191 case AFMT_U8: | |
192 aspec.format = AUDIO_U8; | |
193 break; | |
194 case AFMT_S16_LE: | |
195 aspec.format = AUDIO_S16LSB; | |
196 break; | |
197 case AFMT_S16_BE: | |
198 aspec.format = AUDIO_S16MSB; | |
199 break; | |
200 case AFMT_S8: | |
201 aspec.format = AUDIO_S8; | |
202 break; | |
203 case AFMT_U16_LE: | |
204 aspec.format = AUDIO_U16LSB; | |
205 break; | |
206 case AFMT_U16_BE: | |
207 aspec.format = AUDIO_U16MSB; | |
208 break; | |
209 default: | |
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210 aspec.format = AUDIO_S16LSB; |
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211 ao_data.format = AFMT_S16_LE; |
8027 | 212 mp_msg(MSGT_AO,MSGL_WARN,"SDL: Unsupported audio format: 0x%x.\n", format); |
1066 | 213 } |
214 | |
966 | 215 /* The desired audio frequency in samples-per-second. */ |
216 aspec.freq = rate; | |
217 | |
218 /* Number of channels (mono/stereo) */ | |
219 aspec.channels = channels; | |
220 | |
221 /* 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 | 222 aspec.samples = SAMPLESIZE; |
966 | 223 |
224 /* 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: | |
225 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 | 226 aspec.callback = outputaudio; |
966 | 227 |
228 /* This pointer is passed as the first parameter to the callback function. */ | |
229 aspec.userdata = NULL; | |
230 | |
972 | 231 /* initialize the SDL Audio system */ |
232 if (SDL_Init (SDL_INIT_AUDIO/*|SDL_INIT_NOPARACHUTE*/)) { | |
8027 | 233 mp_msg(MSGT_AO,MSGL_ERR,"SDL: Initializing of SDL Audio failed: %s.\n", SDL_GetError()); |
972 | 234 return 0; |
235 } | |
236 | |
966 | 237 /* Open the audio device and start playing sound! */ |
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238 if(SDL_OpenAudio(&aspec, &obtained) < 0) { |
8027 | 239 mp_msg(MSGT_AO,MSGL_ERR,"SDL: Unable to open audio: %s\n", SDL_GetError()); |
966 | 240 return(0); |
241 } | |
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242 |
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243 /* did we got what we wanted ? */ |
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244 ao_data.channels=obtained.channels; |
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245 ao_data.samplerate=obtained.freq; |
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246 |
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247 switch(obtained.format) { |
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248 case AUDIO_U8 : |
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249 ao_data.format = AFMT_U8; |
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250 break; |
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251 case AUDIO_S16LSB : |
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252 ao_data.format = AFMT_S16_LE; |
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253 break; |
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254 case AUDIO_S16MSB : |
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255 ao_data.format = AFMT_S16_BE; |
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256 break; |
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257 case AUDIO_S8 : |
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258 ao_data.format = AFMT_S8; |
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259 break; |
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260 case AUDIO_U16LSB : |
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261 ao_data.format = AFMT_U16_LE; |
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262 break; |
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263 case AUDIO_U16MSB : |
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264 ao_data.format = AFMT_U16_BE; |
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265 break; |
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266 default: |
8027 | 267 mp_msg(MSGT_AO,MSGL_WARN,"SDL: Unsupported SDL audio format: 0x%x.\n", obtained.format); |
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268 return 0; |
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269 } |
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270 |
8027 | 271 mp_msg(MSGT_AO,MSGL_V,"SDL: buf size = %d\n",obtained.size); |
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272 ao_data.buffersize=obtained.size; |
974 | 273 |
966 | 274 /* unsilence audio, if callback is ready */ |
275 SDL_PauseAudio(0); | |
276 | |
277 return 1; | |
278 } | |
279 | |
280 // close audio device | |
12145 | 281 static void uninit(int immed){ |
8027 | 282 mp_msg(MSGT_AO,MSGL_V,"SDL: Audio Subsystem shutting down!\n"); |
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283 while(buffered_bytes > 0) |
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284 usec_sleep(50000); |
966 | 285 SDL_CloseAudio(); |
972 | 286 SDL_QuitSubSystem(SDL_INIT_AUDIO); |
966 | 287 } |
288 | |
289 // stop playing and empty buffers (for seeking/pause) | |
290 static void reset(){ | |
1066 | 291 |
1091 | 292 //printf("SDL: reset called!\n"); |
1066 | 293 |
972 | 294 /* Reset ring-buffer state */ |
295 buf_read=0; | |
296 buf_write=0; | |
297 buf_read_pos=0; | |
298 buf_write_pos=0; | |
299 | |
300 full_buffers=0; | |
301 buffered_bytes=0; | |
966 | 302 |
303 } | |
304 | |
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305 // stop playing, keep buffers (for pause) |
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306 static void audio_pause() |
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307 { |
1070 | 308 |
1091 | 309 //printf("SDL: audio_pause called!\n"); |
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310 SDL_PauseAudio(1); |
1070 | 311 |
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312 } |
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313 |
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314 // resume playing, after audio_pause() |
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315 static void audio_resume() |
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316 { |
1091 | 317 //printf("SDL: audio_resume called!\n"); |
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318 SDL_PauseAudio(0); |
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319 } |
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320 |
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321 |
966 | 322 // return: how many bytes can be played without blocking |
323 static int get_space(){ | |
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324 return NUM_BUFS*BUFFSIZE - buffered_bytes; |
966 | 325 } |
326 | |
327 // plays 'len' bytes of 'data' | |
328 // it should round it down to outburst*n | |
329 // return: number of bytes played | |
330 static int play(void* data,int len,int flags){ | |
972 | 331 |
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332 len = (len/ao_data.outburst)*ao_data.outburst; |
985 | 333 #if 0 |
972 | 334 int ret; |
966 | 335 |
972 | 336 /* Audio locking prohibits call of outputaudio */ |
966 | 337 SDL_LockAudio(); |
972 | 338 // copy audio stream into ring-buffer |
339 ret = write_buffer(data, len); | |
966 | 340 SDL_UnlockAudio(); |
341 | |
972 | 342 return ret; |
343 #else | |
344 return write_buffer(data, len); | |
345 #endif | |
966 | 346 } |
347 | |
3095 | 348 // return: delay in seconds between first and last sample in buffer |
349 static float get_delay(){ | |
350 return (float)(buffered_bytes + ao_data.buffersize)/(float)ao_data.bps; | |
966 | 351 } |
352 | |
353 | |
354 | |
355 | |
356 | |
357 |