Mercurial > mplayer.hg
annotate libao2/ao_sdl.c @ 12149:2cce4d42f4a8
support for abitrary imaadpcm encoding, ported from ffmpeg
author | alex |
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date | Wed, 07 Apr 2004 12:46:14 +0000 |
parents | 99798c3cdb93 |
children | 75fdb659f5bf |
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 | |
972 | 38 // Samplesize used by the SDLlib AudioSpec struct |
12113 | 39 #ifdef WIN32 |
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40 #define SAMPLESIZE 2048 |
12113 | 41 #else |
42 #define SAMPLESIZE 1024 | |
43 #endif | |
972 | 44 |
45 // General purpose Ring-buffering routines | |
46 | |
974 | 47 #define BUFFSIZE 4096 |
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48 #define NUM_BUFS 8 |
972 | 49 |
50 static unsigned char *buffer[NUM_BUFS]; | |
51 | |
52 static unsigned int buf_read=0; | |
53 static unsigned int buf_write=0; | |
54 static unsigned int buf_read_pos=0; | |
55 static unsigned int buf_write_pos=0; | |
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56 static unsigned char volume=SDL_MIX_MAXVOLUME; |
972 | 57 static int full_buffers=0; |
58 static int buffered_bytes=0; | |
59 | |
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60 |
972 | 61 static int write_buffer(unsigned char* data,int len){ |
62 int len2=0; | |
63 int x; | |
64 while(len>0){ | |
65 if(full_buffers==NUM_BUFS) break; | |
66 x=BUFFSIZE-buf_write_pos; | |
67 if(x>len) x=len; | |
68 memcpy(buffer[buf_write]+buf_write_pos,data+len2,x); | |
69 len2+=x; len-=x; | |
70 buffered_bytes+=x; buf_write_pos+=x; | |
71 if(buf_write_pos>=BUFFSIZE){ | |
72 // block is full, find next! | |
73 buf_write=(buf_write+1)%NUM_BUFS; | |
74 ++full_buffers; | |
75 buf_write_pos=0; | |
76 } | |
77 } | |
78 return len2; | |
79 } | |
80 | |
81 static int read_buffer(unsigned char* data,int len){ | |
82 int len2=0; | |
83 int x; | |
84 while(len>0){ | |
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85 if(buffered_bytes==0) break; // no more data buffered! |
972 | 86 x=BUFFSIZE-buf_read_pos; |
87 if(x>len) x=len; | |
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88 if (x>buffered_bytes) x=buffered_bytes; |
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89 SDL_MixAudio(data+len2,buffer[buf_read]+buf_read_pos,x,volume); |
972 | 90 len2+=x; len-=x; |
91 buffered_bytes-=x; buf_read_pos+=x; | |
92 if(buf_read_pos>=BUFFSIZE){ | |
93 // block is empty, find next! | |
94 buf_read=(buf_read+1)%NUM_BUFS; | |
95 --full_buffers; | |
96 buf_read_pos=0; | |
97 } | |
98 } | |
99 return len2; | |
100 } | |
101 | |
102 // end ring buffer stuff | |
966 | 103 |
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104 #if defined(__MINGW32__) || defined(HPUX) || defined(sgi) || (defined(sun) && defined(__svr4__)) |
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105 /* setenv is missing on win32, solaris, IRIX and HPUX */ |
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106 static void setenv(const char *name, const char *val, int _xx) |
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107 { |
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108 int len = strlen(name) + strlen(val) + 2; |
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109 char *env = malloc(len); |
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110 |
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111 if (env != NULL) { |
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112 strcpy(env, name); |
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113 strcat(env, "="); |
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114 strcat(env, val); |
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115 putenv(env); |
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116 } |
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117 } |
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118 #endif |
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119 |
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120 |
966 | 121 // to set/get/query special features/parameters |
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122 static int control(int cmd,void *arg){ |
6184 | 123 switch (cmd) { |
124 case AOCONTROL_GET_VOLUME: | |
125 { | |
126 ao_control_vol_t* vol = (ao_control_vol_t*)arg; | |
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127 vol->left = vol->right = volume * 100 / SDL_MIX_MAXVOLUME; |
6184 | 128 return CONTROL_OK; |
129 } | |
130 case AOCONTROL_SET_VOLUME: | |
131 { | |
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132 int diff; |
6184 | 133 ao_control_vol_t* vol = (ao_control_vol_t*)arg; |
134 diff = (vol->left+vol->right) / 2; | |
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135 volume = diff * SDL_MIX_MAXVOLUME / 100; |
6184 | 136 return CONTROL_OK; |
137 } | |
138 } | |
139 return -1; | |
966 | 140 } |
141 | |
972 | 142 // SDL Callback function |
143 void outputaudio(void *unused, Uint8 *stream, int len) { | |
144 //SDL_MixAudio(stream, read_buffer(buffers, len), len, SDL_MIX_MAXVOLUME); | |
1066 | 145 //if(!full_buffers) printf("SDL: Buffer underrun!\n"); |
146 | |
972 | 147 read_buffer(stream, len); |
1066 | 148 //printf("SDL: Full Buffers: %i\n", full_buffers); |
966 | 149 } |
150 | |
151 // open & setup audio device | |
152 // return: 1=success 0=fail | |
153 static int init(int rate,int channels,int format,int flags){ | |
154 | |
972 | 155 /* SDL Audio Specifications */ |
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156 SDL_AudioSpec aspec, obtained; |
972 | 157 |
158 int i; | |
159 /* Allocate ring-buffer memory */ | |
983 | 160 for(i=0;i<NUM_BUFS;i++) buffer[i]=(unsigned char *) malloc(BUFFSIZE); |
972 | 161 |
8027 | 162 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 | 163 |
1189 | 164 if(ao_subdevice) { |
165 setenv("SDL_AUDIODRIVER", ao_subdevice, 1); | |
8027 | 166 mp_msg(MSGT_AO,MSGL_INFO,"SDL: using %s audio driver\n", ao_subdevice); |
1189 | 167 } |
3095 | 168 |
7660 | 169 ao_data.channels=channels; |
170 ao_data.samplerate=rate; | |
171 ao_data.format=format; | |
172 | |
3137 | 173 ao_data.bps=channels*rate; |
3095 | 174 if(format != AFMT_U8 && format != AFMT_S8) |
175 ao_data.bps*=2; | |
966 | 176 |
1066 | 177 /* The desired audio format (see SDL_AudioSpec) */ |
178 switch(format) { | |
179 case AFMT_U8: | |
180 aspec.format = AUDIO_U8; | |
181 break; | |
182 case AFMT_S16_LE: | |
183 aspec.format = AUDIO_S16LSB; | |
184 break; | |
185 case AFMT_S16_BE: | |
186 aspec.format = AUDIO_S16MSB; | |
187 break; | |
188 case AFMT_S8: | |
189 aspec.format = AUDIO_S8; | |
190 break; | |
191 case AFMT_U16_LE: | |
192 aspec.format = AUDIO_U16LSB; | |
193 break; | |
194 case AFMT_U16_BE: | |
195 aspec.format = AUDIO_U16MSB; | |
196 break; | |
197 default: | |
8027 | 198 mp_msg(MSGT_AO,MSGL_WARN,"SDL: Unsupported audio format: 0x%x.\n", format); |
1066 | 199 return 0; |
200 } | |
201 | |
966 | 202 /* The desired audio frequency in samples-per-second. */ |
203 aspec.freq = rate; | |
204 | |
205 /* Number of channels (mono/stereo) */ | |
206 aspec.channels = channels; | |
207 | |
208 /* 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 | 209 aspec.samples = SAMPLESIZE; |
966 | 210 |
211 /* 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: | |
212 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 | 213 aspec.callback = outputaudio; |
966 | 214 |
215 /* This pointer is passed as the first parameter to the callback function. */ | |
216 aspec.userdata = NULL; | |
217 | |
972 | 218 /* initialize the SDL Audio system */ |
219 if (SDL_Init (SDL_INIT_AUDIO/*|SDL_INIT_NOPARACHUTE*/)) { | |
8027 | 220 mp_msg(MSGT_AO,MSGL_ERR,"SDL: Initializing of SDL Audio failed: %s.\n", SDL_GetError()); |
972 | 221 return 0; |
222 } | |
223 | |
966 | 224 /* Open the audio device and start playing sound! */ |
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225 if(SDL_OpenAudio(&aspec, &obtained) < 0) { |
8027 | 226 mp_msg(MSGT_AO,MSGL_ERR,"SDL: Unable to open audio: %s\n", SDL_GetError()); |
966 | 227 return(0); |
228 } | |
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229 |
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230 /* did we got what we wanted ? */ |
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231 ao_data.channels=obtained.channels; |
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232 ao_data.samplerate=obtained.freq; |
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233 |
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234 switch(obtained.format) { |
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235 case AUDIO_U8 : |
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236 ao_data.format = AFMT_U8; |
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237 break; |
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238 case AUDIO_S16LSB : |
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239 ao_data.format = AFMT_S16_LE; |
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240 break; |
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241 case AUDIO_S16MSB : |
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242 ao_data.format = AFMT_S16_BE; |
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243 break; |
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244 case AUDIO_S8 : |
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245 ao_data.format = AFMT_S8; |
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246 break; |
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247 case AUDIO_U16LSB : |
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248 ao_data.format = AFMT_U16_LE; |
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249 break; |
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250 case AUDIO_U16MSB : |
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251 ao_data.format = AFMT_U16_BE; |
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252 break; |
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253 default: |
8027 | 254 mp_msg(MSGT_AO,MSGL_WARN,"SDL: Unsupported SDL audio format: 0x%x.\n", obtained.format); |
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255 return 0; |
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256 } |
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257 |
8027 | 258 mp_msg(MSGT_AO,MSGL_V,"SDL: buf size = %d\n",obtained.size); |
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259 ao_data.buffersize=obtained.size; |
974 | 260 |
966 | 261 /* unsilence audio, if callback is ready */ |
262 SDL_PauseAudio(0); | |
263 | |
264 return 1; | |
265 } | |
266 | |
267 // close audio device | |
12145 | 268 static void uninit(int immed){ |
8027 | 269 mp_msg(MSGT_AO,MSGL_V,"SDL: Audio Subsystem shutting down!\n"); |
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270 while(buffered_bytes > 0) |
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271 usec_sleep(50000); |
966 | 272 SDL_CloseAudio(); |
972 | 273 SDL_QuitSubSystem(SDL_INIT_AUDIO); |
966 | 274 } |
275 | |
276 // stop playing and empty buffers (for seeking/pause) | |
277 static void reset(){ | |
1066 | 278 |
1091 | 279 //printf("SDL: reset called!\n"); |
1066 | 280 |
972 | 281 /* Reset ring-buffer state */ |
282 buf_read=0; | |
283 buf_write=0; | |
284 buf_read_pos=0; | |
285 buf_write_pos=0; | |
286 | |
287 full_buffers=0; | |
288 buffered_bytes=0; | |
966 | 289 |
290 } | |
291 | |
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292 // stop playing, keep buffers (for pause) |
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293 static void audio_pause() |
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294 { |
1070 | 295 |
1091 | 296 //printf("SDL: audio_pause called!\n"); |
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297 SDL_PauseAudio(1); |
1070 | 298 |
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299 } |
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300 |
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301 // resume playing, after audio_pause() |
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302 static void audio_resume() |
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303 { |
1091 | 304 //printf("SDL: audio_resume called!\n"); |
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305 SDL_PauseAudio(0); |
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306 } |
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307 |
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308 |
966 | 309 // return: how many bytes can be played without blocking |
310 static int get_space(){ | |
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311 return NUM_BUFS*BUFFSIZE - buffered_bytes; |
966 | 312 } |
313 | |
314 // plays 'len' bytes of 'data' | |
315 // it should round it down to outburst*n | |
316 // return: number of bytes played | |
317 static int play(void* data,int len,int flags){ | |
972 | 318 |
985 | 319 #if 0 |
972 | 320 int ret; |
966 | 321 |
972 | 322 /* Audio locking prohibits call of outputaudio */ |
966 | 323 SDL_LockAudio(); |
972 | 324 // copy audio stream into ring-buffer |
325 ret = write_buffer(data, len); | |
966 | 326 SDL_UnlockAudio(); |
327 | |
972 | 328 return ret; |
329 #else | |
330 return write_buffer(data, len); | |
331 #endif | |
966 | 332 } |
333 | |
3095 | 334 // return: delay in seconds between first and last sample in buffer |
335 static float get_delay(){ | |
336 return (float)(buffered_bytes + ao_data.buffersize)/(float)ao_data.bps; | |
966 | 337 } |
338 | |
339 | |
340 | |
341 | |
342 | |
343 |