Mercurial > mplayer.hg
annotate libao2/ao_esd.c @ 13316:0d17bef9894d
* Changed malloc and strncpy to strdup. Less code.
* More error checking. If malloc or strdup fails, print message and exit.
* Free malloc'd memory when uninit is called.
* Moved default of jpeg_outdir to preinit, so it is always malloc'd and can
easily be freed at uninit.
author | ivo |
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date | Sat, 11 Sep 2004 19:59:31 +0000 |
parents | 99798c3cdb93 |
children | c1955840883d |
rev | line source |
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8572 | 1 /* |
2 * ao_esd - EsounD audio output driver for MPlayer | |
3 * | |
4 * Juergen Keil <jk@tools.de> | |
5 * | |
6 * This driver is distributed under the terms of the GPL | |
7 * | |
8 * TODO / known problems: | |
9 * - does not work well when the esd daemon has autostandby disabled | |
10 * (workaround: run esd with option "-as 2" - fortunatelly this is | |
11 * the default) | |
12 * - plays noise on a linux 2.4.4 kernel with a SB16PCI card, when using | |
13 * a local tcp connection to the esd daemon; there is no noise when using | |
14 * a unix domain socket connection. | |
15 * (there are EIO errors reported by the sound card driver, so this is | |
16 * most likely a linux sound card driver problem) | |
17 */ | |
18 | |
19 #include "../config.h" | |
20 | |
21 #include <sys/types.h> | |
22 #include <sys/time.h> | |
23 #include <sys/socket.h> | |
24 #include <stdio.h> | |
25 #include <string.h> | |
26 #include <unistd.h> | |
27 #include <errno.h> | |
28 #include <fcntl.h> | |
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29 #include <time.h> |
8572 | 30 #ifdef __svr4__ |
31 #include <stropts.h> | |
32 #endif | |
33 #include <esd.h> | |
34 | |
35 #include "audio_out.h" | |
36 #include "audio_out_internal.h" | |
37 #include "afmt.h" | |
38 #include "../config.h" | |
39 #include "../mp_msg.h" | |
40 | |
41 | |
42 #undef ESD_DEBUG | |
43 | |
44 #if ESD_DEBUG | |
45 #define dprintf(...) printf(__VA_ARGS__) | |
46 #else | |
47 #define dprintf(...) /**/ | |
48 #endif | |
49 | |
50 | |
51 #define ESD_CLIENT_NAME "MPlayer" | |
52 #define ESD_MAX_DELAY (1.0f) /* max amount of data buffered in esd (#sec) */ | |
53 | |
54 static ao_info_t info = | |
55 { | |
56 "EsounD audio output", | |
57 "esd", | |
58 "Juergen Keil <jk@tools.de>", | |
59 "" | |
60 }; | |
61 | |
62 LIBAO_EXTERN(esd) | |
63 | |
64 static int esd_fd = -1; | |
65 static int esd_play_fd = -1; | |
66 static esd_server_info_t *esd_svinfo; | |
67 static int esd_latency; | |
68 static int esd_bytes_per_sample; | |
69 static unsigned long esd_samples_written; | |
70 static struct timeval esd_play_start; | |
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71 extern float audio_delay; |
8572 | 72 |
73 /* | |
74 * to set/get/query special features/parameters | |
75 */ | |
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76 static int control(int cmd, void *arg) |
8572 | 77 { |
78 esd_player_info_t *esd_pi; | |
79 esd_info_t *esd_i; | |
80 time_t now; | |
81 static time_t vol_cache_time; | |
82 static ao_control_vol_t vol_cache; | |
83 | |
84 switch (cmd) { | |
85 case AOCONTROL_GET_VOLUME: | |
86 time(&now); | |
87 if (now == vol_cache_time) { | |
88 *(ao_control_vol_t *)arg = vol_cache; | |
89 return CONTROL_OK; | |
90 } | |
91 | |
92 dprintf("esd: get vol\n"); | |
93 if ((esd_i = esd_get_all_info(esd_fd)) == NULL) | |
94 return CONTROL_ERROR; | |
95 | |
96 for (esd_pi = esd_i->player_list; esd_pi != NULL; esd_pi = esd_pi->next) | |
97 if (strcmp(esd_pi->name, ESD_CLIENT_NAME) == 0) | |
98 break; | |
99 | |
100 if (esd_pi != NULL) { | |
101 ao_control_vol_t *vol = (ao_control_vol_t *)arg; | |
102 vol->left = esd_pi->left_vol_scale * 100 / ESD_VOLUME_BASE; | |
103 vol->right = esd_pi->right_vol_scale * 100 / ESD_VOLUME_BASE; | |
104 | |
105 vol_cache = *vol; | |
106 vol_cache_time = now; | |
107 } | |
108 esd_free_all_info(esd_i); | |
109 | |
110 return CONTROL_OK; | |
111 | |
112 case AOCONTROL_SET_VOLUME: | |
113 dprintf("esd: set vol\n"); | |
114 if ((esd_i = esd_get_all_info(esd_fd)) == NULL) | |
115 return CONTROL_ERROR; | |
116 | |
117 for (esd_pi = esd_i->player_list; esd_pi != NULL; esd_pi = esd_pi->next) | |
118 if (strcmp(esd_pi->name, ESD_CLIENT_NAME) == 0) | |
119 break; | |
120 | |
121 if (esd_pi != NULL) { | |
122 ao_control_vol_t *vol = (ao_control_vol_t *)arg; | |
123 esd_set_stream_pan(esd_fd, esd_pi->source_id, | |
124 vol->left * ESD_VOLUME_BASE / 100, | |
125 vol->right * ESD_VOLUME_BASE / 100); | |
126 | |
127 vol_cache = *vol; | |
128 time(&vol_cache_time); | |
129 } | |
130 esd_free_all_info(esd_i); | |
131 return CONTROL_OK; | |
132 | |
133 default: | |
134 return CONTROL_UNKNOWN; | |
135 } | |
136 } | |
137 | |
138 | |
139 /* | |
140 * open & setup audio device | |
141 * return: 1=success 0=fail | |
142 */ | |
143 static int init(int rate_hz, int channels, int format, int flags) | |
144 { | |
145 esd_format_t esd_fmt; | |
146 int bytes_per_sample; | |
147 int fl; | |
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148 char *server = ao_subdevice; /* NULL for localhost */ |
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149 float lag_seconds, lag_net, lag_serv; |
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150 struct timeval proto_start, proto_end; |
8572 | 151 |
152 if (esd_fd < 0) { | |
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153 esd_fd = esd_open_sound(server); |
8572 | 154 if (esd_fd < 0) { |
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155 mp_msg(MSGT_AO, MSGL_ERR, |
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156 "AO: [esd] esd_open_sound failed: %s\n", |
8572 | 157 strerror(errno)); |
158 return 0; | |
159 } | |
160 | |
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161 /* get server info, and measure network latency */ |
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162 gettimeofday(&proto_start, NULL); |
8572 | 163 esd_svinfo = esd_get_server_info(esd_fd); |
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164 if(server) { |
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165 gettimeofday(&proto_end, NULL); |
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166 lag_net = (proto_end.tv_sec - proto_start.tv_sec) + |
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167 (proto_end.tv_usec - proto_start.tv_usec) / 1000000.0; |
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168 lag_net /= 2.0; /* round trip -> one way */ |
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169 } else |
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170 lag_net = 0.0; /* no network lag */ |
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171 |
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172 /* |
8572 | 173 if (esd_svinfo) { |
174 mp_msg(MSGT_AO, MSGL_INFO, "AO: [esd] server info:\n"); | |
175 esd_print_server_info(esd_svinfo); | |
176 } | |
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177 */ |
8572 | 178 } |
179 | |
180 esd_fmt = ESD_STREAM | ESD_PLAY; | |
181 | |
182 #if ESD_RESAMPLES | |
183 /* let the esd daemon convert sample rate */ | |
184 #else | |
185 /* let mplayer's audio filter convert the sample rate */ | |
186 if (esd_svinfo != NULL) | |
187 rate_hz = esd_svinfo->rate; | |
188 #endif | |
189 ao_data.samplerate = rate_hz; | |
190 | |
191 /* EsounD can play mono or stereo */ | |
192 switch (channels) { | |
193 case 1: | |
194 esd_fmt |= ESD_MONO; | |
195 ao_data.channels = bytes_per_sample = 1; | |
196 break; | |
197 default: | |
198 esd_fmt |= ESD_STEREO; | |
199 ao_data.channels = bytes_per_sample = 2; | |
200 break; | |
201 } | |
202 | |
203 /* EsounD can play 8bit unsigned and 16bit signed native */ | |
204 switch (format) { | |
205 case AFMT_S8: | |
206 case AFMT_U8: | |
207 esd_fmt |= ESD_BITS8; | |
208 ao_data.format = AFMT_U8; | |
209 break; | |
210 default: | |
211 esd_fmt |= ESD_BITS16; | |
212 ao_data.format = AFMT_S16_NE; | |
213 bytes_per_sample *= 2; | |
214 break; | |
215 } | |
216 | |
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217 /* modify audio_delay depending on esd_latency |
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218 * latency is number of samples @ 44.1khz stereo 16 bit |
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219 * adjust according to rate_hz & bytes_per_sample |
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220 */ |
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221 #ifdef HAVE_ESD_LATENCY |
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222 esd_latency = esd_get_latency(esd_fd); |
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223 #else |
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224 esd_latency = ((channels == 1 ? 2 : 1) * ESD_DEFAULT_RATE * |
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225 (ESD_BUF_SIZE + 64 * (4.0f / bytes_per_sample)) |
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226 ) / rate_hz; |
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227 esd_latency += ESD_BUF_SIZE * 2; |
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228 #endif |
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229 if(esd_latency > 0) { |
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230 lag_serv = (esd_latency * 4.0f) / (bytes_per_sample * rate_hz); |
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231 lag_seconds = lag_net + lag_serv; |
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232 audio_delay += lag_seconds; |
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233 mp_msg(MSGT_AO, MSGL_INFO, |
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234 "AO: [esd] latency: [server: %0.2fs, net: %0.2fs] " |
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235 "(adjust %0.2fs)\n", lag_serv, lag_net, lag_seconds); |
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236 } |
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237 |
8572 | 238 esd_play_fd = esd_play_stream_fallback(esd_fmt, rate_hz, |
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239 server, ESD_CLIENT_NAME); |
8572 | 240 if (esd_play_fd < 0) { |
241 mp_msg(MSGT_AO, MSGL_ERR, | |
242 "AO: [esd] failed to open esd playback stream: %s\n", | |
243 strerror(errno)); | |
244 return 0; | |
245 } | |
246 | |
247 /* enable non-blocking i/o on the socket connection to the esd server */ | |
248 if ((fl = fcntl(esd_play_fd, F_GETFL)) >= 0) | |
249 fcntl(esd_play_fd, F_SETFL, O_NDELAY|fl); | |
250 | |
251 #if ESD_DEBUG | |
252 { | |
253 int sbuf, rbuf, len; | |
254 len = sizeof(sbuf); | |
255 getsockopt(esd_play_fd, SOL_SOCKET, SO_SNDBUF, &sbuf, &len); | |
256 len = sizeof(rbuf); | |
257 getsockopt(esd_play_fd, SOL_SOCKET, SO_RCVBUF, &rbuf, &len); | |
258 dprintf("esd: send/receive socket buffer space %d/%d bytes\n", | |
259 sbuf, rbuf); | |
260 } | |
261 #endif | |
262 | |
263 ao_data.bps = bytes_per_sample * rate_hz; | |
264 ao_data.outburst = ao_data.bps > 100000 ? 4*ESD_BUF_SIZE : 2*ESD_BUF_SIZE; | |
265 | |
266 esd_play_start.tv_sec = 0; | |
267 esd_samples_written = 0; | |
268 esd_bytes_per_sample = bytes_per_sample; | |
269 | |
270 return 1; | |
271 } | |
272 | |
273 | |
274 /* | |
275 * close audio device | |
276 */ | |
12145 | 277 static void uninit(int immed) |
8572 | 278 { |
279 if (esd_play_fd >= 0) { | |
280 esd_close(esd_play_fd); | |
281 esd_play_fd = -1; | |
282 } | |
283 | |
284 if (esd_svinfo) { | |
285 esd_free_server_info(esd_svinfo); | |
286 esd_svinfo = NULL; | |
287 } | |
288 | |
289 if (esd_fd >= 0) { | |
290 esd_close(esd_fd); | |
291 esd_fd = -1; | |
292 } | |
293 } | |
294 | |
295 | |
296 /* | |
297 * plays 'len' bytes of 'data' | |
298 * it should round it down to outburst*n | |
299 * return: number of bytes played | |
300 */ | |
301 static int play(void* data, int len, int flags) | |
302 { | |
303 int offs; | |
304 int nwritten; | |
305 int nsamples; | |
306 int remainder, n; | |
307 int saved_fl; | |
308 | |
309 /* round down buffersize to a multiple of ESD_BUF_SIZE bytes */ | |
310 len = len / ESD_BUF_SIZE * ESD_BUF_SIZE; | |
311 if (len <= 0) | |
312 return 0; | |
313 | |
314 #define SINGLE_WRITE 0 | |
315 #if SINGLE_WRITE | |
316 nwritten = write(esd_play_fd, data, len); | |
317 #else | |
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318 for (offs = 0, nwritten=0; offs + ESD_BUF_SIZE <= len; offs += ESD_BUF_SIZE) { |
8572 | 319 /* |
320 * note: we're writing to a non-blocking socket here. | |
321 * A partial write means, that the socket buffer is full. | |
322 */ | |
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323 n = write(esd_play_fd, (char*)data + offs, ESD_BUF_SIZE); |
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324 if ( n < 0 ) { |
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325 if ( errno != EAGAIN ) |
8572 | 326 dprintf("esd play: write failed: %s\n", strerror(errno)); |
327 break; | |
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328 } else if ( n != ESD_BUF_SIZE ) { |
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329 nwritten += n; |
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330 break; |
8572 | 331 } else |
332 nwritten += n; | |
333 } | |
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334 #endif |
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335 |
8572 | 336 if (nwritten > 0) { |
337 if (!esd_play_start.tv_sec) | |
338 gettimeofday(&esd_play_start, NULL); | |
339 nsamples = nwritten / esd_bytes_per_sample; | |
340 esd_samples_written += nsamples; | |
341 | |
342 dprintf("esd play: %d %lu\n", nsamples, esd_samples_written); | |
343 } else { | |
344 dprintf("esd play: blocked / %lu\n", esd_samples_written); | |
345 } | |
346 | |
347 return nwritten; | |
348 } | |
349 | |
350 | |
351 /* | |
352 * stop playing, keep buffers (for pause) | |
353 */ | |
354 static void audio_pause() | |
355 { | |
356 /* | |
357 * not possible with esd. the esd daemom will continue playing | |
358 * buffered data (not more than ESD_MAX_DELAY seconds of samples) | |
359 */ | |
360 } | |
361 | |
362 | |
363 /* | |
364 * resume playing, after audio_pause() | |
365 */ | |
366 static void audio_resume() | |
367 { | |
368 /* | |
369 * not possible with esd. | |
370 * | |
371 * Let's hope the pause was long enough that the esd ran out of | |
372 * buffered data; we restart our time based delay computation | |
373 * for an audio resume. | |
374 */ | |
375 esd_play_start.tv_sec = 0; | |
376 esd_samples_written = 0; | |
377 } | |
378 | |
379 | |
380 /* | |
381 * stop playing and empty buffers (for seeking/pause) | |
382 */ | |
383 static void reset() | |
384 { | |
385 #ifdef __svr4__ | |
386 /* throw away data buffered in the esd connection */ | |
387 if (ioctl(esd_play_fd, I_FLUSH, FLUSHW)) | |
388 perror("I_FLUSH"); | |
389 #endif | |
390 } | |
391 | |
392 | |
393 /* | |
394 * return: how many bytes can be played without blocking | |
395 */ | |
396 static int get_space() | |
397 { | |
398 struct timeval tmout; | |
399 fd_set wfds; | |
400 float current_delay; | |
401 int space; | |
402 | |
403 /* | |
404 * Don't buffer too much data in the esd daemon. | |
405 * | |
406 * If we send too much, esd will block in write()s to the sound | |
407 * device, and the consequence is a huge slow down for things like | |
408 * esd_get_all_info(). | |
409 */ | |
410 if ((current_delay = get_delay()) >= ESD_MAX_DELAY) { | |
411 dprintf("esd get_space: too much data buffered\n"); | |
412 return 0; | |
413 } | |
414 | |
415 FD_ZERO(&wfds); | |
416 FD_SET(esd_play_fd, &wfds); | |
417 tmout.tv_sec = 0; | |
418 tmout.tv_usec = 0; | |
419 | |
420 if (select(esd_play_fd + 1, NULL, &wfds, NULL, &tmout) != 1) | |
421 return 0; | |
422 | |
423 if (!FD_ISSET(esd_play_fd, &wfds)) | |
424 return 0; | |
425 | |
426 /* try to fill 50% of the remaining "free" buffer space */ | |
427 space = (ESD_MAX_DELAY - current_delay) * ao_data.bps * 0.5f; | |
428 | |
429 /* round up to next multiple of ESD_BUF_SIZE */ | |
430 space = (space + ESD_BUF_SIZE-1) / ESD_BUF_SIZE * ESD_BUF_SIZE; | |
431 | |
432 dprintf("esd get_space: %d\n", space); | |
433 return space; | |
434 } | |
435 | |
436 | |
437 /* | |
438 * return: delay in seconds between first and last sample in buffer | |
439 */ | |
440 static float get_delay() | |
441 { | |
442 struct timeval now; | |
443 double buffered_samples_time; | |
444 double play_time; | |
445 | |
446 if (!esd_play_start.tv_sec) | |
447 return 0; | |
448 | |
449 buffered_samples_time = (float)esd_samples_written / ao_data.samplerate; | |
450 gettimeofday(&now, NULL); | |
451 play_time = now.tv_sec - esd_play_start.tv_sec; | |
452 play_time += (now.tv_usec - esd_play_start.tv_usec) / 1000000.; | |
453 | |
454 /* dprintf("esd delay: %f %f\n", play_time, buffered_samples_time); */ | |
455 | |
456 if (play_time > buffered_samples_time) { | |
457 dprintf("esd: underflow\n"); | |
458 esd_play_start.tv_sec = 0; | |
459 esd_samples_written = 0; | |
460 return 0; | |
461 } | |
462 | |
463 dprintf("esd: get_delay %f\n", buffered_samples_time - play_time); | |
464 return buffered_samples_time - play_time; | |
465 } |