2 * ALSA 0.9.x-1.x audio output driver
4 * Copyright (C) 2004 Alex Beregszaszi
6 * modified for real ALSA 0.9.0 support by Zsolt Barat <joy@streamminister.de>
7 * additional AC-3 passthrough support by Andy Lo A Foe <andy@alsaplayer.org>
8 * 08/22/2002 iec958-init rewritten and merged with common init, zsolt
9 * 04/13/2004 merged with ao_alsa1.x, fixes provided by Jindrich Makovicka
10 * 04/25/2004 printfs converted to mp_msg, Zsolt.
12 * This file is part of MPlayer.
14 * MPlayer is free software; you can redistribute it and/or modify
15 * it under the terms of the GNU General Public License as published by
16 * the Free Software Foundation; either version 2 of the License, or
17 * (at your option) any later version.
19 * MPlayer is distributed in the hope that it will be useful,
20 * but WITHOUT ANY WARRANTY; without even the implied warranty of
21 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
22 * GNU General Public License for more details.
24 * You should have received a copy of the GNU General Public License along
25 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
26 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
39 #include "subopt-helper.h"
44 #define ALSA_PCM_NEW_HW_PARAMS_API
45 #define ALSA_PCM_NEW_SW_PARAMS_API
47 #ifdef HAVE_SYS_ASOUNDLIB_H
48 #include <sys/asoundlib.h>
49 #elif defined(HAVE_ALSA_ASOUNDLIB_H)
50 #include <alsa/asoundlib.h>
52 #error "asoundlib.h is not in sys/ or alsa/ - please bugreport"
56 #include "audio_out.h"
57 #include "audio_out_internal.h"
58 #include "libaf/af_format.h"
60 static const ao_info_t info
=
62 "ALSA-0.9.x-1.x audio output",
64 "Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
70 static snd_pcm_t
*alsa_handler
;
71 static snd_pcm_format_t alsa_format
;
72 static snd_pcm_hw_params_t
*alsa_hwparams
;
73 static snd_pcm_sw_params_t
*alsa_swparams
;
75 /* 16 sets buffersize to 16 * chunksize is as default 1024
76 * which seems to be good avarge for most situations
77 * so buffersize is 16384 frames by default */
78 static int alsa_fragcount
= 16;
79 static snd_pcm_uframes_t chunk_size
= 1024;
81 static size_t bytes_per_sample
;
83 static int ao_noblock
= 0;
86 static int alsa_can_pause
= 0;
87 static snd_pcm_sframes_t prepause_frames
;
89 #define ALSA_DEVICE_SIZE 256
94 static void alsa_error_handler(const char *file
, int line
, const char *function
,
95 int err
, const char *format
, ...)
100 va_start(va
, format
);
101 vsnprintf(tmp
, sizeof tmp
, format
, va
);
103 tmp
[sizeof tmp
- 1] = '\0';
106 mp_msg(MSGT_AO
, MSGL_ERR
, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
107 file
, line
, function
, tmp
, snd_strerror(err
));
109 mp_msg(MSGT_AO
, MSGL_ERR
, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
110 file
, line
, function
, tmp
);
113 /* to set/get/query special features/parameters */
114 static int control(int cmd
, void *arg
)
117 case AOCONTROL_QUERY_FORMAT
:
119 case AOCONTROL_GET_VOLUME
:
120 case AOCONTROL_SET_VOLUME
:
122 ao_control_vol_t
*vol
= (ao_control_vol_t
*)arg
;
126 snd_mixer_elem_t
*elem
;
127 snd_mixer_selem_id_t
*sid
;
129 static char *mix_name
= "PCM";
130 static char *card
= "default";
131 static int mix_index
= 0;
134 long get_vol
, set_vol
;
137 if(ao_data
.format
== AF_FORMAT_AC3
)
141 char *test_mix_index
;
143 mix_name
= strdup(mixer_channel
);
144 if ((test_mix_index
= strchr(mix_name
, ','))){
147 mix_index
= strtol(test_mix_index
, &test_mix_index
, 0);
149 if (*test_mix_index
){
150 mp_tmsg(MSGT_AO
,MSGL_ERR
,
151 "[AO_ALSA] Invalid mixer index. Defaulting to 0.\n");
156 if(mixer_device
) card
= mixer_device
;
159 snd_mixer_selem_id_alloca(&sid
);
161 //sets simple-mixer index and name
162 snd_mixer_selem_id_set_index(sid
, mix_index
);
163 snd_mixer_selem_id_set_name(sid
, mix_name
);
170 if ((err
= snd_mixer_open(&handle
, 0)) < 0) {
171 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Mixer open error: %s\n", snd_strerror(err
));
172 return CONTROL_ERROR
;
175 if ((err
= snd_mixer_attach(handle
, card
)) < 0) {
176 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Mixer attach %s error: %s\n",
177 card
, snd_strerror(err
));
178 snd_mixer_close(handle
);
179 return CONTROL_ERROR
;
182 if ((err
= snd_mixer_selem_register(handle
, NULL
, NULL
)) < 0) {
183 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Mixer register error: %s\n", snd_strerror(err
));
184 snd_mixer_close(handle
);
185 return CONTROL_ERROR
;
187 err
= snd_mixer_load(handle
);
189 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Mixer load error: %s\n", snd_strerror(err
));
190 snd_mixer_close(handle
);
191 return CONTROL_ERROR
;
194 elem
= snd_mixer_find_selem(handle
, sid
);
196 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to find simple control '%s',%i.\n",
197 snd_mixer_selem_id_get_name(sid
), snd_mixer_selem_id_get_index(sid
));
198 snd_mixer_close(handle
);
199 return CONTROL_ERROR
;
202 snd_mixer_selem_get_playback_volume_range(elem
,&pmin
,&pmax
);
203 f_multi
= (100 / (float)(pmax
- pmin
));
205 if (cmd
== AOCONTROL_SET_VOLUME
) {
207 set_vol
= vol
->left
/ f_multi
+ pmin
+ 0.5;
210 if ((err
= snd_mixer_selem_set_playback_volume(elem
, SND_MIXER_SCHN_FRONT_LEFT
, set_vol
)) < 0) {
211 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Error setting left channel, %s\n",
213 return CONTROL_ERROR
;
215 mp_msg(MSGT_AO
,MSGL_DBG2
,"left=%li, ", set_vol
);
217 set_vol
= vol
->right
/ f_multi
+ pmin
+ 0.5;
219 if ((err
= snd_mixer_selem_set_playback_volume(elem
, SND_MIXER_SCHN_FRONT_RIGHT
, set_vol
)) < 0) {
220 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Error setting right channel, %s\n",
222 return CONTROL_ERROR
;
224 mp_msg(MSGT_AO
,MSGL_DBG2
,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
225 set_vol
, pmin
, pmax
, f_multi
);
227 if (snd_mixer_selem_has_playback_switch(elem
)) {
228 int lmute
= (vol
->left
== 0.0);
229 int rmute
= (vol
->right
== 0.0);
230 if (snd_mixer_selem_has_playback_switch_joined(elem
)) {
231 lmute
= rmute
= lmute
&& rmute
;
233 snd_mixer_selem_set_playback_switch(elem
, SND_MIXER_SCHN_FRONT_RIGHT
, !rmute
);
235 snd_mixer_selem_set_playback_switch(elem
, SND_MIXER_SCHN_FRONT_LEFT
, !lmute
);
239 snd_mixer_selem_get_playback_volume(elem
, SND_MIXER_SCHN_FRONT_LEFT
, &get_vol
);
240 vol
->left
= (get_vol
- pmin
) * f_multi
;
241 snd_mixer_selem_get_playback_volume(elem
, SND_MIXER_SCHN_FRONT_RIGHT
, &get_vol
);
242 vol
->right
= (get_vol
- pmin
) * f_multi
;
244 mp_msg(MSGT_AO
,MSGL_DBG2
,"left=%f, right=%f\n",vol
->left
,vol
->right
);
246 snd_mixer_close(handle
);
251 return CONTROL_UNKNOWN
;
254 static void parse_device (char *dest
, const char *src
, int len
)
257 memmove(dest
, src
, len
);
259 while ((tmp
= strrchr(dest
, '.')))
261 while ((tmp
= strrchr(dest
, '=')))
265 static void print_help (void)
267 mp_tmsg (MSGT_AO
, MSGL_FATAL
,
268 "\n[AO_ALSA] -ao alsa commandline help:\n"\
269 "[AO_ALSA] Example: mplayer -ao alsa:device=hw=0.3\n"\
270 "[AO_ALSA] Sets first card fourth hardware device.\n\n"\
271 "[AO_ALSA] Options:\n"\
272 "[AO_ALSA] noblock\n"\
273 "[AO_ALSA] Opens device in non-blocking mode.\n"\
274 "[AO_ALSA] device=<device-name>\n"\
275 "[AO_ALSA] Sets device (change , to . and : to =)\n");
278 static int str_maxlen(strarg_t
*str
) {
279 if (str
->len
> ALSA_DEVICE_SIZE
)
284 static int try_open_device(const char *device
, int open_mode
, int try_ac3
)
287 char *ac3_device
, *args
;
290 /* to set the non-audio bit, use AES0=6 */
291 len
= strlen(device
);
292 ac3_device
= malloc(len
+ 7 + 1);
295 strcpy(ac3_device
, device
);
296 args
= strchr(ac3_device
, ':');
298 /* no existing parameters: add it behind device name */
299 strcat(ac3_device
, ":AES0=6");
303 while (isspace(*args
));
305 /* ":" but no parameters */
306 strcat(ac3_device
, "AES0=6");
307 } else if (*args
!= '{') {
308 /* a simple list of parameters: add it at the end of the list */
309 strcat(ac3_device
, ",AES0=6");
311 /* parameters in config syntax: add it inside the { } block */
314 while (len
> 0 && isspace(ac3_device
[len
]));
315 if (ac3_device
[len
] == '}')
316 strcpy(ac3_device
+ len
, " AES0=6}");
319 err
= snd_pcm_open(&alsa_handler
, ac3_device
, SND_PCM_STREAM_PLAYBACK
,
323 if (!try_ac3
|| err
< 0)
324 err
= snd_pcm_open(&alsa_handler
, device
, SND_PCM_STREAM_PLAYBACK
,
330 open & setup audio device
331 return: 1=success 0=fail
333 static int init(int rate_hz
, int channels
, int format
, int flags
)
338 snd_pcm_uframes_t bufsize
;
339 snd_pcm_uframes_t boundary
;
341 {"block", OPT_ARG_BOOL
, &block
, NULL
},
342 {"device", OPT_ARG_STR
, &device
, (opt_test_f
)str_maxlen
},
346 char alsa_device
[ALSA_DEVICE_SIZE
+ 1];
347 // make sure alsa_device is null-terminated even when using strncpy etc.
348 memset(alsa_device
, 0, ALSA_DEVICE_SIZE
+ 1);
350 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz
,
353 #if SND_LIB_VERSION >= 0x010005
354 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
356 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR
);
361 snd_lib_error_set_handler(alsa_error_handler
);
363 ao_data
.samplerate
= rate_hz
;
364 ao_data
.format
= format
;
365 ao_data
.channels
= channels
;
370 alsa_format
= SND_PCM_FORMAT_S8
;
373 alsa_format
= SND_PCM_FORMAT_U8
;
375 case AF_FORMAT_U16_LE
:
376 alsa_format
= SND_PCM_FORMAT_U16_LE
;
378 case AF_FORMAT_U16_BE
:
379 alsa_format
= SND_PCM_FORMAT_U16_BE
;
384 case AF_FORMAT_S16_LE
:
385 alsa_format
= SND_PCM_FORMAT_S16_LE
;
390 case AF_FORMAT_S16_BE
:
391 alsa_format
= SND_PCM_FORMAT_S16_BE
;
393 case AF_FORMAT_U32_LE
:
394 alsa_format
= SND_PCM_FORMAT_U32_LE
;
396 case AF_FORMAT_U32_BE
:
397 alsa_format
= SND_PCM_FORMAT_U32_BE
;
399 case AF_FORMAT_S32_LE
:
400 alsa_format
= SND_PCM_FORMAT_S32_LE
;
402 case AF_FORMAT_S32_BE
:
403 alsa_format
= SND_PCM_FORMAT_S32_BE
;
405 case AF_FORMAT_U24_LE
:
406 alsa_format
= SND_PCM_FORMAT_U24_3LE
;
408 case AF_FORMAT_U24_BE
:
409 alsa_format
= SND_PCM_FORMAT_U24_3BE
;
411 case AF_FORMAT_S24_LE
:
412 alsa_format
= SND_PCM_FORMAT_S24_3LE
;
414 case AF_FORMAT_S24_BE
:
415 alsa_format
= SND_PCM_FORMAT_S24_3BE
;
417 case AF_FORMAT_FLOAT_LE
:
418 alsa_format
= SND_PCM_FORMAT_FLOAT_LE
;
420 case AF_FORMAT_FLOAT_BE
:
421 alsa_format
= SND_PCM_FORMAT_FLOAT_BE
;
423 case AF_FORMAT_MU_LAW
:
424 alsa_format
= SND_PCM_FORMAT_MU_LAW
;
426 case AF_FORMAT_A_LAW
:
427 alsa_format
= SND_PCM_FORMAT_A_LAW
;
431 alsa_format
= SND_PCM_FORMAT_MPEG
; //? default should be -1
439 * sets opening sequence for SPDIF
440 * sets also the playback and other switches 'on the fly'
441 * while opening the abstract alias for the spdif subdevice
444 if (format
== AF_FORMAT_AC3
) {
445 device
.str
= "iec958";
446 mp_msg(MSGT_AO
,MSGL_V
,"alsa-spdif-init: playing AC3, %i channels\n", channels
);
449 /* in any case for multichannel playback we should select
455 device
.str
= "default";
456 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: setup for 1/2 channel(s)\n");
459 if (alsa_format
== SND_PCM_FORMAT_FLOAT_LE
)
460 // hack - use the converter plugin
461 device
.str
= "plug:surround40";
463 device
.str
= "surround40";
464 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: device set to surround40\n");
467 if (alsa_format
== SND_PCM_FORMAT_FLOAT_LE
)
468 device
.str
= "plug:surround51";
470 device
.str
= "surround51";
471 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: device set to surround51\n");
474 device
.str
= "default";
475 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] %d channels are not supported.\n",channels
);
477 device
.len
= strlen(device
.str
);
478 if (subopt_parse(ao_subdevice
, subopts
) != 0) {
483 parse_device(alsa_device
, device
.str
, device
.len
);
485 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: using device %s\n", alsa_device
);
487 //setting modes for block or nonblock-mode
489 open_mode
= SND_PCM_NONBLOCK
;
495 //sets buff/chunksize if its set manually
496 if (ao_data
.buffersize
) {
497 switch (ao_data
.buffersize
)
502 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 8192\n");
503 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 512\n");
508 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 8192\n");
509 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 1024\n");
514 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 16384\n");
515 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 512\n");
520 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 16384\n");
521 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 1024\n");
531 //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
532 if ((err
= try_open_device(alsa_device
, open_mode
, format
== AF_FORMAT_AC3
)) < 0)
534 if (err
!= -EBUSY
&& ao_noblock
) {
535 mp_tmsg(MSGT_AO
,MSGL_INFO
,"[AO_ALSA] Open in nonblock-mode failed, trying to open in block-mode.\n");
536 if ((err
= try_open_device(alsa_device
, 0, format
== AF_FORMAT_AC3
)) < 0) {
537 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err
));
541 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err
));
546 if ((err
= snd_pcm_nonblock(alsa_handler
, 0)) < 0) {
547 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AL_ALSA] Error setting block-mode %s.\n", snd_strerror(err
));
549 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: pcm opened in blocking mode\n");
552 snd_pcm_hw_params_alloca(&alsa_hwparams
);
553 snd_pcm_sw_params_alloca(&alsa_swparams
);
555 // setting hw-parameters
556 if ((err
= snd_pcm_hw_params_any(alsa_handler
, alsa_hwparams
)) < 0)
558 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to get initial parameters: %s\n",
563 err
= snd_pcm_hw_params_set_access(alsa_handler
, alsa_hwparams
,
564 SND_PCM_ACCESS_RW_INTERLEAVED
);
566 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to set access type: %s\n",
571 /* workaround for nonsupported formats
572 sets default format to S16_LE if the given formats aren't supported */
573 if ((err
= snd_pcm_hw_params_test_format(alsa_handler
, alsa_hwparams
,
576 mp_tmsg(MSGT_AO
,MSGL_INFO
,
577 "[AO_ALSA] Format %s is not supported by hardware, trying default.\n", af_fmt2str_short(format
));
578 alsa_format
= SND_PCM_FORMAT_S16_LE
;
579 ao_data
.format
= AF_FORMAT_S16_LE
;
582 if ((err
= snd_pcm_hw_params_set_format(alsa_handler
, alsa_hwparams
,
585 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to set format: %s\n",
590 if ((err
= snd_pcm_hw_params_set_channels_near(alsa_handler
, alsa_hwparams
,
591 &ao_data
.channels
)) < 0)
593 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to set channels: %s\n",
598 /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
599 prefer our own resampler */
600 #if SND_LIB_VERSION >= 0x010009
601 if ((err
= snd_pcm_hw_params_set_rate_resample(alsa_handler
, alsa_hwparams
,
604 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to disable resampling: %s\n",
610 if ((err
= snd_pcm_hw_params_set_rate_near(alsa_handler
, alsa_hwparams
,
611 &ao_data
.samplerate
, NULL
)) < 0)
613 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to set samplerate-2: %s\n",
618 bytes_per_sample
= snd_pcm_format_physical_width(alsa_format
) / 8;
619 bytes_per_sample
*= ao_data
.channels
;
620 ao_data
.bps
= ao_data
.samplerate
* bytes_per_sample
;
624 int alsa_buffer_time
= 500000; /* original 60 */
625 int alsa_period_time
;
626 alsa_period_time
= alsa_buffer_time
/4;
627 if ((err
= snd_pcm_hw_params_set_buffer_time_near(alsa_handler
, alsa_hwparams
,
628 &alsa_buffer_time
, NULL
)) < 0)
630 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to set buffer time near: %s\n",
634 alsa_buffer_time
= err
;
636 if ((err
= snd_pcm_hw_params_set_period_time_near(alsa_handler
, alsa_hwparams
,
637 &alsa_period_time
, NULL
)) < 0)
638 /* original: alsa_buffer_time/ao_data.bps */
640 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to set period time: %s\n",
644 mp_tmsg(MSGT_AO
,MSGL_INFO
,"[AO_ALSA] buffer_time: %d, period_time :%d\n",
645 alsa_buffer_time
, err
);
647 #endif//end SET_BUFFERTIME
652 if ((err
= snd_pcm_hw_params_set_period_size_near(alsa_handler
, alsa_hwparams
,
653 &chunk_size
, NULL
)) < 0)
655 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO ALSA] Unable to set period size(%ld): %s\n",
656 chunk_size
, snd_strerror(err
));
660 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set to %li\n", chunk_size
);
662 if ((err
= snd_pcm_hw_params_set_periods_near(alsa_handler
, alsa_hwparams
,
663 &alsa_fragcount
, NULL
)) < 0) {
664 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to set periods: %s\n",
669 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: fragcount=%i\n", alsa_fragcount
);
672 #endif//end SET_CHUNKSIZE
674 /* finally install hardware parameters */
675 if ((err
= snd_pcm_hw_params(alsa_handler
, alsa_hwparams
)) < 0)
677 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to set hw-parameters: %s\n",
681 // end setting hw-params
684 // gets buffersize for control
685 if ((err
= snd_pcm_hw_params_get_buffer_size(alsa_hwparams
, &bufsize
)) < 0)
687 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to get buffersize: %s\n", snd_strerror(err
));
691 ao_data
.buffersize
= bufsize
* bytes_per_sample
;
692 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: got buffersize=%i\n", ao_data
.buffersize
);
695 if ((err
= snd_pcm_hw_params_get_period_size(alsa_hwparams
, &chunk_size
, NULL
)) < 0) {
696 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO ALSA] Unable to get period size: %s\n", snd_strerror(err
));
699 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: got period size %li\n", chunk_size
);
701 ao_data
.outburst
= chunk_size
* bytes_per_sample
;
703 /* setting software parameters */
704 if ((err
= snd_pcm_sw_params_current(alsa_handler
, alsa_swparams
)) < 0) {
705 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to get sw-parameters: %s\n",
709 #if SND_LIB_VERSION >= 0x000901
710 if ((err
= snd_pcm_sw_params_get_boundary(alsa_swparams
, &boundary
)) < 0) {
711 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to get boundary: %s\n",
716 boundary
= 0x7fffffff;
718 /* start playing when one period has been written */
719 if ((err
= snd_pcm_sw_params_set_start_threshold(alsa_handler
, alsa_swparams
, chunk_size
)) < 0) {
720 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to set start threshold: %s\n",
724 /* disable underrun reporting */
725 if ((err
= snd_pcm_sw_params_set_stop_threshold(alsa_handler
, alsa_swparams
, boundary
)) < 0) {
726 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to set stop threshold: %s\n",
730 #if SND_LIB_VERSION >= 0x000901
731 /* play silence when there is an underrun */
732 if ((err
= snd_pcm_sw_params_set_silence_size(alsa_handler
, alsa_swparams
, boundary
)) < 0) {
733 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to set silence size: %s\n",
738 if ((err
= snd_pcm_sw_params(alsa_handler
, alsa_swparams
)) < 0) {
739 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Unable to get sw-parameters: %s\n",
743 /* end setting sw-params */
745 mp_msg(MSGT_AO
,MSGL_V
,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
746 ao_data
.samplerate
, ao_data
.channels
, (int)bytes_per_sample
, ao_data
.buffersize
,
747 snd_pcm_format_description(alsa_format
));
749 } // end switch alsa_handler (spdif)
750 alsa_can_pause
= snd_pcm_hw_params_can_pause(alsa_hwparams
);
755 /* close audio device */
756 static void uninit(int immed
)
763 snd_pcm_drain(alsa_handler
);
765 if ((err
= snd_pcm_close(alsa_handler
)) < 0)
767 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] pcm close error: %s\n", snd_strerror(err
));
772 mp_msg(MSGT_AO
,MSGL_V
,"alsa-uninit: pcm closed\n");
776 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] No handler defined!\n");
780 static void audio_pause(void)
784 if (alsa_can_pause
) {
785 if ((err
= snd_pcm_pause(alsa_handler
, 1)) < 0)
787 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] pcm pause error: %s\n", snd_strerror(err
));
790 mp_msg(MSGT_AO
,MSGL_V
,"alsa-pause: pause supported by hardware\n");
792 if (snd_pcm_delay(alsa_handler
, &prepause_frames
) < 0
793 || prepause_frames
< 0)
796 if ((err
= snd_pcm_drop(alsa_handler
)) < 0)
798 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] pcm drop error: %s\n", snd_strerror(err
));
804 static void audio_resume(void)
808 if (snd_pcm_state(alsa_handler
) == SND_PCM_STATE_SUSPENDED
) {
809 mp_tmsg(MSGT_AO
,MSGL_INFO
,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
810 while ((err
= snd_pcm_resume(alsa_handler
)) == -EAGAIN
) sleep(1);
812 if (alsa_can_pause
) {
813 if ((err
= snd_pcm_pause(alsa_handler
, 0)) < 0)
815 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] pcm resume error: %s\n", snd_strerror(err
));
818 mp_msg(MSGT_AO
,MSGL_V
,"alsa-resume: resume supported by hardware\n");
820 if ((err
= snd_pcm_prepare(alsa_handler
)) < 0)
822 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err
));
825 if (prepause_frames
) {
826 void *silence
= calloc(prepause_frames
, bytes_per_sample
);
827 play(silence
, prepause_frames
* bytes_per_sample
, 0);
833 /* stop playing and empty buffers (for seeking/pause) */
834 static void reset(void)
839 if ((err
= snd_pcm_drop(alsa_handler
)) < 0)
841 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err
));
844 if ((err
= snd_pcm_prepare(alsa_handler
)) < 0)
846 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err
));
853 plays 'len' bytes of 'data'
854 returns: number of bytes played
855 modified last at 29.06.02 by jp
856 thanxs for marius <marius@rospot.com> for giving us the light ;)
859 static int play(void* data
, int len
, int flags
)
861 int num_frames
= len
/ bytes_per_sample
;
862 snd_pcm_sframes_t res
= 0;
864 //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
867 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Device configuration error.");
875 res
= snd_pcm_writei(alsa_handler
, data
, num_frames
);
881 else if (res
== -ESTRPIPE
) { /* suspend */
882 mp_tmsg(MSGT_AO
,MSGL_INFO
,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
883 while ((res
= snd_pcm_resume(alsa_handler
)) == -EAGAIN
)
887 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Write error: %s\n", snd_strerror(res
));
888 mp_tmsg(MSGT_AO
,MSGL_INFO
,"[AO_ALSA] Trying to reset soundcard.\n");
889 if ((res
= snd_pcm_prepare(alsa_handler
)) < 0) {
890 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(res
));
897 return res
< 0 ? res
: res
* bytes_per_sample
;
900 /* how many byes are free in the buffer */
901 static int get_space(void)
903 snd_pcm_status_t
*status
;
906 snd_pcm_status_alloca(&status
);
908 if ((ret
= snd_pcm_status(alsa_handler
, status
)) < 0)
910 mp_tmsg(MSGT_AO
,MSGL_ERR
,"[AO_ALSA] Cannot get pcm status: %s\n", snd_strerror(ret
));
914 unsigned space
= snd_pcm_status_get_avail(status
) * bytes_per_sample
;
915 if (space
> ao_data
.buffersize
) // Buffer underrun?
916 space
= ao_data
.buffersize
;
920 /* delay in seconds between first and last sample in buffer */
921 static float get_delay(void)
924 snd_pcm_sframes_t delay
;
926 if (snd_pcm_delay(alsa_handler
, &delay
) < 0)
930 /* underrun - move the application pointer forward to catch up */
931 #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
932 snd_pcm_forward(alsa_handler
, -delay
);
936 return (float)delay
/ (float)ao_data
.samplerate
;