2 ao_alsa9/1.x - ALSA-0.9.x-1.x output plugin for MPlayer
6 modified for real alsa-0.9.0-support by Zsolt Barat <joy@streamminister.de>
7 additional AC3 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 Any bugreports regarding to this driver are welcome.
25 #include "subopt-helper.h"
30 #define ALSA_PCM_NEW_HW_PARAMS_API
31 #define ALSA_PCM_NEW_SW_PARAMS_API
33 #if HAVE_SYS_ASOUNDLIB_H
34 #include <sys/asoundlib.h>
35 #elif HAVE_ALSA_ASOUNDLIB_H
36 #include <alsa/asoundlib.h>
38 #error "asoundlib.h is not in sys/ or alsa/ - please bugreport"
42 #include "audio_out.h"
43 #include "audio_out_internal.h"
44 #include "libaf/af_format.h"
46 static ao_info_t info
=
48 "ALSA-0.9.x-1.x audio output",
50 "Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
56 static snd_pcm_t
*alsa_handler
;
57 static snd_pcm_format_t alsa_format
;
58 static snd_pcm_hw_params_t
*alsa_hwparams
;
59 static snd_pcm_sw_params_t
*alsa_swparams
;
61 /* 16 sets buffersize to 16 * chunksize is as default 1024
62 * which seems to be good avarge for most situations
63 * so buffersize is 16384 frames by default */
64 static int alsa_fragcount
= 16;
65 static snd_pcm_uframes_t chunk_size
= 1024;
67 static size_t bytes_per_sample
;
69 static int ao_noblock
= 0;
72 static int alsa_can_pause
= 0;
73 static snd_pcm_sframes_t prepause_frames
;
75 #define ALSA_DEVICE_SIZE 256
80 static void alsa_error_handler(const char *file
, int line
, const char *function
,
81 int err
, const char *format
, ...)
87 vsnprintf(tmp
, sizeof tmp
, format
, va
);
89 tmp
[sizeof tmp
- 1] = '\0';
92 mp_msg(MSGT_AO
, MSGL_ERR
, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
93 file
, line
, function
, tmp
, snd_strerror(err
));
95 mp_msg(MSGT_AO
, MSGL_ERR
, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
96 file
, line
, function
, tmp
);
99 /* to set/get/query special features/parameters */
100 static int control(int cmd
, void *arg
)
103 case AOCONTROL_QUERY_FORMAT
:
105 case AOCONTROL_GET_VOLUME
:
106 case AOCONTROL_SET_VOLUME
:
108 ao_control_vol_t
*vol
= (ao_control_vol_t
*)arg
;
112 snd_mixer_elem_t
*elem
;
113 snd_mixer_selem_id_t
*sid
;
115 static char *mix_name
= "PCM";
116 static char *card
= "default";
117 static int mix_index
= 0;
120 long get_vol
, set_vol
;
123 if(ao_data
.format
== AF_FORMAT_AC3
)
127 char *test_mix_index
;
129 mix_name
= strdup(mixer_channel
);
130 if ((test_mix_index
= strchr(mix_name
, ','))){
133 mix_index
= strtol(test_mix_index
, &test_mix_index
, 0);
135 if (*test_mix_index
){
136 mp_msg(MSGT_AO
,MSGL_ERR
,
137 MSGTR_AO_ALSA_InvalidMixerIndexDefaultingToZero
);
142 if(mixer_device
) card
= mixer_device
;
145 snd_mixer_selem_id_alloca(&sid
);
147 //sets simple-mixer index and name
148 snd_mixer_selem_id_set_index(sid
, mix_index
);
149 snd_mixer_selem_id_set_name(sid
, mix_name
);
156 if ((err
= snd_mixer_open(&handle
, 0)) < 0) {
157 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerOpenError
, snd_strerror(err
));
158 return CONTROL_ERROR
;
161 if ((err
= snd_mixer_attach(handle
, card
)) < 0) {
162 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerAttachError
,
163 card
, snd_strerror(err
));
164 snd_mixer_close(handle
);
165 return CONTROL_ERROR
;
168 if ((err
= snd_mixer_selem_register(handle
, NULL
, NULL
)) < 0) {
169 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerRegisterError
, snd_strerror(err
));
170 snd_mixer_close(handle
);
171 return CONTROL_ERROR
;
173 err
= snd_mixer_load(handle
);
175 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerLoadError
, snd_strerror(err
));
176 snd_mixer_close(handle
);
177 return CONTROL_ERROR
;
180 elem
= snd_mixer_find_selem(handle
, sid
);
182 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToFindSimpleControl
,
183 snd_mixer_selem_id_get_name(sid
), snd_mixer_selem_id_get_index(sid
));
184 snd_mixer_close(handle
);
185 return CONTROL_ERROR
;
188 snd_mixer_selem_get_playback_volume_range(elem
,&pmin
,&pmax
);
189 f_multi
= (100 / (float)(pmax
- pmin
));
191 if (cmd
== AOCONTROL_SET_VOLUME
) {
193 set_vol
= vol
->left
/ f_multi
+ pmin
+ 0.5;
196 if ((err
= snd_mixer_selem_set_playback_volume(elem
, SND_MIXER_SCHN_FRONT_LEFT
, set_vol
)) < 0) {
197 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ErrorSettingLeftChannel
,
199 return CONTROL_ERROR
;
201 mp_msg(MSGT_AO
,MSGL_DBG2
,"left=%li, ", set_vol
);
203 set_vol
= vol
->right
/ f_multi
+ pmin
+ 0.5;
205 if ((err
= snd_mixer_selem_set_playback_volume(elem
, SND_MIXER_SCHN_FRONT_RIGHT
, set_vol
)) < 0) {
206 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ErrorSettingRightChannel
,
208 return CONTROL_ERROR
;
210 mp_msg(MSGT_AO
,MSGL_DBG2
,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
211 set_vol
, pmin
, pmax
, f_multi
);
213 if (snd_mixer_selem_has_playback_switch(elem
)) {
214 int lmute
= (vol
->left
== 0.0);
215 int rmute
= (vol
->right
== 0.0);
216 if (snd_mixer_selem_has_playback_switch_joined(elem
)) {
217 lmute
= rmute
= lmute
&& rmute
;
219 snd_mixer_selem_set_playback_switch(elem
, SND_MIXER_SCHN_FRONT_RIGHT
, !rmute
);
221 snd_mixer_selem_set_playback_switch(elem
, SND_MIXER_SCHN_FRONT_LEFT
, !lmute
);
225 snd_mixer_selem_get_playback_volume(elem
, SND_MIXER_SCHN_FRONT_LEFT
, &get_vol
);
226 vol
->left
= (get_vol
- pmin
) * f_multi
;
227 snd_mixer_selem_get_playback_volume(elem
, SND_MIXER_SCHN_FRONT_RIGHT
, &get_vol
);
228 vol
->right
= (get_vol
- pmin
) * f_multi
;
230 mp_msg(MSGT_AO
,MSGL_DBG2
,"left=%f, right=%f\n",vol
->left
,vol
->right
);
232 snd_mixer_close(handle
);
237 return CONTROL_UNKNOWN
;
240 static void parse_device (char *dest
, const char *src
, int len
)
243 memmove(dest
, src
, len
);
245 while ((tmp
= strrchr(dest
, '.')))
247 while ((tmp
= strrchr(dest
, '=')))
251 static void print_help (void)
253 mp_msg (MSGT_AO
, MSGL_FATAL
,
254 MSGTR_AO_ALSA_CommandlineHelp
);
257 static int str_maxlen(strarg_t
*str
) {
258 if (str
->len
> ALSA_DEVICE_SIZE
)
263 static int try_open_device(const char *device
, int open_mode
, int try_ac3
)
266 char *ac3_device
, *args
;
269 /* to set the non-audio bit, use AES0=6 */
270 len
= strlen(device
);
271 ac3_device
= malloc(len
+ 7 + 1);
274 strcpy(ac3_device
, device
);
275 args
= strchr(ac3_device
, ':');
277 /* no existing parameters: add it behind device name */
278 strcat(ac3_device
, ":AES0=6");
282 while (isspace(*args
));
284 /* ":" but no parameters */
285 strcat(ac3_device
, "AES0=6");
286 } else if (*args
!= '{') {
287 /* a simple list of parameters: add it at the end of the list */
288 strcat(ac3_device
, ",AES0=6");
290 /* parameters in config syntax: add it inside the { } block */
293 while (len
> 0 && isspace(ac3_device
[len
]));
294 if (ac3_device
[len
] == '}')
295 strcpy(ac3_device
+ len
, " AES0=6}");
298 err
= snd_pcm_open(&alsa_handler
, ac3_device
, SND_PCM_STREAM_PLAYBACK
,
302 if (!try_ac3
|| err
< 0)
303 err
= snd_pcm_open(&alsa_handler
, device
, SND_PCM_STREAM_PLAYBACK
,
309 open & setup audio device
310 return: 1=success 0=fail
312 static int init(int rate_hz
, int channels
, int format
, int flags
)
317 snd_pcm_uframes_t bufsize
;
318 snd_pcm_uframes_t boundary
;
320 {"block", OPT_ARG_BOOL
, &block
, NULL
},
321 {"device", OPT_ARG_STR
, &device
, (opt_test_f
)str_maxlen
},
325 char alsa_device
[ALSA_DEVICE_SIZE
+ 1];
326 // make sure alsa_device is null-terminated even when using strncpy etc.
327 memset(alsa_device
, 0, ALSA_DEVICE_SIZE
+ 1);
329 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz
,
332 #if SND_LIB_VERSION >= 0x010005
333 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
335 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR
);
340 snd_lib_error_set_handler(alsa_error_handler
);
342 ao_data
.samplerate
= rate_hz
;
343 ao_data
.format
= format
;
344 ao_data
.channels
= channels
;
349 alsa_format
= SND_PCM_FORMAT_S8
;
352 alsa_format
= SND_PCM_FORMAT_U8
;
354 case AF_FORMAT_U16_LE
:
355 alsa_format
= SND_PCM_FORMAT_U16_LE
;
357 case AF_FORMAT_U16_BE
:
358 alsa_format
= SND_PCM_FORMAT_U16_BE
;
360 #ifndef WORDS_BIGENDIAN
363 case AF_FORMAT_S16_LE
:
364 alsa_format
= SND_PCM_FORMAT_S16_LE
;
366 #ifdef WORDS_BIGENDIAN
369 case AF_FORMAT_S16_BE
:
370 alsa_format
= SND_PCM_FORMAT_S16_BE
;
372 case AF_FORMAT_U32_LE
:
373 alsa_format
= SND_PCM_FORMAT_U32_LE
;
375 case AF_FORMAT_U32_BE
:
376 alsa_format
= SND_PCM_FORMAT_U32_BE
;
378 case AF_FORMAT_S32_LE
:
379 alsa_format
= SND_PCM_FORMAT_S32_LE
;
381 case AF_FORMAT_S32_BE
:
382 alsa_format
= SND_PCM_FORMAT_S32_BE
;
384 case AF_FORMAT_FLOAT_LE
:
385 alsa_format
= SND_PCM_FORMAT_FLOAT_LE
;
387 case AF_FORMAT_FLOAT_BE
:
388 alsa_format
= SND_PCM_FORMAT_FLOAT_BE
;
390 case AF_FORMAT_MU_LAW
:
391 alsa_format
= SND_PCM_FORMAT_MU_LAW
;
393 case AF_FORMAT_A_LAW
:
394 alsa_format
= SND_PCM_FORMAT_A_LAW
;
398 alsa_format
= SND_PCM_FORMAT_MPEG
; //? default should be -1
406 * sets opening sequence for SPDIF
407 * sets also the playback and other switches 'on the fly'
408 * while opening the abstract alias for the spdif subdevice
411 if (format
== AF_FORMAT_AC3
) {
412 device
.str
= "iec958";
413 mp_msg(MSGT_AO
,MSGL_V
,"alsa-spdif-init: playing AC3, %i channels\n", channels
);
416 /* in any case for multichannel playback we should select
422 device
.str
= "default";
423 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: setup for 1/2 channel(s)\n");
426 if (alsa_format
== SND_PCM_FORMAT_FLOAT_LE
)
427 // hack - use the converter plugin
428 device
.str
= "plug:surround40";
430 device
.str
= "surround40";
431 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: device set to surround40\n");
434 if (alsa_format
== SND_PCM_FORMAT_FLOAT_LE
)
435 device
.str
= "plug:surround51";
437 device
.str
= "surround51";
438 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: device set to surround51\n");
441 device
.str
= "default";
442 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ChannelsNotSupported
,channels
);
444 device
.len
= strlen(device
.str
);
445 if (subopt_parse(ao_subdevice
, subopts
) != 0) {
450 parse_device(alsa_device
, device
.str
, device
.len
);
452 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: using device %s\n", alsa_device
);
454 //setting modes for block or nonblock-mode
456 open_mode
= SND_PCM_NONBLOCK
;
462 //sets buff/chunksize if its set manually
463 if (ao_data
.buffersize
) {
464 switch (ao_data
.buffersize
)
469 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 8192\n");
470 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 512\n");
475 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 8192\n");
476 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 1024\n");
481 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 16384\n");
482 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 512\n");
487 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 16384\n");
488 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 1024\n");
498 //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
499 if ((err
= try_open_device(alsa_device
, open_mode
, format
== AF_FORMAT_AC3
)) < 0)
501 if (err
!= -EBUSY
&& ao_noblock
) {
502 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_OpenInNonblockModeFailed
);
503 if ((err
= try_open_device(alsa_device
, 0, format
== AF_FORMAT_AC3
)) < 0) {
504 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PlaybackOpenError
, snd_strerror(err
));
508 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PlaybackOpenError
, snd_strerror(err
));
513 if ((err
= snd_pcm_nonblock(alsa_handler
, 0)) < 0) {
514 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ErrorSetBlockMode
, snd_strerror(err
));
516 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: pcm opened in blocking mode\n");
519 snd_pcm_hw_params_alloca(&alsa_hwparams
);
520 snd_pcm_sw_params_alloca(&alsa_swparams
);
522 // setting hw-parameters
523 if ((err
= snd_pcm_hw_params_any(alsa_handler
, alsa_hwparams
)) < 0)
525 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetInitialParameters
,
530 err
= snd_pcm_hw_params_set_access(alsa_handler
, alsa_hwparams
,
531 SND_PCM_ACCESS_RW_INTERLEAVED
);
533 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetAccessType
,
538 /* workaround for nonsupported formats
539 sets default format to S16_LE if the given formats aren't supported */
540 if ((err
= snd_pcm_hw_params_test_format(alsa_handler
, alsa_hwparams
,
543 mp_msg(MSGT_AO
,MSGL_INFO
,
544 MSGTR_AO_ALSA_FormatNotSupportedByHardware
, af_fmt2str_short(format
));
545 alsa_format
= SND_PCM_FORMAT_S16_LE
;
546 ao_data
.format
= AF_FORMAT_S16_LE
;
549 if ((err
= snd_pcm_hw_params_set_format(alsa_handler
, alsa_hwparams
,
552 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetFormat
,
557 if ((err
= snd_pcm_hw_params_set_channels_near(alsa_handler
, alsa_hwparams
,
558 &ao_data
.channels
)) < 0)
560 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetChannels
,
565 /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
566 prefer our own resampler */
567 #if SND_LIB_VERSION >= 0x010009
568 if ((err
= snd_pcm_hw_params_set_rate_resample(alsa_handler
, alsa_hwparams
,
571 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToDisableResampling
,
577 if ((err
= snd_pcm_hw_params_set_rate_near(alsa_handler
, alsa_hwparams
,
578 &ao_data
.samplerate
, NULL
)) < 0)
580 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetSamplerate2
,
585 bytes_per_sample
= snd_pcm_format_physical_width(alsa_format
) / 8;
586 bytes_per_sample
*= ao_data
.channels
;
587 ao_data
.bps
= ao_data
.samplerate
* bytes_per_sample
;
591 int alsa_buffer_time
= 500000; /* original 60 */
592 int alsa_period_time
;
593 alsa_period_time
= alsa_buffer_time
/4;
594 if ((err
= snd_pcm_hw_params_set_buffer_time_near(alsa_handler
, alsa_hwparams
,
595 &alsa_buffer_time
, NULL
)) < 0)
597 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetBufferTimeNear
,
601 alsa_buffer_time
= err
;
603 if ((err
= snd_pcm_hw_params_set_period_time_near(alsa_handler
, alsa_hwparams
,
604 &alsa_period_time
, NULL
)) < 0)
605 /* original: alsa_buffer_time/ao_data.bps */
607 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetPeriodTime
,
611 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_BufferTimePeriodTime
,
612 alsa_buffer_time
, err
);
614 #endif//end SET_BUFFERTIME
619 if ((err
= snd_pcm_hw_params_set_period_size_near(alsa_handler
, alsa_hwparams
,
620 &chunk_size
, NULL
)) < 0)
622 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetPeriodSize
,
623 chunk_size
, snd_strerror(err
));
627 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set to %li\n", chunk_size
);
629 if ((err
= snd_pcm_hw_params_set_periods_near(alsa_handler
, alsa_hwparams
,
630 &alsa_fragcount
, NULL
)) < 0) {
631 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetPeriods
,
636 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: fragcount=%i\n", alsa_fragcount
);
639 #endif//end SET_CHUNKSIZE
641 /* finally install hardware parameters */
642 if ((err
= snd_pcm_hw_params(alsa_handler
, alsa_hwparams
)) < 0)
644 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetHwParameters
,
648 // end setting hw-params
651 // gets buffersize for control
652 if ((err
= snd_pcm_hw_params_get_buffer_size(alsa_hwparams
, &bufsize
)) < 0)
654 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetBufferSize
, snd_strerror(err
));
658 ao_data
.buffersize
= bufsize
* bytes_per_sample
;
659 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: got buffersize=%i\n", ao_data
.buffersize
);
662 if ((err
= snd_pcm_hw_params_get_period_size(alsa_hwparams
, &chunk_size
, NULL
)) < 0) {
663 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetPeriodSize
, snd_strerror(err
));
666 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: got period size %li\n", chunk_size
);
668 ao_data
.outburst
= chunk_size
* bytes_per_sample
;
670 /* setting software parameters */
671 if ((err
= snd_pcm_sw_params_current(alsa_handler
, alsa_swparams
)) < 0) {
672 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetSwParameters
,
676 #if SND_LIB_VERSION >= 0x000901
677 if ((err
= snd_pcm_sw_params_get_boundary(alsa_swparams
, &boundary
)) < 0) {
678 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetBoundary
,
683 boundary
= 0x7fffffff;
685 /* start playing when one period has been written */
686 if ((err
= snd_pcm_sw_params_set_start_threshold(alsa_handler
, alsa_swparams
, chunk_size
)) < 0) {
687 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetStartThreshold
,
691 /* disable underrun reporting */
692 if ((err
= snd_pcm_sw_params_set_stop_threshold(alsa_handler
, alsa_swparams
, boundary
)) < 0) {
693 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetStopThreshold
,
697 #if SND_LIB_VERSION >= 0x000901
698 /* play silence when there is an underrun */
699 if ((err
= snd_pcm_sw_params_set_silence_size(alsa_handler
, alsa_swparams
, boundary
)) < 0) {
700 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetSilenceSize
,
705 if ((err
= snd_pcm_sw_params(alsa_handler
, alsa_swparams
)) < 0) {
706 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetSwParameters
,
710 /* end setting sw-params */
712 mp_msg(MSGT_AO
,MSGL_V
,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
713 ao_data
.samplerate
, ao_data
.channels
, (int)bytes_per_sample
, ao_data
.buffersize
,
714 snd_pcm_format_description(alsa_format
));
716 } // end switch alsa_handler (spdif)
717 alsa_can_pause
= snd_pcm_hw_params_can_pause(alsa_hwparams
);
722 /* close audio device */
723 static void uninit(int immed
)
730 snd_pcm_drain(alsa_handler
);
732 if ((err
= snd_pcm_close(alsa_handler
)) < 0)
734 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmCloseError
, snd_strerror(err
));
739 mp_msg(MSGT_AO
,MSGL_V
,"alsa-uninit: pcm closed\n");
743 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_NoHandlerDefined
);
747 static void audio_pause(void)
751 if (alsa_can_pause
) {
752 if ((err
= snd_pcm_pause(alsa_handler
, 1)) < 0)
754 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPauseError
, snd_strerror(err
));
757 mp_msg(MSGT_AO
,MSGL_V
,"alsa-pause: pause supported by hardware\n");
759 if (snd_pcm_delay(alsa_handler
, &prepause_frames
) < 0
760 || prepause_frames
< 0)
763 if ((err
= snd_pcm_drop(alsa_handler
)) < 0)
765 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmDropError
, snd_strerror(err
));
771 static void audio_resume(void)
775 if (snd_pcm_state(alsa_handler
) == SND_PCM_STATE_SUSPENDED
) {
776 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume
);
777 while ((err
= snd_pcm_resume(alsa_handler
)) == -EAGAIN
) sleep(1);
779 if (alsa_can_pause
) {
780 if ((err
= snd_pcm_pause(alsa_handler
, 0)) < 0)
782 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmResumeError
, snd_strerror(err
));
785 mp_msg(MSGT_AO
,MSGL_V
,"alsa-resume: resume supported by hardware\n");
787 if ((err
= snd_pcm_prepare(alsa_handler
)) < 0)
789 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(err
));
792 if (prepause_frames
) {
793 void *silence
= calloc(prepause_frames
, bytes_per_sample
);
794 play(silence
, prepause_frames
* bytes_per_sample
, 0);
800 /* stop playing and empty buffers (for seeking/pause) */
801 static void reset(void)
806 if ((err
= snd_pcm_drop(alsa_handler
)) < 0)
808 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(err
));
811 if ((err
= snd_pcm_prepare(alsa_handler
)) < 0)
813 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(err
));
820 plays 'len' bytes of 'data'
821 returns: number of bytes played
822 modified last at 29.06.02 by jp
823 thanxs for marius <marius@rospot.com> for giving us the light ;)
826 static int play(void* data
, int len
, int flags
)
828 int num_frames
= len
/ bytes_per_sample
;
829 snd_pcm_sframes_t res
= 0;
831 //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
834 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_DeviceConfigurationError
);
842 res
= snd_pcm_writei(alsa_handler
, data
, num_frames
);
848 else if (res
== -ESTRPIPE
) { /* suspend */
849 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume
);
850 while ((res
= snd_pcm_resume(alsa_handler
)) == -EAGAIN
)
854 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_WriteError
, snd_strerror(res
));
855 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_TryingToResetSoundcard
);
856 if ((res
= snd_pcm_prepare(alsa_handler
)) < 0) {
857 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(res
));
864 return res
< 0 ? res
: res
* bytes_per_sample
;
867 /* how many byes are free in the buffer */
868 static int get_space(void)
870 snd_pcm_status_t
*status
;
873 snd_pcm_status_alloca(&status
);
875 if ((ret
= snd_pcm_status(alsa_handler
, status
)) < 0)
877 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_CannotGetPcmStatus
, snd_strerror(ret
));
881 unsigned space
= snd_pcm_status_get_avail(status
) * bytes_per_sample
;
882 if (space
> ao_data
.buffersize
) // Buffer underrun?
883 space
= ao_data
.buffersize
;
887 /* delay in seconds between first and last sample in buffer */
888 static float get_delay(void)
891 snd_pcm_sframes_t delay
;
893 if (snd_pcm_delay(alsa_handler
, &delay
) < 0)
897 /* underrun - move the application pointer forward to catch up */
898 #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
899 snd_pcm_forward(alsa_handler
, -delay
);
903 return (float)delay
/ (float)ao_data
.samplerate
;