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.
24 #include "subopt-helper.h"
29 #define ALSA_PCM_NEW_HW_PARAMS_API
30 #define ALSA_PCM_NEW_SW_PARAMS_API
32 #if HAVE_SYS_ASOUNDLIB_H
33 #include <sys/asoundlib.h>
34 #elif HAVE_ALSA_ASOUNDLIB_H
35 #include <alsa/asoundlib.h>
37 #error "asoundlib.h is not in sys/ or alsa/ - please bugreport"
41 #include "audio_out.h"
42 #include "audio_out_internal.h"
43 #include "libaf/af_format.h"
45 static ao_info_t info
=
47 "ALSA-0.9.x-1.x audio output",
49 "Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
55 static snd_pcm_t
*alsa_handler
;
56 static snd_pcm_format_t alsa_format
;
57 static snd_pcm_hw_params_t
*alsa_hwparams
;
58 static snd_pcm_sw_params_t
*alsa_swparams
;
60 /* 16 sets buffersize to 16 * chunksize is as default 1024
61 * which seems to be good avarge for most situations
62 * so buffersize is 16384 frames by default */
63 static int alsa_fragcount
= 16;
64 static snd_pcm_uframes_t chunk_size
= 1024;
66 static size_t bytes_per_sample
;
68 static int ao_noblock
= 0;
71 static int alsa_can_pause
= 0;
73 #define ALSA_DEVICE_SIZE 256
78 static void alsa_error_handler(const char *file
, int line
, const char *function
,
79 int err
, const char *format
, ...)
85 vsnprintf(tmp
, sizeof tmp
, format
, va
);
87 tmp
[sizeof tmp
- 1] = '\0';
90 mp_msg(MSGT_AO
, MSGL_ERR
, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
91 file
, line
, function
, tmp
, snd_strerror(err
));
93 mp_msg(MSGT_AO
, MSGL_ERR
, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
94 file
, line
, function
, tmp
);
97 /* to set/get/query special features/parameters */
98 static int control(int cmd
, void *arg
)
101 case AOCONTROL_QUERY_FORMAT
:
103 case AOCONTROL_GET_VOLUME
:
104 case AOCONTROL_SET_VOLUME
:
106 ao_control_vol_t
*vol
= (ao_control_vol_t
*)arg
;
110 snd_mixer_elem_t
*elem
;
111 snd_mixer_selem_id_t
*sid
;
113 static char *mix_name
= "PCM";
114 static char *card
= "default";
115 static int mix_index
= 0;
118 long get_vol
, set_vol
;
122 char *test_mix_index
;
124 mix_name
= strdup(mixer_channel
);
125 if ((test_mix_index
= strchr(mix_name
, ','))){
128 mix_index
= strtol(test_mix_index
, &test_mix_index
, 0);
130 if (*test_mix_index
){
131 mp_msg(MSGT_AO
,MSGL_ERR
,
132 MSGTR_AO_ALSA_InvalidMixerIndexDefaultingToZero
);
137 if(mixer_device
) card
= mixer_device
;
139 if(ao_data
.format
== AF_FORMAT_AC3
)
143 snd_mixer_selem_id_alloca(&sid
);
145 //sets simple-mixer index and name
146 snd_mixer_selem_id_set_index(sid
, mix_index
);
147 snd_mixer_selem_id_set_name(sid
, mix_name
);
154 if ((err
= snd_mixer_open(&handle
, 0)) < 0) {
155 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerOpenError
, snd_strerror(err
));
156 return CONTROL_ERROR
;
159 if ((err
= snd_mixer_attach(handle
, card
)) < 0) {
160 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerAttachError
,
161 card
, snd_strerror(err
));
162 snd_mixer_close(handle
);
163 return CONTROL_ERROR
;
166 if ((err
= snd_mixer_selem_register(handle
, NULL
, NULL
)) < 0) {
167 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerRegisterError
, snd_strerror(err
));
168 snd_mixer_close(handle
);
169 return CONTROL_ERROR
;
171 err
= snd_mixer_load(handle
);
173 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerLoadError
, snd_strerror(err
));
174 snd_mixer_close(handle
);
175 return CONTROL_ERROR
;
178 elem
= snd_mixer_find_selem(handle
, sid
);
180 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToFindSimpleControl
,
181 snd_mixer_selem_id_get_name(sid
), snd_mixer_selem_id_get_index(sid
));
182 snd_mixer_close(handle
);
183 return CONTROL_ERROR
;
186 snd_mixer_selem_get_playback_volume_range(elem
,&pmin
,&pmax
);
187 f_multi
= (100 / (float)(pmax
- pmin
));
189 if (cmd
== AOCONTROL_SET_VOLUME
) {
191 set_vol
= vol
->left
/ f_multi
+ pmin
+ 0.5;
194 if ((err
= snd_mixer_selem_set_playback_volume(elem
, 0, set_vol
)) < 0) {
195 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ErrorSettingLeftChannel
,
197 return CONTROL_ERROR
;
199 mp_msg(MSGT_AO
,MSGL_DBG2
,"left=%li, ", set_vol
);
201 set_vol
= vol
->right
/ f_multi
+ pmin
+ 0.5;
203 if ((err
= snd_mixer_selem_set_playback_volume(elem
, 1, set_vol
)) < 0) {
204 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ErrorSettingRightChannel
,
206 return CONTROL_ERROR
;
208 mp_msg(MSGT_AO
,MSGL_DBG2
,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
209 set_vol
, pmin
, pmax
, f_multi
);
211 if (snd_mixer_selem_has_playback_switch(elem
)) {
212 int lmute
= (vol
->left
== 0.0);
213 int rmute
= (vol
->right
== 0.0);
214 if (snd_mixer_selem_has_playback_switch_joined(elem
)) {
215 lmute
= rmute
= lmute
&& rmute
;
217 snd_mixer_selem_set_playback_switch(elem
, SND_MIXER_SCHN_FRONT_RIGHT
, !rmute
);
219 snd_mixer_selem_set_playback_switch(elem
, SND_MIXER_SCHN_FRONT_LEFT
, !lmute
);
223 snd_mixer_selem_get_playback_volume(elem
, 0, &get_vol
);
224 vol
->left
= (get_vol
- pmin
) * f_multi
;
225 snd_mixer_selem_get_playback_volume(elem
, 1, &get_vol
);
226 vol
->right
= (get_vol
- pmin
) * f_multi
;
228 mp_msg(MSGT_AO
,MSGL_DBG2
,"left=%f, right=%f\n",vol
->left
,vol
->right
);
230 snd_mixer_close(handle
);
235 return(CONTROL_UNKNOWN
);
238 static void parse_device (char *dest
, const char *src
, int len
)
241 memmove(dest
, src
, len
);
243 while ((tmp
= strrchr(dest
, '.')))
245 while ((tmp
= strrchr(dest
, '=')))
249 static void print_help (void)
251 mp_msg (MSGT_AO
, MSGL_FATAL
,
252 MSGTR_AO_ALSA_CommandlineHelp
);
255 static int str_maxlen(strarg_t
*str
) {
256 if (str
->len
> ALSA_DEVICE_SIZE
)
261 static int try_open_device(const char *device
, int open_mode
, int try_ac3
)
264 char *ac3_device
, *args
;
267 /* to set the non-audio bit, use AES0=6 */
268 len
= strlen(device
);
269 ac3_device
= malloc(len
+ 7 + 1);
272 strcpy(ac3_device
, device
);
273 args
= strchr(ac3_device
, ':');
275 /* no existing parameters: add it behind device name */
276 strcat(ac3_device
, ":AES0=6");
280 while (isspace(*args
));
282 /* ":" but no parameters */
283 strcat(ac3_device
, "AES0=6");
284 } else if (*args
!= '{') {
285 /* a simple list of parameters: add it at the end of the list */
286 strcat(ac3_device
, ",AES0=6");
288 /* parameters in config syntax: add it inside the { } block */
291 while (len
> 0 && isspace(ac3_device
[len
]));
292 if (ac3_device
[len
] == '}')
293 strcpy(ac3_device
+ len
, " AES0=6}");
296 err
= snd_pcm_open(&alsa_handler
, ac3_device
, SND_PCM_STREAM_PLAYBACK
,
300 if (!try_ac3
|| err
< 0)
301 err
= snd_pcm_open(&alsa_handler
, device
, SND_PCM_STREAM_PLAYBACK
,
307 open & setup audio device
308 return: 1=success 0=fail
310 static int init(int rate_hz
, int channels
, int format
, int flags
)
315 snd_pcm_uframes_t bufsize
;
316 snd_pcm_uframes_t boundary
;
318 {"block", OPT_ARG_BOOL
, &block
, NULL
},
319 {"device", OPT_ARG_STR
, &device
, (opt_test_f
)str_maxlen
},
323 char alsa_device
[ALSA_DEVICE_SIZE
+ 1];
324 // make sure alsa_device is null-terminated even when using strncpy etc.
325 memset(alsa_device
, 0, ALSA_DEVICE_SIZE
+ 1);
327 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz
,
330 #if SND_LIB_VERSION >= 0x010005
331 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
333 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR
);
336 snd_lib_error_set_handler(alsa_error_handler
);
338 ao_data
.samplerate
= rate_hz
;
339 ao_data
.format
= format
;
340 ao_data
.channels
= channels
;
345 alsa_format
= SND_PCM_FORMAT_S8
;
348 alsa_format
= SND_PCM_FORMAT_U8
;
350 case AF_FORMAT_U16_LE
:
351 alsa_format
= SND_PCM_FORMAT_U16_LE
;
353 case AF_FORMAT_U16_BE
:
354 alsa_format
= SND_PCM_FORMAT_U16_BE
;
356 #ifndef WORDS_BIGENDIAN
359 case AF_FORMAT_S16_LE
:
360 alsa_format
= SND_PCM_FORMAT_S16_LE
;
362 #ifdef WORDS_BIGENDIAN
365 case AF_FORMAT_S16_BE
:
366 alsa_format
= SND_PCM_FORMAT_S16_BE
;
368 case AF_FORMAT_U32_LE
:
369 alsa_format
= SND_PCM_FORMAT_U32_LE
;
371 case AF_FORMAT_U32_BE
:
372 alsa_format
= SND_PCM_FORMAT_U32_BE
;
374 case AF_FORMAT_S32_LE
:
375 alsa_format
= SND_PCM_FORMAT_S32_LE
;
377 case AF_FORMAT_S32_BE
:
378 alsa_format
= SND_PCM_FORMAT_S32_BE
;
380 case AF_FORMAT_FLOAT_LE
:
381 alsa_format
= SND_PCM_FORMAT_FLOAT_LE
;
383 case AF_FORMAT_FLOAT_BE
:
384 alsa_format
= SND_PCM_FORMAT_FLOAT_BE
;
386 case AF_FORMAT_MU_LAW
:
387 alsa_format
= SND_PCM_FORMAT_MU_LAW
;
389 case AF_FORMAT_A_LAW
:
390 alsa_format
= SND_PCM_FORMAT_A_LAW
;
394 alsa_format
= SND_PCM_FORMAT_MPEG
; //? default should be -1
402 * sets opening sequence for SPDIF
403 * sets also the playback and other switches 'on the fly'
404 * while opening the abstract alias for the spdif subdevice
407 if (format
== AF_FORMAT_AC3
) {
408 device
.str
= "iec958";
409 mp_msg(MSGT_AO
,MSGL_V
,"alsa-spdif-init: playing AC3, %i channels\n", channels
);
412 /* in any case for multichannel playback we should select
418 device
.str
= "default";
419 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: setup for 1/2 channel(s)\n");
422 if (alsa_format
== SND_PCM_FORMAT_FLOAT_LE
)
423 // hack - use the converter plugin
424 device
.str
= "plug:surround40";
426 device
.str
= "surround40";
427 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: device set to surround40\n");
430 if (alsa_format
== SND_PCM_FORMAT_FLOAT_LE
)
431 device
.str
= "plug:surround51";
433 device
.str
= "surround51";
434 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: device set to surround51\n");
437 device
.str
= "default";
438 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ChannelsNotSupported
,channels
);
440 device
.len
= strlen(device
.str
);
441 if (subopt_parse(ao_subdevice
, subopts
) != 0) {
446 parse_device(alsa_device
, device
.str
, device
.len
);
448 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: using device %s\n", alsa_device
);
450 //setting modes for block or nonblock-mode
452 open_mode
= SND_PCM_NONBLOCK
;
458 //sets buff/chunksize if its set manually
459 if (ao_data
.buffersize
) {
460 switch (ao_data
.buffersize
)
465 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 8192\n");
466 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 512\n");
471 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 8192\n");
472 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 1024\n");
477 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 16384\n");
478 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 512\n");
483 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 16384\n");
484 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 1024\n");
494 //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
495 if ((err
= try_open_device(alsa_device
, open_mode
, format
== AF_FORMAT_AC3
)) < 0)
497 if (err
!= -EBUSY
&& ao_noblock
) {
498 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_OpenInNonblockModeFailed
);
499 if ((err
= try_open_device(alsa_device
, 0, format
== AF_FORMAT_AC3
)) < 0) {
500 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PlaybackOpenError
, snd_strerror(err
));
504 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PlaybackOpenError
, snd_strerror(err
));
509 if ((err
= snd_pcm_nonblock(alsa_handler
, 0)) < 0) {
510 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ErrorSetBlockMode
, snd_strerror(err
));
512 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: pcm opened in blocking mode\n");
515 snd_pcm_hw_params_alloca(&alsa_hwparams
);
516 snd_pcm_sw_params_alloca(&alsa_swparams
);
518 // setting hw-parameters
519 if ((err
= snd_pcm_hw_params_any(alsa_handler
, alsa_hwparams
)) < 0)
521 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetInitialParameters
,
526 err
= snd_pcm_hw_params_set_access(alsa_handler
, alsa_hwparams
,
527 SND_PCM_ACCESS_RW_INTERLEAVED
);
529 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetAccessType
,
534 /* workaround for nonsupported formats
535 sets default format to S16_LE if the given formats aren't supported */
536 if ((err
= snd_pcm_hw_params_test_format(alsa_handler
, alsa_hwparams
,
539 mp_msg(MSGT_AO
,MSGL_INFO
,
540 MSGTR_AO_ALSA_FormatNotSupportedByHardware
, af_fmt2str_short(format
));
541 alsa_format
= SND_PCM_FORMAT_S16_LE
;
542 ao_data
.format
= AF_FORMAT_S16_LE
;
545 if ((err
= snd_pcm_hw_params_set_format(alsa_handler
, alsa_hwparams
,
548 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetFormat
,
553 if ((err
= snd_pcm_hw_params_set_channels_near(alsa_handler
, alsa_hwparams
,
554 &ao_data
.channels
)) < 0)
556 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetChannels
,
561 /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
562 prefer our own resampler */
563 #if SND_LIB_VERSION >= 0x010009
564 if ((err
= snd_pcm_hw_params_set_rate_resample(alsa_handler
, alsa_hwparams
,
567 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToDisableResampling
,
573 if ((err
= snd_pcm_hw_params_set_rate_near(alsa_handler
, alsa_hwparams
,
574 &ao_data
.samplerate
, NULL
)) < 0)
576 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetSamplerate2
,
581 bytes_per_sample
= snd_pcm_format_physical_width(alsa_format
) / 8;
582 bytes_per_sample
*= ao_data
.channels
;
583 ao_data
.bps
= ao_data
.samplerate
* bytes_per_sample
;
587 int alsa_buffer_time
= 500000; /* original 60 */
588 int alsa_period_time
;
589 alsa_period_time
= alsa_buffer_time
/4;
590 if ((err
= snd_pcm_hw_params_set_buffer_time_near(alsa_handler
, alsa_hwparams
,
591 &alsa_buffer_time
, NULL
)) < 0)
593 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetBufferTimeNear
,
597 alsa_buffer_time
= err
;
599 if ((err
= snd_pcm_hw_params_set_period_time_near(alsa_handler
, alsa_hwparams
,
600 &alsa_period_time
, NULL
)) < 0)
601 /* original: alsa_buffer_time/ao_data.bps */
603 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetPeriodTime
,
607 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_BufferTimePeriodTime
,
608 alsa_buffer_time
, err
);
610 #endif//end SET_BUFFERTIME
615 if ((err
= snd_pcm_hw_params_set_period_size_near(alsa_handler
, alsa_hwparams
,
616 &chunk_size
, NULL
)) < 0)
618 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetPeriodSize
,
619 chunk_size
, snd_strerror(err
));
623 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set to %li\n", chunk_size
);
625 if ((err
= snd_pcm_hw_params_set_periods_near(alsa_handler
, alsa_hwparams
,
626 &alsa_fragcount
, NULL
)) < 0) {
627 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetPeriods
,
632 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: fragcount=%i\n", alsa_fragcount
);
635 #endif//end SET_CHUNKSIZE
637 /* finally install hardware parameters */
638 if ((err
= snd_pcm_hw_params(alsa_handler
, alsa_hwparams
)) < 0)
640 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetHwParameters
,
644 // end setting hw-params
647 // gets buffersize for control
648 if ((err
= snd_pcm_hw_params_get_buffer_size(alsa_hwparams
, &bufsize
)) < 0)
650 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetBufferSize
, snd_strerror(err
));
654 ao_data
.buffersize
= bufsize
* bytes_per_sample
;
655 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: got buffersize=%i\n", ao_data
.buffersize
);
658 if ((err
= snd_pcm_hw_params_get_period_size(alsa_hwparams
, &chunk_size
, NULL
)) < 0) {
659 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetPeriodSize
, snd_strerror(err
));
662 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: got period size %li\n", chunk_size
);
664 ao_data
.outburst
= chunk_size
* bytes_per_sample
;
666 /* setting software parameters */
667 if ((err
= snd_pcm_sw_params_current(alsa_handler
, alsa_swparams
)) < 0) {
668 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetSwParameters
,
672 #if SND_LIB_VERSION >= 0x000901
673 if ((err
= snd_pcm_sw_params_get_boundary(alsa_swparams
, &boundary
)) < 0) {
674 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetBoundary
,
679 boundary
= 0x7fffffff;
681 /* start playing when one period has been written */
682 if ((err
= snd_pcm_sw_params_set_start_threshold(alsa_handler
, alsa_swparams
, chunk_size
)) < 0) {
683 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetStartThreshold
,
687 /* disable underrun reporting */
688 if ((err
= snd_pcm_sw_params_set_stop_threshold(alsa_handler
, alsa_swparams
, boundary
)) < 0) {
689 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetStopThreshold
,
693 #if SND_LIB_VERSION >= 0x000901
694 /* play silence when there is an underrun */
695 if ((err
= snd_pcm_sw_params_set_silence_size(alsa_handler
, alsa_swparams
, boundary
)) < 0) {
696 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetSilenceSize
,
701 if ((err
= snd_pcm_sw_params(alsa_handler
, alsa_swparams
)) < 0) {
702 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetSwParameters
,
706 /* end setting sw-params */
708 mp_msg(MSGT_AO
,MSGL_V
,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
709 ao_data
.samplerate
, ao_data
.channels
, bytes_per_sample
, ao_data
.buffersize
,
710 snd_pcm_format_description(alsa_format
));
712 } // end switch alsa_handler (spdif)
713 alsa_can_pause
= snd_pcm_hw_params_can_pause(alsa_hwparams
);
718 /* close audio device */
719 static void uninit(int immed
)
726 snd_pcm_drain(alsa_handler
);
728 if ((err
= snd_pcm_close(alsa_handler
)) < 0)
730 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmCloseError
, snd_strerror(err
));
735 mp_msg(MSGT_AO
,MSGL_V
,"alsa-uninit: pcm closed\n");
739 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_NoHandlerDefined
);
743 static void audio_pause(void)
747 if (alsa_can_pause
) {
748 if ((err
= snd_pcm_pause(alsa_handler
, 1)) < 0)
750 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPauseError
, snd_strerror(err
));
753 mp_msg(MSGT_AO
,MSGL_V
,"alsa-pause: pause supported by hardware\n");
755 if ((err
= snd_pcm_drop(alsa_handler
)) < 0)
757 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmDropError
, snd_strerror(err
));
763 static void audio_resume(void)
767 if (alsa_can_pause
) {
768 if ((err
= snd_pcm_pause(alsa_handler
, 0)) < 0)
770 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmResumeError
, snd_strerror(err
));
773 mp_msg(MSGT_AO
,MSGL_V
,"alsa-resume: resume supported by hardware\n");
775 if ((err
= snd_pcm_prepare(alsa_handler
)) < 0)
777 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(err
));
783 /* stop playing and empty buffers (for seeking/pause) */
784 static void reset(void)
788 if ((err
= snd_pcm_drop(alsa_handler
)) < 0)
790 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(err
));
793 if ((err
= snd_pcm_prepare(alsa_handler
)) < 0)
795 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(err
));
802 plays 'len' bytes of 'data'
803 returns: number of bytes played
804 modified last at 29.06.02 by jp
805 thanxs for marius <marius@rospot.com> for giving us the light ;)
808 static int play(void* data
, int len
, int flags
)
810 int num_frames
= len
/ bytes_per_sample
;
811 snd_pcm_sframes_t res
= 0;
813 //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
816 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_DeviceConfigurationError
);
824 res
= snd_pcm_writei(alsa_handler
, data
, num_frames
);
830 else if (res
== -ESTRPIPE
) { /* suspend */
831 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume
);
832 while ((res
= snd_pcm_resume(alsa_handler
)) == -EAGAIN
)
836 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_WriteError
, snd_strerror(res
));
837 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_TryingToResetSoundcard
);
838 if ((res
= snd_pcm_prepare(alsa_handler
)) < 0) {
839 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(res
));
846 return res
< 0 ? res
: res
* bytes_per_sample
;
849 /* how many byes are free in the buffer */
850 static int get_space(void)
852 snd_pcm_status_t
*status
;
855 snd_pcm_status_alloca(&status
);
857 if ((ret
= snd_pcm_status(alsa_handler
, status
)) < 0)
859 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_CannotGetPcmStatus
, snd_strerror(ret
));
863 ret
= snd_pcm_status_get_avail(status
) * bytes_per_sample
;
864 if (ret
> ao_data
.buffersize
) // Buffer underrun?
865 ret
= ao_data
.buffersize
;
869 /* delay in seconds between first and last sample in buffer */
870 static float get_delay(void)
873 snd_pcm_sframes_t delay
;
875 if (snd_pcm_delay(alsa_handler
, &delay
) < 0)
879 /* underrun - move the application pointer forward to catch up */
880 #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
881 snd_pcm_forward(alsa_handler
, -delay
);
885 return (float)delay
/ (float)ao_data
.samplerate
;