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;
74 #define ALSA_DEVICE_SIZE 256
79 static void alsa_error_handler(const char *file
, int line
, const char *function
,
80 int err
, const char *format
, ...)
86 vsnprintf(tmp
, sizeof tmp
, format
, va
);
88 tmp
[sizeof tmp
- 1] = '\0';
91 mp_msg(MSGT_AO
, MSGL_ERR
, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
92 file
, line
, function
, tmp
, snd_strerror(err
));
94 mp_msg(MSGT_AO
, MSGL_ERR
, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
95 file
, line
, function
, tmp
);
98 /* to set/get/query special features/parameters */
99 static int control(int cmd
, void *arg
)
102 case AOCONTROL_QUERY_FORMAT
:
104 case AOCONTROL_GET_VOLUME
:
105 case AOCONTROL_SET_VOLUME
:
107 ao_control_vol_t
*vol
= (ao_control_vol_t
*)arg
;
111 snd_mixer_elem_t
*elem
;
112 snd_mixer_selem_id_t
*sid
;
114 static char *mix_name
= "PCM";
115 static char *card
= "default";
116 static int mix_index
= 0;
119 long get_vol
, set_vol
;
123 char *test_mix_index
;
125 mix_name
= strdup(mixer_channel
);
126 if ((test_mix_index
= strchr(mix_name
, ','))){
129 mix_index
= strtol(test_mix_index
, &test_mix_index
, 0);
131 if (*test_mix_index
){
132 mp_msg(MSGT_AO
,MSGL_ERR
,
133 MSGTR_AO_ALSA_InvalidMixerIndexDefaultingToZero
);
138 if(mixer_device
) card
= mixer_device
;
140 if(ao_data
.format
== AF_FORMAT_AC3
)
144 snd_mixer_selem_id_alloca(&sid
);
146 //sets simple-mixer index and name
147 snd_mixer_selem_id_set_index(sid
, mix_index
);
148 snd_mixer_selem_id_set_name(sid
, mix_name
);
155 if ((err
= snd_mixer_open(&handle
, 0)) < 0) {
156 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerOpenError
, snd_strerror(err
));
157 return CONTROL_ERROR
;
160 if ((err
= snd_mixer_attach(handle
, card
)) < 0) {
161 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerAttachError
,
162 card
, snd_strerror(err
));
163 snd_mixer_close(handle
);
164 return CONTROL_ERROR
;
167 if ((err
= snd_mixer_selem_register(handle
, NULL
, NULL
)) < 0) {
168 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerRegisterError
, snd_strerror(err
));
169 snd_mixer_close(handle
);
170 return CONTROL_ERROR
;
172 err
= snd_mixer_load(handle
);
174 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerLoadError
, snd_strerror(err
));
175 snd_mixer_close(handle
);
176 return CONTROL_ERROR
;
179 elem
= snd_mixer_find_selem(handle
, sid
);
181 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToFindSimpleControl
,
182 snd_mixer_selem_id_get_name(sid
), snd_mixer_selem_id_get_index(sid
));
183 snd_mixer_close(handle
);
184 return CONTROL_ERROR
;
187 snd_mixer_selem_get_playback_volume_range(elem
,&pmin
,&pmax
);
188 f_multi
= (100 / (float)(pmax
- pmin
));
190 if (cmd
== AOCONTROL_SET_VOLUME
) {
192 set_vol
= vol
->left
/ f_multi
+ pmin
+ 0.5;
195 if ((err
= snd_mixer_selem_set_playback_volume(elem
, SND_MIXER_SCHN_FRONT_LEFT
, set_vol
)) < 0) {
196 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ErrorSettingLeftChannel
,
198 return CONTROL_ERROR
;
200 mp_msg(MSGT_AO
,MSGL_DBG2
,"left=%li, ", set_vol
);
202 set_vol
= vol
->right
/ f_multi
+ pmin
+ 0.5;
204 if ((err
= snd_mixer_selem_set_playback_volume(elem
, SND_MIXER_SCHN_FRONT_RIGHT
, set_vol
)) < 0) {
205 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ErrorSettingRightChannel
,
207 return CONTROL_ERROR
;
209 mp_msg(MSGT_AO
,MSGL_DBG2
,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
210 set_vol
, pmin
, pmax
, f_multi
);
212 if (snd_mixer_selem_has_playback_switch(elem
)) {
213 int lmute
= (vol
->left
== 0.0);
214 int rmute
= (vol
->right
== 0.0);
215 if (snd_mixer_selem_has_playback_switch_joined(elem
)) {
216 lmute
= rmute
= lmute
&& rmute
;
218 snd_mixer_selem_set_playback_switch(elem
, SND_MIXER_SCHN_FRONT_RIGHT
, !rmute
);
220 snd_mixer_selem_set_playback_switch(elem
, SND_MIXER_SCHN_FRONT_LEFT
, !lmute
);
224 snd_mixer_selem_get_playback_volume(elem
, SND_MIXER_SCHN_FRONT_LEFT
, &get_vol
);
225 vol
->left
= (get_vol
- pmin
) * f_multi
;
226 snd_mixer_selem_get_playback_volume(elem
, SND_MIXER_SCHN_FRONT_RIGHT
, &get_vol
);
227 vol
->right
= (get_vol
- pmin
) * f_multi
;
229 mp_msg(MSGT_AO
,MSGL_DBG2
,"left=%f, right=%f\n",vol
->left
,vol
->right
);
231 snd_mixer_close(handle
);
236 return CONTROL_UNKNOWN
;
239 static void parse_device (char *dest
, const char *src
, int len
)
242 memmove(dest
, src
, len
);
244 while ((tmp
= strrchr(dest
, '.')))
246 while ((tmp
= strrchr(dest
, '=')))
250 static void print_help (void)
252 mp_msg (MSGT_AO
, MSGL_FATAL
,
253 MSGTR_AO_ALSA_CommandlineHelp
);
256 static int str_maxlen(strarg_t
*str
) {
257 if (str
->len
> ALSA_DEVICE_SIZE
)
262 static int try_open_device(const char *device
, int open_mode
, int try_ac3
)
265 char *ac3_device
, *args
;
268 /* to set the non-audio bit, use AES0=6 */
269 len
= strlen(device
);
270 ac3_device
= malloc(len
+ 7 + 1);
273 strcpy(ac3_device
, device
);
274 args
= strchr(ac3_device
, ':');
276 /* no existing parameters: add it behind device name */
277 strcat(ac3_device
, ":AES0=6");
281 while (isspace(*args
));
283 /* ":" but no parameters */
284 strcat(ac3_device
, "AES0=6");
285 } else if (*args
!= '{') {
286 /* a simple list of parameters: add it at the end of the list */
287 strcat(ac3_device
, ",AES0=6");
289 /* parameters in config syntax: add it inside the { } block */
292 while (len
> 0 && isspace(ac3_device
[len
]));
293 if (ac3_device
[len
] == '}')
294 strcpy(ac3_device
+ len
, " AES0=6}");
297 err
= snd_pcm_open(&alsa_handler
, ac3_device
, SND_PCM_STREAM_PLAYBACK
,
301 if (!try_ac3
|| err
< 0)
302 err
= snd_pcm_open(&alsa_handler
, device
, SND_PCM_STREAM_PLAYBACK
,
308 open & setup audio device
309 return: 1=success 0=fail
311 static int init(int rate_hz
, int channels
, int format
, int flags
)
316 snd_pcm_uframes_t bufsize
;
317 snd_pcm_uframes_t boundary
;
319 {"block", OPT_ARG_BOOL
, &block
, NULL
},
320 {"device", OPT_ARG_STR
, &device
, (opt_test_f
)str_maxlen
},
324 char alsa_device
[ALSA_DEVICE_SIZE
+ 1];
325 // make sure alsa_device is null-terminated even when using strncpy etc.
326 memset(alsa_device
, 0, ALSA_DEVICE_SIZE
+ 1);
328 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz
,
331 #if SND_LIB_VERSION >= 0x010005
332 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
334 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR
);
337 snd_lib_error_set_handler(alsa_error_handler
);
339 ao_data
.samplerate
= rate_hz
;
340 ao_data
.format
= format
;
341 ao_data
.channels
= channels
;
346 alsa_format
= SND_PCM_FORMAT_S8
;
349 alsa_format
= SND_PCM_FORMAT_U8
;
351 case AF_FORMAT_U16_LE
:
352 alsa_format
= SND_PCM_FORMAT_U16_LE
;
354 case AF_FORMAT_U16_BE
:
355 alsa_format
= SND_PCM_FORMAT_U16_BE
;
357 #ifndef WORDS_BIGENDIAN
360 case AF_FORMAT_S16_LE
:
361 alsa_format
= SND_PCM_FORMAT_S16_LE
;
363 #ifdef WORDS_BIGENDIAN
366 case AF_FORMAT_S16_BE
:
367 alsa_format
= SND_PCM_FORMAT_S16_BE
;
369 case AF_FORMAT_U32_LE
:
370 alsa_format
= SND_PCM_FORMAT_U32_LE
;
372 case AF_FORMAT_U32_BE
:
373 alsa_format
= SND_PCM_FORMAT_U32_BE
;
375 case AF_FORMAT_S32_LE
:
376 alsa_format
= SND_PCM_FORMAT_S32_LE
;
378 case AF_FORMAT_S32_BE
:
379 alsa_format
= SND_PCM_FORMAT_S32_BE
;
381 case AF_FORMAT_FLOAT_LE
:
382 alsa_format
= SND_PCM_FORMAT_FLOAT_LE
;
384 case AF_FORMAT_FLOAT_BE
:
385 alsa_format
= SND_PCM_FORMAT_FLOAT_BE
;
387 case AF_FORMAT_MU_LAW
:
388 alsa_format
= SND_PCM_FORMAT_MU_LAW
;
390 case AF_FORMAT_A_LAW
:
391 alsa_format
= SND_PCM_FORMAT_A_LAW
;
395 alsa_format
= SND_PCM_FORMAT_MPEG
; //? default should be -1
403 * sets opening sequence for SPDIF
404 * sets also the playback and other switches 'on the fly'
405 * while opening the abstract alias for the spdif subdevice
408 if (format
== AF_FORMAT_AC3
) {
409 device
.str
= "iec958";
410 mp_msg(MSGT_AO
,MSGL_V
,"alsa-spdif-init: playing AC3, %i channels\n", channels
);
413 /* in any case for multichannel playback we should select
419 device
.str
= "default";
420 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: setup for 1/2 channel(s)\n");
423 if (alsa_format
== SND_PCM_FORMAT_FLOAT_LE
)
424 // hack - use the converter plugin
425 device
.str
= "plug:surround40";
427 device
.str
= "surround40";
428 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: device set to surround40\n");
431 if (alsa_format
== SND_PCM_FORMAT_FLOAT_LE
)
432 device
.str
= "plug:surround51";
434 device
.str
= "surround51";
435 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: device set to surround51\n");
438 device
.str
= "default";
439 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ChannelsNotSupported
,channels
);
441 device
.len
= strlen(device
.str
);
442 if (subopt_parse(ao_subdevice
, subopts
) != 0) {
447 parse_device(alsa_device
, device
.str
, device
.len
);
449 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: using device %s\n", alsa_device
);
451 //setting modes for block or nonblock-mode
453 open_mode
= SND_PCM_NONBLOCK
;
459 //sets buff/chunksize if its set manually
460 if (ao_data
.buffersize
) {
461 switch (ao_data
.buffersize
)
466 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 8192\n");
467 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 512\n");
472 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 8192\n");
473 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 1024\n");
478 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 16384\n");
479 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 512\n");
484 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 16384\n");
485 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 1024\n");
495 //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
496 if ((err
= try_open_device(alsa_device
, open_mode
, format
== AF_FORMAT_AC3
)) < 0)
498 if (err
!= -EBUSY
&& ao_noblock
) {
499 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_OpenInNonblockModeFailed
);
500 if ((err
= try_open_device(alsa_device
, 0, format
== AF_FORMAT_AC3
)) < 0) {
501 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PlaybackOpenError
, snd_strerror(err
));
505 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PlaybackOpenError
, snd_strerror(err
));
510 if ((err
= snd_pcm_nonblock(alsa_handler
, 0)) < 0) {
511 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ErrorSetBlockMode
, snd_strerror(err
));
513 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: pcm opened in blocking mode\n");
516 snd_pcm_hw_params_alloca(&alsa_hwparams
);
517 snd_pcm_sw_params_alloca(&alsa_swparams
);
519 // setting hw-parameters
520 if ((err
= snd_pcm_hw_params_any(alsa_handler
, alsa_hwparams
)) < 0)
522 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetInitialParameters
,
527 err
= snd_pcm_hw_params_set_access(alsa_handler
, alsa_hwparams
,
528 SND_PCM_ACCESS_RW_INTERLEAVED
);
530 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetAccessType
,
535 /* workaround for nonsupported formats
536 sets default format to S16_LE if the given formats aren't supported */
537 if ((err
= snd_pcm_hw_params_test_format(alsa_handler
, alsa_hwparams
,
540 mp_msg(MSGT_AO
,MSGL_INFO
,
541 MSGTR_AO_ALSA_FormatNotSupportedByHardware
, af_fmt2str_short(format
));
542 alsa_format
= SND_PCM_FORMAT_S16_LE
;
543 ao_data
.format
= AF_FORMAT_S16_LE
;
546 if ((err
= snd_pcm_hw_params_set_format(alsa_handler
, alsa_hwparams
,
549 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetFormat
,
554 if ((err
= snd_pcm_hw_params_set_channels_near(alsa_handler
, alsa_hwparams
,
555 &ao_data
.channels
)) < 0)
557 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetChannels
,
562 /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
563 prefer our own resampler */
564 #if SND_LIB_VERSION >= 0x010009
565 if ((err
= snd_pcm_hw_params_set_rate_resample(alsa_handler
, alsa_hwparams
,
568 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToDisableResampling
,
574 if ((err
= snd_pcm_hw_params_set_rate_near(alsa_handler
, alsa_hwparams
,
575 &ao_data
.samplerate
, NULL
)) < 0)
577 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetSamplerate2
,
582 bytes_per_sample
= snd_pcm_format_physical_width(alsa_format
) / 8;
583 bytes_per_sample
*= ao_data
.channels
;
584 ao_data
.bps
= ao_data
.samplerate
* bytes_per_sample
;
588 int alsa_buffer_time
= 500000; /* original 60 */
589 int alsa_period_time
;
590 alsa_period_time
= alsa_buffer_time
/4;
591 if ((err
= snd_pcm_hw_params_set_buffer_time_near(alsa_handler
, alsa_hwparams
,
592 &alsa_buffer_time
, NULL
)) < 0)
594 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetBufferTimeNear
,
598 alsa_buffer_time
= err
;
600 if ((err
= snd_pcm_hw_params_set_period_time_near(alsa_handler
, alsa_hwparams
,
601 &alsa_period_time
, NULL
)) < 0)
602 /* original: alsa_buffer_time/ao_data.bps */
604 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetPeriodTime
,
608 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_BufferTimePeriodTime
,
609 alsa_buffer_time
, err
);
611 #endif//end SET_BUFFERTIME
616 if ((err
= snd_pcm_hw_params_set_period_size_near(alsa_handler
, alsa_hwparams
,
617 &chunk_size
, NULL
)) < 0)
619 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetPeriodSize
,
620 chunk_size
, snd_strerror(err
));
624 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set to %li\n", chunk_size
);
626 if ((err
= snd_pcm_hw_params_set_periods_near(alsa_handler
, alsa_hwparams
,
627 &alsa_fragcount
, NULL
)) < 0) {
628 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetPeriods
,
633 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: fragcount=%i\n", alsa_fragcount
);
636 #endif//end SET_CHUNKSIZE
638 /* finally install hardware parameters */
639 if ((err
= snd_pcm_hw_params(alsa_handler
, alsa_hwparams
)) < 0)
641 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetHwParameters
,
645 // end setting hw-params
648 // gets buffersize for control
649 if ((err
= snd_pcm_hw_params_get_buffer_size(alsa_hwparams
, &bufsize
)) < 0)
651 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetBufferSize
, snd_strerror(err
));
655 ao_data
.buffersize
= bufsize
* bytes_per_sample
;
656 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: got buffersize=%i\n", ao_data
.buffersize
);
659 if ((err
= snd_pcm_hw_params_get_period_size(alsa_hwparams
, &chunk_size
, NULL
)) < 0) {
660 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetPeriodSize
, snd_strerror(err
));
663 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: got period size %li\n", chunk_size
);
665 ao_data
.outburst
= chunk_size
* bytes_per_sample
;
667 /* setting software parameters */
668 if ((err
= snd_pcm_sw_params_current(alsa_handler
, alsa_swparams
)) < 0) {
669 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetSwParameters
,
673 #if SND_LIB_VERSION >= 0x000901
674 if ((err
= snd_pcm_sw_params_get_boundary(alsa_swparams
, &boundary
)) < 0) {
675 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetBoundary
,
680 boundary
= 0x7fffffff;
682 /* start playing when one period has been written */
683 if ((err
= snd_pcm_sw_params_set_start_threshold(alsa_handler
, alsa_swparams
, chunk_size
)) < 0) {
684 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetStartThreshold
,
688 /* disable underrun reporting */
689 if ((err
= snd_pcm_sw_params_set_stop_threshold(alsa_handler
, alsa_swparams
, boundary
)) < 0) {
690 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetStopThreshold
,
694 #if SND_LIB_VERSION >= 0x000901
695 /* play silence when there is an underrun */
696 if ((err
= snd_pcm_sw_params_set_silence_size(alsa_handler
, alsa_swparams
, boundary
)) < 0) {
697 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetSilenceSize
,
702 if ((err
= snd_pcm_sw_params(alsa_handler
, alsa_swparams
)) < 0) {
703 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetSwParameters
,
707 /* end setting sw-params */
709 mp_msg(MSGT_AO
,MSGL_V
,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
710 ao_data
.samplerate
, ao_data
.channels
, (int)bytes_per_sample
, ao_data
.buffersize
,
711 snd_pcm_format_description(alsa_format
));
713 } // end switch alsa_handler (spdif)
714 alsa_can_pause
= snd_pcm_hw_params_can_pause(alsa_hwparams
);
719 /* close audio device */
720 static void uninit(int immed
)
727 snd_pcm_drain(alsa_handler
);
729 if ((err
= snd_pcm_close(alsa_handler
)) < 0)
731 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmCloseError
, snd_strerror(err
));
736 mp_msg(MSGT_AO
,MSGL_V
,"alsa-uninit: pcm closed\n");
740 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_NoHandlerDefined
);
744 static void audio_pause(void)
748 if (alsa_can_pause
) {
749 if ((err
= snd_pcm_pause(alsa_handler
, 1)) < 0)
751 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPauseError
, snd_strerror(err
));
754 mp_msg(MSGT_AO
,MSGL_V
,"alsa-pause: pause supported by hardware\n");
756 if ((err
= snd_pcm_drop(alsa_handler
)) < 0)
758 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmDropError
, snd_strerror(err
));
764 static void audio_resume(void)
768 if (snd_pcm_state(alsa_handler
) == SND_PCM_STATE_SUSPENDED
) {
769 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume
);
770 while ((err
= snd_pcm_resume(alsa_handler
)) == -EAGAIN
) sleep(1);
772 if (alsa_can_pause
) {
773 if ((err
= snd_pcm_pause(alsa_handler
, 0)) < 0)
775 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmResumeError
, snd_strerror(err
));
778 mp_msg(MSGT_AO
,MSGL_V
,"alsa-resume: resume supported by hardware\n");
780 if ((err
= snd_pcm_prepare(alsa_handler
)) < 0)
782 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(err
));
788 /* stop playing and empty buffers (for seeking/pause) */
789 static void reset(void)
793 if ((err
= snd_pcm_drop(alsa_handler
)) < 0)
795 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(err
));
798 if ((err
= snd_pcm_prepare(alsa_handler
)) < 0)
800 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(err
));
807 plays 'len' bytes of 'data'
808 returns: number of bytes played
809 modified last at 29.06.02 by jp
810 thanxs for marius <marius@rospot.com> for giving us the light ;)
813 static int play(void* data
, int len
, int flags
)
815 int num_frames
= len
/ bytes_per_sample
;
816 snd_pcm_sframes_t res
= 0;
818 //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
821 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_DeviceConfigurationError
);
829 res
= snd_pcm_writei(alsa_handler
, data
, num_frames
);
835 else if (res
== -ESTRPIPE
) { /* suspend */
836 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume
);
837 while ((res
= snd_pcm_resume(alsa_handler
)) == -EAGAIN
)
841 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_WriteError
, snd_strerror(res
));
842 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_TryingToResetSoundcard
);
843 if ((res
= snd_pcm_prepare(alsa_handler
)) < 0) {
844 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(res
));
851 return res
< 0 ? res
: res
* bytes_per_sample
;
854 /* how many byes are free in the buffer */
855 static int get_space(void)
857 snd_pcm_status_t
*status
;
860 snd_pcm_status_alloca(&status
);
862 if ((ret
= snd_pcm_status(alsa_handler
, status
)) < 0)
864 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_CannotGetPcmStatus
, snd_strerror(ret
));
868 ret
= snd_pcm_status_get_avail(status
) * bytes_per_sample
;
869 if (ret
> ao_data
.buffersize
) // Buffer underrun?
870 ret
= ao_data
.buffersize
;
874 /* delay in seconds between first and last sample in buffer */
875 static float get_delay(void)
878 snd_pcm_sframes_t delay
;
880 if (snd_pcm_delay(alsa_handler
, &delay
) < 0)
884 /* underrun - move the application pointer forward to catch up */
885 #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
886 snd_pcm_forward(alsa_handler
, -delay
);
890 return (float)delay
/ (float)ao_data
.samplerate
;