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_msg(MSGT_AO
,MSGL_ERR
,
151 MSGTR_AO_ALSA_InvalidMixerIndexDefaultingToZero
);
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_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerOpenError
, snd_strerror(err
));
172 return CONTROL_ERROR
;
175 if ((err
= snd_mixer_attach(handle
, card
)) < 0) {
176 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerAttachError
,
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_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerRegisterError
, snd_strerror(err
));
184 snd_mixer_close(handle
);
185 return CONTROL_ERROR
;
187 err
= snd_mixer_load(handle
);
189 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_MixerLoadError
, snd_strerror(err
));
190 snd_mixer_close(handle
);
191 return CONTROL_ERROR
;
194 elem
= snd_mixer_find_selem(handle
, sid
);
196 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToFindSimpleControl
,
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_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ErrorSettingLeftChannel
,
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_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ErrorSettingRightChannel
,
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_msg (MSGT_AO
, MSGL_FATAL
,
268 MSGTR_AO_ALSA_CommandlineHelp
);
271 static int str_maxlen(strarg_t
*str
) {
272 if (str
->len
> ALSA_DEVICE_SIZE
)
277 static int try_open_device(const char *device
, int open_mode
, int try_ac3
)
280 char *ac3_device
, *args
;
283 /* to set the non-audio bit, use AES0=6 */
284 len
= strlen(device
);
285 ac3_device
= malloc(len
+ 7 + 1);
288 strcpy(ac3_device
, device
);
289 args
= strchr(ac3_device
, ':');
291 /* no existing parameters: add it behind device name */
292 strcat(ac3_device
, ":AES0=6");
296 while (isspace(*args
));
298 /* ":" but no parameters */
299 strcat(ac3_device
, "AES0=6");
300 } else if (*args
!= '{') {
301 /* a simple list of parameters: add it at the end of the list */
302 strcat(ac3_device
, ",AES0=6");
304 /* parameters in config syntax: add it inside the { } block */
307 while (len
> 0 && isspace(ac3_device
[len
]));
308 if (ac3_device
[len
] == '}')
309 strcpy(ac3_device
+ len
, " AES0=6}");
312 err
= snd_pcm_open(&alsa_handler
, ac3_device
, SND_PCM_STREAM_PLAYBACK
,
316 if (!try_ac3
|| err
< 0)
317 err
= snd_pcm_open(&alsa_handler
, device
, SND_PCM_STREAM_PLAYBACK
,
323 open & setup audio device
324 return: 1=success 0=fail
326 static int init(int rate_hz
, int channels
, int format
, int flags
)
331 snd_pcm_uframes_t bufsize
;
332 snd_pcm_uframes_t boundary
;
334 {"block", OPT_ARG_BOOL
, &block
, NULL
},
335 {"device", OPT_ARG_STR
, &device
, (opt_test_f
)str_maxlen
},
339 char alsa_device
[ALSA_DEVICE_SIZE
+ 1];
340 // make sure alsa_device is null-terminated even when using strncpy etc.
341 memset(alsa_device
, 0, ALSA_DEVICE_SIZE
+ 1);
343 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz
,
346 #if SND_LIB_VERSION >= 0x010005
347 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
349 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR
);
354 snd_lib_error_set_handler(alsa_error_handler
);
356 ao_data
.samplerate
= rate_hz
;
357 ao_data
.format
= format
;
358 ao_data
.channels
= channels
;
363 alsa_format
= SND_PCM_FORMAT_S8
;
366 alsa_format
= SND_PCM_FORMAT_U8
;
368 case AF_FORMAT_U16_LE
:
369 alsa_format
= SND_PCM_FORMAT_U16_LE
;
371 case AF_FORMAT_U16_BE
:
372 alsa_format
= SND_PCM_FORMAT_U16_BE
;
374 #ifndef WORDS_BIGENDIAN
377 case AF_FORMAT_S16_LE
:
378 alsa_format
= SND_PCM_FORMAT_S16_LE
;
380 #ifdef WORDS_BIGENDIAN
383 case AF_FORMAT_S16_BE
:
384 alsa_format
= SND_PCM_FORMAT_S16_BE
;
386 case AF_FORMAT_U32_LE
:
387 alsa_format
= SND_PCM_FORMAT_U32_LE
;
389 case AF_FORMAT_U32_BE
:
390 alsa_format
= SND_PCM_FORMAT_U32_BE
;
392 case AF_FORMAT_S32_LE
:
393 alsa_format
= SND_PCM_FORMAT_S32_LE
;
395 case AF_FORMAT_S32_BE
:
396 alsa_format
= SND_PCM_FORMAT_S32_BE
;
398 case AF_FORMAT_FLOAT_LE
:
399 alsa_format
= SND_PCM_FORMAT_FLOAT_LE
;
401 case AF_FORMAT_FLOAT_BE
:
402 alsa_format
= SND_PCM_FORMAT_FLOAT_BE
;
404 case AF_FORMAT_MU_LAW
:
405 alsa_format
= SND_PCM_FORMAT_MU_LAW
;
407 case AF_FORMAT_A_LAW
:
408 alsa_format
= SND_PCM_FORMAT_A_LAW
;
412 alsa_format
= SND_PCM_FORMAT_MPEG
; //? default should be -1
420 * sets opening sequence for SPDIF
421 * sets also the playback and other switches 'on the fly'
422 * while opening the abstract alias for the spdif subdevice
425 if (format
== AF_FORMAT_AC3
) {
426 device
.str
= "iec958";
427 mp_msg(MSGT_AO
,MSGL_V
,"alsa-spdif-init: playing AC3, %i channels\n", channels
);
430 /* in any case for multichannel playback we should select
436 device
.str
= "default";
437 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: setup for 1/2 channel(s)\n");
440 if (alsa_format
== SND_PCM_FORMAT_FLOAT_LE
)
441 // hack - use the converter plugin
442 device
.str
= "plug:surround40";
444 device
.str
= "surround40";
445 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: device set to surround40\n");
448 if (alsa_format
== SND_PCM_FORMAT_FLOAT_LE
)
449 device
.str
= "plug:surround51";
451 device
.str
= "surround51";
452 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: device set to surround51\n");
455 device
.str
= "default";
456 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ChannelsNotSupported
,channels
);
458 device
.len
= strlen(device
.str
);
459 if (subopt_parse(ao_subdevice
, subopts
) != 0) {
464 parse_device(alsa_device
, device
.str
, device
.len
);
466 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: using device %s\n", alsa_device
);
468 //setting modes for block or nonblock-mode
470 open_mode
= SND_PCM_NONBLOCK
;
476 //sets buff/chunksize if its set manually
477 if (ao_data
.buffersize
) {
478 switch (ao_data
.buffersize
)
483 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 8192\n");
484 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 512\n");
489 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 8192\n");
490 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 1024\n");
495 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 16384\n");
496 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 512\n");
501 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: buffersize set manually to 16384\n");
502 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set manually to 1024\n");
512 //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
513 if ((err
= try_open_device(alsa_device
, open_mode
, format
== AF_FORMAT_AC3
)) < 0)
515 if (err
!= -EBUSY
&& ao_noblock
) {
516 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_OpenInNonblockModeFailed
);
517 if ((err
= try_open_device(alsa_device
, 0, format
== AF_FORMAT_AC3
)) < 0) {
518 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PlaybackOpenError
, snd_strerror(err
));
522 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PlaybackOpenError
, snd_strerror(err
));
527 if ((err
= snd_pcm_nonblock(alsa_handler
, 0)) < 0) {
528 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_ErrorSetBlockMode
, snd_strerror(err
));
530 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: pcm opened in blocking mode\n");
533 snd_pcm_hw_params_alloca(&alsa_hwparams
);
534 snd_pcm_sw_params_alloca(&alsa_swparams
);
536 // setting hw-parameters
537 if ((err
= snd_pcm_hw_params_any(alsa_handler
, alsa_hwparams
)) < 0)
539 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetInitialParameters
,
544 err
= snd_pcm_hw_params_set_access(alsa_handler
, alsa_hwparams
,
545 SND_PCM_ACCESS_RW_INTERLEAVED
);
547 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetAccessType
,
552 /* workaround for nonsupported formats
553 sets default format to S16_LE if the given formats aren't supported */
554 if ((err
= snd_pcm_hw_params_test_format(alsa_handler
, alsa_hwparams
,
557 mp_msg(MSGT_AO
,MSGL_INFO
,
558 MSGTR_AO_ALSA_FormatNotSupportedByHardware
, af_fmt2str_short(format
));
559 alsa_format
= SND_PCM_FORMAT_S16_LE
;
560 ao_data
.format
= AF_FORMAT_S16_LE
;
563 if ((err
= snd_pcm_hw_params_set_format(alsa_handler
, alsa_hwparams
,
566 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetFormat
,
571 if ((err
= snd_pcm_hw_params_set_channels_near(alsa_handler
, alsa_hwparams
,
572 &ao_data
.channels
)) < 0)
574 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetChannels
,
579 /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
580 prefer our own resampler */
581 #if SND_LIB_VERSION >= 0x010009
582 if ((err
= snd_pcm_hw_params_set_rate_resample(alsa_handler
, alsa_hwparams
,
585 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToDisableResampling
,
591 if ((err
= snd_pcm_hw_params_set_rate_near(alsa_handler
, alsa_hwparams
,
592 &ao_data
.samplerate
, NULL
)) < 0)
594 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetSamplerate2
,
599 bytes_per_sample
= snd_pcm_format_physical_width(alsa_format
) / 8;
600 bytes_per_sample
*= ao_data
.channels
;
601 ao_data
.bps
= ao_data
.samplerate
* bytes_per_sample
;
605 int alsa_buffer_time
= 500000; /* original 60 */
606 int alsa_period_time
;
607 alsa_period_time
= alsa_buffer_time
/4;
608 if ((err
= snd_pcm_hw_params_set_buffer_time_near(alsa_handler
, alsa_hwparams
,
609 &alsa_buffer_time
, NULL
)) < 0)
611 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetBufferTimeNear
,
615 alsa_buffer_time
= err
;
617 if ((err
= snd_pcm_hw_params_set_period_time_near(alsa_handler
, alsa_hwparams
,
618 &alsa_period_time
, NULL
)) < 0)
619 /* original: alsa_buffer_time/ao_data.bps */
621 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetPeriodTime
,
625 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_BufferTimePeriodTime
,
626 alsa_buffer_time
, err
);
628 #endif//end SET_BUFFERTIME
633 if ((err
= snd_pcm_hw_params_set_period_size_near(alsa_handler
, alsa_hwparams
,
634 &chunk_size
, NULL
)) < 0)
636 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetPeriodSize
,
637 chunk_size
, snd_strerror(err
));
641 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: chunksize set to %li\n", chunk_size
);
643 if ((err
= snd_pcm_hw_params_set_periods_near(alsa_handler
, alsa_hwparams
,
644 &alsa_fragcount
, NULL
)) < 0) {
645 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetPeriods
,
650 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: fragcount=%i\n", alsa_fragcount
);
653 #endif//end SET_CHUNKSIZE
655 /* finally install hardware parameters */
656 if ((err
= snd_pcm_hw_params(alsa_handler
, alsa_hwparams
)) < 0)
658 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetHwParameters
,
662 // end setting hw-params
665 // gets buffersize for control
666 if ((err
= snd_pcm_hw_params_get_buffer_size(alsa_hwparams
, &bufsize
)) < 0)
668 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetBufferSize
, snd_strerror(err
));
672 ao_data
.buffersize
= bufsize
* bytes_per_sample
;
673 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: got buffersize=%i\n", ao_data
.buffersize
);
676 if ((err
= snd_pcm_hw_params_get_period_size(alsa_hwparams
, &chunk_size
, NULL
)) < 0) {
677 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetPeriodSize
, snd_strerror(err
));
680 mp_msg(MSGT_AO
,MSGL_V
,"alsa-init: got period size %li\n", chunk_size
);
682 ao_data
.outburst
= chunk_size
* bytes_per_sample
;
684 /* setting software parameters */
685 if ((err
= snd_pcm_sw_params_current(alsa_handler
, alsa_swparams
)) < 0) {
686 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetSwParameters
,
690 #if SND_LIB_VERSION >= 0x000901
691 if ((err
= snd_pcm_sw_params_get_boundary(alsa_swparams
, &boundary
)) < 0) {
692 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetBoundary
,
697 boundary
= 0x7fffffff;
699 /* start playing when one period has been written */
700 if ((err
= snd_pcm_sw_params_set_start_threshold(alsa_handler
, alsa_swparams
, chunk_size
)) < 0) {
701 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetStartThreshold
,
705 /* disable underrun reporting */
706 if ((err
= snd_pcm_sw_params_set_stop_threshold(alsa_handler
, alsa_swparams
, boundary
)) < 0) {
707 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetStopThreshold
,
711 #if SND_LIB_VERSION >= 0x000901
712 /* play silence when there is an underrun */
713 if ((err
= snd_pcm_sw_params_set_silence_size(alsa_handler
, alsa_swparams
, boundary
)) < 0) {
714 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToSetSilenceSize
,
719 if ((err
= snd_pcm_sw_params(alsa_handler
, alsa_swparams
)) < 0) {
720 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_UnableToGetSwParameters
,
724 /* end setting sw-params */
726 mp_msg(MSGT_AO
,MSGL_V
,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
727 ao_data
.samplerate
, ao_data
.channels
, (int)bytes_per_sample
, ao_data
.buffersize
,
728 snd_pcm_format_description(alsa_format
));
730 } // end switch alsa_handler (spdif)
731 alsa_can_pause
= snd_pcm_hw_params_can_pause(alsa_hwparams
);
736 /* close audio device */
737 static void uninit(int immed
)
744 snd_pcm_drain(alsa_handler
);
746 if ((err
= snd_pcm_close(alsa_handler
)) < 0)
748 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmCloseError
, snd_strerror(err
));
753 mp_msg(MSGT_AO
,MSGL_V
,"alsa-uninit: pcm closed\n");
757 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_NoHandlerDefined
);
761 static void audio_pause(void)
765 if (alsa_can_pause
) {
766 if ((err
= snd_pcm_pause(alsa_handler
, 1)) < 0)
768 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPauseError
, snd_strerror(err
));
771 mp_msg(MSGT_AO
,MSGL_V
,"alsa-pause: pause supported by hardware\n");
773 if (snd_pcm_delay(alsa_handler
, &prepause_frames
) < 0
774 || prepause_frames
< 0)
777 if ((err
= snd_pcm_drop(alsa_handler
)) < 0)
779 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmDropError
, snd_strerror(err
));
785 static void audio_resume(void)
789 if (snd_pcm_state(alsa_handler
) == SND_PCM_STATE_SUSPENDED
) {
790 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume
);
791 while ((err
= snd_pcm_resume(alsa_handler
)) == -EAGAIN
) sleep(1);
793 if (alsa_can_pause
) {
794 if ((err
= snd_pcm_pause(alsa_handler
, 0)) < 0)
796 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmResumeError
, snd_strerror(err
));
799 mp_msg(MSGT_AO
,MSGL_V
,"alsa-resume: resume supported by hardware\n");
801 if ((err
= snd_pcm_prepare(alsa_handler
)) < 0)
803 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(err
));
806 if (prepause_frames
) {
807 void *silence
= calloc(prepause_frames
, bytes_per_sample
);
808 play(silence
, prepause_frames
* bytes_per_sample
, 0);
814 /* stop playing and empty buffers (for seeking/pause) */
815 static void reset(void)
820 if ((err
= snd_pcm_drop(alsa_handler
)) < 0)
822 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(err
));
825 if ((err
= snd_pcm_prepare(alsa_handler
)) < 0)
827 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(err
));
834 plays 'len' bytes of 'data'
835 returns: number of bytes played
836 modified last at 29.06.02 by jp
837 thanxs for marius <marius@rospot.com> for giving us the light ;)
840 static int play(void* data
, int len
, int flags
)
842 int num_frames
= len
/ bytes_per_sample
;
843 snd_pcm_sframes_t res
= 0;
845 //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
848 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_DeviceConfigurationError
);
856 res
= snd_pcm_writei(alsa_handler
, data
, num_frames
);
862 else if (res
== -ESTRPIPE
) { /* suspend */
863 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume
);
864 while ((res
= snd_pcm_resume(alsa_handler
)) == -EAGAIN
)
868 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_WriteError
, snd_strerror(res
));
869 mp_msg(MSGT_AO
,MSGL_INFO
,MSGTR_AO_ALSA_TryingToResetSoundcard
);
870 if ((res
= snd_pcm_prepare(alsa_handler
)) < 0) {
871 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_PcmPrepareError
, snd_strerror(res
));
878 return res
< 0 ? res
: res
* bytes_per_sample
;
881 /* how many byes are free in the buffer */
882 static int get_space(void)
884 snd_pcm_status_t
*status
;
887 snd_pcm_status_alloca(&status
);
889 if ((ret
= snd_pcm_status(alsa_handler
, status
)) < 0)
891 mp_msg(MSGT_AO
,MSGL_ERR
,MSGTR_AO_ALSA_CannotGetPcmStatus
, snd_strerror(ret
));
895 unsigned space
= snd_pcm_status_get_avail(status
) * bytes_per_sample
;
896 if (space
> ao_data
.buffersize
) // Buffer underrun?
897 space
= ao_data
.buffersize
;
901 /* delay in seconds between first and last sample in buffer */
902 static float get_delay(void)
905 snd_pcm_sframes_t delay
;
907 if (snd_pcm_delay(alsa_handler
, &delay
) < 0)
911 /* underrun - move the application pointer forward to catch up */
912 #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
913 snd_pcm_forward(alsa_handler
, -delay
);
917 return (float)delay
/ (float)ao_data
.samplerate
;