2 * QEMU ALSA audio driver
4 * Copyright (c) 2005 Vassili Karpov (malc)
6 * Permission is hereby granted, free of charge, to any person obtaining a copy
7 * of this software and associated documentation files (the "Software"), to deal
8 * in the Software without restriction, including without limitation the rights
9 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
10 * copies of the Software, and to permit persons to whom the Software is
11 * furnished to do so, subject to the following conditions:
13 * The above copyright notice and this permission notice shall be included in
14 * all copies or substantial portions of the Software.
16 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
17 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
18 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
19 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
20 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
21 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
24 #include <alsa/asoundlib.h>
25 #include "qemu-common.h"
28 #define AUDIO_CAP "alsa"
29 #include "audio_int.h"
31 typedef struct ALSAVoiceOut
{
37 typedef struct ALSAVoiceIn
{
46 const char *pcm_name_in
;
47 const char *pcm_name_out
;
48 unsigned int buffer_size_in
;
49 unsigned int period_size_in
;
50 unsigned int buffer_size_out
;
51 unsigned int period_size_out
;
52 unsigned int threshold
;
54 int buffer_size_in_overridden
;
55 int period_size_in_overridden
;
57 int buffer_size_out_overridden
;
58 int period_size_out_overridden
;
61 .pcm_name_out
= "default",
62 .pcm_name_in
= "default",
65 struct alsa_params_req
{
70 unsigned int buffer_size
;
71 unsigned int period_size
;
74 struct alsa_params_obt
{
79 snd_pcm_uframes_t samples
;
82 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err
, const char *fmt
, ...)
87 AUD_vlog (AUDIO_CAP
, fmt
, ap
);
90 AUD_log (AUDIO_CAP
, "Reason: %s\n", snd_strerror (err
));
93 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
102 AUD_log (AUDIO_CAP
, "Could not initialize %s\n", typ
);
105 AUD_vlog (AUDIO_CAP
, fmt
, ap
);
108 AUD_log (AUDIO_CAP
, "Reason: %s\n", snd_strerror (err
));
111 static void alsa_anal_close (snd_pcm_t
**handlep
)
113 int err
= snd_pcm_close (*handlep
);
115 alsa_logerr (err
, "Failed to close PCM handle %p\n", *handlep
);
120 static int alsa_write (SWVoiceOut
*sw
, void *buf
, int len
)
122 return audio_pcm_sw_write (sw
, buf
, len
);
125 static snd_pcm_format_t
aud_to_alsafmt (audfmt_e fmt
)
129 return SND_PCM_FORMAT_S8
;
132 return SND_PCM_FORMAT_U8
;
135 return SND_PCM_FORMAT_S16_LE
;
138 return SND_PCM_FORMAT_U16_LE
;
141 return SND_PCM_FORMAT_S32_LE
;
144 return SND_PCM_FORMAT_U32_LE
;
147 dolog ("Internal logic error: Bad audio format %d\n", fmt
);
151 return SND_PCM_FORMAT_U8
;
155 static int alsa_to_audfmt (snd_pcm_format_t alsafmt
, audfmt_e
*fmt
,
159 case SND_PCM_FORMAT_S8
:
164 case SND_PCM_FORMAT_U8
:
169 case SND_PCM_FORMAT_S16_LE
:
174 case SND_PCM_FORMAT_U16_LE
:
179 case SND_PCM_FORMAT_S16_BE
:
184 case SND_PCM_FORMAT_U16_BE
:
189 case SND_PCM_FORMAT_S32_LE
:
194 case SND_PCM_FORMAT_U32_LE
:
199 case SND_PCM_FORMAT_S32_BE
:
204 case SND_PCM_FORMAT_U32_BE
:
210 dolog ("Unrecognized audio format %d\n", alsafmt
);
217 static void alsa_dump_info (struct alsa_params_req
*req
,
218 struct alsa_params_obt
*obt
)
220 dolog ("parameter | requested value | obtained value\n");
221 dolog ("format | %10d | %10d\n", req
->fmt
, obt
->fmt
);
222 dolog ("channels | %10d | %10d\n",
223 req
->nchannels
, obt
->nchannels
);
224 dolog ("frequency | %10d | %10d\n", req
->freq
, obt
->freq
);
225 dolog ("============================================\n");
226 dolog ("requested: buffer size %d period size %d\n",
227 req
->buffer_size
, req
->period_size
);
228 dolog ("obtained: samples %ld\n", obt
->samples
);
231 static void alsa_set_threshold (snd_pcm_t
*handle
, snd_pcm_uframes_t threshold
)
234 snd_pcm_sw_params_t
*sw_params
;
236 snd_pcm_sw_params_alloca (&sw_params
);
238 err
= snd_pcm_sw_params_current (handle
, sw_params
);
240 dolog ("Could not fully initialize DAC\n");
241 alsa_logerr (err
, "Failed to get current software parameters\n");
245 err
= snd_pcm_sw_params_set_start_threshold (handle
, sw_params
, threshold
);
247 dolog ("Could not fully initialize DAC\n");
248 alsa_logerr (err
, "Failed to set software threshold to %ld\n",
253 err
= snd_pcm_sw_params (handle
, sw_params
);
255 dolog ("Could not fully initialize DAC\n");
256 alsa_logerr (err
, "Failed to set software parameters\n");
261 static int alsa_open (int in
, struct alsa_params_req
*req
,
262 struct alsa_params_obt
*obt
, snd_pcm_t
**handlep
)
265 snd_pcm_hw_params_t
*hw_params
;
268 unsigned int freq
, nchannels
;
269 const char *pcm_name
= in
? conf
.pcm_name_in
: conf
.pcm_name_out
;
270 snd_pcm_uframes_t obt_buffer_size
;
271 const char *typ
= in
? "ADC" : "DAC";
272 snd_pcm_format_t obtfmt
;
275 nchannels
= req
->nchannels
;
276 size_in_usec
= req
->size_in_usec
;
278 snd_pcm_hw_params_alloca (&hw_params
);
283 in
? SND_PCM_STREAM_CAPTURE
: SND_PCM_STREAM_PLAYBACK
,
287 alsa_logerr2 (err
, typ
, "Failed to open `%s':\n", pcm_name
);
291 err
= snd_pcm_hw_params_any (handle
, hw_params
);
293 alsa_logerr2 (err
, typ
, "Failed to initialize hardware parameters\n");
297 err
= snd_pcm_hw_params_set_access (
300 SND_PCM_ACCESS_RW_INTERLEAVED
303 alsa_logerr2 (err
, typ
, "Failed to set access type\n");
307 err
= snd_pcm_hw_params_set_format (handle
, hw_params
, req
->fmt
);
308 if (err
< 0 && conf
.verbose
) {
309 alsa_logerr2 (err
, typ
, "Failed to set format %d\n", req
->fmt
);
312 err
= snd_pcm_hw_params_set_rate_near (handle
, hw_params
, &freq
, 0);
314 alsa_logerr2 (err
, typ
, "Failed to set frequency %d\n", req
->freq
);
318 err
= snd_pcm_hw_params_set_channels_near (
324 alsa_logerr2 (err
, typ
, "Failed to set number of channels %d\n",
329 if (nchannels
!= 1 && nchannels
!= 2) {
330 alsa_logerr2 (err
, typ
,
331 "Can not handle obtained number of channels %d\n",
336 if (req
->buffer_size
) {
341 unsigned int btime
= req
->buffer_size
;
343 err
= snd_pcm_hw_params_set_buffer_time_near (
352 snd_pcm_uframes_t bsize
= req
->buffer_size
;
354 err
= snd_pcm_hw_params_set_buffer_size_near (
362 alsa_logerr2 (err
, typ
, "Failed to set buffer %s to %d\n",
363 size_in_usec
? "time" : "size", req
->buffer_size
);
367 if (obt
- req
->buffer_size
)
368 dolog ("Requested buffer %s %u was rejected, using %lu\n",
369 size_in_usec
? "time" : "size", req
->buffer_size
, obt
);
372 if (req
->period_size
) {
377 unsigned int ptime
= req
->period_size
;
379 err
= snd_pcm_hw_params_set_period_time_near (
388 snd_pcm_uframes_t psize
= req
->period_size
;
390 err
= snd_pcm_hw_params_set_buffer_size_near (
399 alsa_logerr2 (err
, typ
, "Failed to set period %s to %d\n",
400 size_in_usec
? "time" : "size", req
->period_size
);
404 if (obt
- req
->period_size
)
405 dolog ("Requested period %s %u was rejected, using %lu\n",
406 size_in_usec
? "time" : "size", req
->period_size
, obt
);
409 err
= snd_pcm_hw_params (handle
, hw_params
);
411 alsa_logerr2 (err
, typ
, "Failed to apply audio parameters\n");
415 err
= snd_pcm_hw_params_get_buffer_size (hw_params
, &obt_buffer_size
);
417 alsa_logerr2 (err
, typ
, "Failed to get buffer size\n");
421 err
= snd_pcm_hw_params_get_format (hw_params
, &obtfmt
);
423 alsa_logerr2 (err
, typ
, "Failed to get format\n");
427 if (alsa_to_audfmt (obtfmt
, &obt
->fmt
, &obt
->endianness
)) {
428 dolog ("Invalid format was returned %d\n", obtfmt
);
432 err
= snd_pcm_prepare (handle
);
434 alsa_logerr2 (err
, typ
, "Could not prepare handle %p\n", handle
);
438 if (!in
&& conf
.threshold
) {
439 snd_pcm_uframes_t threshold
;
442 bytes_per_sec
= freq
<< (nchannels
== 2);
460 threshold
= (conf
.threshold
* bytes_per_sec
) / 1000;
461 alsa_set_threshold (handle
, threshold
);
464 obt
->nchannels
= nchannels
;
466 obt
->samples
= obt_buffer_size
;
471 (obt
->fmt
!= req
->fmt
||
472 obt
->nchannels
!= req
->nchannels
||
473 obt
->freq
!= req
->freq
)) {
474 dolog ("Audio paramters for %s\n", typ
);
475 alsa_dump_info (req
, obt
);
479 alsa_dump_info (req
, obt
);
484 alsa_anal_close (&handle
);
488 static int alsa_recover (snd_pcm_t
*handle
)
490 int err
= snd_pcm_prepare (handle
);
492 alsa_logerr (err
, "Failed to prepare handle %p\n", handle
);
498 static snd_pcm_sframes_t
alsa_get_avail (snd_pcm_t
*handle
)
500 snd_pcm_sframes_t avail
;
502 avail
= snd_pcm_avail_update (handle
);
504 if (avail
== -EPIPE
) {
505 if (!alsa_recover (handle
)) {
506 avail
= snd_pcm_avail_update (handle
);
512 "Could not obtain number of available frames\n");
520 static int alsa_run_out (HWVoiceOut
*hw
)
522 ALSAVoiceOut
*alsa
= (ALSAVoiceOut
*) hw
;
523 int rpos
, live
, decr
;
527 snd_pcm_sframes_t avail
;
529 live
= audio_pcm_hw_get_live_out (hw
);
534 avail
= alsa_get_avail (alsa
->handle
);
536 dolog ("Could not get number of available playback frames\n");
540 decr
= audio_MIN (live
, avail
);
544 int left_till_end_samples
= hw
->samples
- rpos
;
545 int len
= audio_MIN (samples
, left_till_end_samples
);
546 snd_pcm_sframes_t written
;
548 src
= hw
->mix_buf
+ rpos
;
549 dst
= advance (alsa
->pcm_buf
, rpos
<< hw
->info
.shift
);
551 hw
->clip (dst
, src
, len
);
554 written
= snd_pcm_writei (alsa
->handle
, dst
, len
);
560 dolog ("Failed to write %d frames (wrote zero)\n", len
);
565 if (alsa_recover (alsa
->handle
)) {
566 alsa_logerr (written
, "Failed to write %d frames\n",
571 dolog ("Recovering from playback xrun\n");
579 alsa_logerr (written
, "Failed to write %d frames to %p\n",
585 rpos
= (rpos
+ written
) % hw
->samples
;
588 dst
= advance (dst
, written
<< hw
->info
.shift
);
598 static void alsa_fini_out (HWVoiceOut
*hw
)
600 ALSAVoiceOut
*alsa
= (ALSAVoiceOut
*) hw
;
602 ldebug ("alsa_fini\n");
603 alsa_anal_close (&alsa
->handle
);
606 qemu_free (alsa
->pcm_buf
);
607 alsa
->pcm_buf
= NULL
;
611 static int alsa_init_out (HWVoiceOut
*hw
, audsettings_t
*as
)
613 ALSAVoiceOut
*alsa
= (ALSAVoiceOut
*) hw
;
614 struct alsa_params_req req
;
615 struct alsa_params_obt obt
;
617 audsettings_t obt_as
;
619 req
.fmt
= aud_to_alsafmt (as
->fmt
);
621 req
.nchannels
= as
->nchannels
;
622 req
.period_size
= conf
.period_size_out
;
623 req
.buffer_size
= conf
.buffer_size_out
;
624 req
.size_in_usec
= conf
.size_in_usec_in
;
626 if (alsa_open (0, &req
, &obt
, &handle
)) {
630 obt_as
.freq
= obt
.freq
;
631 obt_as
.nchannels
= obt
.nchannels
;
632 obt_as
.fmt
= obt
.fmt
;
633 obt_as
.endianness
= obt
.endianness
;
635 audio_pcm_init_info (&hw
->info
, &obt_as
);
636 hw
->samples
= obt
.samples
;
638 alsa
->pcm_buf
= audio_calloc (AUDIO_FUNC
, obt
.samples
, 1 << hw
->info
.shift
);
639 if (!alsa
->pcm_buf
) {
640 dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
641 hw
->samples
, 1 << hw
->info
.shift
);
642 alsa_anal_close (&handle
);
646 alsa
->handle
= handle
;
650 static int alsa_voice_ctl (snd_pcm_t
*handle
, const char *typ
, int pause
)
655 err
= snd_pcm_drop (handle
);
657 alsa_logerr (err
, "Could not stop %s\n", typ
);
662 err
= snd_pcm_prepare (handle
);
664 alsa_logerr (err
, "Could not prepare handle for %s\n", typ
);
672 static int alsa_ctl_out (HWVoiceOut
*hw
, int cmd
, ...)
674 ALSAVoiceOut
*alsa
= (ALSAVoiceOut
*) hw
;
678 ldebug ("enabling voice\n");
679 return alsa_voice_ctl (alsa
->handle
, "playback", 0);
682 ldebug ("disabling voice\n");
683 return alsa_voice_ctl (alsa
->handle
, "playback", 1);
689 static int alsa_init_in (HWVoiceIn
*hw
, audsettings_t
*as
)
691 ALSAVoiceIn
*alsa
= (ALSAVoiceIn
*) hw
;
692 struct alsa_params_req req
;
693 struct alsa_params_obt obt
;
695 audsettings_t obt_as
;
697 req
.fmt
= aud_to_alsafmt (as
->fmt
);
699 req
.nchannels
= as
->nchannels
;
700 req
.period_size
= conf
.period_size_in
;
701 req
.buffer_size
= conf
.buffer_size_in
;
702 req
.size_in_usec
= conf
.size_in_usec_in
;
704 if (alsa_open (1, &req
, &obt
, &handle
)) {
708 obt_as
.freq
= obt
.freq
;
709 obt_as
.nchannels
= obt
.nchannels
;
710 obt_as
.fmt
= obt
.fmt
;
711 obt_as
.endianness
= obt
.endianness
;
713 audio_pcm_init_info (&hw
->info
, &obt_as
);
714 hw
->samples
= obt
.samples
;
716 alsa
->pcm_buf
= audio_calloc (AUDIO_FUNC
, hw
->samples
, 1 << hw
->info
.shift
);
717 if (!alsa
->pcm_buf
) {
718 dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
719 hw
->samples
, 1 << hw
->info
.shift
);
720 alsa_anal_close (&handle
);
724 alsa
->handle
= handle
;
728 static void alsa_fini_in (HWVoiceIn
*hw
)
730 ALSAVoiceIn
*alsa
= (ALSAVoiceIn
*) hw
;
732 alsa_anal_close (&alsa
->handle
);
735 qemu_free (alsa
->pcm_buf
);
736 alsa
->pcm_buf
= NULL
;
740 static int alsa_run_in (HWVoiceIn
*hw
)
742 ALSAVoiceIn
*alsa
= (ALSAVoiceIn
*) hw
;
743 int hwshift
= hw
->info
.shift
;
745 int live
= audio_pcm_hw_get_live_in (hw
);
746 int dead
= hw
->samples
- live
;
755 snd_pcm_sframes_t avail
;
756 snd_pcm_uframes_t read_samples
= 0;
762 avail
= alsa_get_avail (alsa
->handle
);
764 dolog ("Could not get number of captured frames\n");
768 if (!avail
&& (snd_pcm_state (alsa
->handle
) == SND_PCM_STATE_PREPARED
)) {
772 decr
= audio_MIN (dead
, avail
);
777 if (hw
->wpos
+ decr
> hw
->samples
) {
778 bufs
[0].len
= (hw
->samples
- hw
->wpos
);
779 bufs
[1].len
= (decr
- (hw
->samples
- hw
->wpos
));
785 for (i
= 0; i
< 2; ++i
) {
788 snd_pcm_sframes_t nread
;
789 snd_pcm_uframes_t len
;
793 src
= advance (alsa
->pcm_buf
, bufs
[i
].add
<< hwshift
);
794 dst
= hw
->conv_buf
+ bufs
[i
].add
;
797 nread
= snd_pcm_readi (alsa
->handle
, src
, len
);
803 dolog ("Failed to read %ld frames (read zero)\n", len
);
808 if (alsa_recover (alsa
->handle
)) {
809 alsa_logerr (nread
, "Failed to read %ld frames\n", len
);
813 dolog ("Recovering from capture xrun\n");
823 "Failed to read %ld frames from %p\n",
831 hw
->conv (dst
, src
, nread
, &nominal_volume
);
833 src
= advance (src
, nread
<< hwshift
);
836 read_samples
+= nread
;
842 hw
->wpos
= (hw
->wpos
+ read_samples
) % hw
->samples
;
846 static int alsa_read (SWVoiceIn
*sw
, void *buf
, int size
)
848 return audio_pcm_sw_read (sw
, buf
, size
);
851 static int alsa_ctl_in (HWVoiceIn
*hw
, int cmd
, ...)
853 ALSAVoiceIn
*alsa
= (ALSAVoiceIn
*) hw
;
857 ldebug ("enabling voice\n");
858 return alsa_voice_ctl (alsa
->handle
, "capture", 0);
861 ldebug ("disabling voice\n");
862 return alsa_voice_ctl (alsa
->handle
, "capture", 1);
868 static void *alsa_audio_init (void)
873 static void alsa_audio_fini (void *opaque
)
878 static struct audio_option alsa_options
[] = {
879 {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL
, &conf
.size_in_usec_out
,
880 "DAC period/buffer size in microseconds (otherwise in frames)", NULL
, 0},
881 {"DAC_PERIOD_SIZE", AUD_OPT_INT
, &conf
.period_size_out
,
882 "DAC period size (0 to go with system default)",
883 &conf
.period_size_out_overridden
, 0},
884 {"DAC_BUFFER_SIZE", AUD_OPT_INT
, &conf
.buffer_size_out
,
885 "DAC buffer size (0 to go with system default)",
886 &conf
.buffer_size_out_overridden
, 0},
888 {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL
, &conf
.size_in_usec_in
,
889 "ADC period/buffer size in microseconds (otherwise in frames)", NULL
, 0},
890 {"ADC_PERIOD_SIZE", AUD_OPT_INT
, &conf
.period_size_in
,
891 "ADC period size (0 to go with system default)",
892 &conf
.period_size_in_overridden
, 0},
893 {"ADC_BUFFER_SIZE", AUD_OPT_INT
, &conf
.buffer_size_in
,
894 "ADC buffer size (0 to go with system default)",
895 &conf
.buffer_size_in_overridden
, 0},
897 {"THRESHOLD", AUD_OPT_INT
, &conf
.threshold
,
898 "(undocumented)", NULL
, 0},
900 {"DAC_DEV", AUD_OPT_STR
, &conf
.pcm_name_out
,
901 "DAC device name (for instance dmix)", NULL
, 0},
903 {"ADC_DEV", AUD_OPT_STR
, &conf
.pcm_name_in
,
904 "ADC device name", NULL
, 0},
906 {"VERBOSE", AUD_OPT_BOOL
, &conf
.verbose
,
907 "Behave in a more verbose way", NULL
, 0},
909 {NULL
, 0, NULL
, NULL
, NULL
, 0}
912 static struct audio_pcm_ops alsa_pcm_ops
= {
926 struct audio_driver alsa_audio_driver
= {
927 INIT_FIELD (name
= ) "alsa",
928 INIT_FIELD (descr
= ) "ALSA http://www.alsa-project.org",
929 INIT_FIELD (options
= ) alsa_options
,
930 INIT_FIELD (init
= ) alsa_audio_init
,
931 INIT_FIELD (fini
= ) alsa_audio_fini
,
932 INIT_FIELD (pcm_ops
= ) &alsa_pcm_ops
,
933 INIT_FIELD (can_be_default
= ) 1,
934 INIT_FIELD (max_voices_out
= ) INT_MAX
,
935 INIT_FIELD (max_voices_in
= ) INT_MAX
,
936 INIT_FIELD (voice_size_out
= ) sizeof (ALSAVoiceOut
),
937 INIT_FIELD (voice_size_in
= ) sizeof (ALSAVoiceIn
)