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 #define DEFAULT_BUFFER_SIZE 1024
62 #define DEFAULT_PERIOD_SIZE 256
65 .size_in_usec_out = 1,
67 .pcm_name_out = "default",
68 .pcm_name_in = "default",
70 .buffer_size_in = 400000,
71 .period_size_in = 400000 / 4,
72 .buffer_size_out = 400000,
73 .period_size_out = 400000 / 4,
75 .buffer_size_in = DEFAULT_BUFFER_SIZE * 4,
76 .period_size_in = DEFAULT_PERIOD_SIZE * 4,
77 .buffer_size_out = DEFAULT_BUFFER_SIZE,
78 .period_size_out = DEFAULT_PERIOD_SIZE,
79 .buffer_size_in_overridden = 0,
80 .buffer_size_out_overridden = 0,
81 .period_size_in_overridden = 0,
82 .period_size_out_overridden = 0,
88 struct alsa_params_req {
92 unsigned int buffer_size;
93 unsigned int period_size;
96 struct alsa_params_obt {
100 snd_pcm_uframes_t samples;
103 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
108 AUD_vlog (AUDIO_CAP, fmt, ap);
111 AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
114 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
123 AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
126 AUD_vlog (AUDIO_CAP, fmt, ap);
129 AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
132 static void alsa_anal_close (snd_pcm_t **handlep)
134 int err = snd_pcm_close (*handlep);
136 alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
141 static int alsa_write (SWVoiceOut *sw, void *buf, int len)
143 return audio_pcm_sw_write (sw, buf, len);
146 static int aud_to_alsafmt (audfmt_e fmt)
150 return SND_PCM_FORMAT_S8;
153 return SND_PCM_FORMAT_U8;
156 return SND_PCM_FORMAT_S16_LE;
159 return SND_PCM_FORMAT_U16_LE;
162 return SND_PCM_FORMAT_S32_LE;
165 return SND_PCM_FORMAT_U32_LE;
168 dolog ("Internal logic error: Bad audio format %d\n", fmt);
172 return SND_PCM_FORMAT_U8;
176 static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness)
179 case SND_PCM_FORMAT_S8:
184 case SND_PCM_FORMAT_U8:
189 case SND_PCM_FORMAT_S16_LE:
194 case SND_PCM_FORMAT_U16_LE:
199 case SND_PCM_FORMAT_S16_BE:
204 case SND_PCM_FORMAT_U16_BE:
209 case SND_PCM_FORMAT_S32_LE:
214 case SND_PCM_FORMAT_U32_LE:
219 case SND_PCM_FORMAT_S32_BE:
224 case SND_PCM_FORMAT_U32_BE:
230 dolog ("Unrecognized audio format %d\n", alsafmt);
237 #if defined DEBUG_MISMATCHES || defined DEBUG
238 static void alsa_dump_info (struct alsa_params_req *req,
239 struct alsa_params_obt *obt)
241 dolog ("parameter | requested value | obtained value\n");
242 dolog ("format | %10d | %10d\n", req->fmt, obt->fmt);
243 dolog ("channels | %10d | %10d\n",
244 req->nchannels, obt->nchannels);
245 dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
246 dolog ("============================================\n");
247 dolog ("requested: buffer size %d period size %d\n",
248 req->buffer_size, req->period_size);
249 dolog ("obtained: samples %ld\n", obt->samples);
253 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
256 snd_pcm_sw_params_t *sw_params;
258 snd_pcm_sw_params_alloca (&sw_params);
260 err = snd_pcm_sw_params_current (handle, sw_params);
262 dolog ("Could not fully initialize DAC\n");
263 alsa_logerr (err, "Failed to get current software parameters\n");
267 err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
269 dolog ("Could not fully initialize DAC\n");
270 alsa_logerr (err, "Failed to set software threshold to %ld\n",
275 err = snd_pcm_sw_params (handle, sw_params);
277 dolog ("Could not fully initialize DAC\n");
278 alsa_logerr (err, "Failed to set software parameters\n");
283 static int alsa_open (int in, struct alsa_params_req *req,
284 struct alsa_params_obt *obt, snd_pcm_t **handlep)
287 snd_pcm_hw_params_t *hw_params;
288 int err, freq, nchannels;
289 const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
290 unsigned int period_size, buffer_size;
291 snd_pcm_uframes_t obt_buffer_size;
292 const char *typ = in ? "ADC" : "DAC";
295 period_size = req->period_size;
296 buffer_size = req->buffer_size;
297 nchannels = req->nchannels;
299 snd_pcm_hw_params_alloca (&hw_params);
304 in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
308 alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
312 err = snd_pcm_hw_params_any (handle, hw_params);
314 alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
318 err = snd_pcm_hw_params_set_access (
321 SND_PCM_ACCESS_RW_INTERLEAVED
324 alsa_logerr2 (err, typ, "Failed to set access type\n");
328 err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
330 alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
334 err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
336 alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
340 err = snd_pcm_hw_params_set_channels_near (
346 alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
351 if (nchannels != 1 && nchannels != 2) {
352 alsa_logerr2 (err, typ,
353 "Can not handle obtained number of channels %d\n",
358 if (!((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out))) {
360 buffer_size = DEFAULT_BUFFER_SIZE;
361 period_size= DEFAULT_PERIOD_SIZE;
366 if ((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out)) {
368 err = snd_pcm_hw_params_set_period_time_near (
375 alsa_logerr2 (err, typ,
376 "Failed to set period time %d\n",
382 err = snd_pcm_hw_params_set_buffer_time_near (
390 alsa_logerr2 (err, typ,
391 "Failed to set buffer time %d\n",
398 snd_pcm_uframes_t minval;
401 minval = period_size;
404 err = snd_pcm_hw_params_get_period_size_min (
412 "Could not get minmal period size for %s\n",
417 if (period_size < minval) {
418 if ((in && conf.period_size_in_overridden)
419 || (!in && conf.period_size_out_overridden)) {
420 dolog ("%s period size(%d) is less "
421 "than minmal period size(%ld)\n",
426 period_size = minval;
430 err = snd_pcm_hw_params_set_period_size (
437 alsa_logerr2 (err, typ, "Failed to set period size %d\n",
443 minval = buffer_size;
444 err = snd_pcm_hw_params_get_buffer_size_min (
449 alsa_logerr (err, "Could not get minmal buffer size for %s\n",
453 if (buffer_size < minval) {
454 if ((in && conf.buffer_size_in_overridden)
455 || (!in && conf.buffer_size_out_overridden)) {
457 "%s buffer size(%d) is less "
458 "than minimal buffer size(%ld)\n",
464 buffer_size = minval;
468 err = snd_pcm_hw_params_set_buffer_size (
474 alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
481 dolog ("warning: Buffer size is not set\n");
484 err = snd_pcm_hw_params (handle, hw_params);
486 alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
490 err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
492 alsa_logerr2 (err, typ, "Failed to get buffer size\n");
496 err = snd_pcm_prepare (handle);
498 alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
502 if (!in && conf.threshold) {
503 snd_pcm_uframes_t threshold;
508 << (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16);
510 threshold = (conf.threshold * bytes_per_sec) / 1000;
511 alsa_set_threshold (handle, threshold);
515 obt->nchannels = nchannels;
517 obt->samples = obt_buffer_size;
520 #if defined DEBUG_MISMATCHES || defined DEBUG
521 if (obt->fmt != req->fmt ||
522 obt->nchannels != req->nchannels ||
523 obt->freq != req->freq) {
524 dolog ("Audio paramters mismatch for %s\n", typ);
525 alsa_dump_info (req, obt);
530 alsa_dump_info (req, obt);
535 alsa_anal_close (&handle);
539 static int alsa_recover (snd_pcm_t *handle)
541 int err = snd_pcm_prepare (handle);
543 alsa_logerr (err, "Failed to prepare handle %p\n", handle);
549 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
551 snd_pcm_sframes_t avail;
553 avail = snd_pcm_avail_update (handle);
555 if (avail == -EPIPE) {
556 if (!alsa_recover (handle)) {
557 avail = snd_pcm_avail_update (handle);
563 "Could not obtain number of available frames\n");
571 static int alsa_run_out (HWVoiceOut *hw)
573 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
574 int rpos, live, decr;
578 snd_pcm_sframes_t avail;
580 live = audio_pcm_hw_get_live_out (hw);
585 avail = alsa_get_avail (alsa->handle);
587 dolog ("Could not get number of available playback frames\n");
591 decr = audio_MIN (live, avail);
595 int left_till_end_samples = hw->samples - rpos;
596 int len = audio_MIN (samples, left_till_end_samples);
597 snd_pcm_sframes_t written;
599 src = hw->mix_buf + rpos;
600 dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
602 hw->clip (dst, src, len);
605 written = snd_pcm_writei (alsa->handle, dst, len);
611 dolog ("Failed to write %d frames (wrote zero)\n", len);
616 if (alsa_recover (alsa->handle)) {
617 alsa_logerr (written, "Failed to write %d frames\n",
622 dolog ("Recovering from playback xrun\n");
630 alsa_logerr (written, "Failed to write %d frames to %p\n",
636 rpos = (rpos + written) % hw->samples;
639 dst = advance (dst, written << hw->info.shift);
649 static void alsa_fini_out (HWVoiceOut *hw)
651 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
653 ldebug ("alsa_fini\n");
654 alsa_anal_close (&alsa->handle);
657 qemu_free (alsa->pcm_buf);
658 alsa->pcm_buf = NULL;
662 static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
664 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
665 struct alsa_params_req req;
666 struct alsa_params_obt obt;
667 audfmt_e effective_fmt;
671 audsettings_t obt_as;
673 req.fmt = aud_to_alsafmt (as->fmt);
675 req.nchannels = as->nchannels;
676 req.period_size = conf.period_size_out;
677 req.buffer_size = conf.buffer_size_out;
679 if (alsa_open (0, &req, &obt, &handle)) {
683 err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
685 alsa_anal_close (&handle);
689 obt_as.freq = obt.freq;
690 obt_as.nchannels = obt.nchannels;
691 obt_as.fmt = effective_fmt;
692 obt_as.endianness = endianness;
694 audio_pcm_init_info (&hw->info, &obt_as);
695 hw->samples = obt.samples;
697 alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
698 if (!alsa->pcm_buf) {
699 dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
700 hw->samples, 1 << hw->info.shift);
701 alsa_anal_close (&handle);
705 alsa->handle = handle;
709 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
714 err = snd_pcm_drop (handle);
716 alsa_logerr (err, "Could not stop %s\n", typ);
721 err = snd_pcm_prepare (handle);
723 alsa_logerr (err, "Could not prepare handle for %s\n", typ);
731 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
733 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
737 ldebug ("enabling voice\n");
738 return alsa_voice_ctl (alsa->handle, "playback", 0);
741 ldebug ("disabling voice\n");
742 return alsa_voice_ctl (alsa->handle, "playback", 1);
748 static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
750 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
751 struct alsa_params_req req;
752 struct alsa_params_obt obt;
755 audfmt_e effective_fmt;
757 audsettings_t obt_as;
759 req.fmt = aud_to_alsafmt (as->fmt);
761 req.nchannels = as->nchannels;
762 req.period_size = conf.period_size_in;
763 req.buffer_size = conf.buffer_size_in;
765 if (alsa_open (1, &req, &obt, &handle)) {
769 err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
771 alsa_anal_close (&handle);
775 obt_as.freq = obt.freq;
776 obt_as.nchannels = obt.nchannels;
777 obt_as.fmt = effective_fmt;
778 obt_as.endianness = endianness;
780 audio_pcm_init_info (&hw->info, &obt_as);
781 hw->samples = obt.samples;
783 alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
784 if (!alsa->pcm_buf) {
785 dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
786 hw->samples, 1 << hw->info.shift);
787 alsa_anal_close (&handle);
791 alsa->handle = handle;
795 static void alsa_fini_in (HWVoiceIn *hw)
797 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
799 alsa_anal_close (&alsa->handle);
802 qemu_free (alsa->pcm_buf);
803 alsa->pcm_buf = NULL;
807 static int alsa_run_in (HWVoiceIn *hw)
809 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
810 int hwshift = hw->info.shift;
812 int live = audio_pcm_hw_get_live_in (hw);
813 int dead = hw->samples - live;
822 snd_pcm_sframes_t avail;
823 snd_pcm_uframes_t read_samples = 0;
829 avail = alsa_get_avail (alsa->handle);
831 dolog ("Could not get number of captured frames\n");
835 if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
839 decr = audio_MIN (dead, avail);
844 if (hw->wpos + decr > hw->samples) {
845 bufs[0].len = (hw->samples - hw->wpos);
846 bufs[1].len = (decr - (hw->samples - hw->wpos));
852 for (i = 0; i < 2; ++i) {
855 snd_pcm_sframes_t nread;
856 snd_pcm_uframes_t len;
860 src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
861 dst = hw->conv_buf + bufs[i].add;
864 nread = snd_pcm_readi (alsa->handle, src, len);
870 dolog ("Failed to read %ld frames (read zero)\n", len);
875 if (alsa_recover (alsa->handle)) {
876 alsa_logerr (nread, "Failed to read %ld frames\n", len);
880 dolog ("Recovering from capture xrun\n");
890 "Failed to read %ld frames from %p\n",
898 hw->conv (dst, src, nread, &nominal_volume);
900 src = advance (src, nread << hwshift);
903 read_samples += nread;
909 hw->wpos = (hw->wpos + read_samples) % hw->samples;
913 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
915 return audio_pcm_sw_read (sw, buf, size);
918 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
920 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
924 ldebug ("enabling voice\n");
925 return alsa_voice_ctl (alsa->handle, "capture", 0);
928 ldebug ("disabling voice\n");
929 return alsa_voice_ctl (alsa->handle, "capture", 1);
935 static void *alsa_audio_init (void)
940 static void alsa_audio_fini (void *opaque)
945 static struct audio_option alsa_options[] = {
946 {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
947 "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
948 {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
949 "DAC period size", &conf.period_size_out_overridden, 0},
950 {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
951 "DAC buffer size", &conf.buffer_size_out_overridden, 0},
953 {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
954 "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
955 {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
956 "ADC period size", &conf.period_size_in_overridden, 0},
957 {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
958 "ADC buffer size", &conf.buffer_size_in_overridden, 0},
960 {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
961 "(undocumented)", NULL, 0},
963 {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
964 "DAC device name (for instance dmix)", NULL, 0},
966 {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
967 "ADC device name", NULL, 0},
969 {"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
970 "Behave in a more verbose way", NULL, 0},
972 {NULL, 0, NULL, NULL, NULL, 0}
975 static struct audio_pcm_ops alsa_pcm_ops = {
989 struct audio_driver alsa_audio_driver = {
990 INIT_FIELD (name = ) "alsa",
991 INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
992 INIT_FIELD (options = ) alsa_options,
993 INIT_FIELD (init = ) alsa_audio_init,
994 INIT_FIELD (fini = ) alsa_audio_fini,
995 INIT_FIELD (pcm_ops = ) &alsa_pcm_ops,
996 INIT_FIELD (can_be_default = ) 1,
997 INIT_FIELD (max_voices_out = ) INT_MAX,
998 INIT_FIELD (max_voices_in = ) INT_MAX,
999 INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
1000 INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)