alsaaudio.c@ 1

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1	/*
2	* QEMU ALSA audio driver
3	*
4	* Copyright (c) 2005 Vassili Karpov (malc)
5	*
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:
12	*
13	* The above copyright notice and this permission notice shall be included in
14	* all copies or substantial portions of the Software.
15	*
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
22	* THE SOFTWARE.
23	*/
24	#include <alsa/asoundlib.h>
25
26	#include "Builtins.h"
27	#include "../../vl_vbox.h"
28	#include "audio.h"
29	#include <iprt/alloc.h>
30
31	#define AUDIO_CAP "alsa"
32	#include "audio_int.h"
33
34	typedef struct ALSAVoiceOut {
35	HWVoiceOut hw;
36	void *pcm_buf;
37	snd_pcm_t *handle;
38	} ALSAVoiceOut;
39
40	typedef struct ALSAVoiceIn {
41	HWVoiceIn hw;
42	snd_pcm_t *handle;
43	void *pcm_buf;
44	} ALSAVoiceIn;
45
46	static struct {
47	int size_in_usec_in;
48	int size_in_usec_out;
49	const char *pcm_name_in;
50	const char *pcm_name_out;
51	unsigned int buffer_size_in;
52	unsigned int period_size_in;
53	unsigned int buffer_size_out;
54	unsigned int period_size_out;
55	unsigned int threshold;
56
57	int buffer_size_in_overriden;
58	int period_size_in_overriden;
59
60	int buffer_size_out_overriden;
61	int period_size_out_overriden;
62	int verbose;
63	} conf = {
64	#ifdef HIGH_LATENCY
65	INIT_FIELD (.size_in_usec_in =) 1,
66	INIT_FIELD (.size_in_usec_out =) 1,
67	#else
68	INIT_FIELD (.size_in_usec_in =) 0,
69	INIT_FIELD (.size_in_usec_out =) 0,
70	#endif
71	INIT_FIELD (.pcm_name_out =) "default",
72	INIT_FIELD (.pcm_name_in =) "default",
73	#ifdef HIGH_LATENCY
74	INIT_FIELD (.buffer_size_in =) 400000,
75	INIT_FIELD (.period_size_in =) 400000 / 4,
76	INIT_FIELD (.buffer_size_out =) 400000,
77	INIT_FIELD (.period_size_out =) 400000 / 4,
78	#else
79	#define DEFAULT_BUFFER_SIZE 1024
80	#define DEFAULT_PERIOD_SIZE 256
81	INIT_FIELD (.buffer_size_in =) DEFAULT_BUFFER_SIZE * 4,
82	INIT_FIELD (.period_size_in =) DEFAULT_PERIOD_SIZE * 4,
83	INIT_FIELD (.buffer_size_out =) DEFAULT_BUFFER_SIZE,
84	INIT_FIELD (.period_size_out =) DEFAULT_PERIOD_SIZE,
85	#endif
86	INIT_FIELD (.threshold =) 0,
87	INIT_FIELD (.buffer_size_in_overriden =) 0,
88	INIT_FIELD (.period_size_in_overriden =) 0,
89	INIT_FIELD (.buffer_size_out_overriden =) 0,
90	INIT_FIELD (.period_size_out_overriden =) 0,
91	INIT_FIELD (.verbose =) 0
92	};
93
94	struct alsa_params_req {
95	int freq;
96	audfmt_e fmt;
97	int nchannels;
98	unsigned int buffer_size;
99	unsigned int period_size;
100	};
101
102	struct alsa_params_obt {
103	int freq;
104	audfmt_e fmt;
105	int nchannels;
106	snd_pcm_uframes_t samples;
107	};
108
109	static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
110	{
111	va_list ap;
112
113	va_start (ap, fmt);
114	AUD_vlog (AUDIO_CAP, fmt, ap);
115	va_end (ap);
116
117	AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
118	}
119
120	static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
121	int err,
122	const char *typ,
123	const char *fmt,
124	...
125	)
126	{
127	va_list ap;
128
129	AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
130
131	va_start (ap, fmt);
132	AUD_vlog (AUDIO_CAP, fmt, ap);
133	va_end (ap);
134
135	AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
136	}
137
138	static void alsa_anal_close (snd_pcm_t **handlep)
139	{
140	int err = snd_pcm_close (*handlep);
141	if (err) {
142	alsa_logerr (err, "Failed to close PCM handle %p\n",
143	(void ) handlep);
144	}
145	*handlep = NULL;
146	}
147
148	static int alsa_write (SWVoiceOut sw, void buf, int len)
149	{
150	return audio_pcm_sw_write (sw, buf, len);
151	}
152
153	static int aud_to_alsafmt (audfmt_e fmt)
154	{
155	switch (fmt) {
156	case AUD_FMT_S8:
157	return SND_PCM_FORMAT_S8;
158
159	case AUD_FMT_U8:
160	return SND_PCM_FORMAT_U8;
161
162	case AUD_FMT_S16:
163	return SND_PCM_FORMAT_S16_LE;
164
165	case AUD_FMT_U16:
166	return SND_PCM_FORMAT_U16_LE;
167
168	default:
169	dolog ("Internal logic error: Bad audio format %d\n", fmt);
170	#ifdef DEBUG_AUDIO
171	abort ();
172	#endif
173	return SND_PCM_FORMAT_U8;
174	}
175	}
176
177	static int alsa_to_audfmt (int alsafmt, audfmt_e fmt, int endianness)
178	{
179	switch (alsafmt) {
180	case SND_PCM_FORMAT_S8:
181	*endianness = 0;
182	*fmt = AUD_FMT_S8;
183	break;
184
185	case SND_PCM_FORMAT_U8:
186	*endianness = 0;
187	*fmt = AUD_FMT_U8;
188	break;
189
190	case SND_PCM_FORMAT_S16_LE:
191	*endianness = 0;
192	*fmt = AUD_FMT_S16;
193	break;
194
195	case SND_PCM_FORMAT_U16_LE:
196	*endianness = 0;
197	*fmt = AUD_FMT_U16;
198	break;
199
200	case SND_PCM_FORMAT_S16_BE:
201	*endianness = 1;
202	*fmt = AUD_FMT_S16;
203	break;
204
205	case SND_PCM_FORMAT_U16_BE:
206	*endianness = 1;
207	*fmt = AUD_FMT_U16;
208	break;
209
210	default:
211	dolog ("Unrecognized audio format %d\n", alsafmt);
212	return -1;
213	}
214
215	return 0;
216	}
217
218	#if defined DEBUG_MISMATCHES \|\| defined DEBUG
219	static void alsa_dump_info (struct alsa_params_req *req,
220	struct alsa_params_obt *obt)
221	{
222	dolog ("parameter \| requested value \| obtained value\n");
223	dolog ("format \| %10d \| %10d\n", req->fmt, obt->fmt);
224	dolog ("channels \| %10d \| %10d\n",
225	req->nchannels, obt->nchannels);
226	dolog ("frequency \| %10d \| %10d\n", req->freq, obt->freq);
227	dolog ("============================================\n");
228	dolog ("requested: buffer size %d period size %d\n",
229	req->buffer_size, req->period_size);
230	dolog ("obtained: samples %ld\n", obt->samples);
231	}
232	#endif
233
234	static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
235	{
236	int err;
237	snd_pcm_sw_params_t *sw_params;
238
239	snd_pcm_sw_params_alloca (&sw_params);
240
241	err = snd_pcm_sw_params_current (handle, sw_params);
242	if (err < 0) {
243	dolog ("Could not fully initialize DAC\n");
244	alsa_logerr (err, "Failed to get current software parameters\n");
245	return;
246	}
247
248	err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
249	if (err < 0) {
250	dolog ("Could not fully initialize DAC\n");
251	alsa_logerr (err, "Failed to set software threshold to %ld\n",
252	threshold);
253	return;
254	}
255
256	err = snd_pcm_sw_params (handle, sw_params);
257	if (err < 0) {
258	dolog ("Could not fully initialize DAC\n");
259	alsa_logerr (err, "Failed to set software parameters\n");
260	return;
261	}
262	}
263
264	static int alsa_open (int in, struct alsa_params_req *req,
265	struct alsa_params_obt obt, snd_pcm_t *handlep)
266	{
267	snd_pcm_t *handle;
268	snd_pcm_hw_params_t *hw_params;
269	int err;
270	unsigned int freq, nchannels;
271	const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
272	unsigned int period_size, buffer_size;
273	snd_pcm_uframes_t obt_buffer_size;
274	const char *typ = in ? "ADC" : "DAC";
275
276	freq = req->freq;
277	period_size = req->period_size;
278	buffer_size = req->buffer_size;
279	nchannels = req->nchannels;
280
281	snd_pcm_hw_params_alloca (&hw_params);
282
283	err = snd_pcm_open (
284	&handle,
285	pcm_name,
286	in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
287	SND_PCM_NONBLOCK
288	);
289	if (err < 0) {
290	alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
291	LogRel(("Audio/ALSA: Failed to open '%s' as %s\n", pcm_name, typ));
292	return -1;
293	}
294
295	err = snd_pcm_hw_params_any (handle, hw_params);
296	if (err < 0) {
297	alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
298	goto err;
299	}
300
301	err = snd_pcm_hw_params_set_access (
302	handle,
303	hw_params,
304	SND_PCM_ACCESS_RW_INTERLEAVED
305	);
306	if (err < 0) {
307	alsa_logerr2 (err, typ, "Failed to set access type\n");
308	goto err;
309	}
310
311	err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
312	if (err < 0) {
313	alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
314	goto err;
315	}
316
317	err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
318	if (err < 0) {
319	alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
320	goto err;
321	}
322
323	err = snd_pcm_hw_params_set_channels_near (
324	handle,
325	hw_params,
326	&nchannels
327	);
328	if (err < 0) {
329	alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
330	req->nchannels);
331	goto err;
332	}
333
334	if (nchannels != 1 && nchannels != 2) {
335	alsa_logerr2 (err, typ,
336	"Can not handle obtained number of channels %d\n",
337	nchannels);
338	goto err;
339	}
340
341	if (!((in && conf.size_in_usec_in) \|\| (!in && conf.size_in_usec_out))) {
342	if (!buffer_size) {
343	buffer_size = DEFAULT_BUFFER_SIZE;
344	period_size= DEFAULT_PERIOD_SIZE;
345	}
346	}
347
348	if (buffer_size) {
349	if ((in && conf.size_in_usec_in) \|\| (!in && conf.size_in_usec_out)) {
350	if (period_size) {
351	err = snd_pcm_hw_params_set_period_time_near (
352	handle,
353	hw_params,
354	&period_size,
355	0
356	);
357	if (err < 0) {
358	alsa_logerr2 (err, typ,
359	"Failed to set period time %d\n",
360	req->period_size);
361	goto err;
362	}
363	}
364
365	err = snd_pcm_hw_params_set_buffer_time_near (
366	handle,
367	hw_params,
368	&buffer_size,
369	0
370	);
371
372	if (err < 0) {
373	alsa_logerr2 (err, typ,
374	"Failed to set buffer time %d\n",
375	req->buffer_size);
376	goto err;
377	}
378	}
379	else {
380	int dir;
381	snd_pcm_uframes_t minval;
382
383	if (period_size) {
384	minval = period_size;
385	dir = 0;
386
387	err = snd_pcm_hw_params_get_period_size_min (
388	hw_params,
389	&minval,
390	&dir
391	);
392	if (err < 0) {
393	alsa_logerr (
394	err,
395	"Could not get minmal period size for %s\n",
396	typ
397	);
398	}
399	else {
400	if (period_size < minval) {
401	if ((in && conf.period_size_in_overriden)
402	\|\| (!in && conf.period_size_out_overriden)) {
403	dolog ("%s period size(%d) is less "
404	"than minmal period size(%ld)\n",
405	typ,
406	period_size,
407	minval);
408	}
409	period_size = minval;
410	}
411	}
412
413	err = snd_pcm_hw_params_set_period_size (
414	handle,
415	hw_params,
416	period_size,
417	0
418	);
419	if (err < 0) {
420	alsa_logerr2 (err, typ, "Failed to set period size %d\n",
421	req->period_size);
422	goto err;
423	}
424	}
425
426	minval = buffer_size;
427	err = snd_pcm_hw_params_get_buffer_size_min (
428	hw_params,
429	&minval
430	);
431	if (err < 0) {
432	alsa_logerr (err, "Could not get minmal buffer size for %s\n",
433	typ);
434	}
435	else {
436	if (buffer_size < minval) {
437	if ((in && conf.buffer_size_in_overriden)
438	\|\| (!in && conf.buffer_size_out_overriden)) {
439	dolog (
440	"%s buffer size(%d) is less "
441	"than minimal buffer size(%ld)\n",
442	typ,
443	buffer_size,
444	minval
445	);
446	}
447	buffer_size = minval;
448	}
449	}
450
451	err = snd_pcm_hw_params_set_buffer_size (
452	handle,
453	hw_params,
454	buffer_size
455	);
456	if (err < 0) {
457	alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
458	req->buffer_size);
459	goto err;
460	}
461	}
462	}
463	else {
464	dolog ("warning: Buffer size is not set\n");
465	}
466
467	err = snd_pcm_hw_params (handle, hw_params);
468	if (err < 0) {
469	alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
470	goto err;
471	}
472
473	err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
474	if (err < 0) {
475	alsa_logerr2 (err, typ, "Failed to get buffer size\n");
476	goto err;
477	}
478
479	err = snd_pcm_prepare (handle);
480	if (err < 0) {
481	alsa_logerr2 (err, typ, "Could not prepare handle %p\n",
482	(void *) handle);
483	goto err;
484	}
485
486	if (!in && conf.threshold) {
487	snd_pcm_uframes_t threshold;
488	int bytes_per_sec;
489
490	bytes_per_sec = freq
491	<< (nchannels == 2)
492	<< (req->fmt == AUD_FMT_S16 \|\| req->fmt == AUD_FMT_U16);
493
494	threshold = (conf.threshold * bytes_per_sec) / 1000;
495	alsa_set_threshold (handle, threshold);
496	}
497
498	obt->fmt = req->fmt;
499	obt->nchannels = nchannels;
500	obt->freq = freq;
501	obt->samples = obt_buffer_size;
502	*handlep = handle;
503
504	#if defined DEBUG_MISMATCHES \|\| defined DEBUG
505	if (obt->fmt != req->fmt \|\|
506	obt->nchannels != req->nchannels \|\|
507	obt->freq != req->freq) {
508	dolog ("Audio paramters mismatch for %s\n", typ);
509	alsa_dump_info (req, obt);
510	}
511	#endif
512
513	#ifdef DEBUG
514	alsa_dump_info (req, obt);
515	#endif
516	return 0;
517
518	err:
519	alsa_anal_close (&handle);
520	return -1;
521	}
522
523	static int alsa_recover (snd_pcm_t *handle)
524	{
525	int err = snd_pcm_prepare (handle);
526	if (err < 0) {
527	alsa_logerr (err, "Failed to prepare handle %p\n",
528	(void *) handle);
529	return -1;
530	}
531	return 0;
532	}
533
534	static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
535	{
536	snd_pcm_sframes_t avail;
537
538	avail = snd_pcm_avail_update (handle);
539	if (avail < 0) {
540	if (avail == -EPIPE) {
541	if (!alsa_recover (handle)) {
542	avail = snd_pcm_avail_update (handle);
543	}
544	}
545
546	if (avail < 0) {
547	alsa_logerr (avail,
548	"Could not obtain number of available frames\n");
549	return -1;
550	}
551	}
552
553	return avail;
554	}
555
556	static int alsa_run_out (HWVoiceOut *hw)
557	{
558	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
559	int rpos, live, decr;
560	int samples;
561	uint8_t *dst;
562	st_sample_t *src;
563	snd_pcm_sframes_t avail;
564
565	live = audio_pcm_hw_get_live_out (hw);
566	if (!live) {
567	return 0;
568	}
569
570	avail = alsa_get_avail (alsa->handle);
571	if (avail < 0) {
572	dolog ("Could not get number of available playback frames\n");
573	return 0;
574	}
575
576	decr = audio_MIN (live, avail);
577	samples = decr;
578	rpos = hw->rpos;
579	while (samples) {
580	int left_till_end_samples = hw->samples - rpos;
581	int len = audio_MIN (samples, left_till_end_samples);
582	snd_pcm_sframes_t written;
583
584	src = hw->mix_buf + rpos;
585	dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
586
587	hw->clip (dst, src, len);
588
589	while (len) {
590	written = snd_pcm_writei (alsa->handle, dst, len);
591
592	if (written <= 0) {
593	switch (written) {
594	case 0:
595	if (conf.verbose) {
596	dolog ("Failed to write %d frames (wrote zero)\n", len);
597	}
598	goto exit;
599
600	case -EPIPE:
601	if (alsa_recover (alsa->handle)) {
602	alsa_logerr (written, "Failed to write %d frames\n",
603	len);
604	goto exit;
605	}
606	if (conf.verbose) {
607	dolog ("Recovering from playback xrun\n");
608	}
609	continue;
610
611	case -EAGAIN:
612	goto exit;
613
614	default:
615	alsa_logerr (written, "Failed to write %d frames to %p\n",
616	len, dst);
617	goto exit;
618	}
619	}
620
621	#if 0
622	mixeng_sniff_and_clear (hw, src, dst, written);
623	#endif
624	rpos = (rpos + written) % hw->samples;
625	samples -= written;
626	len -= written;
627	dst = advance (dst, written << hw->info.shift);
628	src += written;
629	}
630	}
631
632	exit:
633	hw->rpos = rpos;
634	return decr;
635	}
636
637	static void alsa_fini_out (HWVoiceOut *hw)
638	{
639	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
640
641	ldebug ("alsa_fini\n");
642	alsa_anal_close (&alsa->handle);
643
644	if (alsa->pcm_buf) {
645	qemu_free (alsa->pcm_buf);
646	alsa->pcm_buf = NULL;
647	}
648	}
649
650	static int alsa_init_out (HWVoiceOut hw, audsettings_t as)
651	{
652	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
653	struct alsa_params_req req;
654	struct alsa_params_obt obt;
655	audfmt_e effective_fmt;
656	int endianness;
657	int err;
658	snd_pcm_t *handle;
659	audsettings_t obt_as;
660
661	req.fmt = aud_to_alsafmt (as->fmt);
662	req.freq = as->freq;
663	req.nchannels = as->nchannels;
664	req.period_size = conf.period_size_out;
665	req.buffer_size = conf.buffer_size_out;
666
667	if (alsa_open (0, &req, &obt, &handle)) {
668	return -1;
669	}
670
671	err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
672	if (err) {
673	alsa_anal_close (&handle);
674	return -1;
675	}
676
677	obt_as.freq = obt.freq;
678	obt_as.nchannels = obt.nchannels;
679	obt_as.fmt = effective_fmt;
680	obt_as.endianness = endianness;
681
682	audio_pcm_init_info (&hw->info, &obt_as);
683	hw->samples = obt.samples;
684
685	alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
686	if (!alsa->pcm_buf) {
687	dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
688	hw->samples, 1 << hw->info.shift);
689	alsa_anal_close (&handle);
690	return -1;
691	}
692
693	alsa->handle = handle;
694	return 0;
695	}
696
697	static int alsa_voice_ctl (snd_pcm_t handle, const char typ, int pause)
698	{
699	int err;
700
701	if (pause) {
702	err = snd_pcm_drop (handle);
703	if (err < 0) {
704	alsa_logerr (err, "Could not stop %s\n", typ);
705	return -1;
706	}
707	}
708	else {
709	err = snd_pcm_prepare (handle);
710	if (err < 0) {
711	alsa_logerr (err, "Could not prepare handle for %s\n", typ);
712	return -1;
713	}
714	}
715
716	return 0;
717	}
718
719	static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
720	{
721	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
722
723	switch (cmd) {
724	case VOICE_ENABLE:
725	ldebug ("enabling voice\n");
726	return alsa_voice_ctl (alsa->handle, "playback", 0);
727
728	case VOICE_DISABLE:
729	ldebug ("disabling voice\n");
730	return alsa_voice_ctl (alsa->handle, "playback", 1);
731	}
732
733	return -1;
734	}
735
736	static int alsa_init_in (HWVoiceIn hw, audsettings_t as)
737	{
738	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
739	struct alsa_params_req req;
740	struct alsa_params_obt obt;
741	int endianness;
742	int err;
743	audfmt_e effective_fmt;
744	snd_pcm_t *handle;
745	audsettings_t obt_as;
746
747	req.fmt = aud_to_alsafmt (as->fmt);
748	req.freq = as->freq;
749	req.nchannels = as->nchannels;
750	req.period_size = conf.period_size_in;
751	req.buffer_size = conf.buffer_size_in;
752
753	if (alsa_open (1, &req, &obt, &handle)) {
754	return -1;
755	}
756
757	err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
758	if (err) {
759	alsa_anal_close (&handle);
760	return -1;
761	}
762
763	obt_as.freq = obt.freq;
764	obt_as.nchannels = obt.nchannels;
765	obt_as.fmt = effective_fmt;
766	obt_as.endianness = endianness;
767
768	audio_pcm_init_info (&hw->info, &obt_as);
769	hw->samples = obt.samples;
770
771	alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
772	if (!alsa->pcm_buf) {
773	dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
774	hw->samples, 1 << hw->info.shift);
775	alsa_anal_close (&handle);
776	return -1;
777	}
778
779	alsa->handle = handle;
780	return 0;
781	}
782
783	static void alsa_fini_in (HWVoiceIn *hw)
784	{
785	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
786
787	alsa_anal_close (&alsa->handle);
788
789	if (alsa->pcm_buf) {
790	qemu_free (alsa->pcm_buf);
791	alsa->pcm_buf = NULL;
792	}
793	}
794
795	static int alsa_run_in (HWVoiceIn *hw)
796	{
797	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
798	int hwshift = hw->info.shift;
799	int i;
800	int live = audio_pcm_hw_get_live_in (hw);
801	int dead = hw->samples - live;
802	int decr;
803	struct {
804	int add;
805	int len;
806	} bufs[2];
807
808	snd_pcm_sframes_t avail;
809	snd_pcm_uframes_t read_samples = 0;
810
811	bufs[0].add = hw->wpos;
812	bufs[0].len = 0;
813	bufs[1].add = 0;
814	bufs[1].len = 0;
815
816	if (!dead) {
817	return 0;
818	}
819
820	avail = alsa_get_avail (alsa->handle);
821	if (avail < 0) {
822	dolog ("Could not get number of captured frames\n");
823	return 0;
824	}
825
826	if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
827	avail = hw->samples;
828	}
829
830	decr = audio_MIN (dead, avail);
831	if (!decr) {
832	return 0;
833	}
834
835	if (hw->wpos + decr > hw->samples) {
836	bufs[0].len = (hw->samples - hw->wpos);
837	bufs[1].len = (decr - (hw->samples - hw->wpos));
838	}
839	else {
840	bufs[0].len = decr;
841	}
842
843	for (i = 0; i < 2; ++i) {
844	void *src;
845	st_sample_t *dst;
846	snd_pcm_sframes_t nread;
847	snd_pcm_uframes_t len;
848
849	len = bufs[i].len;
850
851	src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
852	dst = hw->conv_buf + bufs[i].add;
853
854	while (len) {
855	nread = snd_pcm_readi (alsa->handle, src, len);
856
857	if (nread <= 0) {
858	switch (nread) {
859	case 0:
860	if (conf.verbose) {
861	dolog ("Failed to read %ld frames (read zero)\n", len);
862	}
863	goto exit;
864
865	case -EPIPE:
866	if (alsa_recover (alsa->handle)) {
867	alsa_logerr (nread, "Failed to read %ld frames\n", len);
868	goto exit;
869	}
870	if (conf.verbose) {
871	dolog ("Recovering from capture xrun\n");
872	}
873	continue;
874
875	case -EAGAIN:
876	goto exit;
877
878	default:
879	alsa_logerr (
880	nread,
881	"Failed to read %ld frames from %p\n",
882	len,
883	src
884	);
885	goto exit;
886	}
887	}
888
889	hw->conv (dst, src, nread, &nominal_volume);
890
891	src = advance (src, nread << hwshift);
892	dst += nread;
893
894	read_samples += nread;
895	len -= nread;
896	}
897	}
898
899	exit:
900	hw->wpos = (hw->wpos + read_samples) % hw->samples;
901	return read_samples;
902	}
903
904	static int alsa_read (SWVoiceIn sw, void buf, int size)
905	{
906	return audio_pcm_sw_read (sw, buf, size);
907	}
908
909	static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
910	{
911	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
912
913	switch (cmd) {
914	case VOICE_ENABLE:
915	ldebug ("enabling voice\n");
916	return alsa_voice_ctl (alsa->handle, "capture", 0);
917
918	case VOICE_DISABLE:
919	ldebug ("disabling voice\n");
920	return alsa_voice_ctl (alsa->handle, "capture", 1);
921	}
922
923	return -1;
924	}
925
926	static void *alsa_audio_init (void)
927	{
928	return &conf;
929	}
930
931	static void alsa_audio_fini (void *opaque)
932	{
933	(void) opaque;
934	}
935
936	static struct audio_option alsa_options[] = {
937	{"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
938	"DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
939	{"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
940	"DAC period size", &conf.period_size_out_overriden, 0},
941	{"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
942	"DAC buffer size", &conf.buffer_size_out_overriden, 0},
943
944	{"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
945	"ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
946	{"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
947	"ADC period size", &conf.period_size_in_overriden, 0},
948	{"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
949	"ADC buffer size", &conf.buffer_size_in_overriden, 0},
950
951	{"THRESHOLD", AUD_OPT_INT, &conf.threshold,
952	"(undocumented)", NULL, 0},
953
954	{"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
955	"DAC device name (for instance dmix)", NULL, 0},
956
957	{"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
958	"ADC device name", NULL, 0},
959
960	{"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
961	"Behave in a more verbose way", NULL, 0},
962
963	{NULL, 0, NULL, NULL, NULL, 0}
964	};
965
966	static struct audio_pcm_ops alsa_pcm_ops = {
967	alsa_init_out,
968	alsa_fini_out,
969	alsa_run_out,
970	alsa_write,
971	alsa_ctl_out,
972
973	alsa_init_in,
974	alsa_fini_in,
975	alsa_run_in,
976	alsa_read,
977	alsa_ctl_in
978	};
979
980	struct audio_driver alsa_audio_driver = {
981	INIT_FIELD (name = ) "alsa",
982	INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
983	INIT_FIELD (options = ) alsa_options,
984	INIT_FIELD (init = ) alsa_audio_init,
985	INIT_FIELD (fini = ) alsa_audio_fini,
986	INIT_FIELD (pcm_ops = ) &alsa_pcm_ops,
987	INIT_FIELD (can_be_default = ) 1,
988	INIT_FIELD (max_voices_out = ) INT_MAX,
989	INIT_FIELD (max_voices_in = ) INT_MAX,
990	INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
991	INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)
992	};

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source: vbox/trunk/src/VBox/Devices/Audio/alsaaudio.c@ 1

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