alsaaudio.c@ 55747

Last change on this file since 55747 was 45136, checked in by vboxsync, 12 years ago
Devices/Audio: mangle symbols of libasound and libpulse, otherwise they can be resolved by dynamic libraries if -fvisibility is not supported
Property svn:eol-style set to `native` Property svn:keywords set to `Author Date Id Revision`
File size: 31.9 KB

Line
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	#ifdef VBOX
25	#ifndef DEBUG
26	#define NDEBUG
27	#endif
28	#define LOG_GROUP LOG_GROUP_DEV_AUDIO
29	#include <VBox/log.h>
30	#endif
31
32	#ifdef VBOX
33	# include "alsa_stubs.h"
34	# include "alsa_mangling.h"
35	#endif
36
37	#include <alsa/asoundlib.h>
38
39	#include "VBoxDD.h"
40	#include "vl_vbox.h"
41	#include "audio.h"
42	#include <iprt/alloc.h>
43
44	#define AUDIO_CAP "alsa"
45	#include "audio_int.h"
46
47	typedef struct ALSAVoiceOut {
48	HWVoiceOut hw;
49	void *pcm_buf;
50	snd_pcm_t *handle;
51	} ALSAVoiceOut;
52
53	typedef struct ALSAVoiceIn {
54	HWVoiceIn hw;
55	snd_pcm_t *handle;
56	void *pcm_buf;
57	} ALSAVoiceIn;
58
59	/* latency = period_size * periods / (rate * bytes_per_frame) */
60
61	static struct {
62	int size_in_usec_in;
63	int size_in_usec_out;
64	const char *pcm_name_in;
65	const char *pcm_name_out;
66	unsigned int buffer_size_in;
67	unsigned int period_size_in;
68	unsigned int buffer_size_out;
69	unsigned int period_size_out;
70	unsigned int threshold;
71
72	int buffer_size_in_overriden;
73	int period_size_in_overriden;
74
75	int buffer_size_out_overriden;
76	int period_size_out_overriden;
77	int verbose;
78	} conf = {
79	#ifdef HIGH_LATENCY
80	INIT_FIELD (.size_in_usec_in =) 1,
81	INIT_FIELD (.size_in_usec_out =) 1,
82	#else
83	INIT_FIELD (.size_in_usec_in =) 0,
84	INIT_FIELD (.size_in_usec_out =) 0,
85	#endif
86	INIT_FIELD (.pcm_name_out =) "default",
87	INIT_FIELD (.pcm_name_in =) "default",
88	#ifdef HIGH_LATENCY
89	INIT_FIELD (.buffer_size_in =) 400000,
90	INIT_FIELD (.period_size_in =) 400000 / 4,
91	INIT_FIELD (.buffer_size_out =) 400000,
92	INIT_FIELD (.period_size_out =) 400000 / 4,
93	#else
94	#define DEFAULT_BUFFER_SIZE 1024
95	#define DEFAULT_PERIOD_SIZE 256
96	INIT_FIELD (.buffer_size_in =) DEFAULT_BUFFER_SIZE * 4,
97	INIT_FIELD (.period_size_in =) DEFAULT_PERIOD_SIZE * 4,
98	INIT_FIELD (.buffer_size_out =) DEFAULT_BUFFER_SIZE,
99	INIT_FIELD (.period_size_out =) DEFAULT_PERIOD_SIZE,
100	#endif
101	INIT_FIELD (.threshold =) 0,
102	INIT_FIELD (.buffer_size_in_overriden =) 0,
103	INIT_FIELD (.period_size_in_overriden =) 0,
104	INIT_FIELD (.buffer_size_out_overriden =) 0,
105	INIT_FIELD (.period_size_out_overriden =) 0,
106	INIT_FIELD (.verbose =) 0
107	};
108
109	struct alsa_params_req {
110	int freq;
111	audfmt_e fmt;
112	int nchannels;
113	unsigned long buffer_size;
114	unsigned long period_size;
115	};
116
117	struct alsa_params_obt {
118	int freq;
119	audfmt_e fmt;
120	int nchannels;
121	snd_pcm_uframes_t samples;
122	};
123
124	static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
125	{
126	va_list ap;
127
128	va_start (ap, fmt);
129	AUD_vlog (AUDIO_CAP, fmt, ap);
130	va_end (ap);
131
132	AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
133	}
134
135	static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
136	int err,
137	const char *typ,
138	const char *fmt,
139	...
140	)
141	{
142	va_list ap;
143
144	AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
145
146	va_start (ap, fmt);
147	AUD_vlog (AUDIO_CAP, fmt, ap);
148	va_end (ap);
149
150	AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
151	}
152
153	static void alsa_anal_close (snd_pcm_t **handlep)
154	{
155	int err = snd_pcm_close (*handlep);
156	if (err) {
157	alsa_logerr (err, "Failed to close PCM handle %p\n",
158	(void ) handlep);
159	}
160	*handlep = NULL;
161	}
162
163	static int alsa_write (SWVoiceOut sw, void buf, int len)
164	{
165	return audio_pcm_sw_write (sw, buf, len);
166	}
167
168	static int aud_to_alsafmt (audfmt_e fmt)
169	{
170	switch (fmt) {
171	case AUD_FMT_S8:
172	return SND_PCM_FORMAT_S8;
173
174	case AUD_FMT_U8:
175	return SND_PCM_FORMAT_U8;
176
177	case AUD_FMT_S16:
178	return SND_PCM_FORMAT_S16_LE;
179
180	case AUD_FMT_U16:
181	return SND_PCM_FORMAT_U16_LE;
182
183	case AUD_FMT_S32:
184	return SND_PCM_FORMAT_S32_LE;
185
186	case AUD_FMT_U32:
187	return SND_PCM_FORMAT_U32_LE;
188
189	default:
190	dolog ("Internal logic error: Bad audio format %d\n", fmt);
191	#ifdef DEBUG_AUDIO
192	abort ();
193	#endif
194	return SND_PCM_FORMAT_U8;
195	}
196	}
197
198	static int alsa_to_audfmt (int alsafmt, audfmt_e fmt, int endianness)
199	{
200	switch (alsafmt) {
201	case SND_PCM_FORMAT_S8:
202	*endianness = 0;
203	*fmt = AUD_FMT_S8;
204	break;
205
206	case SND_PCM_FORMAT_U8:
207	*endianness = 0;
208	*fmt = AUD_FMT_U8;
209	break;
210
211	case SND_PCM_FORMAT_S16_LE:
212	*endianness = 0;
213	*fmt = AUD_FMT_S16;
214	break;
215
216	case SND_PCM_FORMAT_U16_LE:
217	*endianness = 0;
218	*fmt = AUD_FMT_U16;
219	break;
220
221	case SND_PCM_FORMAT_S16_BE:
222	*endianness = 1;
223	*fmt = AUD_FMT_S16;
224	break;
225
226	case SND_PCM_FORMAT_U16_BE:
227	*endianness = 1;
228	*fmt = AUD_FMT_U16;
229	break;
230
231	case SND_PCM_FORMAT_S32_LE:
232	*endianness = 0;
233	*fmt = AUD_FMT_S32;
234	break;
235
236	case SND_PCM_FORMAT_U32_LE:
237	*endianness = 0;
238	*fmt = AUD_FMT_U32;
239	break;
240
241	case SND_PCM_FORMAT_S32_BE:
242	*endianness = 1;
243	*fmt = AUD_FMT_S32;
244	break;
245
246	case SND_PCM_FORMAT_U32_BE:
247	*endianness = 1;
248	*fmt = AUD_FMT_U32;
249	break;
250
251	default:
252	dolog ("Unrecognized audio format %d\n", alsafmt);
253	return -1;
254	}
255
256	return 0;
257	}
258
259	#if defined DEBUG_MISMATCHES \|\| defined DEBUG
260	static void alsa_dump_info (struct alsa_params_req *req,
261	struct alsa_params_obt *obt)
262	{
263	dolog ("parameter \| requested value \| obtained value\n");
264	dolog ("format \| %10d \| %10d\n", req->fmt, obt->fmt);
265	dolog ("channels \| %10d \| %10d\n",
266	req->nchannels, obt->nchannels);
267	dolog ("frequency \| %10d \| %10d\n", req->freq, obt->freq);
268	dolog ("============================================\n");
269	dolog ("requested: buffer size %d period size %d\n",
270	req->buffer_size, req->period_size);
271	dolog ("obtained: samples %ld\n", obt->samples);
272	}
273	#endif
274
275	static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
276	{
277	int err;
278	snd_pcm_sw_params_t *sw_params;
279
280	snd_pcm_sw_params_alloca (&sw_params);
281
282	err = snd_pcm_sw_params_current (handle, sw_params);
283	if (err < 0) {
284	dolog ("Could not fully initialize DAC\n");
285	alsa_logerr (err, "Failed to get current software parameters\n");
286	return;
287	}
288
289	err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
290	if (err < 0) {
291	dolog ("Could not fully initialize DAC\n");
292	alsa_logerr (err, "Failed to set software threshold to %ld\n",
293	threshold);
294	return;
295	}
296
297	err = snd_pcm_sw_params (handle, sw_params);
298	if (err < 0) {
299	dolog ("Could not fully initialize DAC\n");
300	alsa_logerr (err, "Failed to set software parameters\n");
301	return;
302	}
303	}
304
305	static int alsa_open (int in, struct alsa_params_req *req,
306	struct alsa_params_obt obt, snd_pcm_t *handlep)
307	{
308	snd_pcm_t *handle;
309	snd_pcm_hw_params_t *hw_params;
310	int err, dir;
311	unsigned int freq, nchannels;
312	const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
313	unsigned int period_size, buffer_size;
314	snd_pcm_uframes_t period_size_f, buffer_size_f;
315	snd_pcm_uframes_t obt_buffer_size, obt_period_size;
316	const char *typ = in ? "ADC" : "DAC";
317
318	freq = req->freq;
319	period_size = req->period_size;
320	buffer_size = req->buffer_size;
321	period_size_f = (snd_pcm_uframes_t)period_size;
322	buffer_size_f = (snd_pcm_uframes_t)buffer_size;
323	nchannels = req->nchannels;
324
325	snd_pcm_hw_params_alloca (&hw_params);
326
327	err = snd_pcm_open (
328	&handle,
329	pcm_name,
330	in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
331	SND_PCM_NONBLOCK
332	);
333	if (err < 0) {
334	#ifndef VBOX
335	alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
336	#else
337	LogRel(("ALSA: Failed to open '%s' as %s\n", pcm_name, typ));
338	#endif
339	return -1;
340	}
341
342	err = snd_pcm_hw_params_any (handle, hw_params);
343	if (err < 0) {
344	#ifndef VBOX
345	alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
346	#else
347	LogRel(("ALSA: Failed to initialize hardware parameters\n"));
348	#endif
349	goto err;
350	}
351
352	err = snd_pcm_hw_params_set_access (
353	handle,
354	hw_params,
355	SND_PCM_ACCESS_RW_INTERLEAVED
356	);
357	if (err < 0) {
358	#ifndef VBOX
359	alsa_logerr2 (err, typ, "Failed to set access type\n");
360	#else
361	LogRel(("ALSA: Failed to set access type\n"));
362	#endif
363	goto err;
364	}
365
366	err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
367	if (err < 0) {
368	#ifndef VBOX
369	alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
370	#else
371	LogRel(("ALSA: Failed to set format %d\n", req->fmt));
372	#endif
373	goto err;
374	}
375
376	err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
377	if (err < 0) {
378	#ifndef VBOX
379	alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
380	#else
381	LogRel(("ALSA: Failed to set frequency %dHz\n", req->freq));
382	#endif
383	goto err;
384	}
385
386	err = snd_pcm_hw_params_set_channels_near (
387	handle,
388	hw_params,
389	&nchannels
390	);
391	if (err < 0) {
392	#ifndef VBOX
393	alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
394	req->nchannels);
395	#else
396	LogRel(("ALSA: Failed to set number of channels to %d\n", req->nchannels));
397	#endif
398	goto err;
399	}
400
401	if (nchannels != 1 && nchannels != 2) {
402	#ifndef VBOX
403	alsa_logerr2 (err, typ,
404	"Can not handle obtained number of channels %d\n",
405	nchannels);
406	#else
407	LogRel(("ALSA: Cannot handle obtained number of channels (%d)\n", nchannels));
408	#endif
409	goto err;
410	}
411
412	if (!((in && conf.size_in_usec_in) \|\| (!in && conf.size_in_usec_out))) {
413	if (!buffer_size) {
414	buffer_size = DEFAULT_BUFFER_SIZE;
415	period_size= DEFAULT_PERIOD_SIZE;
416	}
417	}
418
419	if (buffer_size) {
420	if ((in && conf.size_in_usec_in) \|\| (!in && conf.size_in_usec_out)) {
421	if (period_size) {
422	err = snd_pcm_hw_params_set_period_time_near (
423	handle,
424	hw_params,
425	&period_size,
426	0
427	);
428	if (err < 0) {
429	#ifndef VBOX
430	alsa_logerr2 (err, typ,
431	"Failed to set period time %d\n",
432	req->period_size);
433	#else
434	LogRel(("ALSA: Failed to set period time %d\n", req->period_size));
435	#endif
436	goto err;
437	}
438	}
439
440	err = snd_pcm_hw_params_set_buffer_time_near (
441	handle,
442	hw_params,
443	&buffer_size,
444	0
445	);
446
447	if (err < 0) {
448	#ifndef VBOX
449	alsa_logerr2 (err, typ,
450	"Failed to set buffer time %d\n",
451	req->buffer_size);
452	#else
453	LogRel(("ALSA: Failed to set buffer time %d\n", req->buffer_size));
454	#endif
455	goto err;
456	}
457	}
458	else {
459	snd_pcm_uframes_t minval;
460
461	if (period_size_f) {
462	minval = period_size_f;
463	dir = 0;
464
465	err = snd_pcm_hw_params_get_period_size_min (
466	hw_params,
467	&minval,
468	&dir
469	);
470	if (err < 0) {
471	#ifndef VBOX
472	alsa_logerr (
473	err,
474	"Could not get minimal period size for %s\n",
475	typ
476	);
477	#else
478	LogRel(("ALSA: Could not get minimal period size for %s\n", typ));
479	#endif
480	}
481	else {
482	dolog("minimal period size %ld\n", minval);
483	if (period_size_f < minval) {
484	if ((in && conf.period_size_in_overriden)
485	\|\| (!in && conf.period_size_out_overriden)) {
486	dolog ("%s period size(%d) is less "
487	"than minimal period size(%ld)\n",
488	typ,
489	period_size_f,
490	minval);
491	}
492	period_size_f = minval;
493	}
494	}
495
496	#ifndef VBOX
497	err = snd_pcm_hw_params_set_period_size (
498	handle,
499	hw_params,
500	period_size_f,
501	0
502	);
503	#else
504	err = snd_pcm_hw_params_set_period_size_near (
505	handle,
506	hw_params,
507	&period_size_f,
508	0
509	);
510	#endif
511	dolog("PERIOD_SIZE %d\n", period_size_f);
512	if (err < 0) {
513	#ifndef VBOX
514	alsa_logerr2 (err, typ, "Failed to set period size %d\n",
515	period_size_f);
516	#else
517	LogRel(("ALSA: Failed to set period size %d (%s)\n",
518	period_size_f, snd_strerror(err)));
519	#endif
520	goto err;
521	}
522	}
523
524	#ifdef VBOX
525	/* Calculate default buffer size here since it might have been changed
526	* in the _near functions */
527	buffer_size_f = 4 * period_size_f;
528	#endif
529
530	minval = buffer_size_f;
531	err = snd_pcm_hw_params_get_buffer_size_min (
532	hw_params,
533	&minval
534	);
535	if (err < 0) {
536	#ifndef VBOX
537	alsa_logerr (err, "Could not get minimal buffer size for %s\n",
538	typ);
539	#else
540	LogRel(("ALSA: Could not get minimal buffer size for %s\n", typ));
541	#endif
542	}
543	else {
544	if (buffer_size_f < minval) {
545	if ((in && conf.buffer_size_in_overriden)
546	\|\| (!in && conf.buffer_size_out_overriden)) {
547	dolog (
548	"%s buffer size(%d) is less "
549	"than minimal buffer size(%ld)\n",
550	typ,
551	buffer_size_f,
552	minval
553	);
554	}
555	buffer_size_f = minval;
556	}
557	}
558
559	err = snd_pcm_hw_params_set_buffer_size_near (
560	handle,
561	hw_params,
562	&buffer_size_f
563	);
564	dolog("BUFFER_SIZE %d\n", buffer_size_f);
565	if (err < 0) {
566	#ifndef VBOX
567	alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
568	buffer_size_f);
569	#else
570	LogRel(("ALSA: Failed to set buffer size %d (%s)\n",
571	buffer_size_f, snd_strerror(err)));
572	#endif
573	goto err;
574	}
575	}
576	}
577	else {
578	dolog ("warning: Buffer size is not set\n");
579	}
580
581	err = snd_pcm_hw_params (handle, hw_params);
582	if (err < 0) {
583	#ifndef VBOX
584	alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
585	#else
586	LogRel(("ALSA: Failed to apply audio parameters\n"));
587	#endif
588	goto err;
589	}
590
591	err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
592	if (err < 0) {
593	#ifndef VBOX
594	alsa_logerr2 (err, typ, "Failed to get buffer size\n");
595	#else
596	LogRel(("ALSA: Failed to get buffer size\n"));
597	#endif
598	goto err;
599	}
600
601	#ifdef VBOX
602	dir = 0;
603	err = snd_pcm_hw_params_get_period_size (hw_params, &obt_period_size, &dir);
604	if (err < 0)
605	{
606	LogRel(("ALSA: Failed to get period size\n"));
607	goto err;
608	}
609	LogRel(("ALSA: %s frequency %dHz, period size %ld, buffer size %ld\n",
610	typ, req->freq, obt_period_size, obt_buffer_size));
611	#endif
612
613	err = snd_pcm_prepare (handle);
614	if (err < 0) {
615	alsa_logerr2 (err, typ, "Could not prepare handle %p\n",
616	(void *) handle);
617	goto err;
618	}
619
620	if (!in && conf.threshold) {
621	snd_pcm_uframes_t threshold;
622	int bytes_per_sec;
623
624	bytes_per_sec = freq
625	<< (nchannels == 2)
626	<< (req->fmt == AUD_FMT_S16 \|\| req->fmt == AUD_FMT_U16);
627
628	threshold = (conf.threshold * bytes_per_sec) / 1000;
629	alsa_set_threshold (handle, threshold);
630	}
631
632	obt->fmt = req->fmt;
633	obt->nchannels = nchannels;
634	obt->freq = freq;
635	obt->samples = obt_buffer_size;
636	*handlep = handle;
637
638	#if defined DEBUG_MISMATCHES \|\| defined DEBUG
639	if (obt->fmt != req->fmt \|\|
640	obt->nchannels != req->nchannels \|\|
641	obt->freq != req->freq) {
642	dolog ("Audio parameters mismatch for %s\n", typ);
643	alsa_dump_info (req, obt);
644	}
645	#endif
646
647	#ifdef DEBUG
648	alsa_dump_info (req, obt);
649	#endif
650	return 0;
651
652	err:
653	alsa_anal_close (&handle);
654	return -1;
655	}
656
657	static int alsa_recover (snd_pcm_t *handle)
658	{
659	int err = snd_pcm_prepare (handle);
660	if (err < 0) {
661	alsa_logerr (err, "Failed to prepare handle %p\n",
662	(void *) handle);
663	return -1;
664	}
665	return 0;
666	}
667
668	static int alsa_resume (snd_pcm_t *handle)
669	{
670	int err = snd_pcm_resume (handle);
671	if (err < 0) {
672	#ifndef VBOX
673	alsa_logerr (err, "Failed to resume handle %p\n", handle);
674	#endif
675	return -1;
676	}
677	return 0;
678	}
679
680	static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
681	{
682	snd_pcm_sframes_t avail;
683
684	avail = snd_pcm_avail_update (handle);
685	if (avail < 0) {
686	if (avail == -EPIPE) {
687	if (!alsa_recover (handle)) {
688	avail = snd_pcm_avail_update (handle);
689	}
690	}
691
692	if (avail < 0) {
693	alsa_logerr (avail,
694	"Could not obtain number of available frames\n");
695	return -1;
696	}
697	}
698
699	return avail;
700	}
701
702	static int alsa_run_out (HWVoiceOut *hw)
703	{
704	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
705	int rpos, live, decr;
706	int samples;
707	uint8_t *dst;
708	st_sample_t *src;
709	snd_pcm_sframes_t avail;
710
711	live = audio_pcm_hw_get_live_out (hw);
712	if (!live) {
713	return 0;
714	}
715
716	avail = alsa_get_avail (alsa->handle);
717	if (avail < 0) {
718	dolog ("Could not get number of available playback frames\n");
719	return 0;
720	}
721
722	decr = audio_MIN (live, avail);
723	samples = decr;
724	rpos = hw->rpos;
725	while (samples) {
726	int left_till_end_samples = hw->samples - rpos;
727	int len = audio_MIN (samples, left_till_end_samples);
728	snd_pcm_sframes_t written;
729
730	src = hw->mix_buf + rpos;
731	dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
732
733	hw->clip (dst, src, len);
734
735	while (len) {
736	written = snd_pcm_writei (alsa->handle, dst, len);
737
738	if (written <= 0) {
739	switch (written) {
740	case 0:
741	if (conf.verbose) {
742	dolog ("Failed to write %d frames (wrote zero)\n", len);
743	}
744	goto exit;
745
746	case -EPIPE:
747	if (alsa_recover (alsa->handle)) {
748	alsa_logerr (written, "Failed to write %d frames\n",
749	len);
750	goto exit;
751	}
752	if (conf.verbose) {
753	dolog ("Recovering from playback xrun\n");
754	}
755	continue;
756
757	case -ESTRPIPE:
758	/* stream is suspended and waiting for an
759	application recovery */
760	if (alsa_resume (alsa->handle)) {
761	#ifndef VBOX
762	alsa_logerr (written, "Failed to write %d frames\n", len);
763	#else
764	LogRel(("ALSA: Failed to resume output stream\n"));
765	#endif
766	goto exit;
767	}
768	if (conf.verbose) {
769	dolog ("Resuming suspended output stream\n");
770	}
771	continue;
772
773	case -EAGAIN:
774	goto exit;
775
776	default:
777	alsa_logerr (written, "Failed to write %d frames to %p\n",
778	len, dst);
779	goto exit;
780	}
781	}
782
783	rpos = (rpos + written) % hw->samples;
784	samples -= written;
785	len -= written;
786	dst = advance (dst, written << hw->info.shift);
787	src += written;
788	}
789	}
790
791	exit:
792	hw->rpos = rpos;
793	return decr;
794	}
795
796	static void alsa_fini_out (HWVoiceOut *hw)
797	{
798	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
799
800	ldebug ("alsa_fini\n");
801	alsa_anal_close (&alsa->handle);
802
803	if (alsa->pcm_buf) {
804	qemu_free (alsa->pcm_buf);
805	alsa->pcm_buf = NULL;
806	}
807	}
808
809	static int alsa_init_out (HWVoiceOut hw, audsettings_t as)
810	{
811	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
812	struct alsa_params_req req;
813	struct alsa_params_obt obt;
814	audfmt_e effective_fmt;
815	int endianness;
816	int err;
817	snd_pcm_t *handle;
818	audsettings_t obt_as;
819
820	req.fmt = aud_to_alsafmt (as->fmt);
821	req.freq = as->freq;
822	req.nchannels = as->nchannels;
823	req.period_size = conf.period_size_out;
824	req.buffer_size = conf.buffer_size_out;
825
826	if (alsa_open (0, &req, &obt, &handle)) {
827	return -1;
828	}
829
830	err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
831	if (err) {
832	alsa_anal_close (&handle);
833	return -1;
834	}
835
836	obt_as.freq = obt.freq;
837	obt_as.nchannels = obt.nchannels;
838	obt_as.fmt = effective_fmt;
839	obt_as.endianness = endianness;
840
841	audio_pcm_init_info (&hw->info, &obt_as);
842	hw->samples = obt.samples;
843
844	alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
845	if (!alsa->pcm_buf) {
846	dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
847	hw->samples, 1 << hw->info.shift);
848	alsa_anal_close (&handle);
849	return -1;
850	}
851
852	alsa->handle = handle;
853	return 0;
854	}
855
856	static int alsa_voice_ctl (snd_pcm_t handle, const char typ, int pauseit) /* VBOX: s/pause/pauseit/; -Wshadow */
857	{
858	int err;
859
860	if (pauseit) {
861	err = snd_pcm_drop (handle);
862	if (err < 0) {
863	alsa_logerr (err, "Could not stop %s\n", typ);
864	return -1;
865	}
866	}
867	else {
868	err = snd_pcm_prepare (handle);
869	if (err < 0) {
870	alsa_logerr (err, "Could not prepare handle for %s\n", typ);
871	return -1;
872	}
873	}
874
875	return 0;
876	}
877
878	static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
879	{
880	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
881
882	switch (cmd) {
883	case VOICE_ENABLE:
884	ldebug ("enabling voice\n");
885	return alsa_voice_ctl (alsa->handle, "playback", 0);
886
887	case VOICE_DISABLE:
888	ldebug ("disabling voice\n");
889	return alsa_voice_ctl (alsa->handle, "playback", 1);
890	}
891
892	return -1;
893	}
894
895	static int alsa_init_in (HWVoiceIn hw, audsettings_t as)
896	{
897	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
898	struct alsa_params_req req;
899	struct alsa_params_obt obt;
900	int endianness;
901	int err;
902	audfmt_e effective_fmt;
903	snd_pcm_t *handle;
904	audsettings_t obt_as;
905
906	req.fmt = aud_to_alsafmt (as->fmt);
907	req.freq = as->freq;
908	req.nchannels = as->nchannels;
909	req.period_size = conf.period_size_in;
910	req.buffer_size = conf.buffer_size_in;
911
912	if (alsa_open (1, &req, &obt, &handle)) {
913	return -1;
914	}
915
916	err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
917	if (err) {
918	alsa_anal_close (&handle);
919	return -1;
920	}
921
922	obt_as.freq = obt.freq;
923	obt_as.nchannels = obt.nchannels;
924	obt_as.fmt = effective_fmt;
925	obt_as.endianness = endianness;
926
927	audio_pcm_init_info (&hw->info, &obt_as);
928	hw->samples = obt.samples;
929
930	alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
931	if (!alsa->pcm_buf) {
932	dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
933	hw->samples, 1 << hw->info.shift);
934	alsa_anal_close (&handle);
935	return -1;
936	}
937
938	alsa->handle = handle;
939	return 0;
940	}
941
942	static void alsa_fini_in (HWVoiceIn *hw)
943	{
944	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
945
946	alsa_anal_close (&alsa->handle);
947
948	if (alsa->pcm_buf) {
949	qemu_free (alsa->pcm_buf);
950	alsa->pcm_buf = NULL;
951	}
952	}
953
954	static int alsa_run_in (HWVoiceIn *hw)
955	{
956	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
957	int hwshift = hw->info.shift;
958	int i;
959	int live = audio_pcm_hw_get_live_in (hw);
960	int dead = hw->samples - live;
961	int decr;
962	struct {
963	int add;
964	int len;
965	} bufs[2];
966
967	snd_pcm_sframes_t avail;
968	snd_pcm_uframes_t read_samples = 0;
969
970	bufs[0].add = hw->wpos;
971	bufs[0].len = 0;
972	bufs[1].add = 0;
973	bufs[1].len = 0;
974
975	if (!dead) {
976	return 0;
977	}
978
979	avail = alsa_get_avail (alsa->handle);
980	if (avail < 0) {
981	dolog ("Could not get number of captured frames\n");
982	return 0;
983	}
984
985	if (!avail) {
986	snd_pcm_state_t state;
987	state = snd_pcm_state (alsa->handle);
988	switch (state) {
989	case SND_PCM_STATE_PREPARED:
990	avail = hw->samples;
991	break;
992	case SND_PCM_STATE_SUSPENDED:
993	/* stream is suspended and waiting for an application recovery */
994	if (alsa_resume (alsa->handle)) {
995	#ifndef VBOX
996	dolog ("Failed to resume suspended input stream\n");
997	#else
998	LogRel(("ALSA: Failed to resume input stream\n"));
999	#endif
1000	return 0;
1001	}
1002	if (conf.verbose) {
1003	dolog ("Resuming suspended input stream\n");
1004	}
1005	break;
1006	default:
1007	if (conf.verbose) {
1008	dolog ("No frames available and ALSA state is %d\n", state);
1009	}
1010	return 0;
1011	}
1012	}
1013
1014	decr = audio_MIN (dead, avail);
1015	if (!decr) {
1016	return 0;
1017	}
1018
1019	if (hw->wpos + decr > hw->samples) {
1020	bufs[0].len = (hw->samples - hw->wpos);
1021	bufs[1].len = (decr - (hw->samples - hw->wpos));
1022	}
1023	else {
1024	bufs[0].len = decr;
1025	}
1026
1027	for (i = 0; i < 2; ++i) {
1028	void *src;
1029	st_sample_t *dst;
1030	snd_pcm_sframes_t nread;
1031	snd_pcm_uframes_t len;
1032
1033	len = bufs[i].len;
1034
1035	src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
1036	dst = hw->conv_buf + bufs[i].add;
1037
1038	while (len) {
1039	nread = snd_pcm_readi (alsa->handle, src, len);
1040
1041	if (nread <= 0) {
1042	switch (nread) {
1043	case 0:
1044	if (conf.verbose) {
1045	dolog ("Failed to read %ld frames (read zero)\n", len);
1046	}
1047	goto exit;
1048
1049	case -EPIPE:
1050	if (alsa_recover (alsa->handle)) {
1051	alsa_logerr (nread, "Failed to read %ld frames\n", len);
1052	goto exit;
1053	}
1054	if (conf.verbose) {
1055	dolog ("Recovering from capture xrun\n");
1056	}
1057	continue;
1058
1059	case -EAGAIN:
1060	goto exit;
1061
1062	default:
1063	alsa_logerr (
1064	nread,
1065	"Failed to read %ld frames from %p\n",
1066	len,
1067	src
1068	);
1069	goto exit;
1070	}
1071	}
1072
1073	hw->conv (dst, src, nread, &nominal_volume);
1074
1075	src = advance (src, nread << hwshift);
1076	dst += nread;
1077
1078	read_samples += nread;
1079	len -= nread;
1080	}
1081	}
1082
1083	exit:
1084	hw->wpos = (hw->wpos + read_samples) % hw->samples;
1085	return read_samples;
1086	}
1087
1088	static int alsa_read (SWVoiceIn sw, void buf, int size)
1089	{
1090	return audio_pcm_sw_read (sw, buf, size);
1091	}
1092
1093	static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
1094	{
1095	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
1096
1097	switch (cmd) {
1098	case VOICE_ENABLE:
1099	ldebug ("enabling voice\n");
1100	return alsa_voice_ctl (alsa->handle, "capture", 0);
1101
1102	case VOICE_DISABLE:
1103	ldebug ("disabling voice\n");
1104	return alsa_voice_ctl (alsa->handle, "capture", 1);
1105	}
1106
1107	return -1;
1108	}
1109
1110	#ifdef VBOX
1111	static void alsa_error_handler(const char file, int line, const char function,
1112	int err, const char *fmt, ...)
1113	{
1114	/* ignore */
1115	}
1116	#endif
1117
1118	static void *alsa_audio_init (void)
1119	{
1120	#ifdef VBOX
1121	int rc;
1122
1123	rc = audioLoadAlsaLib();
1124	if (RT_FAILURE(rc)) {
1125	LogRel(("ALSA: Failed to load the ALSA shared library! Error %Rrc\n", rc));
1126	return NULL;
1127	}
1128	snd_lib_error_set_handler (alsa_error_handler);
1129	#endif
1130	return &conf;
1131	}
1132
1133	static void alsa_audio_fini (void *opaque)
1134	{
1135	(void) opaque;
1136	}
1137
1138	static struct audio_option alsa_options[] = {
1139	{"DACSizeInUsec", AUD_OPT_BOOL, &conf.size_in_usec_out,
1140	"DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
1141	{"DACPeriodSize", AUD_OPT_INT, &conf.period_size_out,
1142	"DAC period size", &conf.period_size_out_overriden, 0},
1143	{"DACBufferSize", AUD_OPT_INT, &conf.buffer_size_out,
1144	"DAC buffer size", &conf.buffer_size_out_overriden, 0},
1145
1146	{"ADCSizeInUsec", AUD_OPT_BOOL, &conf.size_in_usec_in,
1147	"ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
1148	{"ADCPeriodSize", AUD_OPT_INT, &conf.period_size_in,
1149	"ADC period size", &conf.period_size_in_overriden, 0},
1150	{"ADCBufferSize", AUD_OPT_INT, &conf.buffer_size_in,
1151	"ADC buffer size", &conf.buffer_size_in_overriden, 0},
1152
1153	{"Threshold", AUD_OPT_INT, &conf.threshold,
1154	"(undocumented)", NULL, 0},
1155
1156	{"DACDev", AUD_OPT_STR, &conf.pcm_name_out,
1157	"DAC device name (for instance dmix)", NULL, 0},
1158
1159	{"ADCDev", AUD_OPT_STR, &conf.pcm_name_in,
1160	"ADC device name", NULL, 0},
1161
1162	{"Verbose", AUD_OPT_BOOL, &conf.verbose,
1163	"Behave in a more verbose way", NULL, 0},
1164
1165	{NULL, 0, NULL, NULL, NULL, 0}
1166	};
1167
1168	static struct audio_pcm_ops alsa_pcm_ops = {
1169	alsa_init_out,
1170	alsa_fini_out,
1171	alsa_run_out,
1172	alsa_write,
1173	alsa_ctl_out,
1174
1175	alsa_init_in,
1176	alsa_fini_in,
1177	alsa_run_in,
1178	alsa_read,
1179	alsa_ctl_in
1180	};
1181
1182	struct audio_driver alsa_audio_driver = {
1183	INIT_FIELD (name = ) "alsa",
1184	INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
1185	INIT_FIELD (options = ) alsa_options,
1186	INIT_FIELD (init = ) alsa_audio_init,
1187	INIT_FIELD (fini = ) alsa_audio_fini,
1188	INIT_FIELD (pcm_ops = ) &alsa_pcm_ops,
1189	INIT_FIELD (can_be_default = ) 1,
1190	INIT_FIELD (max_voices_out = ) INT_MAX,
1191	INIT_FIELD (max_voices_in = ) INT_MAX,
1192	INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
1193	INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)
1194	};

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source: vbox/trunk/src/VBox/Devices/Audio/alsaaudio.c@ 55747

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