alsaaudio.c@ 41371

最後變更在這個檔案從41371是 40844,由 vboxsync 提交於 13 年前
Devices/Audio: Removal of environment variables which acted as audio configuration parameters . Now the audio configuration parameters can be set through SetExtraData. Also, as the configuration parameters are no longer environment variables, there nomenclature
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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	#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	#include <alsa/asoundlib.h>
33
34	#include "VBoxDD.h"
35	#include "vl_vbox.h"
36	#include "audio.h"
37	#ifdef VBOX
38	#include "alsa_stubs.h"
39	#endif
40	#include <iprt/alloc.h>
41
42	#define AUDIO_CAP "alsa"
43	#include "audio_int.h"
44
45	typedef struct ALSAVoiceOut {
46	HWVoiceOut hw;
47	void *pcm_buf;
48	snd_pcm_t *handle;
49	} ALSAVoiceOut;
50
51	typedef struct ALSAVoiceIn {
52	HWVoiceIn hw;
53	snd_pcm_t *handle;
54	void *pcm_buf;
55	} ALSAVoiceIn;
56
57	/* latency = period_size * periods / (rate * bytes_per_frame) */
58
59	static struct {
60	int size_in_usec_in;
61	int size_in_usec_out;
62	const char *pcm_name_in;
63	const char *pcm_name_out;
64	unsigned int buffer_size_in;
65	unsigned int period_size_in;
66	unsigned int buffer_size_out;
67	unsigned int period_size_out;
68	unsigned int threshold;
69
70	int buffer_size_in_overriden;
71	int period_size_in_overriden;
72
73	int buffer_size_out_overriden;
74	int period_size_out_overriden;
75	int verbose;
76	} conf = {
77	#ifdef HIGH_LATENCY
78	INIT_FIELD (.size_in_usec_in =) 1,
79	INIT_FIELD (.size_in_usec_out =) 1,
80	#else
81	INIT_FIELD (.size_in_usec_in =) 0,
82	INIT_FIELD (.size_in_usec_out =) 0,
83	#endif
84	INIT_FIELD (.pcm_name_out =) "default",
85	INIT_FIELD (.pcm_name_in =) "default",
86	#ifdef HIGH_LATENCY
87	INIT_FIELD (.buffer_size_in =) 400000,
88	INIT_FIELD (.period_size_in =) 400000 / 4,
89	INIT_FIELD (.buffer_size_out =) 400000,
90	INIT_FIELD (.period_size_out =) 400000 / 4,
91	#else
92	#define DEFAULT_BUFFER_SIZE 1024
93	#define DEFAULT_PERIOD_SIZE 256
94	INIT_FIELD (.buffer_size_in =) DEFAULT_BUFFER_SIZE * 4,
95	INIT_FIELD (.period_size_in =) DEFAULT_PERIOD_SIZE * 4,
96	INIT_FIELD (.buffer_size_out =) DEFAULT_BUFFER_SIZE,
97	INIT_FIELD (.period_size_out =) DEFAULT_PERIOD_SIZE,
98	#endif
99	INIT_FIELD (.threshold =) 0,
100	INIT_FIELD (.buffer_size_in_overriden =) 0,
101	INIT_FIELD (.period_size_in_overriden =) 0,
102	INIT_FIELD (.buffer_size_out_overriden =) 0,
103	INIT_FIELD (.period_size_out_overriden =) 0,
104	INIT_FIELD (.verbose =) 0
105	};
106
107	struct alsa_params_req {
108	int freq;
109	audfmt_e fmt;
110	int nchannels;
111	unsigned long buffer_size;
112	unsigned long period_size;
113	};
114
115	struct alsa_params_obt {
116	int freq;
117	audfmt_e fmt;
118	int nchannels;
119	snd_pcm_uframes_t samples;
120	};
121
122	static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
123	{
124	va_list ap;
125
126	va_start (ap, fmt);
127	AUD_vlog (AUDIO_CAP, fmt, ap);
128	va_end (ap);
129
130	AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
131	}
132
133	static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
134	int err,
135	const char *typ,
136	const char *fmt,
137	...
138	)
139	{
140	va_list ap;
141
142	AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
143
144	va_start (ap, fmt);
145	AUD_vlog (AUDIO_CAP, fmt, ap);
146	va_end (ap);
147
148	AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
149	}
150
151	static void alsa_anal_close (snd_pcm_t **handlep)
152	{
153	int err = snd_pcm_close (*handlep);
154	if (err) {
155	alsa_logerr (err, "Failed to close PCM handle %p\n",
156	(void ) handlep);
157	}
158	*handlep = NULL;
159	}
160
161	static int alsa_write (SWVoiceOut sw, void buf, int len)
162	{
163	return audio_pcm_sw_write (sw, buf, len);
164	}
165
166	static int aud_to_alsafmt (audfmt_e fmt)
167	{
168	switch (fmt) {
169	case AUD_FMT_S8:
170	return SND_PCM_FORMAT_S8;
171
172	case AUD_FMT_U8:
173	return SND_PCM_FORMAT_U8;
174
175	case AUD_FMT_S16:
176	return SND_PCM_FORMAT_S16_LE;
177
178	case AUD_FMT_U16:
179	return SND_PCM_FORMAT_U16_LE;
180
181	case AUD_FMT_S32:
182	return SND_PCM_FORMAT_S32_LE;
183
184	case AUD_FMT_U32:
185	return SND_PCM_FORMAT_U32_LE;
186
187	default:
188	dolog ("Internal logic error: Bad audio format %d\n", fmt);
189	#ifdef DEBUG_AUDIO
190	abort ();
191	#endif
192	return SND_PCM_FORMAT_U8;
193	}
194	}
195
196	static int alsa_to_audfmt (int alsafmt, audfmt_e fmt, int endianness)
197	{
198	switch (alsafmt) {
199	case SND_PCM_FORMAT_S8:
200	*endianness = 0;
201	*fmt = AUD_FMT_S8;
202	break;
203
204	case SND_PCM_FORMAT_U8:
205	*endianness = 0;
206	*fmt = AUD_FMT_U8;
207	break;
208
209	case SND_PCM_FORMAT_S16_LE:
210	*endianness = 0;
211	*fmt = AUD_FMT_S16;
212	break;
213
214	case SND_PCM_FORMAT_U16_LE:
215	*endianness = 0;
216	*fmt = AUD_FMT_U16;
217	break;
218
219	case SND_PCM_FORMAT_S16_BE:
220	*endianness = 1;
221	*fmt = AUD_FMT_S16;
222	break;
223
224	case SND_PCM_FORMAT_U16_BE:
225	*endianness = 1;
226	*fmt = AUD_FMT_U16;
227	break;
228
229	case SND_PCM_FORMAT_S32_LE:
230	*endianness = 0;
231	*fmt = AUD_FMT_S32;
232	break;
233
234	case SND_PCM_FORMAT_U32_LE:
235	*endianness = 0;
236	*fmt = AUD_FMT_U32;
237	break;
238
239	case SND_PCM_FORMAT_S32_BE:
240	*endianness = 1;
241	*fmt = AUD_FMT_S32;
242	break;
243
244	case SND_PCM_FORMAT_U32_BE:
245	*endianness = 1;
246	*fmt = AUD_FMT_U32;
247	break;
248
249	default:
250	dolog ("Unrecognized audio format %d\n", alsafmt);
251	return -1;
252	}
253
254	return 0;
255	}
256
257	#if defined DEBUG_MISMATCHES \|\| defined DEBUG
258	static void alsa_dump_info (struct alsa_params_req *req,
259	struct alsa_params_obt *obt)
260	{
261	dolog ("parameter \| requested value \| obtained value\n");
262	dolog ("format \| %10d \| %10d\n", req->fmt, obt->fmt);
263	dolog ("channels \| %10d \| %10d\n",
264	req->nchannels, obt->nchannels);
265	dolog ("frequency \| %10d \| %10d\n", req->freq, obt->freq);
266	dolog ("============================================\n");
267	dolog ("requested: buffer size %d period size %d\n",
268	req->buffer_size, req->period_size);
269	dolog ("obtained: samples %ld\n", obt->samples);
270	}
271	#endif
272
273	static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
274	{
275	int err;
276	snd_pcm_sw_params_t *sw_params;
277
278	snd_pcm_sw_params_alloca (&sw_params);
279
280	err = snd_pcm_sw_params_current (handle, sw_params);
281	if (err < 0) {
282	dolog ("Could not fully initialize DAC\n");
283	alsa_logerr (err, "Failed to get current software parameters\n");
284	return;
285	}
286
287	err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
288	if (err < 0) {
289	dolog ("Could not fully initialize DAC\n");
290	alsa_logerr (err, "Failed to set software threshold to %ld\n",
291	threshold);
292	return;
293	}
294
295	err = snd_pcm_sw_params (handle, sw_params);
296	if (err < 0) {
297	dolog ("Could not fully initialize DAC\n");
298	alsa_logerr (err, "Failed to set software parameters\n");
299	return;
300	}
301	}
302
303	static int alsa_open (int in, struct alsa_params_req *req,
304	struct alsa_params_obt obt, snd_pcm_t *handlep)
305	{
306	snd_pcm_t *handle;
307	snd_pcm_hw_params_t *hw_params;
308	int err, dir;
309	unsigned int freq, nchannels;
310	const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
311	unsigned int period_size, buffer_size;
312	snd_pcm_uframes_t period_size_f, buffer_size_f;
313	snd_pcm_uframes_t obt_buffer_size, obt_period_size;
314	const char *typ = in ? "ADC" : "DAC";
315
316	freq = req->freq;
317	period_size = req->period_size;
318	buffer_size = req->buffer_size;
319	period_size_f = (snd_pcm_uframes_t)period_size;
320	buffer_size_f = (snd_pcm_uframes_t)buffer_size;
321	nchannels = req->nchannels;
322
323	snd_pcm_hw_params_alloca (&hw_params);
324
325	err = snd_pcm_open (
326	&handle,
327	pcm_name,
328	in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
329	SND_PCM_NONBLOCK
330	);
331	if (err < 0) {
332	#ifndef VBOX
333	alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
334	#else
335	LogRel(("ALSA: Failed to open '%s' as %s\n", pcm_name, typ));
336	#endif
337	return -1;
338	}
339
340	err = snd_pcm_hw_params_any (handle, hw_params);
341	if (err < 0) {
342	#ifndef VBOX
343	alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
344	#else
345	LogRel(("ALSA: Failed to initialize hardware parameters\n"));
346	#endif
347	goto err;
348	}
349
350	err = snd_pcm_hw_params_set_access (
351	handle,
352	hw_params,
353	SND_PCM_ACCESS_RW_INTERLEAVED
354	);
355	if (err < 0) {
356	#ifndef VBOX
357	alsa_logerr2 (err, typ, "Failed to set access type\n");
358	#else
359	LogRel(("ALSA: Failed to set access type\n"));
360	#endif
361	goto err;
362	}
363
364	err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
365	if (err < 0) {
366	#ifndef VBOX
367	alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
368	#else
369	LogRel(("ALSA: Failed to set format %d\n", req->fmt));
370	#endif
371	goto err;
372	}
373
374	err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
375	if (err < 0) {
376	#ifndef VBOX
377	alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
378	#else
379	LogRel(("ALSA: Failed to set frequency %dHz\n", req->freq));
380	#endif
381	goto err;
382	}
383
384	err = snd_pcm_hw_params_set_channels_near (
385	handle,
386	hw_params,
387	&nchannels
388	);
389	if (err < 0) {
390	#ifndef VBOX
391	alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
392	req->nchannels);
393	#else
394	LogRel(("ALSA: Failed to set number of channels to %d\n", req->nchannels));
395	#endif
396	goto err;
397	}
398
399	if (nchannels != 1 && nchannels != 2) {
400	#ifndef VBOX
401	alsa_logerr2 (err, typ,
402	"Can not handle obtained number of channels %d\n",
403	nchannels);
404	#else
405	LogRel(("ALSA: Cannot handle obtained number of channels (%d)\n", nchannels));
406	#endif
407	goto err;
408	}
409
410	if (!((in && conf.size_in_usec_in) \|\| (!in && conf.size_in_usec_out))) {
411	if (!buffer_size) {
412	buffer_size = DEFAULT_BUFFER_SIZE;
413	period_size= DEFAULT_PERIOD_SIZE;
414	}
415	}
416
417	if (buffer_size) {
418	if ((in && conf.size_in_usec_in) \|\| (!in && conf.size_in_usec_out)) {
419	if (period_size) {
420	err = snd_pcm_hw_params_set_period_time_near (
421	handle,
422	hw_params,
423	&period_size,
424	0
425	);
426	if (err < 0) {
427	#ifndef VBOX
428	alsa_logerr2 (err, typ,
429	"Failed to set period time %d\n",
430	req->period_size);
431	#else
432	LogRel(("ALSA: Failed to set period time %d\n", req->period_size));
433	#endif
434	goto err;
435	}
436	}
437
438	err = snd_pcm_hw_params_set_buffer_time_near (
439	handle,
440	hw_params,
441	&buffer_size,
442	0
443	);
444
445	if (err < 0) {
446	#ifndef VBOX
447	alsa_logerr2 (err, typ,
448	"Failed to set buffer time %d\n",
449	req->buffer_size);
450	#else
451	LogRel(("ALSA: Failed to set buffer time %d\n", req->buffer_size));
452	#endif
453	goto err;
454	}
455	}
456	else {
457	snd_pcm_uframes_t minval;
458
459	if (period_size_f) {
460	minval = period_size_f;
461	dir = 0;
462
463	err = snd_pcm_hw_params_get_period_size_min (
464	hw_params,
465	&minval,
466	&dir
467	);
468	if (err < 0) {
469	#ifndef VBOX
470	alsa_logerr (
471	err,
472	"Could not get minimal period size for %s\n",
473	typ
474	);
475	#else
476	LogRel(("ALSA: Could not get minimal period size for %s\n", typ));
477	#endif
478	}
479	else {
480	dolog("minimal period size %ld\n", minval);
481	if (period_size_f < minval) {
482	if ((in && conf.period_size_in_overriden)
483	\|\| (!in && conf.period_size_out_overriden)) {
484	dolog ("%s period size(%d) is less "
485	"than minimal period size(%ld)\n",
486	typ,
487	period_size_f,
488	minval);
489	}
490	period_size_f = minval;
491	}
492	}
493
494	#ifndef VBOX
495	err = snd_pcm_hw_params_set_period_size (
496	handle,
497	hw_params,
498	period_size_f,
499	0
500	);
501	#else
502	err = snd_pcm_hw_params_set_period_size_near (
503	handle,
504	hw_params,
505	&period_size_f,
506	0
507	);
508	#endif
509	dolog("PERIOD_SIZE %d\n", period_size_f);
510	if (err < 0) {
511	#ifndef VBOX
512	alsa_logerr2 (err, typ, "Failed to set period size %d\n",
513	period_size_f);
514	#else
515	LogRel(("ALSA: Failed to set period size %d (%s)\n",
516	period_size_f, snd_strerror(err)));
517	#endif
518	goto err;
519	}
520	}
521
522	#ifdef VBOX
523	/* Calculate default buffer size here since it might have been changed
524	* in the _near functions */
525	buffer_size_f = 4 * period_size_f;
526	#endif
527
528	minval = buffer_size_f;
529	err = snd_pcm_hw_params_get_buffer_size_min (
530	hw_params,
531	&minval
532	);
533	if (err < 0) {
534	#ifndef VBOX
535	alsa_logerr (err, "Could not get minimal buffer size for %s\n",
536	typ);
537	#else
538	LogRel(("ALSA: Could not get minimal buffer size for %s\n", typ));
539	#endif
540	}
541	else {
542	if (buffer_size_f < minval) {
543	if ((in && conf.buffer_size_in_overriden)
544	\|\| (!in && conf.buffer_size_out_overriden)) {
545	dolog (
546	"%s buffer size(%d) is less "
547	"than minimal buffer size(%ld)\n",
548	typ,
549	buffer_size_f,
550	minval
551	);
552	}
553	buffer_size_f = minval;
554	}
555	}
556
557	err = snd_pcm_hw_params_set_buffer_size_near (
558	handle,
559	hw_params,
560	&buffer_size_f
561	);
562	dolog("BUFFER_SIZE %d\n", buffer_size_f);
563	if (err < 0) {
564	#ifndef VBOX
565	alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
566	buffer_size_f);
567	#else
568	LogRel(("ALSA: Failed to set buffer size %d (%s)\n",
569	buffer_size_f, snd_strerror(err)));
570	#endif
571	goto err;
572	}
573	}
574	}
575	else {
576	dolog ("warning: Buffer size is not set\n");
577	}
578
579	err = snd_pcm_hw_params (handle, hw_params);
580	if (err < 0) {
581	#ifndef VBOX
582	alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
583	#else
584	LogRel(("ALSA: Failed to apply audio parameters\n"));
585	#endif
586	goto err;
587	}
588
589	err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
590	if (err < 0) {
591	#ifndef VBOX
592	alsa_logerr2 (err, typ, "Failed to get buffer size\n");
593	#else
594	LogRel(("ALSA: Failed to get buffer size\n"));
595	#endif
596	goto err;
597	}
598
599	#ifdef VBOX
600	dir = 0;
601	err = snd_pcm_hw_params_get_period_size (hw_params, &obt_period_size, &dir);
602	if (err < 0)
603	{
604	LogRel(("ALSA: Failed to get period size\n"));
605	goto err;
606	}
607	LogRel(("ALSA: %s frequency %dHz, period size %ld, buffer size %ld\n",
608	typ, req->freq, obt_period_size, obt_buffer_size));
609	#endif
610
611	err = snd_pcm_prepare (handle);
612	if (err < 0) {
613	alsa_logerr2 (err, typ, "Could not prepare handle %p\n",
614	(void *) handle);
615	goto err;
616	}
617
618	if (!in && conf.threshold) {
619	snd_pcm_uframes_t threshold;
620	int bytes_per_sec;
621
622	bytes_per_sec = freq
623	<< (nchannels == 2)
624	<< (req->fmt == AUD_FMT_S16 \|\| req->fmt == AUD_FMT_U16);
625
626	threshold = (conf.threshold * bytes_per_sec) / 1000;
627	alsa_set_threshold (handle, threshold);
628	}
629
630	obt->fmt = req->fmt;
631	obt->nchannels = nchannels;
632	obt->freq = freq;
633	obt->samples = obt_buffer_size;
634	*handlep = handle;
635
636	#if defined DEBUG_MISMATCHES \|\| defined DEBUG
637	if (obt->fmt != req->fmt \|\|
638	obt->nchannels != req->nchannels \|\|
639	obt->freq != req->freq) {
640	dolog ("Audio parameters mismatch for %s\n", typ);
641	alsa_dump_info (req, obt);
642	}
643	#endif
644
645	#ifdef DEBUG
646	alsa_dump_info (req, obt);
647	#endif
648	return 0;
649
650	err:
651	alsa_anal_close (&handle);
652	return -1;
653	}
654
655	static int alsa_recover (snd_pcm_t *handle)
656	{
657	int err = snd_pcm_prepare (handle);
658	if (err < 0) {
659	alsa_logerr (err, "Failed to prepare handle %p\n",
660	(void *) handle);
661	return -1;
662	}
663	return 0;
664	}
665
666	static int alsa_resume (snd_pcm_t *handle)
667	{
668	int err = snd_pcm_resume (handle);
669	if (err < 0) {
670	#ifndef VBOX
671	alsa_logerr (err, "Failed to resume handle %p\n", handle);
672	#endif
673	return -1;
674	}
675	return 0;
676	}
677
678	static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
679	{
680	snd_pcm_sframes_t avail;
681
682	avail = snd_pcm_avail_update (handle);
683	if (avail < 0) {
684	if (avail == -EPIPE) {
685	if (!alsa_recover (handle)) {
686	avail = snd_pcm_avail_update (handle);
687	}
688	}
689
690	if (avail < 0) {
691	alsa_logerr (avail,
692	"Could not obtain number of available frames\n");
693	return -1;
694	}
695	}
696
697	return avail;
698	}
699
700	static int alsa_run_out (HWVoiceOut *hw)
701	{
702	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
703	int rpos, live, decr;
704	int samples;
705	uint8_t *dst;
706	st_sample_t *src;
707	snd_pcm_sframes_t avail;
708
709	live = audio_pcm_hw_get_live_out (hw);
710	if (!live) {
711	return 0;
712	}
713
714	avail = alsa_get_avail (alsa->handle);
715	if (avail < 0) {
716	dolog ("Could not get number of available playback frames\n");
717	return 0;
718	}
719
720	decr = audio_MIN (live, avail);
721	samples = decr;
722	rpos = hw->rpos;
723	while (samples) {
724	int left_till_end_samples = hw->samples - rpos;
725	int len = audio_MIN (samples, left_till_end_samples);
726	snd_pcm_sframes_t written;
727
728	src = hw->mix_buf + rpos;
729	dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
730
731	hw->clip (dst, src, len);
732
733	while (len) {
734	written = snd_pcm_writei (alsa->handle, dst, len);
735
736	if (written <= 0) {
737	switch (written) {
738	case 0:
739	if (conf.verbose) {
740	dolog ("Failed to write %d frames (wrote zero)\n", len);
741	}
742	goto exit;
743
744	case -EPIPE:
745	if (alsa_recover (alsa->handle)) {
746	alsa_logerr (written, "Failed to write %d frames\n",
747	len);
748	goto exit;
749	}
750	if (conf.verbose) {
751	dolog ("Recovering from playback xrun\n");
752	}
753	continue;
754
755	case -ESTRPIPE:
756	/* stream is suspended and waiting for an
757	application recovery */
758	if (alsa_resume (alsa->handle)) {
759	#ifndef VBOX
760	alsa_logerr (written, "Failed to write %d frames\n", len);
761	#else
762	LogRel(("ALSA: Failed to resume output stream\n"));
763	#endif
764	goto exit;
765	}
766	if (conf.verbose) {
767	dolog ("Resuming suspended output stream\n");
768	}
769	continue;
770
771	case -EAGAIN:
772	goto exit;
773
774	default:
775	alsa_logerr (written, "Failed to write %d frames to %p\n",
776	len, dst);
777	goto exit;
778	}
779	}
780
781	rpos = (rpos + written) % hw->samples;
782	samples -= written;
783	len -= written;
784	dst = advance (dst, written << hw->info.shift);
785	src += written;
786	}
787	}
788
789	exit:
790	hw->rpos = rpos;
791	return decr;
792	}
793
794	static void alsa_fini_out (HWVoiceOut *hw)
795	{
796	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
797
798	ldebug ("alsa_fini\n");
799	alsa_anal_close (&alsa->handle);
800
801	if (alsa->pcm_buf) {
802	qemu_free (alsa->pcm_buf);
803	alsa->pcm_buf = NULL;
804	}
805	}
806
807	static int alsa_init_out (HWVoiceOut hw, audsettings_t as)
808	{
809	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
810	struct alsa_params_req req;
811	struct alsa_params_obt obt;
812	audfmt_e effective_fmt;
813	int endianness;
814	int err;
815	snd_pcm_t *handle;
816	audsettings_t obt_as;
817
818	req.fmt = aud_to_alsafmt (as->fmt);
819	req.freq = as->freq;
820	req.nchannels = as->nchannels;
821	req.period_size = conf.period_size_out;
822	req.buffer_size = conf.buffer_size_out;
823
824	if (alsa_open (0, &req, &obt, &handle)) {
825	return -1;
826	}
827
828	err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
829	if (err) {
830	alsa_anal_close (&handle);
831	return -1;
832	}
833
834	obt_as.freq = obt.freq;
835	obt_as.nchannels = obt.nchannels;
836	obt_as.fmt = effective_fmt;
837	obt_as.endianness = endianness;
838
839	audio_pcm_init_info (&hw->info, &obt_as);
840	hw->samples = obt.samples;
841
842	alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
843	if (!alsa->pcm_buf) {
844	dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
845	hw->samples, 1 << hw->info.shift);
846	alsa_anal_close (&handle);
847	return -1;
848	}
849
850	alsa->handle = handle;
851	return 0;
852	}
853
854	static int alsa_voice_ctl (snd_pcm_t handle, const char typ, int pauseit) /* VBOX: s/pause/pauseit/; -Wshadow */
855	{
856	int err;
857
858	if (pauseit) {
859	err = snd_pcm_drop (handle);
860	if (err < 0) {
861	alsa_logerr (err, "Could not stop %s\n", typ);
862	return -1;
863	}
864	}
865	else {
866	err = snd_pcm_prepare (handle);
867	if (err < 0) {
868	alsa_logerr (err, "Could not prepare handle for %s\n", typ);
869	return -1;
870	}
871	}
872
873	return 0;
874	}
875
876	static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
877	{
878	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
879
880	switch (cmd) {
881	case VOICE_ENABLE:
882	ldebug ("enabling voice\n");
883	return alsa_voice_ctl (alsa->handle, "playback", 0);
884
885	case VOICE_DISABLE:
886	ldebug ("disabling voice\n");
887	return alsa_voice_ctl (alsa->handle, "playback", 1);
888	}
889
890	return -1;
891	}
892
893	static int alsa_init_in (HWVoiceIn hw, audsettings_t as)
894	{
895	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
896	struct alsa_params_req req;
897	struct alsa_params_obt obt;
898	int endianness;
899	int err;
900	audfmt_e effective_fmt;
901	snd_pcm_t *handle;
902	audsettings_t obt_as;
903
904	req.fmt = aud_to_alsafmt (as->fmt);
905	req.freq = as->freq;
906	req.nchannels = as->nchannels;
907	req.period_size = conf.period_size_in;
908	req.buffer_size = conf.buffer_size_in;
909
910	if (alsa_open (1, &req, &obt, &handle)) {
911	return -1;
912	}
913
914	err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
915	if (err) {
916	alsa_anal_close (&handle);
917	return -1;
918	}
919
920	obt_as.freq = obt.freq;
921	obt_as.nchannels = obt.nchannels;
922	obt_as.fmt = effective_fmt;
923	obt_as.endianness = endianness;
924
925	audio_pcm_init_info (&hw->info, &obt_as);
926	hw->samples = obt.samples;
927
928	alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
929	if (!alsa->pcm_buf) {
930	dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
931	hw->samples, 1 << hw->info.shift);
932	alsa_anal_close (&handle);
933	return -1;
934	}
935
936	alsa->handle = handle;
937	return 0;
938	}
939
940	static void alsa_fini_in (HWVoiceIn *hw)
941	{
942	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
943
944	alsa_anal_close (&alsa->handle);
945
946	if (alsa->pcm_buf) {
947	qemu_free (alsa->pcm_buf);
948	alsa->pcm_buf = NULL;
949	}
950	}
951
952	static int alsa_run_in (HWVoiceIn *hw)
953	{
954	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
955	int hwshift = hw->info.shift;
956	int i;
957	int live = audio_pcm_hw_get_live_in (hw);
958	int dead = hw->samples - live;
959	int decr;
960	struct {
961	int add;
962	int len;
963	} bufs[2];
964
965	snd_pcm_sframes_t avail;
966	snd_pcm_uframes_t read_samples = 0;
967
968	bufs[0].add = hw->wpos;
969	bufs[0].len = 0;
970	bufs[1].add = 0;
971	bufs[1].len = 0;
972
973	if (!dead) {
974	return 0;
975	}
976
977	avail = alsa_get_avail (alsa->handle);
978	if (avail < 0) {
979	dolog ("Could not get number of captured frames\n");
980	return 0;
981	}
982
983	if (!avail) {
984	snd_pcm_state_t state;
985	state = snd_pcm_state (alsa->handle);
986	switch (state) {
987	case SND_PCM_STATE_PREPARED:
988	avail = hw->samples;
989	break;
990	case SND_PCM_STATE_SUSPENDED:
991	/* stream is suspended and waiting for an application recovery */
992	if (alsa_resume (alsa->handle)) {
993	#ifndef VBOX
994	dolog ("Failed to resume suspended input stream\n");
995	#else
996	LogRel(("ALSA: Failed to resume input stream\n"));
997	#endif
998	return 0;
999	}
1000	if (conf.verbose) {
1001	dolog ("Resuming suspended input stream\n");
1002	}
1003	break;
1004	default:
1005	if (conf.verbose) {
1006	dolog ("No frames available and ALSA state is %d\n", state);
1007	}
1008	return 0;
1009	}
1010	}
1011
1012	decr = audio_MIN (dead, avail);
1013	if (!decr) {
1014	return 0;
1015	}
1016
1017	if (hw->wpos + decr > hw->samples) {
1018	bufs[0].len = (hw->samples - hw->wpos);
1019	bufs[1].len = (decr - (hw->samples - hw->wpos));
1020	}
1021	else {
1022	bufs[0].len = decr;
1023	}
1024
1025	for (i = 0; i < 2; ++i) {
1026	void *src;
1027	st_sample_t *dst;
1028	snd_pcm_sframes_t nread;
1029	snd_pcm_uframes_t len;
1030
1031	len = bufs[i].len;
1032
1033	src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
1034	dst = hw->conv_buf + bufs[i].add;
1035
1036	while (len) {
1037	nread = snd_pcm_readi (alsa->handle, src, len);
1038
1039	if (nread <= 0) {
1040	switch (nread) {
1041	case 0:
1042	if (conf.verbose) {
1043	dolog ("Failed to read %ld frames (read zero)\n", len);
1044	}
1045	goto exit;
1046
1047	case -EPIPE:
1048	if (alsa_recover (alsa->handle)) {
1049	alsa_logerr (nread, "Failed to read %ld frames\n", len);
1050	goto exit;
1051	}
1052	if (conf.verbose) {
1053	dolog ("Recovering from capture xrun\n");
1054	}
1055	continue;
1056
1057	case -EAGAIN:
1058	goto exit;
1059
1060	default:
1061	alsa_logerr (
1062	nread,
1063	"Failed to read %ld frames from %p\n",
1064	len,
1065	src
1066	);
1067	goto exit;
1068	}
1069	}
1070
1071	hw->conv (dst, src, nread, &nominal_volume);
1072
1073	src = advance (src, nread << hwshift);
1074	dst += nread;
1075
1076	read_samples += nread;
1077	len -= nread;
1078	}
1079	}
1080
1081	exit:
1082	hw->wpos = (hw->wpos + read_samples) % hw->samples;
1083	return read_samples;
1084	}
1085
1086	static int alsa_read (SWVoiceIn sw, void buf, int size)
1087	{
1088	return audio_pcm_sw_read (sw, buf, size);
1089	}
1090
1091	static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
1092	{
1093	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
1094
1095	switch (cmd) {
1096	case VOICE_ENABLE:
1097	ldebug ("enabling voice\n");
1098	return alsa_voice_ctl (alsa->handle, "capture", 0);
1099
1100	case VOICE_DISABLE:
1101	ldebug ("disabling voice\n");
1102	return alsa_voice_ctl (alsa->handle, "capture", 1);
1103	}
1104
1105	return -1;
1106	}
1107
1108	#ifdef VBOX
1109	static void alsa_error_handler(const char file, int line, const char function,
1110	int err, const char *fmt, ...)
1111	{
1112	/* ignore */
1113	}
1114	#endif
1115
1116	static void *alsa_audio_init (void)
1117	{
1118	#ifdef VBOX
1119	int rc;
1120
1121	rc = audioLoadAlsaLib();
1122	if (RT_FAILURE(rc)) {
1123	LogRel(("ALSA: Failed to load the ALSA shared library! Error %Rrc\n", rc));
1124	return NULL;
1125	}
1126	snd_lib_error_set_handler (alsa_error_handler);
1127	#endif
1128	return &conf;
1129	}
1130
1131	static void alsa_audio_fini (void *opaque)
1132	{
1133	(void) opaque;
1134	}
1135
1136	static struct audio_option alsa_options[] = {
1137	{"DACSizeInUsec", AUD_OPT_BOOL, &conf.size_in_usec_out,
1138	"DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
1139	{"DACPeriodSize", AUD_OPT_INT, &conf.period_size_out,
1140	"DAC period size", &conf.period_size_out_overriden, 0},
1141	{"DACBufferSize", AUD_OPT_INT, &conf.buffer_size_out,
1142	"DAC buffer size", &conf.buffer_size_out_overriden, 0},
1143
1144	{"ADCSizeInUsec", AUD_OPT_BOOL, &conf.size_in_usec_in,
1145	"ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
1146	{"ADCPeriodSize", AUD_OPT_INT, &conf.period_size_in,
1147	"ADC period size", &conf.period_size_in_overriden, 0},
1148	{"ADCBufferSize", AUD_OPT_INT, &conf.buffer_size_in,
1149	"ADC buffer size", &conf.buffer_size_in_overriden, 0},
1150
1151	{"Threshold", AUD_OPT_INT, &conf.threshold,
1152	"(undocumented)", NULL, 0},
1153
1154	{"DACDev", AUD_OPT_STR, &conf.pcm_name_out,
1155	"DAC device name (for instance dmix)", NULL, 0},
1156
1157	{"ADCDev", AUD_OPT_STR, &conf.pcm_name_in,
1158	"ADC device name", NULL, 0},
1159
1160	{"Verbose", AUD_OPT_BOOL, &conf.verbose,
1161	"Behave in a more verbose way", NULL, 0},
1162
1163	{NULL, 0, NULL, NULL, NULL, 0}
1164	};
1165
1166	static struct audio_pcm_ops alsa_pcm_ops = {
1167	alsa_init_out,
1168	alsa_fini_out,
1169	alsa_run_out,
1170	alsa_write,
1171	alsa_ctl_out,
1172
1173	alsa_init_in,
1174	alsa_fini_in,
1175	alsa_run_in,
1176	alsa_read,
1177	alsa_ctl_in
1178	};
1179
1180	struct audio_driver alsa_audio_driver = {
1181	INIT_FIELD (name = ) "alsa",
1182	INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
1183	INIT_FIELD (options = ) alsa_options,
1184	INIT_FIELD (init = ) alsa_audio_init,
1185	INIT_FIELD (fini = ) alsa_audio_fini,
1186	INIT_FIELD (pcm_ops = ) &alsa_pcm_ops,
1187	INIT_FIELD (can_be_default = ) 1,
1188	INIT_FIELD (max_voices_out = ) INT_MAX,
1189	INIT_FIELD (max_voices_in = ) INT_MAX,
1190	INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
1191	INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)
1192	};

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source: vbox/trunk/src/VBox/Devices/Audio/alsaaudio.c@ 41371

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