alsaaudio.c@ 12677

最後變更在這個檔案從12677是 6077,由 vboxsync 提交於 17 年前
pulse/alsa: resolve _all_ symbols during initialization, otherwise incompatible libraries might confuse the sound driver much later during runtime
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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 "Builtins.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 minmal 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 minmal 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 minmal 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 paramters 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 snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
667	{
668	snd_pcm_sframes_t avail;
669
670	avail = snd_pcm_avail_update (handle);
671	if (avail < 0) {
672	if (avail == -EPIPE) {
673	if (!alsa_recover (handle)) {
674	avail = snd_pcm_avail_update (handle);
675	}
676	}
677
678	if (avail < 0) {
679	alsa_logerr (avail,
680	"Could not obtain number of available frames\n");
681	return -1;
682	}
683	}
684
685	return avail;
686	}
687
688	static int alsa_run_out (HWVoiceOut *hw)
689	{
690	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
691	int rpos, live, decr;
692	int samples;
693	uint8_t *dst;
694	st_sample_t *src;
695	snd_pcm_sframes_t avail;
696
697	live = audio_pcm_hw_get_live_out (hw);
698	if (!live) {
699	return 0;
700	}
701
702	avail = alsa_get_avail (alsa->handle);
703	if (avail < 0) {
704	dolog ("Could not get number of available playback frames\n");
705	return 0;
706	}
707
708	decr = audio_MIN (live, avail);
709	samples = decr;
710	rpos = hw->rpos;
711	while (samples) {
712	int left_till_end_samples = hw->samples - rpos;
713	int len = audio_MIN (samples, left_till_end_samples);
714	snd_pcm_sframes_t written;
715
716	src = hw->mix_buf + rpos;
717	dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
718
719	hw->clip (dst, src, len);
720
721	while (len) {
722	written = snd_pcm_writei (alsa->handle, dst, len);
723
724	if (written <= 0) {
725	switch (written) {
726	case 0:
727	if (conf.verbose) {
728	dolog ("Failed to write %d frames (wrote zero)\n", len);
729	}
730	goto exit;
731
732	case -EPIPE:
733	if (alsa_recover (alsa->handle)) {
734	alsa_logerr (written, "Failed to write %d frames\n",
735	len);
736	goto exit;
737	}
738	if (conf.verbose) {
739	dolog ("Recovering from playback xrun\n");
740	}
741	continue;
742
743	case -EAGAIN:
744	goto exit;
745
746	default:
747	alsa_logerr (written, "Failed to write %d frames to %p\n",
748	len, dst);
749	goto exit;
750	}
751	}
752
753	rpos = (rpos + written) % hw->samples;
754	samples -= written;
755	len -= written;
756	dst = advance (dst, written << hw->info.shift);
757	src += written;
758	}
759	}
760
761	exit:
762	hw->rpos = rpos;
763	return decr;
764	}
765
766	static void alsa_fini_out (HWVoiceOut *hw)
767	{
768	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
769
770	ldebug ("alsa_fini\n");
771	alsa_anal_close (&alsa->handle);
772
773	if (alsa->pcm_buf) {
774	qemu_free (alsa->pcm_buf);
775	alsa->pcm_buf = NULL;
776	}
777	}
778
779	static int alsa_init_out (HWVoiceOut hw, audsettings_t as)
780	{
781	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
782	struct alsa_params_req req;
783	struct alsa_params_obt obt;
784	audfmt_e effective_fmt;
785	int endianness;
786	int err;
787	snd_pcm_t *handle;
788	audsettings_t obt_as;
789
790	req.fmt = aud_to_alsafmt (as->fmt);
791	req.freq = as->freq;
792	req.nchannels = as->nchannels;
793	req.period_size = conf.period_size_out;
794	req.buffer_size = conf.buffer_size_out;
795
796	if (alsa_open (0, &req, &obt, &handle)) {
797	return -1;
798	}
799
800	err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
801	if (err) {
802	alsa_anal_close (&handle);
803	return -1;
804	}
805
806	obt_as.freq = obt.freq;
807	obt_as.nchannels = obt.nchannels;
808	obt_as.fmt = effective_fmt;
809	obt_as.endianness = endianness;
810
811	audio_pcm_init_info (&hw->info, &obt_as);
812	hw->samples = obt.samples;
813
814	alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
815	if (!alsa->pcm_buf) {
816	dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
817	hw->samples, 1 << hw->info.shift);
818	alsa_anal_close (&handle);
819	return -1;
820	}
821
822	alsa->handle = handle;
823	return 0;
824	}
825
826	static int alsa_voice_ctl (snd_pcm_t handle, const char typ, int pause)
827	{
828	int err;
829
830	if (pause) {
831	err = snd_pcm_drop (handle);
832	if (err < 0) {
833	alsa_logerr (err, "Could not stop %s\n", typ);
834	return -1;
835	}
836	}
837	else {
838	err = snd_pcm_prepare (handle);
839	if (err < 0) {
840	alsa_logerr (err, "Could not prepare handle for %s\n", typ);
841	return -1;
842	}
843	}
844
845	return 0;
846	}
847
848	static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
849	{
850	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
851
852	switch (cmd) {
853	case VOICE_ENABLE:
854	ldebug ("enabling voice\n");
855	return alsa_voice_ctl (alsa->handle, "playback", 0);
856
857	case VOICE_DISABLE:
858	ldebug ("disabling voice\n");
859	return alsa_voice_ctl (alsa->handle, "playback", 1);
860	}
861
862	return -1;
863	}
864
865	static int alsa_init_in (HWVoiceIn hw, audsettings_t as)
866	{
867	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
868	struct alsa_params_req req;
869	struct alsa_params_obt obt;
870	int endianness;
871	int err;
872	audfmt_e effective_fmt;
873	snd_pcm_t *handle;
874	audsettings_t obt_as;
875
876	req.fmt = aud_to_alsafmt (as->fmt);
877	req.freq = as->freq;
878	req.nchannels = as->nchannels;
879	req.period_size = conf.period_size_in;
880	req.buffer_size = conf.buffer_size_in;
881
882	if (alsa_open (1, &req, &obt, &handle)) {
883	return -1;
884	}
885
886	err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
887	if (err) {
888	alsa_anal_close (&handle);
889	return -1;
890	}
891
892	obt_as.freq = obt.freq;
893	obt_as.nchannels = obt.nchannels;
894	obt_as.fmt = effective_fmt;
895	obt_as.endianness = endianness;
896
897	audio_pcm_init_info (&hw->info, &obt_as);
898	hw->samples = obt.samples;
899
900	alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
901	if (!alsa->pcm_buf) {
902	dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
903	hw->samples, 1 << hw->info.shift);
904	alsa_anal_close (&handle);
905	return -1;
906	}
907
908	alsa->handle = handle;
909	return 0;
910	}
911
912	static void alsa_fini_in (HWVoiceIn *hw)
913	{
914	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
915
916	alsa_anal_close (&alsa->handle);
917
918	if (alsa->pcm_buf) {
919	qemu_free (alsa->pcm_buf);
920	alsa->pcm_buf = NULL;
921	}
922	}
923
924	static int alsa_run_in (HWVoiceIn *hw)
925	{
926	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
927	int hwshift = hw->info.shift;
928	int i;
929	int live = audio_pcm_hw_get_live_in (hw);
930	int dead = hw->samples - live;
931	int decr;
932	struct {
933	int add;
934	int len;
935	} bufs[2];
936
937	snd_pcm_sframes_t avail;
938	snd_pcm_uframes_t read_samples = 0;
939
940	bufs[0].add = hw->wpos;
941	bufs[0].len = 0;
942	bufs[1].add = 0;
943	bufs[1].len = 0;
944
945	if (!dead) {
946	return 0;
947	}
948
949	avail = alsa_get_avail (alsa->handle);
950	if (avail < 0) {
951	dolog ("Could not get number of captured frames\n");
952	return 0;
953	}
954
955	if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
956	avail = hw->samples;
957	}
958
959	decr = audio_MIN (dead, avail);
960	if (!decr) {
961	return 0;
962	}
963
964	if (hw->wpos + decr > hw->samples) {
965	bufs[0].len = (hw->samples - hw->wpos);
966	bufs[1].len = (decr - (hw->samples - hw->wpos));
967	}
968	else {
969	bufs[0].len = decr;
970	}
971
972	for (i = 0; i < 2; ++i) {
973	void *src;
974	st_sample_t *dst;
975	snd_pcm_sframes_t nread;
976	snd_pcm_uframes_t len;
977
978	len = bufs[i].len;
979
980	src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
981	dst = hw->conv_buf + bufs[i].add;
982
983	while (len) {
984	nread = snd_pcm_readi (alsa->handle, src, len);
985
986	if (nread <= 0) {
987	switch (nread) {
988	case 0:
989	if (conf.verbose) {
990	dolog ("Failed to read %ld frames (read zero)\n", len);
991	}
992	goto exit;
993
994	case -EPIPE:
995	if (alsa_recover (alsa->handle)) {
996	alsa_logerr (nread, "Failed to read %ld frames\n", len);
997	goto exit;
998	}
999	if (conf.verbose) {
1000	dolog ("Recovering from capture xrun\n");
1001	}
1002	continue;
1003
1004	case -EAGAIN:
1005	goto exit;
1006
1007	default:
1008	alsa_logerr (
1009	nread,
1010	"Failed to read %ld frames from %p\n",
1011	len,
1012	src
1013	);
1014	goto exit;
1015	}
1016	}
1017
1018	hw->conv (dst, src, nread, &nominal_volume);
1019
1020	src = advance (src, nread << hwshift);
1021	dst += nread;
1022
1023	read_samples += nread;
1024	len -= nread;
1025	}
1026	}
1027
1028	exit:
1029	hw->wpos = (hw->wpos + read_samples) % hw->samples;
1030	return read_samples;
1031	}
1032
1033	static int alsa_read (SWVoiceIn sw, void buf, int size)
1034	{
1035	return audio_pcm_sw_read (sw, buf, size);
1036	}
1037
1038	static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
1039	{
1040	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
1041
1042	switch (cmd) {
1043	case VOICE_ENABLE:
1044	ldebug ("enabling voice\n");
1045	return alsa_voice_ctl (alsa->handle, "capture", 0);
1046
1047	case VOICE_DISABLE:
1048	ldebug ("disabling voice\n");
1049	return alsa_voice_ctl (alsa->handle, "capture", 1);
1050	}
1051
1052	return -1;
1053	}
1054
1055	#ifdef VBOX
1056	static void alsa_error_handler(const char file, int line, const char function,
1057	int err, const char *fmt, ...)
1058	{
1059	/* ignore */
1060	}
1061	#endif
1062
1063	static void *alsa_audio_init (void)
1064	{
1065	#ifdef VBOX
1066	int rc;
1067
1068	rc = audioLoadAlsaLib();
1069	if (RT_FAILURE(rc)) {
1070	LogRel(("ALSA: Failed to load the ALSA shared library! Error %Rrc\n", rc));
1071	return NULL;
1072	}
1073	snd_lib_error_set_handler (alsa_error_handler);
1074	#endif
1075	return &conf;
1076	}
1077
1078	static void alsa_audio_fini (void *opaque)
1079	{
1080	(void) opaque;
1081	}
1082
1083	static struct audio_option alsa_options[] = {
1084	{"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
1085	"DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
1086	{"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
1087	"DAC period size", &conf.period_size_out_overriden, 0},
1088	{"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
1089	"DAC buffer size", &conf.buffer_size_out_overriden, 0},
1090
1091	{"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
1092	"ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
1093	{"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
1094	"ADC period size", &conf.period_size_in_overriden, 0},
1095	{"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
1096	"ADC buffer size", &conf.buffer_size_in_overriden, 0},
1097
1098	{"THRESHOLD", AUD_OPT_INT, &conf.threshold,
1099	"(undocumented)", NULL, 0},
1100
1101	{"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
1102	"DAC device name (for instance dmix)", NULL, 0},
1103
1104	{"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
1105	"ADC device name", NULL, 0},
1106
1107	{"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
1108	"Behave in a more verbose way", NULL, 0},
1109
1110	{NULL, 0, NULL, NULL, NULL, 0}
1111	};
1112
1113	static struct audio_pcm_ops alsa_pcm_ops = {
1114	alsa_init_out,
1115	alsa_fini_out,
1116	alsa_run_out,
1117	alsa_write,
1118	alsa_ctl_out,
1119
1120	alsa_init_in,
1121	alsa_fini_in,
1122	alsa_run_in,
1123	alsa_read,
1124	alsa_ctl_in
1125	};
1126
1127	struct audio_driver alsa_audio_driver = {
1128	INIT_FIELD (name = ) "alsa",
1129	INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
1130	INIT_FIELD (options = ) alsa_options,
1131	INIT_FIELD (init = ) alsa_audio_init,
1132	INIT_FIELD (fini = ) alsa_audio_fini,
1133	INIT_FIELD (pcm_ops = ) &alsa_pcm_ops,
1134	INIT_FIELD (can_be_default = ) 1,
1135	INIT_FIELD (max_voices_out = ) INT_MAX,
1136	INIT_FIELD (max_voices_in = ) INT_MAX,
1137	INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
1138	INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)
1139	};

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source: vbox/trunk/src/VBox/Devices/Audio/alsaaudio.c@ 12677

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