alsaaudio.c@ 980

Last change on this file since 980 was 787, checked in by vboxsync, 18 years ago
made ALSA work again
Property svn:eol-style set to `native` Property svn:keywords set to `Author Date Id Revision`
File size: 28.9 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	#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	/* latency = period_size * periods / (rate * bytes_per_frame) */
47
48	static struct {
49	int size_in_usec_in;
50	int size_in_usec_out;
51	const char *pcm_name_in;
52	const char *pcm_name_out;
53	unsigned int buffer_size_in;
54	unsigned int period_size_in;
55	unsigned int buffer_size_out;
56	unsigned int period_size_out;
57	unsigned int threshold;
58
59	int buffer_size_in_overriden;
60	int period_size_in_overriden;
61
62	int buffer_size_out_overriden;
63	int period_size_out_overriden;
64	int verbose;
65	} conf = {
66	#ifdef HIGH_LATENCY
67	INIT_FIELD (.size_in_usec_in =) 1,
68	INIT_FIELD (.size_in_usec_out =) 1,
69	#else
70	INIT_FIELD (.size_in_usec_in =) 0,
71	INIT_FIELD (.size_in_usec_out =) 0,
72	#endif
73	INIT_FIELD (.pcm_name_out =) "default",
74	INIT_FIELD (.pcm_name_in =) "default",
75	#ifdef HIGH_LATENCY
76	INIT_FIELD (.buffer_size_in =) 400000,
77	INIT_FIELD (.period_size_in =) 400000 / 4,
78	INIT_FIELD (.buffer_size_out =) 400000,
79	INIT_FIELD (.period_size_out =) 400000 / 4,
80	#else
81	#define DEFAULT_BUFFER_SIZE 1024
82	#define DEFAULT_PERIOD_SIZE 256
83	INIT_FIELD (.buffer_size_in =) DEFAULT_BUFFER_SIZE * 4,
84	INIT_FIELD (.period_size_in =) DEFAULT_PERIOD_SIZE * 4,
85	INIT_FIELD (.buffer_size_out =) DEFAULT_BUFFER_SIZE,
86	INIT_FIELD (.period_size_out =) DEFAULT_PERIOD_SIZE,
87	#endif
88	INIT_FIELD (.threshold =) 0,
89	INIT_FIELD (.buffer_size_in_overriden =) 0,
90	INIT_FIELD (.period_size_in_overriden =) 0,
91	INIT_FIELD (.buffer_size_out_overriden =) 0,
92	INIT_FIELD (.period_size_out_overriden =) 0,
93	INIT_FIELD (.verbose =) 0
94	};
95
96	struct alsa_params_req {
97	int freq;
98	audfmt_e fmt;
99	int nchannels;
100	unsigned int buffer_size;
101	unsigned int period_size;
102	};
103
104	struct alsa_params_obt {
105	int freq;
106	audfmt_e fmt;
107	int nchannels;
108	snd_pcm_uframes_t samples;
109	};
110
111	static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
112	{
113	va_list ap;
114
115	va_start (ap, fmt);
116	AUD_vlog (AUDIO_CAP, fmt, ap);
117	va_end (ap);
118
119	AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
120	}
121
122	static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
123	int err,
124	const char *typ,
125	const char *fmt,
126	...
127	)
128	{
129	va_list ap;
130
131	AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
132
133	va_start (ap, fmt);
134	AUD_vlog (AUDIO_CAP, fmt, ap);
135	va_end (ap);
136
137	AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
138	}
139
140	static void alsa_anal_close (snd_pcm_t **handlep)
141	{
142	int err = snd_pcm_close (*handlep);
143	if (err) {
144	alsa_logerr (err, "Failed to close PCM handle %p\n",
145	(void ) handlep);
146	}
147	*handlep = NULL;
148	}
149
150	static int alsa_write (SWVoiceOut sw, void buf, int len)
151	{
152	return audio_pcm_sw_write (sw, buf, len);
153	}
154
155	static int aud_to_alsafmt (audfmt_e fmt)
156	{
157	switch (fmt) {
158	case AUD_FMT_S8:
159	return SND_PCM_FORMAT_S8;
160
161	case AUD_FMT_U8:
162	return SND_PCM_FORMAT_U8;
163
164	case AUD_FMT_S16:
165	return SND_PCM_FORMAT_S16_LE;
166
167	case AUD_FMT_U16:
168	return SND_PCM_FORMAT_U16_LE;
169
170	default:
171	dolog ("Internal logic error: Bad audio format %d\n", fmt);
172	#ifdef DEBUG_AUDIO
173	abort ();
174	#endif
175	return SND_PCM_FORMAT_U8;
176	}
177	}
178
179	static int alsa_to_audfmt (int alsafmt, audfmt_e fmt, int endianness)
180	{
181	switch (alsafmt) {
182	case SND_PCM_FORMAT_S8:
183	*endianness = 0;
184	*fmt = AUD_FMT_S8;
185	break;
186
187	case SND_PCM_FORMAT_U8:
188	*endianness = 0;
189	*fmt = AUD_FMT_U8;
190	break;
191
192	case SND_PCM_FORMAT_S16_LE:
193	*endianness = 0;
194	*fmt = AUD_FMT_S16;
195	break;
196
197	case SND_PCM_FORMAT_U16_LE:
198	*endianness = 0;
199	*fmt = AUD_FMT_U16;
200	break;
201
202	case SND_PCM_FORMAT_S16_BE:
203	*endianness = 1;
204	*fmt = AUD_FMT_S16;
205	break;
206
207	case SND_PCM_FORMAT_U16_BE:
208	*endianness = 1;
209	*fmt = AUD_FMT_U16;
210	break;
211
212	default:
213	dolog ("Unrecognized audio format %d\n", alsafmt);
214	return -1;
215	}
216
217	return 0;
218	}
219
220	#if defined DEBUG_MISMATCHES \|\| defined DEBUG
221	static void alsa_dump_info (struct alsa_params_req *req,
222	struct alsa_params_obt *obt)
223	{
224	dolog ("parameter \| requested value \| obtained value\n");
225	dolog ("format \| %10d \| %10d\n", req->fmt, obt->fmt);
226	dolog ("channels \| %10d \| %10d\n",
227	req->nchannels, obt->nchannels);
228	dolog ("frequency \| %10d \| %10d\n", req->freq, obt->freq);
229	dolog ("============================================\n");
230	dolog ("requested: buffer size %d period size %d\n",
231	req->buffer_size, req->period_size);
232	dolog ("obtained: samples %ld\n", obt->samples);
233	}
234	#endif
235
236	static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
237	{
238	int err;
239	snd_pcm_sw_params_t *sw_params;
240
241	snd_pcm_sw_params_alloca (&sw_params);
242
243	err = snd_pcm_sw_params_current (handle, sw_params);
244	if (err < 0) {
245	dolog ("Could not fully initialize DAC\n");
246	alsa_logerr (err, "Failed to get current software parameters\n");
247	return;
248	}
249
250	err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
251	if (err < 0) {
252	dolog ("Could not fully initialize DAC\n");
253	alsa_logerr (err, "Failed to set software threshold to %ld\n",
254	threshold);
255	return;
256	}
257
258	err = snd_pcm_sw_params (handle, sw_params);
259	if (err < 0) {
260	dolog ("Could not fully initialize DAC\n");
261	alsa_logerr (err, "Failed to set software parameters\n");
262	return;
263	}
264	}
265
266	static int alsa_open (int in, struct alsa_params_req *req,
267	struct alsa_params_obt obt, snd_pcm_t *handlep)
268	{
269	snd_pcm_t *handle;
270	snd_pcm_hw_params_t *hw_params;
271	int err;
272	unsigned int freq, nchannels;
273	const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
274	unsigned int period_size, buffer_size;
275	snd_pcm_uframes_t obt_buffer_size;
276	const char *typ = in ? "ADC" : "DAC";
277
278	freq = req->freq;
279	period_size = req->period_size;
280	buffer_size = req->buffer_size;
281	nchannels = req->nchannels;
282
283	snd_pcm_hw_params_alloca (&hw_params);
284
285	err = snd_pcm_open (
286	&handle,
287	pcm_name,
288	in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
289	SND_PCM_NONBLOCK
290	);
291	if (err < 0) {
292	#ifndef VBOX
293	alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
294	#else
295	LogRel(("Audio/ALSA: Failed to open '%s' as %s\n", pcm_name, typ));
296	#endif
297	return -1;
298	}
299
300	err = snd_pcm_hw_params_any (handle, hw_params);
301	if (err < 0) {
302	#ifndef VBOX
303	alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
304	#else
305	LogRel(("Audio/ALSA: Failed to initialize hardware parameters\n"));
306	#endif
307	goto err;
308	}
309
310	err = snd_pcm_hw_params_set_access (
311	handle,
312	hw_params,
313	SND_PCM_ACCESS_RW_INTERLEAVED
314	);
315	if (err < 0) {
316	#ifndef VBOX
317	alsa_logerr2 (err, typ, "Failed to set access type\n");
318	#else
319	LogRel(("Audio/ALSA: Failed to set access type\n"));
320	#endif
321	goto err;
322	}
323
324	err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
325	if (err < 0) {
326	#ifndef VBOX
327	alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
328	#else
329	LogRel(("Audio/ALSA: Failed to set format %d\n", req->fmt));
330	#endif
331	goto err;
332	}
333
334	err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
335	if (err < 0) {
336	#ifndef VBOX
337	alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
338	#else
339	LogRel(("Audio/ALSA: Failed to set frequency %dHz\n", req->freq));
340	#endif
341	goto err;
342	}
343
344	err = snd_pcm_hw_params_set_channels_near (
345	handle,
346	hw_params,
347	&nchannels
348	);
349	if (err < 0) {
350	#ifndef VBOX
351	alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
352	req->nchannels);
353	#else
354	LogRel(("Audio/ALSA: Failed to set number of channels to %d\n", req->nchannels));
355	#endif
356	goto err;
357	}
358
359	if (nchannels != 1 && nchannels != 2) {
360	#ifndef VBOX
361	alsa_logerr2 (err, typ,
362	"Can not handle obtained number of channels %d\n",
363	nchannels);
364	#else
365	LogRel(("Audio/ALSA: Cannot handle obtained number of channels (%d)\n", nchannels));
366	#endif
367	goto err;
368	}
369
370	if (!((in && conf.size_in_usec_in) \|\| (!in && conf.size_in_usec_out))) {
371	if (!buffer_size) {
372	buffer_size = DEFAULT_BUFFER_SIZE;
373	period_size= DEFAULT_PERIOD_SIZE;
374	}
375	}
376
377	if (buffer_size) {
378	if ((in && conf.size_in_usec_in) \|\| (!in && conf.size_in_usec_out)) {
379	if (period_size) {
380	err = snd_pcm_hw_params_set_period_time_near (
381	handle,
382	hw_params,
383	&period_size,
384	0
385	);
386	if (err < 0) {
387	#ifndef VBOX
388	alsa_logerr2 (err, typ,
389	"Failed to set period time %d\n",
390	req->period_size);
391	#else
392	LogRel(("Audio/ALSA: Failed to set period time %d\n", req->period_size));
393	#endif
394	goto err;
395	}
396	}
397
398	err = snd_pcm_hw_params_set_buffer_time_near (
399	handle,
400	hw_params,
401	&buffer_size,
402	0
403	);
404
405	if (err < 0) {
406	#ifndef VBOX
407	alsa_logerr2 (err, typ,
408	"Failed to set buffer time %d\n",
409	req->buffer_size);
410	#else
411	LogRel(("Audio/ALSA: Failed to set buffer time %d\n", req->buffer_size));
412	#endif
413	goto err;
414	}
415	}
416	else {
417	int dir;
418	snd_pcm_uframes_t minval;
419
420	if (period_size) {
421	minval = period_size;
422	dir = 0;
423
424	err = snd_pcm_hw_params_get_period_size_min (
425	hw_params,
426	&minval,
427	&dir
428	);
429	if (err < 0) {
430	#ifndef VBOX
431	alsa_logerr (
432	err,
433	"Could not get minmal period size for %s\n",
434	typ
435	);
436	#else
437	LogRel(("Audio/ALSA: Could not get minimal period size for %s\n", typ));
438	#endif
439	}
440	else {
441	dolog("minimal period size %ld\n", minval);
442	if (period_size < minval) {
443	if ((in && conf.period_size_in_overriden)
444	\|\| (!in && conf.period_size_out_overriden)) {
445	dolog ("%s period size(%d) is less "
446	"than minmal period size(%ld)\n",
447	typ,
448	period_size,
449	minval);
450	}
451	period_size = minval;
452	}
453	}
454
455	#ifndef VBOX
456	err = snd_pcm_hw_params_set_period_size (
457	handle,
458	hw_params,
459	period_size,
460	0
461	);
462	#else
463	err = snd_pcm_hw_params_set_period_size_near (
464	handle,
465	hw_params,
466	(snd_pcm_uframes_t*)&period_size,
467	0
468	);
469	#endif
470	dolog("PERIOD_SIZE %d\n", period_size);
471	if (err < 0) {
472	#ifndef VBOX
473	alsa_logerr2 (err, typ, "Failed to set period size %d\n",
474	period_size);
475	#else
476	LogRel(("Audio/ALSA: Failed to set period size %d (%s)\n",
477	period_size, snd_strerror(err)));
478	#endif
479	goto err;
480	}
481	}
482
483	#ifdef VBOX
484	/* Calculate default buffer size here since it might have been changed
485	* in the _near functions */
486	buffer_size = 4 * period_size;
487	#endif
488
489	minval = buffer_size;
490	err = snd_pcm_hw_params_get_buffer_size_min (
491	hw_params,
492	&minval
493	);
494	if (err < 0) {
495	#ifndef VBOX
496	alsa_logerr (err, "Could not get minmal buffer size for %s\n",
497	typ);
498	#else
499	LogRel(("Audio/ALSA: Could not get minimal buffer size for %s\n", typ));
500	#endif
501	}
502	else {
503	if (buffer_size < minval) {
504	if ((in && conf.buffer_size_in_overriden)
505	\|\| (!in && conf.buffer_size_out_overriden)) {
506	dolog (
507	"%s buffer size(%d) is less "
508	"than minimal buffer size(%ld)\n",
509	typ,
510	buffer_size,
511	minval
512	);
513	}
514	buffer_size = minval;
515	}
516	}
517
518	err = snd_pcm_hw_params_set_buffer_size_near (
519	handle,
520	hw_params,
521	(snd_pcm_uframes_t*)&buffer_size
522	);
523	dolog("BUFFER_SIZE %d\n", buffer_size);
524	if (err < 0) {
525	#ifndef VBOX
526	alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
527	buffer_size);
528	#else
529	LogRel(("Audio/ALSA: Failed to set buffer size %d (%s)\n",
530	buffer_size, snd_strerror(err)));
531	#endif
532	goto err;
533	}
534	}
535	}
536	else {
537	dolog ("warning: Buffer size is not set\n");
538	}
539
540	err = snd_pcm_hw_params (handle, hw_params);
541	if (err < 0) {
542	#ifndef VBOX
543	alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
544	#else
545	LogRel(("Audio/ALSA: Failed to apply audio parameters\n"));
546	#endif
547	goto err;
548	}
549
550	err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
551	if (err < 0) {
552	#ifndef VBOX
553	alsa_logerr2 (err, typ, "Failed to get buffer size\n");
554	#else
555	LogRel(("Audio/ALSA: Failed to get buffer size\n"));
556	#endif
557	goto err;
558	}
559
560	#ifdef VBOX
561	LogRel(("Audio/ALSA: %s frequency %dHz, period size %d, buffer size %d\n",
562	typ, req->freq, period_size, obt_buffer_size));
563	#endif
564
565	err = snd_pcm_prepare (handle);
566	if (err < 0) {
567	alsa_logerr2 (err, typ, "Could not prepare handle %p\n",
568	(void *) handle);
569	goto err;
570	}
571
572	if (!in && conf.threshold) {
573	snd_pcm_uframes_t threshold;
574	int bytes_per_sec;
575
576	bytes_per_sec = freq
577	<< (nchannels == 2)
578	<< (req->fmt == AUD_FMT_S16 \|\| req->fmt == AUD_FMT_U16);
579
580	threshold = (conf.threshold * bytes_per_sec) / 1000;
581	alsa_set_threshold (handle, threshold);
582	}
583
584	obt->fmt = req->fmt;
585	obt->nchannels = nchannels;
586	obt->freq = freq;
587	obt->samples = obt_buffer_size;
588	*handlep = handle;
589
590	#if defined DEBUG_MISMATCHES \|\| defined DEBUG
591	if (obt->fmt != req->fmt \|\|
592	obt->nchannels != req->nchannels \|\|
593	obt->freq != req->freq) {
594	dolog ("Audio paramters mismatch for %s\n", typ);
595	alsa_dump_info (req, obt);
596	}
597	#endif
598
599	#ifdef DEBUG
600	alsa_dump_info (req, obt);
601	#endif
602	return 0;
603
604	err:
605	alsa_anal_close (&handle);
606	return -1;
607	}
608
609	static int alsa_recover (snd_pcm_t *handle)
610	{
611	int err = snd_pcm_prepare (handle);
612	if (err < 0) {
613	alsa_logerr (err, "Failed to prepare handle %p\n",
614	(void *) handle);
615	return -1;
616	}
617	return 0;
618	}
619
620	static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
621	{
622	snd_pcm_sframes_t avail;
623
624	avail = snd_pcm_avail_update (handle);
625	if (avail < 0) {
626	if (avail == -EPIPE) {
627	if (!alsa_recover (handle)) {
628	avail = snd_pcm_avail_update (handle);
629	}
630	}
631
632	if (avail < 0) {
633	alsa_logerr (avail,
634	"Could not obtain number of available frames\n");
635	return -1;
636	}
637	}
638
639	return avail;
640	}
641
642	static int alsa_run_out (HWVoiceOut *hw)
643	{
644	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
645	int rpos, live, decr;
646	int samples;
647	uint8_t *dst;
648	st_sample_t *src;
649	snd_pcm_sframes_t avail;
650
651	live = audio_pcm_hw_get_live_out (hw);
652	if (!live) {
653	return 0;
654	}
655
656	avail = alsa_get_avail (alsa->handle);
657	if (avail < 0) {
658	dolog ("Could not get number of available playback frames\n");
659	return 0;
660	}
661
662	decr = audio_MIN (live, avail);
663	samples = decr;
664	rpos = hw->rpos;
665	while (samples) {
666	int left_till_end_samples = hw->samples - rpos;
667	int len = audio_MIN (samples, left_till_end_samples);
668	snd_pcm_sframes_t written;
669
670	src = hw->mix_buf + rpos;
671	dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
672
673	hw->clip (dst, src, len);
674
675	while (len) {
676	written = snd_pcm_writei (alsa->handle, dst, len);
677
678	if (written <= 0) {
679	switch (written) {
680	case 0:
681	if (conf.verbose) {
682	dolog ("Failed to write %d frames (wrote zero)\n", len);
683	}
684	goto exit;
685
686	case -EPIPE:
687	if (alsa_recover (alsa->handle)) {
688	alsa_logerr (written, "Failed to write %d frames\n",
689	len);
690	goto exit;
691	}
692	if (conf.verbose) {
693	dolog ("Recovering from playback xrun\n");
694	}
695	continue;
696
697	case -EAGAIN:
698	goto exit;
699
700	default:
701	alsa_logerr (written, "Failed to write %d frames to %p\n",
702	len, dst);
703	goto exit;
704	}
705	}
706
707	rpos = (rpos + written) % hw->samples;
708	samples -= written;
709	len -= written;
710	dst = advance (dst, written << hw->info.shift);
711	src += written;
712	}
713	}
714
715	exit:
716	hw->rpos = rpos;
717	return decr;
718	}
719
720	static void alsa_fini_out (HWVoiceOut *hw)
721	{
722	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
723
724	ldebug ("alsa_fini\n");
725	alsa_anal_close (&alsa->handle);
726
727	if (alsa->pcm_buf) {
728	qemu_free (alsa->pcm_buf);
729	alsa->pcm_buf = NULL;
730	}
731	}
732
733	static int alsa_init_out (HWVoiceOut hw, audsettings_t as)
734	{
735	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
736	struct alsa_params_req req;
737	struct alsa_params_obt obt;
738	audfmt_e effective_fmt;
739	int endianness;
740	int err;
741	snd_pcm_t *handle;
742	audsettings_t obt_as;
743
744	req.fmt = aud_to_alsafmt (as->fmt);
745	req.freq = as->freq;
746	req.nchannels = as->nchannels;
747	req.period_size = conf.period_size_out;
748	req.buffer_size = conf.buffer_size_out;
749
750	if (alsa_open (0, &req, &obt, &handle)) {
751	return -1;
752	}
753
754	err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
755	if (err) {
756	alsa_anal_close (&handle);
757	return -1;
758	}
759
760	obt_as.freq = obt.freq;
761	obt_as.nchannels = obt.nchannels;
762	obt_as.fmt = effective_fmt;
763	obt_as.endianness = endianness;
764
765	audio_pcm_init_info (&hw->info, &obt_as);
766	hw->samples = obt.samples;
767
768	alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
769	if (!alsa->pcm_buf) {
770	dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
771	hw->samples, 1 << hw->info.shift);
772	alsa_anal_close (&handle);
773	return -1;
774	}
775
776	alsa->handle = handle;
777	return 0;
778	}
779
780	static int alsa_voice_ctl (snd_pcm_t handle, const char typ, int pause)
781	{
782	int err;
783
784	if (pause) {
785	err = snd_pcm_drop (handle);
786	if (err < 0) {
787	alsa_logerr (err, "Could not stop %s\n", typ);
788	return -1;
789	}
790	}
791	else {
792	err = snd_pcm_prepare (handle);
793	if (err < 0) {
794	alsa_logerr (err, "Could not prepare handle for %s\n", typ);
795	return -1;
796	}
797	}
798
799	return 0;
800	}
801
802	static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
803	{
804	ALSAVoiceOut alsa = (ALSAVoiceOut ) hw;
805
806	switch (cmd) {
807	case VOICE_ENABLE:
808	ldebug ("enabling voice\n");
809	return alsa_voice_ctl (alsa->handle, "playback", 0);
810
811	case VOICE_DISABLE:
812	ldebug ("disabling voice\n");
813	return alsa_voice_ctl (alsa->handle, "playback", 1);
814	}
815
816	return -1;
817	}
818
819	static int alsa_init_in (HWVoiceIn hw, audsettings_t as)
820	{
821	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
822	struct alsa_params_req req;
823	struct alsa_params_obt obt;
824	int endianness;
825	int err;
826	audfmt_e effective_fmt;
827	snd_pcm_t *handle;
828	audsettings_t obt_as;
829
830	req.fmt = aud_to_alsafmt (as->fmt);
831	req.freq = as->freq;
832	req.nchannels = as->nchannels;
833	req.period_size = conf.period_size_in;
834	req.buffer_size = conf.buffer_size_in;
835
836	if (alsa_open (1, &req, &obt, &handle)) {
837	return -1;
838	}
839
840	err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
841	if (err) {
842	alsa_anal_close (&handle);
843	return -1;
844	}
845
846	obt_as.freq = obt.freq;
847	obt_as.nchannels = obt.nchannels;
848	obt_as.fmt = effective_fmt;
849	obt_as.endianness = endianness;
850
851	audio_pcm_init_info (&hw->info, &obt_as);
852	hw->samples = obt.samples;
853
854	alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
855	if (!alsa->pcm_buf) {
856	dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
857	hw->samples, 1 << hw->info.shift);
858	alsa_anal_close (&handle);
859	return -1;
860	}
861
862	alsa->handle = handle;
863	return 0;
864	}
865
866	static void alsa_fini_in (HWVoiceIn *hw)
867	{
868	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
869
870	alsa_anal_close (&alsa->handle);
871
872	if (alsa->pcm_buf) {
873	qemu_free (alsa->pcm_buf);
874	alsa->pcm_buf = NULL;
875	}
876	}
877
878	static int alsa_run_in (HWVoiceIn *hw)
879	{
880	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
881	int hwshift = hw->info.shift;
882	int i;
883	int live = audio_pcm_hw_get_live_in (hw);
884	int dead = hw->samples - live;
885	int decr;
886	struct {
887	int add;
888	int len;
889	} bufs[2];
890
891	snd_pcm_sframes_t avail;
892	snd_pcm_uframes_t read_samples = 0;
893
894	bufs[0].add = hw->wpos;
895	bufs[0].len = 0;
896	bufs[1].add = 0;
897	bufs[1].len = 0;
898
899	if (!dead) {
900	return 0;
901	}
902
903	avail = alsa_get_avail (alsa->handle);
904	if (avail < 0) {
905	dolog ("Could not get number of captured frames\n");
906	return 0;
907	}
908
909	if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
910	avail = hw->samples;
911	}
912
913	decr = audio_MIN (dead, avail);
914	if (!decr) {
915	return 0;
916	}
917
918	if (hw->wpos + decr > hw->samples) {
919	bufs[0].len = (hw->samples - hw->wpos);
920	bufs[1].len = (decr - (hw->samples - hw->wpos));
921	}
922	else {
923	bufs[0].len = decr;
924	}
925
926	for (i = 0; i < 2; ++i) {
927	void *src;
928	st_sample_t *dst;
929	snd_pcm_sframes_t nread;
930	snd_pcm_uframes_t len;
931
932	len = bufs[i].len;
933
934	src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
935	dst = hw->conv_buf + bufs[i].add;
936
937	while (len) {
938	nread = snd_pcm_readi (alsa->handle, src, len);
939
940	if (nread <= 0) {
941	switch (nread) {
942	case 0:
943	if (conf.verbose) {
944	dolog ("Failed to read %ld frames (read zero)\n", len);
945	}
946	goto exit;
947
948	case -EPIPE:
949	if (alsa_recover (alsa->handle)) {
950	alsa_logerr (nread, "Failed to read %ld frames\n", len);
951	goto exit;
952	}
953	if (conf.verbose) {
954	dolog ("Recovering from capture xrun\n");
955	}
956	continue;
957
958	case -EAGAIN:
959	goto exit;
960
961	default:
962	alsa_logerr (
963	nread,
964	"Failed to read %ld frames from %p\n",
965	len,
966	src
967	);
968	goto exit;
969	}
970	}
971
972	hw->conv (dst, src, nread, &nominal_volume);
973
974	src = advance (src, nread << hwshift);
975	dst += nread;
976
977	read_samples += nread;
978	len -= nread;
979	}
980	}
981
982	exit:
983	hw->wpos = (hw->wpos + read_samples) % hw->samples;
984	return read_samples;
985	}
986
987	static int alsa_read (SWVoiceIn sw, void buf, int size)
988	{
989	return audio_pcm_sw_read (sw, buf, size);
990	}
991
992	static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
993	{
994	ALSAVoiceIn alsa = (ALSAVoiceIn ) hw;
995
996	switch (cmd) {
997	case VOICE_ENABLE:
998	ldebug ("enabling voice\n");
999	return alsa_voice_ctl (alsa->handle, "capture", 0);
1000
1001	case VOICE_DISABLE:
1002	ldebug ("disabling voice\n");
1003	return alsa_voice_ctl (alsa->handle, "capture", 1);
1004	}
1005
1006	return -1;
1007	}
1008
1009	static void *alsa_audio_init (void)
1010	{
1011	return &conf;
1012	}
1013
1014	static void alsa_audio_fini (void *opaque)
1015	{
1016	(void) opaque;
1017	}
1018
1019	static struct audio_option alsa_options[] = {
1020	{"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
1021	"DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
1022	{"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
1023	"DAC period size", &conf.period_size_out_overriden, 0},
1024	{"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
1025	"DAC buffer size", &conf.buffer_size_out_overriden, 0},
1026
1027	{"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
1028	"ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
1029	{"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
1030	"ADC period size", &conf.period_size_in_overriden, 0},
1031	{"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
1032	"ADC buffer size", &conf.buffer_size_in_overriden, 0},
1033
1034	{"THRESHOLD", AUD_OPT_INT, &conf.threshold,
1035	"(undocumented)", NULL, 0},
1036
1037	{"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
1038	"DAC device name (for instance dmix)", NULL, 0},
1039
1040	{"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
1041	"ADC device name", NULL, 0},
1042
1043	{"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
1044	"Behave in a more verbose way", NULL, 0},
1045
1046	{NULL, 0, NULL, NULL, NULL, 0}
1047	};
1048
1049	static struct audio_pcm_ops alsa_pcm_ops = {
1050	alsa_init_out,
1051	alsa_fini_out,
1052	alsa_run_out,
1053	alsa_write,
1054	alsa_ctl_out,
1055
1056	alsa_init_in,
1057	alsa_fini_in,
1058	alsa_run_in,
1059	alsa_read,
1060	alsa_ctl_in
1061	};
1062
1063	struct audio_driver alsa_audio_driver = {
1064	INIT_FIELD (name = ) "alsa",
1065	INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
1066	INIT_FIELD (options = ) alsa_options,
1067	INIT_FIELD (init = ) alsa_audio_init,
1068	INIT_FIELD (fini = ) alsa_audio_fini,
1069	INIT_FIELD (pcm_ops = ) &alsa_pcm_ops,
1070	INIT_FIELD (can_be_default = ) 1,
1071	INIT_FIELD (max_voices_out = ) INT_MAX,
1072	INIT_FIELD (max_voices_in = ) INT_MAX,
1073	INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
1074	INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)
1075	};

Note: See TracBrowser for help on using the repository browser.

source: vbox/trunk/src/VBox/Devices/Audio/alsaaudio.c@ 980

Download in other formats: