DrvHostALSAAudio.cpp@ 55998

Last change on this file since 55998 was 55920, checked in by vboxsync, 10 years ago
PDM/Audio: AudioMixBuffer documentation, renaming.
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1	/* $Id: DrvHostALSAAudio.cpp 55920 2015-05-18 19:11:51Z vboxsync $ */
2	/** @file
3	* VBox audio devices: ALSA audio driver.
4	*/
5
6	/*
7	* Copyright (C) 2006-2015 Oracle Corporation
8	*
9	* This file is part of VirtualBox Open Source Edition (OSE), as
10	* available from http://www.virtualbox.org. This file is free software;
11	* you can redistribute it and/or modify it under the terms of the GNU
12	* General Public License (GPL) as published by the Free Software
13	* Foundation, in version 2 as it comes in the "COPYING" file of the
14	* VirtualBox OSE distribution. VirtualBox OSE is distributed in the
15	* hope that it will be useful, but WITHOUT ANY WARRANTY of any kind.
16	* --------------------------------------------------------------------
17	*
18	* This code is based on: alsaaudio.c
19	*
20	* QEMU ALSA audio driver
21	*
22	* Copyright (c) 2005 Vassili Karpov (malc)
23	*
24	* Permission is hereby granted, free of charge, to any person obtaining a copy
25	* of this software and associated documentation files (the "Software"), to deal
26	* in the Software without restriction, including without limitation the rights
27	* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
28	* copies of the Software, and to permit persons to whom the Software is
29	* furnished to do so, subject to the following conditions:
30	*
31	* The above copyright notice and this permission notice shall be included in
32	* all copies or substantial portions of the Software.
33	*
34	* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
35	* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
36	* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
37	* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
38	* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
39	* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
40	* THE SOFTWARE.
41	*/
42
43	/*******************************************************************************
44	* Header Files *
45	*******************************************************************************/
46
47	#include <iprt/alloc.h>
48	#include <iprt/uuid.h> /* For PDMIBASE_2_PDMDRV. */
49	#include <VBox/vmm/pdmaudioifs.h>
50
51	RT_C_DECLS_BEGIN
52	#include "alsa_stubs.h"
53	#include "alsa_mangling.h"
54	RT_C_DECLS_END
55
56	#include <alsa/asoundlib.h>
57
58	#include "DrvAudio.h"
59	#include "AudioMixBuffer.h"
60
61	#include "VBoxDD.h"
62	#include "vl_vbox.h"
63
64
65
66	#ifdef LOG_GROUP
67	# undef LOG_GROUP
68	#endif
69	#define LOG_GROUP LOG_GROUP_DEV_AUDIO
70	#include <VBox/log.h>
71
72	typedef struct ALSAAUDIOSTREAMIN
73	{
74	PDMAUDIOHSTSTRMIN pStreamIn;
75	snd_pcm_t *phPCM;
76	void *pvBuf;
77	size_t cbBuf;
78	} ALSAAUDIOSTREAMIN, *PALSAAUDIOSTREAMIN;
79
80	typedef struct ALSAAUDIOSTREAMOUT
81	{
82	PDMAUDIOHSTSTRMOUT pStreamOut;
83	snd_pcm_t *phPCM;
84	void *pvBuf;
85	size_t cbBuf;
86	} ALSAAUDIOSTREAMOUT, *PALSAAUDIOSTREAMOUT;
87
88	/* latency = period_size * periods / (rate * bytes_per_frame) */
89
90	typedef struct ALSAAUDIOCFG
91	{
92	int size_in_usec_in;
93	int size_in_usec_out;
94	const char *pcm_name_in;
95	const char *pcm_name_out;
96	unsigned int buffer_size_in;
97	unsigned int period_size_in;
98	unsigned int buffer_size_out;
99	unsigned int period_size_out;
100	unsigned int threshold;
101
102	int buffer_size_in_overriden;
103	int period_size_in_overriden;
104
105	int buffer_size_out_overriden;
106	int period_size_out_overriden;
107
108	} ALSAAUDIOCFG, *PALSAAUDIOCFG;
109
110	static int drvHostALSAAudioRecover(snd_pcm_t *phPCM);
111
112	static ALSAAUDIOCFG s_ALSAConf =
113	{
114	#ifdef HIGH_LATENCY
115	1,
116	1,
117	#else
118	0,
119	0,
120	#endif
121	"default",
122	"default",
123	#ifdef HIGH_LATENCY
124	400000,
125	400000 / 4,
126	400000,
127	400000 / 4,
128	#else
129	# define DEFAULT_BUFFER_SIZE 1024
130	# define DEFAULT_PERIOD_SIZE 256
131	DEFAULT_BUFFER_SIZE * 4,
132	DEFAULT_PERIOD_SIZE * 4,
133	DEFAULT_BUFFER_SIZE,
134	DEFAULT_PERIOD_SIZE,
135	#endif
136	0,
137	0,
138	0,
139	0,
140	0
141	};
142
143	/**
144	* Host Alsa audio driver instance data.
145	* @implements PDMIAUDIOCONNECTOR
146	*/
147	typedef struct DRVHOSTALSAAUDIO
148	{
149	/** Pointer to the driver instance structure. */
150	PPDMDRVINS pDrvIns;
151	/** Pointer to host audio interface. */
152	PDMIHOSTAUDIO IHostAudio;
153	/** Error count for not flooding the release log.
154	* UINT32_MAX for unlimited logging. */
155	uint32_t cLogErrors;
156	} DRVHOSTALSAAUDIO, *PDRVHOSTALSAAUDIO;
157
158	typedef struct ALSAAUDIOSTREAMCFG
159	{
160	unsigned int freq;
161	snd_pcm_format_t fmt;
162	int nchannels;
163	unsigned long buffer_size;
164	unsigned long period_size;
165	snd_pcm_uframes_t samples;
166	} ALSAAUDIOSTREAMCFG, *PALSAAUDIOSTREAMCFG;
167
168	static int drvHostALSAAudioClose(snd_pcm_t **pphPCM)
169	{
170	if (!pphPCM \|\| !*pphPCM)
171	return VINF_SUCCESS;
172
173	int rc;
174	int rc2 = snd_pcm_close(*pphPCM);
175	if (rc2)
176	{
177	LogRel(("ALSA: Closing PCM descriptor failed: %s\n",
178	snd_strerror(rc2)));
179	rc = VERR_GENERAL_FAILURE; /** @todo */
180	}
181	else
182	{
183	*pphPCM = NULL;
184	rc = VINF_SUCCESS;
185	}
186
187	return rc;
188	}
189
190	static snd_pcm_format_t drvHostALSAAudioFmtToALSA(PDMAUDIOFMT fmt)
191	{
192	switch (fmt)
193	{
194	case AUD_FMT_S8:
195	return SND_PCM_FORMAT_S8;
196
197	case AUD_FMT_U8:
198	return SND_PCM_FORMAT_U8;
199
200	case AUD_FMT_S16:
201	return SND_PCM_FORMAT_S16_LE;
202
203	case AUD_FMT_U16:
204	return SND_PCM_FORMAT_U16_LE;
205
206	case AUD_FMT_S32:
207	return SND_PCM_FORMAT_S32_LE;
208
209	case AUD_FMT_U32:
210	return SND_PCM_FORMAT_U32_LE;
211
212	default:
213	break;
214	}
215
216	AssertMsgFailed(("Format %ld not supported\n", fmt));
217	return SND_PCM_FORMAT_U8;
218	}
219
220	static int drvHostALSAAudioALSAToFmt(snd_pcm_format_t fmt,
221	PDMAUDIOFMT pFmt, PDMAUDIOENDIANNESS pEndianness)
222	{
223	AssertPtrReturn(pFmt, VERR_INVALID_POINTER);
224	/* pEndianness is optional. */
225
226	switch (fmt)
227	{
228	case SND_PCM_FORMAT_S8:
229	*pFmt = AUD_FMT_S8;
230	if (pEndianness)
231	*pEndianness = PDMAUDIOENDIANNESS_LITTLE;
232	break;
233
234	case SND_PCM_FORMAT_U8:
235	*pFmt = AUD_FMT_U8;
236	if (pEndianness)
237	*pEndianness = PDMAUDIOENDIANNESS_LITTLE;
238	break;
239
240	case SND_PCM_FORMAT_S16_LE:
241	*pFmt = AUD_FMT_S16;
242	if (pEndianness)
243	*pEndianness = PDMAUDIOENDIANNESS_LITTLE;
244	break;
245
246	case SND_PCM_FORMAT_U16_LE:
247	*pFmt = AUD_FMT_U16;
248	if (pEndianness)
249	*pEndianness = PDMAUDIOENDIANNESS_LITTLE;
250	break;
251
252	case SND_PCM_FORMAT_S16_BE:
253	*pFmt = AUD_FMT_S16;
254	if (pEndianness)
255	*pEndianness = PDMAUDIOENDIANNESS_BIG;
256	break;
257
258	case SND_PCM_FORMAT_U16_BE:
259	*pFmt = AUD_FMT_U16;
260	if (pEndianness)
261	*pEndianness = PDMAUDIOENDIANNESS_BIG;
262	break;
263
264	case SND_PCM_FORMAT_S32_LE:
265	*pFmt = AUD_FMT_S32;
266	if (pEndianness)
267	*pEndianness = PDMAUDIOENDIANNESS_LITTLE;
268	break;
269
270	case SND_PCM_FORMAT_U32_LE:
271	*pFmt = AUD_FMT_U32;
272	if (pEndianness)
273	*pEndianness = PDMAUDIOENDIANNESS_LITTLE;
274	break;
275
276	case SND_PCM_FORMAT_S32_BE:
277	*pFmt = AUD_FMT_S32;
278	if (pEndianness)
279	*pEndianness = PDMAUDIOENDIANNESS_BIG;
280	break;
281
282	case SND_PCM_FORMAT_U32_BE:
283	*pFmt = AUD_FMT_U32;
284	if (pEndianness)
285	*pEndianness = PDMAUDIOENDIANNESS_BIG;
286	break;
287
288	default:
289	AssertMsgFailed(("Format %ld not supported\n", fmt));
290	return VERR_NOT_SUPPORTED;
291	}
292
293	return VINF_SUCCESS;
294	}
295
296	static int drvHostALSAAudioALSAGetShift(snd_pcm_format_t fmt, unsigned *puShift)
297	{
298	AssertPtrReturn(puShift, VERR_INVALID_POINTER);
299
300	switch (fmt)
301	{
302	case SND_PCM_FORMAT_S8:
303	case SND_PCM_FORMAT_U8:
304	*puShift = 0;
305	break;
306
307	case SND_PCM_FORMAT_S16_LE:
308	case SND_PCM_FORMAT_U16_LE:
309	case SND_PCM_FORMAT_S16_BE:
310	case SND_PCM_FORMAT_U16_BE:
311	*puShift = 1;
312	break;
313
314	case SND_PCM_FORMAT_S32_LE:
315	case SND_PCM_FORMAT_U32_LE:
316	case SND_PCM_FORMAT_S32_BE:
317	case SND_PCM_FORMAT_U32_BE:
318	*puShift = 2;
319	break;
320
321	default:
322	AssertMsgFailed(("Format %ld not supported\n", fmt));
323	return VERR_NOT_SUPPORTED;
324	}
325
326	return VINF_SUCCESS;
327	}
328
329	static int drvHostALSAAudioSetThreshold(snd_pcm_t *phPCM,
330	snd_pcm_uframes_t threshold)
331	{
332	snd_pcm_sw_params_t *pSWParms = NULL;
333	snd_pcm_sw_params_alloca(&pSWParms);
334	if (!pSWParms)
335	return VERR_NO_MEMORY;
336
337	int rc;
338	do
339	{
340	int err = snd_pcm_sw_params_current(phPCM, pSWParms);
341	if (err < 0)
342	{
343	LogRel(("ALSA: Failed to get current software parameters for threshold: %s\n",
344	snd_strerror(err)));
345	rc = VERR_ACCESS_DENIED;
346	break;
347	}
348
349	err = snd_pcm_sw_params_set_start_threshold(phPCM, pSWParms, threshold);
350	if (err < 0)
351	{
352	LogRel(("ALSA: Failed to set software threshold to %ld: %s\n",
353	threshold, snd_strerror(err)));
354	rc = VERR_ACCESS_DENIED;
355	break;
356	}
357
358	err = snd_pcm_sw_params(phPCM, pSWParms);
359	if (err < 0)
360	{
361	LogRel(("ALSA: Failed to set new software parameters for threshold: %s\n",
362	snd_strerror(err)));
363	rc = VERR_ACCESS_DENIED;
364	break;
365	}
366
367	LogFlowFunc(("Setting threshold to %RU32\n", threshold));
368	rc = VINF_SUCCESS;
369	}
370	while (0);
371
372	return rc;
373	}
374
375	static int drvHostALSAAudioOpen(bool fIn,
376	PALSAAUDIOSTREAMCFG pCfgReq,
377	PALSAAUDIOSTREAMCFG pCfgObt,
378	snd_pcm_t **pphPCM)
379	{
380	snd_pcm_t *phPCM = NULL;
381	int rc;
382
383	unsigned int cChannels = pCfgReq->nchannels;
384	unsigned int uFreq = pCfgReq->freq;
385	snd_pcm_uframes_t obt_buffer_size;
386
387	do
388	{
389	const char *pszDev = fIn ? s_ALSAConf.pcm_name_in : s_ALSAConf.pcm_name_out;
390	if (!pszDev)
391	{
392	LogRel(("ALSA: Invalid or no %s device name set\n",
393	fIn ? "input" : "output"));
394	rc = VERR_INVALID_PARAMETER;
395	break;
396	}
397
398	int err = snd_pcm_open(&phPCM, pszDev,
399	fIn ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
400	SND_PCM_NONBLOCK);
401	if (err < 0)
402	{
403	LogRel(("ALSA: Failed to open \"%s\" as %s: %s\n", pszDev,
404	fIn ? "ADC" : "DAC", snd_strerror(err)));
405	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
406	break;
407	}
408
409	snd_pcm_hw_params_t *pHWParms;
410	snd_pcm_hw_params_alloca(&pHWParms); /** @todo Check for successful allocation? */
411	err = snd_pcm_hw_params_any(phPCM, pHWParms);
412	if (err < 0)
413	{
414	LogRel(("ALSA: Failed to initialize hardware parameters: %s\n",
415	snd_strerror(err)));
416	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
417	break;
418	}
419
420	err = snd_pcm_hw_params_set_access(phPCM, pHWParms,
421	SND_PCM_ACCESS_RW_INTERLEAVED);
422	if (err < 0)
423	{
424	LogRel(("ALSA: Failed to set access type: %s\n", snd_strerror(err)));
425	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
426	break;
427	}
428
429	err = snd_pcm_hw_params_set_format(phPCM, pHWParms, pCfgReq->fmt);
430	if (err < 0)
431	{
432	LogRel(("ALSA: Failed to set audio format to %d: %s\n",
433	pCfgReq->fmt, snd_strerror(err)));
434	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
435	break;
436	}
437
438	err = snd_pcm_hw_params_set_rate_near(phPCM, pHWParms, &uFreq, 0);
439	if (err < 0)
440	{
441	LogRel(("ALSA: Failed to set frequency to %dHz: %s\n",
442	pCfgReq->freq, snd_strerror(err)));
443	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
444	break;
445	}
446
447	err = snd_pcm_hw_params_set_channels_near(phPCM, pHWParms, &cChannels);
448	if (err < 0)
449	{
450	LogRel(("ALSA: Failed to set number of channels to %d\n", pCfgReq->nchannels));
451	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
452	break;
453	}
454
455	if ( cChannels != 1
456	&& cChannels != 2)
457	{
458	LogRel(("ALSA: Number of audio channels (%u) not supported\n", cChannels));
459	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
460	break;
461	}
462
463	unsigned int period_size = pCfgReq->period_size;
464	unsigned int buffer_size = pCfgReq->buffer_size;
465
466	if ( !((fIn && s_ALSAConf.size_in_usec_in)
467	\|\| (!fIn && s_ALSAConf.size_in_usec_out)))
468	{
469	if (!buffer_size)
470	{
471	buffer_size = DEFAULT_BUFFER_SIZE;
472	period_size = DEFAULT_PERIOD_SIZE;
473	}
474	}
475
476	if (buffer_size)
477	{
478	if ( ( fIn && s_ALSAConf.size_in_usec_in)
479	\|\| (!fIn && s_ALSAConf.size_in_usec_out))
480	{
481	if (period_size)
482	{
483	err = snd_pcm_hw_params_set_period_time_near(phPCM, pHWParms,
484	&period_size, 0);
485	if (err < 0)
486	{
487	LogRel(("ALSA: Failed to set period time %d\n", pCfgReq->period_size));
488	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
489	break;
490	}
491	}
492
493	err = snd_pcm_hw_params_set_buffer_time_near(phPCM, pHWParms,
494	&buffer_size, 0);
495	if (err < 0)
496	{
497	LogRel(("ALSA: Failed to set buffer time %d\n", pCfgReq->buffer_size));
498	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
499	break;
500	}
501	}
502	else
503	{
504	snd_pcm_uframes_t period_size_f = (snd_pcm_uframes_t)period_size;
505	snd_pcm_uframes_t buffer_size_f = (snd_pcm_uframes_t)buffer_size;
506
507	snd_pcm_uframes_t minval;
508
509	if (period_size_f)
510	{
511	minval = period_size_f;
512
513	int dir = 0;
514	err = snd_pcm_hw_params_get_period_size_min(pHWParms,
515	&minval, &dir);
516	if (err < 0)
517	{
518	LogRel(("ALSA: Could not determine minimal period size\n"));
519	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
520	break;
521	}
522	else
523	{
524	LogFunc(("Minimal period size is: %ld\n", minval));
525	if (period_size_f < minval)
526	{
527	if ( ( fIn && s_ALSAConf.period_size_in_overriden)
528	\|\| (!fIn && s_ALSAConf.period_size_out_overriden))
529	{
530	LogFunc(("Period size %RU32 is less than minimal period size %RU32\n",
531	period_size_f, minval));
532	}
533
534	period_size_f = minval;
535	}
536	}
537
538	err = snd_pcm_hw_params_set_period_size_near(phPCM, pHWParms,
539	&period_size_f, 0);
540	LogFunc(("Period size is: %RU32\n", period_size_f));
541	if (err < 0)
542	{
543	LogRel(("ALSA: Failed to set period size %d (%s)\n",
544	period_size_f, snd_strerror(err)));
545	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
546	break;
547	}
548	}
549
550	/* Calculate default buffer size here since it might have been changed
551	* in the _near functions */
552	buffer_size_f = 4 * period_size_f;
553
554	minval = buffer_size_f;
555	err = snd_pcm_hw_params_get_buffer_size_min(pHWParms, &minval);
556	if (err < 0)
557	{
558	LogRel(("ALSA: Could not retrieve minimal buffer size\n"));
559	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
560	break;
561	}
562	else
563	{
564	LogFunc(("Minimal buffer size is: %RU32\n", minval));
565	if (buffer_size_f < minval)
566	{
567	if ( ( fIn && s_ALSAConf.buffer_size_in_overriden)
568	\|\| (!fIn && s_ALSAConf.buffer_size_out_overriden))
569	{
570	LogFunc(("Buffer size %RU32 is less than minimal buffer size %RU32\n",
571	buffer_size_f, minval));
572	}
573
574	buffer_size_f = minval;
575	}
576	}
577
578	err = snd_pcm_hw_params_set_buffer_size_near(phPCM,
579	pHWParms, &buffer_size_f);
580	LogFunc(("Buffer size is: %RU32\n", buffer_size_f));
581	if (err < 0)
582	{
583	LogRel(("ALSA: Failed to set buffer size %d: %s\n",
584	buffer_size_f, snd_strerror(err)));
585	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
586	break;
587	}
588	}
589	}
590	else
591	LogFunc(("Warning: Buffer size is not set\n"));
592
593	err = snd_pcm_hw_params(phPCM, pHWParms);
594	if (err < 0)
595	{
596	LogRel(("ALSA: Failed to apply audio parameters\n"));
597	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
598	break;
599	}
600
601	err = snd_pcm_hw_params_get_buffer_size(pHWParms, &obt_buffer_size);
602	if (err < 0)
603	{
604	LogRel(("ALSA: Failed to get buffer size\n"));
605	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
606	break;
607	}
608
609	snd_pcm_uframes_t obt_period_size;
610	int dir = 0;
611	err = snd_pcm_hw_params_get_period_size(pHWParms, &obt_period_size, &dir);
612	if (err < 0)
613	{
614	LogRel(("ALSA: Failed to get period size\n"));
615	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
616	break;
617	}
618
619	LogFunc(("Freq=%dHz, period size=%RU32, buffer size=%RU32\n",
620	pCfgReq->freq, obt_period_size, obt_buffer_size));
621
622	err = snd_pcm_prepare(phPCM);
623	if (err < 0)
624	{
625	LogRel(("ALSA: Could not prepare hPCM %p\n", (void *)phPCM));
626	rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
627	break;
628	}
629
630	if ( !fIn
631	&& s_ALSAConf.threshold)
632	{
633	unsigned uShift;
634	rc = drvHostALSAAudioALSAGetShift(pCfgReq->fmt, &uShift);
635	if (RT_SUCCESS(rc))
636	{
637	int bytes_per_sec = uFreq
638	<< (cChannels == 2)
639	<< uShift;
640
641	snd_pcm_uframes_t threshold
642	= (s_ALSAConf.threshold * bytes_per_sec) / 1000;
643
644	rc = drvHostALSAAudioSetThreshold(phPCM, threshold);
645	}
646	}
647	else
648	rc = VINF_SUCCESS;
649	}
650	while (0);
651
652	if (RT_SUCCESS(rc))
653	{
654	pCfgObt->fmt = pCfgReq->fmt;
655	pCfgObt->nchannels = cChannels;
656	pCfgObt->freq = uFreq;
657	pCfgObt->samples = obt_buffer_size;
658
659	*pphPCM = phPCM;
660	}
661	else
662	drvHostALSAAudioClose(&phPCM);
663
664	LogFlowFuncLeaveRC(rc);
665	return rc;
666	}
667
668	#ifdef DEBUG
669	static void drvHostALSAAudioErrorHandler(const char file, int line, const char function,
670	int err, const char *fmt, ...)
671	{
672	/** @todo Implement me! */
673	}
674	#endif
675
676	static int drvHostALSAAudioGetAvail(snd_pcm_t phPCM, snd_pcm_sframes_t pFramesAvail)
677	{
678	AssertPtrReturn(phPCM, VERR_INVALID_POINTER);
679	AssertPtrReturn(pFramesAvail, VERR_INVALID_POINTER);
680
681	int rc;
682
683	snd_pcm_sframes_t framesAvail;
684	framesAvail = snd_pcm_avail_update(phPCM);
685	if (framesAvail < 0)
686	{
687	if (framesAvail == -EPIPE)
688	{
689	rc = drvHostALSAAudioRecover(phPCM);
690	if (RT_SUCCESS(rc))
691	framesAvail = snd_pcm_avail_update(phPCM);
692	}
693	else
694	rc = VERR_ACCESS_DENIED; /** @todo Find a better rc. */
695	}
696	else
697	rc = VINF_SUCCESS;
698
699	if (framesAvail >= 0)
700	*pFramesAvail = framesAvail;
701
702	return rc;
703	}
704
705	static int drvHostALSAAudioRecover(snd_pcm_t *phPCM)
706	{
707	AssertPtrReturn(phPCM, VERR_INVALID_POINTER);
708
709	int err = snd_pcm_prepare(phPCM);
710	if (err < 0)
711	{
712	LogFunc(("Failed to recover stream %p: %s\n",
713	phPCM, snd_strerror(err)));
714	return VERR_ACCESS_DENIED; /** @todo Find a better rc. */
715	}
716
717	return VINF_SUCCESS;
718	}
719
720	static int drvHostALSAAudioResume(snd_pcm_t *phPCM)
721	{
722	AssertPtrReturn(phPCM, VERR_INVALID_POINTER);
723
724	int err = snd_pcm_resume(phPCM);
725	if (err < 0)
726	{
727	LogFunc(("Failed to resume stream %p: %s\n",
728	phPCM, snd_strerror(err)));
729	return VERR_ACCESS_DENIED; /** @todo Find a better rc. */
730	}
731
732	return VINF_SUCCESS;
733	}
734
735	static int drvHostALSAAudioStreamCtl(snd_pcm_t *phPCM, bool fPause)
736	{
737	int err;
738	if (fPause)
739	{
740	err = snd_pcm_drop(phPCM);
741	if (err < 0)
742	{
743	LogFlow(("Error stopping stream %p: %s\n",
744	phPCM, snd_strerror(err)));
745	return VERR_ACCESS_DENIED;
746	}
747	}
748	else
749	{
750	err = snd_pcm_prepare (phPCM);
751	if (err < 0)
752	{
753	LogFlow(("Error preparing stream %p: %s\n",
754	phPCM, snd_strerror(err)));
755	return VERR_ACCESS_DENIED;
756	}
757	}
758
759	return VINF_SUCCESS;
760	}
761
762	static DECLCALLBACK(int) drvHostALSAAudioInit(PPDMIHOSTAUDIO pInterface)
763	{
764	NOREF(pInterface);
765
766	LogFlowFuncEnter();
767
768	int rc = audioLoadAlsaLib();
769	if (RT_FAILURE(rc))
770	LogRel(("ALSA: Failed to load the ALSA shared library, rc=%Rrc\n", rc));
771	else
772	{
773	#ifdef DEBUG
774	snd_lib_error_set_handler(drvHostALSAAudioErrorHandler);
775	#endif
776	}
777
778	return rc;
779	}
780
781	static DECLCALLBACK(int) drvHostALSAAudioCaptureIn(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMIN pHstStrmIn,
782	uint32_t *pcSamplesCaptured)
783	{
784	NOREF(pInterface);
785	AssertPtrReturn(pHstStrmIn, VERR_INVALID_POINTER);
786
787	PALSAAUDIOSTREAMIN pThisStrmIn = (PALSAAUDIOSTREAMIN)pHstStrmIn;
788
789	snd_pcm_sframes_t cAvail;
790	int rc = drvHostALSAAudioGetAvail(pThisStrmIn->phPCM, &cAvail);
791	if (RT_FAILURE(rc))
792	{
793	LogFunc(("Error getting number of captured frames, rc=%Rrc\n", rc));
794	return rc;
795	}
796
797	if (!cAvail) /* No data yet? */
798	{
799	snd_pcm_state_t state = snd_pcm_state(pThisStrmIn->phPCM);
800	switch (state)
801	{
802	case SND_PCM_STATE_PREPARED:
803	cAvail = AudioMixBufFree(&pHstStrmIn->MixBuf);
804	break;
805
806	case SND_PCM_STATE_SUSPENDED:
807	{
808	rc = drvHostALSAAudioResume(pThisStrmIn->phPCM);
809	if (RT_FAILURE(rc))
810	break;
811
812	LogFlow(("Resuming suspended input stream\n"));
813	break;
814	}
815
816	default:
817	LogFlow(("No frames available, state=%d\n", state));
818	break;
819	}
820
821	if (!cAvail)
822	{
823	if (pcSamplesCaptured)
824	*pcSamplesCaptured = 0;
825	return VINF_SUCCESS;
826	}
827	}
828
829	Assert(cAvail);
830	size_t cbToRead = AUDIOMIXBUF_S2B(&pHstStrmIn->MixBuf, cAvail);
831
832	LogFlowFunc(("cbToRead=%zu, cAvail=%RI32\n", cbToRead, cAvail));
833
834	uint32_t cWrittenTotal = 0;
835	snd_pcm_uframes_t cToRead;
836	snd_pcm_sframes_t cRead;
837
838	while ( cbToRead
839	&& RT_SUCCESS(rc))
840	{
841	cToRead = RT_MIN(AUDIOMIXBUF_B2S(&pHstStrmIn->MixBuf, cbToRead),
842	AUDIOMIXBUF_B2S(&pHstStrmIn->MixBuf, pThisStrmIn->cbBuf));
843	AssertBreakStmt(cToRead, rc = VERR_NO_DATA);
844	cRead = snd_pcm_readi(pThisStrmIn->phPCM, pThisStrmIn->pvBuf, cToRead);
845	if (cRead <= 0)
846	{
847	switch (cRead)
848	{
849	case 0:
850	{
851	LogFunc(("No input frames available\n"));
852	rc = VERR_ACCESS_DENIED;
853	break;
854	}
855
856	case -EAGAIN:
857	/*
858	* Don't set error here because EAGAIN means there are no further frames
859	* available at the moment, try later. As we might have read some frames
860	* already these need to be processed instead.
861	*/
862	cbToRead = 0;
863	break;
864
865	case -EPIPE:
866	{
867	rc = drvHostALSAAudioRecover(pThisStrmIn->phPCM);
868	if (RT_FAILURE(rc))
869	break;
870
871	LogFlowFunc(("Recovered from capturing\n"));
872	continue;
873	}
874
875	default:
876	LogFunc(("Failed to read input frames: %s\n", snd_strerror(cRead)));
877	rc = VERR_GENERAL_FAILURE; /** @todo Fudge! */
878	break;
879	}
880	}
881	else
882	{
883	uint32_t cWritten;
884	rc = AudioMixBufWriteCirc(&pHstStrmIn->MixBuf,
885	pThisStrmIn->pvBuf, AUDIOMIXBUF_S2B(&pHstStrmIn->MixBuf, cRead),
886	&cWritten);
887	if (RT_FAILURE(rc))
888	break;
889
890	uint32_t cbWritten = AUDIOMIXBUF_S2B(&pHstStrmIn->MixBuf, cWritten);
891
892	Assert(cbToRead >= cbWritten);
893	cbToRead -= cbWritten;
894	cWrittenTotal += cWritten;
895	}
896	}
897
898	if (RT_SUCCESS(rc))
899	{
900	uint32_t cProcessed = 0;
901	if (cWrittenTotal)
902	rc = AudioMixBufMixToParent(&pHstStrmIn->MixBuf, cWrittenTotal,
903	&cProcessed);
904
905	if (pcSamplesCaptured)
906	*pcSamplesCaptured = cWrittenTotal;
907
908	LogFlowFunc(("cWrittenTotal=%RU32 (%RU32 processed), rc=%Rrc\n",
909	cWrittenTotal, cProcessed, rc));
910	}
911
912	LogFlowFuncLeaveRC(rc);
913	return rc;
914	}
915
916	static DECLCALLBACK(int) drvHostALSAAudioPlayOut(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMOUT pHstStrmOut,
917	uint32_t *pcSamplesPlayed)
918	{
919	NOREF(pInterface);
920	AssertPtrReturn(pHstStrmOut, VERR_INVALID_POINTER);
921
922	PALSAAUDIOSTREAMOUT pThisStrmOut = (PALSAAUDIOSTREAMOUT)pHstStrmOut;
923
924	int rc = VINF_SUCCESS;
925	uint32_t cbReadTotal = 0;
926
927	do
928	{
929	snd_pcm_sframes_t cAvail;
930	rc = drvHostALSAAudioGetAvail(pThisStrmOut->phPCM, &cAvail);
931	if (RT_FAILURE(rc))
932	{
933	LogFunc(("Error getting number of playback frames, rc=%Rrc\n", rc));
934	break;
935	}
936
937	size_t cbToRead = RT_MIN(AUDIOMIXBUF_S2B(&pHstStrmOut->MixBuf,
938	cAvail),
939	AUDIOMIXBUF_S2B(&pHstStrmOut->MixBuf,
940	drvAudioHstOutSamplesLive(pHstStrmOut, NULL /* pcStreamsLive */)));
941	LogFlowFunc(("cbToRead=%zu, cbAvail=%zu\n",
942	cbToRead, AUDIOMIXBUF_S2B(&pHstStrmOut->MixBuf, cAvail)));
943
944	uint32_t cRead, cbRead;
945	snd_pcm_sframes_t cWritten;
946	while (cbToRead)
947	{
948	rc = AudioMixBufReadCirc(&pHstStrmOut->MixBuf, pThisStrmOut->pvBuf, cbToRead, &cRead);
949	if (RT_FAILURE(rc))
950	break;
951
952	cbRead = AUDIOMIXBUF_S2B(&pHstStrmOut->MixBuf, cRead);
953	AssertBreak(cbRead);
954
955	cWritten = snd_pcm_writei(pThisStrmOut->phPCM, pThisStrmOut->pvBuf, cRead);
956	if (cWritten <= 0)
957	{
958	switch (cWritten)
959	{
960	case 0:
961	{
962	LogFunc(("Failed to write %RI32 frames\n", cRead));
963	rc = VERR_ACCESS_DENIED;
964	break;
965	}
966
967	case -EPIPE:
968	{
969	rc = drvHostALSAAudioRecover(pThisStrmOut->phPCM);
970	if (RT_FAILURE(rc))
971	break;
972
973	LogFlowFunc(("Recovered from playback\n"));
974	continue;
975	}
976
977	case -ESTRPIPE:
978	{
979	/* Stream was suspended and waiting for a recovery. */
980	rc = drvHostALSAAudioResume(pThisStrmOut->phPCM);
981	if (RT_FAILURE(rc))
982	{
983	LogRel(("ALSA: Failed to resume output stream\n"));
984	break;
985	}
986
987	LogFlowFunc(("Resumed suspended output stream\n"));
988	continue;
989	}
990
991	default:
992	LogFlowFunc(("Failed to write %RI32 output frames, rc=%Rrc\n",
993	cRead, rc));
994	rc = VERR_GENERAL_FAILURE; /** @todo */
995	break;
996	}
997	}
998
999	if (RT_FAILURE(rc))
1000	break;
1001
1002	Assert(cbToRead >= cRead);
1003	cbToRead -= cbRead;
1004	cbReadTotal += cbRead;
1005	}
1006	}
1007	while (0);
1008
1009	if (RT_SUCCESS(rc))
1010	{
1011	uint32_t cReadTotal = AUDIOMIXBUF_B2S(&pHstStrmOut->MixBuf, cbReadTotal);
1012	if (cReadTotal)
1013	AudioMixBufFinish(&pHstStrmOut->MixBuf, cReadTotal);
1014
1015	if (pcSamplesPlayed)
1016	*pcSamplesPlayed = cReadTotal;
1017
1018	LogFlowFunc(("cReadTotal=%RU32 (%RU32 bytes), rc=%Rrc\n",
1019	cReadTotal, cbReadTotal, rc));
1020	}
1021
1022	LogFlowFuncLeaveRC(rc);
1023	return rc;
1024	}
1025
1026	static DECLCALLBACK(int) drvHostALSAAudioFiniIn(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMIN pHstStrmIn)
1027	{
1028	NOREF(pInterface);
1029	AssertPtrReturn(pHstStrmIn, VERR_INVALID_POINTER);
1030
1031	PALSAAUDIOSTREAMIN pThisStrmIn = (PALSAAUDIOSTREAMIN)pHstStrmIn;
1032
1033	drvHostALSAAudioClose(&pThisStrmIn->phPCM);
1034
1035	if (pThisStrmIn->pvBuf)
1036	{
1037	RTMemFree(pThisStrmIn->pvBuf);
1038	pThisStrmIn->pvBuf = NULL;
1039	}
1040
1041	return VINF_SUCCESS;
1042	}
1043
1044	static DECLCALLBACK(int) drvHostALSAAudioFiniOut(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMOUT pHstStrmOut)
1045	{
1046	NOREF(pInterface);
1047	AssertPtrReturn(pHstStrmOut, VERR_INVALID_POINTER);
1048
1049	PALSAAUDIOSTREAMOUT pThisStrmOut = (PALSAAUDIOSTREAMOUT)pHstStrmOut;
1050
1051	drvHostALSAAudioClose(&pThisStrmOut->phPCM);
1052
1053	if (pThisStrmOut->pvBuf)
1054	{
1055	RTMemFree(pThisStrmOut->pvBuf);
1056	pThisStrmOut->pvBuf = NULL;
1057	}
1058
1059	return VINF_SUCCESS;
1060	}
1061
1062	static DECLCALLBACK(int) drvHostALSAAudioInitOut(PPDMIHOSTAUDIO pInterface,
1063	PPDMAUDIOHSTSTRMOUT pHstStrmOut, PPDMAUDIOSTREAMCFG pCfg,
1064	uint32_t *pcSamples)
1065	{
1066	NOREF(pInterface);
1067	AssertPtrReturn(pHstStrmOut, VERR_INVALID_POINTER);
1068	AssertPtrReturn(pCfg, VERR_INVALID_POINTER);
1069
1070	PALSAAUDIOSTREAMOUT pThisStrmOut = (PALSAAUDIOSTREAMOUT)pHstStrmOut;
1071	snd_pcm_t *phPCM = NULL;
1072
1073	int rc;
1074
1075	do
1076	{
1077	ALSAAUDIOSTREAMCFG req;
1078	req.fmt = drvHostALSAAudioFmtToALSA(pCfg->enmFormat);
1079	req.freq = pCfg->uHz;
1080	req.nchannels = pCfg->cChannels;
1081	req.period_size = s_ALSAConf.period_size_out;
1082	req.buffer_size = s_ALSAConf.buffer_size_out;
1083
1084	ALSAAUDIOSTREAMCFG obt;
1085	rc = drvHostALSAAudioOpen(false /* false */, &req, &obt, &phPCM);
1086	if (RT_FAILURE(rc))
1087	break;
1088
1089	PDMAUDIOFMT enmFormat;
1090	PDMAUDIOENDIANNESS enmEnd;
1091	rc = drvHostALSAAudioALSAToFmt(obt.fmt, &enmFormat, &enmEnd);
1092	if (RT_FAILURE(rc))
1093	break;
1094
1095	PDMAUDIOSTREAMCFG streamCfg;
1096	streamCfg.uHz = obt.freq;
1097	streamCfg.cChannels = obt.nchannels;
1098	streamCfg.enmFormat = enmFormat;
1099	streamCfg.enmEndianness = enmEnd;
1100
1101	rc = drvAudioStreamCfgToProps(&streamCfg, &pHstStrmOut->Props);
1102	if (RT_FAILURE(rc))
1103	break;
1104
1105	AssertBreakStmt(obt.samples, rc = VERR_INVALID_PARAMETER);
1106	size_t cbBuf = obt.samples * (1 << pHstStrmOut->Props.cShift);
1107	AssertBreakStmt(cbBuf, rc = VERR_INVALID_PARAMETER);
1108	pThisStrmOut->pvBuf = RTMemAlloc(cbBuf);
1109	if (!pThisStrmOut->pvBuf)
1110	{
1111	LogRel(("ALSA: Not enough memory for output DAC buffer (%RU32 samples, each %d bytes)\n",
1112	obt.samples, 1 << pHstStrmOut->Props.cShift));
1113	rc = VERR_NO_MEMORY;
1114	break;
1115	}
1116
1117	pThisStrmOut->cbBuf = cbBuf;
1118	pThisStrmOut->phPCM = phPCM;
1119
1120	if (pcSamples)
1121	*pcSamples = obt.samples;
1122	}
1123	while (0);
1124
1125	if (RT_FAILURE(rc))
1126	drvHostALSAAudioClose(&phPCM);
1127
1128	LogFlowFuncLeaveRC(rc);
1129	return rc;
1130	}
1131
1132	static DECLCALLBACK(int) drvHostALSAAudioInitIn(PPDMIHOSTAUDIO pInterface,
1133	PPDMAUDIOHSTSTRMIN pHstStrmIn, PPDMAUDIOSTREAMCFG pCfg,
1134	PDMAUDIORECSOURCE enmRecSource,
1135	uint32_t *pcSamples)
1136	{
1137	NOREF(pInterface);
1138	AssertPtrReturn(pHstStrmIn, VERR_INVALID_POINTER);
1139	AssertPtrReturn(pCfg, VERR_INVALID_POINTER);
1140
1141	int rc;
1142
1143	PALSAAUDIOSTREAMIN pThisStrmIn = (PALSAAUDIOSTREAMIN)pHstStrmIn;
1144	snd_pcm_t *phPCM = NULL;
1145
1146	do
1147	{
1148	ALSAAUDIOSTREAMCFG req;
1149	req.fmt = drvHostALSAAudioFmtToALSA(pCfg->enmFormat);
1150	req.freq = pCfg->uHz;
1151	req.nchannels = pCfg->cChannels;
1152	req.period_size = s_ALSAConf.period_size_in;
1153	req.buffer_size = s_ALSAConf.buffer_size_in;
1154
1155	ALSAAUDIOSTREAMCFG obt;
1156	rc = drvHostALSAAudioOpen(true /* fIn */, &req, &obt, &phPCM);
1157	if (RT_FAILURE(rc))
1158	break;
1159
1160	PDMAUDIOFMT enmFormat;
1161	PDMAUDIOENDIANNESS enmEnd;
1162	rc = drvHostALSAAudioALSAToFmt(obt.fmt, &enmFormat, &enmEnd);
1163	if (RT_FAILURE(rc))
1164	break;
1165
1166	PDMAUDIOSTREAMCFG streamCfg;
1167	streamCfg.uHz = obt.freq;
1168	streamCfg.cChannels = obt.nchannels;
1169	streamCfg.enmFormat = enmFormat;
1170	streamCfg.enmEndianness = enmEnd;
1171
1172	rc = drvAudioStreamCfgToProps(&streamCfg, &pHstStrmIn->Props);
1173	if (RT_FAILURE(rc))
1174	break;
1175
1176	AssertBreakStmt(obt.samples, rc = VERR_INVALID_PARAMETER);
1177	size_t cbBuf = obt.samples * (1 << pHstStrmIn->Props.cShift);
1178	AssertBreakStmt(cbBuf, rc = VERR_INVALID_PARAMETER);
1179	pThisStrmIn->pvBuf = RTMemAlloc(cbBuf);
1180	if (!pThisStrmIn->pvBuf)
1181	{
1182	LogRel(("ALSA: Not enough memory for input ADC buffer (%RU32 samples, each %d bytes)\n",
1183	obt.samples, 1 << pHstStrmIn->Props.cShift));
1184	rc = VERR_NO_MEMORY;
1185	break;
1186	}
1187
1188	pThisStrmIn->cbBuf = cbBuf;
1189	pThisStrmIn->phPCM = phPCM;
1190
1191	if (pcSamples)
1192	*pcSamples = obt.samples;
1193	}
1194	while (0);
1195
1196	if (RT_FAILURE(rc))
1197	drvHostALSAAudioClose(&phPCM);
1198
1199	LogFlowFuncLeaveRC(rc);
1200	return rc;
1201	}
1202
1203	static DECLCALLBACK(bool) drvHostALSAAudioIsEnabled(PPDMIHOSTAUDIO pInterface, PDMAUDIODIR enmDir)
1204	{
1205	NOREF(pInterface);
1206	NOREF(enmDir);
1207	return true; /* Always all enabled. */
1208	}
1209
1210	static DECLCALLBACK(int) drvHostALSAAudioControlIn(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMIN pHstStrmIn,
1211	PDMAUDIOSTREAMCMD enmStreamCmd)
1212	{
1213	NOREF(pInterface);
1214	AssertPtrReturn(pHstStrmIn, VERR_INVALID_POINTER);
1215	PALSAAUDIOSTREAMIN pThisStrmIn = (PALSAAUDIOSTREAMIN)pHstStrmIn;
1216
1217	LogFlowFunc(("enmStreamCmd=%ld\n", enmStreamCmd));
1218
1219	int rc;
1220	switch (enmStreamCmd)
1221	{
1222	case PDMAUDIOSTREAMCMD_ENABLE:
1223	rc = drvHostALSAAudioStreamCtl(pThisStrmIn->phPCM, false /* fStop */);
1224	break;
1225
1226	case PDMAUDIOSTREAMCMD_DISABLE:
1227	rc = drvHostALSAAudioStreamCtl(pThisStrmIn->phPCM, true /* fStop */);
1228	break;
1229
1230	default:
1231	AssertMsgFailed(("Invalid command %ld\n", enmStreamCmd));
1232	rc = VERR_INVALID_PARAMETER;
1233	break;
1234	}
1235
1236	return rc;
1237	}
1238
1239	static DECLCALLBACK(int) drvHostALSAAudioControlOut(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMOUT pHstStrmOut,
1240	PDMAUDIOSTREAMCMD enmStreamCmd)
1241	{
1242	NOREF(pInterface);
1243	AssertPtrReturn(pHstStrmOut, VERR_INVALID_POINTER);
1244	PALSAAUDIOSTREAMOUT pThisStrmOut = (PALSAAUDIOSTREAMOUT)pHstStrmOut;
1245
1246	LogFlowFunc(("enmStreamCmd=%ld\n", enmStreamCmd));
1247
1248	int rc;
1249	switch (enmStreamCmd)
1250	{
1251	case PDMAUDIOSTREAMCMD_ENABLE:
1252	rc = drvHostALSAAudioStreamCtl(pThisStrmOut->phPCM, false /* fStop */);
1253	break;
1254
1255	case PDMAUDIOSTREAMCMD_DISABLE:
1256	rc = drvHostALSAAudioStreamCtl(pThisStrmOut->phPCM, true /* fStop */);
1257	break;
1258
1259	default:
1260	AssertMsgFailed(("Invalid command %ld\n", enmStreamCmd));
1261	rc = VERR_INVALID_PARAMETER;
1262	break;
1263	}
1264
1265	return rc;
1266	}
1267
1268	static DECLCALLBACK(int) drvHostALSAAudioGetConf(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDCFG pCfg)
1269	{
1270	NOREF(pInterface);
1271	AssertPtrReturn(pCfg, VERR_INVALID_POINTER);
1272
1273	pCfg->cbStreamOut = sizeof(ALSAAUDIOSTREAMOUT);
1274	pCfg->cbStreamIn = sizeof(ALSAAUDIOSTREAMIN);
1275	pCfg->cMaxHstStrmsOut = INT_MAX;
1276	pCfg->cMaxHstStrmsIn = INT_MAX;
1277
1278	return VINF_SUCCESS;
1279	}
1280
1281	static DECLCALLBACK(void) drvHostALSAAudioShutdown(PPDMIHOSTAUDIO pInterface)
1282	{
1283	NOREF(pInterface);
1284	}
1285
1286	/**
1287	* @interface_method_impl{PDMIBASE,pfnQueryInterface}
1288	*/
1289	static DECLCALLBACK(void ) drvHostALSAAudioQueryInterface(PPDMIBASE pInterface, const char pszIID)
1290	{
1291	PPDMDRVINS pDrvIns = PDMIBASE_2_PDMDRV(pInterface);
1292	PDRVHOSTALSAAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVHOSTALSAAUDIO);
1293	PDMIBASE_RETURN_INTERFACE(pszIID, PDMIBASE, &pDrvIns->IBase);
1294	PDMIBASE_RETURN_INTERFACE(pszIID, PDMIHOSTAUDIO, &pThis->IHostAudio);
1295
1296	return NULL;
1297	}
1298
1299	/**
1300	* Construct a DirectSound Audio driver instance.
1301	*
1302	* @copydoc FNPDMDRVCONSTRUCT
1303	*/
1304	static DECLCALLBACK(int) drvHostAlsaAudioConstruct(PPDMDRVINS pDrvIns, PCFGMNODE pCfg, uint32_t fFlags)
1305	{
1306	PDRVHOSTALSAAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVHOSTALSAAUDIO);
1307	LogRel(("Audio: Initializing ALSA driver\n"));
1308
1309	/*
1310	* Init the static parts.
1311	*/
1312	pThis->pDrvIns = pDrvIns;
1313	/* IBase */
1314	pDrvIns->IBase.pfnQueryInterface = drvHostALSAAudioQueryInterface;
1315	/* IHostAudio */
1316	PDMAUDIO_IHOSTAUDIO_CALLBACKS(drvHostALSAAudio);
1317
1318	return VINF_SUCCESS;
1319	}
1320
1321	/**
1322	* Char driver registration record.
1323	*/
1324	const PDMDRVREG g_DrvHostALSAAudio =
1325	{
1326	/* u32Version */
1327	PDM_DRVREG_VERSION,
1328	/* szName */
1329	"ALSAAudio",
1330	/* szRCMod */
1331	"",
1332	/* szR0Mod */
1333	"",
1334	/* pszDescription */
1335	"ALSA host audio driver",
1336	/* fFlags */
1337	PDM_DRVREG_FLAGS_HOST_BITS_DEFAULT,
1338	/* fClass. */
1339	PDM_DRVREG_CLASS_AUDIO,
1340	/* cMaxInstances */
1341	~0U,
1342	/* cbInstance */
1343	sizeof(DRVHOSTALSAAUDIO),
1344	/* pfnConstruct */
1345	drvHostAlsaAudioConstruct,
1346	/* pfnDestruct */
1347	NULL,
1348	/* pfnRelocate */
1349	NULL,
1350	/* pfnIOCtl */
1351	NULL,
1352	/* pfnPowerOn */
1353	NULL,
1354	/* pfnReset */
1355	NULL,
1356	/* pfnSuspend */
1357	NULL,
1358	/* pfnResume */
1359	NULL,
1360	/* pfnAttach */
1361	NULL,
1362	/* pfnDetach */
1363	NULL,
1364	/* pfnPowerOff */
1365	NULL,
1366	/* pfnSoftReset */
1367	NULL,
1368	/* u32EndVersion */
1369	PDM_DRVREG_VERSION
1370	};
1371
1372	static struct audio_option alsa_options[] =
1373	{
1374	{"DACSizeInUsec", AUD_OPT_BOOL, &s_ALSAConf.size_in_usec_out,
1375	"DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
1376	{"DACPeriodSize", AUD_OPT_INT, &s_ALSAConf.period_size_out,
1377	"DAC period size", &s_ALSAConf.period_size_out_overriden, 0},
1378	{"DACBufferSize", AUD_OPT_INT, &s_ALSAConf.buffer_size_out,
1379	"DAC buffer size", &s_ALSAConf.buffer_size_out_overriden, 0},
1380
1381	{"ADCSizeInUsec", AUD_OPT_BOOL, &s_ALSAConf.size_in_usec_in,
1382	"ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
1383	{"ADCPeriodSize", AUD_OPT_INT, &s_ALSAConf.period_size_in,
1384	"ADC period size", &s_ALSAConf.period_size_in_overriden, 0},
1385	{"ADCBufferSize", AUD_OPT_INT, &s_ALSAConf.buffer_size_in,
1386	"ADC buffer size", &s_ALSAConf.buffer_size_in_overriden, 0},
1387
1388	{"Threshold", AUD_OPT_INT, &s_ALSAConf.threshold,
1389	"(undocumented)", NULL, 0},
1390
1391	{"DACDev", AUD_OPT_STR, &s_ALSAConf.pcm_name_out,
1392	"DAC device name (for instance dmix)", NULL, 0},
1393
1394	{"ADCDev", AUD_OPT_STR, &s_ALSAConf.pcm_name_in,
1395	"ADC device name", NULL, 0},
1396
1397	NULL
1398	};
1399

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source: vbox/trunk/src/VBox/Devices/Audio/DrvHostALSAAudio.cpp@ 55998

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