DrvHostAudioAlsa.cpp@ 101359

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1	/* $Id: DrvHostAudioAlsa.cpp 98103 2023-01-17 14:15:46Z vboxsync $ */
2	/** @file
3	* Host audio driver - Advanced Linux Sound Architecture (ALSA).
4	*/
5
6	/*
7	* Copyright (C) 2006-2023 Oracle and/or its affiliates.
8	*
9	* This file is part of VirtualBox base platform packages, as
10	* available from https://www.virtualbox.org.
11	*
12	* This program is free software; you can redistribute it and/or
13	* modify it under the terms of the GNU General Public License
14	* as published by the Free Software Foundation, in version 3 of the
15	* License.
16	*
17	* This program is distributed in the hope that it will be useful, but
18	* WITHOUT ANY WARRANTY; without even the implied warranty of
19	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
20	* General Public License for more details.
21	*
22	* You should have received a copy of the GNU General Public License
23	* along with this program; if not, see <https://www.gnu.org/licenses>.
24	*
25	* SPDX-License-Identifier: GPL-3.0-only
26	* --------------------------------------------------------------------
27	*
28	* This code is based on: alsaaudio.c
29	*
30	* QEMU ALSA audio driver
31	*
32	* Copyright (c) 2005 Vassili Karpov (malc)
33	*
34	* Permission is hereby granted, free of charge, to any person obtaining a copy
35	* of this software and associated documentation files (the "Software"), to deal
36	* in the Software without restriction, including without limitation the rights
37	* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
38	* copies of the Software, and to permit persons to whom the Software is
39	* furnished to do so, subject to the following conditions:
40	*
41	* The above copyright notice and this permission notice shall be included in
42	* all copies or substantial portions of the Software.
43	*
44	* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
45	* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
46	* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
47	* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
48	* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
49	* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
50	* THE SOFTWARE.
51	*/
52
53
54	/*********************************************************************************************************************************
55	* Header Files *
56	*********************************************************************************************************************************/
57	#define LOG_GROUP LOG_GROUP_DRV_HOST_AUDIO
58	#include <VBox/log.h>
59	#include <iprt/alloc.h>
60	#include <iprt/uuid.h> /* For PDMIBASE_2_PDMDRV. */
61	#include <VBox/vmm/pdmaudioifs.h>
62	#include <VBox/vmm/pdmaudioinline.h>
63	#include <VBox/vmm/pdmaudiohostenuminline.h>
64
65	#include "DrvHostAudioAlsaStubsMangling.h"
66	#include <alsa/asoundlib.h>
67	#include <alsa/control.h> /* For device enumeration. */
68	#include <alsa/version.h>
69	#include "DrvHostAudioAlsaStubs.h"
70
71	#include "VBoxDD.h"
72
73
74	/*********************************************************************************************************************************
75	* Defined Constants And Macros *
76	*********************************************************************************************************************************/
77	/** Maximum number of tries to recover a broken pipe. */
78	#define ALSA_RECOVERY_TRIES_MAX 5
79
80
81	/*********************************************************************************************************************************
82	* Structures *
83	*********************************************************************************************************************************/
84	/**
85	* ALSA host audio specific stream data.
86	*/
87	typedef struct DRVHSTAUDALSASTREAM
88	{
89	/** Common part. */
90	PDMAUDIOBACKENDSTREAM Core;
91
92	/** Handle to the ALSA PCM stream. */
93	snd_pcm_t *hPCM;
94	/** Internal stream offset (for debugging). */
95	uint64_t offInternal;
96
97	/** The stream's acquired configuration. */
98	PDMAUDIOSTREAMCFG Cfg;
99	} DRVHSTAUDALSASTREAM;
100	/** Pointer to the ALSA host audio specific stream data. */
101	typedef DRVHSTAUDALSASTREAM *PDRVHSTAUDALSASTREAM;
102
103
104	/**
105	* Host Alsa audio driver instance data.
106	* @implements PDMIAUDIOCONNECTOR
107	*/
108	typedef struct DRVHSTAUDALSA
109	{
110	/** Pointer to the driver instance structure. */
111	PPDMDRVINS pDrvIns;
112	/** Pointer to host audio interface. */
113	PDMIHOSTAUDIO IHostAudio;
114	/** Error count for not flooding the release log.
115	* UINT32_MAX for unlimited logging. */
116	uint32_t cLogErrors;
117
118	/** Critical section protecting the default device strings. */
119	RTCRITSECT CritSect;
120	/** Default input device name. */
121	char szInputDev[256];
122	/** Default output device name. */
123	char szOutputDev[256];
124	/** Upwards notification interface. */
125	PPDMIHOSTAUDIOPORT pIHostAudioPort;
126	} DRVHSTAUDALSA;
127	/** Pointer to the instance data of an ALSA host audio driver. */
128	typedef DRVHSTAUDALSA *PDRVHSTAUDALSA;
129
130
131
132	/**
133	* Closes an ALSA stream
134	*
135	* @returns VBox status code.
136	* @param phPCM Pointer to the ALSA stream handle to close. Will be set to
137	* NULL.
138	*/
139	static int drvHstAudAlsaStreamClose(snd_pcm_t **phPCM)
140	{
141	if (!phPCM \|\| !*phPCM)
142	return VINF_SUCCESS;
143
144	LogRelFlowFuncEnter();
145
146	int rc;
147	int rc2 = snd_pcm_close(*phPCM);
148	if (rc2 == 0)
149	{
150	*phPCM = NULL;
151	rc = VINF_SUCCESS;
152	}
153	else
154	{
155	rc = RTErrConvertFromErrno(-rc2);
156	LogRel(("ALSA: Closing PCM descriptor failed: %s (%d, %Rrc)\n", snd_strerror(rc2), rc2, rc));
157	}
158
159	LogRelFlowFuncLeaveRC(rc);
160	return rc;
161	}
162
163
164	#ifdef DEBUG
165	static void drvHstAudAlsaDbgErrorHandler(const char file, int line, const char function,
166	int err, const char *fmt, ...)
167	{
168	/** @todo Implement me! */
169	RT_NOREF(file, line, function, err, fmt);
170	}
171	#endif
172
173
174	/**
175	* Tries to recover an ALSA stream.
176	*
177	* @returns VBox status code.
178	* @param hPCM ALSA stream handle.
179	*/
180	static int drvHstAudAlsaStreamRecover(snd_pcm_t *hPCM)
181	{
182	AssertPtrReturn(hPCM, VERR_INVALID_POINTER);
183
184	int rc = snd_pcm_prepare(hPCM);
185	if (rc >= 0)
186	{
187	LogFlowFunc(("Successfully recovered %p.\n", hPCM));
188	return VINF_SUCCESS;
189	}
190	LogFunc(("Failed to recover stream %p: %s (%d)\n", hPCM, snd_strerror(rc), rc));
191	return RTErrConvertFromErrno(-rc);
192	}
193
194
195	/**
196	* Resumes an ALSA stream.
197	*
198	* Used by drvHstAudAlsaHA_StreamPlay() and drvHstAudAlsaHA_StreamCapture().
199	*
200	* @returns VBox status code.
201	* @param hPCM ALSA stream to resume.
202	*/
203	static int drvHstAudAlsaStreamResume(snd_pcm_t *hPCM)
204	{
205	AssertPtrReturn(hPCM, VERR_INVALID_POINTER);
206
207	int rc = snd_pcm_resume(hPCM);
208	if (rc >= 0)
209	{
210	LogFlowFunc(("Successfuly resumed %p.\n", hPCM));
211	return VINF_SUCCESS;
212	}
213	LogFunc(("Failed to resume stream %p: %s (%d)\n", hPCM, snd_strerror(rc), rc));
214	return RTErrConvertFromErrno(-rc);
215	}
216
217
218	/**
219	* @interface_method_impl{PDMIHOSTAUDIO,pfnGetConfig}
220	*/
221	static DECLCALLBACK(int) drvHstAudAlsaHA_GetConfig(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDCFG pBackendCfg)
222	{
223	RT_NOREF(pInterface);
224	AssertPtrReturn(pBackendCfg, VERR_INVALID_POINTER);
225
226	/*
227	* Fill in the config structure.
228	*/
229	RTStrCopy(pBackendCfg->szName, sizeof(pBackendCfg->szName), "ALSA");
230	pBackendCfg->cbStream = sizeof(DRVHSTAUDALSASTREAM);
231	pBackendCfg->fFlags = 0;
232	/* ALSA allows exactly one input and one output used at a time for the selected device(s). */
233	pBackendCfg->cMaxStreamsIn = 1;
234	pBackendCfg->cMaxStreamsOut = 1;
235
236	return VINF_SUCCESS;
237	}
238
239
240	/**
241	* @interface_method_impl{PDMIHOSTAUDIO,pfnGetDevices}
242	*/
243	static DECLCALLBACK(int) drvHstAudAlsaHA_GetDevices(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHOSTENUM pDeviceEnum)
244	{
245	RT_NOREF(pInterface);
246	PDMAudioHostEnumInit(pDeviceEnum);
247
248	char **papszHints = NULL;
249	int rc = snd_device_name_hint(-1 /* All cards /, "pcm", (void**)&papszHints);
250	if (rc == 0)
251	{
252	rc = VINF_SUCCESS;
253	for (size_t iHint = 0; papszHints[iHint] != NULL && RT_SUCCESS(rc); iHint++)
254	{
255	/*
256	* Retrieve the available info:
257	*/
258	const char * const pszHint = papszHints[iHint];
259	char * const pszDev = snd_device_name_get_hint(pszHint, "NAME");
260	char * const pszInOutId = snd_device_name_get_hint(pszHint, "IOID");
261	char * const pszDesc = snd_device_name_get_hint(pszHint, "DESC");
262
263	if (pszDev && RTStrICmpAscii(pszDev, "null") != 0)
264	{
265	/* Detect and log presence of pulse audio plugin. */
266	if (RTStrIStr("pulse", pszDev) != NULL)
267	LogRel(("ALSA: The ALSAAudio plugin for pulse audio is being used (%s).\n", pszDev));
268
269	/*
270	* Add an entry to the enumeration result.
271	* We engage in some trickery here to deal with device names that
272	* are more than 63 characters long.
273	*/
274	size_t const cbId = pszDev ? strlen(pszDev) + 1 : 1;
275	size_t const cbName = pszDesc ? strlen(pszDesc) + 2 + 1 : cbId;
276	PPDMAUDIOHOSTDEV pDev = PDMAudioHostDevAlloc(sizeof(*pDev), cbName, cbId);
277	if (pDev)
278	{
279	RTStrCopy(pDev->pszId, cbId, pszDev);
280	if (pDev->pszId)
281	{
282	pDev->fFlags = PDMAUDIOHOSTDEV_F_NONE;
283	pDev->enmType = PDMAUDIODEVICETYPE_UNKNOWN;
284
285	if (pszInOutId == NULL)
286	{
287	pDev->enmUsage = PDMAUDIODIR_DUPLEX;
288	pDev->cMaxInputChannels = 2;
289	pDev->cMaxOutputChannels = 2;
290	}
291	else if (RTStrICmpAscii(pszInOutId, "Input") == 0)
292	{
293	pDev->enmUsage = PDMAUDIODIR_IN;
294	pDev->cMaxInputChannels = 2;
295	pDev->cMaxOutputChannels = 0;
296	}
297	else
298	{
299	AssertMsg(RTStrICmpAscii(pszInOutId, "Output") == 0, ("%s (%s)\n", pszInOutId, pszHint));
300	pDev->enmUsage = PDMAUDIODIR_OUT;
301	pDev->cMaxInputChannels = 0;
302	pDev->cMaxOutputChannels = 2;
303	}
304
305	if (pszDesc && *pszDesc)
306	{
307	char *pszDesc2 = strchr(pszDesc, '\n');
308	if (!pszDesc2)
309	RTStrCopy(pDev->pszName, cbName, pszDesc);
310	else
311	{
312	*pszDesc2++ = '\0';
313	char *psz;
314	while ((psz = strchr(pszDesc2, '\n')) != NULL)
315	*psz = ' ';
316	RTStrPrintf(pDev->pszName, cbName, "%s (%s)", pszDesc2, pszDesc);
317	}
318	}
319	else
320	RTStrCopy(pDev->pszName, cbName, pszDev);
321
322	PDMAudioHostEnumAppend(pDeviceEnum, pDev);
323
324	LogRel2(("ALSA: Device #%u: '%s' enmDir=%s: %s\n", iHint, pszDev,
325	PDMAudioDirGetName(pDev->enmUsage), pszDesc));
326	}
327	else
328	{
329	PDMAudioHostDevFree(pDev);
330	rc = VERR_NO_STR_MEMORY;
331	}
332	}
333	else
334	rc = VERR_NO_MEMORY;
335	}
336
337	/*
338	* Clean up.
339	*/
340	if (pszInOutId)
341	free(pszInOutId);
342	if (pszDesc)
343	free(pszDesc);
344	if (pszDev)
345	free(pszDev);
346	}
347
348	snd_device_name_free_hint((void **)papszHints);
349
350	if (RT_FAILURE(rc))
351	{
352	PDMAudioHostEnumDelete(pDeviceEnum);
353	PDMAudioHostEnumInit(pDeviceEnum);
354	}
355	}
356	else
357	{
358	int rc2 = RTErrConvertFromErrno(-rc);
359	LogRel2(("ALSA: Error enumerating PCM devices: %Rrc (%d)\n", rc2, rc));
360	rc = rc2;
361	}
362	return rc;
363	}
364
365
366	/**
367	* @interface_method_impl{PDMIHOSTAUDIO,pfnSetDevice}
368	*/
369	static DECLCALLBACK(int) drvHstAudAlsaHA_SetDevice(PPDMIHOSTAUDIO pInterface, PDMAUDIODIR enmDir, const char *pszId)
370	{
371	PDRVHSTAUDALSA pThis = RT_FROM_MEMBER(pInterface, DRVHSTAUDALSA, IHostAudio);
372
373	/*
374	* Validate and normalize input.
375	*/
376	AssertReturn(enmDir == PDMAUDIODIR_IN \|\| enmDir == PDMAUDIODIR_OUT \|\| enmDir == PDMAUDIODIR_DUPLEX, VERR_INVALID_PARAMETER);
377	AssertPtrNullReturn(pszId, VERR_INVALID_POINTER);
378	if (!pszId \|\| !*pszId)
379	pszId = "default";
380	else
381	{
382	size_t cch = strlen(pszId);
383	AssertReturn(cch < sizeof(pThis->szInputDev), VERR_INVALID_NAME);
384	}
385	LogFunc(("enmDir=%d pszId=%s\n", enmDir, pszId));
386
387	/*
388	* Update input.
389	*/
390	if (enmDir == PDMAUDIODIR_IN \|\| enmDir == PDMAUDIODIR_DUPLEX)
391	{
392	int rc = RTCritSectEnter(&pThis->CritSect);
393	AssertRCReturn(rc, rc);
394	if (strcmp(pThis->szInputDev, pszId) == 0)
395	RTCritSectLeave(&pThis->CritSect);
396	else
397	{
398	LogRel(("ALSA: Changing input device: '%s' -> '%s'\n", pThis->szInputDev, pszId));
399	RTStrCopy(pThis->szInputDev, sizeof(pThis->szInputDev), pszId);
400	PPDMIHOSTAUDIOPORT pIHostAudioPort = pThis->pIHostAudioPort;
401	RTCritSectLeave(&pThis->CritSect);
402	if (pIHostAudioPort)
403	{
404	LogFlowFunc(("Notifying parent driver about input device change...\n"));
405	pIHostAudioPort->pfnNotifyDeviceChanged(pIHostAudioPort, PDMAUDIODIR_IN, NULL /pvUser/);
406	}
407	}
408	}
409
410	/*
411	* Update output.
412	*/
413	if (enmDir == PDMAUDIODIR_OUT \|\| enmDir == PDMAUDIODIR_DUPLEX)
414	{
415	int rc = RTCritSectEnter(&pThis->CritSect);
416	AssertRCReturn(rc, rc);
417	if (strcmp(pThis->szOutputDev, pszId) == 0)
418	RTCritSectLeave(&pThis->CritSect);
419	else
420	{
421	LogRel(("ALSA: Changing output device: '%s' -> '%s'\n", pThis->szOutputDev, pszId));
422	RTStrCopy(pThis->szOutputDev, sizeof(pThis->szOutputDev), pszId);
423	PPDMIHOSTAUDIOPORT pIHostAudioPort = pThis->pIHostAudioPort;
424	RTCritSectLeave(&pThis->CritSect);
425	if (pIHostAudioPort)
426	{
427	LogFlowFunc(("Notifying parent driver about output device change...\n"));
428	pIHostAudioPort->pfnNotifyDeviceChanged(pIHostAudioPort, PDMAUDIODIR_OUT, NULL /pvUser/);
429	}
430	}
431	}
432
433	return VINF_SUCCESS;
434	}
435
436
437	/**
438	* @interface_method_impl{PDMIHOSTAUDIO,pfnGetStatus}
439	*/
440	static DECLCALLBACK(PDMAUDIOBACKENDSTS) drvHstAudAlsaHA_GetStatus(PPDMIHOSTAUDIO pInterface, PDMAUDIODIR enmDir)
441	{
442	RT_NOREF(enmDir);
443	AssertPtrReturn(pInterface, PDMAUDIOBACKENDSTS_UNKNOWN);
444
445	return PDMAUDIOBACKENDSTS_RUNNING;
446	}
447
448
449	/**
450	* Converts internal audio PCM properties to an ALSA PCM format.
451	*
452	* @returns Converted ALSA PCM format.
453	* @param pProps Internal audio PCM configuration to convert.
454	*/
455	static snd_pcm_format_t alsaAudioPropsToALSA(PCPDMAUDIOPCMPROPS pProps)
456	{
457	switch (PDMAudioPropsSampleSize(pProps))
458	{
459	case 1:
460	return pProps->fSigned ? SND_PCM_FORMAT_S8 : SND_PCM_FORMAT_U8;
461
462	case 2:
463	if (PDMAudioPropsIsLittleEndian(pProps))
464	return pProps->fSigned ? SND_PCM_FORMAT_S16_LE : SND_PCM_FORMAT_U16_LE;
465	return pProps->fSigned ? SND_PCM_FORMAT_S16_BE : SND_PCM_FORMAT_U16_BE;
466
467	case 4:
468	if (PDMAudioPropsIsLittleEndian(pProps))
469	return pProps->fSigned ? SND_PCM_FORMAT_S32_LE : SND_PCM_FORMAT_U32_LE;
470	return pProps->fSigned ? SND_PCM_FORMAT_S32_BE : SND_PCM_FORMAT_U32_BE;
471
472	default:
473	AssertLogRelMsgFailed(("%RU8 bytes not supported\n", PDMAudioPropsSampleSize(pProps)));
474	return SND_PCM_FORMAT_UNKNOWN;
475	}
476	}
477
478
479	/**
480	* Sets the software parameters of an ALSA stream.
481	*
482	* @returns 0 on success, negative errno on failure.
483	* @param hPCM ALSA stream to set software parameters for.
484	* @param pCfgReq Requested stream configuration (PDM).
485	* @param pCfgAcq The actual stream configuration (PDM). Updated as
486	* needed.
487	*/
488	static int alsaStreamSetSWParams(snd_pcm_t *hPCM, PCPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq)
489	{
490	if (pCfgReq->enmDir == PDMAUDIODIR_IN) /* For input streams there's nothing to do in here right now. */
491	return 0;
492
493	snd_pcm_sw_params_t *pSWParms = NULL;
494	snd_pcm_sw_params_alloca(&pSWParms);
495	AssertReturn(pSWParms, -ENOMEM);
496
497	int err = snd_pcm_sw_params_current(hPCM, pSWParms);
498	AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to get current software parameters: %s\n", snd_strerror(err)), err);
499
500	/* Under normal circumstance, we don't need to set a playback threshold
501	because DrvAudio will do the pre-buffering and hand us everything in
502	one continuous chunk when we should start playing. But since it is
503	configurable, we'll set a reasonable minimum of two DMA periods or
504	max 50 milliseconds (the pAlsaCfgReq->threshold value).
505
506	Of course we also have to make sure the threshold is below the buffer
507	size, or ALSA will never start playing. */
508	unsigned long const cFramesMax = PDMAudioPropsMilliToFrames(&pCfgAcq->Props, 50);
509	unsigned long cFramesThreshold = RT_MIN(pCfgAcq->Backend.cFramesPeriod * 2, cFramesMax);
510	if (cFramesThreshold >= pCfgAcq->Backend.cFramesBufferSize - pCfgAcq->Backend.cFramesBufferSize / 16)
511	cFramesThreshold = pCfgAcq->Backend.cFramesBufferSize - pCfgAcq->Backend.cFramesBufferSize / 16;
512
513	err = snd_pcm_sw_params_set_start_threshold(hPCM, pSWParms, cFramesThreshold);
514	AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set software threshold to %lu: %s\n", cFramesThreshold, snd_strerror(err)), err);
515
516	err = snd_pcm_sw_params_set_avail_min(hPCM, pSWParms, pCfgReq->Backend.cFramesPeriod);
517	AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set available minimum to %u: %s\n",
518	pCfgReq->Backend.cFramesPeriod, snd_strerror(err)), err);
519
520	/* Commit the software parameters: */
521	err = snd_pcm_sw_params(hPCM, pSWParms);
522	AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set new software parameters: %s\n", snd_strerror(err)), err);
523
524	/* Get the actual parameters: */
525	snd_pcm_uframes_t cFramesThresholdActual = cFramesThreshold;
526	err = snd_pcm_sw_params_get_start_threshold(pSWParms, &cFramesThresholdActual);
527	AssertLogRelMsgStmt(err >= 0, ("ALSA: Failed to get start threshold: %s\n", snd_strerror(err)),
528	cFramesThresholdActual = cFramesThreshold);
529
530	LogRel2(("ALSA: SW params: %lu frames threshold, %u frames avail minimum\n",
531	cFramesThresholdActual, pCfgAcq->Backend.cFramesPeriod));
532	return 0;
533	}
534
535
536	/**
537	* Maps a PDM channel ID to an ASLA channel map position.
538	*/
539	static unsigned int drvHstAudAlsaPdmChToAlsa(PDMAUDIOCHANNELID enmId, uint8_t cChannels)
540	{
541	switch (enmId)
542	{
543	case PDMAUDIOCHANNELID_UNKNOWN: return SND_CHMAP_UNKNOWN;
544	case PDMAUDIOCHANNELID_UNUSED_ZERO: return SND_CHMAP_NA;
545	case PDMAUDIOCHANNELID_UNUSED_SILENCE: return SND_CHMAP_NA;
546
547	case PDMAUDIOCHANNELID_FRONT_LEFT: return SND_CHMAP_FL;
548	case PDMAUDIOCHANNELID_FRONT_RIGHT: return SND_CHMAP_FR;
549	case PDMAUDIOCHANNELID_FRONT_CENTER: return cChannels == 1 ? SND_CHMAP_MONO : SND_CHMAP_FC;
550	case PDMAUDIOCHANNELID_LFE: return SND_CHMAP_LFE;
551	case PDMAUDIOCHANNELID_REAR_LEFT: return SND_CHMAP_RL;
552	case PDMAUDIOCHANNELID_REAR_RIGHT: return SND_CHMAP_RR;
553	case PDMAUDIOCHANNELID_FRONT_LEFT_OF_CENTER: return SND_CHMAP_FLC;
554	case PDMAUDIOCHANNELID_FRONT_RIGHT_OF_CENTER: return SND_CHMAP_FRC;
555	case PDMAUDIOCHANNELID_REAR_CENTER: return SND_CHMAP_RC;
556	case PDMAUDIOCHANNELID_SIDE_LEFT: return SND_CHMAP_SL;
557	case PDMAUDIOCHANNELID_SIDE_RIGHT: return SND_CHMAP_SR;
558	case PDMAUDIOCHANNELID_TOP_CENTER: return SND_CHMAP_TC;
559	case PDMAUDIOCHANNELID_FRONT_LEFT_HEIGHT: return SND_CHMAP_TFL;
560	case PDMAUDIOCHANNELID_FRONT_CENTER_HEIGHT: return SND_CHMAP_TFC;
561	case PDMAUDIOCHANNELID_FRONT_RIGHT_HEIGHT: return SND_CHMAP_TFR;
562	case PDMAUDIOCHANNELID_REAR_LEFT_HEIGHT: return SND_CHMAP_TRL;
563	case PDMAUDIOCHANNELID_REAR_CENTER_HEIGHT: return SND_CHMAP_TRC;
564	case PDMAUDIOCHANNELID_REAR_RIGHT_HEIGHT: return SND_CHMAP_TRR;
565
566	case PDMAUDIOCHANNELID_INVALID:
567	case PDMAUDIOCHANNELID_END:
568	case PDMAUDIOCHANNELID_32BIT_HACK:
569	break;
570	}
571	AssertFailed();
572	return SND_CHMAP_NA;
573	}
574
575
576	/**
577	* Sets the hardware parameters of an ALSA stream.
578	*
579	* @returns 0 on success, negative errno on failure.
580	* @param hPCM ALSA stream to set software parameters for.
581	* @param enmAlsaFmt The ALSA format to use.
582	* @param pCfgReq Requested stream configuration (PDM).
583	* @param pCfgAcq The actual stream configuration (PDM). This is assumed
584	* to be a copy of pCfgReq on input, at least for
585	* properties handled here. On output some of the
586	* properties may be updated to match the actual stream
587	* configuration.
588	*/
589	static int alsaStreamSetHwParams(snd_pcm_t *hPCM, snd_pcm_format_t enmAlsaFmt,
590	PCPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq)
591	{
592	/*
593	* Get the current hardware parameters.
594	*/
595	snd_pcm_hw_params_t *pHWParms = NULL;
596	snd_pcm_hw_params_alloca(&pHWParms);
597	AssertReturn(pHWParms, -ENOMEM);
598
599	int err = snd_pcm_hw_params_any(hPCM, pHWParms);
600	AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to initialize hardware parameters: %s\n", snd_strerror(err)), err);
601
602	/*
603	* Modify them according to pAlsaCfgReq.
604	* We update pAlsaCfgObt as we go for parameters set by "near" methods.
605	*/
606	/* We'll use snd_pcm_writei/snd_pcm_readi: */
607	err = snd_pcm_hw_params_set_access(hPCM, pHWParms, SND_PCM_ACCESS_RW_INTERLEAVED);
608	AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set access type: %s\n", snd_strerror(err)), err);
609
610	/* Set the format and frequency. */
611	err = snd_pcm_hw_params_set_format(hPCM, pHWParms, enmAlsaFmt);
612	AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set audio format to %d: %s\n", enmAlsaFmt, snd_strerror(err)), err);
613
614	unsigned int uFreq = PDMAudioPropsHz(&pCfgReq->Props);
615	err = snd_pcm_hw_params_set_rate_near(hPCM, pHWParms, &uFreq, NULL /dir/);
616	AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set frequency to %uHz: %s\n",
617	PDMAudioPropsHz(&pCfgReq->Props), snd_strerror(err)), err);
618	pCfgAcq->Props.uHz = uFreq;
619
620	/* Channel count currently does not change with the mapping translations,
621	as ALSA can express both silent and unknown channel positions. */
622	union
623	{
624	snd_pcm_chmap_t Map;
625	unsigned int padding[1 + PDMAUDIO_MAX_CHANNELS];
626	} u;
627	uint8_t aidSrcChannels[PDMAUDIO_MAX_CHANNELS];
628	unsigned int *aidDstChannels = u.Map.pos;
629	unsigned int cChannels = u.Map.channels = PDMAudioPropsChannels(&pCfgReq->Props);
630	unsigned int iDst = 0;
631	for (unsigned int iSrc = 0; iSrc < cChannels; iSrc++)
632	{
633	uint8_t const idSrc = pCfgReq->Props.aidChannels[iSrc];
634	aidSrcChannels[iDst] = idSrc;
635	aidDstChannels[iDst] = drvHstAudAlsaPdmChToAlsa((PDMAUDIOCHANNELID)idSrc, cChannels);
636	iDst++;
637	}
638	u.Map.channels = cChannels = iDst;
639	for (; iDst < PDMAUDIO_MAX_CHANNELS; iDst++)
640	{
641	aidSrcChannels[iDst] = PDMAUDIOCHANNELID_INVALID;
642	aidDstChannels[iDst] = SND_CHMAP_NA;
643	}
644
645	err = snd_pcm_hw_params_set_channels_near(hPCM, pHWParms, &cChannels);
646	AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set number of channels to %d\n", PDMAudioPropsChannels(&pCfgReq->Props)),
647	err);
648	if (cChannels == PDMAudioPropsChannels(&pCfgReq->Props))
649	memcpy(pCfgAcq->Props.aidChannels, aidSrcChannels, sizeof(pCfgAcq->Props.aidChannels));
650	else
651	{
652	LogRel2(("ALSA: Requested %u channels, got %u\n", u.Map.channels, cChannels));
653	AssertLogRelMsgReturn(cChannels > 0 && cChannels <= PDMAUDIO_MAX_CHANNELS,
654	("ALSA: Unsupported channel count: %u (requested %d)\n",
655	cChannels, PDMAudioPropsChannels(&pCfgReq->Props)), -ERANGE);
656	PDMAudioPropsSetChannels(&pCfgAcq->Props, (uint8_t)cChannels);
657	/** @todo Can we somehow guess channel IDs? snd_pcm_get_chmap? */
658	}
659
660	/* The period size (reportedly frame count per hw interrupt): */
661	int dir = 0;
662	snd_pcm_uframes_t minval = pCfgReq->Backend.cFramesPeriod;
663	err = snd_pcm_hw_params_get_period_size_min(pHWParms, &minval, &dir);
664	AssertLogRelMsgReturn(err >= 0, ("ALSA: Could not determine minimal period size: %s\n", snd_strerror(err)), err);
665
666	snd_pcm_uframes_t period_size_f = pCfgReq->Backend.cFramesPeriod;
667	if (period_size_f < minval)
668	period_size_f = minval;
669	err = snd_pcm_hw_params_set_period_size_near(hPCM, pHWParms, &period_size_f, 0);
670	LogRel2(("ALSA: Period size is: %lu frames (min %lu, requested %u)\n", period_size_f, minval, pCfgReq->Backend.cFramesPeriod));
671	AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set period size %d (%s)\n", period_size_f, snd_strerror(err)), err);
672
673	/* The buffer size: */
674	minval = pCfgReq->Backend.cFramesBufferSize;
675	err = snd_pcm_hw_params_get_buffer_size_min(pHWParms, &minval);
676	AssertLogRelMsgReturn(err >= 0, ("ALSA: Could not retrieve minimal buffer size: %s\n", snd_strerror(err)), err);
677
678	snd_pcm_uframes_t buffer_size_f = pCfgReq->Backend.cFramesBufferSize;
679	if (buffer_size_f < minval)
680	buffer_size_f = minval;
681	err = snd_pcm_hw_params_set_buffer_size_near(hPCM, pHWParms, &buffer_size_f);
682	LogRel2(("ALSA: Buffer size is: %lu frames (min %lu, requested %u)\n", buffer_size_f, minval, pCfgReq->Backend.cFramesBufferSize));
683	AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to set near buffer size %RU32: %s\n", buffer_size_f, snd_strerror(err)), err);
684
685	/*
686	* Set the hardware parameters.
687	*/
688	err = snd_pcm_hw_params(hPCM, pHWParms);
689	AssertLogRelMsgReturn(err >= 0, ("ALSA: Failed to apply audio parameters: %s\n", snd_strerror(err)), err);
690
691	/*
692	* Get relevant parameters and put them in the pAlsaCfgObt structure.
693	*/
694	snd_pcm_uframes_t obt_buffer_size = buffer_size_f;
695	err = snd_pcm_hw_params_get_buffer_size(pHWParms, &obt_buffer_size);
696	AssertLogRelMsgStmt(err >= 0, ("ALSA: Failed to get buffer size: %s\n", snd_strerror(err)), obt_buffer_size = buffer_size_f);
697	pCfgAcq->Backend.cFramesBufferSize = obt_buffer_size;
698
699	snd_pcm_uframes_t obt_period_size = period_size_f;
700	err = snd_pcm_hw_params_get_period_size(pHWParms, &obt_period_size, &dir);
701	AssertLogRelMsgStmt(err >= 0, ("ALSA: Failed to get period size: %s\n", snd_strerror(err)), obt_period_size = period_size_f);
702	pCfgAcq->Backend.cFramesPeriod = obt_period_size;
703
704	LogRel2(("ALSA: HW params: %u Hz, %u frames period, %u frames buffer, %u channel(s), enmAlsaFmt=%d\n",
705	PDMAudioPropsHz(&pCfgAcq->Props), pCfgAcq->Backend.cFramesPeriod, pCfgAcq->Backend.cFramesBufferSize,
706	PDMAudioPropsChannels(&pCfgAcq->Props), enmAlsaFmt));
707
708	#if 0 /* Disabled in the hope to resolve testboxes not being able to drain + crashing when closing the PCM streams. */
709	/*
710	* Channel config (not fatal).
711	*/
712	if (PDMAudioPropsChannels(&pCfgAcq->Props) == PDMAudioPropsChannels(&pCfgReq->Props))
713	{
714	err = snd_pcm_set_chmap(hPCM, &u.Map);
715	if (err < 0)
716	{
717	if (err == -ENXIO)
718	LogRel2(("ALSA: Audio device does not support channel maps, skipping\n"));
719	else
720	LogRel2(("ALSA: snd_pcm_set_chmap failed: %s (%d)\n", snd_strerror(err), err));
721	}
722	}
723	#endif
724
725	return 0;
726	}
727
728
729	/**
730	* Opens (creates) an ALSA stream.
731	*
732	* @returns VBox status code.
733	* @param pThis The alsa driver instance data.
734	* @param enmAlsaFmt The ALSA format to use.
735	* @param pCfgReq Requested configuration to create stream with (PDM).
736	* @param pCfgAcq The actual stream configuration (PDM). This is assumed
737	* to be a copy of pCfgReq on input, at least for
738	* properties handled here. On output some of the
739	* properties may be updated to match the actual stream
740	* configuration.
741	* @param phPCM Where to store the ALSA stream handle on success.
742	*/
743	static int alsaStreamOpen(PDRVHSTAUDALSA pThis, snd_pcm_format_t enmAlsaFmt, PCPDMAUDIOSTREAMCFG pCfgReq,
744	PPDMAUDIOSTREAMCFG pCfgAcq, snd_pcm_t **phPCM)
745	{
746	/*
747	* Open the stream.
748	*/
749	int rc = VERR_AUDIO_STREAM_COULD_NOT_CREATE;
750	const char * const pszType = pCfgReq->enmDir == PDMAUDIODIR_IN ? "input" : "output";
751	const char * const pszDev = pCfgReq->enmDir == PDMAUDIODIR_IN ? pThis->szInputDev : pThis->szOutputDev;
752	snd_pcm_stream_t enmType = pCfgReq->enmDir == PDMAUDIODIR_IN ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK;
753
754	snd_pcm_t *hPCM = NULL;
755	LogRel(("ALSA: Using %s device \"%s\"\n", pszType, pszDev));
756	int err = snd_pcm_open(&hPCM, pszDev, enmType, SND_PCM_NONBLOCK);
757	if (err >= 0)
758	{
759	err = snd_pcm_nonblock(hPCM, 1);
760	if (err >= 0)
761	{
762	/*
763	* Configure hardware stream parameters.
764	*/
765	err = alsaStreamSetHwParams(hPCM, enmAlsaFmt, pCfgReq, pCfgAcq);
766	if (err >= 0)
767	{
768	/*
769	* Prepare it.
770	*/
771	rc = VERR_AUDIO_BACKEND_INIT_FAILED;
772	err = snd_pcm_prepare(hPCM);
773	if (err >= 0)
774	{
775	/*
776	* Configure software stream parameters.
777	*/
778	rc = alsaStreamSetSWParams(hPCM, pCfgReq, pCfgAcq);
779	if (RT_SUCCESS(rc))
780	{
781	*phPCM = hPCM;
782	return VINF_SUCCESS;
783	}
784	}
785	else
786	LogRel(("ALSA: snd_pcm_prepare failed: %s\n", snd_strerror(err)));
787	}
788	}
789	else
790	LogRel(("ALSA: Error setting non-blocking mode for %s stream: %s\n", pszType, snd_strerror(err)));
791	drvHstAudAlsaStreamClose(&hPCM);
792	}
793	else
794	LogRel(("ALSA: Failed to open \"%s\" as %s device: %s\n", pszDev, pszType, snd_strerror(err)));
795	*phPCM = NULL;
796	return rc;
797	}
798
799
800	/**
801	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamCreate}
802	*/
803	static DECLCALLBACK(int) drvHstAudAlsaHA_StreamCreate(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream,
804	PCPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq)
805	{
806	PDRVHSTAUDALSA pThis = RT_FROM_MEMBER(pInterface, DRVHSTAUDALSA, IHostAudio);
807	AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
808	AssertPtrReturn(pStream, VERR_INVALID_POINTER);
809	AssertPtrReturn(pCfgReq, VERR_INVALID_POINTER);
810	AssertPtrReturn(pCfgAcq, VERR_INVALID_POINTER);
811
812	PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream;
813	PDMAudioStrmCfgCopy(&pStreamALSA->Cfg, pCfgReq);
814
815	int rc;
816	snd_pcm_format_t const enmFmt = alsaAudioPropsToALSA(&pCfgReq->Props);
817	if (enmFmt != SND_PCM_FORMAT_UNKNOWN)
818	{
819	rc = alsaStreamOpen(pThis, enmFmt, pCfgReq, pCfgAcq, &pStreamALSA->hPCM);
820	if (RT_SUCCESS(rc))
821	{
822	/* We have no objections to the pre-buffering that DrvAudio applies,
823	only we need to adjust it relative to the actual buffer size. */
824	pCfgAcq->Backend.cFramesPreBuffering = (uint64_t)pCfgReq->Backend.cFramesPreBuffering
825	* pCfgAcq->Backend.cFramesBufferSize
826	/ RT_MAX(pCfgReq->Backend.cFramesBufferSize, 1);
827
828	PDMAudioStrmCfgCopy(&pStreamALSA->Cfg, pCfgAcq);
829	LogFlowFunc(("returns success - hPCM=%p\n", pStreamALSA->hPCM));
830	return rc;
831	}
832	}
833	else
834	rc = VERR_AUDIO_STREAM_COULD_NOT_CREATE;
835	LogFunc(("returns %Rrc\n", rc));
836	return rc;
837	}
838
839
840	/**
841	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamDestroy}
842	*/
843	static DECLCALLBACK(int) drvHstAudAlsaHA_StreamDestroy(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream, bool fImmediate)
844	{
845	RT_NOREF(pInterface);
846	PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream;
847	AssertPtrReturn(pStreamALSA, VERR_INVALID_POINTER);
848	RT_NOREF(fImmediate);
849
850	LogRelFlowFunc(("Stream '%s' state is '%s'\n", pStreamALSA->Cfg.szName, snd_pcm_state_name(snd_pcm_state(pStreamALSA->hPCM))));
851
852	/** @todo r=bird: It's not like we can do much with a bad status... Check
853	* what the caller does... */
854	int rc = drvHstAudAlsaStreamClose(&pStreamALSA->hPCM);
855
856	LogRelFlowFunc(("returns %Rrc\n", rc));
857
858	return rc;
859	}
860
861
862	/**
863	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamEnable}
864	*/
865	static DECLCALLBACK(int) drvHstAudAlsaHA_StreamEnable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
866	{
867	RT_NOREF(pInterface);
868	PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream;
869
870	/*
871	* Prepare the stream.
872	*/
873	int rc = snd_pcm_prepare(pStreamALSA->hPCM);
874	if (rc >= 0)
875	{
876	Assert(snd_pcm_state(pStreamALSA->hPCM) == SND_PCM_STATE_PREPARED);
877
878	/*
879	* Input streams should be started now, whereas output streams must
880	* pre-buffer sufficent data before starting.
881	*/
882	if (pStreamALSA->Cfg.enmDir == PDMAUDIODIR_IN)
883	{
884	rc = snd_pcm_start(pStreamALSA->hPCM);
885	if (rc >= 0)
886	rc = VINF_SUCCESS;
887	else
888	{
889	LogRel(("ALSA: Error starting input stream '%s': %s (%d)\n", pStreamALSA->Cfg.szName, snd_strerror(rc), rc));
890	rc = RTErrConvertFromErrno(-rc);
891	}
892	}
893	else
894	rc = VINF_SUCCESS;
895	}
896	else
897	{
898	LogRel(("ALSA: Error preparing stream '%s': %s (%d)\n", pStreamALSA->Cfg.szName, snd_strerror(rc), rc));
899	rc = RTErrConvertFromErrno(-rc);
900	}
901	LogFlowFunc(("returns %Rrc (state %s)\n", rc, snd_pcm_state_name(snd_pcm_state(pStreamALSA->hPCM))));
902	return rc;
903	}
904
905
906	/**
907	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamDisable}
908	*/
909	static DECLCALLBACK(int) drvHstAudAlsaHA_StreamDisable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
910	{
911	RT_NOREF(pInterface);
912	PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream;
913
914	int rc = snd_pcm_drop(pStreamALSA->hPCM);
915	if (rc >= 0)
916	rc = VINF_SUCCESS;
917	else
918	{
919	LogRel(("ALSA: Error stopping stream '%s': %s (%d)\n", pStreamALSA->Cfg.szName, snd_strerror(rc), rc));
920	rc = RTErrConvertFromErrno(-rc);
921	}
922	LogFlowFunc(("returns %Rrc (state %s)\n", rc, snd_pcm_state_name(snd_pcm_state(pStreamALSA->hPCM))));
923	return rc;
924	}
925
926
927	/**
928	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamPause}
929	*/
930	static DECLCALLBACK(int) drvHstAudAlsaHA_StreamPause(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
931	{
932	/* Same as disable. */
933	/** @todo r=bird: Try use pause and fallback on disable/enable if it isn't
934	* supported or doesn't work. */
935	return drvHstAudAlsaHA_StreamDisable(pInterface, pStream);
936	}
937
938
939	/**
940	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamResume}
941	*/
942	static DECLCALLBACK(int) drvHstAudAlsaHA_StreamResume(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
943	{
944	/* Same as enable. */
945	return drvHstAudAlsaHA_StreamEnable(pInterface, pStream);
946	}
947
948
949	/**
950	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamDrain}
951	*/
952	static DECLCALLBACK(int) drvHstAudAlsaHA_StreamDrain(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
953	{
954	RT_NOREF(pInterface);
955	PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream;
956
957	snd_pcm_state_t const enmState = snd_pcm_state(pStreamALSA->hPCM);
958	LogRelFlowFunc(("Stream '%s' input state: %s (%d)\n", pStreamALSA->Cfg.szName, snd_pcm_state_name(enmState), enmState));
959
960	/* Only for output streams. */
961	AssertReturn(pStreamALSA->Cfg.enmDir == PDMAUDIODIR_OUT, VERR_WRONG_ORDER);
962
963	int rc;
964	switch (enmState)
965	{
966	case SND_PCM_STATE_RUNNING:
967	case SND_PCM_STATE_PREPARED: /* not yet started */
968	{
969	/* Do not change to blocking here! */
970	rc = snd_pcm_drain(pStreamALSA->hPCM);
971	if (rc >= 0 \|\| rc == -EAGAIN)
972	rc = VINF_SUCCESS;
973	else
974	{
975	snd_pcm_state_t const enmState2 = snd_pcm_state(pStreamALSA->hPCM);
976	if (rc == -EPIPE && enmState2 == enmState)
977	{
978	/* Not entirely sure, but possibly an underrun, so just disable the stream. */
979	LogRel2(("ALSA: snd_pcm_drain failed with -EPIPE, stopping stream (%s)\n", pStreamALSA->Cfg.szName));
980	rc = snd_pcm_drop(pStreamALSA->hPCM);
981	if (rc >= 0)
982	rc = VINF_SUCCESS;
983	else
984	{
985	LogRel(("ALSA: Error draining/stopping stream '%s': %s (%d)\n", pStreamALSA->Cfg.szName, snd_strerror(rc), rc));
986	rc = RTErrConvertFromErrno(-rc);
987	}
988	}
989	else
990	{
991	LogRel(("ALSA: Error draining output of '%s': %s (%d; %s -> %s)\n", pStreamALSA->Cfg.szName, snd_strerror(rc),
992	rc, snd_pcm_state_name(enmState), snd_pcm_state_name(enmState2)));
993	rc = RTErrConvertFromErrno(-rc);
994	}
995	}
996	break;
997	}
998
999	default:
1000	rc = VINF_SUCCESS;
1001	break;
1002	}
1003	LogRelFlowFunc(("returns %Rrc (state %s)\n", rc, snd_pcm_state_name(snd_pcm_state(pStreamALSA->hPCM))));
1004	return rc;
1005	}
1006
1007
1008	/**
1009	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetState}
1010	*/
1011	static DECLCALLBACK(PDMHOSTAUDIOSTREAMSTATE) drvHstAudAlsaHA_StreamGetState(PPDMIHOSTAUDIO pInterface,
1012	PPDMAUDIOBACKENDSTREAM pStream)
1013	{
1014	RT_NOREF(pInterface);
1015	PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream;
1016	AssertPtrReturn(pStreamALSA, PDMHOSTAUDIOSTREAMSTATE_INVALID);
1017
1018	PDMHOSTAUDIOSTREAMSTATE enmStreamState = PDMHOSTAUDIOSTREAMSTATE_OKAY;
1019	snd_pcm_state_t enmAlsaState = snd_pcm_state(pStreamALSA->hPCM);
1020	if (enmAlsaState == SND_PCM_STATE_DRAINING)
1021	{
1022	/* We're operating in non-blocking mode, so we must (at least for a demux
1023	config) call snd_pcm_drain again to drive it forward. Otherwise we
1024	might be stuck in the drain state forever. */
1025	Log5Func(("Calling snd_pcm_drain again...\n"));
1026	snd_pcm_drain(pStreamALSA->hPCM);
1027	enmAlsaState = snd_pcm_state(pStreamALSA->hPCM);
1028	}
1029
1030	if (enmAlsaState == SND_PCM_STATE_DRAINING)
1031	enmStreamState = PDMHOSTAUDIOSTREAMSTATE_DRAINING;
1032	#if (((SND_LIB_MAJOR) << 16) \| ((SND_LIB_MAJOR) << 8) \| (SND_LIB_SUBMINOR)) >= 0x10002 /* was added in 1.0.2 */
1033	else if (enmAlsaState == SND_PCM_STATE_DISCONNECTED)
1034	enmStreamState = PDMHOSTAUDIOSTREAMSTATE_NOT_WORKING;
1035	#endif
1036
1037	Log5Func(("Stream '%s': ALSA state=%s -> %s\n",
1038	pStreamALSA->Cfg.szName, snd_pcm_state_name(enmAlsaState), PDMHostAudioStreamStateGetName(enmStreamState) ));
1039	return enmStreamState;
1040	}
1041
1042
1043	/**
1044	* Returns the available audio frames queued.
1045	*
1046	* @returns VBox status code.
1047	* @param hPCM ALSA stream handle.
1048	* @param pcFramesAvail Where to store the available frames.
1049	*/
1050	static int alsaStreamGetAvail(snd_pcm_t hPCM, snd_pcm_sframes_t pcFramesAvail)
1051	{
1052	AssertPtr(hPCM);
1053	AssertPtr(pcFramesAvail);
1054
1055	int rc;
1056	snd_pcm_sframes_t cFramesAvail = snd_pcm_avail_update(hPCM);
1057	if (cFramesAvail > 0)
1058	{
1059	LogFunc(("cFramesAvail=%ld\n", cFramesAvail));
1060	*pcFramesAvail = cFramesAvail;
1061	return VINF_SUCCESS;
1062	}
1063
1064	/*
1065	* We can maybe recover from an EPIPE...
1066	*/
1067	if (cFramesAvail == -EPIPE)
1068	{
1069	rc = drvHstAudAlsaStreamRecover(hPCM);
1070	if (RT_SUCCESS(rc))
1071	{
1072	cFramesAvail = snd_pcm_avail_update(hPCM);
1073	if (cFramesAvail >= 0)
1074	{
1075	LogFunc(("cFramesAvail=%ld\n", cFramesAvail));
1076	*pcFramesAvail = cFramesAvail;
1077	return VINF_SUCCESS;
1078	}
1079	}
1080	else
1081	{
1082	*pcFramesAvail = 0;
1083	return rc;
1084	}
1085	}
1086
1087	rc = RTErrConvertFromErrno(-(int)cFramesAvail);
1088	LogFunc(("failed - cFramesAvail=%ld rc=%Rrc\n", cFramesAvail, rc));
1089	*pcFramesAvail = 0;
1090	return rc;
1091	}
1092
1093
1094	/**
1095	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetPending}
1096	*/
1097	static DECLCALLBACK(uint32_t) drvHstAudAlsaHA_StreamGetPending(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
1098	{
1099	RT_NOREF(pInterface);
1100	PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream;
1101	AssertPtrReturn(pStreamALSA, 0);
1102
1103	/*
1104	* This is only relevant to output streams (input streams can't have
1105	* any pending, unplayed data).
1106	*/
1107	uint32_t cbPending = 0;
1108	if (pStreamALSA->Cfg.enmDir == PDMAUDIODIR_OUT)
1109	{
1110	/*
1111	* Getting the delay (in audio frames) reports the time it will take
1112	* to hear a new sample after all queued samples have been played out.
1113	*
1114	* We use snd_pcm_avail_delay instead of snd_pcm_delay here as it will
1115	* update the buffer positions, and we can use the extra value against
1116	* the buffer size to double check since the delay value may include
1117	* fixed built-in delays in the processing chain and hardware.
1118	*/
1119	snd_pcm_sframes_t cFramesAvail = 0;
1120	snd_pcm_sframes_t cFramesDelay = 0;
1121	int rc = snd_pcm_avail_delay(pStreamALSA->hPCM, &cFramesAvail, &cFramesDelay);
1122
1123	/*
1124	* We now also get the state as the pending value should be zero when
1125	* we're not in a playing state.
1126	*/
1127	snd_pcm_state_t enmState = snd_pcm_state(pStreamALSA->hPCM);
1128	switch (enmState)
1129	{
1130	case SND_PCM_STATE_RUNNING:
1131	case SND_PCM_STATE_DRAINING:
1132	if (rc >= 0)
1133	{
1134	if ((uint32_t)cFramesAvail >= pStreamALSA->Cfg.Backend.cFramesBufferSize)
1135	cbPending = 0;
1136	else
1137	cbPending = PDMAudioPropsFramesToBytes(&pStreamALSA->Cfg.Props, cFramesDelay);
1138	}
1139	break;
1140
1141	default:
1142	break;
1143	}
1144	Log2Func(("returns %u (%#x) - cFramesBufferSize=%RU32 cFramesAvail=%ld cFramesDelay=%ld rc=%d; enmState=%s (%d) \n",
1145	cbPending, cbPending, pStreamALSA->Cfg.Backend.cFramesBufferSize, cFramesAvail, cFramesDelay, rc,
1146	snd_pcm_state_name(enmState), enmState));
1147	}
1148	return cbPending;
1149	}
1150
1151
1152	/**
1153	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetWritable}
1154	*/
1155	static DECLCALLBACK(uint32_t) drvHstAudAlsaHA_StreamGetWritable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
1156	{
1157	RT_NOREF(pInterface);
1158	PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream;
1159
1160	uint32_t cbAvail = 0;
1161	snd_pcm_sframes_t cFramesAvail = 0;
1162	int rc = alsaStreamGetAvail(pStreamALSA->hPCM, &cFramesAvail);
1163	if (RT_SUCCESS(rc))
1164	cbAvail = PDMAudioPropsFramesToBytes(&pStreamALSA->Cfg.Props, cFramesAvail);
1165
1166	return cbAvail;
1167	}
1168
1169
1170	/**
1171	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamPlay}
1172	*/
1173	static DECLCALLBACK(int) drvHstAudAlsaHA_StreamPlay(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream,
1174	const void pvBuf, uint32_t cbBuf, uint32_t pcbWritten)
1175	{
1176	PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream;
1177	AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
1178	AssertPtrReturn(pStream, VERR_INVALID_POINTER);
1179	AssertPtrReturn(pcbWritten, VERR_INVALID_POINTER);
1180	Log4Func(("@%#RX64: pvBuf=%p cbBuf=%#x (%u) state=%s - %s\n", pStreamALSA->offInternal, pvBuf, cbBuf, cbBuf,
1181	snd_pcm_state_name(snd_pcm_state(pStreamALSA->hPCM)), pStreamALSA->Cfg.szName));
1182	if (cbBuf)
1183	AssertPtrReturn(pvBuf, VERR_INVALID_POINTER);
1184	else
1185	{
1186	/* Fend off draining calls. */
1187	*pcbWritten = 0;
1188	return VINF_SUCCESS;
1189	}
1190
1191	/*
1192	* Determine how much we can write (caller actually did this
1193	* already, but we repeat it just to be sure or something).
1194	*/
1195	snd_pcm_sframes_t cFramesAvail;
1196	int rc = alsaStreamGetAvail(pStreamALSA->hPCM, &cFramesAvail);
1197	if (RT_SUCCESS(rc))
1198	{
1199	Assert(cFramesAvail);
1200	if (cFramesAvail)
1201	{
1202	PCPDMAUDIOPCMPROPS pProps = &pStreamALSA->Cfg.Props;
1203	uint32_t cbToWrite = PDMAudioPropsFramesToBytes(pProps, (uint32_t)cFramesAvail);
1204	if (cbToWrite)
1205	{
1206	if (cbToWrite > cbBuf)
1207	cbToWrite = cbBuf;
1208
1209	/*
1210	* Try write the data.
1211	*/
1212	uint32_t cFramesToWrite = PDMAudioPropsBytesToFrames(pProps, cbToWrite);
1213	snd_pcm_sframes_t cFramesWritten = snd_pcm_writei(pStreamALSA->hPCM, pvBuf, cFramesToWrite);
1214	if (cFramesWritten > 0)
1215	{
1216	Log4Func(("snd_pcm_writei w/ cbToWrite=%u -> %ld (frames) [cFramesAvail=%ld]\n",
1217	cbToWrite, cFramesWritten, cFramesAvail));
1218	*pcbWritten = PDMAudioPropsFramesToBytes(pProps, cFramesWritten);
1219	pStreamALSA->offInternal += *pcbWritten;
1220	return VINF_SUCCESS;
1221	}
1222	LogFunc(("snd_pcm_writei w/ cbToWrite=%u -> %ld [cFramesAvail=%ld]\n", cbToWrite, cFramesWritten, cFramesAvail));
1223
1224
1225	/*
1226	* There are a couple of error we can recover from, try to do so.
1227	* Only don't try too many times.
1228	*/
1229	for (unsigned iTry = 0;
1230	(cFramesWritten == -EPIPE \|\| cFramesWritten == -ESTRPIPE) && iTry < ALSA_RECOVERY_TRIES_MAX;
1231	iTry++)
1232	{
1233	if (cFramesWritten == -EPIPE)
1234	{
1235	/* Underrun occurred. */
1236	rc = drvHstAudAlsaStreamRecover(pStreamALSA->hPCM);
1237	if (RT_FAILURE(rc))
1238	break;
1239	LogFlowFunc(("Recovered from playback (iTry=%u)\n", iTry));
1240	}
1241	else
1242	{
1243	/* An suspended event occurred, needs resuming. */
1244	rc = drvHstAudAlsaStreamResume(pStreamALSA->hPCM);
1245	if (RT_FAILURE(rc))
1246	{
1247	LogRel(("ALSA: Failed to resume output stream (iTry=%u, rc=%Rrc)\n", iTry, rc));
1248	break;
1249	}
1250	LogFlowFunc(("Resumed suspended output stream (iTry=%u)\n", iTry));
1251	}
1252
1253	cFramesWritten = snd_pcm_writei(pStreamALSA->hPCM, pvBuf, cFramesToWrite);
1254	if (cFramesWritten > 0)
1255	{
1256	Log4Func(("snd_pcm_writei w/ cbToWrite=%u -> %ld (frames) [cFramesAvail=%ld]\n",
1257	cbToWrite, cFramesWritten, cFramesAvail));
1258	*pcbWritten = PDMAudioPropsFramesToBytes(pProps, cFramesWritten);
1259	pStreamALSA->offInternal += *pcbWritten;
1260	return VINF_SUCCESS;
1261	}
1262	LogFunc(("snd_pcm_writei w/ cbToWrite=%u -> %ld [cFramesAvail=%ld, iTry=%d]\n", cbToWrite, cFramesWritten, cFramesAvail, iTry));
1263	}
1264
1265	/* Make sure we return with an error status. */
1266	if (RT_SUCCESS_NP(rc))
1267	{
1268	if (cFramesWritten == 0)
1269	rc = VERR_ACCESS_DENIED;
1270	else
1271	{
1272	rc = RTErrConvertFromErrno(-(int)cFramesWritten);
1273	LogFunc(("Failed to write %RU32 bytes: %ld (%Rrc)\n", cbToWrite, cFramesWritten, rc));
1274	}
1275	}
1276	}
1277	}
1278	}
1279	else
1280	LogFunc(("Error getting number of playback frames, rc=%Rrc\n", rc));
1281	*pcbWritten = 0;
1282	return rc;
1283	}
1284
1285
1286	/**
1287	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetReadable}
1288	*/
1289	static DECLCALLBACK(uint32_t) drvHstAudAlsaHA_StreamGetReadable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
1290	{
1291	RT_NOREF(pInterface);
1292	PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream;
1293
1294	uint32_t cbAvail = 0;
1295	snd_pcm_sframes_t cFramesAvail = 0;
1296	int rc = alsaStreamGetAvail(pStreamALSA->hPCM, &cFramesAvail);
1297	if (RT_SUCCESS(rc))
1298	cbAvail = PDMAudioPropsFramesToBytes(&pStreamALSA->Cfg.Props, cFramesAvail);
1299
1300	return cbAvail;
1301	}
1302
1303
1304	/**
1305	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamCapture}
1306	*/
1307	static DECLCALLBACK(int) drvHstAudAlsaHA_StreamCapture(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream,
1308	void pvBuf, uint32_t cbBuf, uint32_t pcbRead)
1309	{
1310	RT_NOREF_PV(pInterface);
1311	PDRVHSTAUDALSASTREAM pStreamALSA = (PDRVHSTAUDALSASTREAM)pStream;
1312	AssertPtrReturn(pStreamALSA, VERR_INVALID_POINTER);
1313	AssertPtrReturn(pvBuf, VERR_INVALID_POINTER);
1314	AssertReturn(cbBuf, VERR_INVALID_PARAMETER);
1315	AssertPtrReturn(pcbRead, VERR_INVALID_POINTER);
1316	Log4Func(("@%#RX64: pvBuf=%p cbBuf=%#x (%u) state=%s - %s\n", pStreamALSA->offInternal, pvBuf, cbBuf, cbBuf,
1317	snd_pcm_state_name(snd_pcm_state(pStreamALSA->hPCM)), pStreamALSA->Cfg.szName));
1318
1319	/*
1320	* Figure out how much we can read without trouble (we're doing
1321	* non-blocking reads, but whatever).
1322	*/
1323	snd_pcm_sframes_t cAvail;
1324	int rc = alsaStreamGetAvail(pStreamALSA->hPCM, &cAvail);
1325	if (RT_SUCCESS(rc))
1326	{
1327	if (!cAvail) /* No data yet? */
1328	{
1329	snd_pcm_state_t enmState = snd_pcm_state(pStreamALSA->hPCM);
1330	switch (enmState)
1331	{
1332	case SND_PCM_STATE_PREPARED:
1333	/** @todo r=bird: explain the logic here... */
1334	cAvail = PDMAudioPropsBytesToFrames(&pStreamALSA->Cfg.Props, cbBuf);
1335	break;
1336
1337	case SND_PCM_STATE_SUSPENDED:
1338	rc = drvHstAudAlsaStreamResume(pStreamALSA->hPCM);
1339	if (RT_SUCCESS(rc))
1340	{
1341	LogFlowFunc(("Resumed suspended input stream.\n"));
1342	break;
1343	}
1344	LogFunc(("Failed resuming suspended input stream: %Rrc\n", rc));
1345	return rc;
1346
1347	default:
1348	LogFlow(("No frames available: state=%s (%d)\n", snd_pcm_state_name(enmState), enmState));
1349	break;
1350	}
1351	if (!cAvail)
1352	{
1353	*pcbRead = 0;
1354	return VINF_SUCCESS;
1355	}
1356	}
1357	}
1358	else
1359	{
1360	LogFunc(("Error getting number of captured frames, rc=%Rrc\n", rc));
1361	return rc;
1362	}
1363
1364	size_t cbToRead = PDMAudioPropsFramesToBytes(&pStreamALSA->Cfg.Props, cAvail);
1365	cbToRead = RT_MIN(cbToRead, cbBuf);
1366	LogFlowFunc(("cbToRead=%zu, cAvail=%RI32\n", cbToRead, cAvail));
1367
1368	/*
1369	* Read loop.
1370	*/
1371	uint32_t cbReadTotal = 0;
1372	while (cbToRead > 0)
1373	{
1374	/*
1375	* Do the reading.
1376	*/
1377	snd_pcm_uframes_t const cFramesToRead = PDMAudioPropsBytesToFrames(&pStreamALSA->Cfg.Props, cbToRead);
1378	AssertBreakStmt(cFramesToRead > 0, rc = VERR_NO_DATA);
1379
1380	snd_pcm_sframes_t cFramesRead = snd_pcm_readi(pStreamALSA->hPCM, pvBuf, cFramesToRead);
1381	if (cFramesRead > 0)
1382	{
1383	/*
1384	* We should not run into a full mixer buffer or we lose samples and
1385	* run into an endless loop if ALSA keeps producing samples ("null"
1386	* capture device for example).
1387	*/
1388	uint32_t const cbRead = PDMAudioPropsFramesToBytes(&pStreamALSA->Cfg.Props, cFramesRead);
1389	Assert(cbRead <= cbToRead);
1390
1391	cbToRead -= cbRead;
1392	cbReadTotal += cbRead;
1393	pvBuf = (uint8_t *)pvBuf + cbRead;
1394	pStreamALSA->offInternal += cbRead;
1395	}
1396	else
1397	{
1398	/*
1399	* Try recover from overrun and re-try.
1400	* Other conditions/errors we cannot and will just quit the loop.
1401	*/
1402	if (cFramesRead == -EPIPE)
1403	{
1404	rc = drvHstAudAlsaStreamRecover(pStreamALSA->hPCM);
1405	if (RT_SUCCESS(rc))
1406	{
1407	LogFlowFunc(("Successfully recovered from overrun\n"));
1408	continue;
1409	}
1410	LogFunc(("Failed to recover from overrun: %Rrc\n", rc));
1411	}
1412	else if (cFramesRead == -EAGAIN)
1413	LogFunc(("No input frames available (EAGAIN)\n"));
1414	else if (cFramesRead == 0)
1415	LogFunc(("No input frames available (0)\n"));
1416	else
1417	{
1418	rc = RTErrConvertFromErrno(-(int)cFramesRead);
1419	LogFunc(("Failed to read input frames: %s (%ld, %Rrc)\n", snd_strerror(cFramesRead), cFramesRead, rc));
1420	}
1421
1422	/* If we've read anything, suppress the error. */
1423	if (RT_FAILURE(rc) && cbReadTotal > 0)
1424	{
1425	LogFunc(("Suppressing %Rrc because %#x bytes has been read already\n", rc, cbReadTotal));
1426	rc = VINF_SUCCESS;
1427	}
1428	break;
1429	}
1430	}
1431
1432	LogFlowFunc(("returns %Rrc and %#x (%d) bytes (%u bytes left); state %s\n",
1433	rc, cbReadTotal, cbReadTotal, cbToRead, snd_pcm_state_name(snd_pcm_state(pStreamALSA->hPCM))));
1434	*pcbRead = cbReadTotal;
1435	return rc;
1436	}
1437
1438
1439	/*********************************************************************************************************************************
1440	* PDMIBASE *
1441	*********************************************************************************************************************************/
1442
1443	/**
1444	* @interface_method_impl{PDMIBASE,pfnQueryInterface}
1445	*/
1446	static DECLCALLBACK(void ) drvHstAudAlsaQueryInterface(PPDMIBASE pInterface, const char pszIID)
1447	{
1448	PPDMDRVINS pDrvIns = PDMIBASE_2_PDMDRV(pInterface);
1449	PDRVHSTAUDALSA pThis = PDMINS_2_DATA(pDrvIns, PDRVHSTAUDALSA);
1450	PDMIBASE_RETURN_INTERFACE(pszIID, PDMIBASE, &pDrvIns->IBase);
1451	PDMIBASE_RETURN_INTERFACE(pszIID, PDMIHOSTAUDIO, &pThis->IHostAudio);
1452	return NULL;
1453	}
1454
1455
1456	/*********************************************************************************************************************************
1457	* PDMDRVREG *
1458	*********************************************************************************************************************************/
1459
1460	/**
1461	* @interface_method_impl{PDMDRVREG,pfnDestruct,
1462	* Destructs an ALSA host audio driver instance.}
1463	*/
1464	static DECLCALLBACK(void) drvHstAudAlsaDestruct(PPDMDRVINS pDrvIns)
1465	{
1466	PDMDRV_CHECK_VERSIONS_RETURN_VOID(pDrvIns);
1467	PDRVHSTAUDALSA pThis = PDMINS_2_DATA(pDrvIns, PDRVHSTAUDALSA);
1468	LogFlowFuncEnter();
1469
1470	if (RTCritSectIsInitialized(&pThis->CritSect))
1471	{
1472	RTCritSectEnter(&pThis->CritSect);
1473	pThis->pIHostAudioPort = NULL;
1474	RTCritSectLeave(&pThis->CritSect);
1475	RTCritSectDelete(&pThis->CritSect);
1476	}
1477
1478	LogFlowFuncLeave();
1479	}
1480
1481
1482	/**
1483	* @interface_method_impl{PDMDRVREG,pfnConstruct,
1484	* Construct an ALSA host audio driver instance.}
1485	*/
1486	static DECLCALLBACK(int) drvHstAudAlsaConstruct(PPDMDRVINS pDrvIns, PCFGMNODE pCfg, uint32_t fFlags)
1487	{
1488	RT_NOREF(fFlags);
1489	PDMDRV_CHECK_VERSIONS_RETURN(pDrvIns);
1490	PDRVHSTAUDALSA pThis = PDMINS_2_DATA(pDrvIns, PDRVHSTAUDALSA);
1491	PCPDMDRVHLPR3 pHlp = pDrvIns->pHlpR3;
1492	LogRel(("Audio: Initializing ALSA driver\n"));
1493
1494	/*
1495	* Init the static parts.
1496	*/
1497	pThis->pDrvIns = pDrvIns;
1498	int rc = RTCritSectInit(&pThis->CritSect);
1499	AssertRCReturn(rc, rc);
1500	/* IBase */
1501	pDrvIns->IBase.pfnQueryInterface = drvHstAudAlsaQueryInterface;
1502	/* IHostAudio */
1503	pThis->IHostAudio.pfnGetConfig = drvHstAudAlsaHA_GetConfig;
1504	pThis->IHostAudio.pfnGetDevices = drvHstAudAlsaHA_GetDevices;
1505	pThis->IHostAudio.pfnSetDevice = drvHstAudAlsaHA_SetDevice;
1506	pThis->IHostAudio.pfnGetStatus = drvHstAudAlsaHA_GetStatus;
1507	pThis->IHostAudio.pfnDoOnWorkerThread = NULL;
1508	pThis->IHostAudio.pfnStreamConfigHint = NULL;
1509	pThis->IHostAudio.pfnStreamCreate = drvHstAudAlsaHA_StreamCreate;
1510	pThis->IHostAudio.pfnStreamInitAsync = NULL;
1511	pThis->IHostAudio.pfnStreamDestroy = drvHstAudAlsaHA_StreamDestroy;
1512	pThis->IHostAudio.pfnStreamNotifyDeviceChanged = NULL;
1513	pThis->IHostAudio.pfnStreamEnable = drvHstAudAlsaHA_StreamEnable;
1514	pThis->IHostAudio.pfnStreamDisable = drvHstAudAlsaHA_StreamDisable;
1515	pThis->IHostAudio.pfnStreamPause = drvHstAudAlsaHA_StreamPause;
1516	pThis->IHostAudio.pfnStreamResume = drvHstAudAlsaHA_StreamResume;
1517	pThis->IHostAudio.pfnStreamDrain = drvHstAudAlsaHA_StreamDrain;
1518	pThis->IHostAudio.pfnStreamGetPending = drvHstAudAlsaHA_StreamGetPending;
1519	pThis->IHostAudio.pfnStreamGetState = drvHstAudAlsaHA_StreamGetState;
1520	pThis->IHostAudio.pfnStreamGetWritable = drvHstAudAlsaHA_StreamGetWritable;
1521	pThis->IHostAudio.pfnStreamPlay = drvHstAudAlsaHA_StreamPlay;
1522	pThis->IHostAudio.pfnStreamGetReadable = drvHstAudAlsaHA_StreamGetReadable;
1523	pThis->IHostAudio.pfnStreamCapture = drvHstAudAlsaHA_StreamCapture;
1524
1525	/*
1526	* Read configuration.
1527	*/
1528	PDMDRV_VALIDATE_CONFIG_RETURN(pDrvIns, "OutputDeviceID\|InputDeviceID", "");
1529
1530	rc = pHlp->pfnCFGMQueryStringDef(pCfg, "InputDeviceID", pThis->szInputDev, sizeof(pThis->szInputDev), "default");
1531	AssertRCReturn(rc, rc);
1532	rc = pHlp->pfnCFGMQueryStringDef(pCfg, "OutputDeviceID", pThis->szOutputDev, sizeof(pThis->szOutputDev), "default");
1533	AssertRCReturn(rc, rc);
1534
1535	/*
1536	* Init the alsa library.
1537	*/
1538	rc = audioLoadAlsaLib();
1539	if (RT_FAILURE(rc))
1540	{
1541	LogRel(("ALSA: Failed to load the ALSA shared library: %Rrc\n", rc));
1542	return rc;
1543	}
1544
1545	/*
1546	* Query the notification interface from the driver/device above us.
1547	*/
1548	pThis->pIHostAudioPort = PDMIBASE_QUERY_INTERFACE(pDrvIns->pUpBase, PDMIHOSTAUDIOPORT);
1549	AssertReturn(pThis->pIHostAudioPort, VERR_PDM_MISSING_INTERFACE_ABOVE);
1550
1551	#ifdef DEBUG
1552	/*
1553	* Some debug stuff we don't use for anything at all.
1554	*/
1555	snd_lib_error_set_handler(drvHstAudAlsaDbgErrorHandler);
1556	#endif
1557	return VINF_SUCCESS;
1558	}
1559
1560
1561	/**
1562	* ALSA audio driver registration record.
1563	*/
1564	const PDMDRVREG g_DrvHostALSAAudio =
1565	{
1566	/* u32Version */
1567	PDM_DRVREG_VERSION,
1568	/* szName */
1569	"ALSAAudio",
1570	/* szRCMod */
1571	"",
1572	/* szR0Mod */
1573	"",
1574	/* pszDescription */
1575	"ALSA host audio driver",
1576	/* fFlags */
1577	PDM_DRVREG_FLAGS_HOST_BITS_DEFAULT,
1578	/* fClass. */
1579	PDM_DRVREG_CLASS_AUDIO,
1580	/* cMaxInstances */
1581	~0U,
1582	/* cbInstance */
1583	sizeof(DRVHSTAUDALSA),
1584	/* pfnConstruct */
1585	drvHstAudAlsaConstruct,
1586	/* pfnDestruct */
1587	drvHstAudAlsaDestruct,
1588	/* pfnRelocate */
1589	NULL,
1590	/* pfnIOCtl */
1591	NULL,
1592	/* pfnPowerOn */
1593	NULL,
1594	/* pfnReset */
1595	NULL,
1596	/* pfnSuspend */
1597	NULL,
1598	/* pfnResume */
1599	NULL,
1600	/* pfnAttach */
1601	NULL,
1602	/* pfnDetach */
1603	NULL,
1604	/* pfnPowerOff */
1605	NULL,
1606	/* pfnSoftReset */
1607	NULL,
1608	/* u32EndVersion */
1609	PDM_DRVREG_VERSION
1610	};
1611

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source: vbox/trunk/src/VBox/Devices/Audio/DrvHostAudioAlsa.cpp@ 101359

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