DrvHostALSAAudio.cpp@ 82331

Last change on this file since 82331 was 82255, checked in by vboxsync, 5 years ago
vmm/pdmaudioifs.h: More of the same. bugref:9218
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1	/* $Id: DrvHostALSAAudio.cpp 82255 2019-11-27 23:20:26Z vboxsync $ */
2	/** @file
3	* ALSA audio driver.
4	*/
5
6	/*
7	* Copyright (C) 2006-2019 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	/*********************************************************************************************************************************
45	* Header Files *
46	*********************************************************************************************************************************/
47	#define LOG_GROUP LOG_GROUP_DRV_HOST_AUDIO
48	#include <VBox/log.h>
49	#include <iprt/alloc.h>
50	#include <iprt/uuid.h> /* For PDMIBASE_2_PDMDRV. */
51	#include <VBox/vmm/pdmaudioifs.h>
52
53	RT_C_DECLS_BEGIN
54	#include "alsa_stubs.h"
55	#include "alsa_mangling.h"
56	RT_C_DECLS_END
57
58	#include <alsa/asoundlib.h>
59	#include <alsa/control.h> /* For device enumeration. */
60
61	#include "DrvAudio.h"
62	#include "VBoxDD.h"
63
64
65	/*********************************************************************************************************************************
66	* Defines *
67	*********************************************************************************************************************************/
68
69	/** Makes DRVHOSTALSAAUDIO out of PDMIHOSTAUDIO. */
70	#define PDMIHOSTAUDIO_2_DRVHOSTALSAAUDIO(pInterface) \
71	( (PDRVHOSTALSAAUDIO)((uintptr_t)pInterface - RT_UOFFSETOF(DRVHOSTALSAAUDIO, IHostAudio)) )
72
73
74	/*********************************************************************************************************************************
75	* Structures *
76	*********************************************************************************************************************************/
77
78	typedef struct ALSAAUDIOSTREAM
79	{
80	/** The stream's acquired configuration. */
81	PPDMAUDIOSTREAMCFG pCfg;
82	union
83	{
84	struct
85	{
86	} In;
87	struct
88	{
89	} Out;
90	};
91	snd_pcm_t *phPCM;
92	void *pvBuf;
93	size_t cbBuf;
94	} ALSAAUDIOSTREAM, *PALSAAUDIOSTREAM;
95
96	/* latency = period_size * periods / (rate * bytes_per_frame) */
97
98	static int alsaStreamRecover(snd_pcm_t *phPCM);
99
100	/**
101	* Host Alsa audio driver instance data.
102	* @implements PDMIAUDIOCONNECTOR
103	*/
104	typedef struct DRVHOSTALSAAUDIO
105	{
106	/** Pointer to the driver instance structure. */
107	PPDMDRVINS pDrvIns;
108	/** Pointer to host audio interface. */
109	PDMIHOSTAUDIO IHostAudio;
110	/** Error count for not flooding the release log.
111	* UINT32_MAX for unlimited logging. */
112	uint32_t cLogErrors;
113	} DRVHOSTALSAAUDIO, *PDRVHOSTALSAAUDIO;
114
115	/** Maximum number of tries to recover a broken pipe. */
116	#define ALSA_RECOVERY_TRIES_MAX 5
117
118	typedef struct ALSAAUDIOSTREAMCFG
119	{
120	unsigned int freq;
121	/** PCM sound format. */
122	snd_pcm_format_t fmt;
123	/** PCM data access type. */
124	snd_pcm_access_t access;
125	/** Whether resampling should be performed by alsalib or not. */
126	int resample;
127	int nchannels;
128	/** Buffer size (in audio frames). */
129	unsigned long buffer_size;
130	/** Periods (in audio frames). */
131	unsigned long period_size;
132	/** For playback: Starting to play threshold (in audio frames).
133	* For Capturing: Starting to capture threshold (in audio frames). */
134	unsigned long threshold;
135	} ALSAAUDIOSTREAMCFG, *PALSAAUDIOSTREAMCFG;
136
137
138
139	static snd_pcm_format_t alsaAudioPropsToALSA(PPDMAUDIOPCMPROPS pProps)
140	{
141	switch (pProps->cbSample)
142	{
143	case 1:
144	return pProps->fSigned ? SND_PCM_FORMAT_S8 : SND_PCM_FORMAT_U8;
145
146	case 2:
147	return pProps->fSigned ? SND_PCM_FORMAT_S16_LE : SND_PCM_FORMAT_U16_LE;
148
149	case 4:
150	return pProps->fSigned ? SND_PCM_FORMAT_S32_LE : SND_PCM_FORMAT_U32_LE;
151
152	default:
153	break;
154	}
155
156	AssertMsgFailed(("%RU8 bytes not supported\n", pProps->cbSample));
157	return SND_PCM_FORMAT_U8;
158	}
159
160
161	static int alsaALSAToAudioProps(snd_pcm_format_t fmt, PPDMAUDIOPCMPROPS pProps)
162	{
163	/** @todo r=bird: Why isn't this code setting fSwapEndian for every case? It
164	* seems to make some UNDOCUMENT ASSUMPTIONS about pProps. */
165	switch (fmt)
166	{
167	case SND_PCM_FORMAT_S8:
168	pProps->cbSample = 1;
169	pProps->fSigned = true;
170	break;
171
172	case SND_PCM_FORMAT_U8:
173	pProps->cbSample = 1;
174	pProps->fSigned = false;
175	break;
176
177	case SND_PCM_FORMAT_S16_LE:
178	pProps->cbSample = 2;
179	pProps->fSigned = true;
180	break;
181
182	case SND_PCM_FORMAT_U16_LE:
183	pProps->cbSample = 2;
184	pProps->fSigned = false;
185	break;
186
187	case SND_PCM_FORMAT_S16_BE:
188	pProps->cbSample = 2;
189	pProps->fSigned = true;
190	#ifdef RT_LITTLE_ENDIAN
191	pProps->fSwapEndian = true;
192	#endif
193	break;
194
195	case SND_PCM_FORMAT_U16_BE:
196	pProps->cbSample = 2;
197	pProps->fSigned = false;
198	#ifdef RT_LITTLE_ENDIAN
199	pProps->fSwapEndian = true;
200	#endif
201	break;
202
203	case SND_PCM_FORMAT_S32_LE:
204	pProps->cbSample = 4;
205	pProps->fSigned = true;
206	break;
207
208	case SND_PCM_FORMAT_U32_LE:
209	pProps->cbSample = 4;
210	pProps->fSigned = false;
211	break;
212
213	case SND_PCM_FORMAT_S32_BE:
214	pProps->cbSample = 4;
215	pProps->fSigned = true;
216	#ifdef RT_LITTLE_ENDIAN
217	pProps->fSwapEndian = true;
218	#endif
219	break;
220
221	case SND_PCM_FORMAT_U32_BE:
222	pProps->cbSample = 4;
223	pProps->fSigned = false;
224	#ifdef RT_LITTLE_ENDIAN
225	pProps->fSwapEndian = true;
226	#endif
227	break;
228
229	default:
230	AssertMsgFailedReturn(("Format %d not supported\n", fmt), VERR_NOT_SUPPORTED);
231	}
232
233	Assert(pProps->cbSample > 0);
234	Assert(pProps->cChannels > 0);
235	pProps->cShift = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(pProps->cbSample, pProps->cChannels);
236
237	return VINF_SUCCESS;
238	}
239
240
241	static int alsaStreamSetSWParams(snd_pcm_t *phPCM, bool fIn, PALSAAUDIOSTREAMCFG pCfgReq, PALSAAUDIOSTREAMCFG pCfgObt)
242	{
243	if (fIn) /* For input streams there's nothing to do in here right now. */
244	return VINF_SUCCESS;
245
246	snd_pcm_sw_params_t *pSWParms = NULL;
247	snd_pcm_sw_params_alloca(&pSWParms);
248	if (!pSWParms)
249	return VERR_NO_MEMORY;
250
251	int rc;
252	do
253	{
254	int err = snd_pcm_sw_params_current(phPCM, pSWParms);
255	if (err < 0)
256	{
257	LogRel(("ALSA: Failed to get current software parameters: %s\n", snd_strerror(err)));
258	rc = VERR_ACCESS_DENIED;
259	break;
260	}
261
262	err = snd_pcm_sw_params_set_start_threshold(phPCM, pSWParms, pCfgReq->threshold);
263	if (err < 0)
264	{
265	LogRel(("ALSA: Failed to set software threshold to %ld: %s\n", pCfgReq->threshold, snd_strerror(err)));
266	rc = VERR_ACCESS_DENIED;
267	break;
268	}
269
270	err = snd_pcm_sw_params_set_avail_min(phPCM, pSWParms, pCfgReq->period_size);
271	if (err < 0)
272	{
273	LogRel(("ALSA: Failed to set available minimum to %ld: %s\n", pCfgReq->threshold, snd_strerror(err)));
274	rc = VERR_ACCESS_DENIED;
275	break;
276	}
277
278	err = snd_pcm_sw_params(phPCM, pSWParms);
279	if (err < 0)
280	{
281	LogRel(("ALSA: Failed to set new software parameters: %s\n", snd_strerror(err)));
282	rc = VERR_ACCESS_DENIED;
283	break;
284	}
285
286	err = snd_pcm_sw_params_get_start_threshold(pSWParms, &pCfgObt->threshold);
287	if (err < 0)
288	{
289	LogRel(("ALSA: Failed to get start threshold\n"));
290	rc = VERR_ACCESS_DENIED;
291	break;
292	}
293
294	LogFunc(("Setting threshold to %RU32 frames\n", pCfgObt->threshold));
295	rc = VINF_SUCCESS;
296	}
297	while (0);
298
299	return rc;
300	}
301
302
303	static int alsaStreamClose(snd_pcm_t **pphPCM)
304	{
305	if (!pphPCM \|\| !*pphPCM)
306	return VINF_SUCCESS;
307
308	int rc;
309	int rc2 = snd_pcm_close(*pphPCM);
310	if (rc2)
311	{
312	LogRel(("ALSA: Closing PCM descriptor failed: %s\n", snd_strerror(rc2)));
313	rc = VERR_GENERAL_FAILURE; /** @todo */
314	}
315	else
316	{
317	*pphPCM = NULL;
318	rc = VINF_SUCCESS;
319	}
320
321	LogFlowFuncLeaveRC(rc);
322	return rc;
323	}
324
325
326	static int alsaStreamOpen(bool fIn, PALSAAUDIOSTREAMCFG pCfgReq, PALSAAUDIOSTREAMCFG pCfgObt, snd_pcm_t **pphPCM)
327	{
328	snd_pcm_t *phPCM = NULL;
329
330	int rc = VERR_AUDIO_STREAM_COULD_NOT_CREATE;
331
332	unsigned int cChannels = pCfgReq->nchannels;
333	unsigned int uFreq = pCfgReq->freq;
334	snd_pcm_uframes_t obt_buffer_size;
335
336	do
337	{
338	const char pszDev = "default"; /* @todo Make this configurable through PALSAAUDIOSTREAMCFG. */
339	if (!pszDev)
340	{
341	LogRel(("ALSA: Invalid or no %s device name set\n", fIn ? "input" : "output"));
342	rc = VERR_INVALID_PARAMETER;
343	break;
344	}
345
346	LogRel(("ALSA: Using %s device \"%s\"\n", fIn ? "input" : "output", pszDev));
347
348	int err = snd_pcm_open(&phPCM, pszDev,
349	fIn ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
350	SND_PCM_NONBLOCK);
351	if (err < 0)
352	{
353	LogRel(("ALSA: Failed to open \"%s\" as %s device: %s\n", pszDev, fIn ? "input" : "output", snd_strerror(err)));
354	break;
355	}
356
357	err = snd_pcm_nonblock(phPCM, 1);
358	if (err < 0)
359	{
360	LogRel(("ALSA: Error setting output non-blocking mode: %s\n", snd_strerror(err)));
361	break;
362	}
363
364	snd_pcm_hw_params_t *pHWParms;
365	snd_pcm_hw_params_alloca(&pHWParms); /** @todo Check for successful allocation? */
366	err = snd_pcm_hw_params_any(phPCM, pHWParms);
367	if (err < 0)
368	{
369	LogRel(("ALSA: Failed to initialize hardware parameters: %s\n", snd_strerror(err)));
370	break;
371	}
372
373	err = snd_pcm_hw_params_set_access(phPCM, pHWParms, SND_PCM_ACCESS_RW_INTERLEAVED);
374	if (err < 0)
375	{
376	LogRel(("ALSA: Failed to set access type: %s\n", snd_strerror(err)));
377	break;
378	}
379
380	err = snd_pcm_hw_params_set_format(phPCM, pHWParms, pCfgReq->fmt);
381	if (err < 0)
382	{
383	LogRel(("ALSA: Failed to set audio format to %d: %s\n", pCfgReq->fmt, snd_strerror(err)));
384	break;
385	}
386
387	err = snd_pcm_hw_params_set_rate_near(phPCM, pHWParms, &uFreq, 0);
388	if (err < 0)
389	{
390	LogRel(("ALSA: Failed to set frequency to %uHz: %s\n", pCfgReq->freq, snd_strerror(err)));
391	break;
392	}
393
394	err = snd_pcm_hw_params_set_channels_near(phPCM, pHWParms, &cChannels);
395	if (err < 0)
396	{
397	LogRel(("ALSA: Failed to set number of channels to %d\n", pCfgReq->nchannels));
398	break;
399	}
400
401	if ( cChannels != 1
402	&& cChannels != 2)
403	{
404	LogRel(("ALSA: Number of audio channels (%u) not supported\n", cChannels));
405	break;
406	}
407
408	snd_pcm_uframes_t period_size_f = pCfgReq->period_size;
409	snd_pcm_uframes_t buffer_size_f = pCfgReq->buffer_size;
410
411	snd_pcm_uframes_t minval = period_size_f;
412
413	int dir = 0;
414	err = snd_pcm_hw_params_get_period_size_min(pHWParms, &minval, &dir);
415	if (err < 0)
416	{
417	LogRel(("ALSA: Could not determine minimal period size\n"));
418	break;
419	}
420	else
421	{
422	LogFunc(("Minimal period size is: %ld\n", minval));
423	if (period_size_f < minval)
424	period_size_f = minval;
425	}
426
427	err = snd_pcm_hw_params_set_period_size_near(phPCM, pHWParms, &period_size_f, 0);
428	LogFunc(("Period size is: %RU32\n", period_size_f));
429	if (err < 0)
430	{
431	LogRel(("ALSA: Failed to set period size %d (%s)\n", period_size_f, snd_strerror(err)));
432	break;
433	}
434
435	minval = buffer_size_f;
436	err = snd_pcm_hw_params_get_buffer_size_min(pHWParms, &minval);
437	if (err < 0)
438	{
439	LogRel(("ALSA: Could not retrieve minimal buffer size\n"));
440	break;
441	}
442	else
443	LogFunc(("Minimal buffer size is: %RU32\n", minval));
444
445	err = snd_pcm_hw_params_set_buffer_size_near(phPCM, pHWParms, &buffer_size_f);
446	if (err < 0)
447	{
448	LogRel(("ALSA: Failed to set near buffer size %RU32: %s\n", buffer_size_f, snd_strerror(err)));
449	break;
450	}
451
452	err = snd_pcm_hw_params(phPCM, pHWParms);
453	if (err < 0)
454	{
455	LogRel(("ALSA: Failed to apply audio parameters\n"));
456	break;
457	}
458
459	err = snd_pcm_hw_params_get_buffer_size(pHWParms, &obt_buffer_size);
460	if (err < 0)
461	{
462	LogRel(("ALSA: Failed to get buffer size\n"));
463	break;
464	}
465
466	snd_pcm_uframes_t obt_period_size;
467	err = snd_pcm_hw_params_get_period_size(pHWParms, &obt_period_size, &dir);
468	if (err < 0)
469	{
470	LogRel(("ALSA: Failed to get period size\n"));
471	break;
472	}
473
474	LogRel2(("ALSA: Frequency is %dHz, period size is %RU32 frames, buffer size is %RU32 frames\n",
475	pCfgReq->freq, obt_period_size, obt_buffer_size));
476
477	err = snd_pcm_prepare(phPCM);
478	if (err < 0)
479	{
480	LogRel(("ALSA: Could not prepare hPCM %p\n", (void *)phPCM));
481	rc = VERR_AUDIO_BACKEND_INIT_FAILED;
482	break;
483	}
484
485	rc = alsaStreamSetSWParams(phPCM, fIn, pCfgReq, pCfgObt);
486	if (RT_FAILURE(rc))
487	break;
488
489	pCfgObt->fmt = pCfgReq->fmt;
490	pCfgObt->nchannels = cChannels;
491	pCfgObt->freq = uFreq;
492	pCfgObt->period_size = obt_period_size;
493	pCfgObt->buffer_size = obt_buffer_size;
494
495	rc = VINF_SUCCESS;
496	}
497	while (0);
498
499	if (RT_SUCCESS(rc))
500	{
501	*pphPCM = phPCM;
502	}
503	else
504	alsaStreamClose(&phPCM);
505
506	LogFlowFuncLeaveRC(rc);
507	return rc;
508	}
509
510
511	#ifdef DEBUG
512	static void alsaDbgErrorHandler(const char file, int line, const char function,
513	int err, const char *fmt, ...)
514	{
515	/** @todo Implement me! */
516	RT_NOREF(file, line, function, err, fmt);
517	}
518	#endif
519
520
521	static int alsaStreamGetAvail(snd_pcm_t phPCM, snd_pcm_sframes_t pFramesAvail)
522	{
523	AssertPtrReturn(phPCM, VERR_INVALID_POINTER);
524	/* pFramesAvail is optional. */
525
526	int rc;
527
528	snd_pcm_sframes_t framesAvail = snd_pcm_avail_update(phPCM);
529	if (framesAvail < 0)
530	{
531	if (framesAvail == -EPIPE)
532	{
533	rc = alsaStreamRecover(phPCM);
534	if (RT_SUCCESS(rc))
535	framesAvail = snd_pcm_avail_update(phPCM);
536	}
537	else
538	rc = VERR_ACCESS_DENIED; /** @todo Find a better rc. */
539	}
540	else
541	rc = VINF_SUCCESS;
542
543	if (RT_SUCCESS(rc))
544	{
545	if (pFramesAvail)
546	*pFramesAvail = framesAvail;
547	}
548
549	LogFunc(("cFrames=%ld, rc=%Rrc\n", framesAvail, rc));
550	return rc;
551	}
552
553
554	static int alsaStreamRecover(snd_pcm_t *phPCM)
555	{
556	AssertPtrReturn(phPCM, VERR_INVALID_POINTER);
557
558	int err = snd_pcm_prepare(phPCM);
559	if (err < 0)
560	{
561	LogFunc(("Failed to recover stream %p: %s\n", phPCM, snd_strerror(err)));
562	return VERR_ACCESS_DENIED; /** @todo Find a better rc. */
563	}
564
565	return VINF_SUCCESS;
566	}
567
568
569	static int alsaStreamResume(snd_pcm_t *phPCM)
570	{
571	AssertPtrReturn(phPCM, VERR_INVALID_POINTER);
572
573	int err = snd_pcm_resume(phPCM);
574	if (err < 0)
575	{
576	LogFunc(("Failed to resume stream %p: %s\n", phPCM, snd_strerror(err)));
577	return VERR_ACCESS_DENIED; /** @todo Find a better rc. */
578	}
579
580	return VINF_SUCCESS;
581	}
582
583
584	/**
585	* @interface_method_impl{PDMIHOSTAUDIO,pfnInit}
586	*/
587	static DECLCALLBACK(int) drvHostAlsaAudioHA_Init(PPDMIHOSTAUDIO pInterface)
588	{
589	RT_NOREF(pInterface);
590
591	LogFlowFuncEnter();
592
593	int rc = audioLoadAlsaLib();
594	if (RT_FAILURE(rc))
595	LogRel(("ALSA: Failed to load the ALSA shared library, rc=%Rrc\n", rc));
596	else
597	{
598	#ifdef DEBUG
599	snd_lib_error_set_handler(alsaDbgErrorHandler);
600	#endif
601	}
602
603	return rc;
604	}
605
606
607	/**
608	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamCapture}
609	*/
610	static DECLCALLBACK(int) drvHostAlsaAudioHA_StreamCapture(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream,
611	void pvBuf, uint32_t uBufSize, uint32_t puRead)
612	{
613	AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
614	AssertPtrReturn(pStream, VERR_INVALID_POINTER);
615	AssertPtrReturn(pvBuf, VERR_INVALID_POINTER);
616	AssertReturn(uBufSize, VERR_INVALID_PARAMETER);
617	/* pcbRead is optional. */
618
619	PALSAAUDIOSTREAM pStreamALSA = (PALSAAUDIOSTREAM)pStream;
620
621	snd_pcm_sframes_t cAvail;
622	int rc = alsaStreamGetAvail(pStreamALSA->phPCM, &cAvail);
623	if (RT_FAILURE(rc))
624	{
625	LogFunc(("Error getting number of captured frames, rc=%Rrc\n", rc));
626	return rc;
627	}
628
629	PPDMAUDIOSTREAMCFG pCfg = pStreamALSA->pCfg;
630	AssertPtr(pCfg);
631
632	if (!cAvail) /* No data yet? */
633	{
634	snd_pcm_state_t state = snd_pcm_state(pStreamALSA->phPCM);
635	switch (state)
636	{
637	case SND_PCM_STATE_PREPARED:
638	cAvail = PDMAUDIOSTREAMCFG_B2F(pCfg, uBufSize);
639	break;
640
641	case SND_PCM_STATE_SUSPENDED:
642	{
643	rc = alsaStreamResume(pStreamALSA->phPCM);
644	if (RT_FAILURE(rc))
645	break;
646
647	LogFlow(("Resuming suspended input stream\n"));
648	break;
649	}
650
651	default:
652	LogFlow(("No frames available, state=%d\n", state));
653	break;
654	}
655
656	if (!cAvail)
657	{
658	if (puRead)
659	*puRead = 0;
660	return VINF_SUCCESS;
661	}
662	}
663
664	/*
665	* Check how much we can read from the capture device without overflowing
666	* the mixer buffer.
667	*/
668	size_t cbToRead = RT_MIN((size_t)PDMAUDIOSTREAMCFG_F2B(pCfg, cAvail), uBufSize);
669
670	LogFlowFunc(("cbToRead=%zu, cAvail=%RI32\n", cbToRead, cAvail));
671
672	uint32_t cbReadTotal = 0;
673
674	snd_pcm_uframes_t cToRead;
675	snd_pcm_sframes_t cRead;
676
677	while ( cbToRead
678	&& RT_SUCCESS(rc))
679	{
680	cToRead = RT_MIN(PDMAUDIOSTREAMCFG_B2F(pCfg, cbToRead),
681	PDMAUDIOSTREAMCFG_B2F(pCfg, pStreamALSA->cbBuf));
682	AssertBreakStmt(cToRead, rc = VERR_NO_DATA);
683	cRead = snd_pcm_readi(pStreamALSA->phPCM, pStreamALSA->pvBuf, cToRead);
684	if (cRead <= 0)
685	{
686	switch (cRead)
687	{
688	case 0:
689	{
690	LogFunc(("No input frames available\n"));
691	rc = VERR_ACCESS_DENIED;
692	break;
693	}
694
695	case -EAGAIN:
696	{
697	/*
698	* Don't set error here because EAGAIN means there are no further frames
699	* available at the moment, try later. As we might have read some frames
700	* already these need to be processed instead.
701	*/
702	cbToRead = 0;
703	break;
704	}
705
706	case -EPIPE:
707	{
708	rc = alsaStreamRecover(pStreamALSA->phPCM);
709	if (RT_FAILURE(rc))
710	break;
711
712	LogFlowFunc(("Recovered from capturing\n"));
713	continue;
714	}
715
716	default:
717	{
718	LogFunc(("Failed to read input frames: %s\n", snd_strerror(cRead)));
719	rc = VERR_GENERAL_FAILURE; /** @todo Fudge! */
720	break;
721	}
722	}
723	}
724	else
725	{
726	/*
727	* We should not run into a full mixer buffer or we loose samples and
728	* run into an endless loop if ALSA keeps producing samples ("null"
729	* capture device for example).
730	*/
731	uint32_t cbRead = PDMAUDIOSTREAMCFG_F2B(pCfg, cRead);
732
733	memcpy(pvBuf, pStreamALSA->pvBuf, cbRead);
734
735	Assert(cbToRead >= cbRead);
736	cbToRead -= cbRead;
737	cbReadTotal += cbRead;
738	}
739	}
740
741	if (RT_SUCCESS(rc))
742	{
743	if (puRead)
744	*puRead = cbReadTotal;
745	}
746
747	return rc;
748	}
749
750	/**
751	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamPlay}
752	*/
753	static DECLCALLBACK(int) drvHostAlsaAudioHA_StreamPlay(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream,
754	const void pvBuf, uint32_t uBufSize, uint32_t puWritten)
755	{
756	AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
757	AssertPtrReturn(pStream, VERR_INVALID_POINTER);
758	AssertPtrReturn(pvBuf, VERR_INVALID_POINTER);
759	AssertReturn(uBufSize, VERR_INVALID_PARAMETER);
760	/* puWritten is optional. */
761
762	PALSAAUDIOSTREAM pStreamALSA = (PALSAAUDIOSTREAM)pStream;
763
764	PPDMAUDIOSTREAMCFG pCfg = pStreamALSA->pCfg;
765	AssertPtr(pCfg);
766
767	int rc;
768
769	uint32_t cbWrittenTotal = 0;
770
771	do
772	{
773	snd_pcm_sframes_t csAvail;
774	rc = alsaStreamGetAvail(pStreamALSA->phPCM, &csAvail);
775	if (RT_FAILURE(rc))
776	{
777	LogFunc(("Error getting number of playback frames, rc=%Rrc\n", rc));
778	break;
779	}
780
781	if (!csAvail)
782	break;
783
784	size_t cbToWrite = RT_MIN((unsigned)PDMAUDIOSTREAMCFG_F2B(pCfg, csAvail), pStreamALSA->cbBuf);
785	if (!cbToWrite)
786	break;
787
788	/* Do not write more than available. */
789	if (cbToWrite > uBufSize)
790	cbToWrite = uBufSize;
791
792	memcpy(pStreamALSA->pvBuf, pvBuf, cbToWrite);
793
794	snd_pcm_sframes_t csWritten = 0;
795
796	/* Don't try infinitely on recoverable errors. */
797	unsigned iTry;
798	for (iTry = 0; iTry < ALSA_RECOVERY_TRIES_MAX; iTry++)
799	{
800	csWritten = snd_pcm_writei(pStreamALSA->phPCM, pStreamALSA->pvBuf,
801	PDMAUDIOSTREAMCFG_B2F(pCfg, cbToWrite));
802	if (csWritten <= 0)
803	{
804	switch (csWritten)
805	{
806	case 0:
807	{
808	LogFunc(("Failed to write %zu bytes\n", cbToWrite));
809	rc = VERR_ACCESS_DENIED;
810	break;
811	}
812
813	case -EPIPE:
814	{
815	rc = alsaStreamRecover(pStreamALSA->phPCM);
816	if (RT_FAILURE(rc))
817	break;
818
819	LogFlowFunc(("Recovered from playback\n"));
820	continue;
821	}
822
823	case -ESTRPIPE:
824	{
825	/* Stream was suspended and waiting for a recovery. */
826	rc = alsaStreamResume(pStreamALSA->phPCM);
827	if (RT_FAILURE(rc))
828	{
829	LogRel(("ALSA: Failed to resume output stream\n"));
830	break;
831	}
832
833	LogFlowFunc(("Resumed suspended output stream\n"));
834	continue;
835	}
836
837	default:
838	LogFlowFunc(("Failed to write %RU32 bytes, error unknown\n", cbToWrite));
839	rc = VERR_GENERAL_FAILURE; /** @todo */
840	break;
841	}
842	}
843	else
844	break;
845	} /* For number of tries. */
846
847	if ( iTry == ALSA_RECOVERY_TRIES_MAX
848	&& csWritten <= 0)
849	rc = VERR_BROKEN_PIPE;
850
851	if (RT_FAILURE(rc))
852	break;
853
854	cbWrittenTotal = PDMAUDIOSTREAMCFG_F2B(pCfg, csWritten);
855
856	} while (0);
857
858	if (RT_SUCCESS(rc))
859	{
860	if (puWritten)
861	*puWritten = cbWrittenTotal;
862	}
863
864	return rc;
865	}
866
867
868	static int alsaDestroyStreamIn(PALSAAUDIOSTREAM pStreamALSA)
869	{
870	alsaStreamClose(&pStreamALSA->phPCM);
871
872	if (pStreamALSA->pvBuf)
873	{
874	RTMemFree(pStreamALSA->pvBuf);
875	pStreamALSA->pvBuf = NULL;
876	}
877
878	return VINF_SUCCESS;
879	}
880
881
882	static int alsaDestroyStreamOut(PALSAAUDIOSTREAM pStreamALSA)
883	{
884	alsaStreamClose(&pStreamALSA->phPCM);
885
886	if (pStreamALSA->pvBuf)
887	{
888	RTMemFree(pStreamALSA->pvBuf);
889	pStreamALSA->pvBuf = NULL;
890	}
891
892	return VINF_SUCCESS;
893	}
894
895
896	static int alsaCreateStreamOut(PALSAAUDIOSTREAM pStreamALSA, PPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq)
897	{
898	snd_pcm_t *phPCM = NULL;
899
900	int rc;
901
902	do
903	{
904	ALSAAUDIOSTREAMCFG req;
905	req.fmt = alsaAudioPropsToALSA(&pCfgReq->Props);
906	req.freq = pCfgReq->Props.uHz;
907	req.nchannels = pCfgReq->Props.cChannels;
908	req.period_size = pCfgReq->Backend.cFramesPeriod;
909	req.buffer_size = pCfgReq->Backend.cFramesBufferSize;
910	req.threshold = pCfgReq->Backend.cFramesPreBuffering;
911
912	ALSAAUDIOSTREAMCFG obt;
913	rc = alsaStreamOpen(false /* fIn */, &req, &obt, &phPCM);
914	if (RT_FAILURE(rc))
915	break;
916
917	pCfgAcq->Props.uHz = obt.freq;
918	pCfgAcq->Props.cChannels = obt.nchannels;
919
920	rc = alsaALSAToAudioProps(obt.fmt, &pCfgAcq->Props);
921	if (RT_FAILURE(rc))
922	break;
923
924	pCfgAcq->Backend.cFramesPeriod = obt.period_size;
925	pCfgAcq->Backend.cFramesBufferSize = obt.buffer_size;
926	pCfgAcq->Backend.cFramesPreBuffering = obt.threshold;
927
928	pStreamALSA->cbBuf = pCfgAcq->Backend.cFramesBufferSize * DrvAudioHlpPCMPropsBytesPerFrame(&pCfgAcq->Props);
929	pStreamALSA->pvBuf = RTMemAllocZ(pStreamALSA->cbBuf);
930	if (!pStreamALSA->pvBuf)
931	{
932	LogRel(("ALSA: Not enough memory for output DAC buffer (%zu frames)\n", pCfgAcq->Backend.cFramesBufferSize));
933	rc = VERR_NO_MEMORY;
934	break;
935	}
936
937	pStreamALSA->phPCM = phPCM;
938	}
939	while (0);
940
941	if (RT_FAILURE(rc))
942	alsaStreamClose(&phPCM);
943
944	LogFlowFuncLeaveRC(rc);
945	return rc;
946	}
947
948
949	static int alsaCreateStreamIn(PALSAAUDIOSTREAM pStreamALSA, PPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq)
950	{
951	int rc;
952
953	snd_pcm_t *phPCM = NULL;
954
955	do
956	{
957	ALSAAUDIOSTREAMCFG req;
958	req.fmt = alsaAudioPropsToALSA(&pCfgReq->Props);
959	req.freq = pCfgReq->Props.uHz;
960	req.nchannels = pCfgReq->Props.cChannels;
961	req.period_size = DrvAudioHlpMilliToFrames(50 /* ms /, &pCfgReq->Props); /* @todo Make this configurable. */
962	req.buffer_size = req.period_size * 2; /** @todo Make this configurable. */
963	req.threshold = req.period_size;
964
965	ALSAAUDIOSTREAMCFG obt;
966	rc = alsaStreamOpen(true /* fIn */, &req, &obt, &phPCM);
967	if (RT_FAILURE(rc))
968	break;
969
970	pCfgAcq->Props.uHz = obt.freq;
971	pCfgAcq->Props.cChannels = obt.nchannels;
972
973	rc = alsaALSAToAudioProps(obt.fmt, &pCfgAcq->Props);
974	if (RT_FAILURE(rc))
975	break;
976
977	pCfgAcq->Backend.cFramesPeriod = obt.period_size;
978	pCfgAcq->Backend.cFramesBufferSize = obt.buffer_size;
979	/* No pre-buffering. */
980
981	pStreamALSA->cbBuf = pCfgAcq->Backend.cFramesBufferSize * DrvAudioHlpPCMPropsBytesPerFrame(&pCfgAcq->Props);
982	pStreamALSA->pvBuf = RTMemAlloc(pStreamALSA->cbBuf);
983	if (!pStreamALSA->pvBuf)
984	{
985	LogRel(("ALSA: Not enough memory for input ADC buffer (%zu frames)\n", pCfgAcq->Backend.cFramesBufferSize));
986	rc = VERR_NO_MEMORY;
987	break;
988	}
989
990	pStreamALSA->phPCM = phPCM;
991	}
992	while (0);
993
994	if (RT_FAILURE(rc))
995	alsaStreamClose(&phPCM);
996
997	LogFlowFuncLeaveRC(rc);
998	return rc;
999	}
1000
1001
1002	static int alsaControlStreamIn(PALSAAUDIOSTREAM pStreamALSA, PDMAUDIOSTREAMCMD enmStreamCmd)
1003	{
1004	int rc = VINF_SUCCESS;
1005
1006	int err;
1007
1008	switch (enmStreamCmd)
1009	{
1010	case PDMAUDIOSTREAMCMD_ENABLE:
1011	case PDMAUDIOSTREAMCMD_RESUME:
1012	{
1013	err = snd_pcm_prepare(pStreamALSA->phPCM);
1014	if (err < 0)
1015	{
1016	LogRel(("ALSA: Error preparing input stream: %s\n", snd_strerror(err)));
1017	rc = VERR_ACCESS_DENIED; /** @todo Find a better rc. */
1018	}
1019	else
1020	{
1021	Assert(snd_pcm_state(pStreamALSA->phPCM) == SND_PCM_STATE_PREPARED);
1022
1023	/* Only start the PCM stream for input streams. */
1024	err = snd_pcm_start(pStreamALSA->phPCM);
1025	if (err < 0)
1026	{
1027	LogRel(("ALSA: Error starting input stream: %s\n", snd_strerror(err)));
1028	rc = VERR_ACCESS_DENIED; /** @todo Find a better rc. */
1029	}
1030	}
1031
1032	break;
1033	}
1034
1035	case PDMAUDIOSTREAMCMD_DISABLE:
1036	{
1037	err = snd_pcm_drop(pStreamALSA->phPCM);
1038	if (err < 0)
1039	{
1040	LogRel(("ALSA: Error disabling input stream: %s\n", snd_strerror(err)));
1041	rc = VERR_ACCESS_DENIED; /** @todo Find a better rc. */
1042	}
1043	break;
1044	}
1045
1046	case PDMAUDIOSTREAMCMD_PAUSE:
1047	{
1048	err = snd_pcm_drop(pStreamALSA->phPCM);
1049	if (err < 0)
1050	{
1051	LogRel(("ALSA: Error pausing input stream: %s\n", snd_strerror(err)));
1052	rc = VERR_ACCESS_DENIED; /** @todo Find a better rc. */
1053	}
1054	break;
1055	}
1056
1057	default:
1058	rc = VERR_NOT_SUPPORTED;
1059	break;
1060	}
1061
1062	LogFlowFuncLeaveRC(rc);
1063	return rc;
1064	}
1065
1066
1067	static int alsaControlStreamOut(PALSAAUDIOSTREAM pStreamALSA, PDMAUDIOSTREAMCMD enmStreamCmd)
1068	{
1069	int rc = VINF_SUCCESS;
1070
1071	int err;
1072
1073	switch (enmStreamCmd)
1074	{
1075	case PDMAUDIOSTREAMCMD_ENABLE:
1076	case PDMAUDIOSTREAMCMD_RESUME:
1077	{
1078	err = snd_pcm_prepare(pStreamALSA->phPCM);
1079	if (err < 0)
1080	{
1081	LogRel(("ALSA: Error preparing output stream: %s\n", snd_strerror(err)));
1082	rc = VERR_ACCESS_DENIED; /** @todo Find a better rc. */
1083	}
1084	else
1085	{
1086	Assert(snd_pcm_state(pStreamALSA->phPCM) == SND_PCM_STATE_PREPARED);
1087	}
1088
1089	break;
1090	}
1091
1092	case PDMAUDIOSTREAMCMD_DISABLE:
1093	{
1094	err = snd_pcm_drop(pStreamALSA->phPCM);
1095	if (err < 0)
1096	{
1097	LogRel(("ALSA: Error disabling output stream: %s\n", snd_strerror(err)));
1098	rc = VERR_ACCESS_DENIED; /** @todo Find a better rc. */
1099	}
1100	break;
1101	}
1102
1103	case PDMAUDIOSTREAMCMD_PAUSE:
1104	{
1105	err = snd_pcm_drop(pStreamALSA->phPCM);
1106	if (err < 0)
1107	{
1108	LogRel(("ALSA: Error pausing output stream: %s\n", snd_strerror(err)));
1109	rc = VERR_ACCESS_DENIED; /** @todo Find a better rc. */
1110	}
1111	break;
1112	}
1113
1114	case PDMAUDIOSTREAMCMD_DRAIN:
1115	{
1116	snd_pcm_state_t streamState = snd_pcm_state(pStreamALSA->phPCM);
1117	Log2Func(("Stream state is: %d\n", streamState));
1118
1119	if ( streamState == SND_PCM_STATE_PREPARED
1120	\|\| streamState == SND_PCM_STATE_RUNNING)
1121	{
1122	err = snd_pcm_nonblock(pStreamALSA->phPCM, 0);
1123	if (err < 0)
1124	{
1125	LogRel(("ALSA: Error disabling output non-blocking mode: %s\n", snd_strerror(err)));
1126	rc = VERR_ACCESS_DENIED; /** @todo Find a better rc. */
1127	break;
1128	}
1129
1130	err = snd_pcm_drain(pStreamALSA->phPCM);
1131	if (err < 0)
1132	{
1133	LogRel(("ALSA: Error draining output: %s\n", snd_strerror(err)));
1134	rc = VERR_ACCESS_DENIED; /** @todo Find a better rc. */
1135	break;
1136	}
1137
1138	err = snd_pcm_nonblock(pStreamALSA->phPCM, 1);
1139	if (err < 0)
1140	{
1141	LogRel(("ALSA: Error re-enabling output non-blocking mode: %s\n", snd_strerror(err)));
1142	rc = VERR_ACCESS_DENIED; /** @todo Find a better rc. */
1143	}
1144	}
1145	break;
1146	}
1147
1148	default:
1149	rc = VERR_NOT_SUPPORTED;
1150	break;
1151	}
1152
1153	LogFlowFuncLeaveRC(rc);
1154	return rc;
1155	}
1156
1157
1158	/**
1159	* @interface_method_impl{PDMIHOSTAUDIO,pfnGetConfig}
1160	*/
1161	static DECLCALLBACK(int) drvHostAlsaAudioHA_GetConfig(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDCFG pBackendCfg)
1162	{
1163	RT_NOREF(pInterface);
1164	AssertPtrReturn(pBackendCfg, VERR_INVALID_POINTER);
1165
1166	RTStrPrintf2(pBackendCfg->szName, sizeof(pBackendCfg->szName), "ALSA audio driver");
1167
1168	pBackendCfg->cbStreamIn = sizeof(ALSAAUDIOSTREAM);
1169	pBackendCfg->cbStreamOut = sizeof(ALSAAUDIOSTREAM);
1170
1171	/* Enumerate sound devices. */
1172	char **pszHints;
1173	int err = snd_device_name_hint(-1 /* All cards /, "pcm", (void**)&pszHints);
1174	if (err == 0)
1175	{
1176	char** pszHintCur = pszHints;
1177	while (*pszHintCur != NULL)
1178	{
1179	char pszDev = snd_device_name_get_hint(pszHintCur, "NAME");
1180	bool fSkip = !pszDev
1181	\|\| !RTStrICmp("null", pszDev);
1182	if (fSkip)
1183	{
1184	if (pszDev)
1185	free(pszDev);
1186	pszHintCur++;
1187	continue;
1188	}
1189
1190	char pszIOID = snd_device_name_get_hint(pszHintCur, "IOID");
1191	if (pszIOID)
1192	{
1193	#if 0
1194	if (!RTStrICmp("input", pszIOID))
1195
1196	else if (!RTStrICmp("output", pszIOID))
1197	#endif
1198	}
1199	else /* NULL means bidirectional, input + output. */
1200	{
1201	}
1202
1203	LogRel2(("ALSA: Found %s device: %s\n", pszIOID ? RTStrToLower(pszIOID) : "bidirectional", pszDev));
1204
1205	/* Special case for ALSAAudio. */
1206	if ( pszDev
1207	&& RTStrIStr("pulse", pszDev) != NULL)
1208	LogRel2(("ALSA: ALSAAudio plugin in use\n"));
1209
1210	if (pszIOID)
1211	free(pszIOID);
1212
1213	if (pszDev)
1214	free(pszDev);
1215
1216	pszHintCur++;
1217	}
1218
1219	snd_device_name_free_hint((void **)pszHints);
1220	pszHints = NULL;
1221	}
1222	else
1223	LogRel2(("ALSA: Error enumerating PCM devices: %Rrc (%d)\n", RTErrConvertFromErrno(err), err));
1224
1225	/* ALSA allows exactly one input and one output used at a time for the selected device(s). */
1226	pBackendCfg->cMaxStreamsIn = 1;
1227	pBackendCfg->cMaxStreamsOut = 1;
1228
1229	return VINF_SUCCESS;
1230	}
1231
1232
1233	/**
1234	* @interface_method_impl{PDMIHOSTAUDIO,pfnShutdown}
1235	*/
1236	static DECLCALLBACK(void) drvHostAlsaAudioHA_Shutdown(PPDMIHOSTAUDIO pInterface)
1237	{
1238	RT_NOREF(pInterface);
1239	}
1240
1241
1242	/**
1243	* @interface_method_impl{PDMIHOSTAUDIO,pfnGetStatus}
1244	*/
1245	static DECLCALLBACK(PDMAUDIOBACKENDSTS) drvHostAlsaAudioHA_GetStatus(PPDMIHOSTAUDIO pInterface, PDMAUDIODIR enmDir)
1246	{
1247	RT_NOREF(enmDir);
1248	AssertPtrReturn(pInterface, PDMAUDIOBACKENDSTS_UNKNOWN);
1249
1250	return PDMAUDIOBACKENDSTS_RUNNING;
1251	}
1252
1253
1254	/**
1255	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamCreate}
1256	*/
1257	static DECLCALLBACK(int) drvHostAlsaAudioHA_StreamCreate(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream,
1258	PPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq)
1259	{
1260	AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
1261	AssertPtrReturn(pStream, VERR_INVALID_POINTER);
1262	AssertPtrReturn(pCfgReq, VERR_INVALID_POINTER);
1263	AssertPtrReturn(pCfgAcq, VERR_INVALID_POINTER);
1264
1265	PALSAAUDIOSTREAM pStreamALSA = (PALSAAUDIOSTREAM)pStream;
1266
1267	int rc;
1268	if (pCfgReq->enmDir == PDMAUDIODIR_IN)
1269	rc = alsaCreateStreamIn( pStreamALSA, pCfgReq, pCfgAcq);
1270	else
1271	rc = alsaCreateStreamOut(pStreamALSA, pCfgReq, pCfgAcq);
1272
1273	if (RT_SUCCESS(rc))
1274	{
1275	pStreamALSA->pCfg = DrvAudioHlpStreamCfgDup(pCfgAcq);
1276	if (!pStreamALSA->pCfg)
1277	rc = VERR_NO_MEMORY;
1278	}
1279
1280	return rc;
1281	}
1282
1283
1284	/**
1285	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamDestroy}
1286	*/
1287	static DECLCALLBACK(int) drvHostAlsaAudioHA_StreamDestroy(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
1288	{
1289	AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
1290	AssertPtrReturn(pStream, VERR_INVALID_POINTER);
1291
1292	PALSAAUDIOSTREAM pStreamALSA = (PALSAAUDIOSTREAM)pStream;
1293
1294	if (!pStreamALSA->pCfg) /* Not (yet) configured? Skip. */
1295	return VINF_SUCCESS;
1296
1297	int rc;
1298	if (pStreamALSA->pCfg->enmDir == PDMAUDIODIR_IN)
1299	rc = alsaDestroyStreamIn(pStreamALSA);
1300	else
1301	rc = alsaDestroyStreamOut(pStreamALSA);
1302
1303	if (RT_SUCCESS(rc))
1304	{
1305	DrvAudioHlpStreamCfgFree(pStreamALSA->pCfg);
1306	pStreamALSA->pCfg = NULL;
1307	}
1308
1309	return rc;
1310	}
1311
1312
1313	/**
1314	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamControl}
1315	*/
1316	static DECLCALLBACK(int) drvHostAlsaAudioHA_StreamControl(PPDMIHOSTAUDIO pInterface,
1317	PPDMAUDIOBACKENDSTREAM pStream, PDMAUDIOSTREAMCMD enmStreamCmd)
1318	{
1319	AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
1320	AssertPtrReturn(pStream, VERR_INVALID_POINTER);
1321
1322	PALSAAUDIOSTREAM pStreamALSA = (PALSAAUDIOSTREAM)pStream;
1323
1324	if (!pStreamALSA->pCfg) /* Not (yet) configured? Skip. */
1325	return VINF_SUCCESS;
1326
1327	int rc;
1328	if (pStreamALSA->pCfg->enmDir == PDMAUDIODIR_IN)
1329	rc = alsaControlStreamIn (pStreamALSA, enmStreamCmd);
1330	else
1331	rc = alsaControlStreamOut(pStreamALSA, enmStreamCmd);
1332
1333	return rc;
1334	}
1335
1336
1337	/**
1338	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetReadable}
1339	*/
1340	static DECLCALLBACK(uint32_t) drvHostAlsaAudioHA_StreamGetReadable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
1341	{
1342	RT_NOREF(pInterface);
1343
1344	PALSAAUDIOSTREAM pStreamALSA = (PALSAAUDIOSTREAM)pStream;
1345
1346	uint32_t cbAvail = 0;
1347
1348	snd_pcm_sframes_t cFramesAvail;
1349	int rc = alsaStreamGetAvail(pStreamALSA->phPCM, &cFramesAvail);
1350	if (RT_SUCCESS(rc))
1351	cbAvail = PDMAUDIOSTREAMCFG_F2B(pStreamALSA->pCfg, cFramesAvail);
1352
1353	return cbAvail;
1354	}
1355
1356
1357	/**
1358	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetWritable}
1359	*/
1360	static DECLCALLBACK(uint32_t) drvHostAlsaAudioHA_StreamGetWritable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
1361	{
1362	RT_NOREF(pInterface);
1363
1364	PALSAAUDIOSTREAM pStreamALSA = (PALSAAUDIOSTREAM)pStream;
1365
1366	uint32_t cbAvail = 0;
1367
1368	snd_pcm_sframes_t cFramesAvail;
1369	int rc = alsaStreamGetAvail(pStreamALSA->phPCM, &cFramesAvail);
1370	if (RT_SUCCESS(rc))
1371	cbAvail = PDMAUDIOSTREAMCFG_F2B(pStreamALSA->pCfg, cFramesAvail);
1372
1373	return cbAvail;
1374	}
1375
1376
1377	/**
1378	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetPending}
1379	*/
1380	static DECLCALLBACK(uint32_t) drvHostALSAStreamGetPending(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
1381	{
1382	RT_NOREF(pInterface);
1383	AssertPtrReturn(pStream, 0);
1384
1385	PALSAAUDIOSTREAM pStreamALSA = (PALSAAUDIOSTREAM)pStream;
1386
1387	snd_pcm_sframes_t cFramesDelay = 0;
1388	snd_pcm_state_t enmState = snd_pcm_state(pStreamALSA->phPCM);
1389
1390	int rc = VINF_SUCCESS;
1391
1392	AssertPtr(pStreamALSA->pCfg);
1393	if (pStreamALSA->pCfg->enmDir == PDMAUDIODIR_OUT)
1394	{
1395	/* Getting the delay (in audio frames) reports the time it will take
1396	* to hear a new sample after all queued samples have been played out. */
1397	int rc2 = snd_pcm_delay(pStreamALSA->phPCM, &cFramesDelay);
1398	if (RT_SUCCESS(rc))
1399	rc = rc2;
1400
1401	/* Make sure to check the stream's status.
1402	* If it's anything but SND_PCM_STATE_RUNNING, the delay is meaningless and therefore 0. */
1403	if (enmState != SND_PCM_STATE_RUNNING)
1404	cFramesDelay = 0;
1405	}
1406
1407	/* Note: For input streams we never have pending data left. */
1408
1409	Log2Func(("cFramesDelay=%RI32, enmState=%d, rc=%d\n", cFramesDelay, enmState, rc));
1410
1411	return DrvAudioHlpFramesToBytes(cFramesDelay, &pStreamALSA->pCfg->Props);
1412	}
1413
1414
1415	/**
1416	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetStatus}
1417	*/
1418	static DECLCALLBACK(PDMAUDIOSTREAMSTS) drvHostAlsaAudioHA_StreamGetStatus(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
1419	{
1420	RT_NOREF(pInterface, pStream);
1421
1422	return PDMAUDIOSTREAMSTS_FLAGS_INITIALIZED \| PDMAUDIOSTREAMSTS_FLAGS_ENABLED;
1423	}
1424
1425
1426	/**
1427	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamIterate}
1428	*/
1429	static DECLCALLBACK(int) drvHostAlsaAudioHA_StreamIterate(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
1430	{
1431	AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
1432	AssertPtrReturn(pStream, VERR_INVALID_POINTER);
1433
1434	LogFlowFuncEnter();
1435
1436	/* Nothing to do here for ALSA. */
1437	return VINF_SUCCESS;
1438	}
1439
1440
1441	/**
1442	* @interface_method_impl{PDMIBASE,pfnQueryInterface}
1443	*/
1444	static DECLCALLBACK(void ) drvHostAlsaAudioQueryInterface(PPDMIBASE pInterface, const char pszIID)
1445	{
1446	PPDMDRVINS pDrvIns = PDMIBASE_2_PDMDRV(pInterface);
1447	PDRVHOSTALSAAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVHOSTALSAAUDIO);
1448	PDMIBASE_RETURN_INTERFACE(pszIID, PDMIBASE, &pDrvIns->IBase);
1449	PDMIBASE_RETURN_INTERFACE(pszIID, PDMIHOSTAUDIO, &pThis->IHostAudio);
1450
1451	return NULL;
1452	}
1453
1454
1455	/**
1456	* Construct a DirectSound Audio driver instance.
1457	*
1458	* @copydoc FNPDMDRVCONSTRUCT
1459	*/
1460	static DECLCALLBACK(int) drvHostAlsaAudioConstruct(PPDMDRVINS pDrvIns, PCFGMNODE pCfg, uint32_t fFlags)
1461	{
1462	RT_NOREF(pCfg, fFlags);
1463	PDMDRV_CHECK_VERSIONS_RETURN(pDrvIns);
1464	PDRVHOSTALSAAUDIO pThis = PDMINS_2_DATA(pDrvIns, PDRVHOSTALSAAUDIO);
1465	LogRel(("Audio: Initializing ALSA driver\n"));
1466
1467	/*
1468	* Init the static parts.
1469	*/
1470	pThis->pDrvIns = pDrvIns;
1471	/* IBase */
1472	pDrvIns->IBase.pfnQueryInterface = drvHostAlsaAudioQueryInterface;
1473	/* IHostAudio */
1474	pThis->IHostAudio.pfnInit = drvHostAlsaAudioHA_Init;
1475	pThis->IHostAudio.pfnShutdown = drvHostAlsaAudioHA_Shutdown;
1476	pThis->IHostAudio.pfnGetConfig = drvHostAlsaAudioHA_GetConfig;
1477	pThis->IHostAudio.pfnGetStatus = drvHostAlsaAudioHA_GetStatus;
1478	pThis->IHostAudio.pfnStreamCreate = drvHostAlsaAudioHA_StreamCreate;
1479	pThis->IHostAudio.pfnStreamDestroy = drvHostAlsaAudioHA_StreamDestroy;
1480	pThis->IHostAudio.pfnStreamControl = drvHostAlsaAudioHA_StreamControl;
1481	pThis->IHostAudio.pfnStreamGetReadable = drvHostAlsaAudioHA_StreamGetReadable;
1482	pThis->IHostAudio.pfnStreamGetWritable = drvHostAlsaAudioHA_StreamGetWritable;
1483	pThis->IHostAudio.pfnStreamGetStatus = drvHostAlsaAudioHA_StreamGetStatus;
1484	pThis->IHostAudio.pfnStreamIterate = drvHostAlsaAudioHA_StreamIterate;
1485	pThis->IHostAudio.pfnStreamPlay = drvHostAlsaAudioHA_StreamPlay;
1486	pThis->IHostAudio.pfnStreamCapture = drvHostAlsaAudioHA_StreamCapture;
1487	pThis->IHostAudio.pfnSetCallback = NULL;
1488	pThis->IHostAudio.pfnGetDevices = NULL;
1489	pThis->IHostAudio.pfnStreamGetPending = drvHostALSAStreamGetPending;
1490	pThis->IHostAudio.pfnStreamPlayBegin = NULL;
1491	pThis->IHostAudio.pfnStreamPlayEnd = NULL;
1492	pThis->IHostAudio.pfnStreamCaptureBegin = NULL;
1493	pThis->IHostAudio.pfnStreamCaptureEnd = NULL;
1494
1495
1496	return VINF_SUCCESS;
1497	}
1498
1499
1500	/**
1501	* Char driver registration record.
1502	*/
1503	const PDMDRVREG g_DrvHostALSAAudio =
1504	{
1505	/* u32Version */
1506	PDM_DRVREG_VERSION,
1507	/* szName */
1508	"ALSAAudio",
1509	/* szRCMod */
1510	"",
1511	/* szR0Mod */
1512	"",
1513	/* pszDescription */
1514	"ALSA host audio driver",
1515	/* fFlags */
1516	PDM_DRVREG_FLAGS_HOST_BITS_DEFAULT,
1517	/* fClass. */
1518	PDM_DRVREG_CLASS_AUDIO,
1519	/* cMaxInstances */
1520	~0U,
1521	/* cbInstance */
1522	sizeof(DRVHOSTALSAAUDIO),
1523	/* pfnConstruct */
1524	drvHostAlsaAudioConstruct,
1525	/* pfnDestruct */
1526	NULL,
1527	/* pfnRelocate */
1528	NULL,
1529	/* pfnIOCtl */
1530	NULL,
1531	/* pfnPowerOn */
1532	NULL,
1533	/* pfnReset */
1534	NULL,
1535	/* pfnSuspend */
1536	NULL,
1537	/* pfnResume */
1538	NULL,
1539	/* pfnAttach */
1540	NULL,
1541	/* pfnDetach */
1542	NULL,
1543	/* pfnPowerOff */
1544	NULL,
1545	/* pfnSoftReset */
1546	NULL,
1547	/* u32EndVersion */
1548	PDM_DRVREG_VERSION
1549	};
1550

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source: vbox/trunk/src/VBox/Devices/Audio/DrvHostALSAAudio.cpp@ 82331

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