DrvHostAudioDSound.cpp@ 94368

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1	/* $Id: DrvHostAudioDSound.cpp 93115 2022-01-01 11:31:46Z vboxsync $ */
2	/** @file
3	* Host audio driver - DirectSound (Windows).
4	*/
5
6	/*
7	* Copyright (C) 2006-2022 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
19	/*********************************************************************************************************************************
20	* Header Files *
21	*********************************************************************************************************************************/
22	#define LOG_GROUP LOG_GROUP_DRV_HOST_AUDIO
23	#define INITGUID
24	#include <VBox/log.h>
25	#include <iprt/win/windows.h>
26	#include <dsound.h>
27	#include <mmdeviceapi.h>
28	#include <functiondiscoverykeys_devpkey.h>
29	#include <iprt/win/mmreg.h> /* WAVEFORMATEXTENSIBLE */
30
31	#include <iprt/alloc.h>
32	#include <iprt/system.h>
33	#include <iprt/uuid.h>
34	#include <iprt/utf16.h>
35
36	#include <VBox/vmm/pdmaudioinline.h>
37	#include <VBox/vmm/pdmaudiohostenuminline.h>
38
39	#include "VBoxDD.h"
40
41	#ifdef VBOX_WITH_AUDIO_MMNOTIFICATION_CLIENT
42	# include <new> /* For bad_alloc. */
43	# include "DrvHostAudioDSoundMMNotifClient.h"
44	#endif
45
46
47	/*********************************************************************************************************************************
48	* Defined Constants And Macros *
49	*********************************************************************************************************************************/
50	/*
51	* Optional release logging, which a user can turn on with the
52	* 'VBoxManage debugvm' command.
53	* Debug logging still uses the common Log* macros from VBox.
54	* Messages which always should go to the release log use LogRel.
55	*
56	* @deprecated Use LogRelMax, LogRel2 and LogRel3 directly.
57	*/
58	/** General code behavior. */
59	#define DSLOG(a) do { LogRel2(a); } while(0)
60	/** Something which produce a lot of logging during playback/recording. */
61	#define DSLOGF(a) do { LogRel3(a); } while(0)
62	/** Important messages like errors. Limited in the default release log to avoid log flood. */
63	#define DSLOGREL(a) \
64	do { \
65	static int8_t s_cLogged = 0; \
66	if (s_cLogged < 8) { \
67	++s_cLogged; \
68	LogRel(a); \
69	} else DSLOG(a); \
70	} while (0)
71
72	/** Maximum number of attempts to restore the sound buffer before giving up. */
73	#define DRV_DSOUND_RESTORE_ATTEMPTS_MAX 3
74	#if 0 /** @todo r=bird: What are these for? Nobody is using them... */
75	/** Default input latency (in ms). */
76	#define DRV_DSOUND_DEFAULT_LATENCY_MS_IN 50
77	/** Default output latency (in ms). */
78	#define DRV_DSOUND_DEFAULT_LATENCY_MS_OUT 50
79	#endif
80
81
82	/*********************************************************************************************************************************
83	* Structures and Typedefs *
84	*********************************************************************************************************************************/
85	/* Dynamically load dsound.dll. */
86	typedef HRESULT WINAPI FNDIRECTSOUNDENUMERATEW(LPDSENUMCALLBACKW pDSEnumCallback, PVOID pContext);
87	typedef FNDIRECTSOUNDENUMERATEW *PFNDIRECTSOUNDENUMERATEW;
88	typedef HRESULT WINAPI FNDIRECTSOUNDCAPTUREENUMERATEW(LPDSENUMCALLBACKW pDSEnumCallback, PVOID pContext);
89	typedef FNDIRECTSOUNDCAPTUREENUMERATEW *PFNDIRECTSOUNDCAPTUREENUMERATEW;
90	typedef HRESULT WINAPI FNDIRECTSOUNDCAPTURECREATE8(LPCGUID lpcGUID, LPDIRECTSOUNDCAPTURE8 *lplpDSC, LPUNKNOWN pUnkOuter);
91	typedef FNDIRECTSOUNDCAPTURECREATE8 *PFNDIRECTSOUNDCAPTURECREATE8;
92
93	#define VBOX_DSOUND_MAX_EVENTS 3
94
95	typedef enum DSOUNDEVENT
96	{
97	DSOUNDEVENT_NOTIFY = 0,
98	DSOUNDEVENT_INPUT,
99	DSOUNDEVENT_OUTPUT,
100	} DSOUNDEVENT;
101
102	typedef struct DSOUNDHOSTCFG
103	{
104	RTUUID uuidPlay;
105	LPCGUID pGuidPlay;
106	RTUUID uuidCapture;
107	LPCGUID pGuidCapture;
108	} DSOUNDHOSTCFG, *PDSOUNDHOSTCFG;
109
110	typedef struct DSOUNDSTREAM
111	{
112	/** Common part. */
113	PDMAUDIOBACKENDSTREAM Core;
114	/** Entry in DRVHOSTDSOUND::HeadStreams. */
115	RTLISTNODE ListEntry;
116	/** The stream's acquired configuration. */
117	PDMAUDIOSTREAMCFG Cfg;
118	/** Buffer alignment. */
119	uint8_t uAlign;
120	/** Whether this stream is in an enable state on the DirectSound side. */
121	bool fEnabled;
122	bool afPadding[2];
123	/** Size (in bytes) of the DirectSound buffer. */
124	DWORD cbBufSize;
125	union
126	{
127	struct
128	{
129	/** The actual DirectSound Buffer (DSB) used for the capturing.
130	* This is a secondary buffer and is used as a streaming buffer. */
131	LPDIRECTSOUNDCAPTUREBUFFER8 pDSCB;
132	/** Current read offset (in bytes) within the DSB. */
133	DWORD offReadPos;
134	/** Number of buffer overruns happened. Used for logging. */
135	uint8_t cOverruns;
136	} In;
137	struct
138	{
139	/** The actual DirectSound Buffer (DSB) used for playback.
140	* This is a secondary buffer and is used as a streaming buffer. */
141	LPDIRECTSOUNDBUFFER8 pDSB;
142	/** Current write offset (in bytes) within the DSB.
143	* @note This is needed as the current write position as kept by direct sound
144	* will move ahead if we're too late. */
145	DWORD offWritePos;
146	/** Offset of last play cursor within the DSB when checked for pending. */
147	DWORD offPlayCursorLastPending;
148	/** Offset of last play cursor within the DSB when last played. */
149	DWORD offPlayCursorLastPlayed;
150	/** Total amount (in bytes) written to our internal ring buffer. */
151	uint64_t cbWritten;
152	/** Total amount (in bytes) played (to the DirectSound buffer). */
153	uint64_t cbTransferred;
154	/** Flag indicating whether playback was just (re)started. */
155	bool fFirstTransfer;
156	/** Flag indicating whether this stream is in draining mode, e.g. no new
157	* data is being written to it but DirectSound still needs to be able to
158	* play its remaining (buffered) data. */
159	bool fDrain;
160	/** How much (in bytes) the last transfer from the internal buffer
161	* to the DirectSound buffer was. */
162	uint32_t cbLastTransferred;
163	/** The RTTimeMilliTS() deadline for the draining of this stream. */
164	uint64_t msDrainDeadline;
165	} Out;
166	};
167	/** Timestamp (in ms) of the last transfer from the internal buffer to/from the
168	* DirectSound buffer. */
169	uint64_t msLastTransfer;
170	/** The stream's critical section for synchronizing access. */
171	RTCRITSECT CritSect;
172	/** Used for formatting the current DSound status. */
173	char szStatus[127];
174	/** Fixed zero terminator. */
175	char const chStateZero;
176	} DSOUNDSTREAM, *PDSOUNDSTREAM;
177
178	/**
179	* DirectSound-specific device entry.
180	*/
181	typedef struct DSOUNDDEV
182	{
183	PDMAUDIOHOSTDEV Core;
184	/** The GUID if handy. */
185	GUID Guid;
186	/** The GUID as a string (empty if default). */
187	char szGuid[RTUUID_STR_LENGTH];
188	} DSOUNDDEV;
189	/** Pointer to a DirectSound device entry. */
190	typedef DSOUNDDEV *PDSOUNDDEV;
191
192	/**
193	* Structure for holding a device enumeration context.
194	*/
195	typedef struct DSOUNDENUMCBCTX
196	{
197	/** Enumeration flags. */
198	uint32_t fFlags;
199	/** Pointer to device list to populate. */
200	PPDMAUDIOHOSTENUM pDevEnm;
201	} DSOUNDENUMCBCTX, *PDSOUNDENUMCBCTX;
202
203	typedef struct DRVHOSTDSOUND
204	{
205	/** Pointer to the driver instance structure. */
206	PPDMDRVINS pDrvIns;
207	/** Our audio host audio interface. */
208	PDMIHOSTAUDIO IHostAudio;
209	/** Critical section to serialize access. */
210	RTCRITSECT CritSect;
211	/** DirectSound configuration options. */
212	DSOUNDHOSTCFG Cfg;
213	/** List of devices of last enumeration. */
214	PDMAUDIOHOSTENUM DeviceEnum;
215	/** Whether this backend supports any audio input.
216	* @todo r=bird: This is not actually used for anything. */
217	bool fEnabledIn;
218	/** Whether this backend supports any audio output.
219	* @todo r=bird: This is not actually used for anything. */
220	bool fEnabledOut;
221	/** The Direct Sound playback interface. */
222	LPDIRECTSOUND8 pDS;
223	/** The Direct Sound capturing interface. */
224	LPDIRECTSOUNDCAPTURE8 pDSC;
225	/** List of streams (DSOUNDSTREAM).
226	* Requires CritSect ownership. */
227	RTLISTANCHOR HeadStreams;
228
229	#ifdef VBOX_WITH_AUDIO_MMNOTIFICATION_CLIENT
230	DrvHostAudioDSoundMMNotifClient *m_pNotificationClient;
231	#endif
232	} DRVHOSTDSOUND, *PDRVHOSTDSOUND;
233
234
235	/*********************************************************************************************************************************
236	* Internal Functions *
237	*********************************************************************************************************************************/
238	static HRESULT directSoundPlayRestore(PDRVHOSTDSOUND pThis, LPDIRECTSOUNDBUFFER8 pDSB);
239	static int drvHostDSoundStreamStopPlayback(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, bool fReset);
240
241	static int dsoundDevicesEnumerate(PDRVHOSTDSOUND pThis, PDMAUDIOHOSTENUM pDevEnm, uint32_t fEnum);
242
243	static void dsoundUpdateStatusInternal(PDRVHOSTDSOUND pThis);
244
245
246	#if defined(LOG_ENABLED) \|\| defined(RTLOG_REL_ENABLED)
247	/**
248	* Gets the stream status as a string for logging purposes.
249	*
250	* @returns Status string (pStreamDS->szStatus).
251	* @param pStreamDS The stream to get the status for.
252	*/
253	static const char *drvHostDSoundStreamStatusString(PDSOUNDSTREAM pStreamDS)
254	{
255	/*
256	* Out internal stream status first.
257	*/
258	size_t off;
259	if (pStreamDS->fEnabled)
260	{
261	memcpy(pStreamDS->szStatus, RT_STR_TUPLE("ENABLED "));
262	off = sizeof("ENABLED ") - 1;
263	}
264	else
265	{
266	memcpy(pStreamDS->szStatus, RT_STR_TUPLE("DISABLED"));
267	off = sizeof("DISABLED") - 1;
268	}
269
270	/*
271	* Direction specific stuff, returning with a status DWORD and string mappings for it.
272	*/
273	typedef struct DRVHOSTDSOUNDSFLAGS2STR
274	{
275	const char *pszMnemonic;
276	uint32_t cchMnemonic;
277	uint32_t fFlag;
278	} DRVHOSTDSOUNDSFLAGS2STR;
279	static const DRVHOSTDSOUNDSFLAGS2STR s_aCaptureFlags[] =
280	{
281	{ RT_STR_TUPLE(" CAPTURING"), DSCBSTATUS_CAPTURING },
282	{ RT_STR_TUPLE(" LOOPING"), DSCBSTATUS_LOOPING },
283	};
284	static const DRVHOSTDSOUNDSFLAGS2STR s_aPlaybackFlags[] =
285	{
286	{ RT_STR_TUPLE(" PLAYING"), DSBSTATUS_PLAYING },
287	{ RT_STR_TUPLE(" BUFFERLOST"), DSBSTATUS_BUFFERLOST },
288	{ RT_STR_TUPLE(" LOOPING"), DSBSTATUS_LOOPING },
289	{ RT_STR_TUPLE(" LOCHARDWARE"), DSBSTATUS_LOCHARDWARE },
290	{ RT_STR_TUPLE(" LOCSOFTWARE"), DSBSTATUS_LOCSOFTWARE },
291	{ RT_STR_TUPLE(" TERMINATED"), DSBSTATUS_TERMINATED },
292	};
293	DRVHOSTDSOUNDSFLAGS2STR const *paMappings = NULL;
294	size_t cMappings = 0;
295	DWORD fStatus = 0;
296	if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN)
297	{
298	if (pStreamDS->In.pDSCB)
299	{
300	HRESULT hrc = pStreamDS->In.pDSCB->GetStatus(&fStatus);
301	if (SUCCEEDED(hrc))
302	{
303	paMappings = s_aCaptureFlags;
304	cMappings = RT_ELEMENTS(s_aCaptureFlags);
305	}
306	else
307	RTStrPrintf(&pStreamDS->szStatus[off], sizeof(pStreamDS->szStatus) - off, "GetStatus->%Rhrc", hrc);
308	}
309	else
310	RTStrCopy(&pStreamDS->szStatus[off], sizeof(pStreamDS->szStatus) - off, "NO-DSCB");
311	}
312	else if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT)
313	{
314	if (pStreamDS->Out.fDrain)
315	{
316	memcpy(&pStreamDS->szStatus[off], RT_STR_TUPLE(" DRAINING"));
317	off += sizeof(" DRAINING") - 1;
318	}
319
320	if (pStreamDS->Out.fFirstTransfer)
321	{
322	memcpy(&pStreamDS->szStatus[off], RT_STR_TUPLE(" NOXFER"));
323	off += sizeof(" NOXFER") - 1;
324	}
325
326	if (pStreamDS->Out.pDSB)
327	{
328	HRESULT hrc = pStreamDS->Out.pDSB->GetStatus(&fStatus);
329	if (SUCCEEDED(hrc))
330	{
331	paMappings = s_aPlaybackFlags;
332	cMappings = RT_ELEMENTS(s_aPlaybackFlags);
333	}
334	else
335	RTStrPrintf(&pStreamDS->szStatus[off], sizeof(pStreamDS->szStatus) - off, "GetStatus->%Rhrc", hrc);
336	}
337	else
338	RTStrCopy(&pStreamDS->szStatus[off], sizeof(pStreamDS->szStatus) - off, "NO-DSB");
339	}
340	else
341	RTStrCopy(&pStreamDS->szStatus[off], sizeof(pStreamDS->szStatus) - off, "BAD-DIR");
342
343	/* Format flags. */
344	if (paMappings)
345	{
346	if (fStatus == 0)
347	RTStrCopy(&pStreamDS->szStatus[off], sizeof(pStreamDS->szStatus) - off, " 0");
348	else
349	{
350	for (size_t i = 0; i < cMappings; i++)
351	if (fStatus & paMappings[i].fFlag)
352	{
353	memcpy(&pStreamDS->szStatus[off], paMappings[i].pszMnemonic, paMappings[i].cchMnemonic);
354	off += paMappings[i].cchMnemonic;
355
356	fStatus &= ~paMappings[i].fFlag;
357	if (!fStatus)
358	break;
359	}
360	if (fStatus != 0)
361	off += RTStrPrintf(&pStreamDS->szStatus[off], sizeof(pStreamDS->szStatus) - off, " %#x", fStatus);
362	}
363	}
364
365	/*
366	* Finally, terminate the string. By postponing it this long, it won't be
367	* a big deal if two threads go thru here at the same time as long as the
368	* status is the same.
369	*/
370	Assert(off < sizeof(pStreamDS->szStatus));
371	pStreamDS->szStatus[off] = '\0';
372
373	return pStreamDS->szStatus;
374	}
375	#endif /* LOG_ENABLED \|\| RTLOG_REL_ENABLED */
376
377
378	static DWORD dsoundRingDistance(DWORD offEnd, DWORD offBegin, DWORD cSize)
379	{
380	AssertReturn(offEnd <= cSize, 0);
381	AssertReturn(offBegin <= cSize, 0);
382
383	return offEnd >= offBegin ? offEnd - offBegin : cSize - offBegin + offEnd;
384	}
385
386
387	static char *dsoundGUIDToUtf8StrA(LPCGUID pGUID)
388	{
389	if (pGUID)
390	{
391	LPOLESTR lpOLEStr;
392	HRESULT hr = StringFromCLSID(*pGUID, &lpOLEStr);
393	if (SUCCEEDED(hr))
394	{
395	char *pszGUID;
396	int rc = RTUtf16ToUtf8(lpOLEStr, &pszGUID);
397	CoTaskMemFree(lpOLEStr);
398
399	return RT_SUCCESS(rc) ? pszGUID : NULL;
400	}
401	}
402
403	return RTStrDup("{Default device}");
404	}
405
406
407	static HRESULT directSoundPlayRestore(PDRVHOSTDSOUND pThis, LPDIRECTSOUNDBUFFER8 pDSB)
408	{
409	RT_NOREF(pThis);
410	HRESULT hr = IDirectSoundBuffer8_Restore(pDSB);
411	if (FAILED(hr))
412	DSLOG(("DSound: Restoring playback buffer\n"));
413	else
414	DSLOGREL(("DSound: Restoring playback buffer failed with %Rhrc\n", hr));
415
416	return hr;
417	}
418
419
420	static HRESULT directSoundPlayUnlock(PDRVHOSTDSOUND pThis, LPDIRECTSOUNDBUFFER8 pDSB,
421	PVOID pv1, PVOID pv2,
422	DWORD cb1, DWORD cb2)
423	{
424	RT_NOREF(pThis);
425	HRESULT hr = IDirectSoundBuffer8_Unlock(pDSB, pv1, cb1, pv2, cb2);
426	if (FAILED(hr))
427	DSLOGREL(("DSound: Unlocking playback buffer failed with %Rhrc\n", hr));
428	return hr;
429	}
430
431
432	static HRESULT directSoundPlayLock(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS,
433	DWORD dwOffset, DWORD dwBytes,
434	PVOID ppv1, PVOID ppv2,
435	DWORD pcb1, DWORD pcb2,
436	DWORD dwFlags)
437	{
438	AssertReturn(dwBytes, VERR_INVALID_PARAMETER);
439
440	HRESULT hr = E_FAIL;
441	AssertCompile(DRV_DSOUND_RESTORE_ATTEMPTS_MAX > 0);
442	for (unsigned i = 0; i < DRV_DSOUND_RESTORE_ATTEMPTS_MAX; i++)
443	{
444	PVOID pv1, pv2;
445	DWORD cb1, cb2;
446	hr = IDirectSoundBuffer8_Lock(pStreamDS->Out.pDSB, dwOffset, dwBytes, &pv1, &cb1, &pv2, &cb2, dwFlags);
447	if (SUCCEEDED(hr))
448	{
449	if ( (!pv1 \|\| !(cb1 & pStreamDS->uAlign))
450	&& (!pv2 \|\| !(cb2 & pStreamDS->uAlign)))
451	{
452	if (ppv1)
453	*ppv1 = pv1;
454	if (ppv2)
455	*ppv2 = pv2;
456	if (pcb1)
457	*pcb1 = cb1;
458	if (pcb2)
459	*pcb2 = cb2;
460	return S_OK;
461	}
462	DSLOGREL(("DSound: Locking playback buffer returned misaligned buffer: cb1=%#RX32, cb2=%#RX32 (alignment: %#RX32)\n",
463	pcb1, pcb2, pStreamDS->uAlign));
464	directSoundPlayUnlock(pThis, pStreamDS->Out.pDSB, pv1, pv2, cb1, cb2);
465	return E_FAIL;
466	}
467
468	if (hr != DSERR_BUFFERLOST)
469	break;
470
471	LogFlowFunc(("Locking failed due to lost buffer, restoring ...\n"));
472	directSoundPlayRestore(pThis, pStreamDS->Out.pDSB);
473	}
474
475	DSLOGREL(("DSound: Locking playback buffer failed with %Rhrc (dwOff=%ld, dwBytes=%ld)\n", hr, dwOffset, dwBytes));
476	return hr;
477	}
478
479
480	static HRESULT directSoundCaptureUnlock(LPDIRECTSOUNDCAPTUREBUFFER8 pDSCB,
481	PVOID pv1, PVOID pv2,
482	DWORD cb1, DWORD cb2)
483	{
484	HRESULT hr = IDirectSoundCaptureBuffer8_Unlock(pDSCB, pv1, cb1, pv2, cb2);
485	if (FAILED(hr))
486	DSLOGREL(("DSound: Unlocking capture buffer failed with %Rhrc\n", hr));
487	return hr;
488	}
489
490
491	static HRESULT directSoundCaptureLock(PDSOUNDSTREAM pStreamDS,
492	DWORD dwOffset, DWORD dwBytes,
493	PVOID ppv1, PVOID ppv2,
494	DWORD pcb1, DWORD pcb2,
495	DWORD dwFlags)
496	{
497	PVOID pv1 = NULL;
498	PVOID pv2 = NULL;
499	DWORD cb1 = 0;
500	DWORD cb2 = 0;
501
502	HRESULT hr = IDirectSoundCaptureBuffer8_Lock(pStreamDS->In.pDSCB, dwOffset, dwBytes,
503	&pv1, &cb1, &pv2, &cb2, dwFlags);
504	if (FAILED(hr))
505	{
506	DSLOGREL(("DSound: Locking capture buffer failed with %Rhrc\n", hr));
507	return hr;
508	}
509
510	if ( (pv1 && (cb1 & pStreamDS->uAlign))
511	\|\| (pv2 && (cb2 & pStreamDS->uAlign)))
512	{
513	DSLOGREL(("DSound: Locking capture buffer returned misaligned buffer: cb1=%RI32, cb2=%RI32 (alignment: %RU32)\n",
514	cb1, cb2, pStreamDS->uAlign));
515	directSoundCaptureUnlock(pStreamDS->In.pDSCB, pv1, pv2, cb1, cb2);
516	return E_FAIL;
517	}
518
519	*ppv1 = pv1;
520	*ppv2 = pv2;
521	*pcb1 = cb1;
522	*pcb2 = cb2;
523
524	return S_OK;
525	}
526
527
528	/*
529	* DirectSound playback
530	*/
531
532	/**
533	* Creates a DirectSound playback instance.
534	*
535	* @return HRESULT
536	* @param pGUID GUID of device to create the playback interface for. NULL
537	* for the default device.
538	* @param ppDS Where to return the interface to the created instance.
539	*/
540	static HRESULT drvHostDSoundCreateDSPlaybackInstance(LPCGUID pGUID, LPDIRECTSOUND8 *ppDS)
541	{
542	LogFlowFuncEnter();
543
544	LPDIRECTSOUND8 pDS = NULL;
545	HRESULT hrc = CoCreateInstance(CLSID_DirectSound8, NULL, CLSCTX_ALL, IID_IDirectSound8, (void **)&pDS);
546	if (SUCCEEDED(hrc))
547	{
548	hrc = IDirectSound8_Initialize(pDS, pGUID);
549	if (SUCCEEDED(hrc))
550	{
551	HWND hWnd = GetDesktopWindow();
552	hrc = IDirectSound8_SetCooperativeLevel(pDS, hWnd, DSSCL_PRIORITY);
553	if (SUCCEEDED(hrc))
554	{
555	*ppDS = pDS;
556	LogFlowFunc(("LEAVE S_OK\n"));
557	return S_OK;
558	}
559	LogRelMax(64, ("DSound: Setting cooperative level for (hWnd=%p) failed: %Rhrc\n", hWnd, hrc));
560	}
561	else if (hrc == DSERR_NODRIVER) /* Usually means that no playback devices are attached. */
562	LogRelMax(64, ("DSound: DirectSound playback is currently unavailable\n"));
563	else
564	LogRelMax(64, ("DSound: DirectSound playback initialization failed: %Rhrc\n", hrc));
565
566	IDirectSound8_Release(pDS);
567	}
568	else
569	LogRelMax(64, ("DSound: Creating playback instance failed: %Rhrc\n", hrc));
570
571	LogFlowFunc(("LEAVE %Rhrc\n", hrc));
572	return hrc;
573	}
574
575
576	#if 0 /* not used */
577	static HRESULT directSoundPlayGetStatus(PDRVHOSTDSOUND pThis, LPDIRECTSOUNDBUFFER8 pDSB, DWORD *pdwStatus)
578	{
579	AssertPtrReturn(pThis, E_POINTER);
580	AssertPtrReturn(pDSB, E_POINTER);
581
582	AssertPtrNull(pdwStatus);
583
584	DWORD dwStatus = 0;
585	HRESULT hr = E_FAIL;
586	for (unsigned i = 0; i < DRV_DSOUND_RESTORE_ATTEMPTS_MAX; i++)
587	{
588	hr = IDirectSoundBuffer8_GetStatus(pDSB, &dwStatus);
589	if ( hr == DSERR_BUFFERLOST
590	\|\| ( SUCCEEDED(hr)
591	&& (dwStatus & DSBSTATUS_BUFFERLOST)))
592	{
593	LogFlowFunc(("Getting status failed due to lost buffer, restoring ...\n"));
594	directSoundPlayRestore(pThis, pDSB);
595	}
596	else
597	break;
598	}
599
600	if (SUCCEEDED(hr))
601	{
602	if (pdwStatus)
603	*pdwStatus = dwStatus;
604	}
605	else
606	DSLOGREL(("DSound: Retrieving playback status failed with %Rhrc\n", hr));
607
608	return hr;
609	}
610	#endif
611
612
613	/*
614	* DirectSoundCapture
615	*/
616
617	#if 0 /* unused */
618	static LPCGUID dsoundCaptureSelectDevice(PDRVHOSTDSOUND pThis, PPDMAUDIOSTREAMCFG pCfg)
619	{
620	AssertPtrReturn(pThis, NULL);
621	AssertPtrReturn(pCfg, NULL);
622
623	int rc = VINF_SUCCESS;
624
625	LPCGUID pGUID = pThis->Cfg.pGuidCapture;
626	if (!pGUID)
627	{
628	PDSOUNDDEV pDev = NULL;
629	switch (pCfg->enmPath)
630	{
631	case PDMAUDIOPATH_IN_LINE:
632	/*
633	* At the moment we're only supporting line-in in the HDA emulation,
634	* and line-in + mic-in in the AC'97 emulation both are expected
635	* to use the host's mic-in as well.
636	*
637	* So the fall through here is intentional for now.
638	*/
639	case PDMAUDIOPATH_IN_MIC:
640	pDev = (PDSOUNDDEV)DrvAudioHlpDeviceEnumGetDefaultDevice(&pThis->DeviceEnum, PDMAUDIODIR_IN);
641	break;
642
643	default:
644	AssertFailedStmt(rc = VERR_NOT_SUPPORTED);
645	break;
646	}
647
648	if ( RT_SUCCESS(rc)
649	&& pDev)
650	{
651	DSLOG(("DSound: Guest source '%s' is using host recording device '%s'\n",
652	PDMAudioPathGetName(pCfg->enmPath), pDev->Core.szName));
653	pGUID = &pDev->Guid;
654	}
655	if (RT_FAILURE(rc))
656	{
657	LogRel(("DSound: Selecting recording device failed with %Rrc\n", rc));
658	return NULL;
659	}
660	}
661
662	/* This always has to be in the release log. */
663	char *pszGUID = dsoundGUIDToUtf8StrA(pGUID);
664	LogRel(("DSound: Guest source '%s' is using host recording device with GUID '%s'\n",
665	PDMAudioPathGetName(pCfg->enmPath), pszGUID ? pszGUID: "{?}"));
666	RTStrFree(pszGUID);
667
668	return pGUID;
669	}
670	#endif
671
672
673	/**
674	* Creates a DirectSound capture instance.
675	*
676	* @returns HRESULT
677	* @param pGUID GUID of device to create the capture interface for. NULL
678	* for default.
679	* @param ppDSC Where to return the interface to the created instance.
680	*/
681	static HRESULT drvHostDSoundCreateDSCaptureInstance(LPCGUID pGUID, LPDIRECTSOUNDCAPTURE8 *ppDSC)
682	{
683	LogFlowFuncEnter();
684
685	LPDIRECTSOUNDCAPTURE8 pDSC = NULL;
686	HRESULT hrc = CoCreateInstance(CLSID_DirectSoundCapture8, NULL, CLSCTX_ALL, IID_IDirectSoundCapture8, (void **)&pDSC);
687	if (SUCCEEDED(hrc))
688	{
689	hrc = IDirectSoundCapture_Initialize(pDSC, pGUID);
690	if (SUCCEEDED(hrc))
691	{
692	*ppDSC = pDSC;
693	LogFlowFunc(("LEAVE S_OK\n"));
694	return S_OK;
695	}
696	if (hrc == DSERR_NODRIVER) /* Usually means that no capture devices are attached. */
697	LogRelMax(64, ("DSound: Capture device currently is unavailable\n"));
698	else
699	LogRelMax(64, ("DSound: Initializing capturing device failed: %Rhrc\n", hrc));
700	IDirectSoundCapture_Release(pDSC);
701	}
702	else
703	LogRelMax(64, ("DSound: Creating capture instance failed: %Rhrc\n", hrc));
704
705	LogFlowFunc(("LEAVE %Rhrc\n", hrc));
706	return hrc;
707	}
708
709
710	/**
711	* Updates this host driver's internal status, according to the global, overall input/output
712	* state and all connected (native) audio streams.
713	*
714	* @todo r=bird: This is a 'ing waste of 'ing time! We're doing this everytime
715	* an 'ing stream is created and we doesn't 'ing use the information here
716	* for any darn thing! Given the reported slowness of enumeration and
717	* issues with the 'ing code the only appropriate response is:
718	* AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAARG!!!!!!!
719	*
720	* @param pThis Host audio driver instance.
721	*/
722	static void dsoundUpdateStatusInternal(PDRVHOSTDSOUND pThis)
723	{
724	#if 0 /** @todo r=bird: This isn't doing ANYTHING useful. So, I've just disabled it. */
725	AssertPtrReturnVoid(pThis);
726	LogFlowFuncEnter();
727
728	PDMAudioHostEnumDelete(&pThis->DeviceEnum);
729	int rc = dsoundDevicesEnumerate(pThis, &pThis->DeviceEnum);
730	if (RT_SUCCESS(rc))
731	{
732	#if 0
733	if ( pThis->fEnabledOut != RT_BOOL(cbCtx.cDevOut)
734	\|\| pThis->fEnabledIn != RT_BOOL(cbCtx.cDevIn))
735	{
736	/** @todo Use a registered callback to the audio connector (e.g "OnConfigurationChanged") to
737	* let the connector know that something has changed within the host backend. */
738	}
739	#endif
740	pThis->fEnabledIn = PDMAudioHostEnumCountMatching(&pThis->DeviceEnum, PDMAUDIODIR_IN) != 0;
741	pThis->fEnabledOut = PDMAudioHostEnumCountMatching(&pThis->DeviceEnum, PDMAUDIODIR_OUT) != 0;
742	}
743
744	LogFlowFuncLeaveRC(rc);
745	#else
746	RT_NOREF(pThis);
747	#endif
748	}
749
750
751	/*********************************************************************************************************************************
752	* PDMIHOSTAUDIO *
753	*********************************************************************************************************************************/
754
755	/**
756	* @interface_method_impl{PDMIHOSTAUDIO,pfnGetConfig}
757	*/
758	static DECLCALLBACK(int) drvHostDSoundHA_GetConfig(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDCFG pBackendCfg)
759	{
760	RT_NOREF(pInterface);
761	AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
762	AssertPtrReturn(pBackendCfg, VERR_INVALID_POINTER);
763
764
765	/*
766	* Fill in the config structure.
767	*/
768	RTStrCopy(pBackendCfg->szName, sizeof(pBackendCfg->szName), "DirectSound");
769	pBackendCfg->cbStream = sizeof(DSOUNDSTREAM);
770	pBackendCfg->fFlags = 0;
771	pBackendCfg->cMaxStreamsIn = UINT32_MAX;
772	pBackendCfg->cMaxStreamsOut = UINT32_MAX;
773
774	return VINF_SUCCESS;
775	}
776
777
778	/**
779	* Callback for the playback device enumeration.
780	*
781	* @return TRUE if continuing enumeration, FALSE if not.
782	* @param pGUID Pointer to GUID of enumerated device. Can be NULL.
783	* @param pwszDescription Pointer to (friendly) description of enumerated device.
784	* @param pwszModule Pointer to module name of enumerated device.
785	* @param lpContext Pointer to PDSOUNDENUMCBCTX context for storing the enumerated information.
786	*
787	* @note Carbon copy of drvHostDSoundEnumOldStyleCaptureCallback with OUT direction.
788	*/
789	static BOOL CALLBACK drvHostDSoundEnumOldStylePlaybackCallback(LPGUID pGUID, LPCWSTR pwszDescription,
790	LPCWSTR pwszModule, PVOID lpContext)
791	{
792	PDSOUNDENUMCBCTX pEnumCtx = (PDSOUNDENUMCBCTX)lpContext;
793	AssertPtrReturn(pEnumCtx, FALSE);
794
795	PPDMAUDIOHOSTENUM pDevEnm = pEnumCtx->pDevEnm;
796	AssertPtrReturn(pDevEnm, FALSE);
797
798	AssertPtrNullReturn(pGUID, FALSE); /* pGUID can be NULL for default device(s). */
799	AssertPtrReturn(pwszDescription, FALSE);
800	RT_NOREF(pwszModule); /* Do not care about pwszModule. */
801
802	int rc;
803	size_t const cbName = RTUtf16CalcUtf8Len(pwszDescription) + 1;
804	PDSOUNDDEV pDev = (PDSOUNDDEV)PDMAudioHostDevAlloc(sizeof(DSOUNDDEV), cbName, 0);
805	if (pDev)
806	{
807	pDev->Core.enmUsage = PDMAUDIODIR_OUT;
808	pDev->Core.enmType = PDMAUDIODEVICETYPE_BUILTIN;
809
810	if (pGUID == NULL)
811	pDev->Core.fFlags = PDMAUDIOHOSTDEV_F_DEFAULT_OUT;
812
813	rc = RTUtf16ToUtf8Ex(pwszDescription, RTSTR_MAX, &pDev->Core.pszName, cbName, NULL);
814	if (RT_SUCCESS(rc))
815	{
816	if (!pGUID)
817	pDev->Core.fFlags \|= PDMAUDIOHOSTDEV_F_DEFAULT_OUT;
818	else
819	{
820	memcpy(&pDev->Guid, pGUID, sizeof(pDev->Guid));
821	rc = RTUuidToStr((PCRTUUID)pGUID, pDev->szGuid, sizeof(pDev->szGuid));
822	AssertRC(rc);
823	}
824	pDev->Core.pszId = &pDev->szGuid[0];
825
826	PDMAudioHostEnumAppend(pDevEnm, &pDev->Core);
827
828	/* Note: Querying the actual device information will be done at some
829	* later point in time outside this enumeration callback to prevent
830	* DSound hangs. */
831	return TRUE;
832	}
833	PDMAudioHostDevFree(&pDev->Core);
834	}
835	else
836	rc = VERR_NO_MEMORY;
837
838	LogRel(("DSound: Error enumeration playback device '%ls': rc=%Rrc\n", pwszDescription, rc));
839	return FALSE; /* Abort enumeration. */
840	}
841
842
843	/**
844	* Callback for the capture device enumeration.
845	*
846	* @return TRUE if continuing enumeration, FALSE if not.
847	* @param pGUID Pointer to GUID of enumerated device. Can be NULL.
848	* @param pwszDescription Pointer to (friendly) description of enumerated device.
849	* @param pwszModule Pointer to module name of enumerated device.
850	* @param lpContext Pointer to PDSOUNDENUMCBCTX context for storing the enumerated information.
851	*
852	* @note Carbon copy of drvHostDSoundEnumOldStylePlaybackCallback with IN direction.
853	*/
854	static BOOL CALLBACK drvHostDSoundEnumOldStyleCaptureCallback(LPGUID pGUID, LPCWSTR pwszDescription,
855	LPCWSTR pwszModule, PVOID lpContext)
856	{
857	PDSOUNDENUMCBCTX pEnumCtx = (PDSOUNDENUMCBCTX )lpContext;
858	AssertPtrReturn(pEnumCtx, FALSE);
859
860	PPDMAUDIOHOSTENUM pDevEnm = pEnumCtx->pDevEnm;
861	AssertPtrReturn(pDevEnm, FALSE);
862
863	AssertPtrNullReturn(pGUID, FALSE); /* pGUID can be NULL for default device(s). */
864	AssertPtrReturn(pwszDescription, FALSE);
865	RT_NOREF(pwszModule); /* Do not care about pwszModule. */
866
867	int rc;
868	size_t const cbName = RTUtf16CalcUtf8Len(pwszDescription) + 1;
869	PDSOUNDDEV pDev = (PDSOUNDDEV)PDMAudioHostDevAlloc(sizeof(DSOUNDDEV), cbName, 0);
870	if (pDev)
871	{
872	pDev->Core.enmUsage = PDMAUDIODIR_IN;
873	pDev->Core.enmType = PDMAUDIODEVICETYPE_BUILTIN;
874
875	rc = RTUtf16ToUtf8Ex(pwszDescription, RTSTR_MAX, &pDev->Core.pszName, cbName, NULL);
876	if (RT_SUCCESS(rc))
877	{
878	if (!pGUID)
879	pDev->Core.fFlags \|= PDMAUDIOHOSTDEV_F_DEFAULT_IN;
880	else
881	{
882	memcpy(&pDev->Guid, pGUID, sizeof(pDev->Guid));
883	rc = RTUuidToStr((PCRTUUID)pGUID, pDev->szGuid, sizeof(pDev->szGuid));
884	AssertRC(rc);
885	}
886	pDev->Core.pszId = &pDev->szGuid[0];
887
888	PDMAudioHostEnumAppend(pDevEnm, &pDev->Core);
889
890	/* Note: Querying the actual device information will be done at some
891	* later point in time outside this enumeration callback to prevent
892	* DSound hangs. */
893	return TRUE;
894	}
895	PDMAudioHostDevFree(&pDev->Core);
896	}
897	else
898	rc = VERR_NO_MEMORY;
899
900	LogRel(("DSound: Error enumeration capture device '%ls', rc=%Rrc\n", pwszDescription, rc));
901	return FALSE; /* Abort enumeration. */
902	}
903
904
905	/**
906	* Queries information for a given (DirectSound) device.
907	*
908	* @returns VBox status code.
909	* @param pDev Audio device to query information for.
910	*/
911	static int drvHostDSoundEnumOldStyleQueryDeviceInfo(PDSOUNDDEV pDev)
912	{
913	AssertPtr(pDev);
914	int rc;
915
916	if (pDev->Core.enmUsage == PDMAUDIODIR_OUT)
917	{
918	LPDIRECTSOUND8 pDS;
919	HRESULT hr = drvHostDSoundCreateDSPlaybackInstance(&pDev->Guid, &pDS);
920	if (SUCCEEDED(hr))
921	{
922	DSCAPS DSCaps;
923	RT_ZERO(DSCaps);
924	DSCaps.dwSize = sizeof(DSCAPS);
925	hr = IDirectSound_GetCaps(pDS, &DSCaps);
926	if (SUCCEEDED(hr))
927	{
928	pDev->Core.cMaxOutputChannels = DSCaps.dwFlags & DSCAPS_PRIMARYSTEREO ? 2 : 1;
929
930	DWORD dwSpeakerCfg;
931	hr = IDirectSound_GetSpeakerConfig(pDS, &dwSpeakerCfg);
932	if (SUCCEEDED(hr))
933	{
934	unsigned uSpeakerCount = 0;
935	switch (DSSPEAKER_CONFIG(dwSpeakerCfg))
936	{
937	case DSSPEAKER_MONO: uSpeakerCount = 1; break;
938	case DSSPEAKER_HEADPHONE: uSpeakerCount = 2; break;
939	case DSSPEAKER_STEREO: uSpeakerCount = 2; break;
940	case DSSPEAKER_QUAD: uSpeakerCount = 4; break;
941	case DSSPEAKER_SURROUND: uSpeakerCount = 4; break;
942	case DSSPEAKER_5POINT1: uSpeakerCount = 6; break;
943	case DSSPEAKER_5POINT1_SURROUND: uSpeakerCount = 6; break;
944	case DSSPEAKER_7POINT1: uSpeakerCount = 8; break;
945	case DSSPEAKER_7POINT1_SURROUND: uSpeakerCount = 8; break;
946	default: break;
947	}
948
949	if (uSpeakerCount) /* Do we need to update the channel count? */
950	pDev->Core.cMaxOutputChannels = uSpeakerCount;
951
952	rc = VINF_SUCCESS;
953	}
954	else
955	{
956	LogRel(("DSound: Error retrieving playback device speaker config, hr=%Rhrc\n", hr));
957	rc = VERR_ACCESS_DENIED; /** @todo Fudge! */
958	}
959	}
960	else
961	{
962	LogRel(("DSound: Error retrieving playback device capabilities, hr=%Rhrc\n", hr));
963	rc = VERR_ACCESS_DENIED; /** @todo Fudge! */
964	}
965
966	IDirectSound8_Release(pDS);
967	}
968	else
969	rc = VERR_GENERAL_FAILURE;
970	}
971	else if (pDev->Core.enmUsage == PDMAUDIODIR_IN)
972	{
973	LPDIRECTSOUNDCAPTURE8 pDSC;
974	HRESULT hr = drvHostDSoundCreateDSCaptureInstance(&pDev->Guid, &pDSC);
975	if (SUCCEEDED(hr))
976	{
977	DSCCAPS DSCCaps;
978	RT_ZERO(DSCCaps);
979	DSCCaps.dwSize = sizeof(DSCCAPS);
980	hr = IDirectSoundCapture_GetCaps(pDSC, &DSCCaps);
981	if (SUCCEEDED(hr))
982	{
983	pDev->Core.cMaxInputChannels = DSCCaps.dwChannels;
984	rc = VINF_SUCCESS;
985	}
986	else
987	{
988	LogRel(("DSound: Error retrieving capture device capabilities, hr=%Rhrc\n", hr));
989	rc = VERR_ACCESS_DENIED; /** @todo Fudge! */
990	}
991
992	IDirectSoundCapture_Release(pDSC);
993	}
994	else
995	rc = VERR_GENERAL_FAILURE;
996	}
997	else
998	AssertFailedStmt(rc = VERR_NOT_SUPPORTED);
999
1000	return rc;
1001	}
1002
1003
1004	/**
1005	* Queries information for @a pDevice and adds an entry to the enumeration.
1006	*
1007	* @returns VBox status code.
1008	* @param pDevEnm The enumeration to add the device to.
1009	* @param pDevice The device.
1010	* @param enmType The type of device.
1011	* @param fDefault Whether it's the default device.
1012	*/
1013	static int drvHostDSoundEnumNewStyleAdd(PPDMAUDIOHOSTENUM pDevEnm, IMMDevice *pDevice, EDataFlow enmType, bool fDefault)
1014	{
1015	int rc = VINF_SUCCESS; /* ignore most errors */
1016
1017	/*
1018	* Gather the necessary properties.
1019	*/
1020	IPropertyStore *pProperties = NULL;
1021	HRESULT hrc = pDevice->OpenPropertyStore(STGM_READ, &pProperties);
1022	if (SUCCEEDED(hrc))
1023	{
1024	/* Get the friendly name. */
1025	PROPVARIANT VarName;
1026	PropVariantInit(&VarName);
1027	hrc = pProperties->GetValue(PKEY_Device_FriendlyName, &VarName);
1028	if (SUCCEEDED(hrc))
1029	{
1030	/* Get the DirectSound GUID. */
1031	PROPVARIANT VarGUID;
1032	PropVariantInit(&VarGUID);
1033	hrc = pProperties->GetValue(PKEY_AudioEndpoint_GUID, &VarGUID);
1034	if (SUCCEEDED(hrc))
1035	{
1036	/* Get the device format. */
1037	PROPVARIANT VarFormat;
1038	PropVariantInit(&VarFormat);
1039	hrc = pProperties->GetValue(PKEY_AudioEngine_DeviceFormat, &VarFormat);
1040	if (SUCCEEDED(hrc))
1041	{
1042	WAVEFORMATEX const * const pFormat = (WAVEFORMATEX const *)VarFormat.blob.pBlobData;
1043	AssertPtr(pFormat);
1044
1045	/*
1046	* Create a enumeration entry for it.
1047	*/
1048	size_t const cbName = RTUtf16CalcUtf8Len(VarName.pwszVal) + 1;
1049	PDSOUNDDEV pDev = (PDSOUNDDEV)PDMAudioHostDevAlloc(sizeof(DSOUNDDEV), cbName, 0);
1050	if (pDev)
1051	{
1052	pDev->Core.enmUsage = enmType == eRender ? PDMAUDIODIR_OUT : PDMAUDIODIR_IN;
1053	pDev->Core.enmType = PDMAUDIODEVICETYPE_BUILTIN;
1054	if (fDefault)
1055	pDev->Core.fFlags \|= enmType == eRender
1056	? PDMAUDIOHOSTDEV_F_DEFAULT_OUT : PDMAUDIOHOSTDEV_F_DEFAULT_IN;
1057	if (enmType == eRender)
1058	pDev->Core.cMaxOutputChannels = pFormat->nChannels;
1059	else
1060	pDev->Core.cMaxInputChannels = pFormat->nChannels;
1061
1062	//if (fDefault)
1063	rc = RTUuidFromUtf16((PRTUUID)&pDev->Guid, VarGUID.pwszVal);
1064	if (RT_SUCCESS(rc))
1065	{
1066	rc = RTUuidToStr((PCRTUUID)&pDev->Guid, pDev->szGuid, sizeof(pDev->szGuid));
1067	AssertRC(rc);
1068	pDev->Core.pszId = &pDev->szGuid[0];
1069
1070	rc = RTUtf16ToUtf8Ex(VarName.pwszVal, RTSTR_MAX, &pDev->Core.pszName, cbName, NULL);
1071	if (RT_SUCCESS(rc))
1072	PDMAudioHostEnumAppend(pDevEnm, &pDev->Core);
1073	else
1074	PDMAudioHostDevFree(&pDev->Core);
1075	}
1076	else
1077	{
1078	LogFunc(("RTUuidFromUtf16(%ls): %Rrc\n", VarGUID.pwszVal, rc));
1079	PDMAudioHostDevFree(&pDev->Core);
1080	}
1081	}
1082	else
1083	rc = VERR_NO_MEMORY;
1084	PropVariantClear(&VarFormat);
1085	}
1086	else
1087	LogFunc(("Failed to get PKEY_AudioEngine_DeviceFormat: %Rhrc\n", hrc));
1088	PropVariantClear(&VarGUID);
1089	}
1090	else
1091	LogFunc(("Failed to get PKEY_AudioEndpoint_GUID: %Rhrc\n", hrc));
1092	PropVariantClear(&VarName);
1093	}
1094	else
1095	LogFunc(("Failed to get PKEY_Device_FriendlyName: %Rhrc\n", hrc));
1096	pProperties->Release();
1097	}
1098	else
1099	LogFunc(("OpenPropertyStore failed: %Rhrc\n", hrc));
1100
1101	if (hrc == E_OUTOFMEMORY && RT_SUCCESS_NP(rc))
1102	rc = VERR_NO_MEMORY;
1103	return rc;
1104	}
1105
1106
1107	/**
1108	* Does a (Re-)enumeration of the host's playback + capturing devices.
1109	*
1110	* @return VBox status code.
1111	* @param pDevEnm Where to store the enumerated devices.
1112	*/
1113	static int drvHostDSoundEnumerateDevices(PPDMAUDIOHOSTENUM pDevEnm)
1114	{
1115	DSLOG(("DSound: Enumerating devices ...\n"));
1116
1117	/*
1118	* Use the Vista+ API.
1119	*/
1120	IMMDeviceEnumerator *pEnumerator;
1121	HRESULT hrc = CoCreateInstance(__uuidof(MMDeviceEnumerator), 0, CLSCTX_ALL,
1122	__uuidof(IMMDeviceEnumerator), (void **)&pEnumerator);
1123	if (SUCCEEDED(hrc))
1124	{
1125	int rc = VINF_SUCCESS;
1126	for (unsigned idxPass = 0; idxPass < 2 && RT_SUCCESS(rc); idxPass++)
1127	{
1128	EDataFlow const enmType = idxPass == 0 ? EDataFlow::eRender : EDataFlow::eCapture;
1129
1130	/* Get the default device first. */
1131	IMMDevice *pDefaultDevice = NULL;
1132	hrc = pEnumerator->GetDefaultAudioEndpoint(enmType, eMultimedia, &pDefaultDevice);
1133	if (SUCCEEDED(hrc))
1134	rc = drvHostDSoundEnumNewStyleAdd(pDevEnm, pDefaultDevice, enmType, true);
1135	else
1136	pDefaultDevice = NULL;
1137
1138	/* Enumerate the devices. */
1139	IMMDeviceCollection *pCollection = NULL;
1140	hrc = pEnumerator->EnumAudioEndpoints(enmType, DEVICE_STATE_ACTIVE /\| DEVICE_STATE_UNPLUGGED?/, &pCollection);
1141	if (SUCCEEDED(hrc) && pCollection != NULL)
1142	{
1143	UINT cDevices = 0;
1144	hrc = pCollection->GetCount(&cDevices);
1145	if (SUCCEEDED(hrc))
1146	{
1147	for (UINT idxDevice = 0; idxDevice < cDevices && RT_SUCCESS(rc); idxDevice++)
1148	{
1149	IMMDevice *pDevice = NULL;
1150	hrc = pCollection->Item(idxDevice, &pDevice);
1151	if (SUCCEEDED(hrc) && pDevice)
1152	{
1153	if (pDevice != pDefaultDevice)
1154	rc = drvHostDSoundEnumNewStyleAdd(pDevEnm, pDevice, enmType, false);
1155	pDevice->Release();
1156	}
1157	}
1158	}
1159	pCollection->Release();
1160	}
1161	else
1162	LogRelMax(10, ("EnumAudioEndpoints(%s) failed: %Rhrc\n", idxPass == 0 ? "output" : "input", hrc));
1163
1164	if (pDefaultDevice)
1165	pDefaultDevice->Release();
1166	}
1167	pEnumerator->Release();
1168	if (pDevEnm->cDevices > 0 \|\| RT_FAILURE(rc))
1169	{
1170	DSLOG(("DSound: Enumerating devices done - %u device (%Rrc)\n", pDevEnm->cDevices, rc));
1171	return rc;
1172	}
1173	}
1174
1175	/*
1176	* Fall back to dsound.
1177	*/
1178	/* Resolve symbols once. */
1179	static PFNDIRECTSOUNDENUMERATEW volatile s_pfnDirectSoundEnumerateW = NULL;
1180	static PFNDIRECTSOUNDCAPTUREENUMERATEW volatile s_pfnDirectSoundCaptureEnumerateW = NULL;
1181
1182	PFNDIRECTSOUNDENUMERATEW pfnDirectSoundEnumerateW = s_pfnDirectSoundEnumerateW;
1183	PFNDIRECTSOUNDCAPTUREENUMERATEW pfnDirectSoundCaptureEnumerateW = s_pfnDirectSoundCaptureEnumerateW;
1184	if (!pfnDirectSoundEnumerateW \|\| !pfnDirectSoundCaptureEnumerateW)
1185	{
1186	RTLDRMOD hModDSound = NIL_RTLDRMOD;
1187	int rc = RTLdrLoadSystem("dsound.dll", true /fNoUnload/, &hModDSound);
1188	if (RT_SUCCESS(rc))
1189	{
1190	rc = RTLdrGetSymbol(hModDSound, "DirectSoundEnumerateW", (void **)&pfnDirectSoundEnumerateW);
1191	if (RT_SUCCESS(rc))
1192	s_pfnDirectSoundEnumerateW = pfnDirectSoundEnumerateW;
1193	else
1194	LogRel(("DSound: Failed to get dsound.dll export DirectSoundEnumerateW: %Rrc\n", rc));
1195
1196	rc = RTLdrGetSymbol(hModDSound, "DirectSoundCaptureEnumerateW", (void **)&pfnDirectSoundCaptureEnumerateW);
1197	if (RT_SUCCESS(rc))
1198	s_pfnDirectSoundCaptureEnumerateW = pfnDirectSoundCaptureEnumerateW;
1199	else
1200	LogRel(("DSound: Failed to get dsound.dll export DirectSoundCaptureEnumerateW: %Rrc\n", rc));
1201	RTLdrClose(hModDSound);
1202	}
1203	else
1204	LogRel(("DSound: Unable to load dsound.dll for enumerating devices: %Rrc\n", rc));
1205	if (!pfnDirectSoundEnumerateW && !pfnDirectSoundCaptureEnumerateW)
1206	return rc;
1207	}
1208
1209	/* Common callback context for both playback and capture enumerations: */
1210	DSOUNDENUMCBCTX EnumCtx;
1211	EnumCtx.fFlags = 0;
1212	EnumCtx.pDevEnm = pDevEnm;
1213
1214	/* Enumerate playback devices. */
1215	if (pfnDirectSoundEnumerateW)
1216	{
1217	DSLOG(("DSound: Enumerating playback devices ...\n"));
1218	HRESULT hr = pfnDirectSoundEnumerateW(&drvHostDSoundEnumOldStylePlaybackCallback, &EnumCtx);
1219	if (FAILED(hr))
1220	LogRel(("DSound: Error enumerating host playback devices: %Rhrc\n", hr));
1221	}
1222
1223	/* Enumerate capture devices. */
1224	if (pfnDirectSoundCaptureEnumerateW)
1225	{
1226	DSLOG(("DSound: Enumerating capture devices ...\n"));
1227	HRESULT hr = pfnDirectSoundCaptureEnumerateW(&drvHostDSoundEnumOldStyleCaptureCallback, &EnumCtx);
1228	if (FAILED(hr))
1229	LogRel(("DSound: Error enumerating host capture devices: %Rhrc\n", hr));
1230	}
1231
1232	/*
1233	* Query Information for all enumerated devices.
1234	* Note! This is problematic to do from the enumeration callbacks.
1235	*/
1236	PDSOUNDDEV pDev;
1237	RTListForEach(&pDevEnm->LstDevices, pDev, DSOUNDDEV, Core.ListEntry)
1238	{
1239	drvHostDSoundEnumOldStyleQueryDeviceInfo(pDev); /* ignore rc */
1240	}
1241
1242	DSLOG(("DSound: Enumerating devices done\n"));
1243
1244	return VINF_SUCCESS;
1245	}
1246
1247
1248	/**
1249	* @interface_method_impl{PDMIHOSTAUDIO,pfnGetDevices}
1250	*/
1251	static DECLCALLBACK(int) drvHostDSoundHA_GetDevices(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHOSTENUM pDeviceEnum)
1252	{
1253	RT_NOREF(pInterface);
1254	AssertPtrReturn(pDeviceEnum, VERR_INVALID_POINTER);
1255
1256	PDMAudioHostEnumInit(pDeviceEnum);
1257	int rc = drvHostDSoundEnumerateDevices(pDeviceEnum);
1258	if (RT_FAILURE(rc))
1259	PDMAudioHostEnumDelete(pDeviceEnum);
1260
1261	LogFlowFunc(("Returning %Rrc\n", rc));
1262	return rc;
1263	}
1264
1265
1266	/**
1267	* @interface_method_impl{PDMIHOSTAUDIO,pfnGetStatus}
1268	*/
1269	static DECLCALLBACK(PDMAUDIOBACKENDSTS) drvHostDSoundHA_GetStatus(PPDMIHOSTAUDIO pInterface, PDMAUDIODIR enmDir)
1270	{
1271	RT_NOREF(pInterface, enmDir);
1272	return PDMAUDIOBACKENDSTS_RUNNING;
1273	}
1274
1275
1276	/**
1277	* Converts from PDM stream config to windows WAVEFORMATEXTENSIBLE struct.
1278	*
1279	* @param pCfg The PDM audio stream config to convert from.
1280	* @param pFmt The windows structure to initialize.
1281	*/
1282	static void dsoundWaveFmtFromCfg(PCPDMAUDIOSTREAMCFG pCfg, PWAVEFORMATEXTENSIBLE pFmt)
1283	{
1284	RT_ZERO(*pFmt);
1285	pFmt->Format.wFormatTag = WAVE_FORMAT_PCM;
1286	pFmt->Format.nChannels = PDMAudioPropsChannels(&pCfg->Props);
1287	pFmt->Format.wBitsPerSample = PDMAudioPropsSampleBits(&pCfg->Props);
1288	pFmt->Format.nSamplesPerSec = PDMAudioPropsHz(&pCfg->Props);
1289	pFmt->Format.nBlockAlign = PDMAudioPropsFrameSize(&pCfg->Props);
1290	pFmt->Format.nAvgBytesPerSec = PDMAudioPropsFramesToBytes(&pCfg->Props, PDMAudioPropsHz(&pCfg->Props));
1291	pFmt->Format.cbSize = 0; /* No extra data specified. */
1292
1293	/*
1294	* We need to use the extensible structure if there are more than two channels
1295	* or if the channels have non-standard assignments.
1296	*/
1297	if ( pFmt->Format.nChannels > 2
1298	\|\| ( pFmt->Format.nChannels == 1
1299	? pCfg->Props.aidChannels[0] != PDMAUDIOCHANNELID_MONO
1300	: pCfg->Props.aidChannels[0] != PDMAUDIOCHANNELID_FRONT_LEFT
1301	\|\| pCfg->Props.aidChannels[1] != PDMAUDIOCHANNELID_FRONT_RIGHT))
1302	{
1303	pFmt->Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
1304	pFmt->Format.cbSize = sizeof(*pFmt) - sizeof(pFmt->Format);
1305	pFmt->Samples.wValidBitsPerSample = PDMAudioPropsSampleBits(&pCfg->Props);
1306	pFmt->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
1307	pFmt->dwChannelMask = 0;
1308	unsigned const cSrcChannels = pFmt->Format.nChannels;
1309	for (unsigned i = 0; i < cSrcChannels; i++)
1310	if ( pCfg->Props.aidChannels[i] >= PDMAUDIOCHANNELID_FIRST_STANDARD
1311	&& pCfg->Props.aidChannels[i] < PDMAUDIOCHANNELID_END_STANDARD)
1312	pFmt->dwChannelMask \|= RT_BIT_32(pCfg->Props.aidChannels[i] - PDMAUDIOCHANNELID_FIRST_STANDARD);
1313	else
1314	pFmt->Format.nChannels -= 1;
1315	}
1316	}
1317
1318
1319	/**
1320	* Resets the state of a DirectSound stream, clearing the buffer content.
1321	*
1322	* @param pThis Host audio driver instance.
1323	* @param pStreamDS Stream to reset state for.
1324	*/
1325	static void drvHostDSoundStreamReset(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS)
1326	{
1327	RT_NOREF(pThis);
1328	LogFunc(("Resetting %s\n", pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN ? "capture" : "playback"));
1329
1330	if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN)
1331	{
1332	/*
1333	* Input streams.
1334	*/
1335	LogFunc(("Resetting capture stream '%s'\n", pStreamDS->Cfg.szName));
1336
1337	/* Reset the state: */
1338	pStreamDS->msLastTransfer = 0;
1339	/** @todo r=bird: We set the read position to zero here, but shouldn't we query it
1340	* from the buffer instead given that there isn't any interface for repositioning
1341	* to the start of the buffer as with playback buffers? */
1342	pStreamDS->In.offReadPos = 0;
1343	pStreamDS->In.cOverruns = 0;
1344
1345	/* Clear the buffer content: */
1346	AssertPtr(pStreamDS->In.pDSCB);
1347	if (pStreamDS->In.pDSCB)
1348	{
1349	PVOID pv1 = NULL;
1350	DWORD cb1 = 0;
1351	PVOID pv2 = NULL;
1352	DWORD cb2 = 0;
1353	HRESULT hrc = IDirectSoundCaptureBuffer8_Lock(pStreamDS->In.pDSCB, 0, pStreamDS->cbBufSize,
1354	&pv1, &cb1, &pv2, &cb2, 0 /fFlags/);
1355	if (SUCCEEDED(hrc))
1356	{
1357	PDMAudioPropsClearBuffer(&pStreamDS->Cfg.Props, pv1, cb1, PDMAUDIOPCMPROPS_B2F(&pStreamDS->Cfg.Props, cb1));
1358	if (pv2 && cb2)
1359	PDMAudioPropsClearBuffer(&pStreamDS->Cfg.Props, pv2, cb2, PDMAUDIOPCMPROPS_B2F(&pStreamDS->Cfg.Props, cb2));
1360	hrc = IDirectSoundCaptureBuffer8_Unlock(pStreamDS->In.pDSCB, pv1, cb1, pv2, cb2);
1361	if (FAILED(hrc))
1362	LogRelMaxFunc(64, ("DSound: Unlocking capture buffer '%s' after reset failed: %Rhrc\n",
1363	pStreamDS->Cfg.szName, hrc));
1364	}
1365	else
1366	LogRelMaxFunc(64, ("DSound: Locking capture buffer '%s' for reset failed: %Rhrc\n",
1367	pStreamDS->Cfg.szName, hrc));
1368	}
1369	}
1370	else
1371	{
1372	/*
1373	* Output streams.
1374	*/
1375	Assert(pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT);
1376	LogFunc(("Resetting playback stream '%s'\n", pStreamDS->Cfg.szName));
1377
1378	/* If draining was enagaged, make sure dsound has stopped playing: */
1379	if (pStreamDS->Out.fDrain && pStreamDS->Out.pDSB)
1380	pStreamDS->Out.pDSB->Stop();
1381
1382	/* Reset the internal state: */
1383	pStreamDS->msLastTransfer = 0;
1384	pStreamDS->Out.fFirstTransfer = true;
1385	pStreamDS->Out.fDrain = false;
1386	pStreamDS->Out.cbLastTransferred = 0;
1387	pStreamDS->Out.cbTransferred = 0;
1388	pStreamDS->Out.cbWritten = 0;
1389	pStreamDS->Out.offWritePos = 0;
1390	pStreamDS->Out.offPlayCursorLastPending = 0;
1391	pStreamDS->Out.offPlayCursorLastPlayed = 0;
1392
1393	/* Reset the buffer content and repositioning the buffer to the start of the buffer. */
1394	AssertPtr(pStreamDS->Out.pDSB);
1395	if (pStreamDS->Out.pDSB)
1396	{
1397	HRESULT hrc = IDirectSoundBuffer8_SetCurrentPosition(pStreamDS->Out.pDSB, 0);
1398	if (FAILED(hrc))
1399	LogRelMaxFunc(64, ("DSound: Failed to set buffer position for '%s': %Rhrc\n", pStreamDS->Cfg.szName, hrc));
1400
1401	PVOID pv1 = NULL;
1402	DWORD cb1 = 0;
1403	PVOID pv2 = NULL;
1404	DWORD cb2 = 0;
1405	hrc = IDirectSoundBuffer8_Lock(pStreamDS->Out.pDSB, 0, pStreamDS->cbBufSize, &pv1, &cb1, &pv2, &cb2, 0 /fFlags/);
1406	if (hrc == DSERR_BUFFERLOST)
1407	{
1408	directSoundPlayRestore(pThis, pStreamDS->Out.pDSB);
1409	hrc = IDirectSoundBuffer8_Lock(pStreamDS->Out.pDSB, 0, pStreamDS->cbBufSize, &pv1, &cb1, &pv2, &cb2, 0 /fFlags/);
1410	}
1411	if (SUCCEEDED(hrc))
1412	{
1413	PDMAudioPropsClearBuffer(&pStreamDS->Cfg.Props, pv1, cb1, PDMAUDIOPCMPROPS_B2F(&pStreamDS->Cfg.Props, cb1));
1414	if (pv2 && cb2)
1415	PDMAudioPropsClearBuffer(&pStreamDS->Cfg.Props, pv2, cb2, PDMAUDIOPCMPROPS_B2F(&pStreamDS->Cfg.Props, cb2));
1416
1417	hrc = IDirectSoundBuffer8_Unlock(pStreamDS->Out.pDSB, pv1, cb1, pv2, cb2);
1418	if (FAILED(hrc))
1419	LogRelMaxFunc(64, ("DSound: Unlocking playback buffer '%s' after reset failed: %Rhrc\n",
1420	pStreamDS->Cfg.szName, hrc));
1421	}
1422	else
1423	LogRelMaxFunc(64, ("DSound: Locking playback buffer '%s' for reset failed: %Rhrc\n", pStreamDS->Cfg.szName, hrc));
1424	}
1425	}
1426	}
1427
1428
1429	/**
1430	* Worker for drvHostDSoundHA_StreamCreate that creates caputre stream.
1431	*
1432	* @returns Windows COM status code.
1433	* @param pThis The DSound instance data.
1434	* @param pStreamDS The stream instance data.
1435	* @param pCfgReq The requested stream config (input).
1436	* @param pCfgAcq Where to return the actual stream config. This is a
1437	* copy of @a *pCfgReq when called.
1438	* @param pWaveFmtExt On input the requested stream format. Updated to the
1439	* actual stream format on successful return.
1440	*/
1441	static HRESULT drvHostDSoundStreamCreateCapture(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, PCPDMAUDIOSTREAMCFG pCfgReq,
1442	PPDMAUDIOSTREAMCFG pCfgAcq, WAVEFORMATEXTENSIBLE *pWaveFmtExt)
1443	{
1444	Assert(pStreamDS->In.pDSCB == NULL);
1445	HRESULT hrc;
1446
1447	/*
1448	* Create, initialize and set up a IDirectSoundCapture instance the first time
1449	* we go thru here.
1450	*/
1451	/** @todo bird: Or should we rather just throw this away after we've gotten the
1452	* capture buffer? Old code would just leak it... */
1453	if (pThis->pDSC == NULL)
1454	{
1455	hrc = drvHostDSoundCreateDSCaptureInstance(pThis->Cfg.pGuidCapture, &pThis->pDSC);
1456	if (FAILED(hrc))
1457	return hrc; /* The worker has complained to the release log already. */
1458	}
1459
1460	/*
1461	* Create the capture buffer.
1462	*/
1463	DSCBUFFERDESC BufferDesc =
1464	{
1465	/.dwSize = / sizeof(BufferDesc),
1466	/.dwFlags = / 0,
1467	/.dwBufferBytes =/ PDMAudioPropsFramesToBytes(&pCfgReq->Props, pCfgReq->Backend.cFramesBufferSize),
1468	/.dwReserved = / 0,
1469	/.lpwfxFormat = / &pWaveFmtExt->Format,
1470	/.dwFXCount = / 0,
1471	/.lpDSCFXDesc = / NULL
1472	};
1473
1474	LogRel2(("DSound: Requested capture buffer is %#x B / %u B / %RU64 ms\n", BufferDesc.dwBufferBytes, BufferDesc.dwBufferBytes,
1475	PDMAudioPropsBytesToMilli(&pCfgReq->Props, BufferDesc.dwBufferBytes)));
1476
1477	LPDIRECTSOUNDCAPTUREBUFFER pLegacyDSCB = NULL;
1478	hrc = IDirectSoundCapture_CreateCaptureBuffer(pThis->pDSC, &BufferDesc, &pLegacyDSCB, NULL);
1479	if (FAILED(hrc))
1480	{
1481	LogRelMax(64, ("DSound: Creating capture buffer for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc));
1482	return hrc;
1483	}
1484
1485	/* Get the IDirectSoundCaptureBuffer8 version of the interface. */
1486	hrc = IDirectSoundCaptureBuffer_QueryInterface(pLegacyDSCB, IID_IDirectSoundCaptureBuffer8, (void **)&pStreamDS->In.pDSCB);
1487	IDirectSoundCaptureBuffer_Release(pLegacyDSCB);
1488	if (FAILED(hrc))
1489	{
1490	LogRelMax(64, ("DSound: Querying IID_IDirectSoundCaptureBuffer8 for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc));
1491	return hrc;
1492	}
1493
1494	/*
1495	* Query the actual stream configuration.
1496	*/
1497	#if 0 /** @todo r=bird: WTF was this for? */
1498	DWORD offByteReadPos = 0;
1499	hrc = IDirectSoundCaptureBuffer8_GetCurrentPosition(pStreamDS->In.pDSCB, NULL, &offByteReadPos);
1500	if (FAILED(hrc))
1501	{
1502	offByteReadPos = 0;
1503	DSLOGREL(("DSound: Getting capture position failed with %Rhrc\n", hr));
1504	}
1505	#endif
1506	RT_ZERO(*pWaveFmtExt);
1507	hrc = IDirectSoundCaptureBuffer8_GetFormat(pStreamDS->In.pDSCB, &pWaveFmtExt->Format, sizeof(*pWaveFmtExt), NULL);
1508	if (SUCCEEDED(hrc))
1509	{
1510	/** @todo r=bird: We aren't converting/checking the pWaveFmtX content... */
1511
1512	DSCBCAPS BufferCaps = { /.dwSize = / sizeof(BufferCaps), 0, 0, 0 };
1513	hrc = IDirectSoundCaptureBuffer8_GetCaps(pStreamDS->In.pDSCB, &BufferCaps);
1514	if (SUCCEEDED(hrc))
1515	{
1516	LogRel2(("DSound: Acquired capture buffer capabilities for '%s':\n"
1517	"DSound: dwFlags = %#RX32\n"
1518	"DSound: dwBufferBytes = %#RX32 B / %RU32 B / %RU64 ms\n"
1519	"DSound: dwReserved = %#RX32\n",
1520	pCfgReq->szName, BufferCaps.dwFlags, BufferCaps.dwBufferBytes, BufferCaps.dwBufferBytes,
1521	PDMAudioPropsBytesToMilli(&pCfgReq->Props, BufferCaps.dwBufferBytes), BufferCaps.dwReserved ));
1522
1523	/* Update buffer related stuff: */
1524	pStreamDS->In.offReadPos = 0; /** @todo shouldn't we use offBytReadPos here to "read at the initial capture position"? */
1525	pStreamDS->cbBufSize = BufferCaps.dwBufferBytes;
1526	pCfgAcq->Backend.cFramesBufferSize = PDMAudioPropsBytesToFrames(&pCfgAcq->Props, BufferCaps.dwBufferBytes);
1527
1528	#if 0 /** @todo r=bird: uAlign isn't set anywhere, so this hasn't been checking anything for a while... */
1529	if (bc.dwBufferBytes & pStreamDS->uAlign)
1530	DSLOGREL(("DSound: Capture GetCaps returned misaligned buffer: size %RU32, alignment %RU32\n",
1531	bc.dwBufferBytes, pStreamDS->uAlign + 1));
1532	#endif
1533	LogFlow(("returns S_OK\n"));
1534	return S_OK;
1535	}
1536	LogRelMax(64, ("DSound: Getting capture buffer capabilities for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc));
1537	}
1538	else
1539	LogRelMax(64, ("DSound: Getting capture format for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc));
1540
1541	IDirectSoundCaptureBuffer8_Release(pStreamDS->In.pDSCB);
1542	pStreamDS->In.pDSCB = NULL;
1543	LogFlowFunc(("returns %Rhrc\n", hrc));
1544	return hrc;
1545	}
1546
1547
1548	/**
1549	* Worker for drvHostDSoundHA_StreamCreate that creates playback stream.
1550	*
1551	* @returns Windows COM status code.
1552	* @param pThis The DSound instance data.
1553	* @param pStreamDS The stream instance data.
1554	* @param pCfgReq The requested stream config (input).
1555	* @param pCfgAcq Where to return the actual stream config. This is a
1556	* copy of @a *pCfgReq when called.
1557	* @param pWaveFmtExt On input the requested stream format.
1558	* Updated to the actual stream format on successful
1559	* return.
1560	*/
1561	static HRESULT drvHostDSoundStreamCreatePlayback(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, PCPDMAUDIOSTREAMCFG pCfgReq,
1562	PPDMAUDIOSTREAMCFG pCfgAcq, WAVEFORMATEXTENSIBLE *pWaveFmtExt)
1563	{
1564	Assert(pStreamDS->Out.pDSB == NULL);
1565	HRESULT hrc;
1566
1567	/*
1568	* Create, initialize and set up a DirectSound8 instance the first time
1569	* we go thru here.
1570	*/
1571	/** @todo bird: Or should we rather just throw this away after we've gotten the
1572	* sound buffer? Old code would just leak it... */
1573	if (pThis->pDS == NULL)
1574	{
1575	hrc = drvHostDSoundCreateDSPlaybackInstance(pThis->Cfg.pGuidPlay, &pThis->pDS);
1576	if (FAILED(hrc))
1577	return hrc; /* The worker has complained to the release log already. */
1578	}
1579
1580	/*
1581	* As we reuse our (secondary) buffer for playing out data as it comes in,
1582	* we're using this buffer as a so-called streaming buffer.
1583	*
1584	* See https://msdn.microsoft.com/en-us/library/windows/desktop/ee419014(v=vs.85).aspx
1585	*
1586	* However, as we do not want to use memory on the sound device directly
1587	* (as most modern audio hardware on the host doesn't have this anyway),
1588	* we're not going to use DSBCAPS_STATIC for that.
1589	*
1590	* Instead we're specifying DSBCAPS_LOCSOFTWARE, as this fits the bill
1591	* of copying own buffer data to our secondary's Direct Sound buffer.
1592	*/
1593	DSBUFFERDESC BufferDesc =
1594	{
1595	/.dwSize = / sizeof(BufferDesc),
1596	/.dwFlags = / DSBCAPS_GLOBALFOCUS \| DSBCAPS_GETCURRENTPOSITION2 \| DSBCAPS_LOCSOFTWARE,
1597	/.dwBufferBytes = / PDMAudioPropsFramesToBytes(&pCfgReq->Props, pCfgReq->Backend.cFramesBufferSize),
1598	/.dwReserved = / 0,
1599	/.lpwfxFormat = / &pWaveFmtExt->Format,
1600	/.guid3DAlgorithm = {0, 0, 0, {0,0,0,0, 0,0,0,0}} /
1601	};
1602	LogRel2(("DSound: Requested playback buffer is %#x B / %u B / %RU64 ms\n", BufferDesc.dwBufferBytes, BufferDesc.dwBufferBytes,
1603	PDMAudioPropsBytesToMilli(&pCfgReq->Props, BufferDesc.dwBufferBytes)));
1604
1605	LPDIRECTSOUNDBUFFER pLegacyDSB = NULL;
1606	hrc = IDirectSound8_CreateSoundBuffer(pThis->pDS, &BufferDesc, &pLegacyDSB, NULL);
1607	if (FAILED(hrc))
1608	{
1609	LogRelMax(64, ("DSound: Creating playback sound buffer for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc));
1610	return hrc;
1611	}
1612
1613	/* Get the IDirectSoundBuffer8 version of the interface. */
1614	hrc = IDirectSoundBuffer_QueryInterface(pLegacyDSB, IID_IDirectSoundBuffer8, (PVOID *)&pStreamDS->Out.pDSB);
1615	IDirectSoundBuffer_Release(pLegacyDSB);
1616	if (FAILED(hrc))
1617	{
1618	LogRelMax(64, ("DSound: Querying IID_IDirectSoundBuffer8 for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc));
1619	return hrc;
1620	}
1621
1622	/*
1623	* Query the actual stream parameters, they may differ from what we requested.
1624	*/
1625	RT_ZERO(*pWaveFmtExt);
1626	hrc = IDirectSoundBuffer8_GetFormat(pStreamDS->Out.pDSB, &pWaveFmtExt->Format, sizeof(*pWaveFmtExt), NULL);
1627	if (SUCCEEDED(hrc))
1628	{
1629	/** @todo r=bird: We aren't converting/checking the pWaveFmtX content... */
1630
1631	DSBCAPS BufferCaps = { /.dwSize = / sizeof(BufferCaps), 0, 0, 0, 0 };
1632	hrc = IDirectSoundBuffer8_GetCaps(pStreamDS->Out.pDSB, &BufferCaps);
1633	if (SUCCEEDED(hrc))
1634	{
1635	LogRel2(("DSound: Acquired playback buffer capabilities for '%s':\n"
1636	"DSound: dwFlags = %#RX32\n"
1637	"DSound: dwBufferBytes = %#RX32 B / %RU32 B / %RU64 ms\n"
1638	"DSound: dwUnlockTransferRate = %RU32 KB/s\n"
1639	"DSound: dwPlayCpuOverhead = %RU32%%\n",
1640	pCfgReq->szName, BufferCaps.dwFlags, BufferCaps.dwBufferBytes, BufferCaps.dwBufferBytes,
1641	PDMAudioPropsBytesToMilli(&pCfgReq->Props, BufferCaps.dwBufferBytes),
1642	BufferCaps.dwUnlockTransferRate, BufferCaps.dwPlayCpuOverhead));
1643
1644	/* Update buffer related stuff: */
1645	pStreamDS->cbBufSize = BufferCaps.dwBufferBytes;
1646	pCfgAcq->Backend.cFramesBufferSize = PDMAudioPropsBytesToFrames(&pCfgAcq->Props, BufferCaps.dwBufferBytes);
1647	pCfgAcq->Backend.cFramesPeriod = pCfgAcq->Backend.cFramesBufferSize / 4; /* total fiction */
1648	pCfgAcq->Backend.cFramesPreBuffering = pCfgReq->Backend.cFramesPreBuffering * pCfgAcq->Backend.cFramesBufferSize
1649	/ RT_MAX(pCfgReq->Backend.cFramesBufferSize, 1);
1650
1651	#if 0 /** @todo r=bird: uAlign isn't set anywhere, so this hasn't been checking anything for a while... */
1652	if (bc.dwBufferBytes & pStreamDS->uAlign)
1653	DSLOGREL(("DSound: Playback capabilities returned misaligned buffer: size %RU32, alignment %RU32\n",
1654	bc.dwBufferBytes, pStreamDS->uAlign + 1));
1655	#endif
1656	LogFlow(("returns S_OK\n"));
1657	return S_OK;
1658	}
1659	LogRelMax(64, ("DSound: Getting playback buffer capabilities for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc));
1660	}
1661	else
1662	LogRelMax(64, ("DSound: Getting playback format for '%s' failed: %Rhrc\n", pCfgReq->szName, hrc));
1663
1664	IDirectSoundBuffer8_Release(pStreamDS->Out.pDSB);
1665	pStreamDS->Out.pDSB = NULL;
1666	LogFlowFunc(("returns %Rhrc\n", hrc));
1667	return hrc;
1668	}
1669
1670
1671	/**
1672	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamCreate}
1673	*/
1674	static DECLCALLBACK(int) drvHostDSoundHA_StreamCreate(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream,
1675	PCPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq)
1676	{
1677	PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio);
1678	PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream;
1679	AssertPtrReturn(pStreamDS, VERR_INVALID_POINTER);
1680	AssertPtrReturn(pCfgReq, VERR_INVALID_POINTER);
1681	AssertPtrReturn(pCfgAcq, VERR_INVALID_POINTER);
1682	AssertReturn(pCfgReq->enmDir == PDMAUDIODIR_IN \|\| pCfgReq->enmDir == PDMAUDIODIR_OUT, VERR_INVALID_PARAMETER);
1683	Assert(PDMAudioStrmCfgEquals(pCfgReq, pCfgAcq));
1684
1685	const char * const pszStreamType = pCfgReq->enmDir == PDMAUDIODIR_IN ? "capture" : "playback"; RT_NOREF(pszStreamType);
1686	LogFlowFunc(("enmPath=%s '%s'\n", PDMAudioPathGetName(pCfgReq->enmPath), pCfgReq->szName));
1687	RTListInit(&pStreamDS->ListEntry); /* paranoia */
1688
1689	/* For whatever reason: */
1690	dsoundUpdateStatusInternal(pThis);
1691
1692	/*
1693	* DSound has different COM interfaces for working with input and output
1694	* streams, so we'll quickly part ways here after some common format
1695	* specification setup and logging.
1696	*/
1697	#if defined(RTLOG_REL_ENABLED) \|\| defined(LOG_ENABLED)
1698	char szTmp[64];
1699	#endif
1700	LogRel2(("DSound: Opening %s stream '%s' (%s)\n", pCfgReq->szName, pszStreamType,
1701	PDMAudioPropsToString(&pCfgReq->Props, szTmp, sizeof(szTmp))));
1702
1703	WAVEFORMATEXTENSIBLE WaveFmtExt;
1704	dsoundWaveFmtFromCfg(pCfgReq, &WaveFmtExt);
1705	LogRel2(("DSound: Requested %s format for '%s':\n"
1706	"DSound: wFormatTag = %RU16\n"
1707	"DSound: nChannels = %RU16\n"
1708	"DSound: nSamplesPerSec = %RU32\n"
1709	"DSound: nAvgBytesPerSec = %RU32\n"
1710	"DSound: nBlockAlign = %RU16\n"
1711	"DSound: wBitsPerSample = %RU16\n"
1712	"DSound: cbSize = %RU16\n",
1713	pszStreamType, pCfgReq->szName, WaveFmtExt.Format.wFormatTag, WaveFmtExt.Format.nChannels,
1714	WaveFmtExt.Format.nSamplesPerSec, WaveFmtExt.Format.nAvgBytesPerSec, WaveFmtExt.Format.nBlockAlign,
1715	WaveFmtExt.Format.wBitsPerSample, WaveFmtExt.Format.cbSize));
1716	if (WaveFmtExt.Format.cbSize != 0)
1717	LogRel2(("DSound: dwChannelMask = %#RX32\n"
1718	"DSound: wValidBitsPerSample = %RU16\n",
1719	WaveFmtExt.dwChannelMask, WaveFmtExt.Samples.wValidBitsPerSample));
1720
1721	HRESULT hrc;
1722	if (pCfgReq->enmDir == PDMAUDIODIR_IN)
1723	hrc = drvHostDSoundStreamCreateCapture(pThis, pStreamDS, pCfgReq, pCfgAcq, &WaveFmtExt);
1724	else
1725	hrc = drvHostDSoundStreamCreatePlayback(pThis, pStreamDS, pCfgReq, pCfgAcq, &WaveFmtExt);
1726	int rc;
1727	if (SUCCEEDED(hrc))
1728	{
1729	LogRel2(("DSound: Acquired %s format for '%s':\n"
1730	"DSound: wFormatTag = %RU16\n"
1731	"DSound: nChannels = %RU16\n"
1732	"DSound: nSamplesPerSec = %RU32\n"
1733	"DSound: nAvgBytesPerSec = %RU32\n"
1734	"DSound: nBlockAlign = %RU16\n"
1735	"DSound: wBitsPerSample = %RU16\n"
1736	"DSound: cbSize = %RU16\n",
1737	pszStreamType, pCfgReq->szName, WaveFmtExt.Format.wFormatTag, WaveFmtExt.Format.nChannels,
1738	WaveFmtExt.Format.nSamplesPerSec, WaveFmtExt.Format.nAvgBytesPerSec, WaveFmtExt.Format.nBlockAlign,
1739	WaveFmtExt.Format.wBitsPerSample, WaveFmtExt.Format.cbSize));
1740	if (WaveFmtExt.Format.cbSize != 0)
1741	{
1742	LogRel2(("DSound: dwChannelMask = %#RX32\n"
1743	"DSound: wValidBitsPerSample = %RU16\n",
1744	WaveFmtExt.dwChannelMask, WaveFmtExt.Samples.wValidBitsPerSample));
1745
1746	/* Update the channel count and map here. */
1747	PDMAudioPropsSetChannels(&pCfgAcq->Props, WaveFmtExt.Format.nChannels);
1748	uint8_t idCh = 0;
1749	for (unsigned iBit = 0; iBit < 32 && idCh < WaveFmtExt.Format.nChannels; iBit++)
1750	if (WaveFmtExt.dwChannelMask & RT_BIT_32(iBit))
1751	{
1752	pCfgAcq->Props.aidChannels[idCh] = (unsigned)PDMAUDIOCHANNELID_FIRST_STANDARD + iBit;
1753	idCh++;
1754	}
1755	Assert(idCh == WaveFmtExt.Format.nChannels);
1756	}
1757
1758	/*
1759	* Copy the acquired config and reset the stream (clears the buffer).
1760	*/
1761	PDMAudioStrmCfgCopy(&pStreamDS->Cfg, pCfgAcq);
1762	drvHostDSoundStreamReset(pThis, pStreamDS);
1763
1764	RTCritSectEnter(&pThis->CritSect);
1765	RTListAppend(&pThis->HeadStreams, &pStreamDS->ListEntry);
1766	RTCritSectLeave(&pThis->CritSect);
1767
1768	rc = VINF_SUCCESS;
1769	}
1770	else
1771	rc = VERR_AUDIO_STREAM_COULD_NOT_CREATE;
1772
1773	LogFlowFunc(("returns %Rrc\n", rc));
1774	return rc;
1775	}
1776
1777
1778	/**
1779	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamDestroy}
1780	*/
1781	static DECLCALLBACK(int) drvHostDSoundHA_StreamDestroy(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream,
1782	bool fImmediate)
1783	{
1784	PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio);
1785	PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream;
1786	AssertPtrReturn(pStreamDS, VERR_INVALID_POINTER);
1787	LogFlowFunc(("Stream '%s' {%s}\n", pStreamDS->Cfg.szName, drvHostDSoundStreamStatusString(pStreamDS)));
1788	RT_NOREF(fImmediate);
1789
1790	RTCritSectEnter(&pThis->CritSect);
1791	RTListNodeRemove(&pStreamDS->ListEntry);
1792	RTCritSectLeave(&pThis->CritSect);
1793
1794	if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN)
1795	{
1796	/*
1797	* Input.
1798	*/
1799	if (pStreamDS->In.pDSCB)
1800	{
1801	HRESULT hrc = IDirectSoundCaptureBuffer_Stop(pStreamDS->In.pDSCB);
1802	if (FAILED(hrc))
1803	LogFunc(("IDirectSoundCaptureBuffer_Stop failed: %Rhrc\n", hrc));
1804
1805	drvHostDSoundStreamReset(pThis, pStreamDS);
1806
1807	IDirectSoundCaptureBuffer8_Release(pStreamDS->In.pDSCB);
1808	pStreamDS->In.pDSCB = NULL;
1809	}
1810	}
1811	else
1812	{
1813	/*
1814	* Output.
1815	*/
1816	if (pStreamDS->Out.pDSB)
1817	{
1818	drvHostDSoundStreamStopPlayback(pThis, pStreamDS, true /fReset/);
1819
1820	IDirectSoundBuffer8_Release(pStreamDS->Out.pDSB);
1821	pStreamDS->Out.pDSB = NULL;
1822	}
1823	}
1824
1825	if (RTCritSectIsInitialized(&pStreamDS->CritSect))
1826	RTCritSectDelete(&pStreamDS->CritSect);
1827
1828	return VINF_SUCCESS;
1829	}
1830
1831
1832	/**
1833	* Worker for drvHostDSoundHA_StreamEnable and drvHostDSoundHA_StreamResume.
1834	*
1835	* This will try re-open the capture device if we're having trouble starting it.
1836	*
1837	* @returns VBox status code.
1838	* @param pThis The DSound host audio driver instance data.
1839	* @param pStreamDS The stream instance data.
1840	*/
1841	static int drvHostDSoundStreamCaptureStart(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS)
1842	{
1843	/*
1844	* Check the stream status first.
1845	*/
1846	int rc = VERR_AUDIO_STREAM_NOT_READY;
1847	if (pStreamDS->In.pDSCB)
1848	{
1849	DWORD fStatus = 0;
1850	HRESULT hrc = IDirectSoundCaptureBuffer8_GetStatus(pStreamDS->In.pDSCB, &fStatus);
1851	if (SUCCEEDED(hrc))
1852	{
1853	/*
1854	* Try start capturing if it's not already doing so.
1855	*/
1856	if (!(fStatus & DSCBSTATUS_CAPTURING))
1857	{
1858	LogRel2(("DSound: Starting capture on '%s' ... \n", pStreamDS->Cfg.szName));
1859	hrc = IDirectSoundCaptureBuffer8_Start(pStreamDS->In.pDSCB, DSCBSTART_LOOPING);
1860	if (SUCCEEDED(hrc))
1861	rc = VINF_SUCCESS;
1862	else
1863	{
1864	/*
1865	* Failed to start, try re-create the capture buffer.
1866	*/
1867	LogRelMax(64, ("DSound: Starting to capture on '%s' failed: %Rhrc - will try re-open it ...\n",
1868	pStreamDS->Cfg.szName, hrc));
1869
1870	IDirectSoundCaptureBuffer8_Release(pStreamDS->In.pDSCB);
1871	pStreamDS->In.pDSCB = NULL;
1872
1873	PDMAUDIOSTREAMCFG CfgReq = pStreamDS->Cfg;
1874	PDMAUDIOSTREAMCFG CfgAcq = pStreamDS->Cfg;
1875	WAVEFORMATEXTENSIBLE WaveFmtExt;
1876	dsoundWaveFmtFromCfg(&pStreamDS->Cfg, &WaveFmtExt);
1877	hrc = drvHostDSoundStreamCreateCapture(pThis, pStreamDS, &CfgReq, &CfgAcq, &WaveFmtExt);
1878	if (SUCCEEDED(hrc))
1879	{
1880	PDMAudioStrmCfgCopy(&pStreamDS->Cfg, &CfgAcq);
1881
1882	/*
1883	* Try starting capture again.
1884	*/
1885	LogRel2(("DSound: Starting capture on re-opened '%s' ... \n", pStreamDS->Cfg.szName));
1886	hrc = IDirectSoundCaptureBuffer8_Start(pStreamDS->In.pDSCB, DSCBSTART_LOOPING);
1887	if (SUCCEEDED(hrc))
1888	rc = VINF_SUCCESS;
1889	else
1890	LogRelMax(64, ("DSound: Starting to capture on re-opened '%s' failed: %Rhrc\n",
1891	pStreamDS->Cfg.szName, hrc));
1892	}
1893	else
1894	LogRelMax(64, ("DSound: Re-opening '%s' failed: %Rhrc\n", pStreamDS->Cfg.szName, hrc));
1895	}
1896	}
1897	else
1898	{
1899	LogRel2(("DSound: Already capturing (%#x)\n", fStatus));
1900	AssertFailed();
1901	}
1902	}
1903	else
1904	LogRelMax(64, ("DSound: Retrieving capture status for '%s' failed: %Rhrc\n", pStreamDS->Cfg.szName, hrc));
1905	}
1906	LogFlowFunc(("returns %Rrc\n", rc));
1907	return rc;
1908	}
1909
1910
1911	/**
1912	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamEnable}
1913	*/
1914	static DECLCALLBACK(int) drvHostDSoundHA_StreamEnable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
1915	{
1916	PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio);
1917	PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream;
1918	LogFlowFunc(("Stream '%s' {%s}\n", pStreamDS->Cfg.szName, drvHostDSoundStreamStatusString(pStreamDS)));
1919
1920	/*
1921	* We always reset the buffer before enabling the stream (normally never necessary).
1922	*/
1923	drvHostDSoundStreamReset(pThis, pStreamDS);
1924	pStreamDS->fEnabled = true;
1925
1926	/*
1927	* Input streams will start capturing, while output streams will only start
1928	* playing once we get some audio data to play.
1929	*/
1930	int rc = VINF_SUCCESS;
1931	if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN)
1932	rc = drvHostDSoundStreamCaptureStart(pThis, pStreamDS);
1933	else
1934	Assert(pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT);
1935
1936	LogFlowFunc(("returns %Rrc\n", rc));
1937	return rc;
1938	}
1939
1940
1941	/**
1942	* Worker for drvHostDSoundHA_StreamDestroy, drvHostDSoundHA_StreamDisable and
1943	* drvHostDSoundHA_StreamPause.
1944	*
1945	* @returns VBox status code.
1946	* @param pThis The DSound host audio driver instance data.
1947	* @param pStreamDS The stream instance data.
1948	* @param fReset Whether to reset the buffer and state.
1949	*/
1950	static int drvHostDSoundStreamStopPlayback(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, bool fReset)
1951	{
1952	if (!pStreamDS->Out.pDSB)
1953	return VINF_SUCCESS;
1954
1955	LogRel2(("DSound: Stopping playback of '%s'...\n", pStreamDS->Cfg.szName));
1956	HRESULT hrc = IDirectSoundBuffer8_Stop(pStreamDS->Out.pDSB);
1957	if (FAILED(hrc))
1958	{
1959	LogFunc(("IDirectSoundBuffer8_Stop -> %Rhrc; will attempt restoring the stream...\n", hrc));
1960	directSoundPlayRestore(pThis, pStreamDS->Out.pDSB);
1961	hrc = IDirectSoundBuffer8_Stop(pStreamDS->Out.pDSB);
1962	if (FAILED(hrc))
1963	LogRelMax(64, ("DSound: %s playback of '%s' failed: %Rhrc\n", fReset ? "Stopping" : "Pausing",
1964	pStreamDS->Cfg.szName, hrc));
1965	}
1966	LogRel2(("DSound: Stopped playback of '%s': %Rhrc\n", pStreamDS->Cfg.szName, hrc));
1967
1968	if (fReset)
1969	drvHostDSoundStreamReset(pThis, pStreamDS);
1970	return SUCCEEDED(hrc) ? VINF_SUCCESS : VERR_AUDIO_STREAM_NOT_READY;
1971	}
1972
1973
1974	/**
1975	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamDisable}
1976	*/
1977	static DECLCALLBACK(int) drvHostDSoundHA_StreamDisable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
1978	{
1979	PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio);
1980	PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream;
1981	LogFlowFunc(("cMsLastTransfer=%RI64 ms, stream '%s' {%s} \n",
1982	pStreamDS->msLastTransfer ? RTTimeMilliTS() - pStreamDS->msLastTransfer : -1,
1983	pStreamDS->Cfg.szName, drvHostDSoundStreamStatusString(pStreamDS) ));
1984
1985	/*
1986	* Change the state.
1987	*/
1988	pStreamDS->fEnabled = false;
1989
1990	/*
1991	* Stop the stream and maybe reset the buffer.
1992	*/
1993	int rc = VINF_SUCCESS;
1994	if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN)
1995	{
1996	if (pStreamDS->In.pDSCB)
1997	{
1998	HRESULT hrc = IDirectSoundCaptureBuffer_Stop(pStreamDS->In.pDSCB);
1999	if (SUCCEEDED(hrc))
2000	LogRel3(("DSound: Stopped capture on '%s'.\n", pStreamDS->Cfg.szName));
2001	else
2002	{
2003	LogRelMax(64, ("DSound: Stopping capture on '%s' failed: %Rhrc\n", pStreamDS->Cfg.szName, hrc));
2004	/* Don't report errors up to the caller, as it might just be a capture device change. */
2005	}
2006
2007	/* This isn't strictly speaking necessary since StreamEnable does it too... */
2008	drvHostDSoundStreamReset(pThis, pStreamDS);
2009	}
2010	}
2011	else
2012	{
2013	Assert(pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT);
2014	if (pStreamDS->Out.pDSB)
2015	{
2016	rc = drvHostDSoundStreamStopPlayback(pThis, pStreamDS, true /fReset/);
2017	if (RT_SUCCESS(rc))
2018	LogRel3(("DSound: Stopped playback on '%s'.\n", pStreamDS->Cfg.szName));
2019	}
2020	}
2021
2022	LogFlowFunc(("returns %Rrc {%s}\n", rc, drvHostDSoundStreamStatusString(pStreamDS)));
2023	return rc;
2024	}
2025
2026
2027	/**
2028	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamPause}
2029	*
2030	* @note Basically the same as drvHostDSoundHA_StreamDisable, just w/o the
2031	* buffer resetting and fEnabled change.
2032	*/
2033	static DECLCALLBACK(int) drvHostDSoundHA_StreamPause(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
2034	{
2035	PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio);
2036	PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream;
2037	LogFlowFunc(("cMsLastTransfer=%RI64 ms, stream '%s' {%s} \n",
2038	pStreamDS->msLastTransfer ? RTTimeMilliTS() - pStreamDS->msLastTransfer : -1,
2039	pStreamDS->Cfg.szName, drvHostDSoundStreamStatusString(pStreamDS) ));
2040
2041	/*
2042	* Stop the stream and maybe reset the buffer.
2043	*/
2044	int rc = VINF_SUCCESS;
2045	if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN)
2046	{
2047	if (pStreamDS->In.pDSCB)
2048	{
2049	HRESULT hrc = IDirectSoundCaptureBuffer_Stop(pStreamDS->In.pDSCB);
2050	if (SUCCEEDED(hrc))
2051	LogRel3(("DSound: Stopped capture on '%s'.\n", pStreamDS->Cfg.szName));
2052	else
2053	{
2054	LogRelMax(64, ("DSound: Stopping capture on '%s' failed: %Rhrc\n", pStreamDS->Cfg.szName, hrc));
2055	/* Don't report errors up to the caller, as it might just be a capture device change. */
2056	}
2057	}
2058	}
2059	else
2060	{
2061	Assert(pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT);
2062	if (pStreamDS->Out.pDSB)
2063	{
2064	/* Don't stop draining buffers, we won't be resuming them right.
2065	They'll stop by themselves anyway. */
2066	if (pStreamDS->Out.fDrain)
2067	LogFunc(("Stream '%s' is draining\n", pStreamDS->Cfg.szName));
2068	else
2069	{
2070	rc = drvHostDSoundStreamStopPlayback(pThis, pStreamDS, false /fReset/);
2071	if (RT_SUCCESS(rc))
2072	LogRel3(("DSound: Stopped playback on '%s'.\n", pStreamDS->Cfg.szName));
2073	}
2074	}
2075	}
2076
2077	LogFlowFunc(("returns %Rrc {%s}\n", rc, drvHostDSoundStreamStatusString(pStreamDS)));
2078	return rc;
2079	}
2080
2081
2082	/**
2083	* Worker for drvHostDSoundHA_StreamResume and drvHostDSoundHA_StreamPlay that
2084	* starts playing the DirectSound Buffer.
2085	*
2086	* @returns VBox status code.
2087	* @param pThis Host audio driver instance.
2088	* @param pStreamDS Stream to start playing.
2089	*/
2090	static int directSoundPlayStart(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS)
2091	{
2092	if (!pStreamDS->Out.pDSB)
2093	return VERR_AUDIO_STREAM_NOT_READY;
2094
2095	LogRel2(("DSound: Starting playback of '%s' ...\n", pStreamDS->Cfg.szName));
2096	HRESULT hrc = IDirectSoundBuffer8_Play(pStreamDS->Out.pDSB, 0, 0, DSCBSTART_LOOPING);
2097	if (SUCCEEDED(hrc))
2098	return VINF_SUCCESS;
2099
2100	for (unsigned i = 0; hrc == DSERR_BUFFERLOST && i < DRV_DSOUND_RESTORE_ATTEMPTS_MAX; i++)
2101	{
2102	LogFunc(("Restarting playback failed due to lost buffer, restoring ...\n"));
2103	directSoundPlayRestore(pThis, pStreamDS->Out.pDSB);
2104
2105	hrc = IDirectSoundBuffer8_Play(pStreamDS->Out.pDSB, 0, 0, DSCBSTART_LOOPING);
2106	if (SUCCEEDED(hrc))
2107	return VINF_SUCCESS;
2108	}
2109
2110	LogRelMax(64, ("DSound: Failed to start playback of '%s': %Rhrc\n", pStreamDS->Cfg.szName, hrc));
2111	return VERR_AUDIO_STREAM_NOT_READY;
2112	}
2113
2114
2115	/**
2116	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamResume}
2117	*/
2118	static DECLCALLBACK(int) drvHostDSoundHA_StreamResume(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
2119	{
2120	PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio);
2121	PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream;
2122	LogFlowFunc(("Stream '%s' {%s}\n", pStreamDS->Cfg.szName, drvHostDSoundStreamStatusString(pStreamDS)));
2123
2124	/*
2125	* Input streams will start capturing, while output streams will only start
2126	* playing if we're past the pre-buffering state.
2127	*/
2128	int rc = VINF_SUCCESS;
2129	if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN)
2130	rc = drvHostDSoundStreamCaptureStart(pThis, pStreamDS);
2131	else
2132	{
2133	Assert(pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT);
2134	if (!pStreamDS->Out.fFirstTransfer)
2135	rc = directSoundPlayStart(pThis, pStreamDS);
2136	}
2137
2138	LogFlowFunc(("returns %Rrc {%s}\n", rc, drvHostDSoundStreamStatusString(pStreamDS)));
2139	return rc;
2140	}
2141
2142
2143	/**
2144	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamDrain}
2145	*/
2146	static DECLCALLBACK(int) drvHostDSoundHA_StreamDrain(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
2147	{
2148	PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio);
2149	PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream;
2150	AssertReturn(pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT, VERR_INVALID_PARAMETER);
2151	LogFlowFunc(("cMsLastTransfer=%RI64 ms, stream '%s' {%s} \n",
2152	pStreamDS->msLastTransfer ? RTTimeMilliTS() - pStreamDS->msLastTransfer : -1,
2153	pStreamDS->Cfg.szName, drvHostDSoundStreamStatusString(pStreamDS) ));
2154
2155	/*
2156	* We've started the buffer in looping mode, try switch to non-looping...
2157	*/
2158	int rc = VINF_SUCCESS;
2159	if (pStreamDS->Out.pDSB && !pStreamDS->Out.fDrain)
2160	{
2161	LogRel2(("DSound: Switching playback stream '%s' to drain mode...\n", pStreamDS->Cfg.szName));
2162	HRESULT hrc = IDirectSoundBuffer8_Stop(pStreamDS->Out.pDSB);
2163	if (SUCCEEDED(hrc))
2164	{
2165	hrc = IDirectSoundBuffer8_Play(pStreamDS->Out.pDSB, 0, 0, 0);
2166	if (SUCCEEDED(hrc))
2167	{
2168	uint64_t const msNow = RTTimeMilliTS();
2169	pStreamDS->Out.msDrainDeadline = PDMAudioPropsBytesToMilli(&pStreamDS->Cfg.Props, pStreamDS->cbBufSize) + msNow;
2170	pStreamDS->Out.fDrain = true;
2171	}
2172	else
2173	LogRelMax(64, ("DSound: Failed to restart '%s' in drain mode: %Rhrc\n", pStreamDS->Cfg.szName, hrc));
2174	}
2175	else
2176	{
2177	Log2Func(("drain: IDirectSoundBuffer8_Stop failed: %Rhrc\n", hrc));
2178	directSoundPlayRestore(pThis, pStreamDS->Out.pDSB);
2179
2180	HRESULT hrc2 = IDirectSoundBuffer8_Stop(pStreamDS->Out.pDSB);
2181	if (SUCCEEDED(hrc2))
2182	LogFunc(("Successfully stopped the stream after restoring it. (hrc=%Rhrc)\n", hrc));
2183	else
2184	{
2185	LogRelMax(64, ("DSound: Failed to stop playback stream '%s' for putting into drain mode: %Rhrc (initial), %Rhrc (after restore)\n",
2186	pStreamDS->Cfg.szName, hrc, hrc2));
2187	rc = VERR_AUDIO_STREAM_NOT_READY;
2188	}
2189	}
2190	}
2191	LogFlowFunc(("returns %Rrc {%s}\n", rc, drvHostDSoundStreamStatusString(pStreamDS)));
2192	return rc;
2193	}
2194
2195
2196	/**
2197	* Retrieves the number of free bytes available for writing to a DirectSound output stream.
2198	*
2199	* @return VBox status code. VERR_NOT_AVAILABLE if unable to determine or the
2200	* buffer was not recoverable.
2201	* @param pThis Host audio driver instance.
2202	* @param pStreamDS DirectSound output stream to retrieve number for.
2203	* @param pdwFree Where to return the free amount on success.
2204	* @param poffPlayCursor Where to return the play cursor offset.
2205	*/
2206	static int dsoundGetFreeOut(PDRVHOSTDSOUND pThis, PDSOUNDSTREAM pStreamDS, DWORD pdwFree, DWORD poffPlayCursor)
2207	{
2208	AssertPtrReturn(pThis, VERR_INVALID_POINTER);
2209	AssertPtrReturn(pStreamDS, VERR_INVALID_POINTER);
2210	AssertPtrReturn(pdwFree, VERR_INVALID_POINTER);
2211	AssertPtr(poffPlayCursor);
2212
2213	Assert(pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT); /* Paranoia. */
2214
2215	LPDIRECTSOUNDBUFFER8 pDSB = pStreamDS->Out.pDSB;
2216	AssertPtrReturn(pDSB, VERR_INVALID_POINTER);
2217
2218	HRESULT hr = S_OK;
2219
2220	/* Get the current play position which is used for calculating the free space in the buffer. */
2221	for (unsigned i = 0; i < DRV_DSOUND_RESTORE_ATTEMPTS_MAX; i++)
2222	{
2223	DWORD offPlayCursor = 0;
2224	DWORD offWriteCursor = 0;
2225	hr = IDirectSoundBuffer8_GetCurrentPosition(pDSB, &offPlayCursor, &offWriteCursor);
2226	if (SUCCEEDED(hr))
2227	{
2228	int32_t cbDiff = offWriteCursor - offPlayCursor;
2229	if (cbDiff < 0)
2230	cbDiff += pStreamDS->cbBufSize;
2231
2232	int32_t cbFree = offPlayCursor - pStreamDS->Out.offWritePos;
2233	if (cbFree < 0)
2234	cbFree += pStreamDS->cbBufSize;
2235
2236	if (cbFree > (int32_t)pStreamDS->cbBufSize - cbDiff)
2237	{
2238	/** @todo count/log these. */
2239	pStreamDS->Out.offWritePos = offWriteCursor;
2240	cbFree = pStreamDS->cbBufSize - cbDiff;
2241	}
2242
2243	/* When starting to use a DirectSound buffer, offPlayCursor and offWriteCursor
2244	* both point at position 0, so we won't be able to detect how many bytes
2245	* are writable that way.
2246	*
2247	* So use our per-stream written indicator to see if we just started a stream. */
2248	if (pStreamDS->Out.cbWritten == 0)
2249	cbFree = pStreamDS->cbBufSize;
2250
2251	LogRel4(("DSound: offPlayCursor=%RU32, offWriteCursor=%RU32, offWritePos=%RU32 -> cbFree=%RI32\n",
2252	offPlayCursor, offWriteCursor, pStreamDS->Out.offWritePos, cbFree));
2253
2254	*pdwFree = cbFree;
2255	*poffPlayCursor = offPlayCursor;
2256	return VINF_SUCCESS;
2257	}
2258
2259	if (hr != DSERR_BUFFERLOST) /** @todo MSDN doesn't state this error for GetCurrentPosition(). */
2260	break;
2261
2262	LogFunc(("Getting playing position failed due to lost buffer, restoring ...\n"));
2263
2264	directSoundPlayRestore(pThis, pDSB);
2265	}
2266
2267	if (hr != DSERR_BUFFERLOST) /* Avoid log flooding if the error is still there. */
2268	DSLOGREL(("DSound: Getting current playback position failed with %Rhrc\n", hr));
2269
2270	LogFunc(("Failed with %Rhrc\n", hr));
2271
2272	*poffPlayCursor = pStreamDS->cbBufSize;
2273	return VERR_NOT_AVAILABLE;
2274	}
2275
2276
2277	/**
2278	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetState}
2279	*/
2280	static DECLCALLBACK(PDMHOSTAUDIOSTREAMSTATE) drvHostDSoundHA_StreamGetState(PPDMIHOSTAUDIO pInterface,
2281	PPDMAUDIOBACKENDSTREAM pStream)
2282	{
2283	RT_NOREF(pInterface);
2284	PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream;
2285	AssertPtrReturn(pStreamDS, PDMHOSTAUDIOSTREAMSTATE_INVALID);
2286
2287	if ( pStreamDS->Cfg.enmDir != PDMAUDIODIR_OUT
2288	\|\| !pStreamDS->Out.fDrain)
2289	{
2290	LogFlowFunc(("returns OKAY for '%s' {%s}\n", pStreamDS->Cfg.szName, drvHostDSoundStreamStatusString(pStreamDS)));
2291	return PDMHOSTAUDIOSTREAMSTATE_OKAY;
2292	}
2293	LogFlowFunc(("returns DRAINING for '%s' {%s}\n", pStreamDS->Cfg.szName, drvHostDSoundStreamStatusString(pStreamDS)));
2294	return PDMHOSTAUDIOSTREAMSTATE_DRAINING;
2295	}
2296
2297	#if 0 /* This isn't working as the write cursor is more a function of time than what we do.
2298	Previously we only reported the pre-buffering status anyway, so no harm. */
2299	/**
2300	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetPending}
2301	*/
2302	static DECLCALLBACK(uint32_t) drvHostDSoundHA_StreamGetPending(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
2303	{
2304	/PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio); / RT_NOREF(pInterface);
2305	PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream;
2306	AssertPtrReturn(pStreamDS, 0);
2307	LogFlowFunc(("Stream '%s' {%s}\n", pStreamDS->Cfg.szName, drvHostDSoundStreamStatusString(pStreamDS)));
2308
2309	if (pStreamDS->Cfg.enmDir == PDMAUDIODIR_OUT)
2310	{
2311	/* This is a similar calculation as for StreamGetReadable, only for an output buffer. */
2312	AssertPtr(pStreamDS->In.pDSCB);
2313	DWORD offPlayCursor = 0;
2314	DWORD offWriteCursor = 0;
2315	HRESULT hrc = IDirectSoundBuffer8_GetCurrentPosition(pStreamDS->Out.pDSB, &offPlayCursor, &offWriteCursor);
2316	if (SUCCEEDED(hrc))
2317	{
2318	uint32_t cbPending = dsoundRingDistance(offWriteCursor, offPlayCursor, pStreamDS->cbBufSize);
2319	Log3Func(("cbPending=%RU32\n", cbPending));
2320	return cbPending;
2321	}
2322	AssertMsgFailed(("hrc=%Rhrc\n", hrc));
2323	}
2324	/* else: For input streams we never have any pending data. */
2325
2326	return 0;
2327	}
2328	#endif
2329
2330	/**
2331	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetWritable}
2332	*/
2333	static DECLCALLBACK(uint32_t) drvHostDSoundHA_StreamGetWritable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
2334	{
2335	PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio);
2336	PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream;
2337	AssertPtrReturn(pStreamDS, 0);
2338	LogFlowFunc(("Stream '%s' {%s}\n", pStreamDS->Cfg.szName, drvHostDSoundStreamStatusString(pStreamDS)));
2339
2340	DWORD cbFree = 0;
2341	DWORD offIgn = 0;
2342	int rc = dsoundGetFreeOut(pThis, pStreamDS, &cbFree, &offIgn);
2343	AssertRCReturn(rc, 0);
2344
2345	return cbFree;
2346	}
2347
2348
2349	/**
2350	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamPlay}
2351	*/
2352	static DECLCALLBACK(int) drvHostDSoundHA_StreamPlay(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream,
2353	const void pvBuf, uint32_t cbBuf, uint32_t pcbWritten)
2354	{
2355	PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio);
2356	PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream;
2357	AssertPtrReturn(pStreamDS, 0);
2358	if (cbBuf)
2359	AssertPtrReturn(pvBuf, VERR_INVALID_POINTER);
2360	AssertPtrReturn(pcbWritten, VERR_INVALID_POINTER);
2361
2362	if (pStreamDS->fEnabled)
2363	AssertReturn(pStreamDS->cbBufSize, VERR_INTERNAL_ERROR_2);
2364	else
2365	{
2366	Log2Func(("Skipping disabled stream {%s}\n", drvHostDSoundStreamStatusString(pStreamDS)));
2367	return VINF_SUCCESS;
2368	}
2369	Log4Func(("cbBuf=%#x stream '%s' {%s}\n", cbBuf, pStreamDS->Cfg.szName, drvHostDSoundStreamStatusString(pStreamDS) ));
2370
2371	/** @todo Any condition under which we should call dsoundUpdateStatusInternal(pThis) here?
2372	* The old code thought it did so in case of failure, only it couldn't ever fails, so it never did. */
2373
2374	/*
2375	* Transfer loop.
2376	*/
2377	uint32_t cbWritten = 0;
2378	while (cbBuf > 0)
2379	{
2380	/*
2381	* Figure out how much we can possibly write.
2382	*/
2383	DWORD offPlayCursor = 0;
2384	DWORD cbWritable = 0;
2385	int rc = dsoundGetFreeOut(pThis, pStreamDS, &cbWritable, &offPlayCursor);
2386	AssertRCReturn(rc, rc);
2387	if (cbWritable < pStreamDS->Cfg.Props.cbFrame)
2388	break;
2389
2390	uint32_t const cbToWrite = RT_MIN(cbWritable, cbBuf);
2391	Log3Func(("offPlay=%#x offWritePos=%#x -> cbWritable=%#x cbToWrite=%#x {%s}\n", offPlayCursor, pStreamDS->Out.offWritePos,
2392	cbWritable, cbToWrite, drvHostDSoundStreamStatusString(pStreamDS) ));
2393
2394	/*
2395	* Lock that amount of buffer.
2396	*/
2397	PVOID pv1 = NULL;
2398	DWORD cb1 = 0;
2399	PVOID pv2 = NULL;
2400	DWORD cb2 = 0;
2401	HRESULT hrc = directSoundPlayLock(pThis, pStreamDS, pStreamDS->Out.offWritePos, cbToWrite,
2402	&pv1, &pv2, &cb1, &cb2, 0 /dwFlags/);
2403	AssertMsgReturn(SUCCEEDED(hrc), ("%Rhrc\n", hrc), VERR_ACCESS_DENIED); /** @todo translate these status codes already! */
2404	//AssertMsg(cb1 + cb2 == cbToWrite, ("%#x + %#x vs %#x\n", cb1, cb2, cbToWrite));
2405
2406	/*
2407	* Copy over the data.
2408	*/
2409	memcpy(pv1, pvBuf, cb1);
2410	pvBuf = (uint8_t *)pvBuf + cb1;
2411	cbBuf -= cb1;
2412	cbWritten += cb1;
2413
2414	if (pv2)
2415	{
2416	memcpy(pv2, pvBuf, cb2);
2417	pvBuf = (uint8_t *)pvBuf + cb2;
2418	cbBuf -= cb2;
2419	cbWritten += cb2;
2420	}
2421
2422	/*
2423	* Unlock and update the write position.
2424	*/
2425	directSoundPlayUnlock(pThis, pStreamDS->Out.pDSB, pv1, pv2, cb1, cb2); /** @todo r=bird: pThis + pDSB parameters here for Unlock, but only pThis for Lock. Why? */
2426	pStreamDS->Out.offWritePos = (pStreamDS->Out.offWritePos + cb1 + cb2) % pStreamDS->cbBufSize;
2427
2428	/*
2429	* If this was the first chunk, kick off playing.
2430	*/
2431	if (!pStreamDS->Out.fFirstTransfer)
2432	{ /* likely */ }
2433	else
2434	{
2435	*pcbWritten = cbWritten;
2436	rc = directSoundPlayStart(pThis, pStreamDS);
2437	AssertRCReturn(rc, rc);
2438	pStreamDS->Out.fFirstTransfer = false;
2439	}
2440	}
2441
2442	/*
2443	* Done.
2444	*/
2445	*pcbWritten = cbWritten;
2446
2447	pStreamDS->Out.cbTransferred += cbWritten;
2448	if (cbWritten)
2449	{
2450	uint64_t const msPrev = pStreamDS->msLastTransfer; RT_NOREF(msPrev);
2451	pStreamDS->Out.cbLastTransferred = cbWritten;
2452	pStreamDS->msLastTransfer = RTTimeMilliTS();
2453	LogFlowFunc(("cbLastTransferred=%RU32, msLastTransfer=%RU64 msNow=%RU64 cMsDelta=%RU64 {%s}\n",
2454	cbWritten, msPrev, pStreamDS->msLastTransfer, msPrev ? pStreamDS->msLastTransfer - msPrev : 0,
2455	drvHostDSoundStreamStatusString(pStreamDS) ));
2456	}
2457	else if ( pStreamDS->Out.fDrain
2458	&& RTTimeMilliTS() >= pStreamDS->Out.msDrainDeadline)
2459	{
2460	LogRel2(("DSound: Stopping draining of '%s' {%s} ...\n", pStreamDS->Cfg.szName, drvHostDSoundStreamStatusString(pStreamDS)));
2461	if (pStreamDS->Out.pDSB)
2462	{
2463	HRESULT hrc = IDirectSoundBuffer8_Stop(pStreamDS->Out.pDSB);
2464	if (FAILED(hrc))
2465	LogRelMax(64, ("DSound: Failed to stop draining stream '%s': %Rhrc\n", pStreamDS->Cfg.szName, hrc));
2466	}
2467	pStreamDS->Out.fDrain = false;
2468	pStreamDS->fEnabled = false;
2469	}
2470
2471	return VINF_SUCCESS;
2472	}
2473
2474
2475	/**
2476	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamGetReadable}
2477	*/
2478	static DECLCALLBACK(uint32_t) drvHostDSoundHA_StreamGetReadable(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream)
2479	{
2480	/PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio); / RT_NOREF(pInterface);
2481	PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream;
2482	AssertPtrReturn(pStreamDS, 0);
2483	Assert(pStreamDS->Cfg.enmDir == PDMAUDIODIR_IN);
2484
2485	if (pStreamDS->fEnabled)
2486	{
2487	/* This is the same calculation as for StreamGetPending. */
2488	AssertPtr(pStreamDS->In.pDSCB);
2489	DWORD offCaptureCursor = 0;
2490	DWORD offReadCursor = 0;
2491	HRESULT hrc = IDirectSoundCaptureBuffer_GetCurrentPosition(pStreamDS->In.pDSCB, &offCaptureCursor, &offReadCursor);
2492	if (SUCCEEDED(hrc))
2493	{
2494	uint32_t cbPending = dsoundRingDistance(offCaptureCursor, offReadCursor, pStreamDS->cbBufSize);
2495	Log3Func(("cbPending=%RU32\n", cbPending));
2496	return cbPending;
2497	}
2498	AssertMsgFailed(("hrc=%Rhrc\n", hrc));
2499	}
2500
2501	return 0;
2502	}
2503
2504
2505	/**
2506	* @interface_method_impl{PDMIHOSTAUDIO,pfnStreamCapture}
2507	*/
2508	static DECLCALLBACK(int) drvHostDSoundHA_StreamCapture(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream,
2509	void pvBuf, uint32_t cbBuf, uint32_t pcbRead)
2510	{
2511	/PDRVHOSTDSOUND pThis = RT_FROM_MEMBER(pInterface, DRVHOSTDSOUND, IHostAudio);/ RT_NOREF(pInterface);
2512	PDSOUNDSTREAM pStreamDS = (PDSOUNDSTREAM)pStream;
2513	AssertPtrReturn(pStreamDS, 0);
2514	AssertPtrReturn(pvBuf, VERR_INVALID_POINTER);
2515	AssertReturn(cbBuf, VERR_INVALID_PARAMETER);
2516	AssertPtrReturn(pcbRead, VERR_INVALID_POINTER);
2517
2518	#if 0 /** @todo r=bird: shouldn't we do the same check as for output streams? */
2519	if (pStreamDS->fEnabled)
2520	AssertReturn(pStreamDS->cbBufSize, VERR_INTERNAL_ERROR_2);
2521	else
2522	{
2523	Log2Func(("Stream disabled, skipping\n"));
2524	return VINF_SUCCESS;
2525	}
2526	#endif
2527	Log4Func(("cbBuf=%#x stream '%s' {%s}\n", cbBuf, pStreamDS->Cfg.szName, drvHostDSoundStreamStatusString(pStreamDS) ));
2528
2529	/*
2530	* Read loop.
2531	*/
2532	uint32_t cbRead = 0;
2533	while (cbBuf > 0)
2534	{
2535	/*
2536	* Figure out how much we can read.
2537	*/
2538	DWORD offCaptureCursor = 0;
2539	DWORD offReadCursor = 0;
2540	HRESULT hrc = IDirectSoundCaptureBuffer_GetCurrentPosition(pStreamDS->In.pDSCB, &offCaptureCursor, &offReadCursor);
2541	AssertMsgReturn(SUCCEEDED(hrc), ("%Rhrc\n", hrc), VERR_ACCESS_DENIED); /** @todo translate these status codes already! */
2542	//AssertMsg(offReadCursor == pStreamDS->In.offReadPos, ("%#x %#x\n", offReadCursor, pStreamDS->In.offReadPos));
2543
2544	uint32_t const cbReadable = dsoundRingDistance(offCaptureCursor, pStreamDS->In.offReadPos, pStreamDS->cbBufSize);
2545
2546	if (cbReadable >= pStreamDS->Cfg.Props.cbFrame)
2547	{ /* likely */ }
2548	else
2549	{
2550	if (cbRead > 0)
2551	{ /* likely */ }
2552	else if (pStreamDS->In.cOverruns < 32)
2553	{
2554	pStreamDS->In.cOverruns++;
2555	DSLOG(("DSound: Warning: Buffer full (size is %zu bytes), skipping to record data (overflow #%RU32)\n",
2556	pStreamDS->cbBufSize, pStreamDS->In.cOverruns));
2557	}
2558	break;
2559	}
2560
2561	uint32_t const cbToRead = RT_MIN(cbReadable, cbBuf);
2562	Log3Func(("offCapture=%#x offRead=%#x/%#x -> cbWritable=%#x cbToWrite=%#x {%s}\n", offCaptureCursor, offReadCursor,
2563	pStreamDS->In.offReadPos, cbReadable, cbToRead, drvHostDSoundStreamStatusString(pStreamDS)));
2564
2565	/*
2566	* Lock that amount of buffer.
2567	*/
2568	PVOID pv1 = NULL;
2569	DWORD cb1 = 0;
2570	PVOID pv2 = NULL;
2571	DWORD cb2 = 0;
2572	hrc = directSoundCaptureLock(pStreamDS, pStreamDS->In.offReadPos, cbToRead, &pv1, &pv2, &cb1, &cb2, 0 /dwFlags/);
2573	AssertMsgReturn(SUCCEEDED(hrc), ("%Rhrc\n", hrc), VERR_ACCESS_DENIED); /** @todo translate these status codes already! */
2574	AssertMsg(cb1 + cb2 == cbToRead, ("%#x + %#x vs %#x\n", cb1, cb2, cbToRead));
2575
2576	/*
2577	* Copy over the data.
2578	*/
2579	memcpy(pvBuf, pv1, cb1);
2580	pvBuf = (uint8_t *)pvBuf + cb1;
2581	cbBuf -= cb1;
2582	cbRead += cb1;
2583
2584	if (pv2)
2585	{
2586	memcpy(pvBuf, pv2, cb2);
2587	pvBuf = (uint8_t *)pvBuf + cb2;
2588	cbBuf -= cb2;
2589	cbRead += cb2;
2590	}
2591
2592	/*
2593	* Unlock and update the write position.
2594	*/
2595	directSoundCaptureUnlock(pStreamDS->In.pDSCB, pv1, pv2, cb1, cb2); /** @todo r=bird: pDSB parameter here for Unlock, but pStreamDS for Lock. Why? */
2596	pStreamDS->In.offReadPos = (pStreamDS->In.offReadPos + cb1 + cb2) % pStreamDS->cbBufSize;
2597	}
2598
2599	/*
2600	* Done.
2601	*/
2602	*pcbRead = cbRead;
2603	if (cbRead)
2604	{
2605	uint64_t const msPrev = pStreamDS->msLastTransfer; RT_NOREF(msPrev);
2606	pStreamDS->msLastTransfer = RTTimeMilliTS();
2607	LogFlowFunc(("cbRead=%RU32, msLastTransfer=%RU64 msNow=%RU64 cMsDelta=%RU64 {%s}\n",
2608	cbRead, msPrev, pStreamDS->msLastTransfer, msPrev ? pStreamDS->msLastTransfer - msPrev : 0,
2609	drvHostDSoundStreamStatusString(pStreamDS) ));
2610	}
2611
2612	return VINF_SUCCESS;
2613	}
2614
2615
2616	/*********************************************************************************************************************************
2617	* PDMDRVINS::IBase Interface *
2618	*********************************************************************************************************************************/
2619
2620	/**
2621	* @callback_method_impl{PDMIBASE,pfnQueryInterface}
2622	*/
2623	static DECLCALLBACK(void ) drvHostDSoundQueryInterface(PPDMIBASE pInterface, const char pszIID)
2624	{
2625	PPDMDRVINS pDrvIns = PDMIBASE_2_PDMDRV(pInterface);
2626	PDRVHOSTDSOUND pThis = PDMINS_2_DATA(pDrvIns, PDRVHOSTDSOUND);
2627
2628	PDMIBASE_RETURN_INTERFACE(pszIID, PDMIBASE, &pDrvIns->IBase);
2629	PDMIBASE_RETURN_INTERFACE(pszIID, PDMIHOSTAUDIO, &pThis->IHostAudio);
2630	return NULL;
2631	}
2632
2633
2634	/*********************************************************************************************************************************
2635	* PDMDRVREG Interface *
2636	*********************************************************************************************************************************/
2637
2638	/**
2639	* @callback_method_impl{FNPDMDRVDESTRUCT, pfnDestruct}
2640	*/
2641	static DECLCALLBACK(void) drvHostDSoundDestruct(PPDMDRVINS pDrvIns)
2642	{
2643	PDRVHOSTDSOUND pThis = PDMINS_2_DATA(pDrvIns, PDRVHOSTDSOUND);
2644	PDMDRV_CHECK_VERSIONS_RETURN_VOID(pDrvIns);
2645
2646	LogFlowFuncEnter();
2647
2648	#ifdef VBOX_WITH_AUDIO_MMNOTIFICATION_CLIENT
2649	if (pThis->m_pNotificationClient)
2650	{
2651	pThis->m_pNotificationClient->Unregister();
2652	pThis->m_pNotificationClient->Release();
2653
2654	pThis->m_pNotificationClient = NULL;
2655	}
2656	#endif
2657
2658	PDMAudioHostEnumDelete(&pThis->DeviceEnum);
2659
2660	int rc2 = RTCritSectDelete(&pThis->CritSect);
2661	AssertRC(rc2);
2662
2663	LogFlowFuncLeave();
2664	}
2665
2666
2667	static LPCGUID dsoundConfigQueryGUID(PPDMDRVINS pDrvIns, PCFGMNODE pCfg, const char pszName, RTUUID pUuid)
2668	{
2669	PCPDMDRVHLPR3 pHlp = pDrvIns->pHlpR3;
2670	LPCGUID pGuid = NULL;
2671
2672	char *pszGuid = NULL;
2673	int rc = pHlp->pfnCFGMQueryStringAlloc(pCfg, pszName, &pszGuid);
2674	if (RT_SUCCESS(rc))
2675	{
2676	rc = RTUuidFromStr(pUuid, pszGuid);
2677	if (RT_SUCCESS(rc))
2678	pGuid = (LPCGUID)&pUuid;
2679	else
2680	DSLOGREL(("DSound: Error parsing device GUID for device '%s': %Rrc\n", pszName, rc));
2681
2682	RTStrFree(pszGuid);
2683	}
2684
2685	return pGuid;
2686	}
2687
2688
2689	static void dsoundConfigInit(PDRVHOSTDSOUND pThis, PCFGMNODE pCfg)
2690	{
2691	pThis->Cfg.pGuidPlay = dsoundConfigQueryGUID(pThis->pDrvIns, pCfg, "DeviceGuidOut", &pThis->Cfg.uuidPlay);
2692	pThis->Cfg.pGuidCapture = dsoundConfigQueryGUID(pThis->pDrvIns, pCfg, "DeviceGuidIn", &pThis->Cfg.uuidCapture);
2693
2694	DSLOG(("DSound: Configuration: DeviceGuidOut {%RTuuid}, DeviceGuidIn {%RTuuid}\n",
2695	&pThis->Cfg.uuidPlay,
2696	&pThis->Cfg.uuidCapture));
2697	}
2698
2699
2700	/**
2701	* @callback_method_impl{FNPDMDRVCONSTRUCT,
2702	* Construct a DirectSound Audio driver instance.}
2703	*/
2704	static DECLCALLBACK(int) drvHostDSoundConstruct(PPDMDRVINS pDrvIns, PCFGMNODE pCfg, uint32_t fFlags)
2705	{
2706	PDMDRV_CHECK_VERSIONS_RETURN(pDrvIns);
2707	PDRVHOSTDSOUND pThis = PDMINS_2_DATA(pDrvIns, PDRVHOSTDSOUND);
2708	RT_NOREF(fFlags);
2709	LogRel(("Audio: Initializing DirectSound audio driver\n"));
2710
2711	/*
2712	* Init basic data members and interfaces.
2713	*/
2714	RTListInit(&pThis->HeadStreams);
2715	pThis->pDrvIns = pDrvIns;
2716	/* IBase */
2717	pDrvIns->IBase.pfnQueryInterface = drvHostDSoundQueryInterface;
2718	/* IHostAudio */
2719	pThis->IHostAudio.pfnGetConfig = drvHostDSoundHA_GetConfig;
2720	pThis->IHostAudio.pfnGetDevices = drvHostDSoundHA_GetDevices;
2721	pThis->IHostAudio.pfnSetDevice = NULL;
2722	pThis->IHostAudio.pfnGetStatus = drvHostDSoundHA_GetStatus;
2723	pThis->IHostAudio.pfnDoOnWorkerThread = NULL;
2724	pThis->IHostAudio.pfnStreamConfigHint = NULL;
2725	pThis->IHostAudio.pfnStreamCreate = drvHostDSoundHA_StreamCreate;
2726	pThis->IHostAudio.pfnStreamInitAsync = NULL;
2727	pThis->IHostAudio.pfnStreamDestroy = drvHostDSoundHA_StreamDestroy;
2728	pThis->IHostAudio.pfnStreamNotifyDeviceChanged = NULL;
2729	pThis->IHostAudio.pfnStreamEnable = drvHostDSoundHA_StreamEnable;
2730	pThis->IHostAudio.pfnStreamDisable = drvHostDSoundHA_StreamDisable;
2731	pThis->IHostAudio.pfnStreamPause = drvHostDSoundHA_StreamPause;
2732	pThis->IHostAudio.pfnStreamResume = drvHostDSoundHA_StreamResume;
2733	pThis->IHostAudio.pfnStreamDrain = drvHostDSoundHA_StreamDrain;
2734	pThis->IHostAudio.pfnStreamGetState = drvHostDSoundHA_StreamGetState;
2735	pThis->IHostAudio.pfnStreamGetPending = NULL;
2736	pThis->IHostAudio.pfnStreamGetWritable = drvHostDSoundHA_StreamGetWritable;
2737	pThis->IHostAudio.pfnStreamPlay = drvHostDSoundHA_StreamPlay;
2738	pThis->IHostAudio.pfnStreamGetReadable = drvHostDSoundHA_StreamGetReadable;
2739	pThis->IHostAudio.pfnStreamCapture = drvHostDSoundHA_StreamCapture;
2740
2741	/*
2742	* Init the static parts.
2743	*/
2744	PDMAudioHostEnumInit(&pThis->DeviceEnum);
2745
2746	pThis->fEnabledIn = false;
2747	pThis->fEnabledOut = false;
2748
2749	/*
2750	* Verify that IDirectSound is available.
2751	*/
2752	LPDIRECTSOUND pDirectSound = NULL;
2753	HRESULT hrc = CoCreateInstance(CLSID_DirectSound, NULL, CLSCTX_ALL, IID_IDirectSound, (void **)&pDirectSound);
2754	if (SUCCEEDED(hrc))
2755	IDirectSound_Release(pDirectSound);
2756	else
2757	{
2758	LogRel(("DSound: DirectSound not available: %Rhrc\n", hrc));
2759	return VERR_AUDIO_BACKEND_INIT_FAILED;
2760	}
2761
2762	#ifdef VBOX_WITH_AUDIO_MMNOTIFICATION_CLIENT
2763	/*
2764	* Set up WASAPI device change notifications (Vista+).
2765	*/
2766	if (RTSystemGetNtVersion() >= RTSYSTEM_MAKE_NT_VERSION(6, 0, 0))
2767	{
2768	/* Get the notification interface (from DrvAudio). */
2769	# ifdef VBOX_WITH_AUDIO_CALLBACKS
2770	PPDMIHOSTAUDIOPORT pIHostAudioPort = PDMIBASE_QUERY_INTERFACE(pDrvIns->pUpBase, PDMIHOSTAUDIOPORT);
2771	Assert(pIHostAudioPort);
2772	# else
2773	PPDMIHOSTAUDIOPORT pIHostAudioPort = NULL;
2774	# endif
2775	try
2776	{
2777	pThis->m_pNotificationClient = new DrvHostAudioDSoundMMNotifClient(pIHostAudioPort,
2778	pThis->Cfg.pGuidCapture == NULL,
2779	pThis->Cfg.pGuidPlay == NULL);
2780	}
2781	catch (std::bad_alloc &)
2782	{
2783	return VERR_NO_MEMORY;
2784	}
2785	hrc = pThis->m_pNotificationClient->Initialize();
2786	if (SUCCEEDED(hrc))
2787	{
2788	hrc = pThis->m_pNotificationClient->Register();
2789	if (SUCCEEDED(hrc))
2790	LogRel2(("DSound: Notification client is enabled (ver %#RX64)\n", RTSystemGetNtVersion()));
2791	else
2792	{
2793	LogRel(("DSound: Notification client registration failed: %Rhrc\n", hrc));
2794	return VERR_AUDIO_BACKEND_INIT_FAILED;
2795	}
2796	}
2797	else
2798	{
2799	LogRel(("DSound: Notification client initialization failed: %Rhrc\n", hrc));
2800	return VERR_AUDIO_BACKEND_INIT_FAILED;
2801	}
2802	}
2803	else
2804	LogRel2(("DSound: Notification client is disabled (ver %#RX64)\n", RTSystemGetNtVersion()));
2805	#endif
2806
2807	/*
2808	* Initialize configuration values and critical section.
2809	*/
2810	dsoundConfigInit(pThis, pCfg);
2811	return RTCritSectInit(&pThis->CritSect);
2812	}
2813
2814
2815	/**
2816	* PDM driver registration.
2817	*/
2818	const PDMDRVREG g_DrvHostDSound =
2819	{
2820	/* u32Version */
2821	PDM_DRVREG_VERSION,
2822	/* szName */
2823	"DSoundAudio",
2824	/* szRCMod */
2825	"",
2826	/* szR0Mod */
2827	"",
2828	/* pszDescription */
2829	"DirectSound Audio host driver",
2830	/* fFlags */
2831	PDM_DRVREG_FLAGS_HOST_BITS_DEFAULT,
2832	/* fClass. */
2833	PDM_DRVREG_CLASS_AUDIO,
2834	/* cMaxInstances */
2835	~0U,
2836	/* cbInstance */
2837	sizeof(DRVHOSTDSOUND),
2838	/* pfnConstruct */
2839	drvHostDSoundConstruct,
2840	/* pfnDestruct */
2841	drvHostDSoundDestruct,
2842	/* pfnRelocate */
2843	NULL,
2844	/* pfnIOCtl */
2845	NULL,
2846	/* pfnPowerOn */
2847	NULL,
2848	/* pfnReset */
2849	NULL,
2850	/* pfnSuspend */
2851	NULL,
2852	/* pfnResume */
2853	NULL,
2854	/* pfnAttach */
2855	NULL,
2856	/* pfnDetach */
2857	NULL,
2858	/* pfnPowerOff */
2859	NULL,
2860	/* pfnSoftReset */
2861	NULL,
2862	/* u32EndVersion */
2863	PDM_DRVREG_VERSION
2864	};

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source: vbox/trunk/src/VBox/Devices/Audio/DrvHostAudioDSound.cpp@ 94368

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