vkatCommon.cpp@ 104620

Last change on this file since 104620 was 103570, checked in by vboxsync, 7 months ago
Audio/VKAT: Use a circular buffer when recording data; should make it easier to detect and handle start/stop beacons [build fix].
Property svn:eol-style set to `native` Property svn:keywords set to `Author Date Id Revision`
File size: 72.3 KB

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1	/* $Id: vkatCommon.cpp 103570 2024-02-26 13:12:38Z vboxsync $ */
2	/** @file
3	* Validation Kit Audio Test (VKAT) - Common code.
4	*/
5
6	/*
7	* Copyright (C) 2021-2023 Oracle and/or its affiliates.
8	*
9	* This file is part of VirtualBox base platform packages, as
10	* available from https://www.virtualbox.org.
11	*
12	* This program is free software; you can redistribute it and/or
13	* modify it under the terms of the GNU General Public License
14	* as published by the Free Software Foundation, in version 3 of the
15	* License.
16	*
17	* This program is distributed in the hope that it will be useful, but
18	* WITHOUT ANY WARRANTY; without even the implied warranty of
19	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
20	* General Public License for more details.
21	*
22	* You should have received a copy of the GNU General Public License
23	* along with this program; if not, see <https://www.gnu.org/licenses>.
24	*
25	* The contents of this file may alternatively be used under the terms
26	* of the Common Development and Distribution License Version 1.0
27	* (CDDL), a copy of it is provided in the "COPYING.CDDL" file included
28	* in the VirtualBox distribution, in which case the provisions of the
29	* CDDL are applicable instead of those of the GPL.
30	*
31	* You may elect to license modified versions of this file under the
32	* terms and conditions of either the GPL or the CDDL or both.
33	*
34	* SPDX-License-Identifier: GPL-3.0-only OR CDDL-1.0
35	*/
36
37
38	/*********************************************************************************************************************************
39	* Header Files *
40	*********************************************************************************************************************************/
41	#define LOG_GROUP LOG_GROUP_AUDIO_TEST
42	#include <iprt/log.h>
43
44	#ifdef VBOX_WITH_AUDIO_ALSA
45	# include "DrvHostAudioAlsaStubsMangling.h"
46	# include <alsa/asoundlib.h>
47	# include <alsa/control.h> /* For device enumeration. */
48	# include <alsa/version.h>
49	# include "DrvHostAudioAlsaStubs.h"
50	#endif
51	#ifdef VBOX_WITH_AUDIO_OSS
52	# include <errno.h>
53	# include <fcntl.h>
54	# include <sys/ioctl.h>
55	# include <sys/mman.h>
56	# include <sys/soundcard.h>
57	# include <unistd.h>
58	#endif
59	#ifdef RT_OS_WINDOWS
60	# include <iprt/win/windows.h>
61	# include <iprt/win/audioclient.h>
62	# include <endpointvolume.h> /* For IAudioEndpointVolume. */
63	# include <audiopolicy.h> /* For IAudioSessionManager. */
64	# include <AudioSessionTypes.h>
65	# include <Mmdeviceapi.h>
66	#endif
67
68	#include <iprt/circbuf.h>
69	#include <iprt/ctype.h>
70	#include <iprt/dir.h>
71	#include <iprt/errcore.h>
72	#include <iprt/getopt.h>
73	#include <iprt/message.h>
74	#include <iprt/rand.h>
75	#include <iprt/test.h>
76
77	#include "Audio/AudioHlp.h"
78	#include "Audio/AudioTest.h"
79	#include "Audio/AudioTestService.h"
80	#include "Audio/AudioTestServiceClient.h"
81
82	#include "vkatInternal.h"
83
84
85	/*********************************************************************************************************************************
86	* Defined Constants And Macros *
87	*********************************************************************************************************************************/
88
89
90	/*********************************************************************************************************************************
91	* Internal Functions *
92	*********************************************************************************************************************************/
93	static int audioTestStreamInit(PAUDIOTESTDRVSTACK pDrvStack, PAUDIOTESTSTREAM pStream, PDMAUDIODIR enmDir, PAUDIOTESTIOOPTS pPlayOpt);
94	static int audioTestStreamDestroy(PAUDIOTESTDRVSTACK pDrvStack, PAUDIOTESTSTREAM pStream);
95
96
97	/*********************************************************************************************************************************
98	* Volume handling. *
99	*********************************************************************************************************************************/
100
101	#ifdef VBOX_WITH_AUDIO_ALSA
102	/**
103	* Sets the system's master volume via ALSA, if available.
104	*
105	* @returns VBox status code.
106	* @param uVolPercent Volume (in percent) to set.
107	*/
108	static int audioTestSetMasterVolumeALSA(unsigned uVolPercent)
109	{
110	int rc = audioLoadAlsaLib();
111	if (RT_FAILURE(rc))
112	return rc;
113
114	int err;
115	snd_mixer_t *handle;
116
117	# define ALSA_CHECK_RET(a_Exp, a_Text) \
118	if (!(a_Exp)) \
119	{ \
120	AssertLogRelMsg(a_Exp, a_Text); \
121	if (handle) \
122	snd_mixer_close(handle); \
123	return VERR_GENERAL_FAILURE; \
124	}
125
126	# define ALSA_CHECK_ERR_RET(a_Text) \
127	ALSA_CHECK_RET(err >= 0, a_Text)
128
129	err = snd_mixer_open(&handle, 0 /* Index */);
130	ALSA_CHECK_ERR_RET(("ALSA: Failed to open mixer: %s\n", snd_strerror(err)));
131	err = snd_mixer_attach(handle, "default");
132	ALSA_CHECK_ERR_RET(("ALSA: Failed to attach to default sink: %s\n", snd_strerror(err)));
133	err = snd_mixer_selem_register(handle, NULL, NULL);
134	ALSA_CHECK_ERR_RET(("ALSA: Failed to attach to default sink: %s\n", snd_strerror(err)));
135	err = snd_mixer_load(handle);
136	ALSA_CHECK_ERR_RET(("ALSA: Failed to load mixer: %s\n", snd_strerror(err)));
137
138	snd_mixer_selem_id_t *sid = NULL;
139	snd_mixer_selem_id_alloca(&sid);
140
141	snd_mixer_selem_id_set_index(sid, 0 /* Index */);
142	snd_mixer_selem_id_set_name(sid, "Master");
143
144	snd_mixer_elem_t* elem = snd_mixer_find_selem(handle, sid);
145	ALSA_CHECK_RET(elem != NULL, ("ALSA: Failed to find mixer element: %s\n", snd_strerror(err)));
146
147	long uVolMin, uVolMax;
148	snd_mixer_selem_get_playback_volume_range(elem, &uVolMin, &uVolMax);
149	ALSA_CHECK_ERR_RET(("ALSA: Failed to get playback volume range: %s\n", snd_strerror(err)));
150
151	long const uVol = RT_MIN(uVolPercent, 100) * uVolMax / 100;
152
153	err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, uVol);
154	ALSA_CHECK_ERR_RET(("ALSA: Failed to set playback volume left: %s\n", snd_strerror(err)));
155	err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, uVol);
156	ALSA_CHECK_ERR_RET(("ALSA: Failed to set playback volume right: %s\n", snd_strerror(err)));
157
158	snd_mixer_close(handle);
159
160	return VINF_SUCCESS;
161
162	# undef ALSA_CHECK_RET
163	# undef ALSA_CHECK_ERR_RET
164	}
165	#endif /* VBOX_WITH_AUDIO_ALSA */
166
167	#ifdef VBOX_WITH_AUDIO_OSS
168	/**
169	* Sets the system's master volume via OSS, if available.
170	*
171	* @returns VBox status code.
172	* @param uVolPercent Volume (in percent) to set.
173	*/
174	static int audioTestSetMasterVolumeOSS(unsigned uVolPercent)
175	{
176	int hFile = open("/dev/dsp", O_WRONLY \| O_NONBLOCK, 0);
177	if (hFile == -1)
178	{
179	/* Try opening the mixing device instead. */
180	hFile = open("/dev/mixer", O_RDONLY \| O_NONBLOCK, 0);
181	}
182
183	if (hFile != -1)
184	{
185	/* OSS maps 0 (muted) - 100 (max), so just use uVolPercent unmodified here. */
186	uint16_t uVol = RT_MAKE_U16(uVolPercent, uVolPercent);
187	AssertLogRelMsgReturnStmt(ioctl(hFile, SOUND_MIXER_PCM /* SNDCTL_DSP_SETPLAYVOL */, &uVol) >= 0,
188	("OSS: Failed to set DSP playback volume: %s (%d)\n",
189	strerror(errno), errno), close(hFile), RTErrConvertFromErrno(errno));
190	return VINF_SUCCESS;
191	}
192
193	return VERR_NOT_SUPPORTED;
194	}
195	#endif /* VBOX_WITH_AUDIO_OSS */
196
197	#ifdef RT_OS_WINDOWS
198	static int audioTestSetMasterVolumeWASAPI(unsigned uVolPercent)
199	{
200	HRESULT hr;
201
202	# define WASAPI_CHECK_HR_RET(a_Text) \
203	if (FAILED(hr)) \
204	{ \
205	AssertLogRelMsgFailed(a_Text); \
206	return VERR_GENERAL_FAILURE; \
207	}
208
209	hr = CoInitialize(NULL);
210	WASAPI_CHECK_HR_RET(("CoInitialize() failed, hr=%Rhrc", hr));
211	IMMDeviceEnumerator* pIEnumerator = NULL;
212	hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, CLSCTX_ALL, __uuidof(IMMDeviceEnumerator), (void **)&pIEnumerator);
213	WASAPI_CHECK_HR_RET(("WASAPI: Unable to create IMMDeviceEnumerator, hr=%Rhrc", hr));
214
215	IMMDevice *pIMMDevice = NULL;
216	hr = pIEnumerator->GetDefaultAudioEndpoint(EDataFlow::eRender, ERole::eConsole, &pIMMDevice);
217	WASAPI_CHECK_HR_RET(("WASAPI: Unable to get audio endpoint, hr=%Rhrc", hr));
218	pIEnumerator->Release();
219
220	IAudioEndpointVolume *pIAudioEndpointVolume = NULL;
221	hr = pIMMDevice->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, (void **)&pIAudioEndpointVolume);
222	WASAPI_CHECK_HR_RET(("WASAPI: Unable to activate audio endpoint volume, hr=%Rhrc", hr));
223	pIMMDevice->Release();
224
225	float dbMin, dbMax, dbInc;
226	hr = pIAudioEndpointVolume->GetVolumeRange(&dbMin, &dbMax, &dbInc);
227	WASAPI_CHECK_HR_RET(("WASAPI: Unable to get volume range, hr=%Rhrc", hr));
228
229	float const dbSteps = (dbMax - dbMin) / dbInc;
230	float const dbStepsPerPercent = (dbSteps * dbInc) / 100;
231	float const dbVol = dbMin + (dbStepsPerPercent * (float(RT_MIN(uVolPercent, 100.0))));
232
233	hr = pIAudioEndpointVolume->SetMasterVolumeLevel(dbVol, NULL);
234	WASAPI_CHECK_HR_RET(("WASAPI: Unable to set master volume level, hr=%Rhrc", hr));
235	pIAudioEndpointVolume->Release();
236
237	return VINF_SUCCESS;
238
239	# undef WASAPI_CHECK_HR_RET
240	}
241	#endif /* RT_OS_WINDOWS */
242
243	/**
244	* Sets the system's master volume, if available.
245	*
246	* @returns VBox status code. VERR_NOT_SUPPORTED if not supported.
247	* @param uVolPercent Volume (in percent) to set.
248	*/
249	static int audioTestSetMasterVolume(unsigned uVolPercent)
250	{
251	int rc = VINF_SUCCESS;
252
253	#ifdef VBOX_WITH_AUDIO_ALSA
254	rc = audioTestSetMasterVolumeALSA(uVolPercent);
255	if (RT_SUCCESS(rc))
256	return rc;
257	/* else try OSS (if available) below. */
258	#endif /* VBOX_WITH_AUDIO_ALSA */
259
260	#ifdef VBOX_WITH_AUDIO_OSS
261	rc = audioTestSetMasterVolumeOSS(uVolPercent);
262	if (RT_SUCCESS(rc))
263	return rc;
264	#endif /* VBOX_WITH_AUDIO_OSS */
265
266	#ifdef RT_OS_WINDOWS
267	rc = audioTestSetMasterVolumeWASAPI(uVolPercent);
268	if (RT_SUCCESS(rc))
269	return rc;
270	#endif
271
272	RT_NOREF(rc, uVolPercent);
273	/** @todo Port other platforms. */
274	return VERR_NOT_SUPPORTED;
275	}
276
277
278	/*********************************************************************************************************************************
279	* Device enumeration + handling. *
280	*********************************************************************************************************************************/
281
282	/**
283	* Enumerates audio devices and optionally searches for a specific device.
284	*
285	* @returns VBox status code.
286	* @param pDrvStack Driver stack to use for enumeration.
287	* @param pszDev Device name to search for. Can be NULL if the default device shall be used.
288	* @param ppDev Where to return the pointer of the device enumeration of \a pTstEnv when a
289	* specific device was found.
290	*/
291	int audioTestDevicesEnumerateAndCheck(PAUDIOTESTDRVSTACK pDrvStack, const char pszDev, PPDMAUDIOHOSTDEV ppDev)
292	{
293	RTTestSubF(g_hTest, "Enumerating audio devices and checking for device '%s'", pszDev && *pszDev ? pszDev : "[Default]");
294
295	if (!pDrvStack->pIHostAudio->pfnGetDevices)
296	{
297	RTTestSkipped(g_hTest, "Backend does not support device enumeration, skipping");
298	return VINF_NOT_SUPPORTED;
299	}
300
301	Assert(pszDev == NULL \|\| ppDev);
302
303	if (ppDev)
304	*ppDev = NULL;
305
306	int rc = pDrvStack->pIHostAudio->pfnGetDevices(pDrvStack->pIHostAudio, &pDrvStack->DevEnum);
307	if (RT_SUCCESS(rc))
308	{
309	PPDMAUDIOHOSTDEV pDev;
310	RTListForEach(&pDrvStack->DevEnum.LstDevices, pDev, PDMAUDIOHOSTDEV, ListEntry)
311	{
312	char szFlags[PDMAUDIOHOSTDEV_MAX_FLAGS_STRING_LEN];
313	if (pDev->pszId)
314	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Enum: Device '%s' (ID '%s'):\n", pDev->pszName, pDev->pszId);
315	else
316	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Enum: Device '%s':\n", pDev->pszName);
317	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Enum: Usage = %s\n", PDMAudioDirGetName(pDev->enmUsage));
318	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Enum: Flags = %s\n", PDMAudioHostDevFlagsToString(szFlags, pDev->fFlags));
319	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Enum: Input channels = %RU8\n", pDev->cMaxInputChannels);
320	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Enum: Output channels = %RU8\n", pDev->cMaxOutputChannels);
321
322	if ( (pszDev && *pszDev)
323	&& !RTStrCmp(pDev->pszName, pszDev))
324	{
325	*ppDev = pDev;
326	}
327	}
328	}
329	else
330	RTTestFailed(g_hTest, "Enumerating audio devices failed with %Rrc", rc);
331
332	if (RT_SUCCESS(rc))
333	{
334	if ( (pszDev && *pszDev)
335	&& *ppDev == NULL)
336	{
337	RTTestFailed(g_hTest, "Audio device '%s' not found", pszDev);
338	rc = VERR_NOT_FOUND;
339	}
340	}
341
342	RTTestSubDone(g_hTest);
343	return rc;
344	}
345
346	static int audioTestStreamInit(PAUDIOTESTDRVSTACK pDrvStack, PAUDIOTESTSTREAM pStream,
347	PDMAUDIODIR enmDir, PAUDIOTESTIOOPTS pIoOpts)
348	{
349	int rc;
350
351	if (enmDir == PDMAUDIODIR_IN)
352	rc = audioTestDriverStackStreamCreateInput(pDrvStack, &pIoOpts->Props, pIoOpts->cMsBufferSize,
353	pIoOpts->cMsPreBuffer, pIoOpts->cMsSchedulingHint, &pStream->pStream, &pStream->Cfg);
354	else if (enmDir == PDMAUDIODIR_OUT)
355	rc = audioTestDriverStackStreamCreateOutput(pDrvStack, &pIoOpts->Props, pIoOpts->cMsBufferSize,
356	pIoOpts->cMsPreBuffer, pIoOpts->cMsSchedulingHint, &pStream->pStream, &pStream->Cfg);
357	else
358	rc = VERR_NOT_SUPPORTED;
359
360	if (RT_SUCCESS(rc))
361	{
362	if (!pDrvStack->pIAudioConnector)
363	{
364	pStream->pBackend = &((PAUDIOTESTDRVSTACKSTREAM)pStream->pStream)->Backend;
365	}
366	else
367	pStream->pBackend = NULL;
368
369	/*
370	* Automatically enable the mixer if the PCM properties don't match.
371	*/
372	if ( !pIoOpts->fWithMixer
373	&& !PDMAudioPropsAreEqual(&pIoOpts->Props, &pStream->Cfg.Props))
374	{
375	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Enabling stream mixer\n");
376	pIoOpts->fWithMixer = true;
377	}
378
379	rc = AudioTestMixStreamInit(&pStream->Mix, pDrvStack, pStream->pStream,
380	pIoOpts->fWithMixer ? &pIoOpts->Props : NULL, 100 /* ms /); /* @todo Configure mixer buffer? */
381	}
382
383	if (RT_FAILURE(rc))
384	RTTestFailed(g_hTest, "Initializing %s stream failed with %Rrc", enmDir == PDMAUDIODIR_IN ? "input" : "output", rc);
385
386	return rc;
387	}
388
389	/**
390	* Destroys an audio test stream.
391	*
392	* @returns VBox status code.
393	* @param pDrvStack Driver stack the stream belongs to.
394	* @param pStream Audio stream to destroy.
395	*/
396	static int audioTestStreamDestroy(PAUDIOTESTDRVSTACK pDrvStack, PAUDIOTESTSTREAM pStream)
397	{
398	AssertPtrReturn(pStream, VERR_INVALID_POINTER);
399
400	if (pStream->pStream)
401	{
402	/** @todo Anything else to do here, e.g. test if there are left over samples or some such? */
403
404	audioTestDriverStackStreamDestroy(pDrvStack, pStream->pStream);
405	pStream->pStream = NULL;
406	pStream->pBackend = NULL;
407	}
408
409	AudioTestMixStreamTerm(&pStream->Mix);
410
411	return VINF_SUCCESS;
412	}
413
414
415	/*********************************************************************************************************************************
416	* Test Primitives *
417	*********************************************************************************************************************************/
418
419	/**
420	* Initializes test tone parameters (partly with random values).
421
422	* @param pToneParms Test tone parameters to initialize.
423	*/
424	void audioTestToneParmsInit(PAUDIOTESTTONEPARMS pToneParms)
425	{
426	RT_BZERO(pToneParms, sizeof(AUDIOTESTTONEPARMS));
427
428	/**
429	* Set default (randomized) test tone parameters if not set explicitly.
430	*/
431	pToneParms->dbFreqHz = AudioTestToneGetRandomFreq();
432	pToneParms->msDuration = RTRandU32Ex(200, RT_MS_30SEC);
433	pToneParms->uVolumePercent = 100; /* We always go with maximum volume for now. */
434
435	PDMAudioPropsInit(&pToneParms->Props,
436	2 /* 16-bit /, true / fPcmSigned /, 2 / cPcmChannels /, 44100 / uPcmHz */);
437	}
438
439	/**
440	* Initializes I/O options with some sane default values.
441	*
442	* @param pIoOpts I/O options to initialize.
443	*/
444	void audioTestIoOptsInitDefaults(PAUDIOTESTIOOPTS pIoOpts)
445	{
446	RT_BZERO(pIoOpts, sizeof(AUDIOTESTIOOPTS));
447
448	/* Initialize the PCM properties to some sane values. */
449	PDMAudioPropsInit(&pIoOpts->Props,
450	2 /* 16-bit /, true / fPcmSigned /, 2 / cPcmChannels /, 44100 / uPcmHz */);
451
452	pIoOpts->cMsBufferSize = UINT32_MAX;
453	pIoOpts->cMsPreBuffer = UINT32_MAX;
454	pIoOpts->cMsSchedulingHint = UINT32_MAX;
455	pIoOpts->uVolumePercent = 100; /* Use maximum volume by default. */
456	}
457
458	#if 0 /* Unused */
459	/**
460	* Returns a random scheduling hint (in ms).
461	*/
462	DECLINLINE(uint32_t) audioTestEnvGetRandomSchedulingHint(void)
463	{
464	static const unsigned s_aSchedulingHintsMs[] =
465	{
466	10,
467	25,
468	50,
469	100,
470	200,
471	250
472	};
473
474	return s_aSchedulingHintsMs[RTRandU32Ex(0, RT_ELEMENTS(s_aSchedulingHintsMs) - 1)];
475	}
476	#endif
477
478	/**
479	* Plays a test tone on a specific audio test stream.
480	*
481	* @returns VBox status code.
482	* @param pIoOpts I/O options to use.
483	* @param pTstEnv Test environment to use for running the test.
484	* Optional and can be NULL (for simple playback only).
485	* @param pStream Stream to use for playing the tone.
486	* @param pParms Tone parameters to use.
487	*
488	* @note Blocking function.
489	*/
490	int audioTestPlayTone(PAUDIOTESTIOOPTS pIoOpts, PAUDIOTESTENV pTstEnv, PAUDIOTESTSTREAM pStream, PAUDIOTESTTONEPARMS pParms)
491	{
492	uint32_t const idxTest = (uint8_t)pParms->Hdr.idxTest;
493
494	AUDIOTESTTONE TstTone;
495	AudioTestToneInit(&TstTone, &pStream->Cfg.Props, pParms->dbFreqHz);
496
497	char const *pcszPathOut = NULL;
498	if (pTstEnv)
499	pcszPathOut = pTstEnv->Set.szPathAbs;
500
501	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Playing test tone (tone frequency is %RU16Hz, %RU32ms, %RU8%% volume)\n",
502	idxTest, (uint16_t)pParms->dbFreqHz, pParms->msDuration, pParms->uVolumePercent);
503	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Using %RU32ms stream scheduling hint\n",
504	idxTest, pStream->Cfg.Device.cMsSchedulingHint);
505	if (pcszPathOut)
506	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Writing to '%s'\n", idxTest, pcszPathOut);
507
508	int rc;
509
510	/** @todo Use .WAV here? */
511	AUDIOTESTOBJ Obj;
512	RT_ZERO(Obj); /* Shut up MSVC. */
513	if (pTstEnv)
514	{
515	rc = AudioTestSetObjCreateAndRegister(&pTstEnv->Set, "guest-tone-play.pcm", &Obj);
516	AssertRCReturn(rc, rc);
517	}
518
519	uint8_t const uVolPercent = pIoOpts->uVolumePercent;
520	int rc2 = audioTestSetMasterVolume(uVolPercent);
521	if (RT_FAILURE(rc2))
522	{
523	if (rc2 == VERR_NOT_SUPPORTED)
524	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Setting system's master volume is not supported on this platform, skipping\n");
525	else
526	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Setting system's master volume failed with %Rrc\n", rc2);
527	}
528	else
529	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Set system's master volume to %RU8%%\n", uVolPercent);
530
531	rc = AudioTestMixStreamEnable(&pStream->Mix);
532	if ( RT_SUCCESS(rc)
533	&& AudioTestMixStreamIsOkay(&pStream->Mix))
534	{
535	uint32_t cbToWriteTotal = PDMAudioPropsMilliToBytes(&pStream->Cfg.Props, pParms->msDuration);
536	AssertStmt(cbToWriteTotal, rc = VERR_INVALID_PARAMETER);
537	uint32_t cbWrittenTotal = 0;
538
539	/* We play a pre + post beacon before + after the actual test tone.
540	* We always start with the pre beacon. */
541	AUDIOTESTTONEBEACON Beacon;
542	AudioTestBeaconInit(&Beacon, (uint8_t)pParms->Hdr.idxTest, AUDIOTESTTONEBEACONTYPE_PLAY_PRE, &pStream->Cfg.Props);
543
544	uint32_t const cbBeacon = AudioTestBeaconGetSize(&Beacon);
545	if (cbBeacon)
546	{
547	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Playing 2 x %RU32 bytes pre/post beacons\n",
548	idxTest, cbBeacon);
549
550	if (g_uVerbosity >= 2)
551	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Playing %s beacon ...\n",
552	idxTest, AudioTestBeaconTypeGetName(Beacon.enmType));
553	}
554
555	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Playing %RU32 bytes total (%RU32ms timeout)\n",
556	idxTest, cbToWriteTotal, pTstEnv->msTimeout);
557
558	/* Failsafe if invalid timeout is set. */
559	if ( pTstEnv->msTimeout == 0
560	\|\| pTstEnv->msTimeout == UINT32_MAX)
561	{
562	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Warning! Invalid timeout set (%RU32ms), setting default\n",
563	idxTest, pTstEnv->msTimeout);
564	pTstEnv->msTimeout = AUDIOTEST_TIMEOUT_DEFAULT_MS;
565	}
566
567	AudioTestObjAddMetadataStr(Obj, "test_id=%04RU32\n", pParms->Hdr.idxTest);
568	AudioTestObjAddMetadataStr(Obj, "beacon_type=%RU32\n", (uint32_t)AudioTestBeaconGetType(&Beacon));
569	AudioTestObjAddMetadataStr(Obj, "beacon_pre_bytes=%RU32\n", cbBeacon);
570	AudioTestObjAddMetadataStr(Obj, "beacon_post_bytes=%RU32\n", cbBeacon);
571	AudioTestObjAddMetadataStr(Obj, "stream_to_write_total_bytes=%RU32\n", cbToWriteTotal);
572	AudioTestObjAddMetadataStr(Obj, "stream_period_size_frames=%RU32\n", pStream->Cfg.Backend.cFramesPeriod);
573	AudioTestObjAddMetadataStr(Obj, "stream_buffer_size_frames=%RU32\n", pStream->Cfg.Backend.cFramesBufferSize);
574	AudioTestObjAddMetadataStr(Obj, "stream_prebuf_size_frames=%RU32\n", pStream->Cfg.Backend.cFramesPreBuffering);
575	/* Note: This mostly is provided by backend (e.g. PulseAudio / ALSA / ++) and
576	* has nothing to do with the device emulation scheduling hint. */
577	AudioTestObjAddMetadataStr(Obj, "device_scheduling_hint_ms=%RU32\n", pStream->Cfg.Device.cMsSchedulingHint);
578
579	PAUDIOTESTDRVMIXSTREAM pMix = &pStream->Mix;
580
581	uint32_t const cbPreBuffer = PDMAudioPropsFramesToBytes(pMix->pProps, pStream->Cfg.Backend.cFramesPreBuffering);
582	uint64_t const nsStarted = RTTimeNanoTS();
583	uint64_t nsDonePreBuffering = 0;
584
585	uint64_t offStream = 0;
586	uint64_t nsTimeout = uint64_t(pTstEnv->msTimeout) * RT_NS_1MS_64;
587	uint64_t nsLastMsgCantWrite = 0; /* Timestamp (in ns) when the last message of an unwritable stream was shown. */
588	uint64_t nsLastWrite = 0;
589
590	AUDIOTESTSTATE enmState = AUDIOTESTSTATE_PRE;
591	uint8_t abBuf[_16K];
592
593	for (;;)
594	{
595	uint64_t const nsNow = RTTimeNanoTS();
596	if (!nsLastWrite)
597	nsLastWrite = nsNow;
598
599	/* Pace ourselves a little. */
600	if (offStream >= cbPreBuffer)
601	{
602	if (!nsDonePreBuffering)
603	nsDonePreBuffering = nsNow;
604	uint64_t const cNsWritten = PDMAudioPropsBytesToNano64(pMix->pProps, offStream - cbPreBuffer);
605	uint64_t const cNsElapsed = nsNow - nsStarted;
606	if (cNsWritten > cNsElapsed + RT_NS_10MS)
607	RTThreadSleep((cNsWritten - cNsElapsed - RT_NS_10MS / 2) / RT_NS_1MS);
608	}
609
610	uint32_t cbWritten = 0;
611	uint32_t const cbCanWrite = AudioTestMixStreamGetWritable(&pStream->Mix);
612	if (cbCanWrite)
613	{
614	if (g_uVerbosity >= 4)
615	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Stream is writable with %RU64ms (%RU32 bytes)\n",
616	idxTest, PDMAudioPropsBytesToMilli(pMix->pProps, cbCanWrite), cbCanWrite);
617
618	switch (enmState)
619	{
620	case AUDIOTESTSTATE_PRE:
621	RT_FALL_THROUGH();
622	case AUDIOTESTSTATE_POST:
623	{
624	if (g_uVerbosity >= 4)
625	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: %RU32 bytes (%RU64ms) beacon data remaining\n",
626	idxTest, AudioTestBeaconGetRemaining(&Beacon),
627	PDMAudioPropsBytesToMilli(&pStream->pStream->Cfg.Props, AudioTestBeaconGetRemaining(&Beacon)));
628
629	bool fGoToNextStage = false;
630
631	if ( AudioTestBeaconGetSize(&Beacon)
632	&& !AudioTestBeaconIsComplete(&Beacon))
633	{
634	bool const fStarted = AudioTestBeaconGetRemaining(&Beacon) == AudioTestBeaconGetSize(&Beacon);
635
636	uint32_t const cbBeaconRemaining = AudioTestBeaconGetRemaining(&Beacon);
637	AssertBreakStmt(cbBeaconRemaining, rc = VERR_WRONG_ORDER);
638
639	/* Limit to exactly one beacon (pre or post). */
640	uint32_t const cbToWrite = RT_MIN(sizeof(abBuf), RT_MIN(cbCanWrite, cbBeaconRemaining));
641
642	rc = AudioTestBeaconWrite(&Beacon, abBuf, cbToWrite);
643	if (RT_SUCCESS(rc))
644	{
645	rc = AudioTestMixStreamPlay(&pStream->Mix, abBuf, cbToWrite, &cbWritten);
646	if ( RT_SUCCESS(rc)
647	&& pTstEnv)
648	{
649	/* Also write the beacon data to the test object.
650	* Note: We use cbPlayed here instead of cbToPlay to know if the data actually was
651	* reported as being played by the audio stack. */
652	rc = AudioTestObjWrite(Obj, abBuf, cbWritten);
653	}
654	}
655
656	if ( fStarted
657	&& g_uVerbosity >= 2)
658	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Writing %s beacon begin\n",
659	idxTest, AudioTestBeaconTypeGetName(Beacon.enmType));
660	if (AudioTestBeaconIsComplete(&Beacon))
661	{
662	if (g_uVerbosity >= 2)
663	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Writing %s beacon end\n",
664	idxTest, AudioTestBeaconTypeGetName(Beacon.enmType));
665	fGoToNextStage = true;
666	}
667	}
668	else
669	fGoToNextStage = true;
670
671	if (fGoToNextStage)
672	{
673	if (enmState == AUDIOTESTSTATE_PRE)
674	enmState = AUDIOTESTSTATE_RUN;
675	else if (enmState == AUDIOTESTSTATE_POST)
676	enmState = AUDIOTESTSTATE_DONE;
677	}
678	break;
679	}
680
681	case AUDIOTESTSTATE_RUN:
682	{
683	uint32_t cbToWrite = RT_MIN(sizeof(abBuf), cbCanWrite);
684	cbToWrite = RT_MIN(cbToWrite, cbToWriteTotal - cbWrittenTotal);
685
686	if (g_uVerbosity >= 4)
687	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS,
688	"Test #%RU32: Playing back %RU32 bytes\n", idxTest, cbToWrite);
689
690	if (cbToWrite)
691	{
692	rc = AudioTestToneGenerate(&TstTone, abBuf, cbToWrite, &cbToWrite);
693	if (RT_SUCCESS(rc))
694	{
695	if (pTstEnv)
696	{
697	/* Write stuff to disk before trying to play it. Helps analysis later. */
698	rc = AudioTestObjWrite(Obj, abBuf, cbToWrite);
699	}
700
701	if (RT_SUCCESS(rc))
702	{
703	rc = AudioTestMixStreamPlay(&pStream->Mix, abBuf, cbToWrite, &cbWritten);
704	if (RT_SUCCESS(rc))
705	{
706	AssertBreakStmt(cbWritten <= cbToWrite, rc = VERR_TOO_MUCH_DATA);
707
708	offStream += cbWritten;
709
710	if (cbWritten != cbToWrite)
711	RTTestFailed(g_hTest, "Test #%RU32: Only played %RU32/%RU32 bytes\n",
712	idxTest, cbWritten, cbToWrite);
713
714	if (cbWritten)
715	nsLastWrite = nsNow;
716
717	if (g_uVerbosity >= 4)
718	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS,
719	"Test #%RU32: Played back %RU32 bytes\n", idxTest, cbWritten);
720
721	cbWrittenTotal += cbWritten;
722	}
723	}
724	}
725	}
726
727	if (RT_SUCCESS(rc))
728	{
729	const bool fComplete = cbWrittenTotal >= cbToWriteTotal;
730	if (fComplete)
731	{
732	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Playing back audio data ended\n", idxTest);
733
734	enmState = AUDIOTESTSTATE_POST;
735
736	/* Re-use the beacon object, but this time it's the post beacon. */
737	AudioTestBeaconInit(&Beacon, (uint8_t)idxTest, AUDIOTESTTONEBEACONTYPE_PLAY_POST,
738	&pStream->Cfg.Props);
739	}
740	}
741	else
742	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Playing back failed with %Rrc\n", idxTest, rc);
743	break;
744	}
745
746	case AUDIOTESTSTATE_DONE:
747	{
748	/* Handled below. */
749	break;
750	}
751
752	default:
753	AssertFailed();
754	break;
755	}
756
757	if (RT_FAILURE(rc))
758	break;
759
760	if (enmState == AUDIOTESTSTATE_DONE)
761	break;
762
763	nsLastMsgCantWrite = 0;
764	}
765	else if (AudioTestMixStreamIsOkay(&pStream->Mix))
766	{
767	RTMSINTERVAL const msSleep = RT_MIN(RT_MAX(1, pStream->Cfg.Device.cMsSchedulingHint), 256);
768
769	if ( g_uVerbosity >= 3
770	&& ( !nsLastMsgCantWrite
771	\|\| (nsNow - nsLastMsgCantWrite) > RT_NS_10SEC)) /* Don't spam the output too much. */
772	{
773	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Waiting %RU32ms for stream to be writable again (last write %RU64ns ago) ...\n",
774	idxTest, msSleep, nsNow - nsLastWrite);
775	nsLastMsgCantWrite = nsNow;
776	}
777
778	RTThreadSleep(msSleep);
779	}
780	else
781	AssertFailedBreakStmt(rc = VERR_AUDIO_STREAM_NOT_READY);
782
783	/* Fail-safe in case something screwed up while playing back. */
784	uint64_t const cNsElapsed = nsNow - nsStarted;
785	if (cNsElapsed > nsTimeout)
786	{
787	RTTestFailed(g_hTest, "Test #%RU32: Playback took too long (running %RU64 vs. timeout %RU64), aborting\n",
788	idxTest, cNsElapsed, nsTimeout);
789	rc = VERR_TIMEOUT;
790	}
791
792	if (RT_FAILURE(rc))
793	break;
794	} /* for */
795
796	if (cbWrittenTotal != cbToWriteTotal)
797	{
798	RTTestFailed(g_hTest, "Test #%RU32: Playback ended unexpectedly (%RU32 played, expected %RU32)\n",
799	idxTest, cbWrittenTotal, cbToWriteTotal);
800	rc = cbWrittenTotal > cbToWriteTotal ? VERR_BUFFER_OVERFLOW : VERR_BUFFER_UNDERFLOW;
801	}
802
803	if (RT_SUCCESS(rc))
804	{
805	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Draining stream ...\n", idxTest);
806	rc = AudioTestMixStreamDrain(&pStream->Mix, true /fSync/);
807	}
808	}
809	else
810	rc = VERR_AUDIO_STREAM_NOT_READY;
811
812	if (pTstEnv)
813	{
814	rc2 = AudioTestObjClose(Obj);
815	if (RT_SUCCESS(rc))
816	rc = rc2;
817	}
818
819	if (RT_FAILURE(rc))
820	RTTestFailed(g_hTest, "Test #%RU32: Playing tone failed with %Rrc\n", idxTest, rc);
821
822	return rc;
823	}
824
825	/**
826	* Records a test tone from a specific audio test stream.
827	*
828	* @returns VBox status code.
829	* @param pIoOpts I/O options to use.
830	* @param pTstEnv Test environment to use for running the test.
831	* @param pStream Stream to use for recording the tone.
832	* @param pParms Tone parameters to use.
833	*
834	* @note Blocking function.
835	*/
836	static int audioTestRecordTone(PAUDIOTESTIOOPTS pIoOpts, PAUDIOTESTENV pTstEnv, PAUDIOTESTSTREAM pStream, PAUDIOTESTTONEPARMS pParms)
837	{
838	uint32_t const idxTest = (uint8_t)pParms->Hdr.idxTest;
839
840	const char *pcszPathOut = pTstEnv->Set.szPathAbs;
841
842	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Recording test tone (tone frequency is %RU16Hz, %RU32ms)\n",
843	idxTest, (uint16_t)pParms->dbFreqHz, pParms->msDuration);
844	RTTestPrintf(g_hTest, RTTESTLVL_DEBUG, "Test #%RU32: Writing to '%s'\n", idxTest, pcszPathOut);
845
846	/** @todo Use .WAV here? */
847	AUDIOTESTOBJ Obj;
848	int rc = AudioTestSetObjCreateAndRegister(&pTstEnv->Set, "guest-tone-rec.pcm", &Obj);
849	AssertRCReturn(rc, rc);
850
851	PAUDIOTESTDRVMIXSTREAM pMix = &pStream->Mix;
852
853	rc = AudioTestMixStreamEnable(pMix);
854	if (RT_SUCCESS(rc))
855	{
856	size_t cbToReadTotal = PDMAudioPropsMilliToBytes(&pStream->Cfg.Props, pParms->msDuration);
857	AssertStmt(cbToReadTotal, rc = VERR_INVALID_PARAMETER);
858	size_t cbReadTotal = 0; /* Counts the read test tone data (w/o any beacons). */
859
860	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Recording %RU32 bytes total (%RU32ms timeout)\n",
861	idxTest, cbToReadTotal, pTstEnv->msTimeout);
862
863	/* Failsafe if invalid timeout is set. */
864	if ( pTstEnv->msTimeout == 0
865	\|\| pTstEnv->msTimeout == UINT32_MAX)
866	{
867	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Warning! Invalid timeout set (%RU32ms), setting default\n",
868	idxTest, pTstEnv->msTimeout);
869	pTstEnv->msTimeout = AUDIOTEST_TIMEOUT_DEFAULT_MS;
870	}
871
872	/* We expect a pre + post beacon before + after the actual test tone.
873	* We always start with the pre beacon. */
874	AUDIOTESTTONEBEACON Beacon;
875	AudioTestBeaconInit(&Beacon, (uint8_t)pParms->Hdr.idxTest, AUDIOTESTTONEBEACONTYPE_PLAY_PRE, &pStream->Cfg.Props);
876
877	uint32_t const cbBeacon = AudioTestBeaconGetSize(&Beacon);
878	if (cbBeacon)
879	{
880	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Expecting 2 x %RU32 bytes pre/post beacons\n",
881	idxTest, cbBeacon);
882	if (g_uVerbosity >= 2)
883	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Waiting for %s beacon ...\n",
884	idxTest, AudioTestBeaconTypeGetName(Beacon.enmType));
885	}
886
887	AudioTestObjAddMetadataStr(Obj, "test_id=%04RU32\n", pParms->Hdr.idxTest);
888	AudioTestObjAddMetadataStr(Obj, "beacon_type=%RU32\n", (uint32_t)AudioTestBeaconGetType(&Beacon));
889	AudioTestObjAddMetadataStr(Obj, "beacon_pre_bytes=%RU32\n", cbBeacon);
890	AudioTestObjAddMetadataStr(Obj, "beacon_post_bytes=%RU32\n", cbBeacon);
891	AudioTestObjAddMetadataStr(Obj, "stream_to_record_bytes=%RU32\n", cbToReadTotal);
892	AudioTestObjAddMetadataStr(Obj, "stream_buffer_size_ms=%RU32\n", pIoOpts->cMsBufferSize);
893	AudioTestObjAddMetadataStr(Obj, "stream_prebuf_size_ms=%RU32\n", pIoOpts->cMsPreBuffer);
894	/* Note: This mostly is provided by backend (e.g. PulseAudio / ALSA / ++) and
895	* has nothing to do with the device emulation scheduling hint. */
896	AudioTestObjAddMetadataStr(Obj, "device_scheduling_hint_ms=%RU32\n", pIoOpts->cMsSchedulingHint);
897
898	PRTCIRCBUF pCircBuf;
899	rc = RTCircBufCreate(&pCircBuf, _64K);
900	AssertRCReturn(rc, rc);
901
902	uint64_t const nsStarted = RTTimeNanoTS();
903
904	uint64_t nsTimeout = uint64_t(pTstEnv->msTimeout) * RT_NS_1MS_64;
905	uint64_t nsLastMsgCantRead = 0; /* Timestamp (in ns) when the last message of an unreadable stream was shown. */
906
907	AUDIOTESTSTATE enmState = AUDIOTESTSTATE_PRE;
908
909	while (!g_fTerminate)
910	{
911	uint64_t const nsNow = RTTimeNanoTS();
912
913	void *pvBlock;
914	size_t cbBlock;
915
916	/*
917	* Anything we can capture from the stream?
918	*/
919	uint32_t const cbCanRead = AudioTestMixStreamGetReadable(pMix);
920	if (cbCanRead)
921	{
922	RTCircBufAcquireWriteBlock(pCircBuf, cbCanRead, &pvBlock, &cbBlock);
923
924	uint32_t cbCaptured = 0;
925	if (cbBlock)
926	{
927	rc = AudioTestMixStreamCapture(pMix, pvBlock, (uint32_t)cbBlock, &cbCaptured);
928	if (RT_FAILURE(rc))
929	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Reading from stream failed with %Rrc\n", idxTest, rc);
930
931	if (g_uVerbosity >= 2)
932	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Stream is readable with %RU64ms (%RU32 bytes), captured %RU64ms (%RU32 bytes)\n",
933	idxTest,
934	PDMAudioPropsBytesToMilli(pMix->pProps, cbCanRead), cbCanRead,
935	PDMAudioPropsBytesToMilli(pMix->pProps, cbCaptured), cbCaptured);
936
937	if (RT_FAILURE(rc))
938	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Capturing failed with %Rrc\n",
939	idxTest, rc);
940	}
941
942	RTCircBufReleaseWriteBlock(pCircBuf, cbCaptured);
943	}
944	else if (AudioTestMixStreamIsOkay(pMix))
945	{
946	RTMSINTERVAL const msSleep = RT_MIN(RT_MAX(1, pStream->Cfg.Device.cMsSchedulingHint), 256);
947
948	if ( g_uVerbosity >= 3
949	&& ( !nsLastMsgCantRead
950	\|\| (nsNow - nsLastMsgCantRead) > RT_NS_10SEC)) /* Don't spam the output too much. */
951	{
952	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Waiting %RU32ms for stream to be readable again ...\n",
953	idxTest, msSleep);
954	nsLastMsgCantRead = nsNow;
955	}
956
957	RTThreadSleep(msSleep);
958	}
959
960	/*
961	* Process our buffer.
962	*/
963	size_t cbBlockToAcq;
964	if (pTstEnv->fSelftest) /* For self-test mode we want to have a bit more randomness. */
965	{
966	size_t const u = (uint32_t)RTCircBufUsed(pCircBuf);
967	size_t const r = RTRandU32Ex(1, (uint32_t)RTCircBufSize(pCircBuf));
968	cbBlockToAcq = PDMAudioPropsFloorBytesToFrame(pMix->pProps, RT_MIN((uint32_t)u, (uint32_t)r));
969	}
970	else
971	cbBlockToAcq = PDMAudioPropsFloorBytesToFrame(pMix->pProps, (uint32_t)RTCircBufUsed(pCircBuf));
972
973	RTCircBufAcquireReadBlock(pCircBuf, cbBlockToAcq, &pvBlock, &cbBlock);
974	if (!cbBlock)
975	continue;
976
977	/* Flag indicating whether the whole block we've captured is silence or not. */
978	bool const fIsAllSilence = PDMAudioPropsIsBufferSilence(&pStream->pStream->Cfg.Props, pvBlock, cbBlock);
979	size_t cbRead = 0;
980
981	switch (enmState)
982	{
983	case AUDIOTESTSTATE_PRE:
984	RT_FALL_THROUGH();
985	case AUDIOTESTSTATE_POST:
986	{
987	bool fGoToNextStage = false;
988
989	if ( AudioTestBeaconGetSize(&Beacon)
990	&& !AudioTestBeaconIsComplete(&Beacon))
991	{
992	bool const fStarted = AudioTestBeaconGetRemaining(&Beacon) == AudioTestBeaconGetSize(&Beacon);
993
994	size_t uOff;
995	int rc2 = AudioTestBeaconAddConsecutive(&Beacon, (uint8_t *)pvBlock, cbBlock, &uOff);
996	if ( RT_SUCCESS(rc2)
997	&& uOff)
998	cbRead = uOff;
999
1000	if ( fStarted
1001	&& g_uVerbosity >= 2)
1002	{
1003	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS,
1004	"Test #%RU32: Detection of %s beacon started (%RU64ms recorded so far)\n",
1005	idxTest, AudioTestBeaconTypeGetName(Beacon.enmType),
1006	PDMAudioPropsBytesToMilli(&pStream->pStream->Cfg.Props, (uint32_t)cbReadTotal));
1007	}
1008
1009	if (AudioTestBeaconIsComplete(&Beacon))
1010	{
1011	if (g_uVerbosity >= 2)
1012	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Detection of %s beacon complete\n",
1013	idxTest, AudioTestBeaconTypeGetName(Beacon.enmType));
1014	fGoToNextStage = true;
1015	}
1016	}
1017	else
1018	fGoToNextStage = true;
1019
1020	if (fGoToNextStage)
1021	{
1022	if (enmState == AUDIOTESTSTATE_PRE)
1023	enmState = AUDIOTESTSTATE_RUN;
1024	else if (enmState == AUDIOTESTSTATE_POST)
1025	enmState = AUDIOTESTSTATE_DONE;
1026	}
1027	break;
1028	}
1029
1030	case AUDIOTESTSTATE_RUN:
1031	{
1032	/* Whether we count all silence as recorded data or not.
1033	* Currently we don't, as otherwise consequtively played tones will be cut off in the end. */
1034	bool const fRecord = !fIsAllSilence;
1035	if (fRecord)
1036	{
1037	/* Don't read more than we're told to.
1038	* After the actual test tone data there might come a post beacon which also
1039	* needs to be handled in the AUDIOTESTSTATE_POST state then. */
1040	if (cbReadTotal + cbBlock > cbToReadTotal)
1041	{
1042	AssertBreakStmt(cbToReadTotal >= cbReadTotal, rc = VERR_INTERNAL_ERROR);
1043	cbRead = cbToReadTotal - cbReadTotal;
1044	}
1045	else
1046	cbRead = cbBlock;
1047
1048	cbReadTotal += cbRead;
1049	AssertBreakStmt(cbReadTotal <= cbToReadTotal, rc = VERR_INTERNAL_ERROR);
1050	}
1051
1052	/* Done recording the test tone? */
1053	if (cbReadTotal + cbRead == cbToReadTotal)
1054	{
1055	enmState = AUDIOTESTSTATE_POST;
1056
1057	if (g_uVerbosity >= 2)
1058	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Recording tone data done\n", idxTest);
1059
1060	if (AudioTestBeaconGetSize(&Beacon))
1061	{
1062	/* Re-use the beacon object, but this time it's the post beacon. */
1063	AudioTestBeaconInit(&Beacon, (uint8_t)pParms->Hdr.idxTest, AUDIOTESTTONEBEACONTYPE_PLAY_POST,
1064	&pStream->Cfg.Props);
1065	if (g_uVerbosity >= 2)
1066	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS,
1067	"Test #%RU32: Waiting for %s beacon ...\n",
1068	idxTest, AudioTestBeaconTypeGetName(Beacon.enmType));
1069	}
1070	}
1071	break;
1072	}
1073
1074	case AUDIOTESTSTATE_DONE:
1075	{
1076	/* Nothing to do here. */
1077	break;
1078	}
1079
1080	default:
1081	AssertFailed();
1082	break;
1083	}
1084
1085	if (cbRead)
1086	{
1087	if (g_uVerbosity >= 3)
1088	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Read data (%zu bytes):\n"
1089	"%.*Rhxd\n",
1090	idxTest, cbRead, cbRead, pvBlock);
1091
1092	/* Always write (record) everything, no matter if the current audio contains complete silence or not.
1093	* Might be also become handy later if we want to have a look at start/stop timings and so on. */
1094	rc = AudioTestObjWrite(Obj, pvBlock, cbRead);
1095	AssertRCBreak(rc);
1096	}
1097
1098	if (g_uVerbosity >= 2)
1099	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Processed %RU64ms (%zu bytes)\n",
1100	idxTest, PDMAudioPropsBytesToMilli(pMix->pProps, (uint32_t)cbRead), cbRead);
1101
1102	RTCircBufReleaseReadBlock(pCircBuf, cbRead);
1103
1104	if (enmState == AUDIOTESTSTATE_DONE) /* Bail out when in state "done". */
1105	break;
1106
1107	/* Fail-safe in case something screwed up while playing back. */
1108	uint64_t const cNsElapsed = nsNow - nsStarted;
1109	if (cNsElapsed > nsTimeout)
1110	{
1111	RTTestFailed(g_hTest, "Test #%RU32: Recording took too long (running %RU64 vs. timeout %RU64), aborting\n",
1112	idxTest, cNsElapsed, nsTimeout);
1113	rc = VERR_TIMEOUT;
1114	}
1115
1116	if (RT_FAILURE(rc))
1117	break;
1118	} /* while */
1119
1120	if (g_uVerbosity >= 2)
1121	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test #%RU32: Recorded %zu bytes total\n", idxTest, cbReadTotal);
1122	if (cbReadTotal != cbToReadTotal)
1123	{
1124	RTTestFailed(g_hTest, "Test #%RU32: Recording ended unexpectedly (%zu read, expected %zu)\n",
1125	idxTest, cbReadTotal, cbToReadTotal);
1126	int rc2 = cbReadTotal > cbToReadTotal ? VERR_BUFFER_OVERFLOW : VERR_BUFFER_UNDERFLOW;
1127	if (RT_SUCCESS(rc))
1128	rc = rc2;
1129	}
1130
1131	if (RT_FAILURE(rc))
1132	RTTestFailed(g_hTest, "Test #%RU32: Recording failed (state is '%s')\n", idxTest, AudioTestStateToStr(enmState));
1133
1134	int rc2 = AudioTestMixStreamDisable(pMix);
1135	if (RT_SUCCESS(rc))
1136	rc = rc2;
1137
1138	RTCircBufDestroy(pCircBuf);
1139	}
1140
1141	int rc2 = AudioTestObjClose(Obj);
1142	if (RT_SUCCESS(rc))
1143	rc = rc2;
1144
1145	if (RT_FAILURE(rc))
1146	RTTestFailed(g_hTest, "Test #%RU32: Recording tone done failed with %Rrc\n", idxTest, rc);
1147
1148	return rc;
1149	}
1150
1151
1152	/*********************************************************************************************************************************
1153	* ATS Callback Implementations *
1154	*********************************************************************************************************************************/
1155
1156	/** @copydoc ATSCALLBACKS::pfnHowdy
1157	*
1158	* @note Runs as part of the guest ATS.
1159	*/
1160	static DECLCALLBACK(int) audioTestGstAtsHowdyCallback(void const *pvUser)
1161	{
1162	PATSCALLBACKCTX pCtx = (PATSCALLBACKCTX)pvUser;
1163
1164	AssertReturn(pCtx->cClients <= UINT8_MAX - 1, VERR_BUFFER_OVERFLOW);
1165
1166	pCtx->cClients++;
1167
1168	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "New client connected, now %RU8 total\n", pCtx->cClients);
1169
1170	return VINF_SUCCESS;
1171	}
1172
1173	/** @copydoc ATSCALLBACKS::pfnBye
1174	*
1175	* @note Runs as part of the guest ATS.
1176	*/
1177	static DECLCALLBACK(int) audioTestGstAtsByeCallback(void const *pvUser)
1178	{
1179	PATSCALLBACKCTX pCtx = (PATSCALLBACKCTX)pvUser;
1180
1181	AssertReturn(pCtx->cClients, VERR_WRONG_ORDER);
1182	pCtx->cClients--;
1183
1184	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Client wants to disconnect, %RU8 remaining\n", pCtx->cClients);
1185
1186	if (0 == pCtx->cClients) /* All clients disconnected? Tear things down. */
1187	{
1188	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Last client disconnected, terminating server ...\n");
1189	ASMAtomicWriteBool(&g_fTerminate, true);
1190	}
1191
1192	return VINF_SUCCESS;
1193	}
1194
1195	/** @copydoc ATSCALLBACKS::pfnTestSetBegin
1196	*
1197	* @note Runs as part of the guest ATS.
1198	*/
1199	static DECLCALLBACK(int) audioTestGstAtsTestSetBeginCallback(void const pvUser, const char pszTag)
1200	{
1201	PATSCALLBACKCTX pCtx = (PATSCALLBACKCTX)pvUser;
1202	PAUDIOTESTENV pTstEnv = pCtx->pTstEnv;
1203
1204	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Got request for beginning test set '%s' in '%s'\n", pszTag, pTstEnv->szPathTemp);
1205
1206	return AudioTestSetCreate(&pTstEnv->Set, pTstEnv->szPathTemp, pszTag);
1207	}
1208
1209	/** @copydoc ATSCALLBACKS::pfnTestSetEnd
1210	*
1211	* @note Runs as part of the guest ATS.
1212	*/
1213	static DECLCALLBACK(int) audioTestGstAtsTestSetEndCallback(void const pvUser, const char pszTag)
1214	{
1215	PATSCALLBACKCTX pCtx = (PATSCALLBACKCTX)pvUser;
1216	PAUDIOTESTENV pTstEnv = pCtx->pTstEnv;
1217
1218	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Got request for ending test set '%s'\n", pszTag);
1219
1220	/* Pack up everything to be ready for transmission. */
1221	return audioTestEnvPrologue(pTstEnv, true /* fPack */, pCtx->szTestSetArchive, sizeof(pCtx->szTestSetArchive));
1222	}
1223
1224	/** @copydoc ATSCALLBACKS::pfnTonePlay
1225	*
1226	* @note Runs as part of the guest ATS.
1227	*/
1228	static DECLCALLBACK(int) audioTestGstAtsTonePlayCallback(void const *pvUser, PAUDIOTESTTONEPARMS pToneParms)
1229	{
1230	PATSCALLBACKCTX pCtx = (PATSCALLBACKCTX)pvUser;
1231	PAUDIOTESTENV pTstEnv = pCtx->pTstEnv;
1232	PAUDIOTESTIOOPTS pIoOpts = &pTstEnv->IoOpts;
1233
1234	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Got request for playing test tone #%RU32 (%RU16Hz, %RU32ms) ...\n",
1235	pToneParms->Hdr.idxTest, (uint16_t)pToneParms->dbFreqHz, pToneParms->msDuration);
1236
1237	char szTimeCreated[RTTIME_STR_LEN];
1238	RTTimeToString(&pToneParms->Hdr.tsCreated, szTimeCreated, sizeof(szTimeCreated));
1239	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test created (caller UTC): %s\n", szTimeCreated);
1240
1241	const PAUDIOTESTSTREAM pTstStream = &pTstEnv->aStreams[0]; /** @todo Make this dynamic. */
1242
1243	int rc = audioTestStreamInit(pTstEnv->pDrvStack, pTstStream, PDMAUDIODIR_OUT, pIoOpts);
1244	if (RT_SUCCESS(rc))
1245	{
1246	AUDIOTESTPARMS TstParms;
1247	RT_ZERO(TstParms);
1248	TstParms.enmType = AUDIOTESTTYPE_TESTTONE_PLAY;
1249	TstParms.enmDir = PDMAUDIODIR_OUT;
1250	TstParms.TestTone = *pToneParms;
1251
1252	PAUDIOTESTENTRY pTst;
1253	rc = AudioTestSetTestBegin(&pTstEnv->Set, "Playing test tone", &TstParms, &pTst);
1254	if (RT_SUCCESS(rc))
1255	{
1256	rc = audioTestPlayTone(&pTstEnv->IoOpts, pTstEnv, pTstStream, pToneParms);
1257	if (RT_SUCCESS(rc))
1258	{
1259	AudioTestSetTestDone(pTst);
1260	}
1261	else
1262	AudioTestSetTestFailed(pTst, rc, "Playing tone failed");
1263	}
1264
1265	int rc2 = audioTestStreamDestroy(pTstEnv->pDrvStack, pTstStream);
1266	if (RT_SUCCESS(rc))
1267	rc = rc2;
1268	}
1269	else
1270	RTTestFailed(g_hTest, "Error creating output stream, rc=%Rrc\n", rc);
1271
1272	return rc;
1273	}
1274
1275	/** @copydoc ATSCALLBACKS::pfnToneRecord */
1276	static DECLCALLBACK(int) audioTestGstAtsToneRecordCallback(void const *pvUser, PAUDIOTESTTONEPARMS pToneParms)
1277	{
1278	PATSCALLBACKCTX pCtx = (PATSCALLBACKCTX)pvUser;
1279	PAUDIOTESTENV pTstEnv = pCtx->pTstEnv;
1280	PAUDIOTESTIOOPTS pIoOpts = &pTstEnv->IoOpts;
1281
1282	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Got request for recording test tone #%RU32 (%RU32ms) ...\n",
1283	pToneParms->Hdr.idxTest, pToneParms->msDuration);
1284
1285	char szTimeCreated[RTTIME_STR_LEN];
1286	RTTimeToString(&pToneParms->Hdr.tsCreated, szTimeCreated, sizeof(szTimeCreated));
1287	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test created (caller UTC): %s\n", szTimeCreated);
1288
1289	const PAUDIOTESTSTREAM pTstStream = &pTstEnv->aStreams[0]; /** @todo Make this dynamic. */
1290
1291	int rc = audioTestStreamInit(pTstEnv->pDrvStack, pTstStream, PDMAUDIODIR_IN, pIoOpts);
1292	if (RT_SUCCESS(rc))
1293	{
1294	AUDIOTESTPARMS TstParms;
1295	RT_ZERO(TstParms);
1296	TstParms.enmType = AUDIOTESTTYPE_TESTTONE_RECORD;
1297	TstParms.enmDir = PDMAUDIODIR_IN;
1298	TstParms.TestTone = *pToneParms;
1299
1300	PAUDIOTESTENTRY pTst;
1301	rc = AudioTestSetTestBegin(&pTstEnv->Set, "Recording test tone from host", &TstParms, &pTst);
1302	if (RT_SUCCESS(rc))
1303	{
1304	rc = audioTestRecordTone(pIoOpts, pTstEnv, pTstStream, pToneParms);
1305	if (RT_SUCCESS(rc))
1306	{
1307	AudioTestSetTestDone(pTst);
1308	}
1309	else
1310	AudioTestSetTestFailed(pTst, rc, "Recording tone failed");
1311	}
1312
1313	int rc2 = audioTestStreamDestroy(pTstEnv->pDrvStack, pTstStream);
1314	if (RT_SUCCESS(rc))
1315	rc = rc2;
1316	}
1317	else
1318	RTTestFailed(g_hTest, "Error creating input stream, rc=%Rrc\n", rc);
1319
1320	return rc;
1321	}
1322
1323	/** @copydoc ATSCALLBACKS::pfnTestSetSendBegin */
1324	static DECLCALLBACK(int) audioTestGstAtsTestSetSendBeginCallback(void const pvUser, const char pszTag)
1325	{
1326	RT_NOREF(pszTag);
1327
1328	PATSCALLBACKCTX pCtx = (PATSCALLBACKCTX)pvUser;
1329
1330	if (!RTFileExists(pCtx->szTestSetArchive)) /* Has the archive successfully been created yet? */
1331	return VERR_WRONG_ORDER;
1332
1333	int rc = RTFileOpen(&pCtx->hTestSetArchive, pCtx->szTestSetArchive, RTFILE_O_READ \| RTFILE_O_OPEN \| RTFILE_O_DENY_WRITE);
1334	if (RT_SUCCESS(rc))
1335	{
1336	uint64_t uSize;
1337	rc = RTFileQuerySize(pCtx->hTestSetArchive, &uSize);
1338	if (RT_SUCCESS(rc))
1339	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Sending test set '%s' (%zu bytes)\n", pCtx->szTestSetArchive, uSize);
1340	}
1341
1342	return rc;
1343	}
1344
1345	/** @copydoc ATSCALLBACKS::pfnTestSetSendRead */
1346	static DECLCALLBACK(int) audioTestGstAtsTestSetSendReadCallback(void const *pvUser,
1347	const char pszTag, void pvBuf, size_t cbBuf, size_t *pcbRead)
1348	{
1349	RT_NOREF(pszTag);
1350
1351	PATSCALLBACKCTX pCtx = (PATSCALLBACKCTX)pvUser;
1352
1353	return RTFileRead(pCtx->hTestSetArchive, pvBuf, cbBuf, pcbRead);
1354	}
1355
1356	/** @copydoc ATSCALLBACKS::pfnTestSetSendEnd */
1357	static DECLCALLBACK(int) audioTestGstAtsTestSetSendEndCallback(void const pvUser, const char pszTag)
1358	{
1359	RT_NOREF(pszTag);
1360
1361	PATSCALLBACKCTX pCtx = (PATSCALLBACKCTX)pvUser;
1362
1363	int rc = RTFileClose(pCtx->hTestSetArchive);
1364	if (RT_SUCCESS(rc))
1365	{
1366	pCtx->hTestSetArchive = NIL_RTFILE;
1367	}
1368
1369	return rc;
1370	}
1371
1372
1373	/*********************************************************************************************************************************
1374	* Implementation of audio test environment handling *
1375	*********************************************************************************************************************************/
1376
1377	/**
1378	* Connects an ATS client via TCP/IP to a peer.
1379	*
1380	* @returns VBox status code.
1381	* @param pTstEnv Test environment to use.
1382	* @param pClient Client to connect.
1383	* @param pszWhat Hint of what to connect to where.
1384	* @param pTcpOpts Pointer to TCP options to use.
1385	*/
1386	static int audioTestEnvConnectViaTcp(PAUDIOTESTENV pTstEnv, PATSCLIENT pClient, const char *pszWhat, PAUDIOTESTENVTCPOPTS pTcpOpts)
1387	{
1388	RT_NOREF(pTstEnv);
1389
1390	RTGETOPTUNION Val;
1391	RT_ZERO(Val);
1392
1393	Val.u32 = pTcpOpts->enmConnMode;
1394	int rc = AudioTestSvcClientHandleOption(pClient, ATSTCPOPT_CONN_MODE, &Val);
1395	AssertRCReturn(rc, rc);
1396
1397	if ( pTcpOpts->enmConnMode == ATSCONNMODE_BOTH
1398	\|\| pTcpOpts->enmConnMode == ATSCONNMODE_SERVER)
1399	{
1400	Assert(pTcpOpts->uBindPort); /* Always set by the caller. */
1401	Val.u16 = pTcpOpts->uBindPort;
1402	rc = AudioTestSvcClientHandleOption(pClient, ATSTCPOPT_BIND_PORT, &Val);
1403	AssertRCReturn(rc, rc);
1404
1405	if (pTcpOpts->szBindAddr[0])
1406	{
1407	Val.psz = pTcpOpts->szBindAddr;
1408	rc = AudioTestSvcClientHandleOption(pClient, ATSTCPOPT_BIND_ADDRESS, &Val);
1409	AssertRCReturn(rc, rc);
1410	}
1411	else
1412	{
1413	RTTestFailed(g_hTest, "No bind address specified!\n");
1414	return VERR_INVALID_PARAMETER;
1415	}
1416
1417	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Connecting %s by listening as server at %s:%RU32 ...\n",
1418	pszWhat, pTcpOpts->szBindAddr, pTcpOpts->uBindPort);
1419	}
1420
1421
1422	if ( pTcpOpts->enmConnMode == ATSCONNMODE_BOTH
1423	\|\| pTcpOpts->enmConnMode == ATSCONNMODE_CLIENT)
1424	{
1425	Assert(pTcpOpts->uConnectPort); /* Always set by the caller. */
1426	Val.u16 = pTcpOpts->uConnectPort;
1427	rc = AudioTestSvcClientHandleOption(pClient, ATSTCPOPT_CONNECT_PORT, &Val);
1428	AssertRCReturn(rc, rc);
1429
1430	if (pTcpOpts->szConnectAddr[0])
1431	{
1432	Val.psz = pTcpOpts->szConnectAddr;
1433	rc = AudioTestSvcClientHandleOption(pClient, ATSTCPOPT_CONNECT_ADDRESS, &Val);
1434	AssertRCReturn(rc, rc);
1435	}
1436	else
1437	{
1438	RTTestFailed(g_hTest, "No connect address specified!\n");
1439	return VERR_INVALID_PARAMETER;
1440	}
1441
1442	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Connecting %s by connecting as client to %s:%RU32 ...\n",
1443	pszWhat, pTcpOpts->szConnectAddr, pTcpOpts->uConnectPort);
1444	}
1445
1446	rc = AudioTestSvcClientConnect(pClient);
1447	if (RT_FAILURE(rc))
1448	{
1449	RTTestFailed(g_hTest, "Connecting %s failed with %Rrc\n", pszWhat, rc);
1450	return rc;
1451	}
1452
1453	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Successfully connected %s\n", pszWhat);
1454	return rc;
1455	}
1456
1457	/**
1458	* Configures and starts an ATS TCP/IP server.
1459	*
1460	* @returns VBox status code.
1461	* @param pSrv ATS server instance to configure and start.
1462	* @param pCallbacks ATS callback table to use.
1463	* @param pszDesc Hint of server type which is being started.
1464	* @param pTcpOpts TCP options to use.
1465	*/
1466	static int audioTestEnvConfigureAndStartTcpServer(PATSSERVER pSrv, PCATSCALLBACKS pCallbacks, const char *pszDesc,
1467	PAUDIOTESTENVTCPOPTS pTcpOpts)
1468	{
1469	RTGETOPTUNION Val;
1470	RT_ZERO(Val);
1471
1472	int rc = AudioTestSvcInit(pSrv, pCallbacks);
1473	if (RT_FAILURE(rc))
1474	return rc;
1475
1476	Val.u32 = pTcpOpts->enmConnMode;
1477	rc = AudioTestSvcHandleOption(pSrv, ATSTCPOPT_CONN_MODE, &Val);
1478	AssertRCReturn(rc, rc);
1479
1480	if ( pTcpOpts->enmConnMode == ATSCONNMODE_BOTH
1481	\|\| pTcpOpts->enmConnMode == ATSCONNMODE_SERVER)
1482	{
1483	Assert(pTcpOpts->uBindPort); /* Always set by the caller. */
1484	Val.u16 = pTcpOpts->uBindPort;
1485	rc = AudioTestSvcHandleOption(pSrv, ATSTCPOPT_BIND_PORT, &Val);
1486	AssertRCReturn(rc, rc);
1487
1488	if (pTcpOpts->szBindAddr[0])
1489	{
1490	Val.psz = pTcpOpts->szBindAddr;
1491	rc = AudioTestSvcHandleOption(pSrv, ATSTCPOPT_BIND_ADDRESS, &Val);
1492	AssertRCReturn(rc, rc);
1493	}
1494	else
1495	{
1496	RTTestFailed(g_hTest, "No bind address specified!\n");
1497	return VERR_INVALID_PARAMETER;
1498	}
1499
1500	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Starting server for %s at %s:%RU32 ...\n",
1501	pszDesc, pTcpOpts->szBindAddr, pTcpOpts->uBindPort);
1502	}
1503
1504
1505	if ( pTcpOpts->enmConnMode == ATSCONNMODE_BOTH
1506	\|\| pTcpOpts->enmConnMode == ATSCONNMODE_CLIENT)
1507	{
1508	Assert(pTcpOpts->uConnectPort); /* Always set by the caller. */
1509	Val.u16 = pTcpOpts->uConnectPort;
1510	rc = AudioTestSvcHandleOption(pSrv, ATSTCPOPT_CONNECT_PORT, &Val);
1511	AssertRCReturn(rc, rc);
1512
1513	if (pTcpOpts->szConnectAddr[0])
1514	{
1515	Val.psz = pTcpOpts->szConnectAddr;
1516	rc = AudioTestSvcHandleOption(pSrv, ATSTCPOPT_CONNECT_ADDRESS, &Val);
1517	AssertRCReturn(rc, rc);
1518	}
1519	else
1520	{
1521	RTTestFailed(g_hTest, "No connect address specified!\n");
1522	return VERR_INVALID_PARAMETER;
1523	}
1524
1525	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Starting server for %s by connecting as client to %s:%RU32 ...\n",
1526	pszDesc, pTcpOpts->szConnectAddr, pTcpOpts->uConnectPort);
1527	}
1528
1529	if (RT_SUCCESS(rc))
1530	{
1531	rc = AudioTestSvcStart(pSrv);
1532	if (RT_FAILURE(rc))
1533	RTTestFailed(g_hTest, "Starting server for %s failed with %Rrc\n", pszDesc, rc);
1534	}
1535
1536	return rc;
1537	}
1538
1539	/**
1540	* Initializes an audio test environment.
1541	*
1542	* @param pTstEnv Audio test environment to initialize.
1543	*/
1544	void audioTestEnvInit(PAUDIOTESTENV pTstEnv)
1545	{
1546	RT_BZERO(pTstEnv, sizeof(AUDIOTESTENV));
1547
1548	pTstEnv->msTimeout = AUDIOTEST_TIMEOUT_DEFAULT_MS;
1549
1550	audioTestIoOptsInitDefaults(&pTstEnv->IoOpts);
1551	audioTestToneParmsInit(&pTstEnv->ToneParms);
1552	}
1553
1554	/**
1555	* Creates an audio test environment.
1556	*
1557	* @returns VBox status code.
1558	* @param pTstEnv Audio test environment to create.
1559	* @param pDrvStack Driver stack to use.
1560	*/
1561	int audioTestEnvCreate(PAUDIOTESTENV pTstEnv, PAUDIOTESTDRVSTACK pDrvStack)
1562	{
1563	AssertReturn(PDMAudioPropsAreValid(&pTstEnv->IoOpts.Props), VERR_WRONG_ORDER);
1564
1565	int rc = VINF_SUCCESS;
1566
1567	pTstEnv->pDrvStack = pDrvStack;
1568
1569	/*
1570	* Set sane defaults if not already set.
1571	*/
1572	if (!RTStrNLen(pTstEnv->szTag, sizeof(pTstEnv->szTag)))
1573	rc = AudioTestGenTag(pTstEnv->szTag, sizeof(pTstEnv->szTag));
1574
1575	if (!RTStrNLen(pTstEnv->szPathTemp, sizeof(pTstEnv->szPathTemp)))
1576	{
1577	int rc2 = AudioTestPathGetTemp(pTstEnv->szPathTemp, sizeof(pTstEnv->szPathTemp));
1578	if (RT_SUCCESS(rc))
1579	rc = rc2;
1580	}
1581
1582	if (!RTStrNLen(pTstEnv->szPathOut, sizeof(pTstEnv->szPathOut)))
1583	{
1584	int rc2 = RTPathJoin(pTstEnv->szPathOut, sizeof(pTstEnv->szPathOut), pTstEnv->szPathTemp, "vkat-temp");
1585	if (RT_SUCCESS(rc))
1586	rc = rc2;
1587	}
1588
1589	char szPathTemp[RTPATH_MAX];
1590	if ( RT_SUCCESS(rc)
1591	&& ( !strlen(pTstEnv->szPathTemp)
1592	\|\| !strlen(pTstEnv->szPathOut)))
1593	rc = RTPathTemp(szPathTemp, sizeof(szPathTemp));
1594
1595	if ( RT_SUCCESS(rc)
1596	&& !strlen(pTstEnv->szPathTemp))
1597	rc = RTPathJoin(pTstEnv->szPathTemp, sizeof(pTstEnv->szPathTemp), szPathTemp, "vkat-temp");
1598
1599	if (RT_SUCCESS(rc))
1600	{
1601	rc = RTDirCreate(pTstEnv->szPathTemp, RTFS_UNIX_IRWXU, 0 /* fFlags */);
1602	if (rc == VERR_ALREADY_EXISTS)
1603	rc = VINF_SUCCESS;
1604	}
1605
1606	if ( RT_SUCCESS(rc)
1607	&& !strlen(pTstEnv->szPathOut))
1608	rc = RTPathJoin(pTstEnv->szPathOut, sizeof(pTstEnv->szPathOut), szPathTemp, "vkat");
1609
1610	if (RT_SUCCESS(rc))
1611	{
1612	rc = RTDirCreate(pTstEnv->szPathOut, RTFS_UNIX_IRWXU, 0 /* fFlags */);
1613	if (rc == VERR_ALREADY_EXISTS)
1614	rc = VINF_SUCCESS;
1615	}
1616
1617	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Initializing environment for mode '%s'\n", pTstEnv->enmMode == AUDIOTESTMODE_HOST ? "host" : "guest");
1618	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Using tag '%s'\n", pTstEnv->szTag);
1619	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Output directory is '%s'\n", pTstEnv->szPathOut);
1620	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Temp directory is '%s'\n", pTstEnv->szPathTemp);
1621
1622	if (RT_FAILURE(rc))
1623	{
1624	RTTestFailed(g_hTest, "Initializing test directories failed with %Rrc\n", rc);
1625	return rc;
1626	}
1627
1628	/**
1629	* For NAT'ed VMs we use (default):
1630	* - client mode (uConnectAddr / uConnectPort) on the guest.
1631	* - server mode (uBindAddr / uBindPort) on the host.
1632	*/
1633	if ( !pTstEnv->TcpOpts.szConnectAddr[0]
1634	&& !pTstEnv->TcpOpts.szBindAddr[0])
1635	RTStrCopy(pTstEnv->TcpOpts.szBindAddr, sizeof(pTstEnv->TcpOpts.szBindAddr), "0.0.0.0");
1636
1637	/*
1638	* Determine connection mode based on set variables.
1639	*/
1640	if ( pTstEnv->TcpOpts.szBindAddr[0]
1641	&& pTstEnv->TcpOpts.szConnectAddr[0])
1642	{
1643	pTstEnv->TcpOpts.enmConnMode = ATSCONNMODE_BOTH;
1644	}
1645	else if (pTstEnv->TcpOpts.szBindAddr[0])
1646	pTstEnv->TcpOpts.enmConnMode = ATSCONNMODE_SERVER;
1647	else /* "Reversed mode", i.e. used for NATed VMs. */
1648	pTstEnv->TcpOpts.enmConnMode = ATSCONNMODE_CLIENT;
1649
1650	/* Set a back reference to the test environment for the callback context. */
1651	pTstEnv->CallbackCtx.pTstEnv = pTstEnv;
1652
1653	ATSCALLBACKS Callbacks;
1654	RT_ZERO(Callbacks);
1655	Callbacks.pvUser = &pTstEnv->CallbackCtx;
1656
1657	if (pTstEnv->enmMode == AUDIOTESTMODE_GUEST)
1658	{
1659	Callbacks.pfnHowdy = audioTestGstAtsHowdyCallback;
1660	Callbacks.pfnBye = audioTestGstAtsByeCallback;
1661	Callbacks.pfnTestSetBegin = audioTestGstAtsTestSetBeginCallback;
1662	Callbacks.pfnTestSetEnd = audioTestGstAtsTestSetEndCallback;
1663	Callbacks.pfnTonePlay = audioTestGstAtsTonePlayCallback;
1664	Callbacks.pfnToneRecord = audioTestGstAtsToneRecordCallback;
1665	Callbacks.pfnTestSetSendBegin = audioTestGstAtsTestSetSendBeginCallback;
1666	Callbacks.pfnTestSetSendRead = audioTestGstAtsTestSetSendReadCallback;
1667	Callbacks.pfnTestSetSendEnd = audioTestGstAtsTestSetSendEndCallback;
1668
1669	if (!pTstEnv->TcpOpts.uBindPort)
1670	pTstEnv->TcpOpts.uBindPort = ATS_TCP_DEF_BIND_PORT_GUEST;
1671
1672	if (!pTstEnv->TcpOpts.uConnectPort)
1673	pTstEnv->TcpOpts.uConnectPort = ATS_TCP_DEF_CONNECT_PORT_GUEST;
1674
1675	pTstEnv->pSrv = (PATSSERVER)RTMemAlloc(sizeof(ATSSERVER));
1676	AssertPtrReturn(pTstEnv->pSrv, VERR_NO_MEMORY);
1677
1678	/*
1679	* Start the ATS (Audio Test Service) on the guest side.
1680	* That service then will perform playback and recording operations on the guest, triggered from the host.
1681	*
1682	* When running this in self-test mode, that service also can be run on the host if nothing else is specified.
1683	* Note that we have to bind to "0.0.0.0" by default so that the host can connect to it.
1684	*/
1685	rc = audioTestEnvConfigureAndStartTcpServer(pTstEnv->pSrv, &Callbacks, "guest", &pTstEnv->TcpOpts);
1686	}
1687	else /* Host mode */
1688	{
1689	if (!pTstEnv->TcpOpts.uBindPort)
1690	pTstEnv->TcpOpts.uBindPort = ATS_TCP_DEF_BIND_PORT_HOST;
1691
1692	if (!pTstEnv->TcpOpts.uConnectPort)
1693	pTstEnv->TcpOpts.uConnectPort = ATS_TCP_DEF_CONNECT_PORT_HOST_PORT_FWD;
1694
1695	/**
1696	* Note: Don't set pTstEnv->TcpOpts.szTcpConnectAddr by default here, as this specifies what connection mode
1697	* (client / server / both) we use on the host.
1698	*/
1699
1700	/* We need to start a server on the host so that VMs configured with NAT networking
1701	* can connect to it as well. */
1702	rc = AudioTestSvcClientCreate(&pTstEnv->u.Host.AtsClGuest);
1703	if (RT_SUCCESS(rc))
1704	rc = audioTestEnvConnectViaTcp(pTstEnv, &pTstEnv->u.Host.AtsClGuest,
1705	"host -> guest", &pTstEnv->TcpOpts);
1706	if (RT_SUCCESS(rc))
1707	{
1708	AUDIOTESTENVTCPOPTS ValKitTcpOpts;
1709	RT_ZERO(ValKitTcpOpts);
1710
1711	/* We only connect as client to the Validation Kit audio driver ATS. */
1712	ValKitTcpOpts.enmConnMode = ATSCONNMODE_CLIENT;
1713
1714	/* For now we ASSUME that the Validation Kit audio driver ATS runs on the same host as VKAT (this binary) runs on. */
1715	ValKitTcpOpts.uConnectPort = ATS_TCP_DEF_CONNECT_PORT_VALKIT; /** @todo Make this dynamic. */
1716	RTStrCopy(ValKitTcpOpts.szConnectAddr, sizeof(ValKitTcpOpts.szConnectAddr), ATS_TCP_DEF_CONNECT_HOST_ADDR_STR); /** @todo Ditto. */
1717
1718	rc = AudioTestSvcClientCreate(&pTstEnv->u.Host.AtsClValKit);
1719	if (RT_SUCCESS(rc))
1720	{
1721	rc = audioTestEnvConnectViaTcp(pTstEnv, &pTstEnv->u.Host.AtsClValKit,
1722	"host -> valkit", &ValKitTcpOpts);
1723	if (RT_FAILURE(rc))
1724	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Unable to connect to the Validation Kit audio driver!\n"
1725	"There could be multiple reasons:\n\n"
1726	" - Wrong host being used\n"
1727	" - VirtualBox host version is too old\n"
1728	" - Audio debug mode is not enabled\n"
1729	" - Support for Validation Kit audio driver is not included\n"
1730	" - Firewall / network configuration problem\n");
1731	}
1732	}
1733	}
1734
1735	return rc;
1736	}
1737
1738	/**
1739	* Destroys an audio test environment.
1740	*
1741	* @param pTstEnv Audio test environment to destroy.
1742	*/
1743	void audioTestEnvDestroy(PAUDIOTESTENV pTstEnv)
1744	{
1745	if (!pTstEnv)
1746	return;
1747
1748	/* When in host mode, we need to destroy our ATS clients in order to also let
1749	* the ATS server(s) know we're going to quit. */
1750	if (pTstEnv->enmMode == AUDIOTESTMODE_HOST)
1751	{
1752	AudioTestSvcClientDestroy(&pTstEnv->u.Host.AtsClValKit);
1753	AudioTestSvcClientDestroy(&pTstEnv->u.Host.AtsClGuest);
1754	}
1755
1756	if (pTstEnv->pSrv)
1757	{
1758	int rc2 = AudioTestSvcDestroy(pTstEnv->pSrv);
1759	AssertRC(rc2);
1760
1761	RTMemFree(pTstEnv->pSrv);
1762	pTstEnv->pSrv = NULL;
1763	}
1764
1765	for (unsigned i = 0; i < RT_ELEMENTS(pTstEnv->aStreams); i++)
1766	{
1767	int rc2 = audioTestStreamDestroy(pTstEnv->pDrvStack, &pTstEnv->aStreams[i]);
1768	if (RT_FAILURE(rc2))
1769	RTTestFailed(g_hTest, "Stream destruction for stream #%u failed with %Rrc\n", i, rc2);
1770	}
1771
1772	/* Try cleaning up a bit. */
1773	RTDirRemove(pTstEnv->szPathTemp);
1774	RTDirRemove(pTstEnv->szPathOut);
1775
1776	pTstEnv->pDrvStack = NULL;
1777	}
1778
1779	/**
1780	* Closes, packs up and destroys a test environment.
1781	*
1782	* @returns VBox status code.
1783	* @param pTstEnv Test environment to handle.
1784	* @param fPack Whether to pack the test set up before destroying / wiping it.
1785	* @param pszPackFile Where to store the packed test set file on success. Can be NULL if \a fPack is \c false.
1786	* @param cbPackFile Size (in bytes) of \a pszPackFile. Can be 0 if \a fPack is \c false.
1787	*/
1788	int audioTestEnvPrologue(PAUDIOTESTENV pTstEnv, bool fPack, char *pszPackFile, size_t cbPackFile)
1789	{
1790	/* Close the test set first. */
1791	AudioTestSetClose(&pTstEnv->Set);
1792
1793	int rc = VINF_SUCCESS;
1794
1795	if (fPack)
1796	{
1797	/* Before destroying the test environment, pack up the test set so
1798	* that it's ready for transmission. */
1799	rc = AudioTestSetPack(&pTstEnv->Set, pTstEnv->szPathOut, pszPackFile, cbPackFile);
1800	if (RT_SUCCESS(rc))
1801	RTTestPrintf(g_hTest, RTTESTLVL_ALWAYS, "Test set packed up to '%s'\n", pszPackFile);
1802	}
1803
1804	if (!g_fDrvAudioDebug) /* Don't wipe stuff when debugging. Can be useful for introspecting data. */
1805	/* ignore rc */ AudioTestSetWipe(&pTstEnv->Set);
1806
1807	AudioTestSetDestroy(&pTstEnv->Set);
1808
1809	if (RT_FAILURE(rc))
1810	RTTestFailed(g_hTest, "Test set prologue failed with %Rrc\n", rc);
1811
1812	return rc;
1813	}
1814
1815	/**
1816	* Initializes an audio test parameters set.
1817	*
1818	* @param pTstParms Test parameters set to initialize.
1819	*/
1820	void audioTestParmsInit(PAUDIOTESTPARMS pTstParms)
1821	{
1822	RT_ZERO(*pTstParms);
1823	}
1824
1825	/**
1826	* Destroys an audio test parameters set.
1827	*
1828	* @param pTstParms Test parameters set to destroy.
1829	*/
1830	void audioTestParmsDestroy(PAUDIOTESTPARMS pTstParms)
1831	{
1832	if (!pTstParms)
1833	return;
1834
1835	return;
1836	}
1837

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source: vbox/trunk/src/VBox/ValidationKit/utils/audio/vkatCommon.cpp@ 104620

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