AudioMixBuffer.cpp@ 60066

Last change on this file since 60066 was 57451, checked in by vboxsync, 9 years ago
Audio: Try to fix chopped off samples when playing short sound files.
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1	/* $Id: AudioMixBuffer.cpp 57451 2015-08-19 09:39:17Z vboxsync $ */
2	/** @file
3	* VBox audio: Audio mixing buffer for converting reading/writing audio
4	* samples.
5	*/
6
7	/*
8	* Copyright (C) 2014-2015 Oracle Corporation
9	*
10	* This file is part of VirtualBox Open Source Edition (OSE), as
11	* available from http://www.virtualbox.org. This file is free software;
12	* you can redistribute it and/or modify it under the terms of the GNU
13	* General Public License (GPL) as published by the Free Software
14	* Foundation, in version 2 as it comes in the "COPYING" file of the
15	* VirtualBox OSE distribution. VirtualBox OSE is distributed in the
16	* hope that it will be useful, but WITHOUT ANY WARRANTY of any kind.
17	*/
18	#define LOG_GROUP LOG_GROUP_AUDIO_MIXER_BUFFER
19	#include <VBox/log.h>
20
21	/*
22	* DEBUG_DUMP_PCM_DATA enables dumping the raw PCM data
23	* to a file on the host. Be sure to adjust DEBUG_DUMP_PCM_DATA_PATH
24	* to your needs before using this!
25	*/
26	#ifdef DEBUG
27	//# define DEBUG_DUMP_PCM_DATA
28	# define DEBUG_DUMP_PCM_DATA_PATH "c:\\temp\\"
29	#endif
30
31	#include <iprt/asm-math.h>
32	#include <iprt/assert.h>
33	#ifdef DEBUG_DUMP_PCM_DATA
34	# include <iprt/file.h>
35	#endif
36	#include <iprt/mem.h>
37	#include <iprt/string.h> /* For RT_BZERO. */
38
39	#ifdef TESTCASE
40	# define LOG_ENABLED
41	# include <iprt/stream.h>
42	#endif
43	#include <VBox/err.h>
44
45	#include "AudioMixBuffer.h"
46
47	#if 0
48	# define AUDMIXBUF_LOG(x) LogFlowFunc(x)
49	#else
50	# if defined(TESTCASE)
51	# define AUDMIXBUF_LOG(x) LogFunc(x)
52	# else
53	# define AUDMIXBUF_LOG(x) do {} while (0)
54	# endif
55	#endif
56
57
58	/*
59	* Soft Volume Control
60	*
61	* The external code supplies an 8-bit volume (attenuation) value in the
62	* 0 .. 255 range. This represents 0 to -96dB attenuation where an input
63	* value of 0 corresponds to -96dB and 255 corresponds to 0dB (unchanged).
64	*
65	* Each step thus correspons to 96 / 256 or 0.375dB. Every 6dB (16 steps)
66	* represents doubling the sample value.
67	*
68	* For internal use, the volume control needs to be converted to a 16-bit
69	* (sort of) exponential value between 1 and 65536. This is used with fixed
70	* point arithmetic such that 65536 means 1.0 and 1 means 1/65536.
71	*
72	* For actual volume calculation, 33.31 fixed point is used. Maximum (or
73	* unattenuated) volume is represented as 0x40000000; conveniently, this
74	* value fits into a uint32_t.
75	*
76	* To enable fast processing, the maximum volume must be a power of two
77	* and must not have a sign when converted to int32_t. While 0x80000000
78	* violates these constraints, 0x40000000 does not.
79	*/
80
81
82	/** Logarithmic/exponential volume conversion table. */
83	static uint32_t s_aVolumeConv[256] = {
84	1, 1, 1, 1, 1, 1, 1, 1, /* 7 */
85	1, 2, 2, 2, 2, 2, 2, 2, /* 15 */
86	2, 2, 2, 2, 2, 3, 3, 3, /* 23 */
87	3, 3, 3, 3, 4, 4, 4, 4, /* 31 */
88	4, 4, 5, 5, 5, 5, 5, 6, /* 39 */
89	6, 6, 6, 7, 7, 7, 8, 8, /* 47 */
90	8, 9, 9, 10, 10, 10, 11, 11, /* 55 */
91	12, 12, 13, 13, 14, 15, 15, 16, /* 63 */
92	17, 17, 18, 19, 20, 21, 22, 23, /* 71 */
93	24, 25, 26, 27, 28, 29, 31, 32, /* 79 */
94	33, 35, 36, 38, 40, 41, 43, 45, /* 87 */
95	47, 49, 52, 54, 56, 59, 61, 64, /* 95 */
96	67, 70, 73, 76, 79, 83, 87, 91, /* 103 */
97	95, 99, 103, 108, 112, 117, 123, 128, /* 111 */
98	134, 140, 146, 152, 159, 166, 173, 181, /* 119 */
99	189, 197, 206, 215, 225, 235, 245, 256, /* 127 */
100	267, 279, 292, 304, 318, 332, 347, 362, /* 135 */
101	378, 395, 412, 431, 450, 470, 490, 512, /* 143 */
102	535, 558, 583, 609, 636, 664, 693, 724, /* 151 */
103	756, 790, 825, 861, 899, 939, 981, 1024, /* 159 */
104	1069, 1117, 1166, 1218, 1272, 1328, 1387, 1448, /* 167 */
105	1512, 1579, 1649, 1722, 1798, 1878, 1961, 2048, /* 175 */
106	2139, 2233, 2332, 2435, 2543, 2656, 2774, 2896, /* 183 */
107	3025, 3158, 3298, 3444, 3597, 3756, 3922, 4096, /* 191 */
108	4277, 4467, 4664, 4871, 5087, 5312, 5547, 5793, /* 199 */
109	6049, 6317, 6597, 6889, 7194, 7512, 7845, 8192, /* 207 */
110	8555, 8933, 9329, 9742, 10173, 10624, 11094, 11585, /* 215 */
111	12098, 12634, 13193, 13777, 14387, 15024, 15689, 16384, /* 223 */
112	17109, 17867, 18658, 19484, 20347, 21247, 22188, 23170, /* 231 */
113	24196, 25268, 26386, 27554, 28774, 30048, 31379, 32768, /* 239 */
114	34219, 35734, 37316, 38968, 40693, 42495, 44376, 46341, /* 247 */
115	48393, 50535, 52773, 55109, 57549, 60097, 62757, 65536, /* 255 */
116	};
117
118	/* Bit shift for fixed point conversion. */
119	#define AUDIOMIXBUF_VOL_SHIFT 30
120
121	/* Internal representation of 0dB volume (1.0 in fixed point). */
122	#define AUDIOMIXBUF_VOL_0DB (1 << AUDIOMIXBUF_VOL_SHIFT)
123
124	AssertCompile(AUDIOMIXBUF_VOL_0DB <= 0x40000000); /* Must always hold. */
125	AssertCompile(AUDIOMIXBUF_VOL_0DB == 0x40000000); /* For now -- when only attenuation is used. */
126
127	/**
128	* Structure for holding sample conversion parameters for
129	* the audioMixBufConvFromXXX / audioMixBufConvToXXX macros.
130	*/
131	typedef struct AUDMIXBUF_CONVOPTS
132	{
133	/** Number of audio samples to convert. */
134	uint32_t cSamples;
135	/** Volume to apply during conversion. Pass 0
136	* to convert the original values. May not apply to
137	* all conversion functions. */
138	PDMAUDIOVOLUME Volume;
139	} AUDMIXBUF_CONVOPTS, *PAUDMIXBUF_CONVOPTS;
140
141	/*
142	* When running the audio testcases we want to verfiy
143	* the macro-generated routines separately, so unmark them as being
144	* inlined + static.
145	*/
146	#ifdef TESTCASE
147	# define AUDMIXBUF_MACRO_FN
148	#else
149	# define AUDMIXBUF_MACRO_FN static inline
150	#endif
151
152	#ifdef DEBUG
153	static uint64_t s_cSamplesMixedTotal = 0;
154	static inline void audioMixBufPrint(PPDMAUDIOMIXBUF pMixBuf);
155	#endif
156
157	typedef uint32_t (AUDMIXBUF_FN_CONVFROM) (PPDMAUDIOSAMPLE paDst, const void *pvSrc, uint32_t cbSrc, const PAUDMIXBUF_CONVOPTS pOpts);
158	typedef AUDMIXBUF_FN_CONVFROM *PAUDMIXBUF_FN_CONVFROM;
159
160	typedef void (AUDMIXBUF_FN_CONVTO) (void *pvDst, const PPDMAUDIOSAMPLE paSrc, const PAUDMIXBUF_CONVOPTS pOpts);
161	typedef AUDMIXBUF_FN_CONVTO *PAUDMIXBUF_FN_CONVTO;
162
163	/* Can return VINF_TRY_AGAIN for getting next pointer at beginning (circular) */
164
165	/**
166	* Acquires (reads) a mutable pointer to the mixing buffer's audio samples without
167	* any conversion done.
168	** @todo Rename to AudioMixBufPeek(Mutable/Raw)?
169	** @todo Protect the buffer's data?
170	*
171	* @return IPRT status code.
172	* @param pMixBuf Mixing buffer to acquire audio samples from.
173	* @param cSamplesToRead Number of audio samples to read.
174	* @param ppvSamples Returns a mutable pointer to the buffer's audio sample data.
175	* @param pcSamplesRead Number of audio samples read (acquired).
176	*
177	* @remark This function is not thread safe!
178	*/
179	int AudioMixBufAcquire(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSamplesToRead,
180	PPDMAUDIOSAMPLE ppvSamples, uint32_t pcSamplesRead)
181	{
182	AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER);
183	AssertPtrReturn(ppvSamples, VERR_INVALID_POINTER);
184	AssertPtrReturn(pcSamplesRead, VERR_INVALID_POINTER);
185
186	int rc;
187
188	if (!cSamplesToRead)
189	{
190	*pcSamplesRead = 0;
191	return VINF_SUCCESS;
192	}
193
194	uint32_t cSamplesRead;
195	if (pMixBuf->offReadWrite + cSamplesToRead > pMixBuf->cSamples)
196	{
197	cSamplesRead = pMixBuf->cSamples - pMixBuf->offReadWrite;
198	rc = VINF_TRY_AGAIN;
199	}
200	else
201	{
202	cSamplesRead = cSamplesToRead;
203	rc = VINF_SUCCESS;
204	}
205
206	*ppvSamples = &pMixBuf->pSamples[pMixBuf->offReadWrite];
207	AssertPtr(ppvSamples);
208
209	pMixBuf->offReadWrite = (pMixBuf->offReadWrite + cSamplesRead) % pMixBuf->cSamples;
210	Assert(pMixBuf->offReadWrite <= pMixBuf->cSamples);
211	pMixBuf->cProcessed -= RT_MIN(cSamplesRead, pMixBuf->cProcessed);
212
213	*pcSamplesRead = cSamplesRead;
214
215	return rc;
216	}
217
218	/**
219	* Returns available number of samples for reading.
220	*
221	* @return uint32_t Number of samples available for reading.
222	* @param pMixBuf Mixing buffer to return value for.
223	*/
224	uint32_t AudioMixBufAvail(PPDMAUDIOMIXBUF pMixBuf)
225	{
226	AssertPtrReturn(pMixBuf, true);
227
228	uint32_t cAvail;
229	if (pMixBuf->pParent) /* Child */
230	cAvail = pMixBuf->cMixed;
231	else
232	cAvail = pMixBuf->cProcessed;
233
234	Assert(cAvail <= pMixBuf->cSamples);
235	return cAvail;
236	}
237
238	/**
239	* Clears the entire sample buffer.
240	*
241	* @param pMixBuf Mixing buffer to clear.
242	*
243	*/
244	void AudioMixBufClear(PPDMAUDIOMIXBUF pMixBuf)
245	{
246	AssertPtrReturnVoid(pMixBuf);
247
248	if (pMixBuf->cSamples)
249	RT_BZERO(pMixBuf->pSamples, pMixBuf->cSamples * sizeof(PDMAUDIOSAMPLE));
250	}
251
252	/**
253	* Clears (zeroes) the buffer by a certain amount of (processed) samples and
254	* keeps track to eventually assigned children buffers.
255	*
256	* @param pMixBuf Mixing buffer to clear.
257	* @param cSamplesToClear Number of audio samples to clear.
258	*/
259	void AudioMixBufFinish(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSamplesToClear)
260	{
261	AUDMIXBUF_LOG(("cSamples=%RU32\n", cSamplesToClear));
262	AUDMIXBUF_LOG(("%s: offReadWrite=%RU32, cProcessed=%RU32\n",
263	pMixBuf->pszName, pMixBuf->offReadWrite, pMixBuf->cProcessed));
264
265	PPDMAUDIOMIXBUF pIter;
266	RTListForEach(&pMixBuf->lstBuffers, pIter, PDMAUDIOMIXBUF, Node)
267	{
268	AUDMIXBUF_LOG(("\t%s: cMixed=%RU32 -> %RU32\n",
269	pIter->pszName, pIter->cMixed, pIter->cMixed - cSamplesToClear));
270
271	pIter->cMixed -= RT_MIN(pIter->cMixed, cSamplesToClear);
272	pIter->offReadWrite = 0;
273	}
274
275	uint32_t cLeft = RT_MIN(cSamplesToClear, pMixBuf->cSamples);
276	uint32_t offClear;
277
278	if (cLeft > pMixBuf->offReadWrite) /* Zero end of buffer first (wrap-around). */
279	{
280	AUDMIXBUF_LOG(("Clearing1: %RU32 - %RU32\n",
281	(pMixBuf->cSamples - (cLeft - pMixBuf->offReadWrite)),
282	pMixBuf->cSamples));
283
284	RT_BZERO(pMixBuf->pSamples + (pMixBuf->cSamples - (cLeft - pMixBuf->offReadWrite)),
285	(cLeft - pMixBuf->offReadWrite) * sizeof(PDMAUDIOSAMPLE));
286
287	cLeft -= cLeft - pMixBuf->offReadWrite;
288	offClear = 0;
289	}
290	else
291	offClear = pMixBuf->offReadWrite - cLeft;
292
293	if (cLeft)
294	{
295	AUDMIXBUF_LOG(("Clearing2: %RU32 - %RU32\n",
296	offClear, offClear + cLeft));
297	RT_BZERO(pMixBuf->pSamples + offClear, cLeft * sizeof(PDMAUDIOSAMPLE));
298	}
299	}
300
301	/**
302	* Destroys (uninitializes) a mixing buffer.
303	*
304	* @param pMixBuf Mixing buffer to destroy.
305	*/
306	void AudioMixBufDestroy(PPDMAUDIOMIXBUF pMixBuf)
307	{
308	if (!pMixBuf)
309	return;
310
311	AudioMixBufUnlink(pMixBuf);
312
313	if (pMixBuf->pszName)
314	{
315	AUDMIXBUF_LOG(("%s\n", pMixBuf->pszName));
316
317	RTStrFree(pMixBuf->pszName);
318	pMixBuf->pszName = NULL;
319	}
320
321	if (pMixBuf->pRate)
322	{
323	RTMemFree(pMixBuf->pRate);
324	pMixBuf->pRate = NULL;
325	}
326
327	if (pMixBuf->pSamples)
328	{
329	Assert(pMixBuf->cSamples);
330
331	RTMemFree(pMixBuf->pSamples);
332	pMixBuf->pSamples = NULL;
333	}
334
335	pMixBuf->cSamples = 0;
336	}
337
338	/**
339	* Returns the size (in audio samples) of free audio buffer space.
340	*
341	* @return uint32_t Size (in audio samples) of free audio buffer space.
342	* @param pMixBuf Mixing buffer to return free size for.
343	*/
344	uint32_t AudioMixBufFree(PPDMAUDIOMIXBUF pMixBuf)
345	{
346	AssertPtrReturn(pMixBuf, 0);
347
348	uint32_t cSamplesFree;
349	if (pMixBuf->pParent)
350	{
351	/*
352	* As a linked child buffer we want to know how many samples
353	* already have been consumed by the parent.
354	*/
355	Assert(pMixBuf->cMixed <= pMixBuf->pParent->cSamples);
356	cSamplesFree = pMixBuf->pParent->cSamples - pMixBuf->cMixed;
357	}
358	else /* As a parent. */
359	{
360	Assert(pMixBuf->cSamples >= pMixBuf->cProcessed);
361	cSamplesFree = pMixBuf->cSamples - pMixBuf->cProcessed;
362	}
363
364	AUDMIXBUF_LOG(("%s: cSamplesFree=%RU32\n", pMixBuf->pszName, cSamplesFree));
365	return cSamplesFree;
366	}
367
368	/**
369	* Returns the size (in bytes) of free audio buffer space.
370	*
371	* @return uint32_t Size (in bytes) of free audio buffer space.
372	* @param pMixBuf Mixing buffer to return free size for.
373	*/
374	uint32_t AudioMixBufFreeBytes(PPDMAUDIOMIXBUF pMixBuf)
375	{
376	return AUDIOMIXBUF_S2B(pMixBuf, AudioMixBufFree(pMixBuf));
377	}
378
379	/**
380	* Allocates the internal audio sample buffer.
381	*
382	* @return IPRT status code.
383	* @param pMixBuf Mixing buffer to allocate sample buffer for.
384	* @param cSamples Number of audio samples to allocate.
385	*/
386	static int audioMixBufAlloc(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSamples)
387	{
388	AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER);
389	AssertReturn(cSamples, VERR_INVALID_PARAMETER);
390
391	AUDMIXBUF_LOG(("%s: cSamples=%RU32\n", pMixBuf->pszName, cSamples));
392
393	size_t cbSamples = cSamples * sizeof(PDMAUDIOSAMPLE);
394	if (!cbSamples)
395	return VERR_INVALID_PARAMETER;
396
397	pMixBuf->pSamples = (PPDMAUDIOSAMPLE)RTMemAllocZ(cbSamples);
398	if (!pMixBuf->pSamples)
399	return VERR_NO_MEMORY;
400
401	pMixBuf->cSamples = cSamples;
402
403	return VINF_SUCCESS;
404	}
405
406	/** Note: Enabling this will generate huge logs! */
407	//#define DEBUG_MACROS
408
409	#ifdef DEBUG_MACROS
410	# define AUDMIXBUF_MACRO_LOG(x) AUDMIXBUF_LOG(x)
411	#elif defined(TESTCASE)
412	# define AUDMIXBUF_MACRO_LOG(x) RTPrintf x
413	#else
414	# define AUDMIXBUF_MACRO_LOG(x) do {} while (0)
415	#endif
416
417	/**
418	* Macro for generating the conversion routines from/to different formats.
419	* Be careful what to pass in/out, as most of the macros are optimized for speed and
420	* thus don't do any bounds checking!
421	*
422	* Note: Currently does not handle any endianness conversion yet!
423	*/
424	#define AUDMIXBUF_CONVERT(_aName, _aType, _aMin, _aMax, _aSigned, _aShift) \
425	/* Clips a specific output value to a single sample value. */ \
426	AUDMIXBUF_MACRO_FN int64_t audioMixBufClipFrom##_aName(_aType aVal) \
427	{ \
428	if (_aSigned) \
429	return ((int64_t) aVal) << (32 - _aShift); \
430	return ((int64_t) aVal - ((_aMax >> 1) + 1)) << (32 - _aShift); \
431	} \
432	\
433	/* Clips a single sample value to a specific output value. */ \
434	AUDMIXBUF_MACRO_FN _aType audioMixBufClipTo##_aName(int64_t iVal) \
435	{ \
436	if (iVal >= 0x7fffffff) \
437	return _aMax; \
438	else if (iVal < -INT64_C(0x80000000)) \
439	return _aMin; \
440	\
441	if (_aSigned) \
442	return (_aType) (iVal >> (32 - _aShift)); \
443	return ((_aType) ((iVal >> (32 - _aShift)) + ((_aMax >> 1) + 1))); \
444	} \
445	\
446	AUDMIXBUF_MACRO_FN uint32_t audioMixBufConvFrom##_aName##Stereo(PPDMAUDIOSAMPLE paDst, const void *pvSrc, uint32_t cbSrc, \
447	const PAUDMIXBUF_CONVOPTS pOpts) \
448	{ \
449	_aType pSrc = (_aType )pvSrc; \
450	uint32_t cSamples = (uint32_t)RT_MIN(pOpts->cSamples, cbSrc / sizeof(_aType)); \
451	AUDMIXBUF_MACRO_LOG(("cSamples=%RU32, sizeof(%zu), lVol=%RU32, rVol=%RU32\n", \
452	pOpts->cSamples, sizeof(_aType), pOpts->Volume.uLeft, pOpts->Volume.uRight)); \
453	for (uint32_t i = 0; i < cSamples; i++) \
454	{ \
455	AUDMIXBUF_MACRO_LOG(("%p: l=%RI16, r=%RI16\n", paDst, pSrc, (pSrc + 1))); \
456	paDst->i64LSample = ASMMult2xS32RetS64((int32_t)audioMixBufClipFrom##_aName(*pSrc++), pOpts->Volume.uLeft ) >> AUDIOMIXBUF_VOL_SHIFT; \
457	paDst->i64RSample = ASMMult2xS32RetS64((int32_t)audioMixBufClipFrom##_aName(*pSrc++), pOpts->Volume.uRight) >> AUDIOMIXBUF_VOL_SHIFT; \
458	AUDMIXBUF_MACRO_LOG(("\t-> l=%RI64, r=%RI64\n", paDst->i64LSample, paDst->i64RSample)); \
459	paDst++; \
460	} \
461	\
462	return cSamples; \
463	} \
464	\
465	AUDMIXBUF_MACRO_FN uint32_t audioMixBufConvFrom##_aName##Mono(PPDMAUDIOSAMPLE paDst, const void *pvSrc, uint32_t cbSrc, \
466	const PAUDMIXBUF_CONVOPTS pOpts) \
467	{ \
468	_aType pSrc = (_aType )pvSrc; \
469	uint32_t cSamples = (uint32_t)RT_MIN(pOpts->cSamples, cbSrc / sizeof(_aType)); \
470	AUDMIXBUF_MACRO_LOG(("cSamples=%RU32, sizeof(%zu), lVol=%RU32, rVol=%RU32\n", \
471	cSamples, sizeof(_aType), pOpts->Volume.uLeft, pOpts->Volume.uRight)); \
472	for (uint32_t i = 0; i < cSamples; i++) \
473	{ \
474	AUDMIXBUF_MACRO_LOG(("%p: s=%RI16\n", paDst, *pSrc)); \
475	paDst->i64LSample = ASMMult2xS32RetS64((int32_t)audioMixBufClipFrom##_aName(*pSrc), pOpts->Volume.uLeft) >> AUDIOMIXBUF_VOL_SHIFT; \
476	paDst->i64RSample = ASMMult2xS32RetS64((int32_t)audioMixBufClipFrom##_aName(*pSrc), pOpts->Volume.uRight) >> AUDIOMIXBUF_VOL_SHIFT; \
477	++pSrc; \
478	AUDMIXBUF_MACRO_LOG(("\t-> l=%RI64, r=%RI64\n", paDst->i64LSample, paDst->i64RSample)); \
479	paDst++; \
480	} \
481	\
482	return cSamples; \
483	} \
484	\
485	AUDMIXBUF_MACRO_FN void audioMixBufConvTo##_aName##Stereo(void *pvDst, const PPDMAUDIOSAMPLE paSrc, \
486	const PAUDMIXBUF_CONVOPTS pOpts) \
487	{ \
488	PPDMAUDIOSAMPLE pSrc = paSrc; \
489	_aType pDst = (_aType )pvDst; \
490	_aType l, r; \
491	uint32_t cSamples = pOpts->cSamples; \
492	while (cSamples--) \
493	{ \
494	AUDMIXBUF_MACRO_LOG(("%p: l=%RI64, r=%RI64\n", pSrc, pSrc->i64LSample, pSrc->i64RSample)); \
495	l = audioMixBufClipTo##_aName(pSrc->i64LSample); \
496	r = audioMixBufClipTo##_aName(pSrc->i64RSample); \
497	AUDMIXBUF_MACRO_LOG(("\t-> l=%RI16, r=%RI16\n", l, r)); \
498	*pDst++ = l; \
499	*pDst++ = r; \
500	pSrc++; \
501	} \
502	} \
503	\
504	AUDMIXBUF_MACRO_FN void audioMixBufConvTo##_aName##Mono(void *pvDst, const PPDMAUDIOSAMPLE paSrc, \
505	const PAUDMIXBUF_CONVOPTS pOpts) \
506	{ \
507	PPDMAUDIOSAMPLE pSrc = paSrc; \
508	_aType pDst = (_aType )pvDst; \
509	uint32_t cSamples = pOpts->cSamples; \
510	while (cSamples--) \
511	{ \
512	*pDst++ = audioMixBufClipTo##_aName((pSrc->i64LSample + pSrc->i64RSample) / 2); \
513	pSrc++; \
514	} \
515	}
516
517	/* audioMixBufConvXXXS8: 8 bit, signed. */
518	AUDMIXBUF_CONVERT(S8 /* Name /, int8_t, INT8_MIN / Min /, INT8_MAX / Max /, true / fSigned /, 8 / cShift */)
519	/* audioMixBufConvXXXU8: 8 bit, unsigned. */
520	AUDMIXBUF_CONVERT(U8 /* Name /, uint8_t, 0 / Min /, UINT8_MAX / Max /, false / fSigned /, 8 / cShift */)
521	/* audioMixBufConvXXXS16: 16 bit, signed. */
522	AUDMIXBUF_CONVERT(S16 /* Name /, int16_t, INT16_MIN / Min /, INT16_MAX / Max /, true / fSigned /, 16 / cShift */)
523	/* audioMixBufConvXXXU16: 16 bit, unsigned. */
524	AUDMIXBUF_CONVERT(U16 /* Name /, uint16_t, 0 / Min /, UINT16_MAX / Max /, false / fSigned /, 16 / cShift */)
525	/* audioMixBufConvXXXS32: 32 bit, signed. */
526	AUDMIXBUF_CONVERT(S32 /* Name /, int32_t, INT32_MIN / Min /, INT32_MAX / Max /, true / fSigned /, 32 / cShift */)
527	/* audioMixBufConvXXXU32: 32 bit, unsigned. */
528	AUDMIXBUF_CONVERT(U32 /* Name /, uint32_t, 0 / Min /, UINT32_MAX / Max /, false / fSigned /, 32 / cShift */)
529
530	#undef AUDMIXBUF_CONVERT
531
532	#define AUDMIXBUF_MIXOP(_aName, _aOp) \
533	AUDMIXBUF_MACRO_FN void audioMixBufOp##_aName(PPDMAUDIOSAMPLE paDst, uint32_t cDstSamples, \
534	PPDMAUDIOSAMPLE paSrc, uint32_t cSrcSamples, \
535	PPDMAUDIOSTRMRATE pRate, \
536	uint32_t pcDstWritten, uint32_t pcSrcRead) \
537	{ \
538	AUDMIXBUF_MACRO_LOG(("cSrcSamples=%RU32, cDstSamples=%RU32\n", cSrcSamples, cDstSamples)); \
539	AUDMIXBUF_MACRO_LOG(("pRate=%p: srcOffset=0x%RX32 (%RU32), dstOffset=0x%RX32 (%RU32), dstInc=0x%RX64 (%RU64)\n", \
540	pRate, pRate->srcOffset, pRate->srcOffset, \
541	(uint32_t)(pRate->dstOffset >> 32), (uint32_t)(pRate->dstOffset >> 32), \
542	pRate->dstInc, pRate->dstInc)); \
543	\
544	if (pRate->dstInc == (UINT64_C(1) + UINT32_MAX)) /* No conversion needed? */ \
545	{ \
546	uint32_t cSamples = RT_MIN(cSrcSamples, cDstSamples); \
547	AUDMIXBUF_MACRO_LOG(("cSamples=%RU32\n", cSamples)); \
548	for (uint32_t i = 0; i < cSamples; i++) \
549	{ \
550	paDst[i].i64LSample _aOp paSrc[i].i64LSample; \
551	paDst[i].i64RSample _aOp paSrc[i].i64RSample; \
552	} \
553	\
554	if (pcDstWritten) \
555	*pcDstWritten = cSamples; \
556	if (pcSrcRead) \
557	*pcSrcRead = cSamples; \
558	return; \
559	} \
560	\
561	PPDMAUDIOSAMPLE paSrcStart = paSrc; \
562	PPDMAUDIOSAMPLE paSrcEnd = paSrc + cSrcSamples; \
563	PPDMAUDIOSAMPLE paDstStart = paDst; \
564	PPDMAUDIOSAMPLE paDstEnd = paDst + cDstSamples; \
565	PDMAUDIOSAMPLE samCur = { 0 }; \
566	PDMAUDIOSAMPLE samOut; \
567	PDMAUDIOSAMPLE samLast = pRate->srcSampleLast; \
568	uint64_t lDelta = 0; \
569	\
570	AUDMIXBUF_MACRO_LOG(("Start: paDstEnd=%p - paDstStart=%p -> %zu\n", paDstEnd, paDst, paDstEnd - paDstStart)); \
571	AUDMIXBUF_MACRO_LOG(("Start: paSrcEnd=%p - paSrcStart=%p -> %zu\n", paSrcEnd, paSrc, paSrcEnd - paSrcStart)); \
572	\
573	while (paDst < paDstEnd) \
574	{ \
575	Assert(paSrc <= paSrcEnd); \
576	Assert(paDst <= paDstEnd); \
577	if (paSrc == paSrcEnd) \
578	break; \
579	\
580	lDelta = 0; \
581	while (pRate->srcOffset <= (pRate->dstOffset >> 32)) \
582	{ \
583	Assert(paSrc <= paSrcEnd); \
584	samLast = *paSrc++; \
585	pRate->srcOffset++; \
586	lDelta++; \
587	if (paSrc == paSrcEnd) \
588	break; \
589	} \
590	\
591	Assert(paSrc <= paSrcEnd); \
592	if (paSrc == paSrcEnd) \
593	break; \
594	\
595	samCur = *paSrc; \
596	\
597	/* Interpolate. */ \
598	int64_t iDstOffInt = pRate->dstOffset & UINT32_MAX; \
599	\
600	samOut.i64LSample = (samLast.i64LSample * ((int64_t) (INT64_C(1) << 32) - iDstOffInt) + samCur.i64LSample * iDstOffInt) >> 32; \
601	samOut.i64RSample = (samLast.i64RSample * ((int64_t) (INT64_C(1) << 32) - iDstOffInt) + samCur.i64RSample * iDstOffInt) >> 32; \
602	\
603	paDst->i64LSample _aOp samOut.i64LSample; \
604	paDst->i64RSample _aOp samOut.i64RSample; \
605	\
606	AUDMIXBUF_MACRO_LOG(("\tlDelta=0x%RX64 (%RU64), iDstOffInt=0x%RX64 (%RI64), l=%RI64, r=%RI64 (cur l=%RI64, r=%RI64)\n", \
607	lDelta, lDelta, iDstOffInt, iDstOffInt, \
608	paDst->i64LSample, paDst->i64RSample, \
609	samCur.i64LSample, samCur.i64RSample)); \
610	\
611	paDst++; \
612	pRate->dstOffset += pRate->dstInc; \
613	\
614	AUDMIXBUF_MACRO_LOG(("\t\tpRate->dstOffset=0x%RX32 (%RU32)\n", pRate->dstOffset, pRate->dstOffset >> 32)); \
615	\
616	} \
617	\
618	AUDMIXBUF_MACRO_LOG(("End: paDst=%p - paDstStart=%p -> %zu\n", paDst, paDstStart, paDst - paDstStart)); \
619	AUDMIXBUF_MACRO_LOG(("End: paSrc=%p - paSrcStart=%p -> %zu\n", paSrc, paSrcStart, paSrc - paSrcStart)); \
620	\
621	pRate->srcSampleLast = samLast; \
622	\
623	AUDMIXBUF_MACRO_LOG(("pRate->srcSampleLast l=%RI64, r=%RI64, lDelta=0x%RX64 (%RU64)\n", \
624	pRate->srcSampleLast.i64LSample, pRate->srcSampleLast.i64RSample, lDelta, lDelta)); \
625	\
626	if (pcDstWritten) \
627	*pcDstWritten = paDst - paDstStart; \
628	if (pcSrcRead) \
629	*pcSrcRead = paSrc - paSrcStart; \
630	}
631
632	/* audioMixBufOpAssign: Assigns values from source buffer to destination bufffer, overwriting the destination. */
633	AUDMIXBUF_MIXOP(Assign /* Name /, = / Operation */)
634	/* audioMixBufOpBlend: Blends together the values from both, the source and the destination buffer. */
635	AUDMIXBUF_MIXOP(Blend /* Name /, += / Operation */)
636
637	#undef AUDMIXBUF_MIXOP
638	#undef AUDMIXBUF_MACRO_LOG
639
640	/** Dummy conversion used when the source is muted. */
641	AUDMIXBUF_MACRO_FN uint32_t audioMixBufConvFromSilence(PPDMAUDIOSAMPLE paDst, const void *pvSrc,
642	uint32_t cbSrc, const PAUDMIXBUF_CONVOPTS pOpts)
643	{
644	/* Internally zero always corresponds to silence. */
645	memset(paDst, 0, pOpts->cSamples * sizeof(paDst[0]));
646	return pOpts->cSamples;
647	}
648
649	/**
650	* Looks up the matching conversion (macro) routine for converting
651	* audio samples from a source format.
652	*
653	** @todo Speed up the lookup by binding it to the actual stream state.
654	*
655	* @return PAUDMIXBUF_FN_CONVFROM Function pointer to conversion macro if found, NULL if not supported.
656	* @param enmFmt Audio format to lookup conversion macro for.
657	* @param fMuted Flag determining whether the source is muted.
658	*/
659	static inline PAUDMIXBUF_FN_CONVFROM audioMixBufConvFromLookup(PDMAUDIOMIXBUFFMT enmFmt, bool fMuted)
660	{
661	if (fMuted)
662	return audioMixBufConvFromSilence;
663
664	if (AUDMIXBUF_FMT_SIGNED(enmFmt))
665	{
666	if (AUDMIXBUF_FMT_CHANNELS(enmFmt) == 2)
667	{
668	switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
669	{
670	case 8: return audioMixBufConvFromS8Stereo;
671	case 16: return audioMixBufConvFromS16Stereo;
672	case 32: return audioMixBufConvFromS32Stereo;
673	default: return NULL;
674	}
675	}
676	else if (AUDMIXBUF_FMT_CHANNELS(enmFmt) == 1)
677	{
678	switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
679	{
680	case 8: return audioMixBufConvFromS8Mono;
681	case 16: return audioMixBufConvFromS16Mono;
682	case 32: return audioMixBufConvFromS32Mono;
683	default: return NULL;
684	}
685	}
686	}
687	else /* Unsigned */
688	{
689	if (AUDMIXBUF_FMT_CHANNELS(enmFmt) == 2)
690	{
691	switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
692	{
693	case 8: return audioMixBufConvFromU8Stereo;
694	case 16: return audioMixBufConvFromU16Stereo;
695	case 32: return audioMixBufConvFromU32Stereo;
696	default: return NULL;
697	}
698	}
699	else if (AUDMIXBUF_FMT_CHANNELS(enmFmt) == 1)
700	{
701	switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
702	{
703	case 8: return audioMixBufConvFromU8Mono;
704	case 16: return audioMixBufConvFromU16Mono;
705	case 32: return audioMixBufConvFromU32Mono;
706	default: return NULL;
707	}
708	}
709	}
710
711	return NULL;
712	}
713
714	/**
715	* Looks up the matching conversion (macro) routine for converting
716	* audio samples to a destination format.
717	*
718	** @todo Speed up the lookup by binding it to the actual stream state.
719	*
720	* @return PAUDMIXBUF_FN_CONVTO Function pointer to conversion macro if found, NULL if not supported.
721	* @param enmFmt Audio format to lookup conversion macro for.
722	*/
723	static inline PAUDMIXBUF_FN_CONVTO audioMixBufConvToLookup(PDMAUDIOMIXBUFFMT enmFmt)
724	{
725	if (AUDMIXBUF_FMT_SIGNED(enmFmt))
726	{
727	if (AUDMIXBUF_FMT_CHANNELS(enmFmt) == 2)
728	{
729	switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
730	{
731	case 8: return audioMixBufConvToS8Stereo;
732	case 16: return audioMixBufConvToS16Stereo;
733	case 32: return audioMixBufConvToS32Stereo;
734	default: return NULL;
735	}
736	}
737	else if (AUDMIXBUF_FMT_CHANNELS(enmFmt) == 1)
738	{
739	switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
740	{
741	case 8: return audioMixBufConvToS8Mono;
742	case 16: return audioMixBufConvToS16Mono;
743	case 32: return audioMixBufConvToS32Mono;
744	default: return NULL;
745	}
746	}
747	}
748	else /* Unsigned */
749	{
750	if (AUDMIXBUF_FMT_CHANNELS(enmFmt) == 2)
751	{
752	switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
753	{
754	case 8: return audioMixBufConvToU8Stereo;
755	case 16: return audioMixBufConvToU16Stereo;
756	case 32: return audioMixBufConvToU32Stereo;
757	default: return NULL;
758	}
759	}
760	else if (AUDMIXBUF_FMT_CHANNELS(enmFmt) == 1)
761	{
762	switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
763	{
764	case 8: return audioMixBufConvToU8Mono;
765	case 16: return audioMixBufConvToU16Mono;
766	case 32: return audioMixBufConvToU32Mono;
767	default: return NULL;
768	}
769	}
770	}
771
772	return NULL;
773	}
774
775	/**
776	* Initializes a mixing buffer.
777	*
778	* @return IPRT status code.
779	* @param pMixBuf Mixing buffer to initialize.
780	* @param pszName Name of mixing buffer for easier identification. Optional.
781	* @param pProps PCM audio properties to use for the mixing buffer.
782	* @param cSamples Maximum number of audio samples the mixing buffer can hold.
783	*/
784	int AudioMixBufInit(PPDMAUDIOMIXBUF pMixBuf, const char *pszName, PPDMPCMPROPS pProps, uint32_t cSamples)
785	{
786	AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER);
787	AssertPtrReturn(pszName, VERR_INVALID_POINTER);
788	AssertPtrReturn(pProps, VERR_INVALID_POINTER);
789
790	pMixBuf->pParent = NULL;
791	RTListInit(&pMixBuf->lstBuffers);
792
793	pMixBuf->pSamples = NULL;
794	pMixBuf->cSamples = 0;
795
796	pMixBuf->offReadWrite = 0;
797	pMixBuf->cMixed = 0;
798	pMixBuf->cProcessed = 0;
799
800	/* Set initial volume to max. */
801	pMixBuf->Volume.fMuted = false;
802	pMixBuf->Volume.uLeft = AUDIOMIXBUF_VOL_0DB;
803	pMixBuf->Volume.uRight = AUDIOMIXBUF_VOL_0DB;
804
805	/* Prevent division by zero.
806	* Do a 1:1 conversion according to AUDIOMIXBUF_S2B_RATIO. */
807	pMixBuf->iFreqRatio = 1 << 20;
808
809	pMixBuf->pRate = NULL;
810
811	pMixBuf->AudioFmt = AUDMIXBUF_AUDIO_FMT_MAKE(pProps->uHz,
812	pProps->cChannels,
813	pProps->cBits,
814	pProps->fSigned);
815	pMixBuf->cShift = pProps->cShift;
816	pMixBuf->pszName = RTStrDup(pszName);
817	if (!pMixBuf->pszName)
818	return VERR_NO_MEMORY;
819
820	AUDMIXBUF_LOG(("%s: uHz=%RU32, cChan=%RU8, cBits=%RU8, fSigned=%RTbool\n",
821	pMixBuf->pszName,
822	AUDMIXBUF_FMT_SAMPLE_FREQ(pMixBuf->AudioFmt),
823	AUDMIXBUF_FMT_CHANNELS(pMixBuf->AudioFmt),
824	AUDMIXBUF_FMT_BITS_PER_SAMPLE(pMixBuf->AudioFmt),
825	RT_BOOL(AUDMIXBUF_FMT_SIGNED(pMixBuf->AudioFmt))));
826
827	return audioMixBufAlloc(pMixBuf, cSamples);
828	}
829
830	/**
831	* Returns @true if there are any audio samples available for processing,
832	* @false if not.
833	*
834	* @return bool @true if there are any audio samples available for processing, @false if not.
835	* @param pMixBuf Mixing buffer to return value for.
836	*/
837	bool AudioMixBufIsEmpty(PPDMAUDIOMIXBUF pMixBuf)
838	{
839	AssertPtrReturn(pMixBuf, true);
840
841	if (pMixBuf->pParent)
842	return (pMixBuf->cMixed == 0);
843	return (pMixBuf->cProcessed == 0);
844	}
845
846	/**
847	* Links an audio mixing buffer to a parent mixing buffer. A parent mixing
848	* buffer can have multiple children mixing buffers [1:N], whereas a child only can
849	* have one parent mixing buffer [N:1].
850	*
851	* The mixing direction always goes from the child/children buffer(s) to the
852	* parent buffer.
853	*
854	* For guest audio output the host backend owns the parent mixing buffer, the
855	* device emulation owns the child/children.
856	*
857	* The audio format of each mixing buffer can vary; the internal mixing code
858	* then will autiomatically do the (needed) conversion.
859	*
860	* @return IPRT status code.
861	* @param pMixBuf Mixing buffer to link parent to.
862	* @param pParent Parent mixing buffer to use for linking.
863	*
864	* @remark Circular linking is not allowed.
865	*/
866	int AudioMixBufLinkTo(PPDMAUDIOMIXBUF pMixBuf, PPDMAUDIOMIXBUF pParent)
867	{
868	AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER);
869	AssertPtrReturn(pParent, VERR_INVALID_POINTER);
870
871	AssertMsgReturn(AUDMIXBUF_FMT_SAMPLE_FREQ(pParent->AudioFmt),
872	("Parent sample frequency (Hz) not set\n"), VERR_INVALID_PARAMETER);
873	AssertMsgReturn(AUDMIXBUF_FMT_SAMPLE_FREQ(pMixBuf->AudioFmt),
874	("Buffer sample frequency (Hz) not set\n"), VERR_INVALID_PARAMETER);
875	AssertMsgReturn(pMixBuf != pParent,
876	("Circular linking not allowed\n"), VERR_INVALID_PARAMETER);
877
878	if (pMixBuf->pParent) /* Already linked? */
879	{
880	AUDMIXBUF_LOG(("%s: Already linked to \"%s\"\n",
881	pMixBuf->pszName, pMixBuf->pParent->pszName));
882	return VERR_ACCESS_DENIED;
883	}
884
885	RTListAppend(&pParent->lstBuffers, &pMixBuf->Node);
886	pMixBuf->pParent = pParent;
887
888	/* Calculate the frequency ratio. */
889	pMixBuf->iFreqRatio = ((int64_t)AUDMIXBUF_FMT_SAMPLE_FREQ(pParent->AudioFmt) << 32)
890	/ AUDMIXBUF_FMT_SAMPLE_FREQ(pMixBuf->AudioFmt);
891
892	if (pMixBuf->iFreqRatio == 0) /* Catch division by zero. */
893	pMixBuf->iFreqRatio = 1 << 20; /* Do a 1:1 conversion instead. */
894
895	uint32_t cSamples = (uint32_t)RT_MIN( ((uint64_t)pParent->cSamples << 32)
896	/ pMixBuf->iFreqRatio, _64K /* 64K samples max. */);
897	if (!cSamples)
898	cSamples = pParent->cSamples;
899
900	int rc = VINF_SUCCESS;
901
902	if (cSamples != pMixBuf->cSamples)
903	{
904	AUDMIXBUF_LOG(("%s: Reallocating samples %RU32 -> %RU32\n",
905	pMixBuf->pszName, pMixBuf->cSamples, cSamples));
906
907	uint32_t cbSamples = cSamples * sizeof(PDMAUDIOSAMPLE);
908	Assert(cbSamples);
909	pMixBuf->pSamples = (PPDMAUDIOSAMPLE)RTMemRealloc(pMixBuf->pSamples, cbSamples);
910	if (!pMixBuf->pSamples)
911	rc = VERR_NO_MEMORY;
912
913	if (RT_SUCCESS(rc))
914	{
915	pMixBuf->cSamples = cSamples;
916
917	/* Make sure to zero the reallocated buffer so that it can be
918	* used properly when blending with another buffer later. */
919	RT_BZERO(pMixBuf->pSamples, cbSamples);
920	}
921	}
922
923	if (RT_SUCCESS(rc))
924	{
925	if (!pMixBuf->pRate)
926	{
927	/* Create rate conversion. */
928	pMixBuf->pRate = (PPDMAUDIOSTRMRATE)RTMemAllocZ(sizeof(PDMAUDIOSTRMRATE));
929	if (!pMixBuf->pRate)
930	return VERR_NO_MEMORY;
931	}
932	else
933	RT_BZERO(pMixBuf->pRate, sizeof(PDMAUDIOSTRMRATE));
934
935	pMixBuf->pRate->dstInc = ((uint64_t)AUDMIXBUF_FMT_SAMPLE_FREQ(pMixBuf->AudioFmt) << 32)
936	/ AUDMIXBUF_FMT_SAMPLE_FREQ(pParent->AudioFmt);
937
938	AUDMIXBUF_LOG(("uThisHz=%RU32, uParentHz=%RU32, iFreqRatio=0x%RX64 (%RI64), uRateInc=0x%RX64 (%RU64), cSamples=%RU32 (%RU32 parent)\n",
939	AUDMIXBUF_FMT_SAMPLE_FREQ(pMixBuf->AudioFmt),
940	AUDMIXBUF_FMT_SAMPLE_FREQ(pParent->AudioFmt),
941	pMixBuf->iFreqRatio, pMixBuf->iFreqRatio,
942	pMixBuf->pRate->dstInc, pMixBuf->pRate->dstInc,
943	pMixBuf->cSamples,
944	pParent->cSamples));
945	AUDMIXBUF_LOG(("%s (%RU32Hz) -> %s (%RU32Hz)\n",
946	pMixBuf->pszName, AUDMIXBUF_FMT_SAMPLE_FREQ(pMixBuf->AudioFmt),
947	pMixBuf->pParent->pszName, AUDMIXBUF_FMT_SAMPLE_FREQ(pParent->AudioFmt)));
948	}
949
950	return rc;
951	}
952
953	/**
954	* Returns the number of audio samples mixed (processed) by
955	* the parent mixing buffer.
956	*
957	* @return uint32_t Number of audio samples mixed (processed).
958	* @param pMixBuf Mixing buffer to return number from.
959	*/
960	uint32_t AudioMixBufMixed(PPDMAUDIOMIXBUF pMixBuf)
961	{
962	AssertPtrReturn(pMixBuf, 0);
963
964	AssertMsgReturn(VALID_PTR(pMixBuf->pParent),
965	("Buffer is not linked to a parent buffer\n"),
966	0);
967
968	AUDMIXBUF_LOG(("%s: cMixed=%RU32\n", pMixBuf->pszName, pMixBuf->cMixed));
969	return pMixBuf->cMixed;
970	}
971
972	/**
973	* Mixes audio samples from a source mixing buffer to a destination mixing buffer.
974	*
975	* @return IPRT status code.
976	* @param pDst Destination mixing buffer.
977	* @param pSrc Source mixing buffer.
978	* @param cSamples Number of source audio samples to mix.
979	* @param pcProcessed Number of audio samples successfully mixed.
980	*/
981	static int audioMixBufMixTo(PPDMAUDIOMIXBUF pDst, PPDMAUDIOMIXBUF pSrc, uint32_t cSamples, uint32_t *pcProcessed)
982	{
983	AssertPtrReturn(pDst, VERR_INVALID_POINTER);
984	AssertPtrReturn(pSrc, VERR_INVALID_POINTER);
985	AssertReturn(cSamples, VERR_INVALID_PARAMETER);
986	/* pcProcessed is optional. */
987
988	/* Live samples indicate how many samples there are in the source buffer
989	* which have not been processed yet by the destination buffer. */
990	uint32_t cLive = pSrc->cMixed;
991	if (cLive >= pDst->cSamples)
992	AUDMIXBUF_LOG(("Destination buffer \"%s\" full (%RU32 samples max), live samples = %RU32\n",
993	pDst->pszName, pDst->cSamples, cLive));
994
995	/* Dead samples are the number of samples in the destination buffer which
996	* will not be needed, that is, are not needed in order to process the live
997	* samples of the source buffer. */
998	uint32_t cDead = pDst->cSamples - cLive;
999
1000	uint32_t cToReadTotal = (uint32_t)RT_MIN(cSamples, AUDIOMIXBUF_S2S_RATIO(pSrc, cDead));
1001	uint32_t offRead = 0;
1002
1003	uint32_t cReadTotal = 0;
1004	uint32_t cWrittenTotal = 0;
1005	uint32_t offWrite = (pDst->offReadWrite + cLive) % pDst->cSamples;
1006
1007	AUDMIXBUF_LOG(("pSrc=%s (%RU32 samples), pDst=%s (%RU32 samples), cLive=%RU32, cDead=%RU32, cToReadTotal=%RU32, offWrite=%RU32\n",
1008	pSrc->pszName, pSrc->cSamples, pDst->pszName, pDst->cSamples, cLive, cDead, cToReadTotal, offWrite));
1009
1010	uint32_t cToRead, cToWrite;
1011	uint32_t cWritten, cRead;
1012
1013	while (cToReadTotal)
1014	{
1015	cDead = pDst->cSamples - cLive;
1016
1017	cToRead = cToReadTotal;
1018	cToWrite = RT_MIN(cDead, pDst->cSamples - offWrite);
1019	if (!cToWrite)
1020	{
1021	AUDMIXBUF_LOG(("Warning: Destination buffer \"%s\" full\n", pDst->pszName));
1022	break;
1023	}
1024
1025	Assert(offWrite + cToWrite <= pDst->cSamples);
1026	Assert(offRead + cToRead <= pSrc->cSamples);
1027
1028	AUDMIXBUF_LOG(("\t%RU32Hz -> %RU32Hz\n", AUDMIXBUF_FMT_SAMPLE_FREQ(pSrc->AudioFmt), AUDMIXBUF_FMT_SAMPLE_FREQ(pDst->AudioFmt)));
1029	AUDMIXBUF_LOG(("\tcDead=%RU32, offWrite=%RU32, cToWrite=%RU32, offRead=%RU32, cToRead=%RU32\n",
1030	cDead, offWrite, cToWrite, offRead, cToRead));
1031
1032	audioMixBufOpBlend(pDst->pSamples + offWrite, cToWrite,
1033	pSrc->pSamples + offRead, cToRead,
1034	pSrc->pRate, &cWritten, &cRead);
1035
1036	AUDMIXBUF_LOG(("\t\tcWritten=%RU32, cRead=%RU32\n", cWritten, cRead));
1037
1038	cReadTotal += cRead;
1039	cWrittenTotal += cWritten;
1040
1041	offRead += cRead;
1042	Assert(cToReadTotal >= cRead);
1043	cToReadTotal -= cRead;
1044
1045	offWrite = (offWrite + cWritten) % pDst->cSamples;
1046
1047	cLive += cWritten;
1048	}
1049
1050	pSrc->cMixed += cWrittenTotal;
1051	pDst->cProcessed += cWrittenTotal;
1052	#ifdef DEBUG
1053	s_cSamplesMixedTotal += cWrittenTotal;
1054	audioMixBufPrint(pDst);
1055	#endif
1056
1057	if (pcProcessed)
1058	*pcProcessed = cReadTotal;
1059
1060	AUDMIXBUF_LOG(("cReadTotal=%RU32 (pcProcessed), cWrittenTotal=%RU32, cSrcMixed=%RU32, cDstProc=%RU32\n",
1061	cReadTotal, cWrittenTotal, pSrc->cMixed, pDst->cProcessed));
1062	return VINF_SUCCESS;
1063	}
1064
1065	/**
1066	* Mixes (multiplexes) audio samples to all connected mixing buffer children.
1067	*
1068	* @return IPRT status code.
1069	* @param pMixBuf Mixing buffer to use.
1070	* @param cSamples Number of audio samples to mix to children.
1071	* @param pcProcessed Maximum number of audio samples successfully mixed
1072	* to all children. Optional.
1073	*/
1074	int AudioMixBufMixToChildren(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSamples,
1075	uint32_t *pcProcessed)
1076	{
1077	AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER);
1078
1079	if (!cSamples)
1080	{
1081	if (pcProcessed)
1082	*pcProcessed = 0;
1083	return VINF_SUCCESS;
1084	}
1085
1086	int rc = VINF_SUCCESS;
1087
1088	uint32_t cProcessed;
1089	uint32_t cProcessedMax = 0;
1090
1091	PPDMAUDIOMIXBUF pIter;
1092	RTListForEach(&pMixBuf->lstBuffers, pIter, PDMAUDIOMIXBUF, Node)
1093	{
1094	rc = audioMixBufMixTo(pIter, pMixBuf, cSamples, &cProcessed);
1095	if (RT_FAILURE(rc))
1096	break;
1097
1098	cProcessedMax = RT_MAX(cProcessedMax, cProcessed);
1099	}
1100
1101	if (pcProcessed)
1102	*pcProcessed = cProcessedMax;
1103
1104	return rc;
1105	}
1106
1107	/**
1108	* Mixes audio samples down to the parent mixing buffer.
1109	*
1110	* @return IPRT status code.
1111	* @param pMixBuf Mixing buffer to mix samples down to parent.
1112	* @param cSamples Number of audio samples to mix down.
1113	* @param pcProcessed Number of audio samples successfully processed. Optional.
1114	*/
1115	int AudioMixBufMixToParent(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSamples,
1116	uint32_t *pcProcessed)
1117	{
1118	AssertMsgReturn(VALID_PTR(pMixBuf->pParent),
1119	("Buffer is not linked to a parent buffer\n"),
1120	VERR_INVALID_PARAMETER);
1121
1122	return audioMixBufMixTo(pMixBuf->pParent, pMixBuf, cSamples, pcProcessed);
1123	}
1124
1125	#ifdef DEBUG
1126	/**
1127	* Prints statistics and status of a mixing buffer to the logger.
1128	* For debug versions only.
1129	*
1130	* @return IPRT status code.
1131	* @param pMixBuf Mixing buffer to print.
1132	*/
1133	static inline void audioMixBufPrint(PPDMAUDIOMIXBUF pMixBuf)
1134	{
1135	PPDMAUDIOMIXBUF pParent = pMixBuf;
1136	if (pMixBuf->pParent)
1137	pParent = pMixBuf->pParent;
1138
1139	AUDMIXBUF_LOG(("********************************************\n"));
1140	AUDMIXBUF_LOG(("%s: offReadWrite=%RU32, cProcessed=%RU32, cMixed=%RU32 (BpS=%RU32)\n",
1141	pParent->pszName,
1142	pParent->offReadWrite, pParent->cProcessed, pParent->cMixed,
1143	AUDIOMIXBUF_S2B(pParent, 1)));
1144
1145	PPDMAUDIOMIXBUF pIter;
1146	RTListForEach(&pParent->lstBuffers, pIter, PDMAUDIOMIXBUF, Node)
1147	{
1148	AUDMIXBUF_LOG(("\t%s: offReadWrite=%RU32, cProcessed=%RU32, cMixed=%RU32 (BpS=%RU32)\n",
1149	pIter->pszName,
1150	pIter->offReadWrite, pIter->cProcessed, pIter->cMixed,
1151	AUDIOMIXBUF_S2B(pIter, 1)));
1152	}
1153	AUDMIXBUF_LOG(("Total samples mixed: %RU64\n", s_cSamplesMixedTotal));
1154	AUDMIXBUF_LOG(("********************************************\n"));
1155	}
1156	#endif
1157
1158	/**
1159	* Returns the total number of samples processed.
1160	*
1161	* @return uint32_t
1162	* @param pMixBuf
1163	*/
1164	uint32_t AudioMixBufProcessed(PPDMAUDIOMIXBUF pMixBuf)
1165	{
1166	AssertPtrReturn(pMixBuf, 0);
1167
1168	AUDMIXBUF_LOG(("%s: cProcessed=%RU32\n", pMixBuf->pszName, pMixBuf->cProcessed));
1169	return pMixBuf->cProcessed;
1170	}
1171
1172	/**
1173	* Reads audio samples at a specific offset.
1174	*
1175	* @return IPRT status code.
1176	* @param pMixBuf Mixing buffer to read audio samples from.
1177	* @param offSamples Offset (in audio samples) to start reading from.
1178	* @param pvBuf Pointer to buffer to write output to.
1179	* @param cbBuf Size (in bytes) of buffer to write to.
1180	* @param pcbRead Size (in bytes) of data read. Optional.
1181	*/
1182	int AudioMixBufReadAt(PPDMAUDIOMIXBUF pMixBuf,
1183	uint32_t offSamples,
1184	void *pvBuf, uint32_t cbBuf,
1185	uint32_t *pcbRead)
1186	{
1187	return AudioMixBufReadAtEx(pMixBuf, pMixBuf->AudioFmt,
1188	offSamples, pvBuf, cbBuf, pcbRead);
1189	}
1190
1191	/**
1192	* Reads audio samples at a specific offset.
1193	* If the audio format of the mixing buffer and the requested audio format do
1194	* not match the output will be converted accordingly.
1195	*
1196	* @return IPRT status code.
1197	* @param pMixBuf Mixing buffer to read audio samples from.
1198	* @param enmFmt Audio format to use for output.
1199	* @param offSamples Offset (in audio samples) to start reading from.
1200	* @param pvBuf Pointer to buffer to write output to.
1201	* @param cbBuf Size (in bytes) of buffer to write to.
1202	* @param pcbRead Size (in bytes) of data read. Optional.
1203	*/
1204	int AudioMixBufReadAtEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt,
1205	uint32_t offSamples,
1206	void *pvBuf, uint32_t cbBuf,
1207	uint32_t *pcbRead)
1208	{
1209	AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER);
1210	AssertPtrReturn(pvBuf, VERR_INVALID_POINTER);
1211	/* pcbRead is optional. */
1212
1213	uint32_t cDstSamples = pMixBuf->cSamples;
1214	uint32_t cLive = pMixBuf->cProcessed;
1215
1216	uint32_t cDead = cDstSamples - cLive;
1217	uint32_t cToProcess = (uint32_t)AUDIOMIXBUF_S2S_RATIO(pMixBuf, cDead);
1218	cToProcess = RT_MIN(cToProcess, AUDIOMIXBUF_B2S(pMixBuf, cbBuf));
1219
1220	AUDMIXBUF_LOG(("%s: offSamples=%RU32, cLive=%RU32, cDead=%RU32, cToProcess=%RU32\n",
1221	pMixBuf->pszName, offSamples, cLive, cDead, cToProcess));
1222
1223	int rc;
1224	if (cToProcess)
1225	{
1226	PAUDMIXBUF_FN_CONVTO pConv = audioMixBufConvToLookup(enmFmt);
1227	if (pConv)
1228	{
1229	AUDMIXBUF_CONVOPTS convOpts = { cToProcess, pMixBuf->Volume };
1230	pConv(pvBuf, pMixBuf->pSamples + offSamples, &convOpts);
1231
1232	rc = VINF_SUCCESS;
1233	}
1234	else
1235	rc = VERR_INVALID_PARAMETER;
1236
1237	#ifdef DEBUG
1238	audioMixBufPrint(pMixBuf);
1239	#endif
1240	}
1241	else
1242	rc = VINF_SUCCESS;
1243
1244	if (RT_SUCCESS(rc))
1245	{
1246	if (pcbRead)
1247	*pcbRead = AUDIOMIXBUF_S2B(pMixBuf, cToProcess);
1248	}
1249
1250	AUDMIXBUF_LOG(("cbRead=%RU32, rc=%Rrc\n", AUDIOMIXBUF_S2B(pMixBuf, cToProcess), rc));
1251	return rc;
1252	}
1253
1254	/**
1255	* Reads audio samples. The audio format of the mixing buffer will be used.
1256	*
1257	* @return IPRT status code.
1258	* @param pMixBuf Mixing buffer to read audio samples from.
1259	* @param pvBuf Pointer to buffer to write output to.
1260	* @param cbBuf Size (in bytes) of buffer to write to.
1261	* @param pcRead Number of audio samples read. Optional.
1262	*/
1263	int AudioMixBufReadCirc(PPDMAUDIOMIXBUF pMixBuf,
1264	void pvBuf, uint32_t cbBuf, uint32_t pcRead)
1265	{
1266	return AudioMixBufReadCircEx(pMixBuf, pMixBuf->AudioFmt,
1267	pvBuf, cbBuf, pcRead);
1268	}
1269
1270	/**
1271	* Reads audio samples in a specific audio format.
1272	* If the audio format of the mixing buffer and the requested audio format do
1273	* not match the output will be converted accordingly.
1274	*
1275	* @return IPRT status code.
1276	* @param pMixBuf Mixing buffer to read audio samples from.
1277	* @param enmFmt Audio format to use for output.
1278	* @param pvBuf Pointer to buffer to write output to.
1279	* @param cbBuf Size (in bytes) of buffer to write to.
1280	* @param pcRead Number of audio samples read. Optional.
1281	*/
1282	int AudioMixBufReadCircEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt,
1283	void pvBuf, uint32_t cbBuf, uint32_t pcRead)
1284	{
1285	AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER);
1286	AssertPtrReturn(pvBuf, VERR_INVALID_POINTER);
1287	/* pcbRead is optional. */
1288
1289	if (!cbBuf)
1290	return VINF_SUCCESS;
1291
1292	uint32_t cToRead = RT_MIN(AUDIOMIXBUF_B2S(pMixBuf, cbBuf), pMixBuf->cProcessed);
1293
1294	AUDMIXBUF_LOG(("%s: pvBuf=%p, cbBuf=%zu (%RU32 samples), cToRead=%RU32\n",
1295	pMixBuf->pszName, pvBuf, cbBuf, AUDIOMIXBUF_B2S(pMixBuf, cbBuf), cToRead));
1296
1297	if (!cToRead)
1298	{
1299	#ifdef DEBUG
1300	audioMixBufPrint(pMixBuf);
1301	#endif
1302	if (pcRead)
1303	*pcRead = 0;
1304	return VINF_SUCCESS;
1305	}
1306
1307	PAUDMIXBUF_FN_CONVTO pConv = audioMixBufConvToLookup(enmFmt);
1308	if (!pConv) /* Audio format not supported. */
1309	return VERR_NOT_SUPPORTED;
1310
1311	PPDMAUDIOSAMPLE pSamplesSrc1 = pMixBuf->pSamples + pMixBuf->offReadWrite;
1312	uint32_t cLenSrc1 = cToRead;
1313
1314	PPDMAUDIOSAMPLE pSamplesSrc2 = NULL;
1315	uint32_t cLenSrc2 = 0;
1316
1317	uint32_t offRead = pMixBuf->offReadWrite + cToRead;
1318
1319	/*
1320	* Do we need to wrap around to read all requested data, that is,
1321	* starting at the beginning of our circular buffer? This then will
1322	* be the optional second part to do.
1323	*/
1324	if (offRead >= pMixBuf->cSamples)
1325	{
1326	Assert(pMixBuf->offReadWrite <= pMixBuf->cSamples);
1327	cLenSrc1 = pMixBuf->cSamples - pMixBuf->offReadWrite;
1328
1329	pSamplesSrc2 = pMixBuf->pSamples;
1330	Assert(cToRead >= cLenSrc1);
1331	cLenSrc2 = RT_MIN(cToRead - cLenSrc1, pMixBuf->cSamples);
1332
1333	/* Save new read offset. */
1334	offRead = cLenSrc2;
1335	}
1336
1337	AUDMIXBUF_CONVOPTS convOpts;
1338	convOpts.Volume = pMixBuf->Volume;
1339
1340	/* Anything to do at all? */
1341	int rc = VINF_SUCCESS;
1342	if (cLenSrc1)
1343	{
1344	convOpts.cSamples = cLenSrc1;
1345
1346	AUDMIXBUF_LOG(("P1: offRead=%RU32, cToRead=%RU32\n", pMixBuf->offReadWrite, cLenSrc1));
1347	pConv(pvBuf, pSamplesSrc1, &convOpts);
1348	}
1349
1350	/* Second part present? */
1351	if ( RT_LIKELY(RT_SUCCESS(rc))
1352	&& cLenSrc2)
1353	{
1354	AssertPtr(pSamplesSrc2);
1355
1356	convOpts.cSamples = cLenSrc2;
1357
1358	AUDMIXBUF_LOG(("P2: cToRead=%RU32, offWrite=%RU32 (%zu bytes)\n", cLenSrc2, cLenSrc1,
1359	AUDIOMIXBUF_S2B(pMixBuf, cLenSrc1)));
1360	pConv((uint8_t *)pvBuf + AUDIOMIXBUF_S2B(pMixBuf, cLenSrc1), pSamplesSrc2, &convOpts);
1361	}
1362
1363	if (RT_SUCCESS(rc))
1364	{
1365	#ifdef DEBUG_DUMP_PCM_DATA
1366	RTFILE fh;
1367	rc = RTFileOpen(&fh, DEBUG_DUMP_PCM_DATA_PATH "mixbuf_readcirc.pcm",
1368	RTFILE_O_OPEN_CREATE \| RTFILE_O_APPEND \| RTFILE_O_WRITE \| RTFILE_O_DENY_NONE);
1369	if (RT_SUCCESS(rc))
1370	{
1371	RTFileWrite(fh, pvBuf, AUDIOMIXBUF_S2B(pMixBuf, cLenSrc1 + cLenSrc2), NULL);
1372	RTFileClose(fh);
1373	}
1374	#endif
1375	pMixBuf->offReadWrite = offRead % pMixBuf->cSamples;
1376	pMixBuf->cProcessed -= RT_MIN(cLenSrc1 + cLenSrc2, pMixBuf->cProcessed);
1377
1378	if (pcRead)
1379	*pcRead = cLenSrc1 + cLenSrc2;
1380	}
1381
1382	#ifdef DEBUG
1383	audioMixBufPrint(pMixBuf);
1384	#endif
1385
1386	AUDMIXBUF_LOG(("cRead=%RU32 (%zu bytes), rc=%Rrc\n",
1387	cLenSrc1 + cLenSrc2,
1388	AUDIOMIXBUF_S2B(pMixBuf, cLenSrc1 + cLenSrc2), rc));
1389	return rc;
1390	}
1391
1392	/**
1393	* Resets a mixing buffer.
1394	*
1395	* @param pMixBuf Mixing buffer to reset.
1396	*/
1397	void AudioMixBufReset(PPDMAUDIOMIXBUF pMixBuf)
1398	{
1399	AssertPtrReturnVoid(pMixBuf);
1400
1401	AUDMIXBUF_LOG(("%s\n", pMixBuf->pszName));
1402
1403	pMixBuf->offReadWrite = 0;
1404	pMixBuf->cMixed = 0;
1405	pMixBuf->cProcessed = 0;
1406
1407	AudioMixBufClear(pMixBuf);
1408	}
1409
1410	/**
1411	* Sets the overall (master) volume.
1412	*
1413	* @param pMixBuf Mixing buffer to set volume for.
1414	* @param pVol Pointer to volume structure to set.
1415	*/
1416	void AudioMixBufSetVolume(PPDMAUDIOMIXBUF pMixBuf, PPDMAUDIOVOLUME pVol)
1417	{
1418	AssertPtrReturnVoid(pMixBuf);
1419	AssertPtrReturnVoid(pVol);
1420
1421	LogFlowFunc(("%s: lVol=%RU32, rVol=%RU32\n", pMixBuf->pszName, pVol->uLeft, pVol->uRight));
1422
1423	pMixBuf->Volume.fMuted = pVol->fMuted;
1424	/** @todo Ensure that the input is in the correct range/initialized! */
1425	pMixBuf->Volume.uLeft = s_aVolumeConv[pVol->uLeft & 0xFF] * (AUDIOMIXBUF_VOL_0DB >> 16);
1426	pMixBuf->Volume.uRight = s_aVolumeConv[pVol->uRight & 0xFF] * (AUDIOMIXBUF_VOL_0DB >> 16);
1427
1428	LogFlowFunc(("\t-> lVol=%#RX32, rVol=%#RX32\n", pMixBuf->Volume.uLeft, pMixBuf->Volume.uRight));
1429	}
1430
1431	/**
1432	* Returns the maximum amount of audio samples this buffer can hold.
1433	*
1434	* @return uint32_t Size (in audio samples) the mixing buffer can hold.
1435	* @param pMixBuf Mixing buffer to retrieve maximum for.
1436	*/
1437	uint32_t AudioMixBufSize(PPDMAUDIOMIXBUF pMixBuf)
1438	{
1439	AssertPtrReturn(pMixBuf, 0);
1440	return pMixBuf->cSamples;
1441	}
1442
1443	/**
1444	* Returns the maximum amount of bytes this buffer can hold.
1445	*
1446	* @return uint32_t Size (in bytes) the mixing buffer can hold.
1447	* @param pMixBuf Mixing buffer to retrieve maximum for.
1448	*/
1449	uint32_t AudioMixBufSizeBytes(PPDMAUDIOMIXBUF pMixBuf)
1450	{
1451	AssertPtrReturn(pMixBuf, 0);
1452	return AUDIOMIXBUF_S2B(pMixBuf, pMixBuf->cSamples);
1453	}
1454
1455	/**
1456	* Unlinks a mixing buffer from its parent, if any.
1457	*
1458	* @return IPRT status code.
1459	* @param pMixBuf Mixing buffer to unlink from parent.
1460	*/
1461	void AudioMixBufUnlink(PPDMAUDIOMIXBUF pMixBuf)
1462	{
1463	if (!pMixBuf \|\| !pMixBuf->pszName)
1464	return;
1465
1466	AUDMIXBUF_LOG(("%s\n", pMixBuf->pszName));
1467
1468	if (pMixBuf->pParent)
1469	{
1470	AUDMIXBUF_LOG(("%s: Unlinking from parent \"%s\"\n",
1471	pMixBuf->pszName, pMixBuf->pParent->pszName));
1472
1473	RTListNodeRemove(&pMixBuf->Node);
1474
1475	/* Make sure to reset the parent mixing buffer each time it gets linked
1476	* to a new child. */
1477	AudioMixBufReset(pMixBuf->pParent);
1478	pMixBuf->pParent = NULL;
1479	}
1480
1481	PPDMAUDIOMIXBUF pIter;
1482	while (!RTListIsEmpty(&pMixBuf->lstBuffers))
1483	{
1484	pIter = RTListGetFirst(&pMixBuf->lstBuffers, PDMAUDIOMIXBUF, Node);
1485
1486	AUDMIXBUF_LOG(("\tUnlinking \"%s\"\n", pIter->pszName));
1487
1488	AudioMixBufReset(pIter->pParent);
1489	pIter->pParent = NULL;
1490
1491	RTListNodeRemove(&pIter->Node);
1492	}
1493
1494	if (pMixBuf->pRate)
1495	{
1496	pMixBuf->pRate->dstOffset = pMixBuf->pRate->srcOffset = 0;
1497	pMixBuf->pRate->dstInc = 0;
1498	}
1499
1500	pMixBuf->iFreqRatio = 1; /* Prevent division by zero. */
1501	}
1502
1503	/**
1504	* Writes audio samples at a specific offset.
1505	* The sample format being written must match the format of the mixing buffer.
1506	*
1507	* @return IPRT status code.
1508	* @param pMixBuf Pointer to mixing buffer to write to.
1509	* @param enmFmt Audio format supplied in the buffer.
1510	* @param offSamples Offset (in samples) starting to write at.
1511	* @param pvBuf Pointer to audio buffer to be written.
1512	* @param cbBuf Size (in bytes) of audio buffer.
1513	* @param pcWritten Returns number of audio samples written. Optional.
1514	*/
1515	int AudioMixBufWriteAt(PPDMAUDIOMIXBUF pMixBuf,
1516	uint32_t offSamples,
1517	const void *pvBuf, uint32_t cbBuf,
1518	uint32_t *pcWritten)
1519	{
1520	return AudioMixBufWriteAtEx(pMixBuf, pMixBuf->AudioFmt,
1521	offSamples, pvBuf, cbBuf, pcWritten);
1522	}
1523
1524	/**
1525	* Writes audio samples at a specific offset. The audio sample format
1526	* to be written can be different from the audio format the mixing buffer
1527	* operates on.
1528	*
1529	* @return IPRT status code.
1530	* @param pMixBuf Pointer to mixing buffer to write to.
1531	* @param enmFmt Audio format supplied in the buffer.
1532	* @param offSamples Offset (in samples) starting to write at.
1533	* @param pvBuf Pointer to audio buffer to be written.
1534	* @param cbBuf Size (in bytes) of audio buffer.
1535	* @param pcWritten Returns number of audio samples written. Optional.
1536	*/
1537	int AudioMixBufWriteAtEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt,
1538	uint32_t offSamples,
1539	const void *pvBuf, uint32_t cbBuf,
1540	uint32_t *pcWritten)
1541	{
1542	AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER);
1543	AssertPtrReturn(pvBuf, VERR_INVALID_POINTER);
1544	/* pcWritten is optional. */
1545
1546	uint32_t cDstSamples = pMixBuf->pParent
1547	? pMixBuf->pParent->cSamples : pMixBuf->cSamples;
1548	uint32_t cLive = pMixBuf->cProcessed;
1549
1550	uint32_t cDead = cDstSamples - cLive;
1551	uint32_t cToProcess = (uint32_t)AUDIOMIXBUF_S2S_RATIO(pMixBuf, cDead);
1552	cToProcess = RT_MIN(cToProcess, AUDIOMIXBUF_B2S(pMixBuf, cbBuf));
1553
1554	AUDMIXBUF_LOG(("%s: offSamples=%RU32, cLive=%RU32, cDead=%RU32, cToProcess=%RU32\n",
1555	pMixBuf->pszName, offSamples, cLive, cDead, cToProcess));
1556
1557	if (offSamples + cToProcess > pMixBuf->cSamples)
1558	return VERR_BUFFER_OVERFLOW;
1559
1560	PAUDMIXBUF_FN_CONVFROM pConv = audioMixBufConvFromLookup(enmFmt, pMixBuf->Volume.fMuted);
1561	if (!pConv)
1562	return VERR_NOT_SUPPORTED;
1563
1564	int rc;
1565	uint32_t cWritten;
1566
1567	#ifdef DEBUG_DUMP_PCM_DATA
1568	RTFILE fh;
1569	rc = RTFileOpen(&fh, DEBUG_DUMP_PCM_DATA_PATH "mixbuf_writeat.pcm",
1570	RTFILE_O_OPEN_CREATE \| RTFILE_O_APPEND \| RTFILE_O_WRITE \| RTFILE_O_DENY_NONE);
1571	if (RT_SUCCESS(rc))
1572	{
1573	RTFileWrite(fh, pvBuf, cbBuf, NULL);
1574	RTFileClose(fh);
1575	}
1576	#endif
1577
1578	if (cToProcess)
1579	{
1580	AUDMIXBUF_CONVOPTS convOpts = { cToProcess, pMixBuf->Volume };
1581
1582	cWritten = pConv(pMixBuf->pSamples + offSamples, pvBuf, cbBuf, &convOpts);
1583	#ifdef DEBUG
1584	audioMixBufPrint(pMixBuf);
1585	#endif
1586	rc = cWritten ? VINF_SUCCESS : VERR_GENERAL_FAILURE; /** @todo Fudge! */
1587	}
1588	else
1589	{
1590	cWritten = 0;
1591	rc = VINF_SUCCESS;
1592	}
1593
1594	if (RT_SUCCESS(rc))
1595	{
1596	if (pcWritten)
1597	*pcWritten = cWritten;
1598	}
1599
1600	AUDMIXBUF_LOG(("cWritten=%RU32, rc=%Rrc\n", cWritten, rc));
1601	return rc;
1602	}
1603
1604	/**
1605	* Writes audio samples. The sample format being written must match the
1606	* format of the mixing buffer.
1607	*
1608	* @return IPRT status code.
1609	* @param pMixBuf Pointer to mixing buffer to write to.
1610	* @param pvBuf Pointer to audio buffer to be written.
1611	* @param cbBuf Size (in bytes) of audio buffer.
1612	* @param pcWritten Returns number of audio samples written. Optional.
1613	*/
1614	int AudioMixBufWriteCirc(PPDMAUDIOMIXBUF pMixBuf,
1615	const void *pvBuf, uint32_t cbBuf,
1616	uint32_t *pcWritten)
1617	{
1618	return AudioMixBufWriteCircEx(pMixBuf, pMixBuf->AudioFmt, pvBuf, cbBuf, pcWritten);
1619	}
1620
1621	/**
1622	* Writes audio samples of a specific format.
1623	*
1624	* @return IPRT status code.
1625	* @param pMixBuf Pointer to mixing buffer to write to.
1626	* @param enmFmt Audio format supplied in the buffer.
1627	* @param pvBuf Pointer to audio buffer to be written.
1628	* @param cbBuf Size (in bytes) of audio buffer.
1629	* @param pcWritten Returns number of audio samples written. Optional.
1630	*/
1631	int AudioMixBufWriteCircEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt,
1632	const void *pvBuf, uint32_t cbBuf,
1633	uint32_t *pcWritten)
1634	{
1635	AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER);
1636	AssertPtrReturn(pvBuf, VERR_INVALID_POINTER);
1637	/* pcbWritten is optional. */
1638
1639	if (!cbBuf)
1640	{
1641	if (pcWritten)
1642	*pcWritten = 0;
1643	return VINF_SUCCESS;
1644	}
1645
1646	PPDMAUDIOMIXBUF pParent = pMixBuf->pParent;
1647
1648	AUDMIXBUF_LOG(("%s: enmFmt=%ld, pBuf=%p, cbBuf=%zu, pParent=%p (%RU32)\n",
1649	pMixBuf->pszName, enmFmt, pvBuf, cbBuf, pParent, pParent ? pParent->cSamples : 0));
1650
1651	if ( pParent
1652	&& pParent->cSamples <= pMixBuf->cMixed)
1653	{
1654	if (pcWritten)
1655	*pcWritten = 0;
1656
1657	AUDMIXBUF_LOG(("%s: Parent buffer %s is full\n",
1658	pMixBuf->pszName, pMixBuf->pParent->pszName));
1659
1660	return VINF_SUCCESS;
1661	}
1662
1663	PAUDMIXBUF_FN_CONVFROM pConv = audioMixBufConvFromLookup(enmFmt, pMixBuf->Volume.fMuted);
1664	if (!pConv)
1665	return VERR_NOT_SUPPORTED;
1666
1667	int rc = VINF_SUCCESS;
1668
1669	uint32_t cToWrite = AUDIOMIXBUF_B2S(pMixBuf, cbBuf);
1670	AssertMsg(cToWrite, ("cToWrite is 0 (cbBuf=%zu)\n", cbBuf));
1671
1672	PPDMAUDIOSAMPLE pSamplesDst1 = pMixBuf->pSamples + pMixBuf->offReadWrite;
1673	uint32_t cLenDst1 = cToWrite;
1674
1675	PPDMAUDIOSAMPLE pSamplesDst2 = NULL;
1676	uint32_t cLenDst2 = 0;
1677
1678	uint32_t offWrite = pMixBuf->offReadWrite + cToWrite;
1679
1680	/*
1681	* Do we need to wrap around to write all requested data, that is,
1682	* starting at the beginning of our circular buffer? This then will
1683	* be the optional second part to do.
1684	*/
1685	if (offWrite >= pMixBuf->cSamples)
1686	{
1687	Assert(pMixBuf->offReadWrite <= pMixBuf->cSamples);
1688	cLenDst1 = pMixBuf->cSamples - pMixBuf->offReadWrite;
1689
1690	pSamplesDst2 = pMixBuf->pSamples;
1691	Assert(cToWrite >= cLenDst1);
1692	cLenDst2 = RT_MIN(cToWrite - cLenDst1, pMixBuf->cSamples);
1693
1694	/* Save new read offset. */
1695	offWrite = cLenDst2;
1696	}
1697
1698	uint32_t cWrittenTotal = 0;
1699
1700	AUDMIXBUF_CONVOPTS convOpts;
1701	convOpts.Volume = pMixBuf->Volume;
1702
1703	/* Anything to do at all? */
1704	if (cLenDst1)
1705	{
1706	convOpts.cSamples = cLenDst1;
1707	cWrittenTotal = pConv(pSamplesDst1, pvBuf, cbBuf, &convOpts);
1708	}
1709
1710	/* Second part present? */
1711	if ( RT_LIKELY(RT_SUCCESS(rc))
1712	&& cLenDst2)
1713	{
1714	AssertPtr(pSamplesDst2);
1715
1716	convOpts.cSamples = cLenDst2;
1717	cWrittenTotal += pConv(pSamplesDst2, (uint8_t *)pvBuf + AUDIOMIXBUF_S2B(pMixBuf, cLenDst1), cbBuf, &convOpts);
1718	}
1719
1720	#ifdef DEBUG_DUMP_PCM_DATA
1721	RTFILE fh;
1722	RTFileOpen(&fh, DEBUG_DUMP_PCM_DATA_PATH "mixbuf_writeex.pcm",
1723	RTFILE_O_OPEN_CREATE \| RTFILE_O_APPEND \| RTFILE_O_WRITE \| RTFILE_O_DENY_NONE);
1724	RTFileWrite(fh, pSamplesDst1, AUDIOMIXBUF_S2B(pMixBuf, cLenDst1), NULL);
1725	RTFileClose(fh);
1726	#endif
1727
1728	AUDMIXBUF_LOG(("cLenDst1=%RU32, cLenDst2=%RU32, offWrite=%RU32\n",
1729	cLenDst1, cLenDst2, offWrite));
1730
1731	if (RT_SUCCESS(rc))
1732	{
1733	pMixBuf->offReadWrite = offWrite % pMixBuf->cSamples;
1734	pMixBuf->cProcessed = RT_MIN(pMixBuf->cProcessed + cLenDst1 + cLenDst2,
1735	pMixBuf->cSamples /* Max */);
1736	if (pcWritten)
1737	*pcWritten = cLenDst1 + cLenDst2;
1738	}
1739
1740	#ifdef DEBUG
1741	audioMixBufPrint(pMixBuf);
1742	#endif
1743
1744	AUDMIXBUF_LOG(("cWritten=%RU32 (%zu bytes), rc=%Rrc\n",
1745	cLenDst1 + cLenDst2,
1746	AUDIOMIXBUF_S2B(pMixBuf, cLenDst1 + cLenDst2), rc));
1747	return rc;
1748	}
1749

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source: vbox/trunk/src/VBox/Devices/Audio/AudioMixBuffer.cpp@ 60066

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