Changeset 59987 in vbox for trunk/src/VBox/Devices

Timestamp:

Mar 11, 2016 12:03:37 PM (9 years ago)

Author:

vboxsync

svn:sync-xref-src-repo-rev:

105954

Message:

Audio: Decoupled backend sinks and sources from the maximum of concurrent streams a backend can handle. Also, added some more enumeration code to the ALSA, PulseAudio and OSS backends, which later also can be used for configuration change callbacks. Some function renaming.

Location:

trunk/src/VBox/Devices/Audio

Files:

• DevIchAc97.cpp (modified) (4 diffs)
• DevIchHda.cpp (modified) (2 diffs)
• DrvAudio.cpp (modified) (12 diffs)
• DrvAudio.h (modified) (1 diff)
• DrvHostALSAAudio.cpp (modified) (35 diffs)
• DrvHostCoreAudio.cpp (modified) (32 diffs)
• DrvHostDSound.cpp (modified) (12 diffs)
• DrvHostNullAudio.cpp (modified) (2 diffs)
• DrvHostOSSAudio.cpp (modified) (45 diffs)
• DrvHostPulseAudio.cpp (modified) (51 diffs)
• alsa_mangling.h (modified) (2 diffs)
• alsa_stubs.c (modified) (3 diffs)
• pulse_mangling.h (modified) (2 diffs)
• pulse_stubs.c (modified) (4 diffs)

trunk/src/VBox/Devices/Audio/DevIchAc97.cpp

@@ -642 +642 @@
             pStrmIn = &pDrv->LineIn;
 
-        rc = pDrv->pConnector->pfnCreateIn(pDrv->pConnector, pszDesc, enmRecSource, pCfg, &pStrmIn->pStrmIn);
+        int rc2 = pDrv->pConnector->pfnCreateIn(pDrv->pConnector, pszDesc, enmRecSource, pCfg, &pStrmIn->pStrmIn);
 
         LogFlowFunc(("LUN#%RU8: Created input \"%s\", with rc=%Rrc\n", pDrv->uLUN, pszDesc, rc));
-        if (rc == VINF_SUCCESS) /* Note: Could return VWRN_ALREADY_EXISTS. */
+        if (rc2 == VINF_SUCCESS) /* Note: Could return VWRN_ALREADY_EXISTS. */
         {
             AudioMixerRemoveStream(pSink, pStrmIn->phStrmIn);
-            rc = AudioMixerAddStreamIn(pSink,
-                                       pDrv->pConnector, pStrmIn->pStrmIn,
-                                       0 /* uFlags */, &pStrmIn->phStrmIn);
+            rc2 = AudioMixerAddStreamIn(pSink,
+                                        pDrv->pConnector, pStrmIn->pStrmIn,
+                                        0 /* uFlags */, &pStrmIn->phStrmIn);
         }
 
@@ -2534 +2534 @@
             AssertPtr(pCon);
 
-            bool fValidLineIn = pCon->pfnIsValidIn(pCon, pDrv->LineIn.pStrmIn);
+            bool fValidLineIn = pCon->pfnIsValidIn (pCon, pDrv->LineIn.pStrmIn);
             bool fValidMicIn  = pCon->pfnIsValidIn (pCon, pDrv->MicIn.pStrmIn);
             bool fValidOut    = pCon->pfnIsValidOut(pCon, pDrv->Out.pStrmOut);
@@ -2559 +2559 @@
                 if (RT_SUCCESS(rc2))
                 {
-                    if (backendCfg.cMaxHstStrmsIn)
+                    if (backendCfg.cSources)
                     {
                         /* If the audio backend supports two or more input streams at once,
                          * warn if one of our two inputs (microphone-in and line-in) failed to initialize. */
-                        if (backendCfg.cMaxHstStrmsIn >= 2)
+                        if (backendCfg.cMaxStreamsIn >= 2)
                             fWarn = !fValidLineIn || !fValidMicIn;
                         /* If the audio backend only supports one input stream at once (e.g. pure ALSA, and
                          * *not* ALSA via PulseAudio plugin!), only warn if both of our inputs failed to initialize.
                          * One of the two simply is not in use then. */
-                        else if (backendCfg.cMaxHstStrmsIn == 1)
+                        else if (backendCfg.cMaxStreamsIn == 1)
                             fWarn = !fValidLineIn && !fValidMicIn;
                         /* Don't warn if our backend is not able of supporting any input streams at all. */
@@ -2574 +2574 @@
 
                     if (   !fWarn
-                        && backendCfg.cMaxHstStrmsOut)
+                        && backendCfg.cSinks)
                     {
                         fWarn = !fValidOut;

trunk/src/VBox/Devices/Audio/DevIchHda.cpp

@@ -4807 +4807 @@
                 if (RT_SUCCESS(rc2))
                 {
-                    if (backendCfg.cMaxHstStrmsIn)
+                    if (backendCfg.cSources)
                     {
 #ifdef VBOX_WITH_HDA_MIC_IN
                         /* If the audio backend supports two or more input streams at once,
                          * warn if one of our two inputs (microphone-in and line-in) failed to initialize. */
-                        if (backendCfg.cMaxHstStrmsIn >= 2)
+                        if (backendCfg.cMaxStreamsIn >= 2)
                             fWarn = !fValidLineIn || !fValidMicIn;
                         /* If the audio backend only supports one input stream at once (e.g. pure ALSA, and
                          * *not* ALSA via PulseAudio plugin!), only warn if both of our inputs failed to initialize.
                          * One of the two simply is not in use then. */
-                        else if (backendCfg.cMaxHstStrmsIn == 1)
+                        else if (backendCfg.cMaxStreamsIn == 1)
                             fWarn = !fValidLineIn && !fValidMicIn;
                         /* Don't warn if our backend is not able of supporting any input streams at all. */
@@ -4827 +4827 @@
 
                     if (   !fWarn
-                        && backendCfg.cMaxHstStrmsOut)
+                        && backendCfg.cSinks)
                     {
                         fWarn = !fValidOut;

trunk/src/VBox/Devices/Audio/DrvAudio.cpp

@@ -615 +615 @@
 
             RTMemFree(pHstStrmOut);
-            pThis->cFreeOutputStreams++;
+            pThis->cStreamsFreeOut++;
             return VINF_SUCCESS;
         }
@@ -749 +749 @@
     AssertPtrReturn(pCfg, VERR_INVALID_POINTER);
 
-    if (!pThis->cFreeOutputStreams)
+    if (!pThis->cStreamsFreeOut)
     {
         LogFlowFunc(("Maximum number of host output streams reached\n"));
@@ -801 +801 @@
         {
             RTListPrepend(&pThis->lstHstStrmOut, &pHstStrmOut->Node);
-            pThis->cFreeOutputStreams--;
+            pThis->cStreamsFreeOut--;
         }
 
@@ -1005 +1005 @@
                               PDMAUDIORECSOURCE enmRecSource, PPDMAUDIOHSTSTRMIN *ppHstStrmIn)
 {
-    if (!pThis->cFreeInputStreams)
+    if (!pThis->cStreamsFreeIn)
     {
         LogFlowFunc(("No more input streams free to use, bailing out\n"));
@@ -1057 +1057 @@
         {
             RTListPrepend(&pThis->lstHstStrmIn, &pHstStrmIn->Node);
-            pThis->cFreeInputStreams--;
+            pThis->cStreamsFreeIn--;
         }
 
@@ -1237 +1237 @@
 
             RTMemFree(pHstStrmIn);
-            pThis->cFreeInputStreams++;
+            pThis->cStreamsFreeIn++;
         }
     }
@@ -1422 +1422 @@
         AssertRC(rc);
 
-        if (!pThis->BackendCfg.cMaxHstStrmsOut)
+        if (!pThis->BackendCfg.cMaxStreamsOut)
         {
             int rc2 = RTCritSectLeave(&pThis->CritSect);
@@ -1669 +1669 @@
     }
 
-    uint32_t cMaxHstStrmsOut = pThis->BackendCfg.cMaxHstStrmsOut;
-    size_t cbHstStrmsOut     = pThis->BackendCfg.cbStreamOut;
-
-    if (cbHstStrmsOut)
-    {
-        pThis->cFreeOutputStreams = cMaxHstStrmsOut;
+    if (pThis->BackendCfg.cbStreamOut)
+    {
+        pThis->cStreamsFreeOut = pThis->BackendCfg.cMaxStreamsOut;
     }
     else
-        pThis->cFreeOutputStreams = 0;
-
-    uint32_t cMaxHstStrmsIn = pThis->BackendCfg.cMaxHstStrmsIn;
-    size_t cbHstStrmIn      = pThis->BackendCfg.cbStreamIn;
-
-    if (cbHstStrmIn)
+        pThis->cStreamsFreeOut = 0;
+
+    if (pThis->BackendCfg.cbStreamIn)
     {
         /*
@@ -1689 +1683 @@
          *  - Our HDA emulation currently has only line input (hda.pi).
          */
-        pThis->cFreeInputStreams = cMaxHstStrmsIn;
+        pThis->cStreamsFreeIn = pThis->BackendCfg.cMaxStreamsIn;
     }
     else
-        pThis->cFreeInputStreams = 0;
-
-    LogFlowFunc(("cMaxHstStrmsOut=%RU32 (cb=%zu), cMaxHstStrmsIn=%RU32 (cb=%zu)\n",
-                 cMaxHstStrmsOut, cbHstStrmsOut, cMaxHstStrmsIn, cbHstStrmIn));
-
-    LogFlowFunc(("cFreeInputStreams=%RU8, cFreeOutputStreams=%RU8\n",
-                 pThis->cFreeInputStreams, pThis->cFreeOutputStreams));
-
-    LogRel(("Audio: Host audio backend supports %RU32 output streams and %RU32 input streams at once\n",
-            /* Clamp for logging. Unlimited streams are defined by UINT32_MAX. */
-            RT_MIN(64, cMaxHstStrmsOut), RT_MIN(64, cMaxHstStrmsIn)));
+        pThis->cStreamsFreeIn = 0;
+
+    LogFlowFunc(("cStreamsFreeIn=%RU8, cStreamsFreeOut=%RU8\n", pThis->cStreamsFreeIn, pThis->cStreamsFreeOut));
+
+    LogRel2(("Audio: Host audio backend supports %RU32 input streams and %RU32 output streams at once\n",
+             /* Clamp for logging. Unlimited streams are defined by UINT32_MAX. */
+             RT_MIN(64, pThis->cStreamsFreeIn), RT_MIN(64, pThis->cStreamsFreeOut)));
 
     LogFlowFuncLeave();
@@ -1792 +1782 @@
         /** @todo Check for invalid options? */
 
-        pThis->cFreeOutputStreams = conf.fixed_out.cStreams;
-        pThis->cFreeInputStreams  = conf.fixed_in.cStreams;
-
-        if (!pThis->cFreeOutputStreams)
-            pThis->cFreeOutputStreams = 1;
-
-        if (!pThis->cFreeInputStreams)
-            pThis->cFreeInputStreams = 1;
+        pThis->cStreamsFreeOut = conf.fixed_out.cStreams;
+        pThis->cStreamsFreeIn  = conf.fixed_in.cStreams;
+
+        if (!pThis->cStreamsFreeOut)
+            pThis->cStreamsFreeOut = 1;
+
+        if (!pThis->cStreamsFreeIn)
+            pThis->cStreamsFreeIn = 1;
     }
 
@@ -2055 +2045 @@
             *ppGstStrmIn = pGstStrmIn;
     }
-    else
-    {
-        switch (rc)
-        {
-            case VERR_NO_MORE_HANDLES: /** @todo Find a better rc. */
-                LogRel(("Audio: Skipping to create input stream \"%s\", " \
-                        "as the host audio backend reached its maximum of concurrent audio input streams\n", pszName));
-                break;
-
-            default:
-                break;
-        }
-    }
 
     int rc2 = RTCritSectLeave(&pThis->CritSect);
@@ -2179 +2156 @@
         }
 #endif
-    }
-    else
-    {
-        switch (rc)
-        {
-            case VERR_NO_MORE_HANDLES: /** @todo Find a better rc. */
-                LogRel(("Audio: Skipping to create output stream \"%s\", " \
-                        "as the host audio backend reached its maximum of concurrent audio output streams\n", pszName));
-                break;
-
-            default:
-                break;
-        }
     }
 

trunk/src/VBox/Devices/Audio/DrvAudio.h

@@ -96 +96 @@
     /** Max. number of free input streams.
      *  UINT32_MAX for unlimited streams. */
-    uint32_t                cFreeInputStreams;
+    uint32_t                cStreamsFreeIn;
     /** Max. number of free output streams.
      *  UINT32_MAX for unlimited streams. */
-    uint32_t                cFreeOutputStreams;
+    uint32_t                cStreamsFreeOut;
     /** Audio configuration settings retrieved from the backend. */
     PDMAUDIOBACKENDCFG      BackendCfg;

trunk/src/VBox/Devices/Audio/DrvHostALSAAudio.cpp

@@ -5 +5 @@
 
 /*
- * Copyright (C) 2006-2015 Oracle Corporation
+ * Copyright (C) 2006-2016 Oracle Corporation
  *
  * This file is part of VirtualBox Open Source Edition (OSE), as
@@ -57 +57 @@
 
 #include <alsa/asoundlib.h>
+#include <alsa/control.h> /* For device enumeration. */
 
 #include "DrvAudio.h"
@@ -62 +63 @@
 
 #include "VBoxDD.h"
+
+/*********************************************************************************************************************************
+*   Defines                                                                                                                      *
+*********************************************************************************************************************************/
+
+/** Makes DRVHOSTALSAAUDIO out of PDMIHOSTAUDIO. */
+#define PDMIHOSTAUDIO_2_DRVHOSTALSAAUDIO(pInterface) \
+    ( (PDRVHOSTALSAAUDIO)((uintptr_t)pInterface - RT_OFFSETOF(DRVHOSTALSAAUDIO, IHostAudio)) )
+
+/*********************************************************************************************************************************
+*   Structures                                                                                                                   *
+*********************************************************************************************************************************/
 
 typedef struct ALSAAUDIOSTREAMIN
@@ -101 +114 @@
 } ALSAAUDIOCFG, *PALSAAUDIOCFG;
 
-static int drvHostALSAAudioRecover(snd_pcm_t *phPCM);
+static int alsaStreamRecover(snd_pcm_t *phPCM);
 
 static ALSAAUDIOCFG s_ALSAConf =
@@ -162 +175 @@
 } ALSAAUDIOSTREAMCFG, *PALSAAUDIOSTREAMCFG;
 
-static int drvHostALSAAudioClose(snd_pcm_t **pphPCM)
-{
-    if (!pphPCM || !*pphPCM)
-        return VINF_SUCCESS;
-
-    int rc;
-    int rc2 = snd_pcm_close(*pphPCM);
-    if (rc2)
-    {
-        LogRel(("ALSA: Closing PCM descriptor failed: %s\n", snd_strerror(rc2)));
-        rc = VERR_GENERAL_FAILURE; /** @todo */
-    }
-    else
-    {
-        *pphPCM = NULL;
-        rc = VINF_SUCCESS;
-    }
-
-    return rc;
-}
-
-static snd_pcm_format_t drvHostALSAAudioFmtToALSA(PDMAUDIOFMT fmt)
+
+
+static snd_pcm_format_t alsaAudioFmtToALSA(PDMAUDIOFMT fmt)
 {
     switch (fmt)
@@ -213 +207 @@
 }
 
-static int drvHostALSAAudioALSAToFmt(snd_pcm_format_t fmt,
-                                     PDMAUDIOFMT *pFmt, PDMAUDIOENDIANNESS *pEndianness)
+static int alsaALSAToAudioFmt(snd_pcm_format_t fmt,
+                              PDMAUDIOFMT *pFmt, PDMAUDIOENDIANNESS *pEndianness)
 {
     AssertPtrReturn(pFmt, VERR_INVALID_POINTER);
@@ -289 +283 @@
 }
 
-static int drvHostALSAAudioALSAGetShift(snd_pcm_format_t fmt, unsigned *puShift)
+static int alsaGetSampleShift(snd_pcm_format_t fmt, unsigned *puShift)
 {
     AssertPtrReturn(puShift, VERR_INVALID_POINTER);
@@ -322 +316 @@
 }
 
-static int drvHostALSAAudioSetThreshold(snd_pcm_t *phPCM,
-                                        snd_pcm_uframes_t threshold)
+static int alsaStreamSetThreshold(snd_pcm_t *phPCM, snd_pcm_uframes_t threshold)
 {
     snd_pcm_sw_params_t *pSWParms = NULL;
@@ -368 +361 @@
 }
 
-static int drvHostALSAAudioOpen(bool fIn,
-                                PALSAAUDIOSTREAMCFG pCfgReq,
-                                PALSAAUDIOSTREAMCFG pCfgObt,
-                                snd_pcm_t **pphPCM)
+static int alsaStreamClose(snd_pcm_t **pphPCM)
+{
+    if (!pphPCM || !*pphPCM)
+        return VINF_SUCCESS;
+
+    int rc;
+    int rc2 = snd_pcm_close(*pphPCM);
+    if (rc2)
+    {
+        LogRel(("ALSA: Closing PCM descriptor failed: %s\n", snd_strerror(rc2)));
+        rc = VERR_GENERAL_FAILURE; /** @todo */
+    }
+    else
+    {
+        *pphPCM = NULL;
+        rc = VINF_SUCCESS;
+    }
+
+    return rc;
+}
+
+static int alsaStreamOpen(bool fIn, PALSAAUDIOSTREAMCFG pCfgReq, PALSAAUDIOSTREAMCFG pCfgObt, snd_pcm_t **pphPCM)
 {
     snd_pcm_t *phPCM = NULL;
@@ -624 +635 @@
         {
             unsigned uShift;
-            rc = drvHostALSAAudioALSAGetShift(pCfgReq->fmt, &uShift);
+            rc = alsaGetSampleShift(pCfgReq->fmt, &uShift);
             if (RT_SUCCESS(rc))
             {
@@ -634 +645 @@
                     = (s_ALSAConf.threshold * bytes_per_sec) / 1000;
 
-                rc = drvHostALSAAudioSetThreshold(phPCM, threshold);
+                rc = alsaStreamSetThreshold(phPCM, threshold);
             }
         }
@@ -652 +663 @@
     }
     else
-        drvHostALSAAudioClose(&phPCM);
+        alsaStreamClose(&phPCM);
 
     LogFlowFuncLeaveRC(rc);
@@ -659 +670 @@
 
 #ifdef DEBUG
-static void drvHostALSAAudioErrorHandler(const char *file, int line, const char *function,
-                                         int err, const char *fmt, ...)
+static void alsaDbgErrorHandler(const char *file, int line, const char *function,
+                                int err, const char *fmt, ...)
 {
     /** @todo Implement me! */
@@ -666 +677 @@
 #endif
 
-static int drvHostALSAAudioGetAvail(snd_pcm_t *phPCM, snd_pcm_sframes_t *pFramesAvail)
+static int alsaStreamGetAvail(snd_pcm_t *phPCM, snd_pcm_sframes_t *pFramesAvail)
 {
     AssertPtrReturn(phPCM, VERR_INVALID_POINTER);
@@ -679 +690 @@
         if (framesAvail == -EPIPE)
         {
-            rc = drvHostALSAAudioRecover(phPCM);
+            rc = alsaStreamRecover(phPCM);
             if (RT_SUCCESS(rc))
                 framesAvail = snd_pcm_avail_update(phPCM);
@@ -695 +706 @@
 }
 
-static int drvHostALSAAudioRecover(snd_pcm_t *phPCM)
+static int alsaStreamRecover(snd_pcm_t *phPCM)
 {
     AssertPtrReturn(phPCM, VERR_INVALID_POINTER);
@@ -709 +720 @@
 }
 
-static int drvHostALSAAudioResume(snd_pcm_t *phPCM)
+static int alsaStreamResume(snd_pcm_t *phPCM)
 {
     AssertPtrReturn(phPCM, VERR_INVALID_POINTER);
@@ -760 +771 @@
     {
 #ifdef DEBUG
-        snd_lib_error_set_handler(drvHostALSAAudioErrorHandler);
+        snd_lib_error_set_handler(alsaDbgErrorHandler);
 #endif
     }
@@ -776 +787 @@
 
     snd_pcm_sframes_t cAvail;
-    int rc = drvHostALSAAudioGetAvail(pThisStrmIn->phPCM, &cAvail);
+    int rc = alsaStreamGetAvail(pThisStrmIn->phPCM, &cAvail);
     if (RT_FAILURE(rc))
     {
@@ -794 +805 @@
             case SND_PCM_STATE_SUSPENDED:
             {
-                rc = drvHostALSAAudioResume(pThisStrmIn->phPCM);
+                rc = alsaStreamResume(pThisStrmIn->phPCM);
                 if (RT_FAILURE(rc))
                     break;
@@ -860 +871 @@
                 case -EPIPE:
                 {
-                    rc = drvHostALSAAudioRecover(pThisStrmIn->phPCM);
+                    rc = alsaStreamRecover(pThisStrmIn->phPCM);
                     if (RT_FAILURE(rc))
                         break;
@@ -932 +943 @@
     {
         snd_pcm_sframes_t cAvail;
-        rc = drvHostALSAAudioGetAvail(pThisStrmOut->phPCM, &cAvail);
+        rc = alsaStreamGetAvail(pThisStrmOut->phPCM, &cAvail);
         if (RT_FAILURE(rc))
         {
@@ -975 +986 @@
                         case -EPIPE:
                         {
-                            rc = drvHostALSAAudioRecover(pThisStrmOut->phPCM);
+                            rc = alsaStreamRecover(pThisStrmOut->phPCM);
                             if (RT_FAILURE(rc))
                                 break;
@@ -986 +997 @@
                         {
                             /* Stream was suspended and waiting for a recovery. */
-                            rc = drvHostALSAAudioResume(pThisStrmOut->phPCM);
+                            rc = alsaStreamResume(pThisStrmOut->phPCM);
                             if (RT_FAILURE(rc))
                             {
@@ -1046 +1057 @@
     PALSAAUDIOSTREAMIN pThisStrmIn = (PALSAAUDIOSTREAMIN)pHstStrmIn;
 
-    drvHostALSAAudioClose(&pThisStrmIn->phPCM);
+    alsaStreamClose(&pThisStrmIn->phPCM);
 
     if (pThisStrmIn->pvBuf)
@@ -1064 +1075 @@
     PALSAAUDIOSTREAMOUT pThisStrmOut = (PALSAAUDIOSTREAMOUT)pHstStrmOut;
 
-    drvHostALSAAudioClose(&pThisStrmOut->phPCM);
+    alsaStreamClose(&pThisStrmOut->phPCM);
 
     if (pThisStrmOut->pvBuf)
@@ -1091 +1102 @@
     {
         ALSAAUDIOSTREAMCFG req;
-        req.fmt         = drvHostALSAAudioFmtToALSA(pCfg->enmFormat);
+        req.fmt         = alsaAudioFmtToALSA(pCfg->enmFormat);
         req.freq        = pCfg->uHz;
         req.nchannels   = pCfg->cChannels;
@@ -1098 +1109 @@
 
         ALSAAUDIOSTREAMCFG obt;
-        rc = drvHostALSAAudioOpen(false /* false */, &req, &obt, &phPCM);
+        rc = alsaStreamOpen(false /* false */, &req, &obt, &phPCM);
         if (RT_FAILURE(rc))
             break;
@@ -1104 +1115 @@
         PDMAUDIOFMT enmFormat;
         PDMAUDIOENDIANNESS enmEnd;
-        rc = drvHostALSAAudioALSAToFmt(obt.fmt, &enmFormat, &enmEnd);
+        rc = alsaALSAToAudioFmt(obt.fmt, &enmFormat, &enmEnd);
         if (RT_FAILURE(rc))
             break;
@@ -1139 +1150 @@
 
     if (RT_FAILURE(rc))
-        drvHostALSAAudioClose(&phPCM);
+        alsaStreamClose(&phPCM);
 
     LogFlowFuncLeaveRC(rc);
@@ -1162 +1173 @@
     {
         ALSAAUDIOSTREAMCFG req;
-        req.fmt         = drvHostALSAAudioFmtToALSA(pCfg->enmFormat);
+        req.fmt         = alsaAudioFmtToALSA(pCfg->enmFormat);
         req.freq        = pCfg->uHz;
         req.nchannels   = pCfg->cChannels;
@@ -1169 +1180 @@
 
         ALSAAUDIOSTREAMCFG obt;
-        rc = drvHostALSAAudioOpen(true /* fIn */, &req, &obt, &phPCM);
+        rc = alsaStreamOpen(true /* fIn */, &req, &obt, &phPCM);
         if (RT_FAILURE(rc))
             break;
@@ -1175 +1186 @@
         PDMAUDIOFMT enmFormat;
         PDMAUDIOENDIANNESS enmEnd;
-        rc = drvHostALSAAudioALSAToFmt(obt.fmt, &enmFormat, &enmEnd);
+        rc = alsaALSAToAudioFmt(obt.fmt, &enmFormat, &enmEnd);
         if (RT_FAILURE(rc))
             break;
@@ -1210 +1221 @@
 
     if (RT_FAILURE(rc))
-        drvHostALSAAudioClose(&phPCM);
+        alsaStreamClose(&phPCM);
 
     LogFlowFuncLeaveRC(rc);
@@ -1290 +1301 @@
     AssertPtrReturn(pCfg, VERR_INVALID_POINTER);
 
-    pCfg->cbStreamIn      = sizeof(ALSAAUDIOSTREAMIN);
-    pCfg->cbStreamOut     = sizeof(ALSAAUDIOSTREAMOUT);
+    pCfg->cbStreamIn  = sizeof(ALSAAUDIOSTREAMIN);
+    pCfg->cbStreamOut = sizeof(ALSAAUDIOSTREAMOUT);
+
+    pCfg->cSources    = 0;
+    pCfg->cSinks      = 0;
+
+    /* Enumerate sound devices. */
+    char **pszHints;
+    int err = snd_device_name_hint(-1 /* All cards */, "pcm", (void***)&pszHints);
+    if (err == 0)
+    {
+        char** pszHintCur = pszHints;
+        while (*pszHintCur != NULL)
+        {
+            char *pszDev = snd_device_name_get_hint(*pszHintCur, "NAME");
+            bool fSkip =    !pszDev
+                         || !RTStrICmp("null", pszDev);
+            if (fSkip)
+            {
+                if (pszDev)
+                    free(pszDev);
+                pszHintCur++;
+                continue;
+            }
+
+            char *pszIOID = snd_device_name_get_hint(*pszHintCur, "IOID");
+            if (pszIOID)
+            {
+                if (!RTStrICmp("input", pszIOID))
+                    pCfg->cSources++;
+                else if (!RTStrICmp("output", pszIOID))
+                    pCfg->cSinks++;
+            }
+            else /* NULL means bidirectional, input + output. */
+            {
+                pCfg->cSources++;
+                pCfg->cSinks++;
+            }
+
+            LogRel2(("ALSA: Found %s device: %s\n", pszIOID ?  RTStrToLower(pszIOID) : "bidirectional", pszDev));
+
+            /* Special case for PulseAudio. */
+            if (   pszDev
+                && RTStrIStr("pulse", pszDev) != NULL)
+                LogRel2(("ALSA: PulseAudio plugin in use\n"));
+
+            if (pszIOID)
+                free(pszIOID);
+
+            if (pszDev)
+                free(pszDev);
+
+            pszHintCur++;
+        }
+
+        LogRel2(("ALSA: Found %RU8 host playback devices\n",  pCfg->cSinks));
+        LogRel2(("ALSA: Found %RU8 host capturing devices\n", pCfg->cSources));
+
+        snd_device_name_free_hint((void **)pszHints);
+        pszHints = NULL;
+    }
+    else
+        LogRel2(("ALSA: Error enumerating PCM devices: %Rrc (%d)\n", RTErrConvertFromErrno(err), err));
 
     /* ALSA only allows one input and one output used at a time for
-     * the selected device. */
-    pCfg->cMaxHstStrmsIn  = 1;
-    pCfg->cMaxHstStrmsOut = 1;
+     * the selected device(s). */
+    pCfg->cMaxStreamsIn   = 1;
+    pCfg->cMaxStreamsOut  = 1;
 
     return VINF_SUCCESS;

trunk/src/VBox/Devices/Audio/DrvHostCoreAudio.cpp

@@ -311 +311 @@
 } COREAUDIOSTREAMIN, *PCOREAUDIOSTREAMIN;
 
+
+static int coreAudioInitIn(PPDMAUDIOHSTSTRMIN pHstStrmIn, uint32_t *pcSamples);
+static int coreAudioReinitIn(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMIN pHstStrmIn);
+static int coreAudioInitOut(PPDMAUDIOHSTSTRMOUT pHstStrmOut, uint32_t *pcSamples);
+static int coreAudioReinitOut(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMOUT pHstStrmOut);
+static OSStatus coreAudioPlaybackAudioDevicePropertyChanged(AudioObjectID propertyID, UInt32 nAddresses, const AudioObjectPropertyAddress properties[], void *pvUser);
+static OSStatus coreAudioPlaybackCb(void *pvUser, AudioUnitRenderActionFlags *pActionFlags, const AudioTimeStamp *pAudioTS, UInt32 uBusID, UInt32 cFrames, AudioBufferList* pBufData);
+
 static int drvHostCoreAudioControlIn(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMIN pHstStrmIn, PDMAUDIOSTREAMCMD enmStreamCmd);
-static int drvHostCoreAudioInitInput(PPDMAUDIOHSTSTRMIN pHstStrmIn, uint32_t *pcSamples);
+static int drvHostCoreAudioControlOut(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMOUT pHstStrmOut, PDMAUDIOSTREAMCMD enmStreamCmd);
 static int drvHostCoreAudioFiniIn(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMIN pHstStrmIn);
-static int drvHostCoreAudioReinitInput(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMIN pHstStrmIn);
-
-static int drvHostCoreAudioControlOut(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMOUT pHstStrmOut, PDMAUDIOSTREAMCMD enmStreamCmd);
-static int drvHostCoreAudioInitOutput(PPDMAUDIOHSTSTRMOUT pHstStrmOut, uint32_t *pcSamples);
 static int drvHostCoreAudioFiniOut(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMOUT pHstStrmOut);
-static int drvHostCoreAudioReinitOutput(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMOUT pHstStrmOut);
-static OSStatus drvHostCoreAudioPlaybackAudioDevicePropertyChanged(AudioObjectID propertyID, UInt32 nAddresses, const AudioObjectPropertyAddress properties[], void *pvUser);
-static OSStatus drvHostCoreAudioPlaybackCallback(void *pvUser, AudioUnitRenderActionFlags *pActionFlags, const AudioTimeStamp *pAudioTS, UInt32 uBusID, UInt32 cFrames, AudioBufferList* pBufData);
 
 /* Callback for getting notified when the default input/output device has been changed. */
-static DECLCALLBACK(OSStatus) drvHostCoreAudioDefaultDeviceChanged(AudioObjectID propertyID,
-                                                                   UInt32 nAddresses,
-                                                                   const AudioObjectPropertyAddress properties[],
-                                                                   void *pvUser)
+static DECLCALLBACK(OSStatus) coreAudioDefaultDeviceChanged(AudioObjectID propertyID,
+                                                            UInt32 nAddresses,
+                                                            const AudioObjectPropertyAddress properties[],
+                                                            void *pvUser)
 {
     OSStatus err = noErr;
@@ -397 +398 @@
     }
 
+    /** @todo Implement callback notification here to let the audio connector / device emulation
+     *        know that something has changed. */
+
     return noErr;
 }
 
-static int drvHostCoreAudioReinitInput(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMIN pHstStrmIn)
+static int coreAudioReinitIn(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMIN pHstStrmIn)
 {
     PPDMDRVINS pDrvIns      = PDMIBASE_2_PDMDRV(pInterface);
@@ -409 +413 @@
     drvHostCoreAudioFiniIn(pInterface, &pStreamIn->streamIn);
 
-    drvHostCoreAudioInitInput(&pStreamIn->streamIn, NULL /* pcSamples */);
+    coreAudioInitIn(&pStreamIn->streamIn, NULL /* pcSamples */);
     drvHostCoreAudioControlIn(pInterface, &pStreamIn->streamIn, PDMAUDIOSTREAMCMD_ENABLE);
 
@@ -415 +419 @@
 }
 
-static int drvHostCoreAudioReinitOutput(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMOUT pHstStrmOut)
+static int coreAudioReinitOut(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMOUT pHstStrmOut)
 {
     PPDMDRVINS pDrvIns      = PDMIBASE_2_PDMDRV(pInterface);
@@ -424 +428 @@
     drvHostCoreAudioFiniOut(pInterface, &pStreamOut->streamOut);
 
-    drvHostCoreAudioInitOutput(&pStreamOut->streamOut, NULL /* pcSamples */);
+    coreAudioInitOut(&pStreamOut->streamOut, NULL /* pcSamples */);
     drvHostCoreAudioControlOut(pInterface, &pStreamOut->streamOut, PDMAUDIOSTREAMCMD_ENABLE);
 
@@ -431 +435 @@
 
 /* Callback for getting notified when some of the properties of an audio device has changed. */
-static DECLCALLBACK(OSStatus) drvHostCoreAudioRecordingAudioDevicePropertyChanged(AudioObjectID                     propertyID,
-                                                                                  UInt32                            cAdresses,
-                                                                                  const AudioObjectPropertyAddress  aProperties[],
-                                                                                  void                             *pvUser)
+static DECLCALLBACK(OSStatus) coreAudioRecordingAudioDevicePropertyChanged(AudioObjectID                     propertyID,
+                                                                           UInt32                            cAdresses,
+                                                                           const AudioObjectPropertyAddress  aProperties[],
+                                                                           void                             *pvUser)
 {
     PCOREAUDIOSTREAMIN pStreamIn = (PCOREAUDIOSTREAMIN)pvUser;
@@ -466 +470 @@
 
 /* Callback to convert audio input data from one format to another. */
-static DECLCALLBACK(OSStatus) drvHostCoreAudioConverterCallback(AudioConverterRef              converterID,
-                                                                UInt32                        *pcPackets,
-                                                                AudioBufferList               *pBufData,
-                                                                AudioStreamPacketDescription **ppPacketDesc,
-                                                                void                          *pvUser)
+static DECLCALLBACK(OSStatus) coreAudioConverterCb(AudioConverterRef              converterID,
+                                                   UInt32                        *pcPackets,
+                                                   AudioBufferList               *pBufData,
+                                                   AudioStreamPacketDescription **ppPacketDesc,
+                                                   void                          *pvUser)
 {
     /** @todo Check incoming pointers. */
@@ -521 +525 @@
 
 /* Callback to feed audio input buffer. */
-static DECLCALLBACK(OSStatus) drvHostCoreAudioRecordingCallback(void                       *pvUser,
-                                                                AudioUnitRenderActionFlags *pActionFlags,
-                                                                const AudioTimeStamp       *pAudioTS,
-                                                                UInt32                      uBusID,
-                                                                UInt32                      cFrames,
-                                                                AudioBufferList            *pBufData)
+static DECLCALLBACK(OSStatus) coreAudioRecordingCb(void *pvUser,
+                                                   AudioUnitRenderActionFlags *pActionFlags,
+                                                   const AudioTimeStamp       *pAudioTS,
+                                                   UInt32                      uBusID,
+                                                   UInt32                      cFrames,
+                                                   AudioBufferList            *pBufData)
 {
     PCOREAUDIOSTREAMIN pStreamIn  = (PCOREAUDIOSTREAMIN)pvUser;
@@ -596 +600 @@
                 AudioConverterReset(pStreamIn->converter);
 
-                err = AudioConverterFillComplexBuffer(pStreamIn->converter, drvHostCoreAudioConverterCallback, pStreamIn,
+                err = AudioConverterFillComplexBuffer(pStreamIn->converter, coreAudioConverterCb, pStreamIn,
                                                       &ioOutputDataPacketSize, &tmpList, NULL);
                 if(   err != noErr
@@ -692 +696 @@
 
 /** @todo Eventually split up this function, as this already is huge! */
-static int drvHostCoreAudioInitInput(PPDMAUDIOHSTSTRMIN pHstStrmIn, uint32_t *pcSamples)
+static int coreAudioInitIn(PPDMAUDIOHSTSTRMIN pHstStrmIn, uint32_t *pcSamples)
 {
     OSStatus err = noErr;
@@ -833 +837 @@
     AURenderCallbackStruct cb;
     RT_ZERO(cb);
-    cb.inputProc       = drvHostCoreAudioRecordingCallback;
+    cb.inputProc       = coreAudioRecordingCb;
     cb.inputProcRefCon = pStreamIn;
 
@@ -1023 +1027 @@
         propAdr.mScope    = kAudioUnitScope_Global;
         err = AudioObjectAddPropertyListener(pStreamIn->deviceID, &propAdr,
-                                             drvHostCoreAudioRecordingAudioDevicePropertyChanged, (void *)pStreamIn);
+                                             coreAudioRecordingAudioDevicePropertyChanged, (void *)pStreamIn);
         if (RT_UNLIKELY(err != noErr))
             LogRel(("CoreAudio: Failed to add the processor overload listener for input stream (%RI32)\n", err));
@@ -1030 +1034 @@
         propAdr.mScope    = kAudioUnitScope_Global;
         err = AudioObjectAddPropertyListener(pStreamIn->deviceID, &propAdr,
-                                             drvHostCoreAudioRecordingAudioDevicePropertyChanged, (void *)pStreamIn);
+                                             coreAudioRecordingAudioDevicePropertyChanged, (void *)pStreamIn);
         /* Not fatal. */
         if (RT_UNLIKELY(err != noErr))
@@ -1059 +1063 @@
 
 /** @todo Eventually split up this function, as this already is huge! */
-static int drvHostCoreAudioInitOutput(PPDMAUDIOHSTSTRMOUT pHstStrmOut, uint32_t *pcSamples)
+static int coreAudioInitOut(PPDMAUDIOHSTSTRMOUT pHstStrmOut, uint32_t *pcSamples)
 {
     PCOREAUDIOSTREAMOUT pStreamOut = (PCOREAUDIOSTREAMOUT)pHstStrmOut;
@@ -1188 +1192 @@
     AURenderCallbackStruct cb;
     RT_ZERO(cb);
-    cb.inputProc       = drvHostCoreAudioPlaybackCallback; /* pvUser */
+    cb.inputProc       = coreAudioPlaybackCb; /* pvUser */
     cb.inputProcRefCon = pStreamOut;
 
@@ -1297 +1301 @@
         propAdr.mScope    = kAudioUnitScope_Global;
         err = AudioObjectAddPropertyListener(pStreamOut->deviceID, &propAdr,
-                                             drvHostCoreAudioPlaybackAudioDevicePropertyChanged, (void *)pStreamOut);
+                                             coreAudioPlaybackAudioDevicePropertyChanged, (void *)pStreamOut);
         if (err != noErr)
             LogRel(("CoreAudio: Failed to register processor overload listener for output stream (%RI32)\n", err));
@@ -1305 +1309 @@
         propAdr.mScope    = kAudioUnitScope_Global;
         err = AudioObjectAddPropertyListener(pStreamOut->deviceID, &propAdr,
-                                             drvHostCoreAudioPlaybackAudioDevicePropertyChanged, (void *)pStreamOut);
+                                             coreAudioPlaybackAudioDevicePropertyChanged, (void *)pStreamOut);
         /* Not fatal. */
         if (err != noErr)
@@ -1333 +1337 @@
 }
 
+
+/* Callback for getting notified when some of the properties of an audio device has changed. */
+static DECLCALLBACK(OSStatus) coreAudioPlaybackAudioDevicePropertyChanged(AudioObjectID propertyID,
+                                                                          UInt32 nAddresses,
+                                                                          const AudioObjectPropertyAddress properties[],
+                                                                          void *pvUser)
+{
+    switch (propertyID)
+    {
+#ifdef DEBUG
+        case kAudioDeviceProcessorOverload:
+        {
+            Log2(("CoreAudio: [Output] Processor overload detected!\n"));
+            break;
+        }
+#endif /* DEBUG */
+        default:
+            break;
+    }
+
+    return noErr;
+}
+
+/* Callback to feed audio output buffer. */
+static DECLCALLBACK(OSStatus) coreAudioPlaybackCb(void *pvUser,
+                                                  AudioUnitRenderActionFlags *pActionFlags,
+                                                  const AudioTimeStamp       *pAudioTS,
+                                                  UInt32                      uBusID,
+                                                  UInt32                      cFrames,
+                                                  AudioBufferList            *pBufData)
+{
+    PCOREAUDIOSTREAMOUT pStreamOut = (PCOREAUDIOSTREAMOUT)pvUser;
+    PPDMAUDIOHSTSTRMOUT pHstStrmOut = &pStreamOut->streamOut;
+
+    if (ASMAtomicReadU32(&pStreamOut->status) != CA_STATUS_INIT)
+    {
+        pBufData->mBuffers[0].mDataByteSize = 0;
+        return noErr;
+    }
+
+    /* How much space is used in the ring buffer? */
+    size_t cbDataAvail = RT_MIN(RTCircBufUsed(pStreamOut->pBuf), pBufData->mBuffers[0].mDataByteSize);
+    if (!cbDataAvail)
+    {
+        pBufData->mBuffers[0].mDataByteSize = 0;
+        return noErr;
+    }
+
+    uint8_t *pbSrc = NULL;
+    size_t cbRead = 0;
+    size_t cbToRead;
+    while (cbDataAvail)
+    {
+        /* Try to acquire the necessary block from the ring buffer. */
+        RTCircBufAcquireReadBlock(pStreamOut->pBuf, cbDataAvail, (void **)&pbSrc, &cbToRead);
+
+        /* Break if nothing is used anymore. */
+        if (!cbToRead)
+            break;
+
+        /* Copy the data from our ring buffer to the core audio buffer. */
+        memcpy((uint8_t *)pBufData->mBuffers[0].mData + cbRead, pbSrc, cbToRead);
+
+        /* Release the read buffer, so it could be used for new data. */
+        RTCircBufReleaseReadBlock(pStreamOut->pBuf, cbToRead);
+
+        /* Move offset. */
+        cbRead += cbToRead;
+        Assert(pBufData->mBuffers[0].mDataByteSize >= cbRead);
+
+        Assert(cbToRead <= cbDataAvail);
+        cbDataAvail -= cbToRead;
+    }
+
+    /* Write the bytes to the core audio buffer which where really written. */
+    pBufData->mBuffers[0].mDataByteSize = cbRead;
+
+    LogFlowFunc(("CoreAudio: [Output] Read %zu / %zu bytes\n", cbRead, cbDataAvail));
+
+    return noErr;
+}
+
 static DECLCALLBACK(int) drvHostCoreAudioInit(PPDMIHOSTAUDIO pInterface)
 {
@@ -1356 +1442 @@
     /* Check if the audio device should be reinitialized. If so do it. */
     if (ASMAtomicReadU32(&pStreamIn->status) == CA_STATUS_REINIT)
-        drvHostCoreAudioReinitInput(pInterface, &pStreamIn->streamIn);
+        coreAudioReinitIn(pInterface, &pStreamIn->streamIn);
 
     if (ASMAtomicReadU32(&pStreamIn->status) != CA_STATUS_INIT)
@@ -1428 +1514 @@
 }
 
-/* Callback for getting notified when some of the properties of an audio device has changed. */
-static DECLCALLBACK(OSStatus) drvHostCoreAudioPlaybackAudioDevicePropertyChanged(AudioObjectID propertyID,
-                                                                                 UInt32 nAddresses,
-                                                                                 const AudioObjectPropertyAddress properties[],
-                                                                                 void *pvUser)
-{
-    switch (propertyID)
-    {
-#ifdef DEBUG
-        case kAudioDeviceProcessorOverload:
-        {
-            Log2(("CoreAudio: [Output] Processor overload detected!\n"));
-            break;
-        }
-#endif /* DEBUG */
-        default:
-            break;
-    }
-
-    return noErr;
-}
-
-/* Callback to feed audio output buffer. */
-static DECLCALLBACK(OSStatus) drvHostCoreAudioPlaybackCallback(void                       *pvUser,
-                                                               AudioUnitRenderActionFlags *pActionFlags,
-                                                               const AudioTimeStamp       *pAudioTS,
-                                                               UInt32                      uBusID,
-                                                               UInt32                      cFrames,
-                                                               AudioBufferList            *pBufData)
-{
-    PCOREAUDIOSTREAMOUT pStreamOut = (PCOREAUDIOSTREAMOUT)pvUser;
-    PPDMAUDIOHSTSTRMOUT pHstStrmOut = &pStreamOut->streamOut;
-
-    if (ASMAtomicReadU32(&pStreamOut->status) != CA_STATUS_INIT)
-    {
-        pBufData->mBuffers[0].mDataByteSize = 0;
-        return noErr;
-    }
-
-    /* How much space is used in the ring buffer? */
-    size_t cbDataAvail = RT_MIN(RTCircBufUsed(pStreamOut->pBuf), pBufData->mBuffers[0].mDataByteSize);
-    if (!cbDataAvail)
-    {
-        pBufData->mBuffers[0].mDataByteSize = 0;
-        return noErr;
-    }
-
-    uint8_t *pbSrc = NULL;
-    size_t cbRead = 0;
-    size_t cbToRead;
-    while (cbDataAvail)
-    {
-        /* Try to acquire the necessary block from the ring buffer. */
-        RTCircBufAcquireReadBlock(pStreamOut->pBuf, cbDataAvail, (void **)&pbSrc, &cbToRead);
-
-        /* Break if nothing is used anymore. */
-        if (!cbToRead)
-            break;
-
-        /* Copy the data from our ring buffer to the core audio buffer. */
-        memcpy((uint8_t *)pBufData->mBuffers[0].mData + cbRead, pbSrc, cbToRead);
-
-        /* Release the read buffer, so it could be used for new data. */
-        RTCircBufReleaseReadBlock(pStreamOut->pBuf, cbToRead);
-
-        /* Move offset. */
-        cbRead += cbToRead;
-        Assert(pBufData->mBuffers[0].mDataByteSize >= cbRead);
-
-        Assert(cbToRead <= cbDataAvail);
-        cbDataAvail -= cbToRead;
-    }
-
-    /* Write the bytes to the core audio buffer which where really written. */
-    pBufData->mBuffers[0].mDataByteSize = cbRead;
-
-    LogFlowFunc(("CoreAudio: [Output] Read %zu / %zu bytes\n", cbRead, cbDataAvail));
-
-    return noErr;
-}
-
 static DECLCALLBACK(int) drvHostCoreAudioPlayOut(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMOUT pHstStrmOut,
                                                  uint32_t *pcSamplesPlayed)
@@ -1519 +1524 @@
     /* Check if the audio device should be reinitialized. If so do it. */
     if (ASMAtomicReadU32(&pStreamOut->status) == CA_STATUS_REINIT)
-        drvHostCoreAudioReinitOutput(pInterface, &pStreamOut->streamOut);
+        coreAudioReinitOut(pInterface, &pStreamOut->streamOut);
 
     /* Not much else to do here. */
@@ -1772 +1777 @@
 #ifdef DEBUG
         err = AudioObjectRemovePropertyListener(pStreamIn->deviceID, &propAdr,
-                                                drvHostCoreAudioRecordingAudioDevicePropertyChanged, pStreamIn);
+                                                coreAudioRecordingAudioDevicePropertyChanged, pStreamIn);
         /* Not Fatal */
         if (RT_UNLIKELY(err != noErr))
@@ -1780 +1785 @@
         propAdr.mSelector = kAudioDevicePropertyNominalSampleRate;
         err = AudioObjectRemovePropertyListener(pStreamIn->deviceID, &propAdr,
-                                                drvHostCoreAudioRecordingAudioDevicePropertyChanged, pStreamIn);
+                                                coreAudioRecordingAudioDevicePropertyChanged, pStreamIn);
         /* Not Fatal */
         if (RT_UNLIKELY(err != noErr))
@@ -1789 +1794 @@
             propAdr.mSelector = kAudioHardwarePropertyDefaultInputDevice;
             err = AudioObjectRemovePropertyListener(kAudioObjectSystemObject, &propAdr,
-                                                    drvHostCoreAudioDefaultDeviceChanged, pStreamIn);
+                                                    coreAudioDefaultDeviceChanged, pStreamIn);
             if (RT_LIKELY(err == noErr))
             {
@@ -1872 +1877 @@
 #ifdef DEBUG
         err = AudioObjectRemovePropertyListener(pStreamOut->deviceID, &propAdr,
-                                                drvHostCoreAudioPlaybackAudioDevicePropertyChanged, pStreamOut);
+                                                coreAudioPlaybackAudioDevicePropertyChanged, pStreamOut);
         /* Not Fatal */
         if (RT_UNLIKELY(err != noErr))
@@ -1880 +1885 @@
         propAdr.mSelector = kAudioDevicePropertyNominalSampleRate;
         err = AudioObjectRemovePropertyListener(pStreamOut->deviceID, &propAdr,
-                                                drvHostCoreAudioPlaybackAudioDevicePropertyChanged, pStreamOut);
+                                                coreAudioPlaybackAudioDevicePropertyChanged, pStreamOut);
         /* Not Fatal */
         if (RT_UNLIKELY(err != noErr))
@@ -1891 +1896 @@
             propAdr.mElement  = kAudioObjectPropertyElementMaster;
             err = AudioObjectRemovePropertyListener(kAudioObjectSystemObject, &propAdr,
-                                                    drvHostCoreAudioDefaultDeviceChanged, pStreamOut);
+                                                    coreAudioDefaultDeviceChanged, pStreamOut);
             if (RT_LIKELY(err == noErr))
             {
@@ -1962 +1967 @@
         }
 #endif
-        rc = drvHostCoreAudioInitInput(&pStreamIn->streamIn, pcSamples);
+        rc = coreAudioInitIn(&pStreamIn->streamIn, pcSamples);
     }
 
@@ -1973 +1978 @@
                                                    kAudioObjectPropertyElementMaster };
             OSStatus err = AudioObjectAddPropertyListener(kAudioObjectSystemObject, &propAdr,
-                                                          drvHostCoreAudioDefaultDeviceChanged, (void *)pStreamIn);
+                                                          coreAudioDefaultDeviceChanged, (void *)pStreamIn);
             /* Not fatal. */
             if (RT_LIKELY(err == noErr))
@@ -2023 +2028 @@
 #endif
 
-    rc = drvHostCoreAudioInitOutput(pHstStrmOut, pcSamples);
+    rc = coreAudioInitOut(pHstStrmOut, pcSamples);
     if (RT_FAILURE(rc))
         return rc;
@@ -2033 +2038 @@
                                                kAudioObjectPropertyElementMaster };
         err = AudioObjectAddPropertyListener(kAudioObjectSystemObject, &propAdr,
-                                             drvHostCoreAudioDefaultDeviceChanged, (void *)pStreamOut);
+                                             coreAudioDefaultDeviceChanged, (void *)pStreamOut);
         /* Not fatal. */
         if (RT_LIKELY(err == noErr))
@@ -2053 +2058 @@
 }
 
-static DECLCALLBACK(int) drvHostCoreAudioGetConf(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDCFG pAudioConf)
-{
-    pAudioConf->cbStreamOut     = sizeof(COREAUDIOSTREAMOUT);
-    pAudioConf->cbStreamIn      = sizeof(COREAUDIOSTREAMIN);
-    pAudioConf->cMaxHstStrmsOut = 1;
-    pAudioConf->cMaxHstStrmsIn  = 2;
+static DECLCALLBACK(int) drvHostCoreAudioGetConf(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDCFG pCfg)
+{
+    NOREF(pInterface);
+    AssertPtrReturn(pCfg, VERR_INVALID_POINTER);
+
+    LogFlowFuncEnter();
+
+    pCfg->cbStreamIn      = sizeof(COREAUDIOSTREAMIN);
+    pCfg->cbStreamOut     = sizeof(COREAUDIOSTREAMOUT);
+    pCfg->cMaxStreamsIn   = UINT32_MAX;
+    pCfg->cMaxStreamsOut  = UINT32_MAX;
+
+    /** @todo Implement a proper device detection. */
+    pCfg->cDevsIn         = 1;
+    pCfg->cDevsOut        = 1;
 
     return VINF_SUCCESS;

trunk/src/VBox/Devices/Audio/DrvHostDSound.cpp

@@ -91 +91 @@
 {
     PDMAUDIOHSTSTRMOUT   strmOut; /* Always must come first! */
-    LPDIRECTSOUND8       pDS;
+    LPDIRECTSOUND8       pDS;     /** @todo Move this out of this structure! Not required per-stream (e.g. for multi-channel). */
     LPDIRECTSOUNDBUFFER8 pDSB;
     DWORD                cbPlayWritePos;
@@ -160 +160 @@
 typedef struct DSOUNDENUMCBCTX
 {
+    /** Pointer to host backend driver. */
     PDRVHOSTDSOUND      pDrv;
-    PPDMAUDIOBACKENDCFG pCfg;
     /** Enumeration flags. */
     uint32_t            fFlags;
+    /** Number of found input devices. */
+    uint8_t             cDevIn;
+    /** Number of found output devices. */
+    uint8_t             cDevOut;
 } DSOUNDENUMCBCTX, *PDSOUNDENUMCBCTX;
 
@@ -173 +177 @@
 } DSOUNDDEV, *PDSOUNDDEV;
 
+/*********************************************************************************************************************************
+*   Defines                                                                                                                      *
+*********************************************************************************************************************************/
+
 /** Maximum number of attempts to restore the sound buffer before giving up. */
 #define DRV_DSOUND_RESTORE_ATTEMPTS_MAX         3
@@ -179 +187 @@
 #define PDMIHOSTAUDIO_2_DRVHOSTDSOUND(pInterface) \
     ( (PDRVHOSTDSOUND)((uintptr_t)pInterface - RT_OFFSETOF(DRVHOSTDSOUND, IHostAudio)) )
+
+/*********************************************************************************************************************************
+*   Prototypes                                                                                                                   *
+*********************************************************************************************************************************/
 
 static HRESULT directSoundPlayRestore(PDRVHOSTDSOUND pThis, LPDIRECTSOUNDBUFFER8 pDSB);
@@ -1243 +1255 @@
     AssertPtrReturn(pCtx, FALSE);
     AssertPtrReturn(pCtx->pDrv, FALSE);
-    AssertPtrReturn(pCtx->pCfg, FALSE);
 
     if (!lpGUID)
@@ -1259 +1270 @@
         return FALSE; /* Abort enumeration. */
 
-    pCtx->pCfg->cMaxHstStrmsOut++;
+    pCtx->cDevOut++;
 
     return TRUE;
@@ -1270 +1281 @@
     AssertPtrReturn(pCtx, FALSE);
     AssertPtrReturn(pCtx->pDrv, FALSE);
-    AssertPtrReturn(pCtx->pCfg, FALSE);
 
     if (!lpGUID)
@@ -1283 +1293 @@
         return FALSE; /* Abort enumeration. */
 
-    pCtx->pCfg->cMaxHstStrmsIn++;
+    pCtx->cDevIn++;
 
     return TRUE;
@@ -1293 +1303 @@
  * @return  IPRT status code.
  * @param   pThis               Host audio driver instance.
- * @param   pCfg                Where to store the enumeration results.
+ * @param   pEnmCtx             Enumeration context to use.
  * @param   fEnum               Enumeration flags.
  */
-static int dsoundDevicesEnumerate(PDRVHOSTDSOUND pThis, PPDMAUDIOBACKENDCFG pCfg, uint32_t fEnum)
-{
-    AssertPtrReturn(pThis, VERR_INVALID_POINTER);
-    AssertPtrReturn(pCfg,  VERR_INVALID_POINTER);
+static int dsoundDevicesEnumerate(PDRVHOSTDSOUND pThis, PDSOUNDENUMCBCTX pEnmCtx, uint32_t fEnum)
+{
+    AssertPtrReturn(pThis,   VERR_INVALID_POINTER);
+    AssertPtrReturn(pEnmCtx, VERR_INVALID_POINTER);
 
     dsoundDevicesClear(pThis);
-
-    pCfg->cMaxHstStrmsOut = 0;
-    pCfg->cMaxHstStrmsIn  = 0;
 
     RTLDRMOD hDSound = NULL;
@@ -1319 +1326 @@
         if (RT_SUCCESS(rc))
         {
-            DSOUNDENUMCBCTX ctx = { pThis, pCfg, fEnum };
-
-            HRESULT hr = pfnDirectSoundEnumerateW(&dsoundDevicesEnumCbPlayback, &ctx);
+            HRESULT hr = pfnDirectSoundEnumerateW(&dsoundDevicesEnumCbPlayback, pEnmCtx);
             if (FAILED(hr))
                 LogRel2(("DSound: Error enumerating host playback devices: %Rhrc\n", hr));
 
-            hr = pfnDirectSoundCaptureEnumerateW(&dsoundDevicesEnumCbCapture, &ctx);
+            hr = pfnDirectSoundCaptureEnumerateW(&dsoundDevicesEnumCbCapture, pEnmCtx);
             if (FAILED(hr))
                 LogRel2(("DSound: Error enumerating host capturing devices: %Rhrc\n", hr));
+
+            if (fEnum & DSOUNDENUMCBFLAGS_LOG)
+            {
+                LogRel2(("DSound: Found %RU8 host playback devices\n",  pEnmCtx->cDevOut));
+                LogRel2(("DSound: Found %RU8 host capturing devices\n", pEnmCtx->cDevIn));
+            }
         }
 
@@ -1357 +1368 @@
     RT_ZERO(Cfg);
 
-    Cfg.cbStreamOut = sizeof(DSOUNDSTREAMOUT);
-    Cfg.cbStreamIn  = sizeof(DSOUNDSTREAMIN);
-
-    int rc = dsoundDevicesEnumerate(pThis, &Cfg, fEnum);
-    AssertRC(rc);
-
+    Cfg.cbStreamOut     = sizeof(DSOUNDSTREAMOUT);
+    Cfg.cbStreamIn      = sizeof(DSOUNDSTREAMIN);
+
+    DSOUNDENUMCBCTX cbCtx = { pThis, fEnum, 0, 0 };
+
+    int rc = dsoundDevicesEnumerate(pThis, &cbCtx, fEnum);
+    if (RT_SUCCESS(rc))
+    {
 #ifdef VBOX_WITH_AUDIO_CALLBACKS
-    if (   pThis->fEnabledOut != RT_BOOL(Cfg.cMaxHstStrmsOut)
-        || pThis->fEnabledIn  != RT_BOOL(Cfg.cMaxHstStrmsIn))
-    {
-        /** @todo Use a registered callback to the audio connector (e.g "OnConfigurationChanged") to
-         *        let the connector know that something has changed within the host backend. */
-    }
+        if (   pThis->fEnabledOut != RT_BOOL(cbCtx.cDevOut)
+            || pThis->fEnabledIn  != RT_BOOL(cbCtx.cDevIn))
+        {
+            /** @todo Use a registered callback to the audio connector (e.g "OnConfigurationChanged") to
+             *        let the connector know that something has changed within the host backend. */
+        }
 #else
-    pThis->fEnabledOut = RT_BOOL(Cfg.cMaxHstStrmsOut);
-    pThis->fEnabledIn  = RT_BOOL(Cfg.cMaxHstStrmsIn);
+        pThis->fEnabledOut = RT_BOOL(cbCtx.cDevOut);
+        pThis->fEnabledIn  = RT_BOOL(cbCtx.cDevIn);
 #endif
 
-    if (pCfg)
-        memcpy(pCfg, &Cfg, sizeof(PDMAUDIOBACKENDCFG));
+        Cfg.cSources       = cbCtx.cDevIn;
+        Cfg.cSinks         = cbCtx.cDevOut;
+        Cfg.cMaxStreamsIn  = UINT32_MAX;
+        Cfg.cMaxStreamsOut = UINT32_MAX;
+
+        if (pCfg)
+            memcpy(pCfg, &Cfg, sizeof(PDMAUDIOBACKENDCFG));
+    }
 
     LogFlowFuncLeaveRC(rc);
@@ -2096 +2115 @@
 #endif
 
-        PDMAUDIOBACKENDCFG Cfg;
-        dsoundUpdateStatusInternalEx(pThis, &Cfg, DSOUNDENUMCBFLAGS_LOG /* fEnum */);
-
-        DSLOGREL(("DSound: Found %RU32 host playback devices\n",  Cfg.cMaxHstStrmsOut));
-        DSLOGREL(("DSound: Found %RU32 host capturing devices\n", Cfg.cMaxHstStrmsIn));
+        dsoundUpdateStatusInternalEx(pThis, NULL /* pCfg */, DSOUNDENUMCBFLAGS_LOG /* fEnum */);
     }
     else

trunk/src/VBox/Devices/Audio/DrvHostNullAudio.cpp

@@ -6 +6 @@
 
 /*
- * Copyright (C) 2006-2015 Oracle Corporation
+ * Copyright (C) 2006-2016 Oracle Corporation
  *
  * This file is part of VirtualBox Open Source Edition (OSE), as
@@ -87 +87 @@
     AssertPtrReturn(pCfg, VERR_INVALID_POINTER);
 
-    pCfg->cbStreamOut = sizeof(NULLAUDIOSTREAMOUT);
-    pCfg->cbStreamIn  = sizeof(NULLAUDIOSTREAMIN);
-
-    pCfg->cMaxHstStrmsOut = 1; /* Output */
-    pCfg->cMaxHstStrmsIn  = 2; /* Line input + microphone input. */
+    pCfg->cbStreamOut    = sizeof(NULLAUDIOSTREAMOUT);
+    pCfg->cbStreamIn     = sizeof(NULLAUDIOSTREAMIN);
+
+    /* The NULL backend has exactly one input source and one output sink. */
+    pCfg->cSources       = 1;
+    pCfg->cSinks         = 1;
+
+    pCfg->cMaxStreamsOut = 1; /* Output */
+    pCfg->cMaxStreamsIn  = 2; /* Line input + microphone input. */
 
     return VINF_SUCCESS;

trunk/src/VBox/Devices/Audio/DrvHostOSSAudio.cpp

@@ -5 +5 @@
 
 /*
- * Copyright (C) 2014-2015 Oracle Corporation
+ * Copyright (C) 2014-2016 Oracle Corporation
  *
  * This file is part of VirtualBox Open Source Edition (OSE), as
@@ -34 +34 @@
 #include <VBox/vmm/pdmaudioifs.h>
 
+/*********************************************************************************************************************************
+*   Defines                                                                                                                      *
+*********************************************************************************************************************************/
+
+#if ((SOUND_VERSION > 360) && (defined(OSS_SYSINFO)))
+/* OSS > 3.6 has a new syscall available for querying a bit more detailed information
+ * about OSS' audio capabilities. */
+# define VBOX_WITH_OSS_SYSINFO 1
+#endif
+
+/** Makes DRVHOSTOSSAUDIO out of PDMIHOSTAUDIO. */
+#define PDMIHOSTAUDIO_2_DRVHOSTOSSAUDIO(pInterface) \
+    ( (PDRVHOSTOSSAUDIO)((uintptr_t)pInterface - RT_OFFSETOF(DRVHOSTOSSAUDIO, IHostAudio)) )
+
+/*********************************************************************************************************************************
+*   Structures                                                                                                                   *
+*********************************************************************************************************************************/
+
 /**
  * OSS host audio driver instance data.
@@ -67 +85 @@
     int                cbFragmentSize;
     /** Own PCM buffer. */
-    void              *pvPCMBuf;
+    void              *pvBuf;
     /** Size (in bytes) of own PCM buffer. */
-    size_t             cbPCMBuf;
+    size_t             cbBuf;
     int                old_optr;
 } OSSAUDIOSTREAMIN, *POSSAUDIOSTREAMIN;
@@ -86 +104 @@
 #endif
     /** Own PCM buffer in case memory mapping is unavailable. */
-    void               *pvPCMBuf;
+    void               *pvBuf;
     /** Size (in bytes) of own PCM buffer. */
-    size_t              cbPCMBuf;
+    size_t              cbBuf;
     int                 old_optr;
 } OSSAUDIOSTREAMOUT, *POSSAUDIOSTREAMOUT;
@@ -133 +151 @@
 }
 
-static int drvHostOSSAudioFmtToOSS(PDMAUDIOFMT fmt)
+static int ossAudioFmtToOSS(PDMAUDIOFMT fmt)
 {
     switch (fmt)
@@ -157 +175 @@
 }
 
-static int drvHostOSSAudioOSSToFmt(int fmt,
-                                   PDMAUDIOFMT *pFmt, PDMAUDIOENDIANNESS *pENDIANNESS)
+static int ossOSSToAudioFmt(int fmt, PDMAUDIOFMT *pFmt, PDMAUDIOENDIANNESS *pENDIANNESS)
 {
     switch (fmt)
@@ -206 +223 @@
 }
 
-static int drvHostOSSAudioClose(int *phFile)
+static int ossStreamClose(int *phFile)
 {
     if (!phFile || !*phFile)
@@ -214 +231 @@
     if (close(*phFile))
     {
-        LogRel(("OSS: Closing descriptor failed: %s\n",
-                strerror(errno)));
+        LogRel(("OSS: Closing stream failed: %s\n", strerror(errno)));
         rc = VERR_GENERAL_FAILURE; /** @todo */
     }
@@ -227 +243 @@
 }
 
-static int drvHostOSSAudioOpen(bool fIn,
-                               POSSAUDIOSTREAMCFG pReq, POSSAUDIOSTREAMCFG pObt,
-                               int *phFile)
-{
-    AssertPtrReturn(pReq, VERR_INVALID_POINTER);
-    AssertPtrReturn(pObt, VERR_INVALID_POINTER);
+static int ossStreamOpen(const char *pszDev, int fOpen, POSSAUDIOSTREAMCFG pReq, POSSAUDIOSTREAMCFG pObt, int *phFile)
+{
+    AssertPtrReturn(pszDev, VERR_INVALID_POINTER);
+    AssertPtrReturn(pReq,   VERR_INVALID_POINTER);
+    AssertPtrReturn(pObt,   VERR_INVALID_POINTER);
     AssertPtrReturn(phFile, VERR_INVALID_POINTER);
 
     int rc;
-    int hFile;
-
+
+    int hFile = -1;
     do
     {
-        const char *pszDev = fIn ? s_OSSConf.devpath_in : s_OSSConf.devpath_out;
-        if (!pszDev)
-        {
-            LogRel(("OSS: Invalid or no %s device name set\n",
-                    fIn ? "input" : "output"));
-            rc = VERR_INVALID_PARAMETER;
-            break;
-        }
-
-        hFile = open(pszDev, (fIn ? O_RDONLY : O_WRONLY) | O_NONBLOCK);
+        hFile = open(pszDev, fOpen);
         if (hFile == -1)
         {
-            LogRel(("OSS: Failed to open %s: %s(%d)\n", pszDev, strerror(errno), errno));
+            LogRel(("OSS: Failed to open %s: %s (%d)\n", pszDev, strerror(errno), errno));
             rc = RTErrConvertFromErrno(errno);
             break;
         }
 
-        int iFormat = drvHostOSSAudioFmtToOSS(pReq->enmFormat);
+        int iFormat = ossAudioFmtToOSS(pReq->enmFormat);
         if (ioctl(hFile, SNDCTL_DSP_SAMPLESIZE, &iFormat))
         {
-            LogRel(("OSS: Failed to set audio format to %ld errno=%s(%d)\n", iFormat, strerror(errno), errno));
+            LogRel(("OSS: Failed to set audio format to %ld: %s (%d)\n", iFormat, strerror(errno), errno));
             rc = RTErrConvertFromErrno(errno);
             break;
@@ -268 +274 @@
         if (ioctl(hFile, SNDCTL_DSP_CHANNELS, &cChannels))
         {
-            LogRel(("OSS: Failed to set number of audio channels (%d): %s(%d)\n", pReq->cChannels, strerror(errno), errno));
+            LogRel(("OSS: Failed to set number of audio channels (%d): %s (%d)\n", pReq->cChannels, strerror(errno), errno));
             rc = RTErrConvertFromErrno(errno);
             break;
@@ -276 +282 @@
         if (ioctl(hFile, SNDCTL_DSP_SPEED, &freq))
         {
-            LogRel(("OSS: Failed to set audio frequency (%dHZ): %s(%d)\n", pReq->uFreq, strerror(errno), errno));
+            LogRel(("OSS: Failed to set audio frequency (%dHZ): %s (%d)\n", pReq->uFreq, strerror(errno), errno));
             rc = RTErrConvertFromErrno(errno);
             break;
@@ -285 +291 @@
         if (ioctl(hFile, SNDCTL_DSP_NONBLOCK))
         {
-            LogRel(("OSS: Failed to set non-blocking mode: %s(%d)\n", strerror(errno), errno));
+            LogRel(("OSS: Failed to set non-blocking mode: %s (%d)\n", strerror(errno), errno));
             rc = RTErrConvertFromErrno(errno);
             break;
@@ -293 +299 @@
         if (ioctl(hFile, SNDCTL_DSP_SETFRAGMENT, &mmmmssss))
         {
-            LogRel(("OSS: Failed to set %RU16 fragments to %RU32 bytes each: %s(%d)\n",
+            LogRel(("OSS: Failed to set %RU16 fragments to %RU32 bytes each: %s (%d)\n",
                     pReq->cFragments, pReq->cbFragmentSize, strerror(errno), errno));
             rc = RTErrConvertFromErrno(errno);
@@ -300 +306 @@
 
         audio_buf_info abinfo;
-        if (ioctl(hFile, fIn ? SNDCTL_DSP_GETISPACE : SNDCTL_DSP_GETOSPACE, &abinfo))
-        {
-            LogRel(("OSS: Failed to retrieve buffer length: %s(%d)\n", strerror(errno), errno));
+        if (ioctl(hFile, (fOpen & O_RDONLY) ? SNDCTL_DSP_GETISPACE : SNDCTL_DSP_GETOSPACE, &abinfo))
+        {
+            LogRel(("OSS: Failed to retrieve buffer length: %s (%d)\n", strerror(errno), errno));
             rc = RTErrConvertFromErrno(errno);
             break;
         }
 
-        rc = drvHostOSSAudioOSSToFmt(iFormat, &pObt->enmFormat, &pObt->enmENDIANNESS);
+        rc = ossOSSToAudioFmt(iFormat, &pObt->enmFormat, &pObt->enmENDIANNESS);
         if (RT_SUCCESS(rc))
         {
@@ -321 +327 @@
 
     if (RT_FAILURE(rc))
-        drvHostOSSAudioClose(&hFile);
+        ossStreamClose(&hFile);
 
     LogFlowFuncLeaveRC(rc);
@@ -362 +368 @@
         {
             DrvAudioClearBuf(&pHstStrmOut->Props,
-                             pThisStrmOut->pvPCMBuf, pThisStrmOut->cbPCMBuf, AudioMixBufSize(&pHstStrmOut->MixBuf));
+                             pThisStrmOut->pvBuf, pThisStrmOut->cbBuf, AudioMixBufSize(&pHstStrmOut->MixBuf));
 
             mask = PCM_ENABLE_OUTPUT;
@@ -412 +418 @@
     AssertPtrReturn(pHstStrmIn, VERR_INVALID_POINTER);
 
-    POSSAUDIOSTREAMIN pThisStrmIn = (POSSAUDIOSTREAMIN)pHstStrmIn;
+    POSSAUDIOSTREAMIN pStrm = (POSSAUDIOSTREAMIN)pHstStrmIn;
 
     int rc = VINF_SUCCESS;
-    size_t cbToRead = RT_MIN(pThisStrmIn->cbPCMBuf,
+    size_t cbToRead = RT_MIN(pStrm->cbBuf,
                              AudioMixBufFreeBytes(&pHstStrmIn->MixBuf));
 
@@ -427 +433 @@
     while (cbToRead)
     {
-        cbTemp = RT_MIN(cbToRead, pThisStrmIn->cbPCMBuf);
+        cbTemp = RT_MIN(cbToRead, pStrm->cbBuf);
         AssertBreakStmt(cbTemp, rc = VERR_NO_DATA);
-        cbRead = read(pThisStrmIn->hFile, (uint8_t *)pThisStrmIn->pvPCMBuf + offWrite, cbTemp);
-
-        LogFlowFunc(("cbRead=%zi, cbTemp=%RU32, cbToRead=%zu\n",
-                     cbRead, cbTemp, cbToRead));
+        cbRead = read(pStrm->hFile, (uint8_t *)pStrm->pvBuf + offWrite, cbTemp);
+
+        LogFlowFunc(("cbRead=%zi, cbTemp=%RU32, cbToRead=%zu\n", cbRead, cbTemp, cbToRead));
 
         if (cbRead < 0)
@@ -451 +456 @@
 
                 default:
-                    LogFlowFunc(("Failed to read %zu input frames, rc=%Rrc\n",
-                                 cbTemp, rc));
-                    rc = VERR_GENERAL_FAILURE; /** @todo */
+                    LogFlowFunc(("Failed to read %zu input frames, rc=%Rrc\n", cbTemp, rc));
+                    rc = VERR_GENERAL_FAILURE; /** @todo Fix this. */
                     break;
             }
@@ -463 +467 @@
         {
             uint32_t cWritten;
-            rc = AudioMixBufWriteCirc(&pHstStrmIn->MixBuf,
-                                      pThisStrmIn->pvPCMBuf, cbRead,
-                                      &cWritten);
+            rc = AudioMixBufWriteCirc(&pHstStrmIn->MixBuf, pStrm->pvBuf, cbRead, &cWritten);
             if (RT_FAILURE(rc))
                 break;
@@ -487 +489 @@
         uint32_t cProcessed = 0;
         if (cWrittenTotal)
-            rc = AudioMixBufMixToParent(&pHstStrmIn->MixBuf, cWrittenTotal,
-                                        &cProcessed);
+            rc = AudioMixBufMixToParent(&pHstStrmIn->MixBuf, cWrittenTotal, &cProcessed);
 
         if (pcSamplesCaptured)
@@ -506 +507 @@
     AssertPtrReturn(pHstStrmIn, VERR_INVALID_POINTER);
 
-    POSSAUDIOSTREAMIN pThisStrmIn = (POSSAUDIOSTREAMIN)pHstStrmIn;
+    POSSAUDIOSTREAMIN pStrm = (POSSAUDIOSTREAMIN)pHstStrmIn;
 
     LogFlowFuncEnter();
 
-    if (pThisStrmIn->pvPCMBuf)
-    {
-        Assert(pThisStrmIn->cbPCMBuf);
-
-        RTMemFree(pThisStrmIn->pvPCMBuf);
-        pThisStrmIn->pvPCMBuf = NULL;
-    }
-
-    pThisStrmIn->cbPCMBuf = 0;
+    if (pStrm->pvBuf)
+    {
+        Assert(pStrm->cbBuf);
+
+        RTMemFree(pStrm->pvBuf);
+        pStrm->pvBuf = NULL;
+    }
+
+    pStrm->cbBuf = 0;
+
+    ossStreamClose(&pStrm->hFile);
 
     return VINF_SUCCESS;
@@ -528 +531 @@
     AssertPtrReturn(pHstStrmOut, VERR_INVALID_POINTER);
 
-    POSSAUDIOSTREAMOUT pThisStrmOut = (POSSAUDIOSTREAMOUT)pHstStrmOut;
+    POSSAUDIOSTREAMOUT pStrm = (POSSAUDIOSTREAMOUT)pHstStrmOut;
 
     LogFlowFuncEnter();
 
 #ifndef RT_OS_L4
-    if (!pThisStrmOut->fMemMapped)
-    {
-        if (pThisStrmOut->pvPCMBuf)
-        {
-            Assert(pThisStrmOut->cbPCMBuf);
-
-            RTMemFree(pThisStrmOut->pvPCMBuf);
-            pThisStrmOut->pvPCMBuf = NULL;
-        }
-
-        pThisStrmOut->cbPCMBuf = 0;
-    }
-#endif
+    if (pStrm->fMemMapped)
+    {
+        if (pStrm->pvBuf)
+        {
+            Assert(pStrm->cbBuf);
+
+            int rc2 = munmap(pStrm->pvBuf, pStrm->cbBuf);
+            if (rc2 == 0)
+            {
+                pStrm->pvBuf      = NULL;
+                pStrm->cbBuf      = 0;
+
+                pStrm->fMemMapped = false;
+            }
+            else
+                LogRel(("OSS: Failed to memory unmap playback buffer on close: %s\n", strerror(errno)));
+        }
+    }
+    else
+    {
+#endif
+        if (pStrm->pvBuf)
+        {
+            Assert(pStrm->cbBuf);
+
+            RTMemFree(pStrm->pvBuf);
+            pStrm->pvBuf = NULL;
+        }
+
+        pStrm->cbBuf = 0;
+#ifndef RT_OS_L4
+    }
+#endif
+
+    ossStreamClose(&pStrm->hFile);
 
     return VINF_SUCCESS;
@@ -554 +579 @@
     NOREF(pInterface);
 
+    pCfg->cbStreamIn  = sizeof(OSSAUDIOSTREAMIN);
     pCfg->cbStreamOut = sizeof(OSSAUDIOSTREAMOUT);
-    pCfg->cbStreamIn = sizeof(OSSAUDIOSTREAMIN);
-    pCfg->cMaxHstStrmsOut = INT_MAX;
-    pCfg->cMaxHstStrmsIn = INT_MAX;
+
+    pCfg->cSources    = 0;
+    pCfg->cSinks      = 0;
+
+    int hFile = open("/dev/dsp", O_WRONLY | O_NONBLOCK, 0);
+    if (hFile == -1)
+    {
+        /* Try opening the mixing device instead. */
+        hFile = open("/dev/mixer", O_RDONLY | O_NONBLOCK, 0);
+    }
+
+    int ossVer = -1;
+
+#ifdef VBOX_WITH_OSS_SYSINFO
+    oss_sysinfo ossInfo;
+    RT_ZERO(ossInfo);
+#endif
+
+    if (hFile != -1)
+    {
+        int err = ioctl(hFile, OSS_GETVERSION, &ossVer);
+        if (err == 0)
+        {
+            LogRel2(("OSS: Using version: %d\n", ossVer));
+#ifdef VBOX_WITH_OSS_SYSINFO
+            err = ioctl(hFile, OSS_SYSINFO, &ossInfo);
+            if (err == 0)
+            {
+                LogRel2(("OSS: Number of DSPs: %d\n", ossVer.numaudios));
+                LogRel2(("OSS: Number of mixers: %d\n", ossVer.nummixers));
+                /** @todo Determine number of input devices + output devices. */
+            }
+            else
+            {
+#endif
+                /* Since we cannot query anything, assume that we have at least
+                 * one input and one output if we found "/dev/dsp" or "/dev/mixer". */
+                pCfg->cSources        = 1;
+                pCfg->cSinks          = 1;
+
+                pCfg->cMaxStreamsIn   = UINT32_MAX; /* Line in + microphone in. */
+                pCfg->cMaxStreamsOut  = UINT32_MAX;
+#ifdef VBOX_WITH_OSS_SYSINFO
+            }
+#endif
+        }
+        else
+            LogRel(("OSS: Unable to determine installed version: %s (%d)\n", strerror(err), err));
+    }
+    else
+        LogRel(("OSS: No devices found, audio is not available\n"));
 
     return VINF_SUCCESS;
@@ -567 +641 @@
                                                uint32_t *pcSamples)
 {
-    NOREF(pInterface);
+    AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
     AssertPtrReturn(pHstStrmIn, VERR_INVALID_POINTER);
-    AssertPtrReturn(pCfg, VERR_INVALID_POINTER);
-
-    POSSAUDIOSTREAMIN pThisStrmIn = (POSSAUDIOSTREAMIN)pHstStrmIn;
+    AssertPtrReturn(pCfg,       VERR_INVALID_POINTER);
+
+    PDRVHOSTOSSAUDIO  pThis = PDMIHOSTAUDIO_2_DRVHOSTOSSAUDIO(pInterface);
+    POSSAUDIOSTREAMIN pStrm = (POSSAUDIOSTREAMIN)pHstStrmIn;
 
     int rc;
@@ -587 +662 @@
         reqStream.cbFragmentSize = s_OSSConf.fragsize;
 
-        rc = drvHostOSSAudioOpen(true /* fIn */,
-                                 &reqStream, &obtStream, &hFile);
+        rc = ossStreamOpen(s_OSSConf.devpath_in, O_RDONLY | O_NONBLOCK, &reqStream, &obtStream, &hFile);
         if (RT_SUCCESS(rc))
         {
             if (obtStream.cFragments * obtStream.cbFragmentSize & pHstStrmIn->Props.uAlign)
-                LogRel(("OSS: Warning: Misaligned DAC output buffer: Size = %zu, Alignment = %u\n",
+                LogRel(("OSS: Warning: Misaligned capturing buffer: Size = %zu, Alignment = %u\n",
                         obtStream.cFragments * obtStream.cbFragmentSize,
                         pHstStrmIn->Props.uAlign + 1));
-
-            pThisStrmIn->hFile = hFile;
 
             PDMAUDIOSTREAMCFG streamCfg;
@@ -617 +689 @@
         {
             size_t cbSample = (1 << pHstStrmIn->Props.cShift);
-            size_t cbBuf    = cSamples * cbSample;
-            pThisStrmIn->pvPCMBuf = RTMemAlloc(cbBuf);
-            if (!pThisStrmIn->pvPCMBuf)
-            {
-                LogRel(("OSS: Failed allocating ADC buffer with %RU32 samples (%zu bytes per sample)\n", cSamples, cbSample));
+
+            size_t cbBuf = cSamples * cbSample;
+            void  *pvBuf = RTMemAlloc(cbBuf);
+            if (!pvBuf)
+            {
+                LogRel(("OSS: Failed allocating capturing buffer with %RU32 samples (%zu bytes per sample)\n",
+                        cSamples, cbSample));
                 rc = VERR_NO_MEMORY;
-            }
-
-            pThisStrmIn->cbPCMBuf = cbBuf;
+                break;
+            }
+
+            pStrm->hFile = hFile;
+            pStrm->pvBuf = pvBuf;
+            pStrm->cbBuf = cbBuf;
 
             if (pcSamples)
@@ -634 +711 @@
 
     if (RT_FAILURE(rc))
-        drvHostOSSAudioClose(&hFile);
+        ossStreamClose(&hFile);
 
     LogFlowFuncLeaveRC(rc);
@@ -644 +721 @@
                                                 uint32_t *pcSamples)
 {
-    NOREF(pInterface);
+    AssertPtrReturn(pInterface,  VERR_INVALID_POINTER);
     AssertPtrReturn(pHstStrmOut, VERR_INVALID_POINTER);
-    AssertPtrReturn(pCfg, VERR_INVALID_POINTER);
-
-    POSSAUDIOSTREAMOUT pThisStrmOut = (POSSAUDIOSTREAMOUT)pHstStrmOut;
+    AssertPtrReturn(pCfg,        VERR_INVALID_POINTER);
+
+    PDRVHOSTOSSAUDIO   pThis = PDMIHOSTAUDIO_2_DRVHOSTOSSAUDIO(pInterface);
+    POSSAUDIOSTREAMOUT pStrm = (POSSAUDIOSTREAMOUT)pHstStrmOut;
 
     int rc;
@@ -664 +742 @@
         reqStream.cbFragmentSize = s_OSSConf.fragsize;
 
-        rc = drvHostOSSAudioOpen(false /* fIn */,
-                                 &reqStream, &obtStream, &hFile);
+        rc = ossStreamOpen(s_OSSConf.devpath_out, O_WRONLY | O_NONBLOCK, &reqStream, &obtStream, &hFile);
         if (RT_SUCCESS(rc))
         {
             if (obtStream.cFragments * obtStream.cbFragmentSize & pHstStrmOut->Props.uAlign)
-                LogRel(("OSS: Warning: Misaligned DAC output buffer: Size = %zu, Alignment = %u\n",
+                LogRel(("OSS: Warning: Misaligned playback buffer: Size = %zu, Alignment = %u\n",
                         obtStream.cFragments * obtStream.cbFragmentSize,
                         pHstStrmOut->Props.uAlign + 1));
-
-            pThisStrmOut->hFile = hFile;
 
             PDMAUDIOSTREAMCFG streamCfg;
@@ -689 +764 @@
         if (RT_SUCCESS(rc))
         {
+            pStrm->fMemMapped = false;
+
+            size_t cbSamples =  cSamples << pHstStrmOut->Props.cShift;
+            Assert(cbSamples);
+
 #ifndef RT_OS_L4
-            pThisStrmOut->fMemMapped = false;
             if (s_OSSConf.try_mmap)
             {
-                pThisStrmOut->pvPCMBuf = mmap(0, cSamples << pHstStrmOut->Props.cShift,
-                                              PROT_READ | PROT_WRITE, MAP_SHARED, hFile, 0);
-                if (pThisStrmOut->pvPCMBuf == MAP_FAILED)
+                pStrm->pvBuf = mmap(0, cbSamples, PROT_READ | PROT_WRITE, MAP_SHARED, hFile, 0);
+                if (pStrm->pvBuf == MAP_FAILED)
                 {
-                    LogRel(("OSS: Failed to memory map %zu bytes of DAC output file: %s\n",
-                            cSamples << pHstStrmOut->Props.cShift, strerror(errno)));
+                    LogRel(("OSS: Failed to memory map %zu bytes of playback buffer: %s\n", cbSamples, strerror(errno)));
                     rc = RTErrConvertFromErrno(errno);
                     break;
@@ -707 +784 @@
                     if (ioctl(hFile, SNDCTL_DSP_SETTRIGGER, &mask) < 0)
                     {
-                        LogRel(("OSS: Failed to retrieve initial trigger mask: %s\n",
-                                strerror(errno)));
+                        LogRel(("OSS: Failed to retrieve initial trigger mask for playback buffer: %s\n", strerror(errno)));
                         rc = RTErrConvertFromErrno(errno);
                         /* Note: No break here, need to unmap file first! */
@@ -717 +793 @@
                         if (ioctl (hFile, SNDCTL_DSP_SETTRIGGER, &mask) < 0)
                         {
-                            LogRel(("OSS: Failed to retrieve PCM_ENABLE_OUTPUT mask: %s\n",
-                                    strerror(errno)));
+                            LogRel(("OSS: Failed to retrieve PCM_ENABLE_OUTPUT mask: %s\n", strerror(errno)));
                             rc = RTErrConvertFromErrno(errno);
                             /* Note: No break here, need to unmap file first! */
                         }
                         else
-                            pThisStrmOut->fMemMapped = true;
+                            pStrm->fMemMapped = true;
                     }
 
-                    if (!pThisStrmOut->fMemMapped)
+                    if (RT_FAILURE(rc))
                     {
-                        int rc2 = munmap(pThisStrmOut->pvPCMBuf,
-                                         cSamples << pHstStrmOut->Props.cShift);
+                        int rc2 = munmap(pStrm->pvBuf, cbSamples);
                         if (rc2)
-                            LogRel(("OSS: Failed to unmap DAC output file: %s\n", strerror(errno)));
+                            LogRel(("OSS: Failed to memory unmap playback buffer: %s\n", strerror(errno)));
                         break;
                     }
@@ -740 +814 @@
             /* Memory mapping failed above? Try allocating an own buffer. */
 #ifndef RT_OS_L4
-            if (!pThisStrmOut->fMemMapped)
-            {
-#endif
-                size_t cbSample = (1 << pHstStrmOut->Props.cShift);
-                size_t cbPCMBuf = cSamples * cbSample;
-
-                LogFlowFunc(("cSamples=%RU32\n", cSamples));
-
-                pThisStrmOut->pvPCMBuf = RTMemAlloc(cbPCMBuf);
-                if (!pThisStrmOut->pvPCMBuf)
+            if (!pStrm->fMemMapped)
+            {
+#endif
+                void *pvBuf = RTMemAlloc(cbSamples);
+                if (!pvBuf)
                 {
-                    LogRel(("OSS: Failed allocating DAC buffer with %RU32 samples (%zu bytes per sample)\n", cSamples, cbSample));
+                    LogRel(("OSS: Failed allocating playback buffer with %RU32 samples (%zu bytes)\n", cSamples, cbSamples));
                     rc = VERR_NO_MEMORY;
                     break;
                 }
 
-                pThisStrmOut->cbPCMBuf = cbPCMBuf;
+                pStrm->hFile = hFile;
+                pStrm->pvBuf = pvBuf;
+                pStrm->cbBuf = cbSamples;
 #ifndef RT_OS_L4
             }
@@ -767 +838 @@
 
     if (RT_FAILURE(rc))
-        drvHostOSSAudioClose(&hFile);
+        ossStreamClose(&hFile);
 
     LogFlowFuncLeaveRC(rc);
@@ -786 +857 @@
     AssertPtrReturn(pHstStrmOut, VERR_INVALID_POINTER);
 
-    POSSAUDIOSTREAMOUT pThisStrmOut = (POSSAUDIOSTREAMOUT)pHstStrmOut;
+    POSSAUDIOSTREAMOUT pStrm = (POSSAUDIOSTREAMOUT)pHstStrmOut;
 
     int rc = VINF_SUCCESS;
@@ -800 +871 @@
 
 #ifndef RT_OS_L4
-        if (pThisStrmOut->fMemMapped)
+        if (pStrm->fMemMapped)
         {
             /* Get current playback pointer. */
-            int rc2 = ioctl(pThisStrmOut->hFile, SNDCTL_DSP_GETOPTR, &cntinfo);
+            int rc2 = ioctl(pStrm->hFile, SNDCTL_DSP_GETOPTR, &cntinfo);
             if (!rc2)
             {
@@ -813 +884 @@
 
             /* Nothing to play? */
-            if (cntinfo.ptr == pThisStrmOut->old_optr)
+            if (cntinfo.ptr == pStrm->old_optr)
                 break;
 
             int cbData;
-            if (cntinfo.ptr > pThisStrmOut->old_optr)
-                cbData = cntinfo.ptr - pThisStrmOut->old_optr;
+            if (cntinfo.ptr > pStrm->old_optr)
+                cbData = cntinfo.ptr - pStrm->old_optr;
             else
-                cbData = cbBuf + cntinfo.ptr - pThisStrmOut->old_optr;
+                cbData = cbBuf + cntinfo.ptr - pStrm->old_optr;
             Assert(cbData);
 
@@ -830 +901 @@
 #endif
             audio_buf_info abinfo;
-            int rc2 = ioctl(pThisStrmOut->hFile, SNDCTL_DSP_GETOSPACE, &abinfo);
+            int rc2 = ioctl(pStrm->hFile, SNDCTL_DSP_GETOSPACE, &abinfo);
             if (rc2 < 0)
             {
-                LogRel(("OSS: Failed to retrieve current playback buffer: %s\n",
-                        strerror(errno)));
+                LogRel(("OSS: Failed to retrieve current playback buffer: %s\n", strerror(errno)));
                 rc = RTErrConvertFromErrno(errno);
                 break;
@@ -841 +911 @@
             if ((size_t)abinfo.bytes > cbBuf)
             {
-                LogFlowFunc(("Warning: Invalid available size, size=%d, bufsize=%d\n",
-                             abinfo.bytes, cbBuf));
+                LogFlowFunc(("Warning: Invalid available size, size=%d, bufsize=%zu\n", abinfo.bytes, cbBuf));
                 abinfo.bytes = cbBuf;
                 /* Keep going. */
@@ -849 +918 @@
             if (abinfo.bytes < 0)
             {
-                LogFlowFunc(("Warning: Invalid available size, size=%d, bufsize=%d\n",
-                             abinfo.bytes, cbBuf));
+                LogFlowFunc(("Warning: Invalid available size, size=%d, bufsize=%zu\n", abinfo.bytes, cbBuf));
                 rc = VERR_INVALID_PARAMETER;
                 break;
             }
 
-            cToRead = RT_MIN((uint32_t)AUDIOMIXBUF_B2S(&pHstStrmOut->MixBuf, abinfo.bytes),
-                             cLive);
+            cToRead = RT_MIN((uint32_t)AUDIOMIXBUF_B2S(&pHstStrmOut->MixBuf, abinfo.bytes), cLive);
             if (!cToRead)
                 break;
@@ -868 +935 @@
         while (cbToRead)
         {
-            rc = AudioMixBufReadCirc(&pHstStrmOut->MixBuf,
-                                     pThisStrmOut->pvPCMBuf, cbToRead, &cRead);
+            rc = AudioMixBufReadCirc(&pHstStrmOut->MixBuf, pStrm->pvBuf, cbToRead, &cRead);
             if (RT_FAILURE(rc))
                 break;
 
             cbRead = AUDIOMIXBUF_S2B(&pHstStrmOut->MixBuf, cRead);
-            ssize_t cbWritten = write(pThisStrmOut->hFile, pThisStrmOut->pvPCMBuf,
-                                      cbRead);
+            ssize_t cbWritten = write(pStrm->hFile, pStrm->pvBuf, cbRead);
             if (cbWritten == -1)
             {
@@ -884 +949 @@
 
             Assert(cbToRead >= cbRead);
-            cbToRead -= cbRead;
+            cbToRead    -= cbRead;
             cbReadTotal += cbRead;
         }
@@ -890 +955 @@
 #ifndef RT_OS_L4
         /* Update read pointer. */
-        if (pThisStrmOut->fMemMapped)
-            pThisStrmOut->old_optr = cntinfo.ptr;
+        if (pStrm->fMemMapped)
+            pStrm->old_optr = cntinfo.ptr;
 #endif
 
@@ -905 +970 @@
             *pcSamplesPlayed = cReadTotal;
 
-        LogFlowFunc(("cReadTotal=%RU32 (%RU32 bytes), rc=%Rrc\n",
-                     cReadTotal, cbReadTotal, rc));
+        LogFlowFunc(("cReadTotal=%RU32 (%RU32 bytes), rc=%Rrc\n", cReadTotal, cbReadTotal, rc));
     }
 

trunk/src/VBox/Devices/Audio/DrvHostPulseAudio.cpp

@@ -16 +16 @@
  */
 
-
 /*********************************************************************************************************************************
 *   Header Files                                                                                                                 *
@@ -41 +40 @@
 #include "VBoxDD.h"
 
+/*********************************************************************************************************************************
+*   Defines                                                                                                                      *
+*********************************************************************************************************************************/
 #define VBOX_PULSEAUDIO_MAX_LOG_REL_ERRORS 32 /** @todo Make this configurable thru driver options. */
 
@@ -51 +53 @@
 #endif
 
-/*
- * We use a g_pMainLoop in a separate thread g_pContext. We have to call functions for
- * manipulating objects either from callback functions or we have to protect
- * these functions by pa_threaded_mainloop_lock() / pa_threaded_mainloop_unlock().
- */
-static struct pa_threaded_mainloop *g_pMainLoop;
-static struct pa_context           *g_pContext;
-static volatile bool                g_fAbortMainLoop;
+/** No flags specified. */
+#define PULSEAUDIOENUMCBFLAGS_NONE          0
+/** (Release) log found devices. */
+#define PULSEAUDIOENUMCBFLAGS_LOG           RT_BIT(0)
+
+/** Makes DRVHOSTPULSEAUDIO out of PDMIHOSTAUDIO. */
+#define PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface) \
+    ( (PDRVHOSTPULSEAUDIO)((uintptr_t)pInterface - RT_OFFSETOF(DRVHOSTPULSEAUDIO, IHostAudio)) )
+
+/*********************************************************************************************************************************
+*   Structures                                                                                                                   *
+*********************************************************************************************************************************/
 
 /**
@@ -67 +73 @@
 {
     /** Pointer to the driver instance structure. */
-    PPDMDRVINS         pDrvIns;
+    PPDMDRVINS            pDrvIns;
+    /** Pointer to PulseAudio's threaded main loop. */
+    pa_threaded_mainloop *pMainLoop;
+    /**
+    * Pointer to our PulseAudio context.
+    * Note: We use a pMainLoop in a separate thread (pContext).
+    *       So either use callback functions or protect these functions
+    *       by pa_threaded_mainloop_lock() / pa_threaded_mainloop_unlock().
+    */
+    pa_context           *pContext;
+    /** Shutdown indicator. */
+    bool                  fLoopWait;
     /** Pointer to host audio interface. */
-    PDMIHOSTAUDIO      IHostAudio;
+    PDMIHOSTAUDIO         IHostAudio;
     /** Error count for not flooding the release log.
-     *  UINT32_MAX for unlimited logging. */
-    uint32_t           cLogErrors;
-    /** Configuration option: stream name. */
-    char               *pszStreamName;
+     *  Specify UINT32_MAX for unlimited logging. */
+    uint32_t              cLogErrors;
+    /** Configuration option: stream name. Optional. */
+    char                 *pszStreamName;
 } DRVHOSTPULSEAUDIO, *PDRVHOSTPULSEAUDIO;
 
@@ -124 +141 @@
 };
 
-/** Makes DRVHOSTPULSEAUDIO out of PDMIHOSTAUDIO. */
-#define PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface) \
-    ( (PDRVHOSTPULSEAUDIO)((uintptr_t)pInterface - RT_OFFSETOF(DRVHOSTPULSEAUDIO, IHostAudio)) )
-
-static int  drvHostPulseAudioError(PDRVHOSTPULSEAUDIO pThis, const char *szMsg);
-static void drvHostPulseAudioCbSuccess(pa_stream *pStream, int fSuccess, void *pvContext);
+/**
+ * Callback context for server enumeration callbacks.
+ */
+typedef struct PULSEAUDIOENUMCBCTX
+{
+    /** Pointer to host backend driver. */
+    PDRVHOSTPULSEAUDIO  pDrv;
+    /** Enumeration flags. */
+    uint32_t            fFlags;
+    /** Number of found input devices. */
+    uint8_t             cDevIn;
+    /** Number of found output devices. */
+    uint8_t             cDevOut;
+    /** Name of default sink being used. Must be free'd using RTStrFree(). */
+    char               *pszDefaultSink;
+    /** Name of default source being used. Must be free'd using RTStrFree(). */
+    char               *pszDefaultSource;
+} PULSEAUDIOENUMCBCTX, *PPULSEAUDIOENUMCBCTX;
+
+/*********************************************************************************************************************************
+*   Prototypes                                                                                                                   *
+*********************************************************************************************************************************/
+
+static int  paEnumerate(PDRVHOSTPULSEAUDIO pThis, PPDMAUDIOBACKENDCFG pCfg, uint32_t fEnum);
+static int  paError(PDRVHOSTPULSEAUDIO pThis, const char *szMsg);
+static void paStreamCbSuccess(pa_stream *pStream, int fSuccess, void *pvContext);
 
 /**
@@ -135 +172 @@
  * mainloop might not have been entered yet.
  */
-static void drvHostPulseAudioAbortMainLoop(void)
-{
-    g_fAbortMainLoop = true;
-    pa_threaded_mainloop_signal(g_pMainLoop, 0);
-}
-
-static pa_sample_format_t drvHostPulseAudioFmtToPulse(PDMAUDIOFMT fmt)
+static void paSignalWaiter(PDRVHOSTPULSEAUDIO pThis)
+{
+    pThis->fLoopWait = true;
+    pa_threaded_mainloop_signal(pThis->pMainLoop, 0);
+}
+
+static pa_sample_format_t paFmtToPulse(PDMAUDIOFMT fmt)
 {
     switch (fmt)
@@ -163 +200 @@
 }
 
-static int drvHostPulseAudioPulseToFmt(pa_sample_format_t pulsefmt,
-                                       PDMAUDIOFMT *pFmt, PDMAUDIOENDIANNESS *pEndianness)
+static int paPulseToFmt(pa_sample_format_t pulsefmt,
+                        PDMAUDIOFMT *pFmt, PDMAUDIOENDIANNESS *pEndianness)
 {
     switch (pulsefmt)
@@ -208 +245 @@
  * Synchronously wait until an operation completed.
  */
-static int drvHostPulseAudioWaitFor(pa_operation *pOP, RTMSINTERVAL cMsTimeout)
-{
-    AssertPtrReturn(pOP, VERR_INVALID_POINTER);
+static int paWaitForEx(PDRVHOSTPULSEAUDIO pThis, pa_operation *pOP, RTMSINTERVAL cMsTimeout)
+{
+    AssertPtrReturn(pThis, VERR_INVALID_POINTER);
+    AssertPtrReturn(pOP,   VERR_INVALID_POINTER);
 
     int rc = VINF_SUCCESS;
-    if (pOP)
-    {
-        uint64_t u64StartMs = RTTimeMilliTS();
-        while (pa_operation_get_state(pOP) == PA_OPERATION_RUNNING)
-        {
-            if (!g_fAbortMainLoop)
-                pa_threaded_mainloop_wait(g_pMainLoop);
-            g_fAbortMainLoop = false;
-
-            uint64_t u64ElapsedMs = RTTimeMilliTS() - u64StartMs;
-            if (u64ElapsedMs >= cMsTimeout)
-            {
-                rc = VERR_TIMEOUT;
-                break;
-            }
-        }
-
-        pa_operation_unref(pOP);
-    }
+
+    uint64_t u64StartMs = RTTimeMilliTS();
+    while (pa_operation_get_state(pOP) == PA_OPERATION_RUNNING)
+    {
+        if (!pThis->fLoopWait)
+        {
+            AssertPtr(pThis->pMainLoop);
+            pa_threaded_mainloop_wait(pThis->pMainLoop);
+        }
+        pThis->fLoopWait = false;
+
+        uint64_t u64ElapsedMs = RTTimeMilliTS() - u64StartMs;
+        if (u64ElapsedMs >= cMsTimeout)
+        {
+            rc = VERR_TIMEOUT;
+            break;
+        }
+    }
+
+    pa_operation_unref(pOP);
 
     return rc;
+}
+
+static int paWaitFor(PDRVHOSTPULSEAUDIO pThis, pa_operation *pOP)
+{
+    return paWaitForEx(pThis, pOP, 10 * 1000 /* 10s timeout */);
 }
 
@@ -239 +283 @@
  * Context status changed.
  */
-static void drvHostPulseAudioCbCtxState(pa_context *pContext, void *pvUser)
-{
-    AssertPtrReturnVoid(pContext);
-    NOREF(pvUser);
-
-    switch (pa_context_get_state(pContext))
+static void paContextCbStateChanged(pa_context *pCtx, void *pvUser)
+{
+    AssertPtrReturnVoid(pCtx);
+
+    PDRVHOSTPULSEAUDIO pThis = (PDRVHOSTPULSEAUDIO)pvUser;
+    AssertPtrReturnVoid(pThis);
+
+    switch (pa_context_get_state(pCtx))
     {
         case PA_CONTEXT_READY:
         case PA_CONTEXT_TERMINATED:
-            drvHostPulseAudioAbortMainLoop();
+            paSignalWaiter(pThis);
             break;
 
         case PA_CONTEXT_FAILED:
-            LogRel(("PulseAudio: Audio input/output stopped!\n"));
-            drvHostPulseAudioAbortMainLoop();
+            LogRel(("PulseAudio: Audio context has failed, stopping\n"));
+            paSignalWaiter(pThis);
             break;
 
@@ -264 +310 @@
  * Callback called when our pa_stream_drain operation was completed.
  */
-static void drvHostPulseAudioCbStreamDrain(pa_stream *pStream, int fSuccess, void *pvContext)
+static void paStreamCbDrain(pa_stream *pStream, int fSuccess, void *pvUser)
 {
     AssertPtrReturnVoid(pStream);
 
-    PPULSEAUDIOSTREAM pStrm = (PPULSEAUDIOSTREAM)pvContext;
+    PPULSEAUDIOSTREAM pStrm = (PPULSEAUDIOSTREAM)pvUser;
     AssertPtrReturnVoid(pStrm);
 
@@ -275 +321 @@
     {
         pa_operation_unref(pa_stream_cork(pStream, 1,
-                                          drvHostPulseAudioCbSuccess, pvContext));
+                                          paStreamCbSuccess, pvUser));
     }
     else
-        drvHostPulseAudioError(pStrm->pDrv, "Failed to drain stream");
+        paError(pStrm->pDrv, "Failed to drain stream");
 
     pa_operation_unref(pStrm->pDrainOp);
@@ -287 +333 @@
  * Stream status changed.
  */
-static void drvHostPulseAudioCbStreamState(pa_stream *pStream, void *pvContext)
+static void paStreamCbStateChanged(pa_stream *pStream, void *pvUser)
 {
     AssertPtrReturnVoid(pStream);
-    NOREF(pvContext);
+
+    PDRVHOSTPULSEAUDIO pThis = (PDRVHOSTPULSEAUDIO)pvUser;
+    AssertPtrReturnVoid(pThis);
 
     switch (pa_stream_get_state(pStream))
@@ -297 +345 @@
         case PA_STREAM_FAILED:
         case PA_STREAM_TERMINATED:
-            drvHostPulseAudioAbortMainLoop();
+            paSignalWaiter(pThis);
             break;
 
@@ -305 +353 @@
 }
 
-static void drvHostPulseAudioCbSuccess(pa_stream *pStream, int fSuccess, void *pvContext)
+static void paStreamCbSuccess(pa_stream *pStream, int fSuccess, void *pvUser)
 {
     AssertPtrReturnVoid(pStream);
 
-    PPULSEAUDIOSTREAM pStrm = (PPULSEAUDIOSTREAM)pvContext;
+    PPULSEAUDIOSTREAM pStrm = (PPULSEAUDIOSTREAM)pvUser;
     AssertPtrReturnVoid(pStrm);
 
@@ -315 +363 @@
 
     if (fSuccess)
-        drvHostPulseAudioAbortMainLoop();
+        paSignalWaiter(pStrm->pDrv);
     else
-        drvHostPulseAudioError(pStrm->pDrv, "Failed to finish stream operation");
-}
-
-static int drvHostPulseAudioOpen(bool fIn, const char *pszName,
-                                 pa_sample_spec *pSampleSpec, pa_buffer_attr *pBufAttr,
-                                 pa_stream **ppStream)
-{
-    AssertPtrReturn(pszName, VERR_INVALID_POINTER);
+        paError(pStrm->pDrv, "Failed to finish stream operation");
+}
+
+static int paStreamOpen(PDRVHOSTPULSEAUDIO pThis, bool fIn, const char *pszName,
+                        pa_sample_spec *pSampleSpec, pa_buffer_attr *pBufAttr,
+                        pa_stream **ppStream)
+{
+    AssertPtrReturn(pThis,       VERR_INVALID_POINTER);
+    AssertPtrReturn(pszName,     VERR_INVALID_POINTER);
     AssertPtrReturn(pSampleSpec, VERR_INVALID_POINTER);
-    AssertPtrReturn(pBufAttr, VERR_INVALID_POINTER);
-    AssertPtrReturn(ppStream, VERR_INVALID_POINTER);
+    AssertPtrReturn(pBufAttr,    VERR_INVALID_POINTER);
+    AssertPtrReturn(ppStream,    VERR_INVALID_POINTER);
 
     if (!pa_sample_spec_valid(pSampleSpec))
@@ -345 +394 @@
              pa_sample_format_to_string(pSampleSpec->format)));
 
-    pa_threaded_mainloop_lock(g_pMainLoop);
+    pa_threaded_mainloop_lock(pThis->pMainLoop);
 
     do
     {
-        if (!(pStream = pa_stream_new(g_pContext, pszName, pSampleSpec,
+        if (!(pStream = pa_stream_new(pThis->pContext, pszName, pSampleSpec,
                                       NULL /* pa_channel_map */)))
         {
@@ -357 +406 @@
         }
 
-        pa_stream_set_state_callback(pStream, drvHostPulseAudioCbStreamState, NULL);
+        pa_stream_set_state_callback(pStream, paStreamCbStateChanged, pThis);
 
 #if PA_API_VERSION >= 12
@@ -379 +428 @@
             {
                 LogRel(("PulseAudio: Could not connect input stream \"%s\": %s\n",
-                        pszName, pa_strerror(pa_context_errno(g_pContext))));
+                        pszName, pa_strerror(pa_context_errno(pThis->pContext))));
                 rc = VERR_AUDIO_BACKEND_INIT_FAILED;
                 break;
@@ -393 +442 @@
             {
                 LogRel(("PulseAudio: Could not connect playback stream \"%s\": %s\n",
-                        pszName, pa_strerror(pa_context_errno(g_pContext))));
+                        pszName, pa_strerror(pa_context_errno(pThis->pContext))));
                 rc = VERR_AUDIO_BACKEND_INIT_FAILED;
                 break;
@@ -402 +451 @@
         for (;;)
         {
-            if (!g_fAbortMainLoop)
-                pa_threaded_mainloop_wait(g_pMainLoop);
-            g_fAbortMainLoop = false;
-
-            pa_stream_state_t sstate = pa_stream_get_state(pStream);
-            if (sstate == PA_STREAM_READY)
+            if (!pThis->fLoopWait)
+                pa_threaded_mainloop_wait(pThis->pMainLoop);
+            pThis->fLoopWait = false;
+
+            pa_stream_state_t streamSt = pa_stream_get_state(pStream);
+            if (streamSt == PA_STREAM_READY)
                 break;
-            else if (   sstate == PA_STREAM_FAILED
-                     || sstate == PA_STREAM_TERMINATED)
+            else if (   streamSt == PA_STREAM_FAILED
+                     || streamSt == PA_STREAM_TERMINATED)
             {
-                LogRel(("PulseAudio: Failed to initialize stream \"%s\" (state %ld)\n",
-                        pszName, sstate));
+                LogRel(("PulseAudio: Failed to initialize stream \"%s\" (state %ld)\n", pszName, streamSt));
                 rc = VERR_AUDIO_BACKEND_INIT_FAILED;
                 break;
@@ -440 +488 @@
         pa_stream_disconnect(pStream);
 
-    pa_threaded_mainloop_unlock(g_pMainLoop);
+    pa_threaded_mainloop_unlock(pThis->pMainLoop);
 
     if (RT_FAILURE(rc))
@@ -456 +504 @@
 static DECLCALLBACK(int) drvHostPulseAudioInit(PPDMIHOSTAUDIO pInterface)
 {
-    NOREF(pInterface);
+    AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
+
+    PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface);
 
     LogFlowFuncEnter();
@@ -467 +517 @@
     }
 
+    pThis->fLoopWait = false;
+    pThis->pMainLoop = NULL;
+
     bool fLocked = false;
 
     do
     {
-        if (!(g_pMainLoop = pa_threaded_mainloop_new()))
+        if (!(pThis->pMainLoop = pa_threaded_mainloop_new()))
         {
             LogRel(("PulseAudio: Failed to allocate main loop: %s\n",
-                     pa_strerror(pa_context_errno(g_pContext))));
+                     pa_strerror(pa_context_errno(pThis->pContext))));
             rc = VERR_NO_MEMORY;
             break;
         }
 
-        if (!(g_pContext = pa_context_new(pa_threaded_mainloop_get_api(g_pMainLoop), "VirtualBox")))
+        if (!(pThis->pContext = pa_context_new(pa_threaded_mainloop_get_api(pThis->pMainLoop), "VirtualBox")))
         {
             LogRel(("PulseAudio: Failed to allocate context: %s\n",
-                     pa_strerror(pa_context_errno(g_pContext))));
+                     pa_strerror(pa_context_errno(pThis->pContext))));
             rc = VERR_NO_MEMORY;
             break;
         }
 
-        if (pa_threaded_mainloop_start(g_pMainLoop) < 0)
+        if (pa_threaded_mainloop_start(pThis->pMainLoop) < 0)
         {
             LogRel(("PulseAudio: Failed to start threaded mainloop: %s\n",
-                     pa_strerror(pa_context_errno(g_pContext))));
+                     pa_strerror(pa_context_errno(pThis->pContext))));
             rc = VERR_AUDIO_BACKEND_INIT_FAILED;
             break;
         }
 
-        g_fAbortMainLoop = false;
-        pa_context_set_state_callback(g_pContext, drvHostPulseAudioCbCtxState, NULL);
-        pa_threaded_mainloop_lock(g_pMainLoop);
+        /* Install a global callback to known if something happens to our acquired context. */
+        pa_context_set_state_callback(pThis->pContext, paContextCbStateChanged, pThis /* pvUserData */);
+
+        pa_threaded_mainloop_lock(pThis->pMainLoop);
         fLocked = true;
 
-        if (pa_context_connect(g_pContext, NULL /* pszServer */,
+        if (pa_context_connect(pThis->pContext, NULL /* pszServer */,
                                PA_CONTEXT_NOFLAGS, NULL) < 0)
         {
             LogRel(("PulseAudio: Failed to connect to server: %s\n",
-                     pa_strerror(pa_context_errno(g_pContext))));
+                     pa_strerror(pa_context_errno(pThis->pContext))));
             rc = VERR_AUDIO_BACKEND_INIT_FAILED;
             break;
         }
 
-        /* Wait until the g_pContext is ready */
+        /* Wait until the pThis->pContext is ready. */
         for (;;)
         {
-            if (!g_fAbortMainLoop)
-                pa_threaded_mainloop_wait(g_pMainLoop);
-            g_fAbortMainLoop = false;
-
-            pa_context_state_t cstate = pa_context_get_state(g_pContext);
+            if (!pThis->fLoopWait)
+                pa_threaded_mainloop_wait(pThis->pMainLoop);
+            pThis->fLoopWait = false;
+
+            pa_context_state_t cstate = pa_context_get_state(pThis->pContext);
             if (cstate == PA_CONTEXT_READY)
                 break;
@@ -531 +585 @@
 
     if (fLocked)
-        pa_threaded_mainloop_unlock(g_pMainLoop);
+        pa_threaded_mainloop_unlock(pThis->pMainLoop);
 
     if (RT_FAILURE(rc))
     {
-        if (g_pMainLoop)
-            pa_threaded_mainloop_stop(g_pMainLoop);
-
-        if (g_pContext)
-        {
-            pa_context_disconnect(g_pContext);
-            pa_context_unref(g_pContext);
-            g_pContext = NULL;
-        }
-
-        if (g_pMainLoop)
-        {
-            pa_threaded_mainloop_free(g_pMainLoop);
-            g_pMainLoop = NULL;
+        if (pThis->pMainLoop)
+            pa_threaded_mainloop_stop(pThis->pMainLoop);
+
+        if (pThis->pContext)
+        {
+            pa_context_disconnect(pThis->pContext);
+            pa_context_unref(pThis->pContext);
+            pThis->pContext = NULL;
+        }
+
+        if (pThis->pMainLoop)
+        {
+            pa_threaded_mainloop_free(pThis->pMainLoop);
+            pThis->pMainLoop = NULL;
         }
     }
@@ -560 +614 @@
                                                   uint32_t *pcSamples)
 {
-    AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
+    AssertPtrReturn(pInterface,  VERR_INVALID_POINTER);
     AssertPtrReturn(pHstStrmOut, VERR_INVALID_POINTER);
-    AssertPtrReturn(pCfg, VERR_INVALID_POINTER);
+    AssertPtrReturn(pCfg,        VERR_INVALID_POINTER);
     /* pcSamples is optional. */
 
-    PDRVHOSTPULSEAUDIO pDrv = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface);
-    PPULSEAUDIOSTREAM pThisStrmOut = (PPULSEAUDIOSTREAM)pHstStrmOut;
+    PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface);
+    PPULSEAUDIOSTREAM  pStrm = (PPULSEAUDIOSTREAM)pHstStrmOut;
 
     LogFlowFuncEnter();
 
-    pThisStrmOut->pDrainOp            = NULL;
-
-    pThisStrmOut->SampleSpec.format   = drvHostPulseAudioFmtToPulse(pCfg->enmFormat);
-    pThisStrmOut->SampleSpec.rate     = pCfg->uHz;
-    pThisStrmOut->SampleSpec.channels = pCfg->cChannels;
+    pStrm->pDrainOp            = NULL;
+
+    pStrm->SampleSpec.format   = paFmtToPulse(pCfg->enmFormat);
+    pStrm->SampleSpec.rate     = pCfg->uHz;
+    pStrm->SampleSpec.channels = pCfg->cChannels;
 
     /* Note that setting maxlength to -1 does not work on PulseAudio servers
      * older than 0.9.10. So use the suggested value of 3/2 of tlength */
-    pThisStrmOut->BufAttr.tlength     =   (pa_bytes_per_second(&pThisStrmOut->SampleSpec)
+    pStrm->BufAttr.tlength     =   (pa_bytes_per_second(&pStrm->SampleSpec)
                                         * s_pulseCfg.buffer_msecs_out) / 1000;
-    pThisStrmOut->BufAttr.maxlength   = (pThisStrmOut->BufAttr.tlength * 3) / 2;
-    pThisStrmOut->BufAttr.prebuf      = -1; /* Same as tlength */
-    pThisStrmOut->BufAttr.minreq      = -1; /* Pulse should set something sensible for minreq on it's own */
+    pStrm->BufAttr.maxlength   = (pStrm->BufAttr.tlength * 3) / 2;
+    pStrm->BufAttr.prebuf      = -1; /* Same as tlength */
+    pStrm->BufAttr.minreq      = -1; /* Pulse should set something sensible for minreq on it's own */
 
     /* Note that the struct BufAttr is updated to the obtained values after this call! */
     char achName[64];
-    RTStrPrintf(achName, sizeof(achName), "%.32s (out)", pDrv->pszStreamName);
-    int rc = drvHostPulseAudioOpen(false /* fIn */, achName, &pThisStrmOut->SampleSpec, &pThisStrmOut->BufAttr,
-                                   &pThisStrmOut->pStream);
+    RTStrPrintf(achName, sizeof(achName), "%.32s (out)", pThis->pszStreamName);
+
+    int rc = paStreamOpen(pThis, false /* fIn */, achName, &pStrm->SampleSpec, &pStrm->BufAttr, &pStrm->pStream);
     if (RT_FAILURE(rc))
         return rc;
 
     PDMAUDIOSTREAMCFG streamCfg;
-    rc = drvHostPulseAudioPulseToFmt(pThisStrmOut->SampleSpec.format,
-                                     &streamCfg.enmFormat, &streamCfg.enmEndianness);
+    rc = paPulseToFmt(pStrm->SampleSpec.format,
+                      &streamCfg.enmFormat, &streamCfg.enmEndianness);
     if (RT_FAILURE(rc))
     {
-        LogRel(("PulseAudio: Cannot find audio output format %ld\n", pThisStrmOut->SampleSpec.format));
+        LogRel(("PulseAudio: Cannot find audio output format %ld\n", pStrm->SampleSpec.format));
         return rc;
     }
 
-    streamCfg.uHz       = pThisStrmOut->SampleSpec.rate;
-    streamCfg.cChannels = pThisStrmOut->SampleSpec.channels;
+    streamCfg.uHz       = pStrm->SampleSpec.rate;
+    streamCfg.cChannels = pStrm->SampleSpec.channels;
 
     rc = DrvAudioStreamCfgToProps(&streamCfg, &pHstStrmOut->Props);
     if (RT_SUCCESS(rc))
     {
-        uint32_t cbBuf  = RT_MIN(pThisStrmOut->BufAttr.tlength * 2,
-                                 pThisStrmOut->BufAttr.maxlength); /** @todo Make this configurable! */
+        uint32_t cbBuf  = RT_MIN(pStrm->BufAttr.tlength * 2,
+                                 pStrm->BufAttr.maxlength); /** @todo Make this configurable! */
         if (cbBuf)
         {
-            pThisStrmOut->pvPCMBuf = RTMemAllocZ(cbBuf);
-            if (pThisStrmOut->pvPCMBuf)
+            pStrm->pvPCMBuf = RTMemAllocZ(cbBuf);
+            if (pStrm->pvPCMBuf)
             {
-                pThisStrmOut->cbPCMBuf = cbBuf;
+                pStrm->cbPCMBuf = cbBuf;
 
                 uint32_t cSamples = cbBuf >> pHstStrmOut->Props.cShift;
@@ -621 +675 @@
 
                 /* Save pointer to driver instance. */
-                pThisStrmOut->pDrv = pDrv;
+                pStrm->pDrv = pThis;
 
                 LogFunc(("cbBuf=%RU32, cSamples=%RU32\n", cbBuf, cSamples));
@@ -653 +707 @@
     /* pcSamples is optional. */
 
-    PDRVHOSTPULSEAUDIO pDrv = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface);
-    PPULSEAUDIOSTREAM pThisStrmIn = (PPULSEAUDIOSTREAM)pHstStrmIn;
+    PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface);
+    PPULSEAUDIOSTREAM  pStrm = (PPULSEAUDIOSTREAM)pHstStrmIn;
 
     LogFunc(("enmRecSrc=%ld\n", enmRecSource));
 
-    pThisStrmIn->SampleSpec.format   = drvHostPulseAudioFmtToPulse(pCfg->enmFormat);
-    pThisStrmIn->SampleSpec.rate     = pCfg->uHz;
-    pThisStrmIn->SampleSpec.channels = pCfg->cChannels;
+    pStrm->SampleSpec.format   = paFmtToPulse(pCfg->enmFormat);
+    pStrm->SampleSpec.rate     = pCfg->uHz;
+    pStrm->SampleSpec.channels = pCfg->cChannels;
 
     /* XXX check these values */
-    pThisStrmIn->BufAttr.fragsize    = (pa_bytes_per_second(&pThisStrmIn->SampleSpec)
+    pStrm->BufAttr.fragsize    = (pa_bytes_per_second(&pStrm->SampleSpec)
                                    * s_pulseCfg.buffer_msecs_in) / 1000;
-    pThisStrmIn->BufAttr.maxlength   = (pThisStrmIn->BufAttr.fragsize * 3) / 2;
+    pStrm->BufAttr.maxlength   = (pStrm->BufAttr.fragsize * 3) / 2;
     /* Note: Other members of pa_buffer_attr are ignored for record streams. */
 
     char achName[64];
-    RTStrPrintf(achName, sizeof(achName), "%.32s (in)", pDrv->pszStreamName);
-    int rc = drvHostPulseAudioOpen(true /* fIn */, achName, &pThisStrmIn->SampleSpec, &pThisStrmIn->BufAttr,
-                                   &pThisStrmIn->pStream);
+    RTStrPrintf(achName, sizeof(achName), "%.32s (in)", pThis->pszStreamName);
+
+    int rc = paStreamOpen(pThis, true /* fIn */, achName, &pStrm->SampleSpec, &pStrm->BufAttr,
+                          &pStrm->pStream);
     if (RT_FAILURE(rc))
         return rc;
 
     PDMAUDIOSTREAMCFG streamCfg;
-    rc = drvHostPulseAudioPulseToFmt(pThisStrmIn->SampleSpec.format, &streamCfg.enmFormat,
-                                     &streamCfg.enmEndianness);
+    rc = paPulseToFmt(pStrm->SampleSpec.format, &streamCfg.enmFormat,
+                      &streamCfg.enmEndianness);
     if (RT_FAILURE(rc))
     {
-        LogRel(("PulseAudio: Cannot find audio capture format %ld\n", pThisStrmIn->SampleSpec.format));
+        LogRel(("PulseAudio: Cannot find audio capture format %ld\n", pStrm->SampleSpec.format));
         return rc;
     }
 
-    streamCfg.uHz       = pThisStrmIn->SampleSpec.rate;
-    streamCfg.cChannels = pThisStrmIn->SampleSpec.channels;
+    streamCfg.uHz       = pStrm->SampleSpec.rate;
+    streamCfg.cChannels = pStrm->SampleSpec.channels;
 
     rc = DrvAudioStreamCfgToProps(&streamCfg, &pHstStrmIn->Props);
     if (RT_SUCCESS(rc))
     {
-        uint32_t cSamples = RT_MIN(pThisStrmIn->BufAttr.fragsize * 10, pThisStrmIn->BufAttr.maxlength)
+        uint32_t cSamples = RT_MIN(pStrm->BufAttr.fragsize * 10, pStrm->BufAttr.maxlength)
                             >> pHstStrmIn->Props.cShift;
         LogFunc(("cShift=%RU8, cSamples=%RU32\n", pHstStrmIn->Props.cShift, cSamples));
@@ -698 +753 @@
 
         /* Save pointer to driver instance. */
-        pThisStrmIn->pDrv = pDrv;
-
-        pThisStrmIn->pu8PeekBuf = NULL;
+        pStrm->pDrv = pThis;
+
+        pStrm->pu8PeekBuf = NULL;
     }
 
@@ -710 +765 @@
                                                     uint32_t *pcSamplesCaptured)
 {
-    NOREF(pInterface);
+    AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
     AssertPtrReturn(pHstStrmIn, VERR_INVALID_POINTER);
     /* pcSamplesPlayed is optional. */
 
-    PPULSEAUDIOSTREAM pThisStrmIn = (PPULSEAUDIOSTREAM)pHstStrmIn;
+    PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface);
+    PPULSEAUDIOSTREAM  pStrm = (PPULSEAUDIOSTREAM)pHstStrmIn;
 
     /* We should only call pa_stream_readable_size() once and trust the first value. */
-    pa_threaded_mainloop_lock(g_pMainLoop);
-    size_t cbAvail = pa_stream_readable_size(pThisStrmIn->pStream);
-    pa_threaded_mainloop_unlock(g_pMainLoop);
+    pa_threaded_mainloop_lock(pThis->pMainLoop);
+    size_t cbAvail = pa_stream_readable_size(pStrm->pStream);
+    pa_threaded_mainloop_unlock(pThis->pMainLoop);
 
     if (cbAvail == (size_t)-1)
-        return drvHostPulseAudioError(pThisStrmIn->pDrv, "Failed to determine input data size");
+        return paError(pStrm->pDrv, "Failed to determine input data size");
 
     /* If the buffer was not dropped last call, add what remains. */
-    if (pThisStrmIn->pu8PeekBuf)
-    {
-        Assert(pThisStrmIn->cbPeekBuf >= pThisStrmIn->offPeekBuf);
-        cbAvail += (pThisStrmIn->cbPeekBuf - pThisStrmIn->offPeekBuf);
+    if (pStrm->pu8PeekBuf)
+    {
+        Assert(pStrm->cbPeekBuf >= pStrm->offPeekBuf);
+        cbAvail += (pStrm->cbPeekBuf - pStrm->offPeekBuf);
     }
 
@@ -739 +795 @@
 
     int rc = VINF_SUCCESS;
+
     size_t cbToRead = RT_MIN(cbAvail, AudioMixBufFreeBytes(&pHstStrmIn->MixBuf));
 
     LogFlowFunc(("cbToRead=%zu, cbAvail=%zu, offPeekBuf=%zu, cbPeekBuf=%zu\n",
-                 cbToRead, cbAvail, pThisStrmIn->offPeekBuf, pThisStrmIn->cbPeekBuf));
-
-    size_t offWrite = 0;
+                 cbToRead, cbAvail, pStrm->offPeekBuf, pStrm->cbPeekBuf));
+
+    size_t   offWrite      = 0;
     uint32_t cWrittenTotal = 0;
 
@@ -750 +807 @@
     {
         /* If there is no data, do another peek. */
-        if (!pThisStrmIn->pu8PeekBuf)
-        {
-            pa_threaded_mainloop_lock(g_pMainLoop);
-            pa_stream_peek(pThisStrmIn->pStream,
-                           (const void**)&pThisStrmIn->pu8PeekBuf, &pThisStrmIn->cbPeekBuf);
-            pa_threaded_mainloop_unlock(g_pMainLoop);
-
-            pThisStrmIn->offPeekBuf = 0;
+        if (!pStrm->pu8PeekBuf)
+        {
+            pa_threaded_mainloop_lock(pThis->pMainLoop);
+            pa_stream_peek(pStrm->pStream,
+                           (const void**)&pStrm->pu8PeekBuf, &pStrm->cbPeekBuf);
+            pa_threaded_mainloop_unlock(pThis->pMainLoop);
+
+            pStrm->offPeekBuf = 0;
 
             /* No data anymore?
              * Note: If there's a data hole (cbPeekBuf then contains the length of the hole)
              *       we need to drop the stream lateron. */
-            if (   !pThisStrmIn->pu8PeekBuf
-                && !pThisStrmIn->cbPeekBuf)
+            if (   !pStrm->pu8PeekBuf
+                && !pStrm->cbPeekBuf)
             {
                 break;
@@ -769 +826 @@
         }
 
-        Assert(pThisStrmIn->cbPeekBuf >= pThisStrmIn->offPeekBuf);
-        size_t cbToWrite = RT_MIN(pThisStrmIn->cbPeekBuf - pThisStrmIn->offPeekBuf, cbToRead);
+        Assert(pStrm->cbPeekBuf >= pStrm->offPeekBuf);
+        size_t cbToWrite = RT_MIN(pStrm->cbPeekBuf - pStrm->offPeekBuf, cbToRead);
 
         LogFlowFunc(("cbToRead=%zu, cbToWrite=%zu, offPeekBuf=%zu, cbPeekBuf=%zu, pu8PeekBuf=%p\n",
                      cbToRead, cbToWrite,
-                     pThisStrmIn->offPeekBuf, pThisStrmIn->cbPeekBuf, pThisStrmIn->pu8PeekBuf));
+                     pStrm->offPeekBuf, pStrm->cbPeekBuf, pStrm->pu8PeekBuf));
 
         if (cbToWrite)
@@ -780 +837 @@
             uint32_t cWritten;
             rc = AudioMixBufWriteCirc(&pHstStrmIn->MixBuf,
-                                      pThisStrmIn->pu8PeekBuf + pThisStrmIn->offPeekBuf,
+                                      pStrm->pu8PeekBuf + pStrm->offPeekBuf,
                                       cbToWrite, &cWritten);
             if (RT_FAILURE(rc))
@@ -790 +847 @@
             cbToRead -= cbWritten;
             cWrittenTotal += cWritten;
-            pThisStrmIn->offPeekBuf += cbWritten;
+            pStrm->offPeekBuf += cbWritten;
         }
 
@@ -796 +853 @@
                !cbToWrite
             /* Was there a hole in the peeking buffer? Drop it. */
-            || !pThisStrmIn->pu8PeekBuf
+            || !pStrm->pu8PeekBuf
             /* If the buffer is done, drop it. */
-            || pThisStrmIn->offPeekBuf == pThisStrmIn->cbPeekBuf)
-        {
-            pa_threaded_mainloop_lock(g_pMainLoop);
-            pa_stream_drop(pThisStrmIn->pStream);
-            pa_threaded_mainloop_unlock(g_pMainLoop);
-
-            pThisStrmIn->pu8PeekBuf = NULL;
+            || pStrm->offPeekBuf == pStrm->cbPeekBuf)
+        {
+            pa_threaded_mainloop_lock(pThis->pMainLoop);
+            pa_stream_drop(pStrm->pStream);
+            pa_threaded_mainloop_unlock(pThis->pMainLoop);
+
+            pStrm->pu8PeekBuf = NULL;
         }
     }
@@ -829 +886 @@
                                                   uint32_t *pcSamplesPlayed)
 {
-    NOREF(pInterface);
+    AssertPtrReturn(pInterface,  VERR_INVALID_POINTER);
     AssertPtrReturn(pHstStrmOut, VERR_INVALID_POINTER);
     /* pcSamplesPlayed is optional. */
 
-    PPULSEAUDIOSTREAM pThisStrmOut = (PPULSEAUDIOSTREAM)pHstStrmOut;
+    PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface);
+    PPULSEAUDIOSTREAM  pStrm = (PPULSEAUDIOSTREAM)pHstStrmOut;
 
     int rc = VINF_SUCCESS;
@@ -848 +906 @@
     }
 
-    pa_threaded_mainloop_lock(g_pMainLoop);
+    pa_threaded_mainloop_lock(pThis->pMainLoop);
 
     do
     {
-        size_t cbWriteable = pa_stream_writable_size(pThisStrmOut->pStream);
+        size_t cbWriteable = pa_stream_writable_size(pStrm->pStream);
         if (cbWriteable == (size_t)-1)
         {
-            rc = drvHostPulseAudioError(pThisStrmOut->pDrv, "Failed to determine output data size");
+            rc = paError(pStrm->pDrv, "Failed to determine output data size");
             break;
         }
@@ -868 +926 @@
         while (cbToRead)
         {
-            rc = AudioMixBufReadCirc(&pHstStrmOut->MixBuf, pThisStrmOut->pvPCMBuf,
-                                     RT_MIN(cbToRead, pThisStrmOut->cbPCMBuf), &cRead);
+            rc = AudioMixBufReadCirc(&pHstStrmOut->MixBuf, pStrm->pvPCMBuf,
+                                     RT_MIN(cbToRead, pStrm->cbPCMBuf), &cRead);
             if (   !cRead
                 || RT_FAILURE(rc))
@@ -877 +935 @@
 
             cbRead = AUDIOMIXBUF_S2B(&pHstStrmOut->MixBuf, cRead);
-            if (pa_stream_write(pThisStrmOut->pStream, pThisStrmOut->pvPCMBuf, cbRead, NULL /* Cleanup callback */,
+            if (pa_stream_write(pStrm->pStream, pStrm->pvPCMBuf, cbRead, NULL /* Cleanup callback */,
                                 0, PA_SEEK_RELATIVE) < 0)
             {
-                rc = drvHostPulseAudioError(pThisStrmOut->pDrv, "Failed to write to output stream");
+                rc = paError(pStrm->pDrv, "Failed to write to output stream");
                 break;
             }
@@ -894 +952 @@
     } while (0);
 
-    pa_threaded_mainloop_unlock(g_pMainLoop);
+    pa_threaded_mainloop_unlock(pThis->pMainLoop);
 
     if (RT_SUCCESS(rc))
@@ -913 +971 @@
 
 /** @todo Implement va handling. */
-static int drvHostPulseAudioError(PDRVHOSTPULSEAUDIO pThis, const char *szMsg)
+static int paError(PDRVHOSTPULSEAUDIO pThis, const char *szMsg)
 {
     AssertPtrReturn(pThis, VERR_INVALID_POINTER);
@@ -920 +978 @@
     if (pThis->cLogErrors++ < VBOX_PULSEAUDIO_MAX_LOG_REL_ERRORS)
     {
-        int rc2 = pa_context_errno(g_pContext);
+        int rc2 = pa_context_errno(pThis->pContext);
         LogRel(("PulseAudio: %s: %s\n", szMsg, pa_strerror(rc2)));
     }
@@ -928 +986 @@
 }
 
+static DECLCALLBACK(void) paEnumSinkCb(pa_context *pCtx, const pa_sink_info *pInfo, int eol, void *pvUserData)
+{
+    if (eol != 0)
+        return;
+
+    AssertPtrReturnVoid(pCtx);
+    AssertPtrReturnVoid(pInfo);
+
+    PPULSEAUDIOENUMCBCTX pCbCtx = (PPULSEAUDIOENUMCBCTX)pvUserData;
+    AssertPtrReturnVoid(pCbCtx);
+    AssertPtrReturnVoid(pCbCtx->pDrv);
+
+    LogRel2(("PulseAudio: Using output sink '%s'\n", pInfo->name));
+
+    /** @todo Store sinks + channel mapping in callback context as soon as we have surround support. */
+    pCbCtx->cDevOut++;
+
+    pa_threaded_mainloop_signal(pCbCtx->pDrv->pMainLoop, 0);
+}
+
+static DECLCALLBACK(void) paEnumSourceCb(pa_context *pCtx, const pa_source_info *pInfo, int eol, void *pvUserData)
+{
+    if (eol != 0)
+        return;
+
+    AssertPtrReturnVoid(pCtx);
+    AssertPtrReturnVoid(pInfo);
+
+    PPULSEAUDIOENUMCBCTX pCbCtx = (PPULSEAUDIOENUMCBCTX)pvUserData;
+    AssertPtrReturnVoid(pCbCtx);
+    AssertPtrReturnVoid(pCbCtx->pDrv);
+
+    LogRel2(("PulseAudio: Using input source '%s'\n", pInfo->name));
+
+    /** @todo Store sources + channel mapping in callback context as soon as we have surround support. */
+    pCbCtx->cDevIn++;
+
+    pa_threaded_mainloop_signal(pCbCtx->pDrv->pMainLoop, 0);
+}
+
+static DECLCALLBACK(void) paEnumServerCb(pa_context *pCtx, const pa_server_info *pInfo, void *pvUserData)
+{
+    AssertPtrReturnVoid(pCtx);
+    AssertPtrReturnVoid(pInfo);
+
+    PPULSEAUDIOENUMCBCTX pCbCtx = (PPULSEAUDIOENUMCBCTX)pvUserData;
+    AssertPtrReturnVoid(pCbCtx);
+
+    PDRVHOSTPULSEAUDIO pThis    = pCbCtx->pDrv;
+    AssertPtrReturnVoid(pThis);
+
+    if (pInfo->default_sink_name)
+    {
+        Assert(RTStrIsValidEncoding(pInfo->default_sink_name));
+        pCbCtx->pszDefaultSink   = RTStrDup(pInfo->default_sink_name);
+    }
+
+    if (pInfo->default_sink_name)
+    {
+        Assert(RTStrIsValidEncoding(pInfo->default_source_name));
+        pCbCtx->pszDefaultSource = RTStrDup(pInfo->default_source_name);
+    }
+
+    pa_threaded_mainloop_signal(pThis->pMainLoop, 0);
+}
+
+static int paEnumerate(PDRVHOSTPULSEAUDIO pThis, PPDMAUDIOBACKENDCFG pCfg, uint32_t fEnum)
+{
+    AssertPtrReturn(pThis, VERR_INVALID_POINTER);
+    AssertPtrReturn(pCfg,  VERR_INVALID_POINTER);
+
+    PDMAUDIOBACKENDCFG Cfg;
+    RT_ZERO(Cfg);
+
+    Cfg.cbStreamOut    = sizeof(PULSEAUDIOSTREAM);
+    Cfg.cbStreamIn     = sizeof(PULSEAUDIOSTREAM);
+    Cfg.cMaxStreamsOut = UINT32_MAX;
+    Cfg.cMaxStreamsIn  = UINT32_MAX;
+
+    PULSEAUDIOENUMCBCTX cbCtx;
+    RT_ZERO(cbCtx);
+
+    cbCtx.pDrv   = pThis;
+    cbCtx.fFlags = fEnum;
+
+    bool fLog = (fEnum & PULSEAUDIOENUMCBFLAGS_LOG);
+
+    int rc = paWaitFor(pThis, pa_context_get_server_info(pThis->pContext, paEnumServerCb, &cbCtx));
+    if (RT_SUCCESS(rc))
+    {
+        if (cbCtx.pszDefaultSink)
+        {
+            if (fLog)
+                LogRel2(("PulseAudio: Default output sink is '%s'\n", cbCtx.pszDefaultSink));
+
+            rc = paWaitFor(pThis, pa_context_get_sink_info_by_name(pThis->pContext, cbCtx.pszDefaultSink,
+                                                                   paEnumSinkCb, &cbCtx));
+            if (   RT_FAILURE(rc)
+                && fLog)
+            {
+                LogRel(("PulseAudio: Error enumerating properties for default output sink '%s'\n", cbCtx.pszDefaultSink));
+            }
+        }
+        else if (fLog)
+            LogRel2(("PulseAudio: No default output sink found\n"));
+
+        if (RT_SUCCESS(rc))
+        {
+            if (cbCtx.pszDefaultSource)
+            {
+                if (fLog)
+                    LogRel2(("PulseAudio: Default input source is '%s'\n", cbCtx.pszDefaultSource));
+
+                rc = paWaitFor(pThis, pa_context_get_source_info_by_name(pThis->pContext, cbCtx.pszDefaultSource,
+                                                                         paEnumSourceCb, &cbCtx));
+                if (   RT_FAILURE(rc)
+                    && fLog)
+                {
+                    LogRel(("PulseAudio: Error enumerating properties for default input source '%s'\n", cbCtx.pszDefaultSource));
+                }
+            }
+            else if (fLog)
+                LogRel2(("PulseAudio: No default input source found\n"));
+        }
+
+        if (RT_SUCCESS(rc))
+        {
+            Cfg.cSinks   = cbCtx.cDevOut;
+            Cfg.cSources = cbCtx.cDevIn;
+
+            if (fLog)
+            {
+                LogRel2(("PulseAudio: Found %RU8 host playback device(s)\n",  cbCtx.cDevOut));
+                LogRel2(("PulseAudio: Found %RU8 host capturing device(s)\n", cbCtx.cDevIn));
+            }
+
+            if (pCfg)
+                memcpy(pCfg, &Cfg, sizeof(PDMAUDIOBACKENDCFG));
+        }
+
+        if (cbCtx.pszDefaultSink)
+        {
+            RTStrFree(cbCtx.pszDefaultSink);
+            cbCtx.pszDefaultSink = NULL;
+        }
+
+        if (cbCtx.pszDefaultSource)
+        {
+            RTStrFree(cbCtx.pszDefaultSource);
+            cbCtx.pszDefaultSource = NULL;
+        }
+    }
+    else if (fLog)
+        LogRel(("PulseAudio: Error enumerating PulseAudio server properties\n"));
+
+    LogFlowFuncLeaveRC(rc);
+    return rc;
+}
+
 static DECLCALLBACK(int) drvHostPulseAudioFiniIn(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMIN pHstStrmIn)
 {
-    NOREF(pInterface);
+    AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
     AssertPtrReturn(pHstStrmIn, VERR_INVALID_POINTER);
 
+    PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface);
+    PPULSEAUDIOSTREAM  pStrm = (PPULSEAUDIOSTREAM)pHstStrmIn;
+
     LogFlowFuncEnter();
 
-    PPULSEAUDIOSTREAM pThisStrmIn = (PPULSEAUDIOSTREAM)pHstStrmIn;
-    if (pThisStrmIn->pStream)
-    {
-        pa_threaded_mainloop_lock(g_pMainLoop);
-        pa_stream_disconnect(pThisStrmIn->pStream);
-        pa_stream_unref(pThisStrmIn->pStream);
-        pa_threaded_mainloop_unlock(g_pMainLoop);
-
-        pThisStrmIn->pStream = NULL;
+    if (pStrm->pStream)
+    {
+        pa_threaded_mainloop_lock(pThis->pMainLoop);
+        pa_stream_disconnect(pStrm->pStream);
+        pa_stream_unref(pStrm->pStream);
+        pa_threaded_mainloop_unlock(pThis->pMainLoop);
+
+        pStrm->pStream = NULL;
     }
 
@@ -951 +1170 @@
 static DECLCALLBACK(int) drvHostPulseAudioFiniOut(PPDMIHOSTAUDIO pInterface, PPDMAUDIOHSTSTRMOUT pHstStrmOut)
 {
-    NOREF(pInterface);
+    AssertPtrReturn(pInterface,  VERR_INVALID_POINTER);
     AssertPtrReturn(pHstStrmOut, VERR_INVALID_POINTER);
 
+    PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface);
+    PPULSEAUDIOSTREAM  pStrm = (PPULSEAUDIOSTREAM)pHstStrmOut;
+
     LogFlowFuncEnter();
 
-    PPULSEAUDIOSTREAM pThisStrmOut = (PPULSEAUDIOSTREAM)pHstStrmOut;
-    if (pThisStrmOut->pStream)
-    {
-        pa_threaded_mainloop_lock(g_pMainLoop);
-        pa_stream_disconnect(pThisStrmOut->pStream);
-        pa_stream_unref(pThisStrmOut->pStream);
-        pa_threaded_mainloop_unlock(g_pMainLoop);
-
-        pThisStrmOut->pStream = NULL;
-    }
-
-    if (pThisStrmOut->pvPCMBuf)
-    {
-        RTMemFree(pThisStrmOut->pvPCMBuf);
-        pThisStrmOut->pvPCMBuf = NULL;
-
-        pThisStrmOut->cbPCMBuf = 0;
+    if (pStrm->pStream)
+    {
+        pa_threaded_mainloop_lock(pThis->pMainLoop);
+        pa_stream_disconnect(pStrm->pStream);
+        pa_stream_unref(pStrm->pStream);
+        pa_threaded_mainloop_unlock(pThis->pMainLoop);
+
+        pStrm->pStream = NULL;
+    }
+
+    if (pStrm->pvPCMBuf)
+    {
+        RTMemFree(pStrm->pvPCMBuf);
+        pStrm->pvPCMBuf = NULL;
+        pStrm->cbPCMBuf = 0;
     }
 
@@ -981 +1201 @@
                                                      PPDMAUDIOHSTSTRMOUT pHstStrmOut, PDMAUDIOSTREAMCMD enmStreamCmd)
 {
-    NOREF(pInterface);
+    AssertPtrReturn(pInterface , VERR_INVALID_POINTER);
     AssertPtrReturn(pHstStrmOut, VERR_INVALID_POINTER);
 
-    PPULSEAUDIOSTREAM pThisStrmOut = (PPULSEAUDIOSTREAM)pHstStrmOut;
+    PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface);
+    PPULSEAUDIOSTREAM  pStrm = (PPULSEAUDIOSTREAM)pHstStrmOut;
+
     int rc = VINF_SUCCESS;
 
@@ -994 +1216 @@
         case PDMAUDIOSTREAMCMD_RESUME:
         {
-            pa_threaded_mainloop_lock(g_pMainLoop);
-
-            if (   pThisStrmOut->pDrainOp
-                && pa_operation_get_state(pThisStrmOut->pDrainOp) != PA_OPERATION_DONE)
+            pa_threaded_mainloop_lock(pThis->pMainLoop);
+
+            if (   pStrm->pDrainOp
+                && pa_operation_get_state(pStrm->pDrainOp) != PA_OPERATION_DONE)
             {
-                pa_operation_cancel(pThisStrmOut->pDrainOp);
-                pa_operation_unref(pThisStrmOut->pDrainOp);
-
-                pThisStrmOut->pDrainOp = NULL;
+                pa_operation_cancel(pStrm->pDrainOp);
+                pa_operation_unref(pStrm->pDrainOp);
+
+                pStrm->pDrainOp = NULL;
             }
             else
             {
-                /* This should return immediately. */
-                rc = drvHostPulseAudioWaitFor(pa_stream_cork(pThisStrmOut->pStream, 0,
-                                                             drvHostPulseAudioCbSuccess, pThisStrmOut),
-                                              15 * 1000 /* 15s timeout */);
+                rc = paWaitFor(pThis, pa_stream_cork(pStrm->pStream, 0, paStreamCbSuccess, pStrm));
             }
 
-            pa_threaded_mainloop_unlock(g_pMainLoop);
+            pa_threaded_mainloop_unlock(pThis->pMainLoop);
             break;
         }
@@ -1021 +1240 @@
             /* Pause audio output (the Pause bit of the AC97 x_CR register is set).
              * Note that we must return immediately from here! */
-            pa_threaded_mainloop_lock(g_pMainLoop);
-            if (!pThisStrmOut->pDrainOp)
+            pa_threaded_mainloop_lock(pThis->pMainLoop);
+            if (!pStrm->pDrainOp)
             {
-                /* This should return immediately. */
-                rc = drvHostPulseAudioWaitFor(pa_stream_trigger(pThisStrmOut->pStream,
-                                                                drvHostPulseAudioCbSuccess, pThisStrmOut),
-                                              15 * 1000 /* 15s timeout */);
+                rc = paWaitFor(pThis, pa_stream_trigger(pStrm->pStream, paStreamCbSuccess, pStrm));
                 if (RT_LIKELY(RT_SUCCESS(rc)))
-                    pThisStrmOut->pDrainOp = pa_stream_drain(pThisStrmOut->pStream,
-                                                             drvHostPulseAudioCbStreamDrain, pThisStrmOut);
+                    pStrm->pDrainOp = pa_stream_drain(pStrm->pStream, paStreamCbDrain, pStrm);
             }
-            pa_threaded_mainloop_unlock(g_pMainLoop);
+            pa_threaded_mainloop_unlock(pThis->pMainLoop);
             break;
         }
@@ -1049 +1264 @@
                                                     PDMAUDIOSTREAMCMD enmStreamCmd)
 {
-    NOREF(pInterface);
+    AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
     AssertPtrReturn(pHstStrmIn, VERR_INVALID_POINTER);
 
-    PPULSEAUDIOSTREAM pThisStrmIn = (PPULSEAUDIOSTREAM)pHstStrmIn;
+    PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface);
+    PPULSEAUDIOSTREAM  pStrm = (PPULSEAUDIOSTREAM)pHstStrmIn;
+
     int rc = VINF_SUCCESS;
 
@@ -1062 +1279 @@
         case PDMAUDIOSTREAMCMD_RESUME:
         {
-            pa_threaded_mainloop_lock(g_pMainLoop);
-            /* This should return immediately. */
-            rc = drvHostPulseAudioWaitFor(pa_stream_cork(pThisStrmIn->pStream, 0 /* Play / resume */,
-                                                         drvHostPulseAudioCbSuccess, pThisStrmIn),
-                                          15 * 1000 /* 15s timeout */);
-            pa_threaded_mainloop_unlock(g_pMainLoop);
+            pa_threaded_mainloop_lock(pThis->pMainLoop);
+            rc = paWaitFor(pThis, pa_stream_cork(pStrm->pStream, 0 /* Play / resume */, paStreamCbSuccess, pStrm));
+            pa_threaded_mainloop_unlock(pThis->pMainLoop);
             break;
         }
@@ -1074 +1288 @@
         case PDMAUDIOSTREAMCMD_PAUSE:
         {
-            pa_threaded_mainloop_lock(g_pMainLoop);
-            if (pThisStrmIn->pu8PeekBuf) /* Do we need to drop the peek buffer?*/
+            pa_threaded_mainloop_lock(pThis->pMainLoop);
+            if (pStrm->pu8PeekBuf) /* Do we need to drop the peek buffer?*/
             {
-                pa_stream_drop(pThisStrmIn->pStream);
-                pThisStrmIn->pu8PeekBuf = NULL;
+                pa_stream_drop(pStrm->pStream);
+                pStrm->pu8PeekBuf = NULL;
             }
-            /* This should return immediately. */
-            rc = drvHostPulseAudioWaitFor(pa_stream_cork(pThisStrmIn->pStream, 1 /* Stop / pause */,
-                                                         drvHostPulseAudioCbSuccess, pThisStrmIn),
-                                          15 * 1000 /* 15s timeout */);
-            pa_threaded_mainloop_unlock(g_pMainLoop);
+
+            rc = paWaitFor(pThis, pa_stream_cork(pStrm->pStream, 1 /* Stop / pause */, paStreamCbSuccess, pStrm));
+            pa_threaded_mainloop_unlock(pThis->pMainLoop);
             break;
         }
@@ -1099 +1311 @@
 static DECLCALLBACK(int) drvHostPulseAudioGetConf(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDCFG pCfg)
 {
-    NOREF(pInterface);
-    AssertPtrReturn(pCfg, VERR_INVALID_POINTER);
-
-    pCfg->cbStreamOut     = sizeof(PULSEAUDIOSTREAM);
-    pCfg->cbStreamIn      = sizeof(PULSEAUDIOSTREAM);
-    pCfg->cMaxHstStrmsOut = UINT32_MAX;
-    pCfg->cMaxHstStrmsIn  = UINT32_MAX;
-
-    return VINF_SUCCESS;
+    AssertPtrReturn(pInterface, VERR_INVALID_POINTER);
+    AssertPtrReturn(pCfg,       VERR_INVALID_POINTER);
+
+    PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface);
+
+    return paEnumerate(pThis, pCfg, PULSEAUDIOENUMCBFLAGS_LOG /* fEnum */);
 }
 
 static DECLCALLBACK(void) drvHostPulseAudioShutdown(PPDMIHOSTAUDIO pInterface)
 {
-    NOREF(pInterface);
+    AssertPtrReturnVoid(pInterface);
+
+    PDRVHOSTPULSEAUDIO pThis = PDMIHOSTAUDIO_2_DRVHOSTPULSEAUDIO(pInterface);
 
     LogFlowFuncEnter();
 
-    if (g_pMainLoop)
-        pa_threaded_mainloop_stop(g_pMainLoop);
-
-    if (g_pContext)
-    {
-        pa_context_disconnect(g_pContext);
-        pa_context_unref(g_pContext);
-        g_pContext = NULL;
-    }
-
-    if (g_pMainLoop)
-    {
-        pa_threaded_mainloop_free(g_pMainLoop);
-        g_pMainLoop = NULL;
+    if (pThis->pMainLoop)
+        pa_threaded_mainloop_stop(pThis->pMainLoop);
+
+    if (pThis->pContext)
+    {
+        pa_context_disconnect(pThis->pContext);
+        pa_context_unref(pThis->pContext);
+        pThis->pContext = NULL;
+    }
+
+    if (pThis->pMainLoop)
+    {
+        pa_threaded_mainloop_free(pThis->pMainLoop);
+        pThis->pMainLoop = NULL;
     }
 

trunk/src/VBox/Devices/Audio/alsa_mangling.h

@@ -6 +6 @@
 
 /*
- * Copyright (C) 2013-2015 Oracle Corporation
+ * Copyright (C) 2013-2016 Oracle Corporation
  *
  * This file is part of VirtualBox Open Source Edition (OSE), as
@@ -21 +21 @@
 
 #define ALSA_MANGLER(symbol) VBox_##symbol
+
+#define snd_device_name_hint                    ALSA_MANGLER(snd_device_name_hint)
+#define snd_device_name_get_hint                ALSA_MANGLER(snd_device_name_get_hint)
+#define snd_device_name_free_hint               ALSA_MANGLER(snd_device_name_free_hint)
 
 #define snd_pcm_hw_params_any                   ALSA_MANGLER(snd_pcm_hw_params_any)

trunk/src/VBox/Devices/Audio/alsa_stubs.c

@@ -5 +5 @@
 
 /*
- * Copyright (C) 2006-2015 Oracle Corporation
+ * Copyright (C) 2006-2016 Oracle Corporation
  *
  * This file is part of VirtualBox Open Source Edition (OSE), as
@@ -35 +35 @@
         return pfn_ ## function shortsig; \
     }
+
+PROXY_STUB(snd_device_name_hint, int,
+           (int card, const char *iface, void ***hints),
+           (card, iface, hints))
+PROXY_STUB(snd_device_name_free_hint, int,
+           (void ***hints),
+           (hints))
+PROXY_STUB(snd_device_name_get_hint, char *,
+           (const void *hint, const char *id),
+           (hint, id))
 
 PROXY_STUB(snd_pcm_hw_params_any, int,
@@ -118 +128 @@
 static SHARED_FUNC SharedFuncs[] =
 {
+    ELEMENT(snd_device_name_hint),
+    ELEMENT(snd_device_name_get_hint),
+    ELEMENT(snd_device_name_free_hint),
+
     ELEMENT(snd_pcm_hw_params_any),
     ELEMENT(snd_pcm_close),

trunk/src/VBox/Devices/Audio/pulse_mangling.h

@@ -6 +6 @@
 
 /*
- * Copyright (C) 2013-2015 Oracle Corporation
+ * Copyright (C) 2013-2016 Oracle Corporation
  *
  * This file is part of VirtualBox Open Source Edition (OSE), as
@@ -42 +42 @@
 #define pa_context_connect                      PULSE_MANGLER(pa_context_connect)
 #define pa_context_disconnect                   PULSE_MANGLER(pa_context_disconnect)
+#define pa_context_get_server_info              PULSE_MANGLER(pa_context_get_server_info)
+#define pa_context_get_sink_info_by_name        PULSE_MANGLER(pa_context_get_sink_info_by_name)
+#define pa_context_get_source_info_by_name      PULSE_MANGLER(pa_context_get_source_info_by_name)
 #define pa_context_get_state                    PULSE_MANGLER(pa_context_get_state)
 #define pa_context_unref                        PULSE_MANGLER(pa_context_unref)

trunk/src/VBox/Devices/Audio/pulse_stubs.c

@@ -5 +5 @@
 
 /*
- * Copyright (C) 2006-2015 Oracle Corporation
+ * Copyright (C) 2006-2016 Oracle Corporation
  *
  * This file is part of VirtualBox Open Source Edition (OSE), as
@@ -111 +111 @@
                 (p))
 PROXY_STUB     (pa_context_connect, int,
-                (pa_context *c, const char *server, pa_context_flags_t flags,
-                 const pa_spawn_api *api),
+                (pa_context *c, const char *server, pa_context_flags_t flags, const pa_spawn_api *api),
                 (c, server, flags, api))
 PROXY_STUB_VOID(pa_context_disconnect,
                 (pa_context *c),
                 (c))
+PROXY_STUB     (pa_context_get_server_info, pa_operation*,
+                (pa_context *c, const char *name, pa_server_info_cb_t cb, void *userdata),
+                (c, name, cb, userdata))
+PROXY_STUB     (pa_context_get_sink_info_by_name, pa_operation*,
+                (pa_context *c, const char *name, pa_sink_info_cb_t cb, void *userdata),
+                (c, name, cb, userdata))
+PROXY_STUB     (pa_context_get_source_info_by_name, pa_operation*,
+                (pa_context *c, const char *name, pa_source_info_cb_t cb, void *userdata),
+                (c, name, cb, userdata))
 PROXY_STUB     (pa_context_get_state, pa_context_state_t,
                 (pa_context *c),
@@ -220 +228 @@
     ELEMENT(pa_context_connect),
     ELEMENT(pa_context_disconnect),
+    ELEMENT(pa_context_get_server_info),
+    ELEMENT(pa_context_get_sink_info_by_name),
+    ELEMENT(pa_context_get_source_info_by_name),
     ELEMENT(pa_context_get_state),
     ELEMENT(pa_context_unref),
@@ -285 +296 @@
     return rc;
 }
-