Changeset 88269 in vbox for trunk/src/VBox/Devices/Audio

Timestamp:

Mar 24, 2021 11:45:54 AM (4 years ago)

Author:

vboxsync

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

143474

Message:

Audio: Made sure PDMAUDIOPCMPROPS is initialized using a helper function or the initializer macro, and that vital changes are made using setting helper functions. There are now two derived fields (frame size and shift count) that must be maintained, so this was the sanest way of doing it. Added a raw flag to PDMAUDIOPCMPROPS for VRDE/VRDP, since it wants the raw mixer content and we need a way of expressing this (PDMAUDIOSTREAMLAYOUT isn't the right place). The mixer buffers now uses PDMAUDIOPCMPROPS rather than the weird 32-bit format contraption for picking conversion functions. Simplify the drvAudioStreamPlay code by eliminating the PDMAUDIOSTREAMLAYOUT_RAW special case. bugref:9890

Location:

trunk/src/VBox/Devices/Audio

Files:

• AudioHlp.cpp (modified) (7 diffs)
• AudioMixBuffer.cpp (modified) (26 diffs)
• AudioMixBuffer.h (modified) (5 diffs)
• AudioMixer.cpp (modified) (3 diffs)
• DevHda.cpp (modified) (10 diffs)
• DevHdaCommon.cpp (modified) (5 diffs)
• DevHdaStream.cpp (modified) (1 diff)
• DevHdaStreamMap.cpp (modified) (9 diffs)
• DevIchAc97.cpp (modified) (4 diffs)
• DevSB16.cpp (modified) (2 diffs)
• DrvAudio.cpp (modified) (16 diffs)
• DrvHostAudioAlsa.cpp (modified) (8 diffs)
• DrvHostAudioCoreAudio.cpp (modified) (4 diffs)
• DrvHostAudioDSound.cpp (modified) (5 diffs)
• DrvHostAudioDebug.cpp (modified) (4 diffs)
• DrvHostAudioOss.cpp (modified) (4 diffs)
• DrvHostAudioPulseAudio.cpp (modified) (10 diffs)
• testcase/tstAudioMixBuffer.cpp (modified) (13 diffs)

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

@@ -179 +179 @@
 bool AudioHlpStreamCfgIsValid(PCPDMAUDIOSTREAMCFG pCfg)
 {
-    AssertPtrReturn(pCfg, false);
-
-    AssertReturn(PDMAudioStrmCfgIsValid(pCfg), false);
-
-    bool fValid = (   pCfg->enmDir == PDMAUDIODIR_IN
-                   || pCfg->enmDir == PDMAUDIODIR_OUT);
-
-    fValid &= (   pCfg->enmLayout == PDMAUDIOSTREAMLAYOUT_NON_INTERLEAVED
-               || pCfg->enmLayout == PDMAUDIOSTREAMLAYOUT_RAW);
-
-    if (fValid)
-        fValid = AudioHlpPcmPropsAreValid(&pCfg->Props);
-
-    return fValid;
+    /* Ugly! HDA attach code calls us with uninitialized (all zero) config. */
+    if (   pCfg->enmLayout != PDMAUDIOSTREAMLAYOUT_INVALID
+        || PDMAudioPropsHz(&pCfg->Props) != 0)
+    {
+        if (PDMAudioStrmCfgIsValid(pCfg))
+        {
+            if (   pCfg->enmDir == PDMAUDIODIR_IN
+                || pCfg->enmDir == PDMAUDIODIR_OUT)
+            {
+                if (   pCfg->enmLayout == PDMAUDIOSTREAMLAYOUT_NON_INTERLEAVED
+                    || pCfg->enmLayout == PDMAUDIOSTREAMLAYOUT_RAW)
+                    return AudioHlpPcmPropsAreValid(&pCfg->Props);
+            }
+        }
+    }
+    return false;
 }
 
@@ -219 +221 @@
  *        function will flag such properties as not valid.
  *
- * @todo  r=bird: See note and explain properly.
+ * @todo  r=bird: See note and explain properly. Perhaps rename to
+ *        AudioHlpPcmPropsAreValidAndSupported?
  *
  * @returns @c true if the properties are valid, @c false if not.
@@ -227 +230 @@
 {
     AssertPtrReturn(pProps, false);
-
     AssertReturn(PDMAudioPropsAreValid(pProps), false);
 
-    /** @todo r=bird: This code is cannot make up its mind whether to return on
-     *        false, or whether to return at the end. (hint: just return
-     *        immediately, duh.) */
-
     /* Minimum 1 channel (mono), maximum 7.1 (= 8) channels. */
-    bool fValid = (   pProps->cChannels >= 1
-                   && pProps->cChannels <= 8);
-
-    if (fValid)
-    {
-        switch (pProps->cbSample)
+    if (PDMAudioPropsChannels(pProps) >= 1 && PDMAudioPropsChannels(pProps) <= 8)
+    {
+        switch (PDMAudioPropsSampleSize(pProps))
         {
             case 1: /* 8 bit */
-               if (pProps->fSigned)
-                   fValid = false;
+               if (PDMAudioPropsIsSigned(pProps))
+                   return false;
                break;
             case 2: /* 16 bit */
-                if (!pProps->fSigned)
-                    fValid = false;
+                if (!PDMAudioPropsIsSigned(pProps))
+                    return false;
                 break;
             /** @todo Do we need support for 24 bit samples? */
             case 4: /* 32 bit */
-                if (!pProps->fSigned)
-                    fValid = false;
+                if (!PDMAudioPropsIsSigned(pProps))
+                    return false;
+                break;
+            case 8: /* 64-bit raw */
+                if (   !PDMAudioPropsIsSigned(pProps)
+                    || !pProps->fRaw)
+                    return false;
                 break;
             default:
-                fValid = false;
-                break;
-        }
-    }
-
-    if (!fValid)
-        return false;
-
-    fValid &= pProps->uHz > 0;
-    fValid &= pProps->cShift == PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(pProps->cbSample, pProps->cChannels);
-    fValid &= pProps->fSwapEndian == false; /** @todo Handling Big Endian audio data is not supported yet. */
-
-    return fValid;
+                return false;
+        }
+
+        if (pProps->uHz > 0)
+        {
+            if (!pProps->fSwapEndian) /** @todo Handling Big Endian audio data is not supported yet. */
+                return true;
+        }
+    }
+    return false;
 }
 
@@ -520 +517 @@
     /** @todo Validate fOpen flags. */
     AssertPtrReturn(pProps,  VERR_INVALID_POINTER);
+    Assert(PDMAudioPropsAreValid(pProps));
 
     int rc;
@@ -529 +527 @@
     else if (pFile->enmType == PDMAUDIOFILETYPE_WAV)
     {
-        Assert(pProps->cChannels);
-        Assert(pProps->uHz);
-        Assert(pProps->cbSample);
-
         pFile->pvData = (PAUDIOWAVFILEDATA)RTMemAllocZ(sizeof(AUDIOWAVFILEDATA));
         if (pFile->pvData)
@@ -549 +543 @@
             pData->Hdr.u32Size1         = 16; /* Means PCM. */
             pData->Hdr.u16AudioFormat   = 1;  /* PCM, linear quantization. */
-            pData->Hdr.u16NumChannels   = pProps->cChannels;
+            pData->Hdr.u16NumChannels   = PDMAudioPropsChannels(pProps);
             pData->Hdr.u32SampleRate    = pProps->uHz;
             pData->Hdr.u32ByteRate      = PDMAudioPropsGetBitrate(pProps) / 8;
-            pData->Hdr.u16BlockAlign    = pProps->cChannels * pProps->cbSample;
-            pData->Hdr.u16BitsPerSample = pProps->cbSample * 8;
+            pData->Hdr.u16BlockAlign    = PDMAudioPropsFrameSize(pProps);
+            pData->Hdr.u16BitsPerSample = PDMAudioPropsSampleBits(pProps);
 
             /* Data chunk. */
@@ -584 +578 @@
 
     if (RT_SUCCESS(rc))
-    {
         LogRel2(("Audio: Opened file '%s'\n", pFile->szName));
-    }
     else
         LogRel(("Audio: Failed opening file '%s', rc=%Rrc\n", pFile->szName, rc));

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

@@ -47 +47 @@
 #endif
 #include <VBox/err.h>
+#include <VBox/vmm/pdmaudioinline.h>
 
 #include "AudioMixBuffer.h"
@@ -53 +54 @@
 # ifdef DEBUG
 #  define AUDMIXBUF_LOG(x) LogFlowFunc(x)
+#  define AUDMIXBUF_LOG_ENABLED
 # else
-# define AUDMIXBUF_LOG(x) do {} while (0)
+#  define AUDMIXBUF_LOG(x) do {} while (0)
 # endif
 #else /* VBOX_AUDIO_TESTCASE */
 # define AUDMIXBUF_LOG(x) RTPrintf x
+# define AUDMIXBUF_LOG_ENABLED
 #endif
 
@@ -136 +139 @@
 
 /**
- * Peeks for audio frames without any conversion done.
- * This will get the raw frame data out of a mixing buffer.
- *
- * @return  IPRT status code or VINF_AUDIO_MORE_DATA_AVAILABLE if more data is available to read.
- *
- * @param   pMixBuf                 Mixing buffer to acquire audio frames from.
- * @param   cFramesToRead           Number of audio frames to read.
- * @param   paFrameBuf              Buffer where to store the returned audio frames.
- * @param   cFrameBuf               Size (in frames) of the buffer to store audio frames into.
- * @param   pcFramesRead            Returns number of read audio frames. Optional.
- *
- * @remark  This function is not thread safe!
- */
-/** @todo r=bird: This isn't a 'ing Peek function, it's a Read function!
- *        Aaaaaaaaaaaaaaaaaaaaaaaaaaaaaarg!!!!!!!!!!!!!!!!!!! */
-int AudioMixBufPeek(PPDMAUDIOMIXBUF pMixBuf, uint32_t cFramesToRead,
-                    PPDMAUDIOFRAME paFrameBuf, uint32_t cFrameBuf, uint32_t *pcFramesRead)
-{
-    AssertPtrReturn(pMixBuf,    VERR_INVALID_POINTER);
-    AssertPtrReturn(paFrameBuf, VERR_INVALID_POINTER);
-    AssertReturn(cFrameBuf,     VERR_INVALID_PARAMETER);
-    /* pcRead is optional. */
-
-    int rc;
-
-    if (!cFramesToRead)
-    {
-        if (pcFramesRead)
-            *pcFramesRead = 0;
-        return VINF_SUCCESS;
-    }
-
-    uint32_t cRead;
-    if (pMixBuf->offRead + cFramesToRead > pMixBuf->cFrames)
-    {
-        cRead = pMixBuf->cFrames - pMixBuf->offRead;
-        rc = VINF_AUDIO_MORE_DATA_AVAILABLE;
-    }
-    else
-    {
-        cRead = cFramesToRead;
-        rc = VINF_SUCCESS;
-    }
-
-    if (cRead > cFrameBuf)
-    {
-        cRead = cFrameBuf;
-        rc = VINF_AUDIO_MORE_DATA_AVAILABLE;
-    }
-
-    if (cRead)
-    {
-        memcpy(paFrameBuf, &pMixBuf->pFrames[pMixBuf->offRead], sizeof(PDMAUDIOFRAME) * cRead);
-
-        pMixBuf->offRead = (pMixBuf->offRead + cRead) % pMixBuf->cFrames;
-        Assert(pMixBuf->offRead <= pMixBuf->cFrames);
-        pMixBuf->cUsed  -= RT_MIN(cRead, pMixBuf->cUsed);
-    }
-
-    if (pcFramesRead)
-        *pcFramesRead = cRead;
-
-    return rc;
-}
-
-/**
  * Returns a mutable pointer to the mixing buffer's audio frame buffer for writing raw
  * audio frames.
@@ -213 +150 @@
  * @remark  This function is not thread safe!
  */
+/** @todo r=bird: This isn't a 'ing Peek function, it's a Read function!
+ *        Aaaaaaaaaaaaaaaaaaaaaaaaaaaaaarg!!!!!!!!!!!!!!!!!!! */
 int AudioMixBufPeekMutable(PPDMAUDIOMIXBUF pMixBuf, uint32_t cFrames,
                            PPDMAUDIOFRAME *ppvFrames, uint32_t *pcFramesToWrite)
@@ -281 +220 @@
     AssertStmt(cFramesToClear <= pMixBuf->cFrames, cFramesToClear = pMixBuf->cFrames);
 
+/** @todo r=bird: Why isn't this done when reading/releaseing ? */
     PPDMAUDIOMIXBUF pIter;
     RTListForEach(&pMixBuf->lstChildren, pIter, PDMAUDIOMIXBUF, Node)
@@ -291 +231 @@
     }
 
+/** @todo r=bird: waste of time? */
     uint32_t cClearOff;
     uint32_t cClearLen;
@@ -559 +500 @@
 
 #undef AUDMIXBUF_CONVERT
+
+/*
+ * Manually coded signed 64-bit conversion.
+ */
+#if 0
+DECLCALLBACK(uint32_t) audioMixBufConvFromS64Stereo(PPDMAUDIOFRAME paDst, const void *pvSrc, uint32_t cbSrc,
+                                                    PCPDMAUDMIXBUFCONVOPTS pOpts)
+{
+    _aType const *pSrc = (_aType const *)pvSrc;
+    uint32_t cFrames = RT_MIN(pOpts->cFrames, cbSrc / sizeof(_aType));
+    AUDMIXBUF_MACRO_LOG(("cFrames=%RU32, BpS=%zu, lVol=%RU32, rVol=%RU32\n",
+                         pOpts->cFrames, sizeof(_aType), pOpts->From.Volume.uLeft, pOpts->From.Volume.uRight));
+    for (uint32_t i = 0; i < cFrames; i++)
+    {
+        paDst->i64LSample = ASMMult2xS32RetS64((int32_t)audioMixBufClipFrom##_aName(*pSrc++), pOpts->From.Volume.uLeft ) >> AUDIOMIXBUF_VOL_SHIFT; \
+        paDst->i64RSample = ASMMult2xS32RetS64((int32_t)audioMixBufClipFrom##_aName(*pSrc++), pOpts->From.Volume.uRight) >> AUDIOMIXBUF_VOL_SHIFT; \
+        paDst++;
+    }
+
+    return cFrames;
+}
+#endif
+
+DECLCALLBACK(void) audioMixBufConvToRawS64Stereo(void *pvDst, PCPDMAUDIOFRAME paSrc, PCPDMAUDMIXBUFCONVOPTS pOpts)
+{
+    AssertCompile(sizeof(paSrc[0]) == sizeof(int64_t) * 2);
+    memcpy(pvDst, paSrc, sizeof(int64_t) * 2 * pOpts->cFrames);
+}
+
+
+
 
 #define AUDMIXBUF_MIXOP(_aName, _aOp) \
@@ -674 +646 @@
 
 /**
- * Looks up the matching conversion (macro) routine for converting
- * audio frames from a source format.
- *
- ** @todo Speed up the lookup by binding it to the actual stream state.
- *
- * @return  PAUDMIXBUF_FN_CONVFROM  Function pointer to conversion macro if found, NULL if not supported.
- * @param   enmFmt                  Audio format to lookup conversion macro for.
- */
-static PFNPDMAUDIOMIXBUFCONVFROM audioMixBufConvFromLookup(PDMAUDIOMIXBUFFMT enmFmt)
-{
-    if (AUDMIXBUF_FMT_SIGNED(enmFmt))
-    {
-        if (AUDMIXBUF_FMT_CHANNELS(enmFmt) == 2)
+ * Looks up the matching conversion function for converting audio frames from a
+ * source format.
+ *
+ * @returns Pointer to matching conversion function, NULL if not supported.
+ * @param   pProp       The audio format to find a "from" converter for.
+ */
+static PFNPDMAUDIOMIXBUFCONVFROM audioMixBufConvFromLookup(PCPDMAUDIOPCMPROPS pProps)
+{
+    if (PDMAudioPropsIsSigned(pProps))
+    {
+        switch (PDMAudioPropsChannels(pProps))
         {
-            switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
-            {
-                case 8:  return audioMixBufConvFromS8Stereo;
-                case 16: return audioMixBufConvFromS16Stereo;
-                case 32: return audioMixBufConvFromS32Stereo;
-                default: return NULL;
-            }
+            case 2:
+                switch (PDMAudioPropsSampleSize(pProps))
+                {
+                    case 1:  return audioMixBufConvFromS8Stereo;
+                    case 2:  return audioMixBufConvFromS16Stereo;
+                    case 4:  return audioMixBufConvFromS32Stereo;
+                    //case 8:  return pProps->fRaw ? audioMixBufConvToRawS64Stereo : NULL;
+                    default: return NULL;
+                }
+
+            case 1:
+                switch (PDMAudioPropsSampleSize(pProps))
+                {
+                    case 1:  return audioMixBufConvFromS8Mono;
+                    case 2:  return audioMixBufConvFromS16Mono;
+                    case 4:  return audioMixBufConvFromS32Mono;
+                    default: return NULL;
+                }
+            default:
+                return NULL;
         }
-        else
+    }
+    else /* Unsigned */
+    {
+        switch (PDMAudioPropsChannels(pProps))
         {
-            switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
-            {
-                case 8:  return audioMixBufConvFromS8Mono;
-                case 16: return audioMixBufConvFromS16Mono;
-                case 32: return audioMixBufConvFromS32Mono;
-                default: return NULL;
-            }
+            case 2:
+                switch (PDMAudioPropsSampleSize(pProps))
+                {
+                    case 1:  return audioMixBufConvFromU8Stereo;
+                    case 2:  return audioMixBufConvFromU16Stereo;
+                    case 4:  return audioMixBufConvFromU32Stereo;
+                    default: return NULL;
+                }
+
+            case 1:
+                switch (PDMAudioPropsSampleSize(pProps))
+                {
+                    case 1:  return audioMixBufConvFromU8Mono;
+                    case 2:  return audioMixBufConvFromU16Mono;
+                    case 4:  return audioMixBufConvFromU32Mono;
+                    default: return NULL;
+                }
+            default:
+                return NULL;
         }
     }
+    /* not reached */
+}
+
+/**
+ * Looks up the matching conversion function for converting audio frames to a
+ * destination format.
+ *
+ * @returns Pointer to matching conversion function, NULL if not supported.
+ * @param   pProp       The audio format to find a "to" converter for.
+ */
+static PFNPDMAUDIOMIXBUFCONVTO audioMixBufConvToLookup(PCPDMAUDIOPCMPROPS pProps)
+{
+    if (PDMAudioPropsIsSigned(pProps))
+    {
+        switch (PDMAudioPropsChannels(pProps))
+        {
+            case 2:
+                switch (PDMAudioPropsSampleSize(pProps))
+                {
+                    case 1:  return audioMixBufConvToS8Stereo;
+                    case 2:  return audioMixBufConvToS16Stereo;
+                    case 4:  return audioMixBufConvToS32Stereo;
+                    case 8:  return pProps->fRaw ? audioMixBufConvToRawS64Stereo : NULL;
+                    default: return NULL;
+                }
+
+            case 1:
+                switch (PDMAudioPropsSampleSize(pProps))
+                {
+                    case 1:  return audioMixBufConvToS8Mono;
+                    case 2:  return audioMixBufConvToS16Mono;
+                    case 4:  return audioMixBufConvToS32Mono;
+                    default: return NULL;
+                }
+            default:
+                return NULL;
+        }
+    }
     else /* Unsigned */
     {
-        if (AUDMIXBUF_FMT_CHANNELS(enmFmt) == 2)
+        switch (PDMAudioPropsChannels(pProps))
         {
-            switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
-            {
-                case 8:  return audioMixBufConvFromU8Stereo;
-                case 16: return audioMixBufConvFromU16Stereo;
-                case 32: return audioMixBufConvFromU32Stereo;
-                default: return NULL;
-            }
-        }
-        else
-        {
-            switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
-            {
-                case 8:  return audioMixBufConvFromU8Mono;
-                case 16: return audioMixBufConvFromU16Mono;
-                case 32: return audioMixBufConvFromU32Mono;
-                default: return NULL;
-            }
-        }
-    }
-    /* not reached */
-}
-
-/**
- * Looks up the matching conversion (macro) routine for converting
- * audio frames to a destination format.
- *
- ** @todo Speed up the lookup by binding it to the actual stream state.
- *
- * @return  PAUDMIXBUF_FN_CONVTO    Function pointer to conversion macro if found, NULL if not supported.
- * @param   enmFmt                  Audio format to lookup conversion macro for.
- */
-static PFNPDMAUDIOMIXBUFCONVTO audioMixBufConvToLookup(PDMAUDIOMIXBUFFMT enmFmt)
-{
-    if (AUDMIXBUF_FMT_SIGNED(enmFmt))
-    {
-        if (AUDMIXBUF_FMT_CHANNELS(enmFmt) == 2)
-        {
-            switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
-            {
-                case 8:  return audioMixBufConvToS8Stereo;
-                case 16: return audioMixBufConvToS16Stereo;
-                case 32: return audioMixBufConvToS32Stereo;
-                default: return NULL;
-            }
-        }
-        else
-        {
-            switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
-            {
-                case 8:  return audioMixBufConvToS8Mono;
-                case 16: return audioMixBufConvToS16Mono;
-                case 32: return audioMixBufConvToS32Mono;
-                default: return NULL;
-            }
-        }
-    }
-    else /* Unsigned */
-    {
-        if (AUDMIXBUF_FMT_CHANNELS(enmFmt) == 2)
-        {
-            switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
-            {
-                case 8:  return audioMixBufConvToU8Stereo;
-                case 16: return audioMixBufConvToU16Stereo;
-                case 32: return audioMixBufConvToU32Stereo;
-                default: return NULL;
-            }
-        }
-        else
-        {
-            switch (AUDMIXBUF_FMT_BITS_PER_SAMPLE(enmFmt))
-            {
-                case 8:  return audioMixBufConvToU8Mono;
-                case 16: return audioMixBufConvToU16Mono;
-                case 32: return audioMixBufConvToU32Mono;
-                default: return NULL;
-            }
+            case 2:
+                switch (PDMAudioPropsSampleSize(pProps))
+                {
+                    case 1:  return audioMixBufConvToU8Stereo;
+                    case 2:  return audioMixBufConvToU16Stereo;
+                    case 4:  return audioMixBufConvToU32Stereo;
+                    default: return NULL;
+                }
+
+            case 1:
+                switch (PDMAudioPropsSampleSize(pProps))
+                {
+                    case 1:  return audioMixBufConvToU8Mono;
+                    case 2:  return audioMixBufConvToU16Mono;
+                    case 4:  return audioMixBufConvToU32Mono;
+                    default: return NULL;
+                }
+            default:
+                return NULL;
         }
     }
@@ -826 +804 @@
  * @param   cFrames                 Maximum number of audio frames the mixing buffer can hold.
  */
-int AudioMixBufInit(PPDMAUDIOMIXBUF pMixBuf, const char *pszName, PPDMAUDIOPCMPROPS pProps, uint32_t cFrames)
+int AudioMixBufInit(PPDMAUDIOMIXBUF pMixBuf, const char *pszName, PCPDMAUDIOPCMPROPS pProps, uint32_t cFrames)
 {
     AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER);
     AssertPtrReturn(pszName, VERR_INVALID_POINTER);
     AssertPtrReturn(pProps,  VERR_INVALID_POINTER);
+    Assert(PDMAudioPropsAreValid(pProps));
 
     pMixBuf->uMagic  = PDMAUDIOMIXBUF_MAGIC;
@@ -857 +836 @@
     pMixBuf->pRate = NULL;
 
+    /** @todo r=bird: Why invent a new representation for the mixer?  See also
+     *        comment in pdmaudioifs.h about missing MAKE macros. */
     pMixBuf->uAudioFmt = AUDMIXBUF_AUDIO_FMT_MAKE(pProps->uHz,
-                                                 pProps->cChannels,
-                                                 pProps->cbSample * 8 /* Bit */,
-                                                 pProps->fSigned);
-
-    pMixBuf->pfnConvFrom = audioMixBufConvFromLookup(pMixBuf->uAudioFmt);
-    pMixBuf->pfnConvTo   = audioMixBufConvToLookup(pMixBuf->uAudioFmt);
-
-    pMixBuf->cShift = pProps->cShift;
+                                                  PDMAudioPropsChannels(pProps),
+                                                  PDMAudioPropsSampleBits(pProps),
+                                                  pProps->fSigned);
+
+    pMixBuf->Props       = *pProps;
+    pMixBuf->pfnConvFrom = audioMixBufConvFromLookup(pProps);
+    pMixBuf->pfnConvTo   = audioMixBufConvToLookup(pProps);
+
     pMixBuf->pszName = RTStrDup(pszName);
     if (!pMixBuf->pszName)
         return VERR_NO_MEMORY;
 
-    AUDMIXBUF_LOG(("%s: uHz=%RU32, cChan=%RU8, cBits=%RU8, fSigned=%RTbool\n",
-                   pMixBuf->pszName,
-                   AUDMIXBUF_FMT_SAMPLE_FREQ(pMixBuf->uAudioFmt),
-                   AUDMIXBUF_FMT_CHANNELS(pMixBuf->uAudioFmt),
-                   AUDMIXBUF_FMT_BITS_PER_SAMPLE(pMixBuf->uAudioFmt),
-                   RT_BOOL(AUDMIXBUF_FMT_SIGNED(pMixBuf->uAudioFmt))));
+
+#ifdef AUDMIXBUF_LOG_ENABLED
+    char szTmp[PDMAUDIOPROPSTOSTRING_MAX];
+    AUDMIXBUF_LOG(("%s: %s\n", pMixBuf->pszName, PDMAudioPropsToString(pProps, szTmp, sizeof(szTmp))));
+#endif
 
     return audioMixBufAlloc(pMixBuf, cFrames);
@@ -1456 +1436 @@
  * @param   pcbRead                 Size (in bytes) of data read. Optional.
  */
-int AudioMixBufReadAt(PPDMAUDIOMIXBUF pMixBuf,
-                      uint32_t offFrames,
-                      void *pvBuf, uint32_t cbBuf,
-                      uint32_t *pcbRead)
-{
-    return AudioMixBufReadAtEx(pMixBuf, pMixBuf->uAudioFmt,
-                               offFrames, pvBuf, cbBuf, pcbRead);
+int AudioMixBufReadAt(PPDMAUDIOMIXBUF pMixBuf, uint32_t offFrames, void *pvBuf, uint32_t cbBuf, uint32_t *pcbRead)
+{
+    return AudioMixBufReadAtEx(pMixBuf, &pMixBuf->Props, offFrames, pvBuf, cbBuf, pcbRead);
 }
 
@@ -1471 +1447 @@
  *
  * @return  IPRT status code.
- * @param   pMixBuf                 Mixing buffer to read audio frames from.
- * @param   enmFmt                  Audio format to use for output.
- * @param   offFrames               Offset (in audio frames) to start reading from.
- * @param   pvBuf                   Pointer to buffer to write output to.
- * @param   cbBuf                   Size (in bytes) of buffer to write to.
- * @param   pcbRead                 Size (in bytes) of data read. Optional.
- */
-int AudioMixBufReadAtEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt,
-                        uint32_t offFrames,
-                        void *pvBuf, uint32_t cbBuf,
-                        uint32_t *pcbRead)
+ * @param   pMixBuf     Mixing buffer to read audio frames from.
+ * @param   pDstProps   The target format.
+ * @param   offFrames   Offset (in audio frames) to start reading from.
+ * @param   pvBuf       Pointer to buffer to write output to.
+ * @param   cbBuf       Size (in bytes) of buffer to write to.
+ * @param   pcbRead     Size (in bytes) of data read. Optional.
+ */
+int AudioMixBufReadAtEx(PPDMAUDIOMIXBUF pMixBuf, PCPDMAUDIOPCMPROPS pDstProps,
+                        uint32_t offFrames, void *pvBuf, uint32_t cbBuf, uint32_t *pcbRead)
 {
     AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER);
@@ -1501 +1475 @@
     {
         PFNPDMAUDIOMIXBUFCONVTO pfnConvTo = NULL;
-        if (pMixBuf->uAudioFmt != enmFmt)
-            pfnConvTo = audioMixBufConvToLookup(enmFmt);
+        if (PDMAudioPropsAreEqual(&pMixBuf->Props, pDstProps))
+            pfnConvTo = pMixBuf->pfnConvTo;
         else
-            pfnConvTo = pMixBuf->pfnConvTo;
-
+            pfnConvTo = audioMixBufConvToLookup(pDstProps);
         if (pfnConvTo)
         {
@@ -1552 +1525 @@
 int AudioMixBufAcquireReadBlock(PPDMAUDIOMIXBUF pMixBuf, void *pvBuf, uint32_t cbBuf, uint32_t *pcAcquiredFrames)
 {
-    return AudioMixBufAcquireReadBlockEx(pMixBuf, pMixBuf->uAudioFmt, pvBuf, cbBuf, pcAcquiredFrames);
+    return AudioMixBufAcquireReadBlockEx(pMixBuf, &pMixBuf->Props, pvBuf, cbBuf, pcAcquiredFrames);
 }
 
 /**
  * Reads audio frames in a specific audio format.
+ *
  * If the audio format of the mixing buffer and the requested audio format do
  * not match the output will be converted accordingly.
  *
- * @return  IPRT status code.
+ * @returns VBox status code.
  * @param   pMixBuf             Mixing buffer to read audio frames from.
- * @param   enmFmt              Audio format to use for output.
+ * @param   pDstProps           The target format.
  * @param   pvBuf               Pointer to buffer to write output to.
  * @param   cbBuf               Size (in bytes) of buffer to write to.
@@ -1568 +1542 @@
  *                              the block that was acquired.
  */
-int AudioMixBufAcquireReadBlockEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt,
+int AudioMixBufAcquireReadBlockEx(PPDMAUDIOMIXBUF pMixBuf, PCPDMAUDIOPCMPROPS pDstProps,
                                   void *pvBuf, uint32_t cbBuf, uint32_t *pcAcquiredFrames)
 {
@@ -1581 +1555 @@
     uint32_t cFramesToRead = RT_MIN(pMixBuf->cUsed, AUDIOMIXBUF_B2F(pMixBuf, cbBuf));
 
-    AUDMIXBUF_LOG(("%s: cbBuf=%RU32 (%RU32 frames), cFramesToRead=%RU32, fmtSrc=0x%x, fmtDst=0x%x\n",
-                   pMixBuf->pszName, cbBuf, AUDIOMIXBUF_B2F(pMixBuf, cbBuf), cFramesToRead, pMixBuf->uAudioFmt, enmFmt));
-
+#ifdef AUDMIXBUF_LOG_ENABLED
+    char szTmp1[PDMAUDIOPROPSTOSTRING_MAX], szTmp2[PDMAUDIOPROPSTOSTRING_MAX];
+#endif
+    AUDMIXBUF_LOG(("%s: cbBuf=%RU32 (%RU32 frames), cFramesToRead=%RU32, MixBuf=%s, pDstProps=%s\n",
+                   pMixBuf->pszName, cbBuf, AUDIOMIXBUF_B2F(pMixBuf, cbBuf), cFramesToRead,
+                   PDMAudioPropsToString(&pMixBuf->Props, szTmp1, sizeof(szTmp1)),
+                   PDMAudioPropsToString(pDstProps, szTmp2, sizeof(szTmp2))));
     if (!cFramesToRead)
     {
@@ -1594 +1572 @@
 
     PFNPDMAUDIOMIXBUFCONVTO pfnConvTo;
-    if (pMixBuf->uAudioFmt == enmFmt)
+    if (PDMAudioPropsAreEqual(&pMixBuf->Props, pDstProps))
         pfnConvTo = pMixBuf->pfnConvTo;
     else
-        pfnConvTo = audioMixBufConvToLookup(enmFmt);
+        pfnConvTo = audioMixBufConvToLookup(pDstProps);
     AssertReturn(pfnConvTo, VERR_NOT_SUPPORTED);
 
@@ -1799 +1777 @@
 int AudioMixBufWriteAt(PPDMAUDIOMIXBUF pMixBuf, uint32_t offFrames, const void *pvBuf, uint32_t cbBuf, uint32_t *pcWritten)
 {
-    return AudioMixBufWriteAtEx(pMixBuf, pMixBuf->uAudioFmt, offFrames, pvBuf, cbBuf, pcWritten);
+    return AudioMixBufWriteAtEx(pMixBuf, &pMixBuf->Props, offFrames, pvBuf, cbBuf, pcWritten);
 }
 
@@ -1812 +1790 @@
  *
  * @return  IPRT status code.
- * @param   pMixBuf                 Pointer to mixing buffer to write to.
- * @param   enmFmt                  Audio format supplied in the buffer.
- * @param   offFrames               Offset (in frames) starting to write at.
- * @param   pvBuf                   Pointer to audio buffer to be written.
- * @param   cbBuf                   Size (in bytes) of audio buffer.
- * @param   pcWritten               Returns number of audio frames written. Optional.
- */
-int AudioMixBufWriteAtEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt,
-                         uint32_t offFrames, const void *pvBuf, uint32_t cbBuf,
-                         uint32_t *pcWritten)
+ * @param   pMixBuf     Pointer to mixing buffer to write to.
+ * @param   pSrcProps   The source format.
+ * @param   offFrames   Offset (in frames) starting to write at.
+ * @param   pvBuf       Pointer to audio buffer to be written.
+ * @param   cbBuf       Size (in bytes) of audio buffer.
+ * @param   pcWritten   Returns number of audio frames written. Optional.
+ */
+int AudioMixBufWriteAtEx(PPDMAUDIOMIXBUF pMixBuf, PCPDMAUDIOPCMPROPS pSrcProps,
+                         uint32_t offFrames, const void *pvBuf, uint32_t cbBuf, uint32_t *pcWritten)
 {
     AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER);
@@ -1860 +1837 @@
     if (!pMixBuf->Volume.fMuted)
     {
-        if (pMixBuf->uAudioFmt != enmFmt)
-            pfnConvFrom = audioMixBufConvFromLookup(enmFmt);
+        if (PDMAudioPropsAreEqual(&pMixBuf->Props, pSrcProps))
+            pfnConvFrom = pMixBuf->pfnConvFrom;
         else
-            pfnConvFrom = pMixBuf->pfnConvFrom;
+            pfnConvFrom = audioMixBufConvFromLookup(pSrcProps);
+        AssertReturn(pfnConvFrom, VERR_NOT_SUPPORTED);
     }
     else
@@ -1871 +1849 @@
 
     uint32_t cWritten;
-    if (   pfnConvFrom
-        && cToWrite)
+    if (cToWrite)
     {
         PDMAUDMIXBUFCONVOPTS convOpts;
 
-        convOpts.cFrames           = cToWrite;
+        convOpts.cFrames            = cToWrite;
         convOpts.From.Volume.fMuted = pMixBuf->Volume.fMuted;
         convOpts.From.Volume.uLeft  = pMixBuf->Volume.uLeft;
@@ -1884 +1861 @@
     }
     else
-    {
         cWritten = 0;
-        if (!pfnConvFrom)
-        {
-            AssertFailed();
-            rc = VERR_NOT_SUPPORTED;
-        }
-    }
 
     AUDMIXBUF_LOG(("%s: offFrames=%RU32, cbBuf=%RU32, cToWrite=%RU32 (%zu bytes), cWritten=%RU32 (%zu bytes), rc=%Rrc\n",
@@ -1929 +1899 @@
  * @param   pcWritten               Returns number of audio frames written. Optional.
  */
-int AudioMixBufWriteCirc(PPDMAUDIOMIXBUF pMixBuf,
-                         const void *pvBuf, uint32_t cbBuf,
-                         uint32_t *pcWritten)
-{
-    return AudioMixBufWriteCircEx(pMixBuf, pMixBuf->uAudioFmt, pvBuf, cbBuf, pcWritten);
+int AudioMixBufWriteCirc(PPDMAUDIOMIXBUF pMixBuf, const void *pvBuf, uint32_t cbBuf, uint32_t *pcWritten)
+{
+    return AudioMixBufWriteCircEx(pMixBuf, &pMixBuf->Props, pvBuf, cbBuf, pcWritten);
 }
 
@@ -1941 +1909 @@
  *
  * @return  IPRT status code, or VERR_BUFFER_OVERFLOW no space is available for writing anymore.
- * @param   pMixBuf                 Pointer to mixing buffer to write to.
- * @param   enmFmt                  Audio format supplied in the buffer.
- * @param   pvBuf                   Pointer to audio buffer to be written.
- * @param   cbBuf                   Size (in bytes) of audio buffer.
- * @param   pcWritten               Returns number of audio frames written. Optional.
- */
-int AudioMixBufWriteCircEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt,
+ * @param   pMixBuf     Pointer to mixing buffer to write to.
+ * @param   pSrcProps   The source format.
+ * @param   enmFmt      Audio format supplied in the buffer.
+ * @param   pvBuf       Pointer to audio buffer to be written.
+ * @param   cbBuf       Size (in bytes) of audio buffer.
+ * @param   pcWritten   Returns number of audio frames written. Optional.
+ */
+int AudioMixBufWriteCircEx(PPDMAUDIOMIXBUF pMixBuf, PCPDMAUDIOPCMPROPS pSrcProps,
                            const void *pvBuf, uint32_t cbBuf, uint32_t *pcWritten)
 {
     AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER);
     AssertPtrReturn(pvBuf,   VERR_INVALID_POINTER);
-    /* pcbWritten is optional. */
+    AssertPtrNullReturn(pcWritten, VERR_INVALID_POINTER);
 
     if (!cbBuf)
@@ -1970 +1939 @@
     if (!pMixBuf->Volume.fMuted)
     {
-        if (pMixBuf->uAudioFmt != enmFmt)
-            pfnConvFrom = audioMixBufConvFromLookup(enmFmt);
+        if (PDMAudioPropsAreEqual(&pMixBuf->Props, pSrcProps))
+            pfnConvFrom = pMixBuf->pfnConvFrom;
         else
-            pfnConvFrom = pMixBuf->pfnConvFrom;
+            pfnConvFrom = audioMixBufConvFromLookup(pSrcProps);
+        AssertReturn(pfnConvFrom, VERR_NOT_SUPPORTED);
     }
     else
         pfnConvFrom = &audioMixBufConvFromSilence;
-
-    if (!pfnConvFrom)
-    {
-        AssertFailed();
-        return VERR_NOT_SUPPORTED;
-    }
 
     int rc = VINF_SUCCESS;
@@ -2034 +1998 @@
         *pcWritten = cWritten;
 
-    AUDMIXBUF_LOG(("%s: enmFmt=0x%x, cbBuf=%RU32 (%RU32 frames), cWritten=%RU32, rc=%Rrc\n",
-                   pMixBuf->pszName, enmFmt, cbBuf, AUDIOMIXBUF_B2F(pMixBuf, cbBuf), cWritten, rc));
+#ifdef AUDMIXBUF_LOG_ENABLED
+    char szTmp[PDMAUDIOPROPSTOSTRING_MAX];
+#endif
+    AUDMIXBUF_LOG(("%s: pSrcProps=%s, cbBuf=%RU32 (%RU32 frames), cWritten=%RU32, rc=%Rrc\n", pMixBuf->pszName,
+                   PDMAudioPropsToString(pSrcProps, szTmp, sizeof(szTmp)), cbBuf, AUDIOMIXBUF_B2F(pMixBuf, cbBuf), cWritten, rc));
     return rc;
 }

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

@@ -26 +26 @@
 
 /** Constructs 32 bit value for given frequency, number of channels, bits per sample and signed bit.
- *  Note: This currently matches 1:1 the VRDE encoding -- this might change in the future, so better don't rely on this fact! */
+ * @note This currently matches 1:1 the VRDE encoding -- this might change in the future, so better don't rely on this fact! */
 #define AUDMIXBUF_AUDIO_FMT_MAKE(freq, c, bps, s) ((((s) & 0x1) << 28) + (((bps) & 0xFF) << 20) + (((c) & 0xF) << 16) + ((freq) & 0xFFFF))
 
@@ -40 +40 @@
 #define AUDMIXBUF_FMT_BYTES_PER_SAMPLE(a) ((AUDMIXBUF_AUDIO_FMT_BITS_PER_SAMPLE(a) + 7) / 8)
 
-/** Converts frames to bytes. */
-#define AUDIOMIXBUF_F2B(pBuf, frames) ((frames) << (pBuf)->cShift)
+/** Converts (audio) frames to bytes. */
+#define AUDIOMIXBUF_F2B(a_pMixBuf, a_cFrames)   PDMAUDIOPCMPROPS_F2B(&(a_pMixBuf)->Props, a_cFrames)
+/** Converts bytes to (audio) frames.
+ * @note Does *not* take the conversion ratio into account. */
+#define AUDIOMIXBUF_B2F(a_pMixBuf, a_cb)        PDMAUDIOPCMPROPS_B2F(&(a_pMixBuf)->Props, a_cb)
+
 /** Converts frames to bytes, respecting the conversion ratio to
  *  a linked buffer. */
-#define AUDIOMIXBUF_F2B_RATIO(pBuf, frames) ((((int64_t) frames << 32) / (pBuf)->iFreqRatio) << (pBuf)->cShift)
-/** Converts bytes to frames, *not* taking the conversion ratio
- *  into account. */
-#define AUDIOMIXBUF_B2F(pBuf, cb)  (cb >> (pBuf)->cShift)
-/** Converts number of frames according to the buffer's ratio. */
-#define AUDIOMIXBUF_F2F_RATIO(pBuf, frames)  (((int64_t) frames << 32) / (pBuf)->iFreqRatio)
+#define AUDIOMIXBUF_F2B_RATIO(a_pMixBuf, a_cFrames) AUDIOMIXBUF_F2B(a_pMixBuf, AUDIOMIXBUF_F2F_RATIO(a_pMixBuf, a_cFrames))
+/** Converts number of frames according to the buffer's ratio.
+ * @todo r=bird: Why the *signed* cast?  */
+#define AUDIOMIXBUF_F2F_RATIO(a_pMixBuf, a_cFrames) (((int64_t)(a_cFrames) << 32) / (a_pMixBuf)->iFreqRatio)
 
 
@@ -58 +60 @@
 uint32_t AudioMixBufFree(PPDMAUDIOMIXBUF pMixBuf);
 uint32_t AudioMixBufFreeBytes(PPDMAUDIOMIXBUF pMixBuf);
-int AudioMixBufInit(PPDMAUDIOMIXBUF pMixBuf, const char *pszName, PPDMAUDIOPCMPROPS pProps, uint32_t cFrames);
+int AudioMixBufInit(PPDMAUDIOMIXBUF pMixBuf, const char *pszName, PCPDMAUDIOPCMPROPS pProps, uint32_t cFrames);
 bool AudioMixBufIsEmpty(PPDMAUDIOMIXBUF pMixBuf);
 int AudioMixBufLinkTo(PPDMAUDIOMIXBUF pMixBuf, PPDMAUDIOMIXBUF pParent);
@@ -64 +66 @@
 int AudioMixBufMixToParent(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSrcFrames, uint32_t *pcSrcMixed);
 int AudioMixBufMixToParentEx(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSrcOffset, uint32_t cSrcFrames, uint32_t *pcSrcMixed);
-int AudioMixBufPeek(PPDMAUDIOMIXBUF pMixBuf, uint32_t cFramesToRead, PPDMAUDIOFRAME paSampleBuf, uint32_t cSampleBuf, uint32_t *pcFramesRead);
 int AudioMixBufPeekMutable(PPDMAUDIOMIXBUF pMixBuf, uint32_t cFramesToRead, PPDMAUDIOFRAME *ppvSamples, uint32_t *pcFramesRead);
 uint32_t AudioMixBufUsed(PPDMAUDIOMIXBUF pMixBuf);
 uint32_t AudioMixBufUsedBytes(PPDMAUDIOMIXBUF pMixBuf);
-int AudioMixBufReadAt(PPDMAUDIOMIXBUF pMixBuf, uint32_t offSamples, void *pvBuf, uint32_t cbBuf, uint32_t *pcbRead);
-int AudioMixBufReadAtEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt, uint32_t offSamples, void *pvBuf, uint32_t cbBuf, uint32_t *pcbRead);
-int AudioMixBufAcquireReadBlock(PPDMAUDIOMIXBUF pMixBuf, void *pvBuf, uint32_t cbBuf, uint32_t *pcAcquiredFrames);
-int AudioMixBufAcquireReadBlockEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt, void *pvBuf, uint32_t cbBuf, uint32_t *pcAcquiredFrames);
+int         AudioMixBufReadAt(PPDMAUDIOMIXBUF pMixBuf, uint32_t offFrames, void *pvBuf, uint32_t cbBuf, uint32_t *pcbRead);
+int         AudioMixBufReadAtEx(PPDMAUDIOMIXBUF pMixBuf, PCPDMAUDIOPCMPROPS pDstProps, uint32_t offFrames,
+                                void *pvBuf, uint32_t cbBuf, uint32_t *pcbRead);
+int         AudioMixBufAcquireReadBlock(PPDMAUDIOMIXBUF pMixBuf, void *pvBuf, uint32_t cbBuf, uint32_t *pcAcquiredFrames);
+int         AudioMixBufAcquireReadBlockEx(PPDMAUDIOMIXBUF pMixBuf, PCPDMAUDIOPCMPROPS pDstProps,
+                                          void *pvBuf, uint32_t cbBuf, uint32_t *pcAcquiredFrames);
 void AudioMixBufReleaseReadBlock(PPDMAUDIOMIXBUF pMixBuf, uint32_t cFrames);
 uint32_t AudioMixBufReadPos(PPDMAUDIOMIXBUF pMixBuf);
@@ -79 +82 @@
 uint32_t AudioMixBufSizeBytes(PPDMAUDIOMIXBUF pMixBuf);
 void AudioMixBufUnlink(PPDMAUDIOMIXBUF pMixBuf);
-int AudioMixBufWriteAt(PPDMAUDIOMIXBUF pMixBuf, uint32_t offSamples, const void *pvBuf, uint32_t cbBuf, uint32_t *pcWritten);
-int AudioMixBufWriteAtEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt, uint32_t offSamples, const void *pvBuf, uint32_t cbBuf, uint32_t *pcWritten);
-int AudioMixBufWriteCirc(PPDMAUDIOMIXBUF pMixBuf, const void *pvBuf, uint32_t cbBuf, uint32_t *pcWritten);
-int AudioMixBufWriteCircEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt, const void *pvBuf, uint32_t cbBuf, uint32_t *pcWritten);
+int         AudioMixBufWriteAt(PPDMAUDIOMIXBUF pMixBuf, uint32_t offSamples, const void *pvBuf, uint32_t cbBuf, uint32_t *pcWritten);
+int         AudioMixBufWriteAtEx(PPDMAUDIOMIXBUF pMixBuf, PCPDMAUDIOPCMPROPS pSrcProps, uint32_t offFrames,
+                                 const void *pvBuf, uint32_t cbBuf, uint32_t *pcWritten);
+int         AudioMixBufWriteCirc(PPDMAUDIOMIXBUF pMixBuf, const void *pvBuf, uint32_t cbBuf, uint32_t *pcWritten);
+int         AudioMixBufWriteCircEx(PPDMAUDIOMIXBUF pMixBuf, PCPDMAUDIOPCMPROPS pSrcProps,
+                                   const void *pvBuf,uint32_t cbBuf, uint32_t *pcWritten);
 uint32_t AudioMixBufWritePos(PPDMAUDIOMIXBUF pMixBuf);
 

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

@@ -264 +264 @@
         }
         else
-            RTMemFree(pMixer);
+            RTMemFree(pMixer); /** @todo leaks pszName due to badly structured code */
     }
     else
@@ -611 +611 @@
         return rc;
 
-    LogFlowFunc(("[%s] fFlags=0x%x (enmDir=%ld, %u bits, %RU8 channels, %RU32Hz)\n",
-                 pSink->pszName, fFlags, pCfg->enmDir, pCfg->Props.cbSample * 8, pCfg->Props.cChannels, pCfg->Props.uHz));
+    LogFlowFunc(("[%s] fFlags=0x%x (enmDir=%ld, %u bits, %RU8 channels, %RU32Hz)\n", pSink->pszName, fFlags, pCfg->enmDir,
+                 PDMAudioPropsSampleBits(&pCfg->Props), PDMAudioPropsChannels(&pCfg->Props), pCfg->Props.uHz));
 
     /*
@@ -1488 +1488 @@
 
     if (pSink->PCMProps.uHz)
-        LogFlowFunc(("[%s] Old format: %u bit, %RU8 channels, %RU32Hz\n",
-                     pSink->pszName, pSink->PCMProps.cbSample * 8, pSink->PCMProps.cChannels, pSink->PCMProps.uHz));
+        LogFlowFunc(("[%s] Old format: %u bit, %RU8 channels, %RU32Hz\n", pSink->pszName,
+                     PDMAudioPropsSampleBits(&pSink->PCMProps), PDMAudioPropsChannels(&pSink->PCMProps), pSink->PCMProps.uHz));
 
     memcpy(&pSink->PCMProps, pPCMProps, sizeof(PDMAUDIOPCMPROPS));
 
-    LogFlowFunc(("[%s] New format %u bit, %RU8 channels, %RU32Hz\n",
-                 pSink->pszName, pSink->PCMProps.cbSample * 8, pSink->PCMProps.cChannels, pSink->PCMProps.uHz));
+    LogFlowFunc(("[%s] New format %u bit, %RU8 channels, %RU32Hz\n", pSink->pszName, PDMAudioPropsSampleBits(&pSink->PCMProps),
+                 PDMAudioPropsChannels(&pSink->PCMProps), pSink->PCMProps.uHz));
 
     /* Also update the sink's mixing buffer format. */

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

@@ -1449 +1449 @@
 
 # ifdef LOG_ENABLED
-                PDMAUDIOPCMPROPS Props;
-                rc2 = hdaR3SDFMTToPCMProps(HDA_STREAM_REG(pThis, FMT, uSD), &Props);
-                AssertRC(rc2);
-                LogFunc(("[SD%RU8] %RU32Hz, %RU8bit, %RU8 channel(s)\n",
-                         uSD, Props.uHz, Props.cbSample * 8 /* Bit */, Props.cChannels));
+                if (LogIsEnabled())
+                {
+                    PDMAUDIOPCMPROPS Props = { 0 };
+                    rc2 = hdaR3SDFMTToPCMProps(HDA_STREAM_REG(pThis, FMT, uSD), &Props); AssertRC(rc2);
+                    LogFunc(("[SD%RU8] %RU32Hz, %RU8bit, %RU8 channel(s)\n",
+                             uSD, Props.uHz, PDMAudioPropsSampleBits(&Props), PDMAudioPropsChannels(&Props)));
+                }
 # endif
                 /* (Re-)initialize the stream with current values. */
@@ -1667 +1669 @@
 
     /** @todo Make the following configurable through mixer API and/or CFGM? */
-    switch (pCfg->Props.cChannels)
+    switch (PDMAudioPropsGetChannels(&pCfg->Props))
     {
         case 3:  /* 2.1: Front (Stereo) + LFE. */
@@ -1706 +1708 @@
     if (rc == VERR_NOT_SUPPORTED)
     {
-        LogRel2(("HDA: Warning: Unsupported channel count (%RU8), falling back to stereo channels (2)\n", pCfg->Props.cChannels));
+        LogRel2(("HDA: Warning: Unsupported channel count (%RU8), falling back to stereo channels (2)\n", pCfg->Props.cChannelsX));
 
         /* Fall back to 2 channels (see below in fUseFront block). */
@@ -1723 +1725 @@
             pCfg->u.enmDst          = PDMAUDIOPLAYBACKDST_FRONT;
             pCfg->enmLayout         = PDMAUDIOSTREAMLAYOUT_NON_INTERLEAVED;
-
-            pCfg->Props.cShift      = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(pCfg->Props.cbSample, pCfg->Props.cChannels);
+            /// @todo PDMAudioPropsSetChannels(&pCfg->Props, 2); ?
 
             rc = hdaR3CodecAddStream(pThisCC->pCodec, PDMAUDIOMIXERCTL_FRONT, pCfg);
@@ -1737 +1738 @@
             pCfg->u.enmDst          = PDMAUDIOPLAYBACKDST_CENTER_LFE;
             pCfg->enmLayout         = PDMAUDIOSTREAMLAYOUT_NON_INTERLEAVED;
-
-            pCfg->Props.cChannels   = (fUseCenter && fUseLFE) ? 2 : 1;
-            pCfg->Props.cShift      = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(pCfg->Props.cbSample, pCfg->Props.cChannels);
+            PDMAudioPropsSetChannels(&pCfg->Props, fUseCenter && fUseLFE ? 2 : 1);
 
             rc = hdaR3CodecAddStream(pThisCC->pCodec, PDMAUDIOMIXERCTL_CENTER_LFE, pCfg);
@@ -1751 +1750 @@
             pCfg->u.enmDst          = PDMAUDIOPLAYBACKDST_REAR;
             pCfg->enmLayout         = PDMAUDIOSTREAMLAYOUT_NON_INTERLEAVED;
-
-            pCfg->Props.cChannels   = 2;
-            pCfg->Props.cShift      = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(pCfg->Props.cbSample, pCfg->Props.cChannels);
+            PDMAudioPropsSetChannels(&pCfg->Props, 2);
 
             rc = hdaR3CodecAddStream(pThisCC->pCodec, PDMAUDIOMIXERCTL_REAR, pCfg);
@@ -1912 +1909 @@
     int rc2 = hdaR3SDFMTToPCMProps(RT_LO_U16(u32Value), &Props);
     AssertRC(rc2);
-    LogFunc(("[SD%RU8] Set to %#x (%RU32Hz, %RU8bit, %RU8 channel(s))\n",
-             HDA_SD_NUM_FROM_REG(pThis, FMT, iReg), u32Value, Props.uHz, Props.cbSample * 8 /* Bit */, Props.cChannels));
+    LogFunc(("[SD%RU8] Set to %#x (%RU32Hz, %RU8bit, %RU8 channel(s))\n", HDA_SD_NUM_FROM_REG(pThis, FMT, iReg), u32Value,
+             PDMAudioPropsHz(&Props), PDMAudioPropsSampleBits(&Props), PDMAudioPropsChannels(&Props)));
 
     /*
@@ -2319 +2316 @@
     AssertPtrReturn(pCfg,     VERR_INVALID_POINTER);
 
-    LogFunc(("szSink=%s, szStream=%s, cChannels=%RU8\n", pMixSink->pszName, pCfg->szName, pCfg->Props.cChannels));
+    LogFunc(("szSink=%s, szStream=%s, cChannels=%RU8\n", pMixSink->pszName, pCfg->szName, PDMAudioPropsChannels(&pCfg->Props)));
 
     PPDMAUDIOSTREAMCFG pStreamCfg = PDMAudioStrmCfgDup(pCfg);
@@ -5356 +5353 @@
         PDMDevHlpSTAMRegisterF(pDevIns, &pThis->aStreams[idxStream].State.Cfg.Props.uHz, STAMTYPE_U32, STAMVISIBILITY_USED, STAMUNIT_BYTES,
                                "The stream frequency.",                     "Stream%u/Cfg/Hz", idxStream);
-        PDMDevHlpSTAMRegisterF(pDevIns, &pThis->aStreams[idxStream].State.Cfg.Props.cChannels, STAMTYPE_U8, STAMVISIBILITY_USED, STAMUNIT_BYTES,
+        PDMDevHlpSTAMRegisterF(pDevIns, &pThis->aStreams[idxStream].State.Cfg.Props.cbFrame, STAMTYPE_U8, STAMVISIBILITY_USED, STAMUNIT_BYTES,
+                               "The number of channels.",                   "Stream%u/Cfg/FrameSize-Host", idxStream);
+        PDMDevHlpSTAMRegisterF(pDevIns, &pThisCC->aStreams[idxStream].State.Mapping.GuestProps.cbFrame, STAMTYPE_U8, STAMVISIBILITY_USED, STAMUNIT_BYTES,
+                               "The number of channels.",                   "Stream%u/Cfg/FrameSize-Guest", idxStream);
+#if 0 /** @todo this would require some callback */
+        PDMDevHlpSTAMRegisterF(pDevIns, &pThis->aStreams[idxStream].State.Cfg.Props.cChannelsX, STAMTYPE_U8, STAMVISIBILITY_USED, STAMUNIT_BYTES,
                                "The number of channels.",                   "Stream%u/Cfg/Channels-Host", idxStream);
         PDMDevHlpSTAMRegisterF(pDevIns, &pThisCC->aStreams[idxStream].State.Mapping.GuestProps.cChannels, STAMTYPE_U8, STAMVISIBILITY_USED, STAMUNIT_BYTES,
@@ -5362 +5364 @@
         PDMDevHlpSTAMRegisterF(pDevIns, &pThis->aStreams[idxStream].State.Cfg.Props.cbSample, STAMTYPE_U8, STAMVISIBILITY_USED, STAMUNIT_BYTES,
                                "The size of a sample (per channel).",       "Stream%u/Cfg/cbSample", idxStream);
+#endif
     }
 

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

@@ -22 +22 @@
 *   Header Files                                                                                                                 *
 *********************************************************************************************************************************/
+#define LOG_GROUP LOG_GROUP_DEV_HDA
 #include <iprt/assert.h>
 #include <iprt/errcore.h>
@@ -27 +28 @@
 
 #include <VBox/AssertGuest.h>
-
-#define LOG_GROUP LOG_GROUP_DEV_HDA
+#include <VBox/vmm/pdmaudioinline.h>
+
 #include <VBox/log.h>
 
@@ -34 +35 @@
 #include "DevHdaCommon.h"
 #include "DevHdaStream.h"
+
 
 
@@ -304 +306 @@
     }
 
-    uint8_t cBytes = 0;
+    uint8_t cbSample = 0;
     switch (EXTRACT_VALUE(u16SDFMT, HDA_SDFMT_BITS_MASK, HDA_SDFMT_BITS_SHIFT))
     {
         case 0:
-            cBytes = 1;
+            cbSample = 1;
             break;
         case 1:
-            cBytes = 2;
+            cbSample = 2;
             break;
         case 4:
-            cBytes = 4;
+            cbSample = 4;
             break;
         default:
@@ -324 +326 @@
 
     if (RT_SUCCESS(rc))
-    {
-        RT_BZERO(pProps, sizeof(PDMAUDIOPCMPROPS));
-
-        pProps->cbSample  = cBytes;
-        pProps->fSigned   = true;
-        pProps->cChannels = (u16SDFMT & 0xf) + 1;
-        pProps->uHz       = u32Hz * u32HzMult / u32HzDiv;
-        pProps->cShift    = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(pProps->cbSample, pProps->cChannels);
-    }
+        PDMAudioPropsInit(pProps, cbSample, true /*fSigned*/, (u16SDFMT & 0xf) + 1 /*cChannels*/, u32Hz * u32HzMult / u32HzDiv);
 
 # undef EXTRACT_VALUE

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

@@ -739 +739 @@
 
     /* Serious paranoia: */
-    ASSERT_GUEST_LOGREL_MSG_STMT(cbCircBuf % (pCfg->Props.cbSample * pCfg->Props.cChannels) == 0,
+    ASSERT_GUEST_LOGREL_MSG_STMT(cbCircBuf % PDMAudioPropsFrameSize(&pCfg->Props) == 0,
                                  ("Ring buffer size (%RU32) for stream #%RU8 not aligned to the (host) frame size (%RU8)\n",
-                                  cbCircBuf, uSD, pCfg->Props.cbSample * pCfg->Props.cChannels),
+                                  cbCircBuf, uSD, PDMAudioPropsFrameSize(&pCfg->Props)),
                                  rc = VERR_INVALID_PARAMETER);
     ASSERT_GUEST_LOGREL_MSG_STMT(cbCircBuf, ("Ring buffer size for stream #%RU8 is invalid\n", uSD),

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

@@ -209 +209 @@
      *        - have a simple (raw) data layout
      *        - work in a non-striped fashion, e.g. interleaved (only on one SDn, not spread over multiple SDns) */
-    if  (   pProps->cChannels == 1  /* Mono */
-         || pProps->cChannels == 2  /* Stereo */
-         || pProps->cChannels == 4  /* Quadrophonic */
-         || pProps->cChannels == 6) /* Surround (5.1) */
+    uint8_t cChannels = PDMAudioPropsChannels(pProps);
+    if  (   cChannels == 1  /* Mono */
+         || cChannels == 2  /* Stereo */
+         || cChannels == 4  /* Quadrophonic */
+         || cChannels == 6) /* Surround (5.1) */
     {
         /*
@@ -218 +219 @@
          */
         memcpy(&pMap->GuestProps, pProps, sizeof(PDMAUDIOPCMPROPS));
-        if (pProps->cChannels != cHostChannels)
+        if (cChannels != cHostChannels)
         {
-            if (pProps->cChannels == 1)
+            if (cChannels == 1)
                 LogRelMax(32, ("HDA: Warning: Guest mono, host stereo.\n"));
-            else if (cHostChannels == 1 && pProps->cChannels == 2)
+            else if (cHostChannels == 1 && cChannels == 2)
                 LogRelMax(32, ("HDA: Warning: Host mono, guest stereo.\n"));
             else
 #ifndef VBOX_WITH_AUDIO_HDA_51_SURROUND
                 LogRelMax(32, ("HDA: Warning: Guest configured %u channels, host only supports %u. Ignoring additional channels.\n",
-                               pProps->cChannels, cHostChannels));
+                               cChannels, cHostChannels));
 #else
 # error reconsider the above logic
 #endif
-            pProps->cChannels = cHostChannels;
-            pProps->cShift    = PDMAUDIOPCMPROPS_MAKE_SHIFT(pProps);
+            PDMAudioPropsSetChannels(pProps, cHostChannels);
         }
 
@@ -238 +238 @@
          * Pick conversion functions.
          */
-        Assert(pMap->GuestProps.cbSample == pProps->cbSample);
+        Assert(PDMAudioPropsSampleSize(&pMap->GuestProps) == PDMAudioPropsSampleSize(pProps));
 
         /* If the channel count matches, we can use the memcpy converters: */
-        if (pProps->cChannels == pMap->GuestProps.cChannels)
+        if (PDMAudioPropsChannels(pProps) == PDMAudioPropsChannels(&pMap->GuestProps))
         {
             pMap->pfnGuestToHost = hdaR3StreamMap_GenericCopy;
@@ -252 +252 @@
                first two channels and map this onto the host stereo ones. */
             AssertReturn(cHostChannels == 2, VERR_NOT_SUPPORTED);
-            switch (pMap->GuestProps.cbSample)
+            switch (PDMAudioPropsSampleSize(&pMap->GuestProps))
             {
                 case 2:
-                    if (pMap->GuestProps.cChannels > 1)
+                    if (PDMAudioPropsChannels(&pMap->GuestProps) > 1)
                     {
                         pMap->pfnGuestToHost = hdaR3StreamMap_G2H_GenericS16_NonMono2Stereo;
@@ -268 +268 @@
 
                 case 4:
-                    if (pMap->GuestProps.cChannels > 1)
+                    if (PDMAudioPropsChannels(&pMap->GuestProps) > 1)
                     {
                         pMap->pfnGuestToHost = hdaR3StreamMap_G2H_GenericS32_NonMono2Stereo;
@@ -281 +281 @@
 
                 default:
-                    AssertMsgFailedReturn(("cbSample=%u\n", pMap->GuestProps.cbSample), VERR_NOT_SUPPORTED);
+                    AssertMsgFailedReturn(("cbSample=%u\n", PDMAudioPropsChannels(&pMap->GuestProps)), VERR_NOT_SUPPORTED);
             }
             pMap->fMappingNeeded = true;
@@ -297 +297 @@
             pMapLR->aenmIDs[0]  = PDMAUDIOSTREAMCHANNELID_FRONT_LEFT;
             pMapLR->aenmIDs[1]  = PDMAUDIOSTREAMCHANNELID_FRONT_RIGHT;
-            pMapLR->cbFrame     = pProps->cbSample * pProps->cChannels;
-            pMapLR->cbStep      = pProps->cbSample * 2 /* Front left + Front right channels */;
+            pMapLR->cbFrame     = PDMAudioPropsFrameSize(pProps);
+            pMapLR->cbStep      = PDMAudioPropsSampleSize(pProps) * 2 /* Front left + Front right channels */;
             pMapLR->offFirst    = 0;
             pMapLR->offNext     = pMapLR->offFirst;
@@ -334 +334 @@
     hdaR3StreamMapReset(pMap);
 
-    pMap->cbGuestFrame = pProps->cChannels * pProps->cbSample;
+    pMap->cbGuestFrame = PDMAudioPropsFrameSize(pProps);
     int rc = hdaR3StreamMapSetup(pMap, pProps, cHostChannels);
     if (RT_SUCCESS(rc))
@@ -346 +346 @@
         {
             LogFunc(("cChannels=%RU8, cBytes=%RU8 -> cbGuestFrame=%RU32\n",
-                     pProps->cChannels, pProps->cbSample, pMap->cbGuestFrame));
+                     PDMAudioPropsChannels(pProps), PDMAudioPropsSampleSize(pProps), pMap->cbGuestFrame));
 
             Assert(pMap->cbGuestFrame); /* Frame size must not be 0. */

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

@@ -2137 +2137 @@
 static int ichac97R3StreamOpen(PAC97STATE pThis, PAC97STATER3 pThisCC, PAC97STREAM pStream, PAC97STREAMR3 pStreamCC, bool fForce)
 {
-    PDMAUDIOSTREAMCFG Cfg;
+    int                 rc = VINF_SUCCESS;
+    PAUDMIXSINK         pMixSink;
+    PDMAUDIOSTREAMCFG   Cfg;
     RT_ZERO(Cfg);
-    Cfg.Props.cChannels = 2;
-    Cfg.Props.cbSample  = 2 /* 16-bit */;
-    Cfg.Props.fSigned   = true;
-    Cfg.Props.cShift    = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(Cfg.Props.cbSample, Cfg.Props.cChannels);
-
-    int rc = VINF_SUCCESS;
-    PAUDMIXSINK pMixSink;
     switch (pStream->u8SD)
     {
         case AC97SOUNDSOURCE_PI_INDEX:
         {
-            Cfg.Props.uHz   = ichac97MixerGet(pThis, AC97_PCM_LR_ADC_Rate);
+            PDMAudioPropsInit(&Cfg.Props, 2 /*16-bit*/, true /*signed*/, 2 /*stereo*/,
+                              ichac97MixerGet(pThis, AC97_PCM_LR_ADC_Rate));
             Cfg.enmDir      = PDMAUDIODIR_IN;
             Cfg.u.enmSrc    = PDMAUDIORECSRC_LINE;
@@ -2162 +2158 @@
         case AC97SOUNDSOURCE_MC_INDEX:
         {
-            Cfg.Props.uHz   = ichac97MixerGet(pThis, AC97_MIC_ADC_Rate);
+            PDMAudioPropsInit(&Cfg.Props, 2 /*16-bit*/, true /*signed*/, 2 /*stereo*/,
+                              ichac97MixerGet(pThis, AC97_MIC_ADC_Rate));
             Cfg.enmDir      = PDMAUDIODIR_IN;
             Cfg.u.enmSrc    = PDMAUDIORECSRC_MIC;
@@ -2174 +2171 @@
         case AC97SOUNDSOURCE_PO_INDEX:
         {
-            Cfg.Props.uHz   = ichac97MixerGet(pThis, AC97_PCM_Front_DAC_Rate);
+            PDMAudioPropsInit(&Cfg.Props, 2 /*16-bit*/, true /*signed*/, 2 /*stereo*/,
+                              ichac97MixerGet(pThis, AC97_PCM_Front_DAC_Rate));
             Cfg.enmDir      = PDMAUDIODIR_OUT;
             Cfg.u.enmDst    = PDMAUDIOPLAYBACKDST_FRONT;
@@ -2197 +2195 @@
             || fForce)
         {
-            LogRel2(("AC97: (Re-)Opening stream '%s' (%RU32Hz, %RU8 channels, %s%RU8)\n",
-                     Cfg.szName, Cfg.Props.uHz, Cfg.Props.cChannels, Cfg.Props.fSigned ? "S" : "U", Cfg.Props.cbSample * 8));
+            LogRel2(("AC97: (Re-)Opening stream '%s' (%RU32Hz, %RU8 channels, %s%RU8)\n", Cfg.szName, Cfg.Props.uHz,
+                     PDMAudioPropsChannels(&Cfg.Props), Cfg.Props.fSigned ? "S" : "U",  PDMAudioPropsSampleBits(&Cfg.Props)));
 
             LogFlowFunc(("[SD%RU8] uHz=%RU32\n", pStream->u8SD, Cfg.Props.uHz));

trunk/src/VBox/Devices/Audio/DevSB16.cpp

@@ -910 +910 @@
     pCfg->u.enmDst        = PDMAUDIOPLAYBACKDST_FRONT;
     pCfg->enmLayout       = PDMAUDIOSTREAMLAYOUT_NON_INTERLEAVED;
-
-    pCfg->Props.uHz       = pThis->freq;
-    pCfg->Props.cChannels = 1; /* Mono */
-    pCfg->Props.cbSample  = 1 /* 8-bit */;
-    pCfg->Props.fSigned   = false;
-    pCfg->Props.cShift    = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(pCfg->Props.cbSample, pCfg->Props.cChannels);
+    PDMAudioPropsInit(&pCfg->Props, 1 /*8-bit*/, false /*fSigned*/, 1 /*mono*/, pThis->freq);
 
     AssertCompile(sizeof(pCfg->szName) >= sizeof("Output"));
@@ -1833 +1828 @@
         PDMAUDIOSTREAMCFG Cfg;
         RT_ZERO(Cfg);
-
-        Cfg.Props.uHz       = pThis->freq;
-        Cfg.Props.cChannels = 1 << pThis->fmt_stereo;
-        Cfg.Props.cbSample  = pThis->fmt_bits / 8;
-        Cfg.Props.fSigned   = RT_BOOL(pThis->fmt_signed);
-        Cfg.Props.cShift    = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(Cfg.Props.cbSample, Cfg.Props.cChannels);
+        PDMAudioPropsInit(&Cfg.Props, pThis->fmt_bits / 8, pThis->fmt_signed != 0, 1 << pThis->fmt_stereo, pThis->freq);
 
         if (!PDMAudioStrmCfgMatchesProps(&Cfg, &pThis->Out.Cfg.Props))

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

@@ -716 +716 @@
 
     LogRel2(("Audio: Creating stream '%s'\n", pStream->szName));
-    LogRel2(("Audio: Guest %s format for '%s': %RU32Hz, %u%s, %RU8 %s\n",
+    LogRel2(("Audio: Guest %s format for '%s': %RU32Hz, %u%s, %RU8 channel%s\n",
              pCfgGuest->enmDir == PDMAUDIODIR_IN ? "recording" : "playback", pStream->szName,
-             pCfgGuest->Props.uHz, pCfgGuest->Props.cbSample * 8, pCfgGuest->Props.fSigned ? "S" : "U",
-             pCfgGuest->Props.cChannels, pCfgGuest->Props.cChannels == 1 ? "Channel" : "Channels"));
-    LogRel2(("Audio: Requested host %s format for '%s': %RU32Hz, %u%s, %RU8 %s\n",
+             pCfgGuest->Props.uHz, PDMAudioPropsSampleBits(&pCfgGuest->Props), pCfgGuest->Props.fSigned ? "S" : "U",
+             PDMAudioPropsChannels(&pCfgGuest->Props), PDMAudioPropsChannels(&pCfgGuest->Props) == 1 ? "" : "s"));
+    LogRel2(("Audio: Requested host %s format for '%s': %RU32Hz, %u%s, %RU8 channel%s\n",
              pCfgHost->enmDir == PDMAUDIODIR_IN ? "recording" : "playback", pStream->szName,
-             pCfgHost->Props.uHz, pCfgHost->Props.cbSample * 8, pCfgHost->Props.fSigned ? "S" : "U",
-             pCfgHost->Props.cChannels, pCfgHost->Props.cChannels == 1 ? "Channel" : "Channels"));
+             pCfgHost->Props.uHz, PDMAudioPropsSampleBits(&pCfgHost->Props), pCfgHost->Props.fSigned ? "S" : "U",
+             PDMAudioPropsChannels(&pCfgHost->Props), PDMAudioPropsChannels(&pCfgHost->Props) == 1 ? "" : "s"));
 
     PDMAUDIOSTREAMCFG CfgHostAcq;
@@ -730 +730 @@
         return rc;
 
-#ifdef LOG_ENABLED
     LogFunc(("[%s] Acquired host format:\n",  pStream->szName));
     PDMAudioStrmCfgLog(&CfgHostAcq);
-#endif
-
-    LogRel2(("Audio: Acquired host %s format for '%s': %RU32Hz, %u%s, %RU8 %s\n",
+    LogRel2(("Audio: Acquired host %s format for '%s': %RU32Hz, %u%s, %RU8 channel%s\n",
              CfgHostAcq.enmDir == PDMAUDIODIR_IN ? "recording" : "playback",  pStream->szName,
-             CfgHostAcq.Props.uHz, CfgHostAcq.Props.cbSample * 8, CfgHostAcq.Props.fSigned ? "S" : "U",
-             CfgHostAcq.Props.cChannels, CfgHostAcq.Props.cChannels == 1 ? "Channel" : "Channels"));
+             CfgHostAcq.Props.uHz, PDMAudioPropsSampleBits(&CfgHostAcq.Props), CfgHostAcq.Props.fSigned ? "S" : "U",
+             PDMAudioPropsChannels(&CfgHostAcq.Props), PDMAudioPropsChannels(&CfgHostAcq.Props) == 1 ? "" : "s"));
+    Assert(PDMAudioPropsAreValid(&CfgHostAcq.Props));
 
     /* Let the user know if the backend changed some of the tweakable values. */
@@ -755 +753 @@
                  PDMAudioPropsFramesToMilli(&pCfgHost->Props, pCfgHost->Backend.cFramesPreBuffering), pCfgHost->Backend.cFramesPreBuffering,
                  PDMAudioPropsFramesToMilli(&CfgHostAcq.Props, CfgHostAcq.Backend.cFramesPreBuffering), CfgHostAcq.Backend.cFramesPreBuffering));
+
     /*
      * Configure host buffers.
@@ -801 +800 @@
     AudioMixBufDestroy(&pStream->Host.MixBuf);
 
-    /* Make sure to (re-)set the host buffer's shift size. */
-    CfgHostAcq.Props.cShift = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(CfgHostAcq.Props.cbSample, CfgHostAcq.Props.cChannels);
-
     rc = AudioMixBufInit(&pStream->Host.MixBuf, pStream->szName, &CfgHostAcq.Props, CfgHostAcq.Backend.cFramesBufferSize);
     AssertRC(rc);
@@ -825 +821 @@
     /* Set the guests's default audio data layout. */
     pCfgGuest->enmLayout = PDMAUDIOSTREAMLAYOUT_NON_INTERLEAVED;
-
-    /* Make sure to (re-)set the guest buffer's shift size. */
-    pCfgGuest->Props.cShift = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(pCfgGuest->Props.cbSample, pCfgGuest->Props.cChannels);
 
     rc = AudioMixBufInit(&pStream->Guest.MixBuf, pStream->szName, &pCfgGuest->Props, CfgHostAcq.Backend.cFramesBufferSize);
@@ -1448 +1441 @@
 
 /**
- * Worker for drvAudioStreamPlay that plays non-interleaved data.
+ * Worker for drvAudioStreamPlay that does the actual playing.
  *
  * @returns VBox status code.
  * @param   pThis           The audio driver instance data.
  * @param   pStream         The stream to play.
- * @param   cFramesToPlay   Number of audio frames to play.
+ * @param   cFramesToPlay   Number of audio frames to play.  The backend is
+ *                          supposed to have buffer space for this.
  * @param   pcFramesPlayed  Where to return the number of audio frames played.
  */
-static int drvAudioStreamPlayNonInterleaved(PDRVAUDIO pThis, PPDMAUDIOSTREAM pStream,
-                                            uint32_t cFramesToPlay, uint32_t *pcFramesPlayed)
+static int drvAudioStreamPlayDoIt(PDRVAUDIO pThis, PPDMAUDIOSTREAM pStream, uint32_t cFramesToPlay, uint32_t *pcFramesPlayed)
 {
     Assert(pStream->enmDir == PDMAUDIODIR_OUT);
-    Assert(pStream->Host.Cfg.enmLayout == PDMAUDIOSTREAMLAYOUT_NON_INTERLEAVED);
 
     /*
@@ -1490 +1482 @@
         if (RT_SUCCESS(rc))
         {
-            if (cbPlayed)
-            {
-                if (pThis->Out.Cfg.Dbg.fEnabled)
-                    AudioHlpFileWrite(pStream->Out.Dbg.pFilePlayNonInterleaved, abChunk, cbPlayed, 0 /* fFlags */);
-
-                if (cbRead != cbPlayed)
-                    LogRel2(("Audio: Host stream '%s' played wrong amount (%RU32 bytes read but played %RU32)\n",
-                             pStream->szName, cbRead, cbPlayed));
-
-                cFramesPlayed  = AUDIOMIXBUF_B2F(&pStream->Host.MixBuf, cbPlayed);
-                AssertStmt(cFramesLeft >= cFramesPlayed, cFramesPlayed = cFramesLeft);
-                cFramesLeft   -= cFramesPlayed;
-            }
-            else
-            {
-                /** @todo r=bird: If the backend is doing non-blocking writes, we'll probably
-                 *        be spinning like crazy here...  The ALSA backend is non-blocking.   */
-            }
+            if (pThis->Out.Cfg.Dbg.fEnabled)
+                AudioHlpFileWrite(pStream->Out.Dbg.pFilePlayNonInterleaved, abChunk, cbPlayed, 0 /* fFlags */);
+
+            if (cbRead != cbPlayed)
+                LogRel2(("Audio: Host stream '%s' played wrong amount (%RU32 bytes read but played %RU32)\n",
+                         pStream->szName, cbRead, cbPlayed));
+
+            cFramesPlayed  = AUDIOMIXBUF_B2F(&pStream->Host.MixBuf, cbPlayed);
+            AssertStmt(cFramesLeft >= cFramesPlayed, cFramesPlayed = cFramesLeft);
+            cFramesLeft   -= cFramesPlayed;
         }
 
@@ -1513 +1497 @@
 
         AssertRCBreak(rc); /* (this is here for Acquire/Release symmetry - which isn't at all necessary) */
+        AssertBreak(cbPlayed > 0); /* (ditto) */
     }
 
     Log3Func(("[%s] Played %RU32/%RU32 frames, rc=%Rrc\n", pStream->szName, cFramesToPlay - cFramesLeft, cFramesToPlay, rc));
     *pcFramesPlayed = cFramesToPlay - cFramesLeft;
-    return rc;
-}
-
-/**
- * Plays an audio host output stream which has been configured for raw audio (layout) data.
- *
- * @returns VBox status code.
- * @param   pThis               Pointer to driver instance.
- * @param   pStream             Stream to play.
- * @param   cFramesToPlay       Number of audio frames to play.
- * @param   pcFramesPlayed      Where to return number of audio frames played.
- */
-static int drvAudioStreamPlayRaw(PDRVAUDIO pThis, PPDMAUDIOSTREAM pStream, uint32_t cFramesToPlay, uint32_t *pcFramesPlayed)
-{
-    Assert(pStream->enmDir == PDMAUDIODIR_OUT);
-    Assert(pStream->Host.Cfg.enmLayout == PDMAUDIOSTREAMLAYOUT_RAW);
-
-    /*
-     * ...
-     */
-    int      rc          = VINF_SUCCESS;
-    uint32_t cFramesLeft = cFramesToPlay;
-    while (cFramesLeft > 0)
-    {
-        PDMAUDIOFRAME   aFrames[_2K];
-        uint32_t        cFramesRead = 0;
-        /** @todo r=bird: Peek functions do _not_ generally drop stuff from what they're
-         *        peeking into.  We normally name such functions Read.  (Oh man, am I
-         *        getting tired of this kind of crap.) */
-        rc = AudioMixBufPeek(&pStream->Host.MixBuf, cFramesLeft, aFrames,
-                             RT_MIN(cFramesLeft, RT_ELEMENTS(aFrames)), &cFramesRead);
-        if (RT_SUCCESS(rc))
-        {
-            Assert(cFramesRead <= RT_ELEMENTS(aFrames));
-            if (cFramesRead)
-            {
-                uint32_t cbPlayed = 0;
-                rc = pThis->pHostDrvAudio->pfnStreamPlay(pThis->pHostDrvAudio, pStream->pvBackend,
-                                                         aFrames, cFramesRead * sizeof(aFrames[0]), &cbPlayed);
-                AssertRCBreak(rc);
-                /** @todo r=bird: This is a natural follow up to the non-peeking peek crap
-                 *        above.  It works because VRDE is the only consumer and it always
-                 *        processes all that it gets. */
-                AssertBreakStmt(cbPlayed == cFramesRead * sizeof(aFrames[0]), rc = VERR_INTERNAL_ERROR_2);
-
-                Assert(cFramesRead <= cFramesLeft);
-                cFramesLeft -= cFramesRead;
-            }
-            else
-            {
-/** @todo r=bird: This looks totally loopy.  Why should another call return data
- *        when the first one didn't? */
-                if (rc == VINF_AUDIO_MORE_DATA_AVAILABLE) /* Do another peeking round if there is more data available. */
-                    continue;
-                break;
-            }
-        }
-        else
-            break;
-    }
-
-    *pcFramesPlayed = cFramesToPlay - cFramesLeft;
-    Log3Func(("[%s] Played %RU32/%RU32 frames, rc=%Rrc\n", pStream->szName, cFramesToPlay - cFramesLeft, cFramesToPlay, rc));
     return rc;
 }
@@ -1675 +1597 @@
         if (cFramesLive >= pStream->Host.Cfg.Backend.cFramesPreBuffering)
         {
-            LogRel2(("Audio: Stream '%s' buffering complete\n", pStream->szName));
-            Log3Func(("[%s] Dbg: Buffering complete!\n", pStream->szName));
+            LogRel2(("Audio: Stream '%s' buffering complete!\n", pStream->szName));
             pStream->fThresholdReached = fJustStarted = true;
         }
@@ -1690 +1611 @@
                  && (pStream->fStatus & PDMAUDIOSTREAMSTS_FLAGS_PENDING_DISABLE))
         {
-            LogRel2(("Audio: Stream '%s' buffering complete (short sound)\n", pStream->szName));
-            Log3Func(("[%s] Dbg: Buffering complete (short)!\n", pStream->szName));
+            LogRel2(("Audio: Stream '%s' buffering complete (short sound)!\n", pStream->szName));
             pStream->fThresholdReached = fJustStarted = true;
         }
@@ -1756 +1676 @@
          *        https://stackoverflow.com/questions/17879933/whats-the-interleaved-audio
          */
-        if (RT_LIKELY(pStream->Host.Cfg.enmLayout == PDMAUDIOSTREAMLAYOUT_NON_INTERLEAVED))
-            rc = drvAudioStreamPlayNonInterleaved(pThis, pStream, cFramesToPlay, pcFramesPlayed);
-        else if (pStream->Host.Cfg.enmLayout == PDMAUDIOSTREAMLAYOUT_RAW)
-            rc = drvAudioStreamPlayRaw(pThis, pStream, cFramesToPlay, pcFramesPlayed);
-        else
-            AssertFailedStmt(rc = VERR_NOT_IMPLEMENTED);
+        rc = drvAudioStreamPlayDoIt(pThis, pStream, cFramesToPlay, pcFramesPlayed);
 
         if (pThis->pHostDrvAudio->pfnStreamPlayEnd)
@@ -3085 +3000 @@
      * PCM
      */
-    if (pDrvCfg->Props.cbSample) /* Anything set via custom extra-data? */
-    {
-        pCfgReq->Props.cbSample = pDrvCfg->Props.cbSample;
-        LogRel2(("Audio: Using custom sample size of %RU8 bytes for stream '%s'\n", pCfgReq->Props.cbSample, pStream->szName));
+    if (PDMAudioPropsSampleSize(&pDrvCfg->Props) != 0) /* Anything set via custom extra-data? */
+    {
+        PDMAudioPropsSetSampleSize(&pCfgReq->Props, PDMAudioPropsSampleSize(&pDrvCfg->Props));
+        LogRel2(("Audio: Using custom sample size of %RU8 bytes for stream '%s'\n",
+                 PDMAudioPropsSampleSize(&pCfgReq->Props), pStream->szName));
     }
 
@@ -3111 +3027 @@
     }
 
-    if (pDrvCfg->Props.cChannels) /* Anything set via custom extra-data? */
-    {
-        pCfgReq->Props.cChannels = pDrvCfg->Props.cChannels;
-        LogRel2(("Audio: Using custom %RU8 channel(s) for stream '%s'\n", pCfgReq->Props.cChannels, pStream->szName));
-    }
-
-    /* Make sure to (re-)set the host buffer's shift size. */
-    pCfgReq->Props.cShift = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(pCfgReq->Props.cbSample, pCfgReq->Props.cChannels);
+    if (PDMAudioPropsChannels(&pDrvCfg->Props) != 0) /* Anything set via custom extra-data? */
+    {
+        PDMAudioPropsSetChannels(&pCfgReq->Props, PDMAudioPropsChannels(&pDrvCfg->Props));
+        LogRel2(("Audio: Using custom %RU8 channel(s) for stream '%s'\n", PDMAudioPropsChannels(&pDrvCfg->Props), pStream->szName));
+    }
 
     /* Validate PCM properties. */
@@ -3237 +3150 @@
 
     /* Validate acquired configuration. */
-    if (!AudioHlpStreamCfgIsValid(pCfgAcq))
-    {
-        LogRel(("Audio: Creating stream '%s' returned an invalid backend configuration, skipping\n", pStream->szName));
-        return VERR_INVALID_PARAMETER;
-    }
+    char szTmp[PDMAUDIOPROPSTOSTRING_MAX];
+    AssertLogRelMsgReturn(AudioHlpStreamCfgIsValid(pCfgAcq),
+                          ("Audio: Creating stream '%s' returned an invalid backend configuration (%s), skipping\n",
+                           pStream->szName, PDMAudioPropsToString(&pCfgAcq->Props, szTmp, sizeof(szTmp))),
+                          VERR_INVALID_PARAMETER);
 
     /* Let the user know that the backend changed one of the values requested above. */
@@ -3637 +3550 @@
     }
 
+    PDMDrvHlpSTAMDeregister(pDrvIns, &pThis->Out.StatsReBuffering);
 #ifdef VBOX_WITH_STATISTICS
     PDMDrvHlpSTAMDeregister(pDrvIns, &pThis->Stats.TotalStreamsActive);
@@ -3787 +3701 @@
         AssertRCReturn(rc, rc);
 
-        QUERY_VAL_RET(8,  "PCMSampleBit",        &pAudioCfg->Props.cbSample,    0,
-                         pAudioCfg->Props.cbSample == 0
-                      || pAudioCfg->Props.cbSample == 8
-                      || pAudioCfg->Props.cbSample == 16
-                      || pAudioCfg->Props.cbSample == 32
-                      || pAudioCfg->Props.cbSample == 64,
+        uint8_t cSampleBits = 0;
+        QUERY_VAL_RET(8,  "PCMSampleBit",        &cSampleBits,                  0,
+                         cSampleBits == 0
+                      || cSampleBits == 8
+                      || cSampleBits == 16
+                      || cSampleBits == 32
+                      || cSampleBits == 64,
                       "Must be either 0, 8, 16, 32 or 64");
-        pAudioCfg->Props.cbSample /= 8;
-
-        QUERY_VAL_RET(8,  "PCMSampleChannels",   &pAudioCfg->Props.cChannels,   0,
-                      pAudioCfg->Props.cChannels <= 16, "Max 16");
+        if (cSampleBits)
+            PDMAudioPropsSetSampleSize(&pAudioCfg->Props, cSampleBits / 8);
+
+        uint8_t cChannels;
+        QUERY_VAL_RET(8,  "PCMSampleChannels",   &cChannels,                    0, cChannels <= 16, "Max 16");
+        if (cChannels)
+            PDMAudioPropsSetChannels(&pAudioCfg->Props, cChannels);
 
         QUERY_VAL_RET(32, "PCMSampleHz",         &pAudioCfg->Props.uHz,         0,

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

@@ -149 +149 @@
 static snd_pcm_format_t alsaAudioPropsToALSA(PPDMAUDIOPCMPROPS pProps)
 {
-    switch (pProps->cbSample)
+    switch (PDMAudioPropsSampleSize(pProps))
     {
         case 1:
@@ -155 +155 @@
 
         case 2:
-            return pProps->fSigned ? SND_PCM_FORMAT_S16_LE : SND_PCM_FORMAT_U16_LE;
+            if (PDMAudioPropsIsLittleEndian(pProps))
+                return pProps->fSigned ? SND_PCM_FORMAT_S16_LE : SND_PCM_FORMAT_U16_LE;
+            return pProps->fSigned ? SND_PCM_FORMAT_S16_BE : SND_PCM_FORMAT_U16_BE;
 
         case 4:
-            return pProps->fSigned ? SND_PCM_FORMAT_S32_LE : SND_PCM_FORMAT_U32_LE;
+            if (PDMAudioPropsIsLittleEndian(pProps))
+                return pProps->fSigned ? SND_PCM_FORMAT_S32_LE : SND_PCM_FORMAT_U32_LE;
+            return pProps->fSigned ? SND_PCM_FORMAT_S32_BE : SND_PCM_FORMAT_U32_BE;
 
         default:
-            break;
-    }
-
-    AssertMsgFailed(("%RU8 bytes not supported\n", pProps->cbSample));
-    return SND_PCM_FORMAT_U8;
+            AssertMsgFailed(("%RU8 bytes not supported\n", PDMAudioPropsSampleSize(pProps)));
+            return SND_PCM_FORMAT_U8;
+    }
 }
 
@@ -173 +175 @@
  *
  * @returns VBox status code.
- * @param   fmt                 ALSA PCM format to convert.
- * @param   pProps              Where to store the converted PCM properties on success.
- */
-static int alsaALSAToAudioProps(snd_pcm_format_t fmt, PPDMAUDIOPCMPROPS pProps)
-{
+ * @param   pProps      Where to store the converted PCM properties on success.
+ * @param   fmt         ALSA PCM format to convert.
+ * @param   cChannels   Number of channels.
+ * @param   uHz         Frequency.
+ */
+static int alsaALSAToAudioProps(PPDMAUDIOPCMPROPS pProps, snd_pcm_format_t fmt, int cChannels, unsigned uHz)
+{
+    AssertReturn(cChannels > 0, VERR_INVALID_PARAMETER);
+    AssertReturn(cChannels < 16, VERR_INVALID_PARAMETER);
     switch (fmt)
     {
         case SND_PCM_FORMAT_S8:
-            pProps->cbSample    = 1;
-            pProps->fSigned     = true;
-            pProps->fSwapEndian = false;
+            PDMAudioPropsInit(pProps, 1 /*8-bit*/,  true /*signed*/, cChannels, uHz);
             break;
 
         case SND_PCM_FORMAT_U8:
-            pProps->cbSample    = 1;
-            pProps->fSigned     = false;
-            pProps->fSwapEndian = false;
+            PDMAudioPropsInit(pProps, 1 /*8-bit*/, false /*signed*/, cChannels, uHz);
             break;
 
         case SND_PCM_FORMAT_S16_LE:
-            pProps->cbSample    = 2;
-            pProps->fSigned     = true;
-            pProps->fSwapEndian = false;
+            PDMAudioPropsInitEx(pProps, 2 /*16-bit*/,  true /*signed*/, cChannels, uHz, true /*fLittleEndian*/, false /*fRaw*/);
             break;
 
         case SND_PCM_FORMAT_U16_LE:
-            pProps->cbSample    = 2;
-            pProps->fSigned     = false;
-            pProps->fSwapEndian = false;
+            PDMAudioPropsInitEx(pProps, 2 /*16-bit*/, false /*signed*/, cChannels, uHz, true /*fLittleEndian*/, false /*fRaw*/);
             break;
 
         case SND_PCM_FORMAT_S16_BE:
-            pProps->cbSample    = 2;
-            pProps->fSigned     = true;
-#ifdef RT_LITTLE_ENDIAN
-            pProps->fSwapEndian = true;
-#endif
+            PDMAudioPropsInitEx(pProps, 2 /*16-bit*/,  true /*signed*/, cChannels, uHz, false /*fLittleEndian*/, false /*fRaw*/);
             break;
 
         case SND_PCM_FORMAT_U16_BE:
-            pProps->cbSample    = 2;
-            pProps->fSigned     = false;
-#ifdef RT_LITTLE_ENDIAN
-            pProps->fSwapEndian = true;
-#endif
+            PDMAudioPropsInitEx(pProps, 2 /*16-bit*/, false /*signed*/, cChannels, uHz, false /*fLittleEndian*/, false /*fRaw*/);
             break;
 
         case SND_PCM_FORMAT_S32_LE:
-            pProps->cbSample    = 4;
-            pProps->fSigned     = true;
-            pProps->fSwapEndian = false;
+            PDMAudioPropsInitEx(pProps, 4 /*32-bit*/,  true /*signed*/, cChannels, uHz, true /*fLittleEndian*/, false /*fRaw*/);
             break;
 
         case SND_PCM_FORMAT_U32_LE:
-            pProps->cbSample    = 4;
-            pProps->fSigned     = false;
-            pProps->fSwapEndian = false;
+            PDMAudioPropsInitEx(pProps, 4 /*32-bit*/, false /*signed*/, cChannels, uHz, true /*fLittleEndian*/, false /*fRaw*/);
             break;
 
         case SND_PCM_FORMAT_S32_BE:
-            pProps->cbSample    = 4;
-            pProps->fSigned     = true;
-#ifdef RT_LITTLE_ENDIAN
-            pProps->fSwapEndian = true;
-#endif
+            PDMAudioPropsInitEx(pProps, 4 /*32-bit*/,  true /*signed*/, cChannels, uHz, false /*fLittleEndian*/, false /*fRaw*/);
             break;
 
         case SND_PCM_FORMAT_U32_BE:
-            pProps->cbSample    = 4;
-            pProps->fSigned     = false;
-#ifdef RT_LITTLE_ENDIAN
-            pProps->fSwapEndian = true;
-#endif
+            PDMAudioPropsInitEx(pProps, 4 /*32-bit*/, false /*signed*/, cChannels, uHz, false /*fLittleEndian*/, false /*fRaw*/);
             break;
 
@@ -251 +229 @@
             AssertMsgFailedReturn(("Format %d not supported\n", fmt), VERR_NOT_SUPPORTED);
     }
-
-    AssertReturn(pProps->cbSample > 0,  VERR_NOT_SUPPORTED);
-    AssertReturn(pProps->cChannels > 0, VERR_INVALID_PARAMETER);
-
-    pProps->cShift = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(pProps->cbSample, pProps->cChannels);
-
     return VINF_SUCCESS;
 }
@@ -937 +909 @@
         ALSAAUDIOSTREAMCFG req;
         req.fmt         = alsaAudioPropsToALSA(&pCfgReq->Props);
-        req.freq        = pCfgReq->Props.uHz;
-        req.nchannels   = pCfgReq->Props.cChannels;
+        req.freq        = PDMAudioPropsHz(&pCfgReq->Props);
+        req.nchannels   = PDMAudioPropsChannels(&pCfgReq->Props);
         req.period_size = pCfgReq->Backend.cFramesPeriod;
         req.buffer_size = pCfgReq->Backend.cFramesBufferSize;
@@ -948 +920 @@
             break;
 
-        pCfgAcq->Props.uHz       = obt.freq;
-        pCfgAcq->Props.cChannels = obt.nchannels;
-
-        rc = alsaALSAToAudioProps(obt.fmt, &pCfgAcq->Props);
+        rc = alsaALSAToAudioProps(&pCfgAcq->Props, obt.fmt, obt.nchannels, obt.freq);
         if (RT_FAILURE(rc))
             break;
@@ -1000 +969 @@
         ALSAAUDIOSTREAMCFG req;
         req.fmt         = alsaAudioPropsToALSA(&pCfgReq->Props);
-        req.freq        = pCfgReq->Props.uHz;
-        req.nchannels   = pCfgReq->Props.cChannels;
+        req.freq        = PDMAudioPropsHz(&pCfgReq->Props);
+        req.nchannels   = PDMAudioPropsChannels(&pCfgReq->Props);
         req.period_size = PDMAudioPropsMilliToFrames(&pCfgReq->Props, 50 /*ms*/); /** @todo Make this configurable. */
         req.buffer_size = req.period_size * 2; /** @todo Make this configurable. */
@@ -1011 +980 @@
             break;
 
-        pCfgAcq->Props.uHz       = obt.freq;
-        pCfgAcq->Props.cChannels = obt.nchannels;
-
-        rc = alsaALSAToAudioProps(obt.fmt, &pCfgAcq->Props);
+        rc = alsaALSAToAudioProps(&pCfgAcq->Props, obt.fmt, obt.nchannels, obt.freq);
         if (RT_FAILURE(rc))
             break;

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

@@ -150 +150 @@
     char pszSampleRate[32];
     LogRel2(("CoreAudio: %s description:\n", pszDesc));
-    LogRel2(("CoreAudio:\tFormat ID: %RU32 (%c%c%c%c)\n", pASBD->mFormatID,
+    LogRel2(("CoreAudio:  Format ID: %RU32 (%c%c%c%c)\n", pASBD->mFormatID,
              RT_BYTE4(pASBD->mFormatID), RT_BYTE3(pASBD->mFormatID),
              RT_BYTE2(pASBD->mFormatID), RT_BYTE1(pASBD->mFormatID)));
-    LogRel2(("CoreAudio:\tFlags: %RU32", pASBD->mFormatFlags));
+    LogRel2(("CoreAudio:  Flags: %RU32", pASBD->mFormatFlags));
     if (pASBD->mFormatFlags & kAudioFormatFlagIsFloat)
         LogRel2((" Float"));
@@ -171 +171 @@
         LogRel2((" AllClear"));
     LogRel2(("\n"));
-    snprintf(pszSampleRate, 32, "%.2f", (float)pASBD->mSampleRate); /** @todo r=andy Use RTStrPrint*. */
-    LogRel2(("CoreAudio:\tSampleRate      : %s\n", pszSampleRate));
-    LogRel2(("CoreAudio:\tChannelsPerFrame: %RU32\n", pASBD->mChannelsPerFrame));
-    LogRel2(("CoreAudio:\tFramesPerPacket : %RU32\n", pASBD->mFramesPerPacket));
-    LogRel2(("CoreAudio:\tBitsPerChannel  : %RU32\n", pASBD->mBitsPerChannel));
-    LogRel2(("CoreAudio:\tBytesPerFrame   : %RU32\n", pASBD->mBytesPerFrame));
-    LogRel2(("CoreAudio:\tBytesPerPacket  : %RU32\n", pASBD->mBytesPerPacket));
-}
-
-static void coreAudioPCMPropsToASBD(PDMAUDIOPCMPROPS *pPCMProps, AudioStreamBasicDescription *pASBD)
-{
-    AssertPtrReturnVoid(pPCMProps);
+    LogRel2(("CoreAudio:  SampleRate      : %RU64.%02u Hz\n",
+             (uint64_t)pASBD->mSampleRate, (unsigned)(pASBD->mSampleRate * 100) % 100));
+    LogRel2(("CoreAudio:  ChannelsPerFrame: %RU32\n", pASBD->mChannelsPerFrame));
+    LogRel2(("CoreAudio:  FramesPerPacket : %RU32\n", pASBD->mFramesPerPacket));
+    LogRel2(("CoreAudio:  BitsPerChannel  : %RU32\n", pASBD->mBitsPerChannel));
+    LogRel2(("CoreAudio:  BytesPerFrame   : %RU32\n", pASBD->mBytesPerFrame));
+    LogRel2(("CoreAudio:  BytesPerPacket  : %RU32\n", pASBD->mBytesPerPacket));
+}
+
+static void coreAudioPCMPropsToASBD(PCPDMAUDIOPCMPROPS pProps, AudioStreamBasicDescription *pASBD)
+{
+    AssertPtrReturnVoid(pProps);
     AssertPtrReturnVoid(pASBD);
 
@@ -189 +189 @@
     pASBD->mFormatID         = kAudioFormatLinearPCM;
     pASBD->mFormatFlags      = kAudioFormatFlagIsPacked;
+    if (pProps->fSigned)
+        pASBD->mFormatFlags |= kAudioFormatFlagIsSignedInteger;
+    if (PDMAudioPropsIsBigEndian(pProps))
+        pASBD->mFormatFlags |= kAudioFormatFlagIsBigEndian;
+    pASBD->mSampleRate       = PDMAudioPropsHz(pProps);
+    pASBD->mChannelsPerFrame = PDMAudioPropsChannels(pProps);
+    pASBD->mBitsPerChannel   = PDMAudioPropsSampleBits(pProps);
+    pASBD->mBytesPerFrame    = PDMAudioPropsFrameSize(pProps);
     pASBD->mFramesPerPacket  = 1; /* For uncompressed audio, set this to 1. */
-    pASBD->mSampleRate       = (Float64)pPCMProps->uHz;
-    pASBD->mChannelsPerFrame = pPCMProps->cChannels;
-    pASBD->mBitsPerChannel   = pPCMProps->cbSample * 8;
-    if (pPCMProps->fSigned)
-        pASBD->mFormatFlags |= kAudioFormatFlagIsSignedInteger;
-    pASBD->mBytesPerFrame    = pASBD->mChannelsPerFrame * (pASBD->mBitsPerChannel / 8);
-    pASBD->mBytesPerPacket   = pASBD->mFramesPerPacket * pASBD->mBytesPerFrame;
+    pASBD->mBytesPerPacket   = PDMAudioPropsFrameSize(pProps) * pASBD->mFramesPerPacket;
 }
 
@@ -205 +207 @@
     AssertPtrReturn(pCfg,  VERR_INVALID_PARAMETER);
 
-    pCfg->Props.cChannels = pASBD->mChannelsPerFrame;
-    pCfg->Props.uHz       = (uint32_t)pASBD->mSampleRate;
-    AssertMsg(!(pASBD->mBitsPerChannel & 7), ("%u\n", pASBD->mBitsPerChannel));
-    pCfg->Props.cbSample  = pASBD->mBitsPerChannel / 8;
-    pCfg->Props.fSigned   = RT_BOOL(pASBD->mFormatFlags & kAudioFormatFlagIsSignedInteger);
-    pCfg->Props.cShift    = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(pCfg->Props.cbSample, pCfg->Props.cChannels);
-    /** @todo r=bird: pCfg->Props.fSwapEndian is not initialized here!  */
-
+    AssertLogRelMsgReturn(pASBD->mChannelsPerFrame > 0, VERR_NOT_SUPPORTED);
+    AssertLogRelMsgReturn(pASBD->mChannelsPerFrame < 16, VERR_NOT_SUPPORTED);
+    AssertLogRelMsgReturn(   pASBD->mBitsPerChannel == 8
+                          || pASBD->mBitsPerChannel == 16
+                          || pASBD->mBitsPerChannel == 32,
+                          ("%u\n", pASBD->mBitsPerChannel), VERR_NOT_SUPPORTED);
+    AssertLogRelMsgReturn(!(pASBD->mFormatFlags & (kAudioFormatFlagIsFloat /** @todo more we don't like?*/)),
+                          ("%#x\n", pASBD->mFormatFlags), VERR_NOT_SUPPORTED);
+
+    PDMAudioPropsInitEx(&pCfg->Props,
+                        pASBD->mBitsPerChannel / 8,
+                        RT_BOOL(pASBD->mFormatFlags & kAudioFormatFlagIsSignedInteger),
+                        pASBD->mChannelsPerFrame,
+                        (uint32_t)pASBD->mSampleRate,
+                        RT_BOOL(pASBD->mFormatFlags & kAudioFormatFlagIsBigEndian),
+                        false /*fRaw*/);
     return VINF_SUCCESS;
 }

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

@@ -269 +269 @@
 
     pFmt->wFormatTag      = WAVE_FORMAT_PCM;
-    pFmt->nChannels       = pCfg->Props.cChannels;
-    pFmt->wBitsPerSample  = pCfg->Props.cbSample * 8;
-    pFmt->nSamplesPerSec  = pCfg->Props.uHz;
-    pFmt->nBlockAlign     = pFmt->nChannels * pFmt->wBitsPerSample / 8;
-    pFmt->nAvgBytesPerSec = pFmt->nSamplesPerSec * pFmt->nBlockAlign;
+    pFmt->nChannels       = PDMAudioPropsChannels(&pCfg->Props);
+    pFmt->wBitsPerSample  = PDMAudioPropsSampleBits(&pCfg->Props);
+    pFmt->nSamplesPerSec  = PDMAudioPropsHz(&pCfg->Props);
+    pFmt->nBlockAlign     = PDMAudioPropsFrameSize(&pCfg->Props);
+    pFmt->nAvgBytesPerSec = PDMAudioPropsFramesToBytes(&pCfg->Props, PDMAudioPropsHz(&pCfg->Props));
     pFmt->cbSize          = 0; /* No extra data specified. */
 
@@ -614 +614 @@
     AssertPtrReturn(pCfgAcq,   E_POINTER);
 
+/** @todo r=bird: I cannot see any code populating pCfgAcq... */
+
     LogFlowFuncEnter();
 
     Assert(pStreamDS->Out.pDSB == NULL);
 
-    DSLOG(("DSound: Opening playback stream (uHz=%RU32, cChannels=%RU8, cBits=%u, fSigned=%RTbool)\n",
-           pCfgReq->Props.uHz, pCfgReq->Props.cChannels, pCfgReq->Props.cbSample * 8, pCfgReq->Props.fSigned));
+    DSLOG(("DSound: Opening playback stream (uHz=%RU32, cChannels=%RU8, cBits=%u, fSigned=%RTbool)\n", pCfgReq->Props.uHz,
+           PDMAudioPropsChannels(&pCfgReq->Props), PDMAudioPropsSampleBits(&pCfgReq->Props), pCfgReq->Props.fSigned));
 
     WAVEFORMATEX wfx;
@@ -1369 +1371 @@
     AssertPtrReturn(pCfgAcq,   E_POINTER);
 
+    /** @todo r=bird: I cannot see any code populating pCfgAcq... */
+
     LogFlowFuncEnter();
 
     Assert(pStreamDS->In.pDSCB == NULL);
 
-    DSLOG(("DSound: Opening capturing stream (uHz=%RU32, cChannels=%RU8, cBits=%u, fSigned=%RTbool)\n",
-           pCfgReq->Props.uHz, pCfgReq->Props.cChannels, pCfgReq->Props.cbSample * 8, pCfgReq->Props.fSigned));
+    DSLOG(("DSound: Opening capturing stream (uHz=%RU32, cChannels=%RU8, cBits=%u, fSigned=%RTbool)\n", pCfgReq->Props.uHz,
+           PDMAudioPropsChannels(&pCfgReq->Props), PDMAudioPropsSampleBits(&pCfgReq->Props), pCfgReq->Props.fSigned));
 
     WAVEFORMATEX wfx;
@@ -2067 +2071 @@
              pStreamDS, pCfgReq, PDMAudioRecSrcGetName(pCfgReq->u.enmSrc)));
 
-    int rc = VINF_SUCCESS;
 
     /* Try to open capture in case the device is already there. */
+    int rc;
     HRESULT hr = directSoundCaptureOpen(pThis, pStreamDS, pCfgReq, pCfgAcq);
     if (SUCCEEDED(hr))
@@ -2363 +2367 @@
     if (RT_SUCCESS(rc))
     {
+        /** @todo already copied   */
         rc = PDMAudioStrmCfgCopy(&pStreamDS->Cfg, pCfgAcq);
         if (RT_SUCCESS(rc))

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

@@ -16 +16 @@
  */
 
-#include <iprt/alloc.h>
+#include <iprt/mem.h>
 #include <iprt/rand.h>
 #include <iprt/uuid.h> /* For PDMIBASE_2_PDMDRV. */
@@ -211 +211 @@
  */
 static DECLCALLBACK(int) drvHostDebugAudioHA_StreamPlay(PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream,
-                                                        const void *pvBuf, uint32_t uBufSize, uint32_t *puWritten)
-{
-    RT_NOREF(pInterface);
-    PDEBUGAUDIOSTREAM  pStreamDbg = (PDEBUGAUDIOSTREAM)pStream;
-
-    int rc = AudioHlpFileWrite(pStreamDbg->pFile, pvBuf, uBufSize, 0 /* fFlags */);
+                                                        const void *pvBuf, uint32_t cbBuf, uint32_t *pcbWritten)
+{
+    PDEBUGAUDIOSTREAM pStreamDbg = (PDEBUGAUDIOSTREAM)pStream;
+    RT_NOREF(pInterface);
+
+    int rc = AudioHlpFileWrite(pStreamDbg->pFile, pvBuf, cbBuf, 0 /* fFlags */);
     if (RT_FAILURE(rc))
     {
@@ -223 +223 @@
     }
 
-    if (puWritten)
-        *puWritten = uBufSize;
-
+    *pcbWritten = cbBuf;
     return VINF_SUCCESS;
 }
@@ -243 +241 @@
     AssertPtr(pCfg);
 
-    Assert(uBufSize % pCfg->Props.cbSample == 0);
+    Assert(uBufSize % PDMAudioPropsSampleSize(&pCfg->Props) == 0);
+    size_t const cSamples = uBufSize / PDMAudioPropsSampleSize(&pCfg->Props);
 
     uint16_t *paBuf = (uint16_t *)pvBuf;
 
     /* Generate a simple mono sine wave. */
-    for (size_t i = 0; i < uBufSize / pCfg->Props.cbSample; i++)
+    for (size_t i = 0; i < cSamples; i++)
     {
         paBuf[i] = 32760 * sin((2.f * float(3.1415) * pStreamDbg->In.uFreqHz) / pCfg->Props.uHz * pStreamDbg->In.uSample);

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

@@ -151 +151 @@
 
 
-static int ossOSSToAudioProps(int fmt, PPDMAUDIOPCMPROPS pProps)
-{
-    RT_BZERO(pProps, sizeof(PDMAUDIOPCMPROPS));
-
-    /** @todo r=bird: What's the assumption about the incoming pProps?  Code is
-     *        clearly ASSUMING something about how it's initialized, but even so,
-     *        the fSwapEndian isn't correct in a portable way. */
+static int ossOSSToAudioProps(PPDMAUDIOPCMPROPS pProps, int fmt, int cChannels, int uHz)
+{
     switch (fmt)
     {
         case AFMT_S8:
-            pProps->cbSample    = 1;
-            pProps->fSigned     = true;
+            PDMAudioPropsInit(pProps, 1 /*8-bit*/, true /*signed*/, cChannels, uHz);
             break;
 
         case AFMT_U8:
-            pProps->cbSample    = 1;
-            pProps->fSigned     = false;
+            PDMAudioPropsInit(pProps, 1 /*8-bit*/, false /*signed*/, cChannels, uHz);
             break;
 
         case AFMT_S16_LE:
-            pProps->cbSample    = 2;
-            pProps->fSigned     = true;
+            PDMAudioPropsInitEx(pProps, 2 /*16-bit*/, true /*signed*/, cChannels, uHz, true /*fLittleEndian*/, false /*fRaw*/);
             break;
 
         case AFMT_U16_LE:
-            pProps->cbSample    = 2;
-            pProps->fSigned     = false;
-            break;
-
-       case AFMT_S16_BE:
-            pProps->cbSample    = 2;
-            pProps->fSigned     = true;
-#ifdef RT_LITTLE_ENDIAN
-            pProps->fSwapEndian = true;
-#endif
+            PDMAudioPropsInitEx(pProps, 2 /*16-bit*/, false /*signed*/, cChannels, uHz, true /*fLittleEndian*/, false /*fRaw*/);
+            break;
+
+        case AFMT_S16_BE:
+            PDMAudioPropsInitEx(pProps, 2 /*16-bit*/, true /*signed*/, cChannels, uHz, false /*fLittleEndian*/, false /*fRaw*/);
             break;
 
         case AFMT_U16_BE:
-            pProps->cbSample    = 2;
-            pProps->fSigned     = false;
-#ifdef RT_LITTLE_ENDIAN
-            pProps->fSwapEndian = true;
-#endif
+            PDMAudioPropsInitEx(pProps, 2 /*16-bit*/, false /*signed*/, cChannels, uHz, false /*fLittleEndian*/, false /*fRaw*/);
             break;
 
@@ -240 +223 @@
 
         int iFormat;
-        switch (pOSSReq->Props.cbSample)
+        switch (PDMAudioPropsSampleSize(&pOSSReq->Props))
         {
             case 1:
@@ -266 +249 @@
         }
 
-        int cChannels = pOSSReq->Props.cChannels;
+        int cChannels = PDMAudioPropsChannels(&pOSSReq->Props);
         if (ioctl(fdFile, SNDCTL_DSP_CHANNELS, &cChannels))
         {
             LogRel(("OSS: Failed to set number of audio channels (%RU8): %s (%d)\n",
-                    pOSSReq->Props.cChannels, strerror(errno), errno));
+                    PDMAudioPropsChannels(&pOSSReq->Props), strerror(errno), errno));
             break;
         }
@@ -311 +294 @@
         }
 
-        rc = ossOSSToAudioProps(iFormat, &pOSSAcq->Props);
+        rc = ossOSSToAudioProps(&pOSSAcq->Props, iFormat, cChannels, freq);
         if (RT_SUCCESS(rc))
         {
-            pOSSAcq->Props.cChannels = cChannels;
-            pOSSAcq->Props.uHz       = freq;
-            pOSSAcq->Props.cShift    = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(pOSSAcq->Props.cbSample, pOSSAcq->Props.cChannels);
-
             pOSSAcq->cFragments      = abinfo.fragstotal;
             pOSSAcq->cbFragmentSize  = abinfo.fragsize;

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

@@ -217 +217 @@
 static pa_sample_format_t paAudioPropsToPulse(PPDMAUDIOPCMPROPS pProps)
 {
-    switch (pProps->cbSample)
+    switch (PDMAudioPropsSampleSize(pProps))
     {
         case 1:
@@ -226 +226 @@
         case 2:
             if (pProps->fSigned)
-                return PA_SAMPLE_S16LE;
+                return PDMAudioPropsIsLittleEndian(pProps) ? PA_SAMPLE_S16LE : PA_SAMPLE_S16BE;
             break;
 
@@ -232 +232 @@
         case 4:
             if (pProps->fSigned)
-                return PA_SAMPLE_S32LE;
+                return PDMAudioPropsIsLittleEndian(pProps) ? PA_SAMPLE_S32LE : PA_SAMPLE_S32BE;
             break;
 #endif
-
-        default:
-            break;
-    }
-
-    AssertMsgFailed(("%RU8%s not supported\n", pProps->cbSample, pProps->fSigned ? "S" : "U"));
+    }
+
+    AssertMsgFailed(("%RU8%s not supported\n", PDMAudioPropsSampleSize(pProps), pProps->fSigned ? "S" : "U"));
     return PA_SAMPLE_INVALID;
 }
 
 
-static int paPulseToAudioProps(pa_sample_format_t pulsefmt, PPDMAUDIOPCMPROPS pProps)
-{
-    /** @todo r=bird: You are assuming undocumented stuff about
-     *        pProps->fSwapEndian. */
+static int paPulseToAudioProps(PPDMAUDIOPCMPROPS pProps, pa_sample_format_t pulsefmt, uint8_t cChannels, uint32_t uHz)
+{
+    AssertReturn(cChannels > 0, VERR_INVALID_PARAMETER);
+    AssertReturn(cChannels < 16, VERR_INVALID_PARAMETER);
+
     switch (pulsefmt)
     {
         case PA_SAMPLE_U8:
-            pProps->cbSample    = 1;
-            pProps->fSigned     = false;
+            PDMAudioPropsInit(pProps, 1 /*8-bit*/, false /*signed*/, cChannels, uHz);
             break;
 
         case PA_SAMPLE_S16LE:
-            pProps->cbSample    = 2;
-            pProps->fSigned     = true;
+            PDMAudioPropsInitEx(pProps, 2 /*16-bit*/, true /*signed*/, cChannels, uHz, true /*fLittleEndian*/, false /*fRaw*/);
             break;
 
         case PA_SAMPLE_S16BE:
-            pProps->cbSample    = 2;
-            pProps->fSigned     = true;
-            /** @todo Handle Endianess. */
+            PDMAudioPropsInitEx(pProps, 2 /*16-bit*/, true /*signed*/, cChannels, uHz, false /*fLittleEndian*/, false /*fRaw*/);
             break;
 
 #ifdef PA_SAMPLE_S32LE
         case PA_SAMPLE_S32LE:
-            pProps->cbSample    = 4;
-            pProps->fSigned     = true;
+            PDMAudioPropsInitEx(pProps, 4 /*32-bit*/, true /*signed*/, cChannels, uHz, true /*fLittleEndian*/, false /*fRaw*/);
             break;
 #endif
@@ -276 +269 @@
 #ifdef PA_SAMPLE_S32BE
         case PA_SAMPLE_S32BE:
-            pProps->cbSample    = 4;
-            pProps->fSigned     = true;
-            /** @todo Handle Endianess. */
+            PDMAudioPropsInitEx(pProps, 4 /*32-bit*/, true /*signed*/, cChannels, uHz, false /*fLittleEndian*/, false /*fRaw*/);
             break;
 #endif
 
         default:
-            AssertLogRelMsgFailed(("PulseAudio: Format (%ld) not supported\n", pulsefmt));
+            AssertLogRelMsgFailed(("PulseAudio: Format (%d) not supported\n", pulsefmt));
             return VERR_NOT_SUPPORTED;
     }
@@ -789 +780 @@
 
     pStreamPA->SampleSpec.format   = paAudioPropsToPulse(&pCfgReq->Props);
-    pStreamPA->SampleSpec.rate     = pCfgReq->Props.uHz;
-    pStreamPA->SampleSpec.channels = pCfgReq->Props.cChannels;
+    pStreamPA->SampleSpec.rate     = PDMAudioPropsHz(&pCfgReq->Props);
+    pStreamPA->SampleSpec.channels = PDMAudioPropsChannels(&pCfgReq->Props);
 
     pStreamPA->curLatencyUs        = PDMAudioPropsFramesToMilli(&pCfgReq->Props, pCfgReq->Backend.cFramesBufferSize) * RT_US_1MS;
@@ -816 +807 @@
         return rc;
 
-    rc = paPulseToAudioProps(pStreamPA->SampleSpec.format, &pCfgAcq->Props);
+    rc = paPulseToAudioProps(&pCfgAcq->Props, pStreamPA->SampleSpec.format,
+                             pStreamPA->SampleSpec.channels, pStreamPA->SampleSpec.rate);
     if (RT_FAILURE(rc))
     {
@@ -822 +814 @@
         return rc;
     }
-
-    pCfgAcq->Props.uHz       = pStreamPA->SampleSpec.rate;
-    pCfgAcq->Props.cChannels = pStreamPA->SampleSpec.channels;
-    pCfgAcq->Props.cShift    = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(pCfgAcq->Props.cbSample, pCfgAcq->Props.cChannels);
 
     LogFunc(("Acquired: BufAttr tlength=%RU32, maxLength=%RU32, minReq=%RU32\n",
@@ -844 +832 @@
 {
     pStreamPA->SampleSpec.format   = paAudioPropsToPulse(&pCfgReq->Props);
-    pStreamPA->SampleSpec.rate     = pCfgReq->Props.uHz;
-    pStreamPA->SampleSpec.channels = pCfgReq->Props.cChannels;
+    pStreamPA->SampleSpec.rate     = PDMAudioPropsHz(&pCfgReq->Props);
+    pStreamPA->SampleSpec.channels = PDMAudioPropsChannels(&pCfgReq->Props);
 
     pStreamPA->BufAttr.fragsize    = PDMAudioPropsFramesToBytes(&pCfgReq->Props, pCfgReq->Backend.cFramesPeriod);
@@ -860 +848 @@
         return rc;
 
-    rc = paPulseToAudioProps(pStreamPA->SampleSpec.format, &pCfgAcq->Props);
+    rc = paPulseToAudioProps(&pCfgAcq->Props, pStreamPA->SampleSpec.format,
+                             pStreamPA->SampleSpec.channels, pStreamPA->SampleSpec.rate);
     if (RT_FAILURE(rc))
     {
@@ -869 +858 @@
     pStreamPA->pDrv       = pThis;
     pStreamPA->pu8PeekBuf = NULL;
-
-    pCfgAcq->Props.uHz         = pStreamPA->SampleSpec.rate;
-    pCfgAcq->Props.cChannels   = pStreamPA->SampleSpec.channels;
 
     pCfgAcq->Backend.cFramesPeriod       = PDMAUDIOSTREAMCFG_B2F(pCfgAcq, pStreamPA->BufAttr.fragsize);

trunk/src/VBox/Devices/Audio/testcase/tstAudioMixBuffer.cpp

@@ -38 +38 @@
     RTTestSub(hTest, "Basics");
 
-    const PDMAUDIOPCMPROPS Cfg441StereoS16 = PDMAUDIOPCMPROPS_INITIALIZOR(
+    const PDMAUDIOPCMPROPS Cfg441StereoS16 = PDMAUDIOPCMPROPS_INITIALIZER(
         /* a_cb: */             2,
         /* a_fSigned: */        true,
         /* a_cChannels: */      2,
         /* a_uHz: */            44100,
-        /* a_cShift: */         PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(2 /* cb */, 2 /* cChannels */),
         /* a_fSwapEndian: */    false
     );
-    const PDMAUDIOPCMPROPS Cfg441StereoU16 = PDMAUDIOPCMPROPS_INITIALIZOR(
+    const PDMAUDIOPCMPROPS Cfg441StereoU16 = PDMAUDIOPCMPROPS_INITIALIZER(
         /* a_cb: */             2,
         /* a_fSigned: */        false,
         /* a_cChannels: */      2,
         /* a_uHz: */            44100,
-        /* a_cShift: */         PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(2 /* cb */, 2 /* cChannels */),
         /* a_fSwapEndian: */    false
     );
-    const PDMAUDIOPCMPROPS Cfg441StereoU32 = PDMAUDIOPCMPROPS_INITIALIZOR(
+    const PDMAUDIOPCMPROPS Cfg441StereoU32 = PDMAUDIOPCMPROPS_INITIALIZER(
         /* a_cb: */             4,
         /* a_fSigned: */        false,
         /* a_cChannels: */      2,
         /* a_uHz: */            44100,
-        /* a_cShift: */         PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(4 /* cb */, 2 /* cChannels */),
         /* a_fSwapEndian: */    false
     );
@@ -191 +188 @@
 
     /* 44100Hz, 2 Channels, S16 */
-    PDMAUDIOPCMPROPS config = PDMAUDIOPCMPROPS_INITIALIZOR(
+    PDMAUDIOPCMPROPS config = PDMAUDIOPCMPROPS_INITIALIZER(
         2,                                                                  /* Bytes */
         true,                                                               /* Signed */
         2,                                                                  /* Channels */
         44100,                                                              /* Hz */
-        PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(2 /* Bytes */, 2 /* Channels */), /* Shift */
         false                                                               /* Swap Endian */
     );
@@ -312 +308 @@
 
     /* 44100Hz, 2 Channels, S16 */
-    PDMAUDIOPCMPROPS cfg_p = PDMAUDIOPCMPROPS_INITIALIZOR(
+    PDMAUDIOPCMPROPS cfg_p = PDMAUDIOPCMPROPS_INITIALIZER(
         2,                                                                  /* Bytes */
         true,                                                               /* Signed */
         2,                                                                  /* Channels */
         44100,                                                              /* Hz */
-        PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(2 /* Bytes */, 2 /* Channels */), /* Shift */
         false                                                               /* Swap Endian */
     );
@@ -327 +322 @@
 
     /* 22050Hz, 2 Channels, S16 */
-    PDMAUDIOPCMPROPS cfg_c1 = PDMAUDIOPCMPROPS_INITIALIZOR(/* Upmixing to parent */
+    PDMAUDIOPCMPROPS cfg_c1 = PDMAUDIOPCMPROPS_INITIALIZER(/* Upmixing to parent */
         2,                                                                  /* Bytes */
         true,                                                               /* Signed */
         2,                                                                  /* Channels */
         22050,                                                              /* Hz */
-        PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(2 /* Bytes */, 2 /* Channels */), /* Shift */
         false                                                               /* Swap Endian */
     );
@@ -346 +340 @@
 
     /* 48000Hz, 2 Channels, S16 */
-    PDMAUDIOPCMPROPS cfg_c2 = PDMAUDIOPCMPROPS_INITIALIZOR(/* Downmixing to parent */
+    PDMAUDIOPCMPROPS cfg_c2 = PDMAUDIOPCMPROPS_INITIALIZER(/* Downmixing to parent */
         2,                                                                  /* Bytes */
         true,                                                               /* Signed */
         2,                                                                  /* Channels */
         48000,                                                              /* Hz */
-        PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(2 /* Bytes */, 2 /* Channels */), /* Shift */
         false                                                               /* Swap Endian */
     );
@@ -460 +453 @@
 
     /* 44100Hz, 1 Channel, U8 */
-    PDMAUDIOPCMPROPS cfg_p = PDMAUDIOPCMPROPS_INITIALIZOR(
+    PDMAUDIOPCMPROPS cfg_p = PDMAUDIOPCMPROPS_INITIALIZER(
         1,                                                                  /* Bytes */
         false,                                                              /* Signed */
         1,                                                                  /* Channels */
         44100,                                                              /* Hz */
-        PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(1 /* Bytes */, 1 /* Channels */), /* Shift */
         false                                                               /* Swap Endian */
     );
@@ -483 +475 @@
 
     /* 22050Hz, 1 Channel, U8 */
-    PDMAUDIOPCMPROPS cfg_c = PDMAUDIOPCMPROPS_INITIALIZOR( /* Upmixing to parent */
+    PDMAUDIOPCMPROPS cfg_c = PDMAUDIOPCMPROPS_INITIALIZER( /* Upmixing to parent */
         1,                                                                  /* Bytes */
         false,                                                              /* Signed */
         1,                                                                  /* Channels */
         22050,                                                              /* Hz */
-        PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(1 /* Bytes */, 1 /* Channels */), /* Shift */
         false                                                               /* Swap Endian */
     );
@@ -514 +505 @@
 
     /**** 8-bit unsigned samples ****/
-    RTTestPrintf(hTest, RTTESTLVL_DEBUG, "Conversion test %uHz %uch 8-bit\n", cfg_c.uHz, cfg_c.cChannels);
+    RTTestPrintf(hTest, RTTESTLVL_DEBUG, "Conversion test %uHz %uch 8-bit\n", cfg_c.uHz, PDMAudioPropsChannels(&cfg_c));
     RTTESTI_CHECK_RC_OK(AudioMixBufWriteCirc(&child, &aFrames8U, sizeof(aFrames8U), &cFramesWritten));
     RTTESTI_CHECK_MSG(cFramesWritten == cFramesChild, ("Child: Expected %RU32 written frames, got %RU32\n", cFramesChild, cFramesWritten));
@@ -565 +556 @@
 
     /* 44100Hz, 1 Channel, S16 */
-    PDMAUDIOPCMPROPS cfg_p = PDMAUDIOPCMPROPS_INITIALIZOR(
+    PDMAUDIOPCMPROPS cfg_p = PDMAUDIOPCMPROPS_INITIALIZER(
         2,                                                                  /* Bytes */
         true,                                                               /* Signed */
         1,                                                                  /* Channels */
         44100,                                                              /* Hz */
-        PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(2 /* Bytes */, 1 /* Channels */), /* Shift */
         false                                                               /* Swap Endian */
     );
@@ -580 +570 @@
 
     /* 22050Hz, 1 Channel, S16 */
-    PDMAUDIOPCMPROPS cfg_c = PDMAUDIOPCMPROPS_INITIALIZOR( /* Upmixing to parent */
+    PDMAUDIOPCMPROPS cfg_c = PDMAUDIOPCMPROPS_INITIALIZER( /* Upmixing to parent */
         2,                                                                  /* Bytes */
         true,                                                               /* Signed */
         1,                                                                  /* Channels */
         22050,                                                              /* Hz */
-        PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(2 /* Bytes */, 1 /* Channels */), /* Shift */
         false                                                               /* Swap Endian */
     );
@@ -611 +600 @@
 
     /**** 16-bit signed samples ****/
-    RTTestPrintf(hTest, RTTESTLVL_DEBUG, "Conversion test %uHz %uch 16-bit\n", cfg_c.uHz, cfg_c.cChannels);
+    RTTestPrintf(hTest, RTTESTLVL_DEBUG, "Conversion test %uHz %uch 16-bit\n", cfg_c.uHz, PDMAudioPropsChannels(&cfg_c));
     RTTESTI_CHECK_RC_OK(AudioMixBufWriteCirc(&child, &aFrames16S, sizeof(aFrames16S), &cFramesWritten));
     RTTESTI_CHECK_MSG(cFramesWritten == cFramesChild, ("Child: Expected %RU32 written frames, got %RU32\n", cFramesChild, cFramesWritten));
@@ -662 +651 @@
     /* Same for parent/child. */
     /* 44100Hz, 2 Channels, S16 */
-    PDMAUDIOPCMPROPS cfg = PDMAUDIOPCMPROPS_INITIALIZOR(
+    PDMAUDIOPCMPROPS cfg = PDMAUDIOPCMPROPS_INITIALIZER(
         2,                                                                  /* Bytes */
         true,                                                               /* Signed */
         2,                                                                  /* Channels */
         44100,                                                              /* Hz */
-        PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(2 /* Bytes */, 2 /* Channels */), /* Shift */
         false                                                               /* Swap Endian */
     );
@@ -699 +687 @@
 
     /**** Volume control test ****/
-    RTTestPrintf(hTest, RTTESTLVL_DEBUG, "Volume control test %uHz %uch \n", cfg.uHz, cfg.cChannels);
+    RTTestPrintf(hTest, RTTESTLVL_DEBUG, "Volume control test %uHz %uch \n", cfg.uHz, PDMAudioPropsChannels(&cfg));
 
     /* 1) Full volume/0dB attenuation (255). */