Changeset 55462 in vbox for trunk/src/VBox/Devices

Timestamp:

Apr 27, 2015 8:01:05 PM (10 years ago)

Author:

vboxsync

Message:

Audio: Fixed minor format conversion errors, added a testcase (16-bit signed test only so far).

Location:

trunk/src/VBox/Devices/Audio

Files:

• AudioMixBuffer.cpp (modified) (9 diffs)
• testcase/tstAudioMixBuffer.cpp (modified) (2 diffs)

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

@@ -74 +74 @@
  *
  * For actual volume calculation, 33.31 fixed point is used. Maximum (or
- * unattenuated) volume is represented as 0x80000000; conveniently, this
+ * unattenuated) volume is represented as 0x40000000; conveniently, this
  * value fits into a uint32_t.
+ *
+ * To enable fast processing, the maximum volume must be a power of two
+ * and must not have a sign when converted to int32_t. While 0x80000000
+ * violates these constraints, 0x40000000 does not.
  */
 
@@ -115 +119 @@
 };
 
+/* Bit shift for fixed point conversion. */
+#define VOL_SHIFT       30
+
+/* Internal representation of 0dB volume (1.0 in fixed point). */
+#define VOL_0DB         (1 << VOL_SHIFT)
+
+AssertCompile(VOL_0DB <= 0x40000000);   /* Must always hold. */
+AssertCompile(VOL_0DB == 0x40000000);   /* For now -- when only attenuation is used. */
+
 /**
  * Structure for holding sample conversion parameters for
@@ -343 +356 @@
     AUDMIXBUF_MACRO_FN _aType audioMixBufClipTo##_aName(int64_t iVal) \
     { \
-        if (iVal >= 0x7f000000) \
+        if (iVal >= 0x7fffffff) \
             return _aMax; \
-        else if (iVal < -2147483648LL) \
+        else if (iVal < -INT64_C(0x80000000)) \
             return _aMin; \
         \
@@ -363 +376 @@
         { \
             AUDMIXBUF_MACRO_LOG(("%p: l=%RI16, r=%RI16\n", paDst, *pSrc, *(pSrc + 1))); \
-            paDst->i64LSample = ASMMult2xS32RetS64((int32_t)audioMixBufClipFrom##_aName(*pSrc++), pOpts->Volume.uLeft ) >> 31; \
-            paDst->i64RSample = ASMMult2xS32RetS64((int32_t)audioMixBufClipFrom##_aName(*pSrc++), pOpts->Volume.uRight) >> 31; \
+            paDst->i64LSample = ASMMult2xS32RetS64((int32_t)audioMixBufClipFrom##_aName(*pSrc++), pOpts->Volume.uLeft ) >> VOL_SHIFT; \
+            paDst->i64RSample = ASMMult2xS32RetS64((int32_t)audioMixBufClipFrom##_aName(*pSrc++), pOpts->Volume.uRight) >> VOL_SHIFT; \
             AUDMIXBUF_MACRO_LOG(("\t-> l=%RI64, r=%RI64\n", paDst->i64LSample, paDst->i64RSample)); \
             paDst++; \
@@ -382 +395 @@
         { \
             AUDMIXBUF_MACRO_LOG(("%p: s=%RI16\n", paDst, *pSrc)); \
-            paDst->i64LSample = ASMMult2xS32RetS64((int32_t)audioMixBufClipFrom##_aName(*pSrc++), pOpts->Volume.uLeft) >> 31; \
+            paDst->i64LSample = ASMMult2xS32RetS64((int32_t)audioMixBufClipFrom##_aName(*pSrc++), pOpts->Volume.uLeft) >> VOL_SHIFT; \
             paDst->i64RSample = paDst->i64LSample; \
             AUDMIXBUF_MACRO_LOG(("\t-> l=%RI64, r=%RI64\n", paDst->i64LSample, paDst->i64RSample)); \
@@ -418 +431 @@
         while (cSamples--) \
         { \
-            *pDst++ = audioMixBufClipTo##_aName(pSrc->i64LSample + pSrc->i64RSample); \
+            *pDst++ = audioMixBufClipTo##_aName((pSrc->i64LSample + pSrc->i64RSample) / 2); \
             pSrc++; \
         } \
@@ -506 +519 @@
             int64_t iDstOffInt = pRate->dstOffset & UINT32_MAX; \
             \
-            samOut.i64LSample = (samLast.i64LSample * ((int64_t) UINT32_MAX - iDstOffInt) + samCur.i64LSample * iDstOffInt) >> 32; \
-            samOut.i64RSample = (samLast.i64RSample * ((int64_t) UINT32_MAX - iDstOffInt) + samCur.i64RSample * iDstOffInt) >> 32; \
+            samOut.i64LSample = (samLast.i64LSample * ((int64_t) (INT64_C(1) << 32) - iDstOffInt) + samCur.i64LSample * iDstOffInt) >> 32; \
+            samOut.i64RSample = (samLast.i64RSample * ((int64_t) (INT64_C(1) << 32) - iDstOffInt) + samCur.i64RSample * iDstOffInt) >> 32; \
             \
             paDst->i64LSample _aOp samOut.i64LSample; \
@@ -709 +722 @@
     /* Set initial volume to max. */
     pMixBuf->Volume.fMuted = false;
-    pMixBuf->Volume.uLeft  = 0x7FFFFFFF;
-    pMixBuf->Volume.uRight = 0x7FFFFFFF;
+    pMixBuf->Volume.uLeft  = VOL_0DB;
+    pMixBuf->Volume.uRight = VOL_0DB;
 
     /* Prevent division by zero.
@@ -1196 +1209 @@
 
     pMixBuf->Volume.fMuted = pVol->fMuted;
-    pMixBuf->Volume.uLeft  = (UINT64_C(0x80000000) * pVol->uLeft) / 255;
-    pMixBuf->Volume.uRight = (UINT64_C(0x80000000) * pVol->uRight) / 255;
     //@todo: Ensure that the input is in the correct range/initialized!
-    pMixBuf->Volume.uLeft  = aVolumeConv[pVol->uLeft  & 0xFF] * UINT64_C(0x8000);
-    pMixBuf->Volume.uRight = aVolumeConv[pVol->uRight & 0xFF] * UINT64_C(0x8000);
+    pMixBuf->Volume.uLeft  = aVolumeConv[pVol->uLeft  & 0xFF] * UINT32_C(VOL_0DB >> 16);
+    pMixBuf->Volume.uRight = aVolumeConv[pVol->uRight & 0xFF] * UINT32_C(VOL_0DB >> 16);
 
     LogFlowFunc(("\t-> lVol=%#RX32, rVol=%#RX32\n", pMixBuf->Volume.uLeft, pMixBuf->Volume.uRight));

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

@@ -236 +236 @@
 }
 
+static int tstConversion(RTTEST hTest)
+{
+    unsigned        i;
+    uint32_t        cBufSize = 256;
+    PDMPCMPROPS     props;
+
+
+    RTTestSubF(hTest, "Sample conversion");
+
+    PDMAUDIOSTREAMCFG cfg_p =
+    {
+        44100,                   /* Hz */
+        1                        /* Channels */,
+        AUD_FMT_S16              /* Format */,
+        PDMAUDIOENDIANNESS_LITTLE /* ENDIANNESS */
+    };
+
+    int rc = drvAudioStreamCfgToProps(&cfg_p, &props);
+    AssertRC(rc);
+
+    PDMAUDIOMIXBUF parent;
+    RTTESTI_CHECK_RC_OK(audioMixBufInit(&parent, "Parent", &props, cBufSize));
+
+    /* Child uses half the sample rate; that ensures the mixing engine can't
+     * take shortcuts and performs conversion. Because conversion to double
+     * the sample rate effectively inserts one additional sample between every
+     * two source samples, N source samples will be converted to N * 2 - 1
+     * samples. However, the last source sample will be saved for later
+     * interpolation and not immediately output.
+     */
+    PDMAUDIOSTREAMCFG cfg_c =   /* Upmixing to parent */
+    {
+        22050,                   /* Hz */
+        1                        /* Channels */,
+        AUD_FMT_S16              /* Format */,
+        PDMAUDIOENDIANNESS_LITTLE /* ENDIANNESS */
+    };
+
+    rc = drvAudioStreamCfgToProps(&cfg_c, &props);
+    AssertRC(rc);
+
+    PDMAUDIOMIXBUF child;
+    RTTESTI_CHECK_RC_OK(audioMixBufInit(&child, "Child", &props, cBufSize));
+    RTTESTI_CHECK_RC_OK(audioMixBufLinkTo(&child, &parent));
+
+    /*
+     * Writing + mixing from child -> parent, sequential.
+     */
+    uint32_t cbBuf = 256;
+    char pvBuf[256];
+    int16_t samples[16] = { 0xAA, 0xBB, INT16_MIN, INT16_MIN + 1, INT16_MIN / 2, -3, -2, -1,
+                            0, 1, 2, 3, INT16_MAX / 2, INT16_MAX - 1, INT16_MAX, 0 };
+    uint32_t read, written, mixed, temp;
+
+    uint32_t cChildFree     = cBufSize;
+    uint32_t cChildMixed    = 0;
+    uint32_t cSamplesChild  = 16;
+    uint32_t cSamplesParent = cSamplesChild * 2 - 2;
+    uint32_t cSamplesRead   = 0;
+
+    RTTestPrintf(hTest, RTTESTLVL_DEBUG, "Conversion test %uHz %uch\n", cfg_c.uHz, cfg_c.cChannels);
+    RTTESTI_CHECK_RC_OK(audioMixBufWriteAt(&child, 0, &samples, sizeof(samples), &written));
+    RTTESTI_CHECK_MSG(written == cSamplesChild, ("Child: Expected %RU32 written samples, got %RU32\n", cSamplesChild, written));
+    RTTESTI_CHECK_RC_OK(audioMixBufMixToParent(&child, written, &mixed));
+    temp = audioMixBufProcessed(&parent);
+    RTTESTI_CHECK_MSG(audioMixBufMixed(&child) == temp, ("Child: Expected %RU32 mixed samples, got %RU32\n", audioMixBufMixed(&child), temp));
+
+    RTTESTI_CHECK(audioMixBufProcessed(&parent) == audioMixBufMixed(&child));
+
+    for (;;)
+    {
+        RTTESTI_CHECK_RC_OK_BREAK(audioMixBufReadCirc(&parent, pvBuf, cbBuf, &read));
+        if (!read)
+            break;
+        cSamplesRead += read;
+        audioMixBufFinish(&parent, read);
+    }
+    RTTESTI_CHECK_MSG(cSamplesRead == cSamplesParent, ("Parent: Expected %RU32 mixed samples, got %RU32\n", cSamplesParent, cSamplesRead));
+
+    /* Check that the samples came out unharmed. Every other sample is interpolated and we ignore it. */
+    int16_t *pSrc16 = &samples[0];
+    int16_t *pDst16 = (int16_t *)pvBuf;
+
+    for (i = 0; i < cSamplesChild; ++i)
+    {
+        RTTESTI_CHECK_MSG(*pSrc16 == *pDst16, ("index %u: Dst=%d, Src=%d\n", i, *pDst16, *pSrc16));
+        pSrc16 += 1;
+        pDst16 += 2;
+    }
+
+    RTTESTI_CHECK(audioMixBufProcessed(&parent) == 0);
+    RTTESTI_CHECK(audioMixBufMixed(&child) == 0);
+
+    return RTTestSubErrorCount(hTest) ? VERR_GENERAL_FAILURE : VINF_SUCCESS;
+}
+
 int main(int argc, char **argv)
 {
@@ -252 +348 @@
     if (RT_SUCCESS(rc))
         rc = tstParentChild(hTest);
+    if (RT_SUCCESS(rc))
+        rc = tstConversion(hTest);
 
     /*