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[OMPT-svn-commit] SF.net SVN: modplug:[1043] trunk/OpenMPT
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From: <sag...@us...> - 2011-09-18 23:40:33
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Revision: 1043
http://modplug.svn.sourceforge.net/modplug/?rev=1043&view=rev
Author: saga-games
Date: 2011-09-18 23:40:27 +0000 (Sun, 18 Sep 2011)
Log Message:
-----------
[Mod] Global volume is now applied before internal DSPs (http://bugs.openmpt.org/view.php?id=72).
[Mod] OpenMPT: Version is now 1.20.00.24
Modified Paths:
--------------
trunk/OpenMPT/mptrack/version.h
trunk/OpenMPT/soundlib/Sndfile.h
trunk/OpenMPT/soundlib/Sndmix.cpp
Modified: trunk/OpenMPT/mptrack/version.h
===================================================================
--- trunk/OpenMPT/mptrack/version.h 2011-09-18 21:12:51 UTC (rev 1042)
+++ trunk/OpenMPT/mptrack/version.h 2011-09-18 23:40:27 UTC (rev 1043)
@@ -15,7 +15,7 @@
#define VER_MAJORMAJOR 1
#define VER_MAJOR 20
#define VER_MINOR 00
-#define VER_MINORMINOR 23
+#define VER_MINORMINOR 24
//Creates version number from version parts that appears in version string.
//For example MAKE_VERSION_NUMERIC(1,17,02,28) gives version number of
Modified: trunk/OpenMPT/soundlib/Sndfile.h
===================================================================
--- trunk/OpenMPT/soundlib/Sndfile.h 2011-09-18 21:12:51 UTC (rev 1042)
+++ trunk/OpenMPT/soundlib/Sndfile.h 2011-09-18 23:40:27 UTC (rev 1043)
@@ -1088,7 +1088,7 @@
#ifdef MODPLUG_TRACKER
void ProcessMidiOut(CHANNELINDEX nChn, MODCHANNEL *pChn); //rewbs.VSTdelay : added arg.
#endif
- void ApplyGlobalVolume(int SoundBuffer[], long lTotalSampleCount);
+ void ApplyGlobalVolume(int SoundBuffer[], int RearBuffer[], long lTotalSampleCount);
// Static helper functions
public:
Modified: trunk/OpenMPT/soundlib/Sndmix.cpp
===================================================================
--- trunk/OpenMPT/soundlib/Sndmix.cpp 2011-09-18 21:12:51 UTC (rev 1042)
+++ trunk/OpenMPT/soundlib/Sndmix.cpp 2011-09-18 23:40:27 UTC (rev 1043)
@@ -287,33 +287,38 @@
LPCONVERTPROC pCvt = X86_Convert32To8;
UINT lRead, lMax, lSampleSize, lCount, lSampleCount, nStat=0;
UINT nMaxPlugins;
-
+
nMaxPlugins = MAX_MIXPLUGINS;
while ((nMaxPlugins > 0) && (!m_MixPlugins[nMaxPlugins-1].pMixPlugin)) nMaxPlugins--;
m_nMixStat = 0;
+
lSampleSize = gnChannels;
if (gnBitsPerSample == 16) { lSampleSize *= 2; pCvt = X86_Convert32To16; }
#ifndef FASTSOUNDLIB
else if (gnBitsPerSample == 24) { lSampleSize *= 3; pCvt = X86_Convert32To24; }
else if (gnBitsPerSample == 32) { lSampleSize *= 4; pCvt = X86_Convert32To32; }
-#endif
+#endif // FASTSOUNDLIB
+
lMax = cbBuffer / lSampleSize;
if ((!lMax) || (!lpBuffer) || (!m_nChannels)) return 0;
lRead = lMax;
if (m_dwSongFlags & SONG_ENDREACHED)
goto MixDone;
+
while (lRead > 0)
{
// Update Channel Data
if (!m_nBufferCount)
{
+
#ifndef FASTSOUNDLIB
if (m_dwSongFlags & SONG_FADINGSONG)
{
m_dwSongFlags |= SONG_ENDREACHED;
m_nBufferCount = lRead;
} else
-#endif
+#endif // FASTSOUNDLIB
+
if (ReadNote())
{
// Save pattern cue points for WAV rendering here (if we reached a new pattern, that is.)
@@ -334,9 +339,10 @@
break;
}
#endif // MODPLUG_TRACKER
+
#ifndef FASTSOUNDLIB
if (!FadeSong(FADESONGDELAY) || m_bIsRendering) //rewbs: disable song fade when rendering.
-#endif
+#endif // FASTSOUNDLIB
{
m_dwSongFlags |= SONG_ENDREACHED;
if (lRead == lMax || m_bIsRendering) //rewbs: don't complete buffer when rendering
@@ -345,15 +351,19 @@
}
}
}
+
lCount = m_nBufferCount;
if (lCount > MIXBUFFERSIZE) lCount = MIXBUFFERSIZE;
if (lCount > lRead) lCount = lRead;
if (!lCount)
break;
+
lSampleCount = lCount;
- #ifndef NO_REVERB
+
+#ifndef NO_REVERB
gnReverbSend = 0;
- #endif
+#endif // NO_REVERB
+
// Resetting sound buffer
X86_StereoFill(MixSoundBuffer, lSampleCount, &gnDryROfsVol, &gnDryLOfsVol);
@@ -362,23 +372,42 @@
{
lSampleCount *= 2;
m_nMixStat += CreateStereoMix(lCount);
- #ifndef NO_REVERB
+
+#ifndef NO_REVERB
ProcessReverb(lCount);
- #endif
+#endif // NO_REVERB
+
if (nMaxPlugins) ProcessPlugins(lCount);
+
+ // Apply global volume
+ if (m_pConfig->getGlobalVolumeAppliesToMaster())
+ {
+ ApplyGlobalVolume(MixSoundBuffer, MixRearBuffer, lSampleCount);
+ }
+
ProcessStereoDSP(lCount);
} else
{
m_nMixStat += CreateStereoMix(lCount);
- #ifndef NO_REVERB
+
+#ifndef NO_REVERB
ProcessReverb(lCount);
- #endif
+#endif // NO_REVERB
+
if (nMaxPlugins) ProcessPlugins(lCount);
X86_MonoFromStereo(MixSoundBuffer, lCount);
+
+ // Apply global volume
+ if (m_pConfig->getGlobalVolumeAppliesToMaster())
+ {
+ ApplyGlobalVolume(MixSoundBuffer, nullptr, lSampleCount);
+ }
+
ProcessMonoDSP(lCount);
}
+
+#ifdef ENABLE_EQ
// Graphic Equalizer
-#ifdef ENABLE_EQ
if (gdwSoundSetup & SNDMIX_EQ)
{
if (gnChannels >= 2)
@@ -386,13 +415,17 @@
else
EQMono(MixSoundBuffer, lCount);
}
-#endif
+#endif // ENABLE_EQ
+
nStat++;
+
#ifndef NO_AGC
// Automatic Gain Control
if (gdwSoundSetup & SNDMIX_AGC) ProcessAGC(lSampleCount);
-#endif
- UINT lTotalSampleCount = lSampleCount;
+#endif // NO_AGC
+
+ UINT lTotalSampleCount = lSampleCount; // Including rear channels
+
#ifndef FASTSOUNDLIB
// Multichannel
if (gnChannels > 2)
@@ -400,6 +433,9 @@
X86_InterleaveFrontRear(MixSoundBuffer, MixRearBuffer, lSampleCount);
lTotalSampleCount *= 2;
}
+#endif // FASTSOUNDLIB
+
+#ifndef FASTSOUNDLIB
// Noise Shaping
if (gnBitsPerSample <= 16)
{
@@ -408,16 +444,13 @@
X86_Dither(MixSoundBuffer, lTotalSampleCount, gnBitsPerSample);
}
- //Apply global volume
- if (m_pConfig->getGlobalVolumeAppliesToMaster()) {
- ApplyGlobalVolume(MixSoundBuffer, lTotalSampleCount);
- }
-
// Hook Function
- if (gpSndMixHook) { //Currently only used for VU Meter, so it's OK to do it after global Vol.
+ if (gpSndMixHook)
+ {
+ //Currently only used for VU Meter, so it's OK to do it after global Vol.
gpSndMixHook(MixSoundBuffer, lTotalSampleCount, gnChannels);
}
-#endif
+#endif // FASTSOUNDLIB
// Perform clipping
lpBuffer += pCvt(lpBuffer, MixSoundBuffer, lTotalSampleCount);
@@ -1597,7 +1630,7 @@
*/
const int vibpos = pChn->nAutoVibPos & 0xFF;
int adepth = pChn->nAutoVibDepth; // (1)
- adepth += pSmp->nVibSweep & 0xFF; // (2 & 3)
+ adepth += pSmp->nVibSweep; // (2 & 3)
adepth = min(adepth, (int)(pSmp->nVibDepth << 8));
pChn->nAutoVibDepth = adepth; // (5)
adepth >>= 8; // (4)
@@ -2529,52 +2562,61 @@
#endif
-void CSoundFile::ApplyGlobalVolume(int SoundBuffer[], long lTotalSampleCount)
-//---------------------------------------------------------------------------
+void CSoundFile::ApplyGlobalVolume(int SoundBuffer[], int RearBuffer[], long lTotalSampleCount)
+//---------------------------------------------------------------------------------------------
{
- long delta = 0;
long step = 0;
if (m_nGlobalVolumeDestination != m_nGlobalVolume)
{
- //user has provided new global volume
+ // User has provided new global volume
const bool rampUp = m_nGlobalVolumeDestination > m_nGlobalVolume;
m_nGlobalVolumeDestination = m_nGlobalVolume;
m_nSamplesToGlobalVolRampDest = rampUp ? gnVolumeRampUpSamples : gnVolumeRampDownSamples;
}
- long rampLength = m_nSamplesToGlobalVolRampDest;
+ const long rampLength = m_nSamplesToGlobalVolRampDest;
if (m_nSamplesToGlobalVolRampDest > 0)
{
- // still some ramping left to do.
- long highResGlobalVolumeDestination = static_cast<long>(m_nGlobalVolumeDestination)<<VOLUMERAMPPRECISION;
+ // Still some ramping left to do.
+ long highResGlobalVolumeDestination = static_cast<long>(m_nGlobalVolumeDestination) << VOLUMERAMPPRECISION;
- delta = highResGlobalVolumeDestination - m_lHighResRampingGlobalVolume;
+ const long delta = highResGlobalVolumeDestination - m_lHighResRampingGlobalVolume;
step = delta / static_cast<long>(m_nSamplesToGlobalVolRampDest);
- UINT maxStep = max(50, (10000 / (rampLength + 1))); // define max step size as some factor of user defined ramping value: the lower the value, the more likely the click.
- while(abs(step) > maxStep) // if step is too big (might cause click), extend ramp length.
+ UINT maxStep = max(50, (10000 / (rampLength + 1))); // Define max step size as some factor of user defined ramping value: the lower the value, the more likely the click.
+ while(abs(step) > maxStep) // If step is too big (might cause click), extend ramp length.
{
m_nSamplesToGlobalVolRampDest += rampLength;
step = delta / static_cast<long>(m_nSamplesToGlobalVolRampDest);
}
}
- for (int pos = 0; pos < lTotalSampleCount; pos++)
+ const long highResVolume = m_lHighResRampingGlobalVolume;
+ const UINT samplesToRamp = m_nSamplesToGlobalVolRampDest;
+
+ // SoundBuffer has interleaved left/right channels for the front channels; RearBuffer has the rear left/right channels.
+ // So we process the pairs independently for ramping.
+ for (int pairs = max(gnChannels / 2, 1); pairs > 0; pairs--)
{
+ int *sample = (pairs == 1) ? SoundBuffer : RearBuffer;
+ m_lHighResRampingGlobalVolume = highResVolume;
+ m_nSamplesToGlobalVolRampDest = samplesToRamp;
- if (m_nSamplesToGlobalVolRampDest > 0)
+ for (int pos = lTotalSampleCount; pos > 0; pos--, sample++)
{
- //ramping required
- m_lHighResRampingGlobalVolume += step;
- SoundBuffer[pos] = _muldiv(SoundBuffer[pos], m_lHighResRampingGlobalVolume, MAX_GLOBAL_VOLUME<<VOLUMERAMPPRECISION);
- m_nSamplesToGlobalVolRampDest--;
- } else
- {
- SoundBuffer[pos] = _muldiv(SoundBuffer[pos], m_nGlobalVolume, MAX_GLOBAL_VOLUME);
- m_lHighResRampingGlobalVolume = m_nGlobalVolume<<VOLUMERAMPPRECISION;
+ if (m_nSamplesToGlobalVolRampDest > 0)
+ {
+ // Ramping required
+ m_lHighResRampingGlobalVolume += step;
+ *sample = _muldiv(*sample, m_lHighResRampingGlobalVolume, MAX_GLOBAL_VOLUME << VOLUMERAMPPRECISION);
+ m_nSamplesToGlobalVolRampDest--;
+ } else
+ {
+ *sample = _muldiv(*sample, m_nGlobalVolume, MAX_GLOBAL_VOLUME);
+ m_lHighResRampingGlobalVolume = m_nGlobalVolume << VOLUMERAMPPRECISION;
+ }
}
-
}
}
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