Switch to F32 samples, make volume scale more linear in sound..

src/common/SoundSDL2.cxx

@@ -32,13 +32,6 @@
 #include "EmulationTiming.hxx"
 #include "audio/SimpleResampler.hxx"
 
-namespace {
-  inline Int16 applyVolume(Int16 sample, Int32 volumeFactor)
-  {
-    return static_cast<Int16>(static_cast<Int32>(sample) * volumeFactor / 0xffff);
-  }
-}
-
 // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
 SoundSDL2::SoundSDL2(OSystem& osystem)
   : Sound(osystem),
@@ -50,20 +43,13 @@ SoundSDL2::SoundSDL2(OSystem& osystem)
 {
   myOSystem.logMessage("SoundSDL2::SoundSDL2 started ...", 2);
 
-#ifdef BSPF_WINDOWS
-  // TODO - remove the following code once we convert to the new sound
-  //        core, and use 32-bit floating point samples and do
-  //        our own resampling
-  SDL_setenv("SDL_AUDIODRIVER", "directsound", true);
-#endif
-
   // The sound system is opened only once per program run, to eliminate
   // issues with opening and closing it multiple times
   // This fixes a bug most prevalent with ATI video cards in Windows,
   // whereby sound stopped working after the first video change
   SDL_AudioSpec desired;
   desired.freq   = myOSystem.settings().getInt("freq");
-  desired.format = AUDIO_S16SYS;
+  desired.format = AUDIO_F32SYS;
   desired.channels = 2;
   desired.samples  = myOSystem.settings().getInt("fragsize");
   desired.callback = callback;
@@ -191,7 +177,7 @@ void SoundSDL2::setVolume(Int32 percent)
     myVolume = percent;
 
     SDL_LockAudio();
-    myVolumeFactor = static_cast<Int32>(floor(static_cast<double>(0xffff) * static_cast<double>(myVolume) / 100.));
+    myVolumeFactor = std::pow(static_cast<float>(percent) / 100.f, 2.f);
     SDL_UnlockAudio();
   }
 }
@@ -235,11 +221,11 @@ uInt32 SoundSDL2::getSampleRate() const
 }
 
 // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
-void SoundSDL2::processFragment(Int16* stream, uInt32 length)
+void SoundSDL2::processFragment(float* stream, uInt32 length)
 {
   myResampler->fillFragment(stream, length);
 
-  for (uInt32 i = 0; i < length; i++) stream[i] = applyVolume(stream[i], myVolumeFactor);
+  for (uInt32 i = 0; i < length; i++) stream[i] = stream[i] * myVolumeFactor;
 }
 
 // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
@@ -272,7 +258,7 @@ void SoundSDL2::callback(void* udata, uInt8* stream, int len)
   SoundSDL2* self = static_cast<SoundSDL2*>(udata);
 
   if (self->myAudioQueue)
-    self->processFragment(reinterpret_cast<Int16*>(stream), len >> 1);
+    self->processFragment(reinterpret_cast<float*>(stream), len >> 2);
   else
     SDL_memset(stream, 0, len);
 }

src/common/SoundSDL2.hxx

@@ -111,7 +111,7 @@ class SoundSDL2 : public Sound
       @param stream  Pointer to the start of the fragment
       @param length  Length of the fragment
     */
-    void processFragment(Int16* stream, uInt32 length);
+    void processFragment(float* stream, uInt32 length);
 
   private:
 
@@ -123,7 +123,7 @@ class SoundSDL2 : public Sound
 
     // Current volume as a percentage (0 - 100)
     uInt32 myVolume;
-    Int32 myVolumeFactor;
+    float myVolumeFactor;
 
     // Audio specification structure
     SDL_AudioSpec myHardwareSpec;

src/common/audio/Resampler.hxx

@@ -55,7 +55,7 @@ class Resampler {
       myNextFragmentCallback(nextFragmentCallback)
     {}
 
-    virtual void fillFragment(Int16* fragment, uInt32 length) = 0;
+    virtual void fillFragment(float* fragment, uInt32 length) = 0;
 
     virtual ~Resampler() {}
 

src/common/audio/SimpleResampler.cxx

@@ -31,7 +31,7 @@ SimpleResampler::SimpleResampler(
 {}
 
 // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
-void SimpleResampler::fillFragment(Int16* fragment, uInt32 length)
+void SimpleResampler::fillFragment(float* fragment, uInt32 length)
 {
   if (myIsUnderrun) {
     Int16* nextFragment = myNextFragmentCallback();
@@ -44,7 +44,7 @@ void SimpleResampler::fillFragment(Int16* fragment, uInt32 length)
   }
 
   if (!myCurrentFragment) {
-    memset(fragment, 0, 2 * length);
+    memset(fragment, 0, sizeof(float) * length);
     return;
   }
 
@@ -74,18 +74,23 @@ void SimpleResampler::fillFragment(Int16* fragment, uInt32 length)
       }
     }
 
-    if (myFormatTo.stereo) {
-      if (myFormatFrom.stereo) {
-        fragment[2*i] = myCurrentFragment[2*myFragmentIndex];
-        fragment[2*i + 1] = myCurrentFragment[2*myFragmentIndex + 1];
+    if (myFormatFrom.stereo) {
+      float sampleL = static_cast<float>(myCurrentFragment[2*myFragmentIndex]) / static_cast<float>(0x7fff);
+      float sampleR = static_cast<float>(myCurrentFragment[2*myFragmentIndex + 1]) / static_cast<float>(0x7fff);
+
+      if (myFormatTo.stereo) {
+        fragment[2*i] = sampleL;
+        fragment[2*i + 1] = sampleR;
       }
       else
-        fragment[2*i] = fragment[2*i + 1] = myCurrentFragment[myFragmentIndex];
+        fragment[i] = (sampleL + sampleR) / 2.f;
     } else {
-      if (myFormatFrom.stereo)
-        fragment[i] = (myCurrentFragment[2*myFragmentIndex] / 2) + (myCurrentFragment[2*myFragmentIndex + 1] / 2);
+      float sample = static_cast<float>(myCurrentFragment[myFragmentIndex] / static_cast<float>(0x7fff));
+
+      if (myFormatTo.stereo)
+        fragment[2*i] = fragment[2*i + 1] = sample;
       else
-        fragment[i] = myCurrentFragment[myFragmentIndex];
+        fragment[i] = sample;
     }
   }
 }

src/common/audio/SimpleResampler.hxx

@@ -28,7 +28,7 @@ class SimpleResampler : public Resampler {
       Resampler::NextFragmentCallback NextFragmentCallback
     );
 
-    virtual void fillFragment(Int16* fragment, uInt32 length);
+    virtual void fillFragment(float* fragment, uInt32 length);
 
   private: