Fix minor compile warnings generated by clang:

- mostly change pointer -> 0 to use 'nullptr'
 - some commenting and formatting fixes

src/common/AudioQueue.cxx

@@ -93,7 +93,7 @@ Int16* AudioQueue::enqueue(Int16* fragment)
     if (!myFirstFragmentForEnqueue) throw runtime_error("enqueue called empty");
 
     newFragment = myFirstFragmentForEnqueue;
-    myFirstFragmentForEnqueue = 0;
+    myFirstFragmentForEnqueue = nullptr;
 
     return newFragment;
   }
@@ -115,13 +115,13 @@ Int16* AudioQueue::dequeue(Int16* fragment)
 {
   lock_guard<mutex> guard(myMutex);
 
-  if (mySize == 0) return 0;
+  if (mySize == 0) return nullptr;
 
   if (!fragment) {
     if (!myFirstFragmentForDequeue) throw runtime_error("dequeue called empty");
 
     fragment = myFirstFragmentForDequeue;
-    myFirstFragmentForDequeue = 0;
+    myFirstFragmentForDequeue = nullptr;
   }
 
   Int16* nextFragment = myFragmentQueue.at(myNextFragment);

src/common/AudioQueue.hxx

@@ -41,7 +41,7 @@ class AudioQueue
        Create a new AudioQueue.
 
        @param fragmentSize  The size (in stereo / mono samples) of each fragment
-       @param capacaity         The number of fragments that can be queued before wrapping.
+       @param capacity      The number of fragments that can be queued before wrapping.
        @param isStereo      Whether samples are stereo or mono.
        @param sampleRate    The sample rate. This is not used, but can be queried.
      */
@@ -83,7 +83,7 @@ class AudioQueue
       @param fragment   The returned fragment. This must be empty on the first call (when
                         there is nothing to return)
      */
-    Int16* enqueue(Int16* fragment = 0);
+    Int16* enqueue(Int16* fragment = nullptr);
 
     /**
       Dequeue a fragment for playback and return the played fragment. This may
@@ -93,7 +93,7 @@ class AudioQueue
       @param fragment  The returned fragment. This must be empty on the first call (when
                        there is nothing to return).
      */
-    Int16* dequeue(Int16* fragment = 0);
+    Int16* dequeue(Int16* fragment = nullptr);
 
     /**
       Return the currently playing fragment without drawing a new one. This is called

src/common/SoundSDL2.cxx

@@ -39,8 +39,7 @@ SoundSDL2::SoundSDL2(OSystem& osystem)
     myIsInitializedFlag(false),
     myVolume(100),
     myVolumeFactor(0xffff),
-    myAudioQueue(0),
-    myCurrentFragment(0)
+    myCurrentFragment(nullptr)
 {
   myOSystem.logMessage("SoundSDL2::SoundSDL2 started ...", 2);
 
@@ -103,9 +102,10 @@ void SoundSDL2::setEnabled(bool state)
 }
 
 // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
-void SoundSDL2::open(shared_ptr<AudioQueue> audioQueue, EmulationTiming* emulationTiming)
+void SoundSDL2::open(shared_ptr<AudioQueue> audioQueue,
+                     EmulationTiming* emulationTiming)
 {
-  this->emulationTiming = emulationTiming;
+  myEmulationTiming = emulationTiming;
 
   myOSystem.logMessage("SoundSDL2::open started ...", 2);
   mute(true);
@@ -118,7 +118,7 @@ void SoundSDL2::open(shared_ptr<AudioQueue> audioQueue, EmulationTiming* emulati
 
   myAudioQueue = audioQueue;
   myUnderrun = true;
-  myCurrentFragment = 0;
+  myCurrentFragment = nullptr;
 
   // Adjust volume to that defined in settings
   setVolume(myOSystem.settings().getInt("volume"));
@@ -149,8 +149,8 @@ void SoundSDL2::close()
   mute(true);
 
   if (myAudioQueue) myAudioQueue->closeSink(myCurrentFragment);
-  myAudioQueue = 0;
-  myCurrentFragment = 0;
+  myAudioQueue.reset();
+  myCurrentFragment = nullptr;
 
   myOSystem.logMessage("SoundSDL2::close", 2);
 
@@ -167,7 +167,8 @@ void SoundSDL2::mute(bool state)
 
 // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
 void SoundSDL2::reset()
-{}
+{
+}
 
 // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
 void SoundSDL2::setVolume(Int32 percent)
@@ -233,14 +234,15 @@ void SoundSDL2::processFragment(float* stream, uInt32 length)
 void SoundSDL2::initResampler()
 {
   Resampler::NextFragmentCallback nextFragmentCallback = [this] () -> Int16* {
-    Int16* nextFragment = 0;
+    Int16* nextFragment = nullptr;
 
     if (myUnderrun)
-      nextFragment = myAudioQueue->size() > emulationTiming->prebufferFragmentCount() ? myAudioQueue->dequeue(myCurrentFragment) : 0;
+      nextFragment = myAudioQueue->size() > myEmulationTiming->prebufferFragmentCount() ?
+          myAudioQueue->dequeue(myCurrentFragment) : nullptr;
     else
       nextFragment = myAudioQueue->dequeue(myCurrentFragment);
 
-    myUnderrun = nextFragment == 0;
+    myUnderrun = nextFragment == nullptr;
     if (nextFragment) myCurrentFragment = nextFragment;
 
     return nextFragment;

src/common/SoundSDL2.hxx

@@ -33,7 +33,7 @@ class EmulationTiming;
 /**
   This class implements the sound API for SDL.
 
-  @author Stephen Anthony and Bradford W. Mott
+  @author Stephen Anthony and Christian Speckner (DirtyHairy)
 */
 class SoundSDL2 : public Sound
 {
@@ -130,7 +130,7 @@ class SoundSDL2 : public Sound
 
     shared_ptr<AudioQueue> myAudioQueue;
 
-    EmulationTiming* emulationTiming;
+    EmulationTiming* myEmulationTiming;
 
     Int16* myCurrentFragment;
     bool myUnderrun;

src/common/audio/LanczosResampler.cxx

@@ -75,10 +75,10 @@ LanczosResampler::LanczosResampler(
   myKernelSize(2 * kernelParameter),
   myCurrentKernelIndex(0),
   myKernelParameter(kernelParameter),
-  myBuffer(0),
-  myBufferL(0),
-  myBufferR(0),
-  myCurrentFragment(0),
+  myBuffer(nullptr),
+  myBufferL(nullptr),
+  myBufferR(nullptr),
+  myCurrentFragment(nullptr),
   myFragmentIndex(0),
   myIsUnderrun(true),
   myTimeIndex(0)

src/common/audio/LanczosResampler.hxx

@@ -22,7 +22,8 @@
 #include "Resampler.hxx"
 #include "ConvolutionBuffer.hxx"
 
-class LanczosResampler : public Resampler {
+class LanczosResampler : public Resampler
+{
   public:
     LanczosResampler(
       Resampler::Format formatFrom,

src/common/audio/Resampler.hxx

@@ -30,10 +30,10 @@ class Resampler {
     class Format {
       public:
 
-        Format(uInt32 sampleRate, uInt32 fragmentSize, bool stereo) :
-          sampleRate(sampleRate),
-          fragmentSize(fragmentSize),
-          stereo(stereo)
+        Format(uInt32 f_sampleRate, uInt32 f_fragmentSize, bool f_stereo) :
+          sampleRate(f_sampleRate),
+          fragmentSize(f_fragmentSize),
+          stereo(f_stereo)
         {}
 
       public:

src/common/audio/SimpleResampler.cxx

@@ -22,13 +22,13 @@ SimpleResampler::SimpleResampler(
   Resampler::Format formatFrom,
   Resampler::Format formatTo,
   Resampler::NextFragmentCallback nextFragmentCallback)
-:
-  Resampler(formatFrom, formatTo, nextFragmentCallback),
-  myCurrentFragment(0),
+  : Resampler(formatFrom, formatTo, nextFragmentCallback),
+    myCurrentFragment(nullptr),
     myTimeIndex(0),
     myFragmentIndex(0),
     myIsUnderrun(true)
-{}
+{
+}
 
 // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
 void SimpleResampler::fillFragment(float* fragment, uInt32 length)

src/common/audio/SimpleResampler.hxx

@@ -21,7 +21,8 @@
 #include "bspf.hxx"
 #include "Resampler.hxx"
 
-class SimpleResampler : public Resampler {
+class SimpleResampler : public Resampler
+{
   public:
     SimpleResampler(
       Resampler::Format formatFrom,

src/emucore/M6502.hxx

@@ -110,7 +110,7 @@ class M6502 : public Serializable
       is executed, someone stops execution, or an error occurs.  Answers
       true iff execution stops normally.
 
-      @param number Indicates the number of cycles to execute. Not that the actual
+      @param cycles Indicates the number of cycles to execute. Not that the actual
                     granularity of the CPU is instructions, so this is only accurate up to
                     a couple of cycles
       @return true iff execution stops normally

src/emucore/tia/Audio.cxx

@@ -35,8 +35,8 @@ namespace {
 
 // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
 Audio::Audio()
-  : myAudioQueue(0),
-    myCurrentFragment(0)
+  : myAudioQueue(nullptr),
+    myCurrentFragment(nullptr)
 {
   for (uInt8 i = 0; i <= 0x1e; i++) myMixingTableSum[i] = mixingTableEntry(i, 0x1e);
   for (uInt8 i = 0; i <= 0x0f; i++) myMixingTableIndividual[i] = mixingTableEntry(i, 0x0f);
@@ -65,7 +65,8 @@ void Audio::setAudioQueue(shared_ptr<AudioQueue> queue)
 
 // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
 void Audio::tick()
-{ switch (myCounter) {
+{
+  switch (myCounter) {
     case 9:
     case 81:
       myChannel0.phase0();