audio fixes and normalization between hardware types

sdl/sdl-speaker.cpp

@@ -10,8 +10,8 @@ extern "C"
 };
 
 // What values do we use for logical speaker-high and speaker-low?
-#define HIGHVAL (0x1FFF)
-#define LOWVAL (-(0x1FFF))
+#define HIGHVAL (0x4FFF)
+#define LOWVAL (-(0x4FFF))
 
 #include "globals.h"
 
@@ -21,11 +21,13 @@ extern "C"
 
 #define WATERLEVEL SDLSIZE
 
+#define AUDIO_SAMPLE_RATE_EXACT 44100
+
 // FIXME: Globals; ick.
 static volatile uint32_t bufIdx = 0;
 static volatile short soundBuf[CACHEMULTIPLIER*SDLSIZE];
 static pthread_mutex_t togmutex = PTHREAD_MUTEX_INITIALIZER;
-static volatile uint32_t skippedSamples = 0;
+static volatile uint64_t skippedSamples = 0;
 #define SAMPLEBYTES sizeof(short)
 
 volatile uint8_t audioRunning = 0;
@@ -63,7 +65,7 @@ static void audioCallback(void *unused, Uint8 *stream, int len)
   
   static short lastKnownSample = 0; // saved for when the apple is quiescent
 
-  if (bufIdx >= SDLSIZE) {
+  if (bufIdx >= SDLSIZE) { // technically 'len/SAMPLEBYTES' but it should always be constant I think?
     memcpy(stream, (void *)soundBuf, SDLSIZE*SAMPLEBYTES);
     lastKnownSample = stream[SDLSIZE-1];
 
@@ -138,7 +140,7 @@ void SDLSpeaker::begin()
   SDL_AudioSpec audioDevice;
   SDL_AudioSpec audioActual;
   SDL_memset(&audioDevice, 0, sizeof(audioDevice));
-  audioDevice.freq = 44100; // count of 16-bit samples
+  audioDevice.freq = AUDIO_SAMPLE_RATE_EXACT; // count of 16-bit samples
   audioDevice.format = AUDIO_S16;
   audioDevice.channels = 1;
   audioDevice.samples = SDLSIZE; // SDLSIZE 16-bit samples @ 44100Hz: 4096 is about 1/10th second out of sync
@@ -161,13 +163,13 @@ void SDLSpeaker::toggle(int64_t c)
 {
   pthread_mutex_lock(&togmutex);
 
-  int64_t expectedCycleNumber = (float)c * (float)44100 / (float)g_speed;
+  int64_t expectedCycleNumber = (float)c * (float)AUDIO_SAMPLE_RATE_EXACT / (float)g_speed;
   if (lastFilledTime == 0) {
     lastFilledTime = expectedCycleNumber;
   }
   // This subtracts skippedSamples because those were filled automatically
   // by the audioCallback when we had no data.
-  int32_t audioBufferSamples = expectedCycleNumber - lastFilledTime - skippedSamples;
+  int64_t audioBufferSamples = expectedCycleNumber - lastFilledTime - skippedSamples;
   // If audioBufferSamples < 0, then we need to keep some
   // skippedSamples for later; otherwise we can keep moving forward.
   if (audioBufferSamples < 0) {

teensy/teensy-speaker.cpp

@@ -24,13 +24,8 @@ AudioConnection         patchCord4(mixer1, 0, i2s, 0);
 #define BUFSIZE (4096)
 #define CACHEMULTIPLIER 2
 static volatile uint32_t bufIdx; // 0 .. BUFSIZE-1
-static volatile uint32_t skippedSamples; // Who knows where this will
-					 // wind up (FIXME: eventual
-					 // rollover means we need a
-					 // way to purge the queue
-					 // when it's quiescent for
-					 // too long & restart all the
-					 // constants)
+static volatile uint64_t skippedSamples; // Who knows where this will
+					 // wind up
 static volatile uint8_t audioRunning = 0; // FIXME: needs constants abstracted
 static volatile int64_t lastFilledTime = 0;
 
@@ -45,7 +40,8 @@ static bool toggleState = false;
 TeensySpeaker::TeensySpeaker(uint8_t sda, uint8_t scl) : PhysicalSpeaker()
 {
   toggleState = false;
-  mixerValue = numMixed = 0;
+  mixerValue = 0x80;
+
   AudioMemory(8);
 }
 
@@ -55,8 +51,8 @@ TeensySpeaker::~TeensySpeaker()
 
 void TeensySpeaker::begin()
 {
-  mixer1.gain(0, 0.1f); // left channel
-  mixer1.gain(1, 0.1f); // right channel
+  mixer1.gain(0, 0.1f); // left channel (off for now)
+  mixer1.gain(1, 0.1f); // right channel (full volume for now)
 
   memset(soundBuf, 0, sizeof(soundBuf));
   
@@ -83,7 +79,7 @@ void TeensySpeaker::toggle(int64_t c)
   // and we have filled to cycle number lastFilledTime. So how many do
   // we need?  This subtracts skippedSamples because those were filled
   // automatically by the audioCallback when we had no data.
-  int32_t audioBufferSamples = expectedCycleNumber - lastFilledTime - skippedSamples;
+  int64_t audioBufferSamples = expectedCycleNumber - lastFilledTime - skippedSamples;
   // If audioBufferSamples < 0, then we need to keep some
   // skippedSamples for later; otherwise we can keep moving forward.
   if (audioBufferSamples < 0) {
@@ -184,10 +180,7 @@ void TeensyAudio::update(void)
     goto done;
   }
 
-  // FROM THE SOUND OF IT, something below this line isn't filling buffers
-  // completely; or something isn't filling soundBuf completely in toggle().
-  
-  static short lastKnownSample = 0;
+  static short lastKnownSample = 0; // saved for when the apple is quiescent
   
   if (bufIdx >= AUDIO_BLOCK_SAMPLES) {
     memcpy(stream, (void *)soundBuf, AUDIO_BLOCK_SAMPLES * SAMPLEBYTES);
@@ -208,6 +201,7 @@ void TeensyAudio::update(void)
       for (int32_t i=0; i<AUDIO_BLOCK_SAMPLES-bufIdx; i++) {
 	stream[i+bufIdx] = lastKnownSample;
       }
+      bufIdx = 0;
     } else {
       // No big deal - buffer underrun might just mean nothing is
       // trying to play audio right now.

teensy/teensy-speaker.h

@@ -27,8 +27,8 @@ class TeensySpeaker : public PhysicalSpeaker {
 
  private:
 
-  uint32_t mixerValue;
-  uint8_t numMixed;
+   uint8_t mixerValue;
+   bool toggleState;
 };
 
 #endif