Various changes to improve code readability, added mouse hiding.

[virtualjaguar] / src / dac.cpp

diff --git a/src/dac.cpp b/src/dac.cpp
index b7a781091ebea15cba7f346f0b263b3e838e5280..f08295448ceca78c4392aeb38c3461b1d680be55 100644 (file)
--- a/src/dac.cpp
+++ b/src/dac.cpp
@@ -77,8 +77,8 @@
 
 static SDL_AudioSpec desired;
 static bool SDLSoundInitialized;
-static uint8_t SCLKFrequencyDivider = 19;                      // Default is roughly 22 KHz (20774 Hz in NTSC mode)
-/*static*/ uint16_t serialMode = 0;
+//static uint8_t SCLKFrequencyDivider = 19;                    // Default is roughly 22 KHz (20774 Hz in NTSC mode)
+// /*static*/ uint16_t serialMode = 0;
 
 // Private function prototypes
 
@@ -117,6 +117,7 @@ void DACInit(void)
        }
 
        ltxd = lrxd = desired.silence;
+       sclk = 19;                                                                      // Default is roughly 22 KHz
 
        uint32_t riscClockRate = (vjs.hardwareTypeNTSC ? RISC_CLOCK_RATE_NTSC : RISC_CLOCK_RATE_PAL);
        uint32_t cyclesPerSample = riscClockRate / DAC_AUDIO_RATE;
@@ -194,8 +195,8 @@ void SDLSoundCallback(void * userdata, Uint8 * buffer, int length)
 
        // The length of time we're dealing with here is 1/48000 s, so we multiply this
        // by the number of cycles per second to get the number of cycles for one sample.
-       uint32_t riscClockRate = (vjs.hardwareTypeNTSC ? RISC_CLOCK_RATE_NTSC : RISC_CLOCK_RATE_PAL);
-       uint32_t cyclesPerSample = riscClockRate / DAC_AUDIO_RATE;
+//     uint32_t riscClockRate = (vjs.hardwareTypeNTSC ? RISC_CLOCK_RATE_NTSC : RISC_CLOCK_RATE_PAL);
+//     uint32_t cyclesPerSample = riscClockRate / DAC_AUDIO_RATE;
        // This is the length of time
 //     timePerSample = (1000000.0 / (double)riscClockRate) * ();
 
@@ -203,12 +204,14 @@ void SDLSoundCallback(void * userdata, Uint8 * buffer, int length)
 
        bufferIndex = 0;
        sampleBuffer = buffer;
+// If length is the length of the sample buffer in BYTES, then shouldn't the # of
+// samples be / 4? No, because we bump the sample count by 2, so this is OK.
        numberOfSamples = length / 2;
        bufferDone = false;
 
        SetCallbackTime(DSPSampleCallback, 1000000.0 / (double)DAC_AUDIO_RATE, EVENT_JERRY);
 
-       // These timings are tied to NTSC, need to fix that in event.cpp/h!
+       // These timings are tied to NTSC, need to fix that in event.cpp/h! [FIXED]
        do
        {
                double timeToNextEvent = GetTimeToNextEvent(EVENT_JERRY);
@@ -281,14 +284,17 @@ void DACWriteWord(uint32_t offset, uint16_t data, uint32_t who/*= UNKNOWN*/)
        }
        else if (offset == SCLK + 2)                                    // Sample rate
        {
-               WriteLog("DAC: Writing %u to SCLK...\n", data);
+               WriteLog("DAC: Writing %u to SCLK (by %s)...\n", data, whoName[who]);
 
-               if ((uint8_t)data != SCLKFrequencyDivider)
-                       SCLKFrequencyDivider = (uint8_t)data;
+               sclk = data & 0xFF;
+               JERRYI2SInterruptTimer = -1;
+               RemoveCallback(JERRYI2SCallback);
+               JERRYI2SCallback();
        }
        else if (offset == SMODE + 2)
        {
-               serialMode = data;
+//             serialMode = data;
+               smode = data;
                WriteLog("DAC: %s writing to SMODE. Bits: %s%s%s%s%s%s [68K PC=%08X]\n", whoName[who],
                        (data & 0x01 ? "INTERNAL " : ""), (data & 0x02 ? "MODE " : ""),
                        (data & 0x04 ? "WSEN " : ""), (data & 0x08 ? "RISING " : ""),