X-Git-Url: http://shamusworld.gotdns.org/cgi-bin/gitweb.cgi?a=blobdiff_plain;f=src%2Fdac.cpp;h=f08295448ceca78c4392aeb38c3461b1d680be55;hb=7228359373eb7602c26f7b098d6b2271ff5727a1;hp=617918a13f725f248f9403a5bf6741706a77e4a8;hpb=f30bf746981a99079e766b0d4e9de5391a4175ff;p=virtualjaguar diff --git a/src/dac.cpp b/src/dac.cpp index 617918a..f082954 100644 --- a/src/dac.cpp +++ b/src/dac.cpp @@ -11,6 +11,7 @@ // Who When What // --- ---------- ------------------------------------------------------------- // JLH 01/16/2010 Created this log ;-) +// JLH 04/30/2012 Changed SDL audio handler to run JERRY // // Need to set up defaults that the BIOS sets for the SSI here in DACInit()... !!! FIX !!! @@ -18,27 +19,45 @@ // work correctly...! Perhaps just need to set up SSI stuff so BUTCH doesn't get // confused... -// ALSO: Need to implement some form of proper locking to replace the clusterfuck -// that is the current spinlock implementation. Since the DSP is a separate -// entity, could we get away with running it in the sound IRQ? - -// ALSO: It may be a good idea to physically separate the left and right buffers -// to prevent things like the DSP filling only one side and such. Do such -// mono modes exist on the Jag? Seems to according to Super Burnout. +// After testing on a real Jaguar, it seems clear that the I2S interrupt drives +// the audio subsystem. So while you can drive the audio at a *slower* rate than +// set by SCLK, you can't drive it any *faster*. Also note, that if the I2S +// interrupt is not enabled/running on the DSP, then there is no audio. Also, +// audio can be muted by clearing bit 8 of JOYSTICK (JOY1). +// +// Approach: We can run the DSP in the host system's audio IRQ, by running the +// DSP for the alloted time (depending on the host buffer size & sample rate) +// by simply reading the L/R_I2S (L/RTXD) registers at regular intervals. We +// would also have to time the I2S/TIMER0/TIMER1 interrupts in the DSP as well. +// This way, we can run the host audio IRQ at, say, 48 KHz and not have to care +// so much about SCLK and running a separate buffer and all the attendant +// garbage that comes with that awful approach. +// +// There would still be potential gotchas, as the SCLK can theoretically drive +// the I2S at 26590906 / 2 (for SCLK == 0) = 13.3 MHz which corresponds to an +// audio rate 416 KHz (dividing the I2S rate by 32, for 16-bit stereo). It +// seems doubtful that anything useful could come of such a high rate, and we +// can probably safely ignore any such ridiculously high audio rates. It won't +// sound the same as on a real Jaguar, but who cares? :-) #include "dac.h" #include "SDL.h" -//#include "gui.h" +#include "cdrom.h" +#include "dsp.h" +#include "event.h" +#include "jerry.h" #include "jaguar.h" #include "log.h" -#include "m68k.h" +#include "m68000/m68kinterface.h" //#include "memory.h" #include "settings.h" + //#define DEBUG_DAC -#define BUFFER_SIZE 0x10000 // Make the DAC buffers 64K x 16 bits +#define BUFFER_SIZE 0x10000 // Make the DAC buffers 64K x 16 bits +#define DAC_AUDIO_RATE 48000 // Set the audio rate to 48 KHz // Jaguar memory locations @@ -51,24 +70,21 @@ // Global variables -//uint16 lrxd, rrxd; // I2S ports (into Jaguar) +// These are defined in memory.h/cpp +//uint16_t lrxd, rrxd; // I2S ports (into Jaguar) // Local variables -static uint32 LeftFIFOHeadPtr, LeftFIFOTailPtr, RightFIFOHeadPtr, RightFIFOTailPtr; static SDL_AudioSpec desired; static bool SDLSoundInitialized; - -// We can get away with using native endian here because we can tell SDL to use the native -// endian when looking at the sample buffer, i.e., no need to worry about it. - -static uint16 DACBuffer[BUFFER_SIZE]; -static uint8 SCLKFrequencyDivider = 19; // Default is roughly 22 KHz (20774 Hz in NTSC mode) -/*static*/ uint16 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 void SDLSoundCallback(void * userdata, Uint8 * buffer, int length); +void DSPSampleCallback(void); + // // Initialize the SDL sound system @@ -77,41 +93,57 @@ void DACInit(void) { SDLSoundInitialized = false; - if (!vjs.audioEnabled) +// if (!vjs.audioEnabled) + if (!vjs.DSPEnabled) { - WriteLog("DAC: Host audio playback disabled.\n"); + WriteLog("DAC: DSP/host audio playback disabled.\n"); return; } -// memory_malloc_secure((void **)&DACBuffer, BUFFER_SIZE * sizeof(uint16), "DAC buffer"); -// DACBuffer = (uint16 *)memory_malloc(BUFFER_SIZE * sizeof(uint16), "DAC buffer"); - - desired.freq = GetCalculatedFrequency(); // SDL will do conversion on the fly, if it can't get the exact rate. Nice! - desired.format = AUDIO_S16SYS; // This uses the native endian (for portability)... + desired.freq = DAC_AUDIO_RATE; + desired.format = AUDIO_S16SYS; desired.channels = 2; -// desired.samples = 4096; // Let's try a 4K buffer (can always go lower) - desired.samples = 2048; // Let's try a 2K buffer (can always go lower) + desired.samples = 2048; // 2K buffer = audio delay of 42.67 ms (@ 48 KHz) desired.callback = SDLSoundCallback; - if (SDL_OpenAudio(&desired, NULL) < 0) // NULL means SDL guarantees what we want + if (SDL_OpenAudio(&desired, NULL) < 0) // NULL means SDL guarantees what we want WriteLog("DAC: Failed to initialize SDL sound...\n"); else { SDLSoundInitialized = true; DACReset(); - SDL_PauseAudio(false); // Start playback! - WriteLog("DAC: Successfully initialized.\n"); + SDL_PauseAudio(false); // Start playback! + WriteLog("DAC: Successfully initialized. Sample rate: %u\n", desired.freq); } + + 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; + WriteLog("DAC: RISC clock = %u, cyclesPerSample = %u\n", riscClockRate, cyclesPerSample); } + // // Reset the sound buffer FIFOs // void DACReset(void) { - LeftFIFOHeadPtr = LeftFIFOTailPtr = 0, RightFIFOHeadPtr = RightFIFOTailPtr = 1; +// LeftFIFOHeadPtr = LeftFIFOTailPtr = 0, RightFIFOHeadPtr = RightFIFOTailPtr = 1; + ltxd = lrxd = desired.silence; +} + + +// +// Pause/unpause the SDL audio thread +// +void DACPauseAudioThread(bool state/*= true*/) +{ + SDL_PauseAudio(state); } + // // Close down the SDL sound subsystem // @@ -123,309 +155,146 @@ void DACDone(void) SDL_CloseAudio(); } -// memory_free(DACBuffer); WriteLog("DAC: Done.\n"); } + +// Approach: Run the DSP for however many cycles needed to correspond to whatever sample rate +// we've set the audio to run at. So, e.g., if we run it at 48 KHz, then we would run the DSP +// for however much time it takes to fill the buffer. So with a 2K buffer, this would correspond +// to running the DSP for 0.042666... seconds. At 26590906 Hz, this would correspond to +// running the DSP for 1134545 cycles. You would then sample the L/RTXD registers every +// 1134545 / 2048 = 554 cycles to fill the buffer. You would also have to manage interrupt +// timing as well (generating them at the proper times), but that shouldn't be too difficult... +// If the DSP isn't running, then fill the buffer with L/RTXD and exit. + // // SDL callback routine to fill audio buffer // // Note: The samples are packed in the buffer in 16 bit left/16 bit right pairs. +// Also, length is the length of the buffer in BYTES // +static Uint8 * sampleBuffer; +static int bufferIndex = 0; +static int numberOfSamples = 0; +static bool bufferDone = false; void SDLSoundCallback(void * userdata, Uint8 * buffer, int length) { - // Clear the buffer to silence, in case the DAC buffer is empty (or short) -//This causes choppy sound... Ick. - memset(buffer, desired.silence, length); -//WriteLog("DAC: Inside callback...\n"); - if (LeftFIFOHeadPtr != LeftFIFOTailPtr) + // 1st, check to see if the DSP is running. If not, fill the buffer with L/RXTD and exit. + + if (!DSPIsRunning()) { -//WriteLog("DAC: About to write some data!\n"); - int numLeftSamplesReady - = (LeftFIFOTailPtr + (LeftFIFOTailPtr < LeftFIFOHeadPtr ? BUFFER_SIZE : 0)) - - LeftFIFOHeadPtr; - int numRightSamplesReady - = (RightFIFOTailPtr + (RightFIFOTailPtr < RightFIFOHeadPtr ? BUFFER_SIZE : 0)) - - RightFIFOHeadPtr; -//This waits for the slower side to catch up. If writing only one side, then this -//causes the buffer not to drain... - int numSamplesReady - = (numLeftSamplesReady < numRightSamplesReady - ? numLeftSamplesReady : numRightSamplesReady);//Hmm. * 2; - -//Kludge, until I can figure out WTF is going on WRT Super Burnout. -if (numLeftSamplesReady == 0 || numRightSamplesReady == 0) - numSamplesReady = numLeftSamplesReady + numRightSamplesReady; - -//The numbers look good--it's just that the DSP can't get enough samples in the DAC buffer! -//WriteLog("DAC: Left/RightFIFOHeadPtr: %u/%u, Left/RightFIFOTailPtr: %u/%u\n", LeftFIFOHeadPtr, RightFIFOHeadPtr, LeftFIFOTailPtr, RightFIFOTailPtr); -//WriteLog(" numLeft/RightSamplesReady: %i/%i, numSamplesReady: %i, length of buffer: %i\n", numLeftSamplesReady, numRightSamplesReady, numSamplesReady, length); - -/* if (numSamplesReady > length) - numSamplesReady = length;//*/ - if (numSamplesReady > length / 2) // length / 2 because we're comparing 16-bit lengths - numSamplesReady = length / 2; -//else -// WriteLog(" Not enough samples to fill the buffer (short by %u L/R samples)...\n", (length / 2) - numSamplesReady); -//WriteLog("DAC: %u samples ready.\n", numSamplesReady); - - // Actually, it's a bit more involved than this, but this is the general idea: -// memcpy(buffer, DACBuffer, length); - for(int i=0; i Left/RightFIFOHeadPtr: %04X/%04X, Left/RightFIFOTailPtr: %04X/%04X\n", LeftFIFOHeadPtr, RightFIFOHeadPtr, LeftFIFOTailPtr, RightFIFOTailPtr); + for(int i=0; i<(length/2); i+=2) + { + ((uint16_t *)buffer)[i + 0] = ltxd; + ((uint16_t *)buffer)[i + 1] = rtxd; + } + + return; } -//Hmm. Seems that the SDL buffer isn't being starved by the DAC buffer... -// else -// WriteLog("DAC: Silence...!\n"); -} -// -// Calculate the frequency of SCLK * 32 using the divider -// -int GetCalculatedFrequency(void) -{ - int systemClockFrequency = (vjs.hardwareTypeNTSC ? RISC_CLOCK_RATE_NTSC : RISC_CLOCK_RATE_PAL); + // 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; + // This is the length of time +// timePerSample = (1000000.0 / (double)riscClockRate) * (); - // We divide by 32 here in order to find the frequency of 32 SCLKs in a row (transferring - // 16 bits of left data + 16 bits of right data = 32 bits, 1 SCLK = 1 bit transferred). - return systemClockFrequency / (32 * (2 * (SCLKFrequencyDivider + 1))); -} + // Now, run the DSP for that length of time for each sample we need to make -static int oldFreq = 0; + 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; -void DACSetNewFrequency(int freq) -{ - if (freq == oldFreq) - return; + SetCallbackTime(DSPSampleCallback, 1000000.0 / (double)DAC_AUDIO_RATE, EVENT_JERRY); - oldFreq = freq; + // These timings are tied to NTSC, need to fix that in event.cpp/h! [FIXED] + do + { + double timeToNextEvent = GetTimeToNextEvent(EVENT_JERRY); - // Should do some sanity checking on the frequency... + if (vjs.DSPEnabled) + { + if (vjs.usePipelinedDSP) + DSPExecP2(USEC_TO_RISC_CYCLES(timeToNextEvent)); + else + DSPExec(USEC_TO_RISC_CYCLES(timeToNextEvent)); + } - if (SDLSoundInitialized) - SDL_CloseAudio(); + HandleNextEvent(EVENT_JERRY); + } + while (!bufferDone); +} - desired.freq = freq;// SDL will do conversion on the fly, if it can't get the exact rate. Nice! - WriteLog("DAC: Changing sample rate to %u Hz!\n", desired.freq); - if (SDLSoundInitialized) +void DSPSampleCallback(void) +{ + ((uint16_t *)sampleBuffer)[bufferIndex + 0] = ltxd; + ((uint16_t *)sampleBuffer)[bufferIndex + 1] = rtxd; + bufferIndex += 2; + + if (bufferIndex == numberOfSamples) { - if (SDL_OpenAudio(&desired, NULL) < 0) // NULL means SDL guarantees what we want - { -// This is bad, Bad, BAD !!! DON'T ABORT BECAUSE WE DIDN'T GET OUR FREQ! !!! FIX !!! -#warning !!! FIX !!! Aborting because of SDL audio problem is bad! - WriteLog("DAC: Failed to initialize SDL sound: %s.\nDesired freq: %u\nShutting down!\n", SDL_GetError(), desired.freq); -// LogDone(); -// exit(1); -#warning "Reimplement GUICrashGracefully!" -// GUICrashGracefully("Failed to initialize SDL sound!"); - return; - } + bufferDone = true; + return; } - DACReset(); + SetCallbackTime(DSPSampleCallback, 1000000.0 / (double)DAC_AUDIO_RATE, EVENT_JERRY); +} - if (SDLSoundInitialized) - SDL_PauseAudio(false); // Start playback! + +#if 0 +// +// Calculate the frequency of SCLK * 32 using the divider +// +int GetCalculatedFrequency(void) +{ + int systemClockFrequency = (vjs.hardwareTypeNTSC ? RISC_CLOCK_RATE_NTSC : RISC_CLOCK_RATE_PAL); + + // We divide by 32 here in order to find the frequency of 32 SCLKs in a row (transferring + // 16 bits of left data + 16 bits of right data = 32 bits, 1 SCLK = 1 bit transferred). + return systemClockFrequency / (32 * (2 * (SCLKFrequencyDivider + 1))); } +#endif + // // LTXD/RTXD/SCLK/SMODE ($F1A148/4C/50/54) // -void DACWriteByte(uint32 offset, uint8 data, uint32 who/*= UNKNOWN*/) +void DACWriteByte(uint32_t offset, uint8_t data, uint32_t who/*= UNKNOWN*/) { WriteLog("DAC: %s writing BYTE %02X at %08X\n", whoName[who], data, offset); if (offset == SCLK + 3) - DACWriteWord(offset - 3, (uint16)data); + DACWriteWord(offset - 3, (uint16_t)data); } -void DACWriteWord(uint32 offset, uint16 data, uint32 who/*= UNKNOWN*/) + +void DACWriteWord(uint32_t offset, uint16_t data, uint32_t who/*= UNKNOWN*/) { if (offset == LTXD + 2) { - if (!SDLSoundInitialized) - return; - // Spin until buffer has been drained (for too fast processors!)... -//Small problem--if Head == 0 and Tail == buffer end, then this will fail... !!! FIX !!! -//[DONE] - // Also, we're taking advantage of the fact that the buffer is a multiple of two - // in this check... -uint32 spin = 0; - while (((LeftFIFOTailPtr + 2) & (BUFFER_SIZE - 1)) == LeftFIFOHeadPtr)//; - { -spin++; -//if ((spin & 0x0FFFFFFF) == 0) -// WriteLog("Tail=%X, Head=%X, BUFFER_SIZE-1=%X\n", RightFIFOTailPtr, RightFIFOHeadPtr, BUFFER_SIZE - 1); - -if (spin == 0xFFFF0000) -{ -uint32 ltail = LeftFIFOTailPtr, lhead = LeftFIFOHeadPtr; -WriteLog("Tail=%X, Head=%X", ltail, lhead); - - WriteLog("\nStuck in left DAC spinlock! Aborting!\n"); - WriteLog("LTail=%X, LHead=%X, BUFFER_SIZE-1=%X\n", LeftFIFOTailPtr, LeftFIFOHeadPtr, BUFFER_SIZE - 1); - WriteLog("RTail=%X, RHead=%X, BUFFER_SIZE-1=%X\n", RightFIFOTailPtr, RightFIFOHeadPtr, BUFFER_SIZE - 1); - WriteLog("From while: Tail=%X, Head=%X", (LeftFIFOTailPtr + 2) & (BUFFER_SIZE - 1), LeftFIFOHeadPtr); -// LogDone(); -// exit(0); -#warning "Reimplement GUICrashGracefully!" -// GUICrashGracefully("Stuck in left DAC spinlock!"); - return; -} - }//*/ - - SDL_LockAudio(); // Is it necessary to do this? Mebbe. - // We use a circular buffer 'cause it's easy. Note that the callback function - // takes care of dumping audio to the soundcard...! Also note that we're writing - // the samples in the buffer in an interleaved L/R format. - LeftFIFOTailPtr = (LeftFIFOTailPtr + 2) % BUFFER_SIZE; - DACBuffer[LeftFIFOTailPtr] = data; - SDL_UnlockAudio(); + ltxd = data; } else if (offset == RTXD + 2) { - if (!SDLSoundInitialized) - return; -/* -Here's what's happening now: - -Stuck in right DAC spinlock! -Aborting! - -Tail=681, Head=681, BUFFER_SIZE-1=FFFF -From while: Tail=683, Head=681 - -????? What the FUCK ????? - -& when I uncomment the lines below spin++; it *doesn't* lock here... WTF????? - -I think it was missing parentheses causing the fuckup... Seems to work now... - -Except for Super Burnout now...! Aarrrgggghhhhh! - -Tail=AC, Head=AE -Stuck in left DAC spinlock! Aborting! -Tail=AC, Head=AE, BUFFER_SIZE-1=FFFF -From while: Tail=AE, Head=AE - -So it's *really* stuck here in the left FIFO. Figure out why!!! - -Prolly 'cause it doesn't set the sample rate right away--betcha it works with the BIOS... -It gets farther, but then locks here (weird!): - -Tail=2564, Head=2566 -Stuck in left DAC spinlock! Aborting! -Tail=2564, Head=2566, BUFFER_SIZE-1=FFFF -From while: Tail=2566, Head=2566 - -Weird--recompile with more WriteLog() entries and it *doesn't* lock... -Yeah, because there was no DSP running. Duh! - -Tail=AC, Head=AE -Stuck in left DAC spinlock! Aborting! -LTail=AC, LHead=AE, BUFFER_SIZE-1=FFFF -RTail=AF, RHead=AF, BUFFER_SIZE-1=FFFF -From while: Tail=AE, Head=AE - -Odd: The right FIFO is empty, but the left FIFO is full! -And this is what is causing the lockup--the DAC callback waits for the side with -less samples ready and in this case it's the right channel (that never fills up) -that it's waiting for...! - -Okay, with the kludge in place for the right channel not being filled, we select -a track and then it locks here: - -Tail=60D8, Head=60DA -Stuck in left DAC spinlock! Aborting! -LTail=60D8, LHead=60D8, BUFFER_SIZE-1=FFFF -RTail=DB, RHead=60D9, BUFFER_SIZE-1=FFFF -From while: Tail=60DA, Head=60D8 -*/ -#warning Spinlock problem--!!! FIX !!! -#warning Odd: The right FIFO is empty, but the left FIFO is full! - // Spin until buffer has been drained (for too fast processors!)... -uint32 spin = 0; - while (((RightFIFOTailPtr + 2) & (BUFFER_SIZE - 1)) == RightFIFOHeadPtr)//; - { -spin++; -//if ((spin & 0x0FFFFFFF) == 0) -// WriteLog("Tail=%X, Head=%X, BUFFER_SIZE-1=%X\n", RightFIFOTailPtr, RightFIFOHeadPtr, BUFFER_SIZE - 1); - -if (spin == 0xFFFF0000) -{ -uint32 rtail = RightFIFOTailPtr, rhead = RightFIFOHeadPtr; -WriteLog("Tail=%X, Head=%X", rtail, rhead); - - WriteLog("\nStuck in right DAC spinlock! Aborting!\n"); - WriteLog("LTail=%X, LHead=%X, BUFFER_SIZE-1=%X\n", LeftFIFOTailPtr, LeftFIFOHeadPtr, BUFFER_SIZE - 1); - WriteLog("RTail=%X, RHead=%X, BUFFER_SIZE-1=%X\n", RightFIFOTailPtr, RightFIFOHeadPtr, BUFFER_SIZE - 1); - WriteLog("From while: Tail=%X, Head=%X", (RightFIFOTailPtr + 2) & (BUFFER_SIZE - 1), RightFIFOHeadPtr); -// LogDone(); -// exit(0); -#warning "Reimplement GUICrashGracefully!" -// GUICrashGracefully("Stuck in right DAC spinlock!"); - return; -} - }//*/ - - SDL_LockAudio(); - RightFIFOTailPtr = (RightFIFOTailPtr + 2) % BUFFER_SIZE; - DACBuffer[RightFIFOTailPtr] = data; - SDL_UnlockAudio(); -/*#ifdef DEBUG_DAC - else - WriteLog("DAC: Ran into FIFO's right tail pointer!\n"); -#endif*/ + rtxd = data; } else if (offset == SCLK + 2) // Sample rate { - WriteLog("DAC: Writing %u to SCLK...\n", data); - if ((uint8)data != SCLKFrequencyDivider) - { - SCLKFrequencyDivider = (uint8)data; -//Of course a better way would be to query the hardware to find the upper limit... - if (data > 7) // Anything less than 8 is too high! - { - if (SDLSoundInitialized) - SDL_CloseAudio(); - - desired.freq = GetCalculatedFrequency();// SDL will do conversion on the fly, if it can't get the exact rate. Nice! - WriteLog("DAC: Changing sample rate to %u Hz!\n", desired.freq); - - if (SDLSoundInitialized) - { - if (SDL_OpenAudio(&desired, NULL) < 0) // NULL means SDL guarantees what we want - { -// This is bad, Bad, BAD !!! DON'T ABORT BECAUSE WE DIDN'T GET OUR FREQ! !!! FIX !!! -#warning !!! FIX !!! Aborting because of SDL audio problem is bad! - WriteLog("DAC: Failed to initialize SDL sound: %s.\nDesired freq: %u\nShutting down!\n", SDL_GetError(), desired.freq); -// LogDone(); -// exit(1); -#warning "Reimplement GUICrashGracefully!" -// GUICrashGracefully("Failed to initialize SDL sound!"); - return; - } - } - - DACReset(); - - if (SDLSoundInitialized) - SDL_PauseAudio(false); // Start playback! - } - } + WriteLog("DAC: Writing %u to SCLK (by %s)...\n", data, whoName[who]); + + 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 " : ""), @@ -434,17 +303,19 @@ WriteLog("Tail=%X, Head=%X", rtail, rhead); } } + // // LRXD/RRXD/SSTAT ($F1A148/4C/50) // -uint8 DACReadByte(uint32 offset, uint32 who/*= UNKNOWN*/) +uint8_t DACReadByte(uint32_t offset, uint32_t who/*= UNKNOWN*/) { // WriteLog("DAC: %s reading byte from %08X\n", whoName[who], offset); return 0xFF; } -//static uint16 fakeWord = 0; -uint16 DACReadWord(uint32 offset, uint32 who/*= UNKNOWN*/) + +//static uint16_t fakeWord = 0; +uint16_t DACReadWord(uint32_t offset, uint32_t who/*= UNKNOWN*/) { // WriteLog("DAC: %s reading word from %08X\n", whoName[who], offset); // return 0xFFFF; @@ -463,3 +334,4 @@ uint16 DACReadWord(uint32 offset, uint32 who/*= UNKNOWN*/) return 0xFFFF; // May need SSTAT as well... (but may be a Jaguar II only feature) } +