X-Git-Url: http://shamusworld.gotdns.org/cgi-bin/gitweb.cgi?a=blobdiff_plain;f=src%2Fdac.cpp;h=4e35d3bbb0b0e3e540a37c17bce2c49666cff63f;hb=4ff423c8b9d28ce594f33574959f4fc78a4d417c;hp=2ae3e69595f8f4708fb8caef3a106b895ad9aed4;hpb=67a5f1a40072983cf87ae2093ca95c271d14e706;p=virtualjaguar diff --git a/src/dac.cpp b/src/dac.cpp index 2ae3e69..4e35d3b 100644 --- a/src/dac.cpp +++ b/src/dac.cpp @@ -1,16 +1,20 @@ // // DAC (really, Synchronous Serial Interface) Handler // -// by cal2 +// Original by Cal2 // GCC/SDL port by Niels Wagenaar (Linux/WIN32) and Caz (BeOS) // Rewritten by James L. Hammons // -#include +//#include +#include "SDL.h" #include "jaguar.h" +#include "settings.h" #include "dac.h" -#define BUFFER_SIZE 0x8000 // Make the DAC buffers 32K x 16 bits +//#define DEBUG_DAC + +#define BUFFER_SIZE 0x10000 // Make the DAC buffers 64K x 16 bits // Jaguar memory locations @@ -37,7 +41,7 @@ void SDLSoundCallback(void * userdata, Uint8 * buffer, int length); int GetCalculatedFrequency(void); // -// Initialize the SDL sound system (?) (!) +// Initialize the SDL sound system // void DACInit(void) { @@ -46,7 +50,8 @@ void DACInit(void) 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.channels = 2; - desired.samples = 4096; // Let's try a 4K buffer (can always go lower) +// 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.callback = SDLSoundCallback; if (SDL_OpenAudio(&desired, NULL) < 0) // NULL means SDL guarantees what we want @@ -70,12 +75,13 @@ void DACReset(void) } // -// Close down the SDL sound subsystem (?) (!) +// Close down the SDL sound subsystem // void DACDone(void) { SDL_PauseAudio(true); SDL_CloseAudio(); + memory_free(DACBuffer); WriteLog("DAC: Done.\n"); } @@ -117,15 +123,15 @@ void SDLSoundCallback(void * userdata, Uint8 * buffer, int length) // 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: %u/%u, Left/RightFIFOTailPtr: %u/%u\n", LeftFIFOHeadPtr, RightFIFOHeadPtr, LeftFIFOTailPtr, RightFIFOTailPtr); } //Hmm. Seems that the SDL buffer isn't being starved by the DAC buffer... @@ -138,8 +144,7 @@ void SDLSoundCallback(void * userdata, Uint8 * buffer, int length) // int GetCalculatedFrequency(void) { - extern bool hardwareTypeNTSC; - int systemClockFrequency = (hardwareTypeNTSC ? RISC_CLOCK_RATE_NTSC : RISC_CLOCK_RATE_PAL); + 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). @@ -160,37 +165,47 @@ void DACWriteWord(uint32 offset, uint16 data) { if (offset == LTXD + 2) { - if (LeftFIFOTailPtr + 2 != LeftFIFOHeadPtr) - { - 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; -// Aaron's code does this, but I don't know why... -//Flipping this bit makes the audio MUCH louder. Need to look at the amplitude of the -//waveform to see if any massaging is needed here... -//Looks like a cheap & dirty way to convert signed samples to unsigned... -// DACBuffer[LeftFIFOTailPtr] = data ^ 0x8000; - SDL_UnlockAudio(); - } - else - WriteLog("DAC: Ran into FIFO's left tail pointer!\n"); + // 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... + while ((LeftFIFOTailPtr + 2) & (BUFFER_SIZE - 1) == LeftFIFOHeadPtr); + + 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(); } else if (offset == RTXD + 2) { - if (RightFIFOTailPtr + 2 != RightFIFOHeadPtr) - { - SDL_LockAudio(); - RightFIFOTailPtr = (RightFIFOTailPtr + 2) % BUFFER_SIZE; - DACBuffer[RightFIFOTailPtr] = data; -// Aaron's code does this, but I don't know why... -// DACBuffer[RightFIFOTailPtr] = data ^ 0x8000; - SDL_UnlockAudio(); - } + // Spin until buffer has been drained (for too fast processors!)... +//uint32 spin = 0; + while ((RightFIFOTailPtr + 2) & (BUFFER_SIZE - 1) == RightFIFOHeadPtr); +/* { +spin++; +if (spin == 0x10000000) +{ + WriteLog("\nStuck in right DAC spinlock! Tail=%u, Head=%u\nAborting!\n", RightFIFOTailPtr, RightFIFOHeadPtr); + log_done(); + exit(0); +} + }*/ + +//This is wrong if (RightFIFOTailPtr + 2 != RightFIFOHeadPtr) +// { + 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*/ } else if (offset == SCLK + 2) // Sample rate { @@ -199,7 +214,7 @@ void DACWriteWord(uint32 offset, uint16 data) { SCLKFrequencyDivider = (uint8)data; //Of course a better way would be to query the hardware to find the upper limit... - if (data > 7) // Anything less is too high! + if (data > 7) // Anything less than 8 is too high! { SDL_CloseAudio(); desired.freq = GetCalculatedFrequency();// SDL will do conversion on the fly, if it can't get the exact rate. Nice!