X-Git-Url: http://shamusworld.gotdns.org/cgi-bin/gitweb.cgi?a=blobdiff_plain;f=src%2Fdac.cpp;h=e85d858cd4e979698ed3382491fe49920e3e343d;hb=432a82c9d473fc33c4fe6b9d2db60db0d84c6612;hp=8c60b5ff710771c3b83c0801e1b9c32e9d4e78b2;hpb=19197c6b601dbfd415ded7ac4918813e1805ca39;p=virtualjaguar diff --git a/src/dac.cpp b/src/dac.cpp index 8c60b5f..e85d858 100644 --- a/src/dac.cpp +++ b/src/dac.cpp @@ -1,7 +1,7 @@ // // 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 // @@ -9,10 +9,11 @@ #include #include "jaguar.h" #include "dac.h" +#include "settings.h" //#define DEBUG_DAC -#define BUFFER_SIZE 0x8000 // Make the DAC buffers 32K x 16 bits +#define BUFFER_SIZE 0x10000 // Make the DAC buffers 64K x 16 bits // Jaguar memory locations @@ -39,7 +40,7 @@ void SDLSoundCallback(void * userdata, Uint8 * buffer, int length); int GetCalculatedFrequency(void); // -// Initialize the SDL sound system (?) (!) +// Initialize the SDL sound system // void DACInit(void) { @@ -48,7 +49,8 @@ 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 @@ -72,7 +74,7 @@ void DACReset(void) } // -// Close down the SDL sound subsystem (?) (!) +// Close down the SDL sound subsystem // void DACDone(void) { @@ -119,15 +121,15 @@ 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... @@ -140,8 +142,7 @@ 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). @@ -162,41 +163,47 @@ 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(); - } -#ifdef DEBUG_DAC - else - WriteLog("DAC: Ran into FIFO's left tail pointer!\n"); -#endif + // 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(); - } -#ifdef DEBUG_DAC + // 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 +#endif*/ } else if (offset == SCLK + 2) // Sample rate {