//
// Originally by David Raingeard
// GCC/SDL port by Niels Wagenaar (Linux/WIN32) and Caz (BeOS)
-// Rewritten by James L. Hammons
+// Rewritten by James Hammons
+// (C) 2010 Underground Software
//
+// JLH = James Hammons <jlhamm@acm.org>
+//
+// Who When What
+// --- ---------- -------------------------------------------------------------
+// JLH 01/16/2010 Created this log ;-)
+//
+
+// Need to set up defaults that the BIOS sets for the SSI here in DACInit()... !!! FIX !!!
+// or something like that... Seems like it already does, but it doesn't seem to
+// 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 "m68k.h"
+//#include "gui.h"
#include "jaguar.h"
+#include "log.h"
+#include "m68k.h"
+//#include "memory.h"
#include "settings.h"
-#include "dac.h"
//#define DEBUG_DAC
// Global variables
-uint16 lrxd, rrxd; // I2S ports (into Jaguar)
+//uint16 lrxd, rrxd; // I2S ports (into Jaguar)
// Local variables
static uint32 LeftFIFOHeadPtr, LeftFIFOTailPtr, RightFIFOHeadPtr, RightFIFOTailPtr;
static SDL_AudioSpec desired;
-static bool SDLSoundInitialized = false;
+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;
+static uint16 DACBuffer[BUFFER_SIZE];
static uint8 SCLKFrequencyDivider = 19; // Default is roughly 22 KHz (20774 Hz in NTSC mode)
/*static*/ uint16 serialMode = 0;
// Private function prototypes
void SDLSoundCallback(void * userdata, Uint8 * buffer, int length);
-int GetCalculatedFrequency(void);
//
// Initialize the SDL sound system
//
void DACInit(void)
{
- memory_malloc_secure((void **)&DACBuffer, BUFFER_SIZE * sizeof(uint16), "DAC buffer");
+ SDLSoundInitialized = false;
+
+ if (!vjs.audioEnabled)
+ {
+ WriteLog("DAC: 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.callback = SDLSoundCallback;
if (SDL_OpenAudio(&desired, NULL) < 0) // NULL means SDL guarantees what we want
- {
-// WriteLog("DAC: Failed to initialize SDL sound. Shutting down!\n");
-// log_done();
-// exit(1);
WriteLog("DAC: Failed to initialize SDL sound...\n");
- }
else
{
SDLSoundInitialized = true;
SDL_CloseAudio();
}
- memory_free(DACBuffer);
+// 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
//
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)
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);
// Could also use (as long as BUFFER_SIZE is a multiple of 2):
// LeftFIFOHeadPtr = (LeftFIFOHeadPtr + numSamplesReady) & (BUFFER_SIZE - 1);
// RightFIFOHeadPtr = (RightFIFOHeadPtr + numSamplesReady) & (BUFFER_SIZE - 1);
-//WriteLog(" -> Left/RightFIFOHeadPtr: %u/%u, Left/RightFIFOTailPtr: %u/%u\n", LeftFIFOHeadPtr, RightFIFOHeadPtr, LeftFIFOTailPtr, RightFIFOTailPtr);
+//WriteLog(" -> Left/RightFIFOHeadPtr: %04X/%04X, Left/RightFIFOTailPtr: %04X/%04X\n", LeftFIFOHeadPtr, RightFIFOHeadPtr, LeftFIFOTailPtr, RightFIFOTailPtr);
}
//Hmm. Seems that the SDL buffer isn't being starved by the DAC buffer...
// else
return systemClockFrequency / (32 * (2 * (SCLKFrequencyDivider + 1)));
}
+static int oldFreq = 0;
+
+void DACSetNewFrequency(int freq)
+{
+ if (freq == oldFreq)
+ return;
+
+ oldFreq = freq;
+
+ // Should do some sanity checking on the frequency...
+
+ if (SDLSoundInitialized)
+ SDL_CloseAudio();
+
+ 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)
+ {
+ 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!
+}
+
//
// LTXD/RTXD/SCLK/SMODE ($F1A148/4C/50/54)
//
{
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...
- while ((LeftFIFOTailPtr + 2) & (BUFFER_SIZE - 1) == LeftFIFOHeadPtr);
+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
}
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);
-/* {
+uint32 spin = 0;
+ while (((RightFIFOTailPtr + 2) & (BUFFER_SIZE - 1)) == RightFIFOHeadPtr)//;
+ {
spin++;
-if (spin == 0x10000000)
+//if ((spin & 0x0FFFFFFF) == 0)
+// WriteLog("Tail=%X, Head=%X, BUFFER_SIZE-1=%X\n", RightFIFOTailPtr, RightFIFOHeadPtr, BUFFER_SIZE - 1);
+
+if (spin == 0xFFFF0000)
{
- WriteLog("\nStuck in right DAC spinlock! Tail=%u, Head=%u\nAborting!\n", RightFIFOTailPtr, RightFIFOHeadPtr);
- log_done();
- exit(0);
+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;
}
- }*/
+ }//*/
-//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");
{
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);
- log_done();
- exit(1);
+// LogDone();
+// exit(1);
+#warning "Reimplement GUICrashGracefully!"
+// GUICrashGracefully("Failed to initialize SDL sound!");
+ return;
}
}
else if (offset == RRXD + 2)
return rrxd;
- return 0xFFFF; // May need SSTAT as well... (but may be a Jaguar II only feature)
+ return 0xFFFF; // May need SSTAT as well... (but may be a Jaguar II only feature)
}
projects / virtualjaguar / blobdiff