Added appropriate audio thread pausing to the GUI layer.

[virtualjaguar] / src / dac.cpp

//
// DAC (really, Synchronous Serial Interface) Handler
//
// Originally by David Raingeard
// GCC/SDL port by Niels Wagenaar (Linux/WIN32) and Caz (BeOS)
// Rewritten by James Hammons
// (C) 2010 Underground Software
//
// JLH = James Hammons <jlhamm@acm.org>
//
// 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 !!!
// 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?

// 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 "cdrom.h"
#include "dsp.h"
#include "event.h"
#include "jerry.h"
#include "jaguar.h"
#include "log.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 DAC_AUDIO_RATE          48000                           // Set the audio rate to 48 KHz

// Jaguar memory locations

#define LTXD                    0xF1A148
#define RTXD                    0xF1A14C
#define LRXD                    0xF1A148
#define RRXD                    0xF1A14C
#define SCLK                    0xF1A150
#define SMODE                   0xF1A154

// Global variables

// These are defined in memory.h/cpp
//uint16 lrxd, rrxd;                                                    // I2S ports (into Jaguar)

// Local variables

static SDL_AudioSpec desired;
static bool SDLSoundInitialized;
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);
void DSPSampleCallback(void);


//
// Initialize the SDL sound system
//
void DACInit(void)
{
        SDLSoundInitialized = false;

//      if (!vjs.audioEnabled)
        if (!vjs.DSPEnabled)
        {
                WriteLog("DAC: DSP/host audio playback disabled.\n");
                return;
        }

        desired.freq = DAC_AUDIO_RATE;
        desired.format = AUDIO_S16SYS;
        desired.channels = 2;
        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
                WriteLog("DAC: Failed to initialize SDL sound...\n");
        else
        {
                SDLSoundInitialized = true;
                DACReset();
                SDL_PauseAudio(false);                                  // Start playback!
                WriteLog("DAC: Successfully initialized. Sample rate: %u\n", desired.freq);
        }

        ltxd = lrxd = desired.silence;

        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;
        ltxd = lrxd = desired.silence;
}


//
// Pause/unpause the SDL audio thread
//
void DACPauseAudioThread(bool state/*= true*/)
{
                SDL_PauseAudio(state);
}


//
// Close down the SDL sound subsystem
//
void DACDone(void)
{
        if (SDLSoundInitialized)
        {
                SDL_PauseAudio(true);
                SDL_CloseAudio();
        }

        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)
{
        // 1st, check to see if the DSP is running. If not, fill the buffer with L/RXTD and exit.

        if (!DSPIsRunning())
        {
                for(int i=0; i<(length/2); i+=2)
                {
                        ((uint16_t *)buffer)[i + 0] = ltxd;
                        ((uint16_t *)buffer)[i + 1] = rtxd;
                }

                return;
        }

        // 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) * ();

        // Now, run the DSP for that length of time for each sample we need to make

        bufferIndex = 0;
        sampleBuffer = buffer;
        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!
        do
        {
                double timeToNextEvent = GetTimeToNextEvent(EVENT_JERRY);

                if (vjs.DSPEnabled)
                {
                        if (vjs.usePipelinedDSP)
                                DSPExecP2(USEC_TO_RISC_CYCLES(timeToNextEvent));
                        else
                                DSPExec(USEC_TO_RISC_CYCLES(timeToNextEvent));
                }

                HandleNextEvent(EVENT_JERRY);
        }
        while (!bufferDone);
}


void DSPSampleCallback(void)
{
        ((uint16_t *)sampleBuffer)[bufferIndex + 0] = ltxd;
        ((uint16_t *)sampleBuffer)[bufferIndex + 1] = rtxd;
        bufferIndex += 2;

        if (bufferIndex == numberOfSamples)
        {
                bufferDone = true;
                return;
        }

        SetCallbackTime(DSPSampleCallback, 1000000.0 / (double)DAC_AUDIO_RATE, EVENT_JERRY);
}


#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*/)
{
        WriteLog("DAC: %s writing BYTE %02X at %08X\n", whoName[who], data, offset);
        if (offset == SCLK + 3)
                DACWriteWord(offset - 3, (uint16)data);
}


void DACWriteWord(uint32 offset, uint16 data, uint32 who/*= UNKNOWN*/)
{
        if (offset == LTXD + 2)
        {
                ltxd = data;
        }
        else if (offset == RTXD + 2)
        {
                rtxd = data;
        }
        else if (offset == SCLK + 2)                                    // Sample rate
        {
                WriteLog("DAC: Writing %u to SCLK...\n", data);

                if ((uint8)data != SCLKFrequencyDivider)
                        SCLKFrequencyDivider = (uint8)data;
        }
        else if (offset == SMODE + 2)
        {
                serialMode = 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 " : ""),
                        (data & 0x10 ? "FALLING " : ""), (data & 0x20 ? "EVERYWORD" : ""),
                        m68k_get_reg(NULL, M68K_REG_PC));
        }
}


//
// LRXD/RRXD/SSTAT ($F1A148/4C/50)
//
uint8 DACReadByte(uint32 offset, uint32 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*/)
{
//      WriteLog("DAC: %s reading word from %08X\n", whoName[who], offset);
//      return 0xFFFF;
//      WriteLog("DAC: %s reading WORD %04X from %08X\n", whoName[who], fakeWord, offset);
//      return fakeWord++;
//NOTE: This only works if a bunch of things are set in BUTCH which we currently don't
//      check for. !!! FIX !!!
// Partially fixed: We check for I2SCNTRL in the JERRY I2S routine...
//      return GetWordFromButchSSI(offset, who);
        if (offset == LRXD || offset == RRXD)
                return 0x0000;
        else if (offset == LRXD + 2)
                return lrxd;
        else if (offset == RRXD + 2)
                return rrxd;

        return 0xFFFF;  // May need SSTAT as well... (but may be a Jaguar II only feature)
}