Minor refactoring on V65C02 core (fixed EA_ABS_* macros), changed sound to

[apple2] / src / sound.cpp

//
// Sound Interface
//
// by James L. Hammons
// (C) 2005 Underground Software
//
// JLH = James L. Hammons <jlhamm@acm.org>
//
// WHO  WHEN        WHAT
// ---  ----------  ------------------------------------------------------------
// JLH  12/02/2005  Fixed a problem with sound callback thread signaling the
//                  main thread
// JLH  12/03/2005  Fixed sound callback dropping samples when the sample buffer
//                  is shorter than the callback sample buffer
//

// STILL TO DO:
//
// - Figure out why it's losing samples (Bard's Tale) [DONE]
//

#include "sound.h"

#include <string.h>                                                             // For memset, memcpy
#include <SDL.h>
#include "log.h"


#define AMPLITUDE       (32)                                            // -32 - +32 seems to be plenty loud!

// Global variables


// Local variables

static SDL_AudioSpec desired;
static bool soundInitialized = false;
static bool speakerState;
static uint8 soundBuffer[4096];
static uint32 soundBufferPos;
static uint32 sampleBase;
static SDL_cond * conditional = NULL;
static SDL_mutex * mutex = NULL;

// Private function prototypes

static void SDLSoundCallback(void * userdata, Uint8 * buffer, int length);

//
// Initialize the SDL sound system
//
void SoundInit(void)
{
// To weed out problems for now...
#if 0
return;
#endif

        desired.freq = 44100;                                           // SDL will do conversion on the fly, if it can't get the exact rate. Nice!
        desired.format = AUDIO_S8;                                      // This uses the native endian (for portability)...
//      desired.format = AUDIO_S16SYS;                          // This uses the native endian (for portability)...
        desired.channels = 1;
//      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 = 1024;                                         // Let's try a 1K buffer (can always go lower)
        desired.callback = SDLSoundCallback;

        if (SDL_OpenAudio(&desired, NULL) < 0)          // NULL means SDL guarantees what we want
        {
                WriteLog("Sound: Failed to initialize SDL sound.\n");
                return;
        }

        conditional = SDL_CreateCond();
        mutex = SDL_CreateMutex();
        SDL_mutexP(mutex);                                                      // Must lock the mutex for the cond to work properly...
        soundBufferPos = 0;
        sampleBase = 0;

        SDL_PauseAudio(false);                                          // Start playback!
        soundInitialized = true;
        WriteLog("Sound: Successfully initialized.\n");
}

//
// Close down the SDL sound subsystem
//
void SoundDone(void)
{
        if (soundInitialized)
        {
                SDL_PauseAudio(true);
                SDL_CloseAudio();
                SDL_DestroyCond(conditional);
                SDL_DestroyMutex(mutex);
                WriteLog("Sound: Done.\n");
        }
}

//
// Sound card callback handler
//
static void SDLSoundCallback(void * userdata, Uint8 * buffer, int length)
{
        // The sound buffer should only starve when starting which will cause it to
        // lag behind the emulation at most by around 1 frame...
        // (Actually, this should never happen since we fill the buffer beforehand.)
        // (But, then again, if the sound hasn't been toggled for a while, then this
        //  makes perfect sense as the buffer won't have been filled at all!)

        if (soundBufferPos < (uint32)length)            // The sound buffer is starved...
        {
//printf("Sound buffer starved!\n");
//fflush(stdout);
                for(uint32 i=0; i<soundBufferPos; i++)
                        buffer[i] = soundBuffer[i];

                // Fill buffer with last value
                memset(buffer + soundBufferPos, (uint8)(speakerState ? AMPLITUDE : -AMPLITUDE), length - soundBufferPos);
                soundBufferPos = 0;                                             // Reset soundBufferPos to start of buffer...
                sampleBase = 0;                                                 // & sampleBase...
//Ick. This should never happen!
SDL_CondSignal(conditional);                    // Wake up any threads waiting for the buffer to drain...
                return;                                                                 // & bail!
        }

        memcpy(buffer, soundBuffer, length);            // Fill sound buffer with frame buffered sound
        soundBufferPos -= length;
        sampleBase -= length;

//      if (soundBufferPos > 0)
//              memcpy(soundBuffer, soundBuffer + length, soundBufferPos);      // Move current buffer down to start
//      memcpy(soundBuffer, soundBuffer + length, length);
        // Move current buffer down to start
        for(uint32 i=0; i<soundBufferPos; i++)
                soundBuffer[i] = soundBuffer[length + i];

//      lastValue = buffer[length - 1];
        SDL_CondSignal(conditional);                            // Wake up any threads waiting for the buffer to drain...
}

// Need some interface functions here to take care of flipping the
// waveform at the correct time in the sound stream...

/*
Maybe set up a buffer 1 frame long (44100 / 60 = 735 bytes per frame)

Hmm. That's smaller than the sound buffer 2048 bytes... (About 2.75 frames needed to fill)

So... I guess what we could do is this:

-- Execute V65C02 for one frame. The read/writes at I/O address $C030 fill up the buffer
   to the current time position.
-- The sound callback function copies the pertinent area out of the buffer, resets
   the time position back (or copies data down from what it took out)
*/

void ToggleSpeaker(uint32 time)
{
        if (!soundInitialized)
                return;

#if 0
if (time > 95085)//(time & 0x80000000)
{
        WriteLog("ToggleSpeaker() given bad time value: %08X (%u)!\n", time, time);
//      fflush(stdout);
}
#endif

        // 1.024 MHz / 60 = 17066.6... cycles (23.2199 cycles per sample)
        // Need the last frame position in order to calculate correctly...
        // (or do we?)

        SDL_LockAudio();
        uint32 currentPos = sampleBase + (uint32)((double)time / 23.2199);

        if (currentPos > 4095)
        {
#if 0
WriteLog("ToggleSpeaker() about to go into spinlock at time: %08X (%u) (sampleBase=%u)!\n", time, time, sampleBase);
#endif
// Still hanging on this spinlock...
// That could be because the "time" value is too high and so the buffer will NEVER be
// empty enough...
// Now that we're using a conditional, it seems to be working OK--though not perfectly...
/*
ToggleSpeaker() about to go into spinlock at time: 00004011 (16401) (sampleBase=3504)!
16401 -> 706 samples, 3504 + 706 = 4210

And it still thrashed the sound even though it didn't run into a spinlock...

Seems like it's OK now that I've fixed the buffer-less-than-length bug...
*/
                SDL_UnlockAudio();
                SDL_CondWait(conditional, mutex);
                SDL_LockAudio();
                currentPos = sampleBase + (uint32)((double)time / 23.2199);
#if 0
WriteLog("--> after spinlock (sampleBase=%u)...\n", sampleBase);
#endif
        }

        int8 sample = (speakerState ? AMPLITUDE : -AMPLITUDE);

        while (soundBufferPos < currentPos)
                soundBuffer[soundBufferPos++] = (uint8)sample;

        speakerState = !speakerState;
        SDL_UnlockAudio();
}

void HandleSoundAtFrameEdge(void)
{
        if (!soundInitialized)
                return;

        SDL_LockAudio();
        sampleBase += 735;
        SDL_UnlockAudio();
}