[apple2] / src / sound.cpp

//
// Sound Interface
//
// by James Hammons
// (C) 2005 Underground Software
//
// JLH = James 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]
// - Figure out why it's playing too fast [DONE]
//

#include "sound.h"

#include <string.h>                     // For memset, memcpy
#include <SDL2/SDL.h>
#include "ay8910.h"
#include "log.h"

// Useful defines

//#define DEBUG

#define SAMPLES_PER_FRAME       (SAMPLE_RATE / 60.0)
#define CYCLES_PER_SAMPLE       (1024000.0 / SAMPLE_RATE)
// 32K ought to be enough for anybody
#define SOUND_BUFFER_SIZE       (32768)

// Global variables


// Local variables

static SDL_AudioSpec desired, obtained;
static SDL_AudioDeviceID device;
static bool soundInitialized = false;
static bool speakerState = false;
static uint16_t soundBuffer[SOUND_BUFFER_SIZE];
static uint32_t soundBufferPos;
//static uint64_t lastToggleCycles;
//static SDL_cond * conditional = NULL;
//static SDL_mutex * mutex = NULL;
//static SDL_mutex * mutex2 = NULL;
static uint16_t sample;
static uint8_t ampPtr = 12;                                             // Start with -2047 - +2047
static int16_t amplitude[17] = { 0, 1, 2, 3, 7, 15, 31, 63, 127, 255,
        511, 1023, 2047, 4095, 8191, 16383, 32767 };

// Private function prototypes

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


//
// Initialize the SDL sound system
//
void SoundInit(void)
{
        SDL_zero(desired);
        desired.freq = SAMPLE_RATE;             // SDL will do conversion on the fly, if it can't get the exact rate. Nice!
        desired.format = AUDIO_U16SYS;  // This uses the native endian (for portability)...
        desired.channels = 1;
        desired.samples = 512;                  // Let's try a 1/2K buffer
        desired.callback = SDLSoundCallback;

        device = SDL_OpenAudioDevice(NULL, 0, &desired, &obtained, 0);

        if (device == 0)
        {
                WriteLog("Sound: Failed to initialize SDL sound.\n");
                return;
        }

//      conditional = SDL_CreateCond();
//      mutex = SDL_CreateMutex();
//      mutex2 = SDL_CreateMutex();// Let's try real signalling...
        soundBufferPos = 0;
//      lastToggleCycles = 0;
        sample = desired.silence;       // ? wilwok ? yes

        SDL_PauseAudioDevice(device, 0);                        // Start playback!
        soundInitialized = true;
        WriteLog("Sound: Successfully initialized.\n");
}


//
// Close down the SDL sound subsystem
//
void SoundDone(void)
{
        if (soundInitialized)
        {
                SDL_PauseAudioDevice(device, 1);
                SDL_CloseAudioDevice(device);
//              SDL_DestroyCond(conditional);
//              SDL_DestroyMutex(mutex);
//              SDL_DestroyMutex(mutex2);
                WriteLog("Sound: Done.\n");
        }
}


void SoundPause(void)
{
        if (soundInitialized)
                SDL_PauseAudioDevice(device, 1);
}


void SoundResume(void)
{
        if (soundInitialized)
                SDL_PauseAudioDevice(device, 0);
}


//
// Sound card callback handler
//
static uint32_t sndFrmCnt = 0;
static uint32_t lastStarve = 0;
static void SDLSoundCallback(void * /*userdata*/, Uint8 * buffer8, int length8)
{
sndFrmCnt++;
//WriteLog("SDLSoundCallback(): begin (soundBufferPos=%i)\n", soundBufferPos);

        // Let's try using a mutex for shared resource consumption...
//Actually, I think Lock/UnlockAudio() does this already...
//WriteLog("SDLSoundCallback: soundBufferPos = %i\n", soundBufferPos);
//      SDL_mutexP(mutex2);

        // Recast this as a 16-bit type...
        uint16_t * buffer = (uint16_t *)buffer8;
        uint32_t length = (uint32_t)length8 / 2;

//WriteLog("SDLSoundCallback(): filling buffer...\n");
        if (soundBufferPos < length)
        {
WriteLog("*** Sound buffer starved (%d short) *** [%d delta %d]\n", length - soundBufferPos, sndFrmCnt, sndFrmCnt - lastStarve);
lastStarve = sndFrmCnt;
#if 1
                for(uint32_t i=0; i<length; i++)
                        buffer[i] = desired.silence;
#else
                // The sound buffer is starved...
                for(uint32_t i=0; i<soundBufferPos; i++)
                        buffer[i] = soundBuffer[i];

                // Fill buffer with last value
                for(uint32_t i=soundBufferPos; i<length; i++)
                        buffer[i] = sample;

                // Reset soundBufferPos to start of buffer...
                soundBufferPos = 0;
#endif
        }
        else
        {
                // Fill sound buffer with frame buffered sound
                for(uint32_t i=0; i<length; i++)
                        buffer[i] = soundBuffer[i];

                soundBufferPos -= length;

                // Move current buffer down to start
                for(uint32_t i=0; i<soundBufferPos; i++)
                        soundBuffer[i] = soundBuffer[length + i];
        }

        // Free the mutex...
//WriteLog("SDLSoundCallback(): SDL_mutexV(mutex2)\n");
//      SDL_mutexV(mutex2);
        // Wake up any threads waiting for the buffer to drain...
//      SDL_CondSignal(conditional);
//WriteLog("SDLSoundCallback(): end\n");
}


// This is called by the main CPU thread every ~21.666 cycles.
void WriteSampleToBuffer(void)
{
#ifdef USE_NEW_AY8910
        uint16_t s1 = AYGetSample(0);
        uint16_t s2 = AYGetSample(1);
        uint16_t adjustedMockingboard = s1 + s2;
#else
        int16_t s1, s2, s3, s4, s5, s6;
        int16_t * bufPtrs[6] = { &s1, &s2, &s3, &s4, &s5, &s6 };
        AY8910Update(0, bufPtrs, 1);
        AY8910Update(1, &bufPtrs[3], 1);
        int16_t adjustedMockingboard = (s1 / 8) + (s2 / 8) + (s3 / 8)
                + (s4 / 8) + (s5 / 8) + (s6 / 8);
#endif
//need to do this *before* mixing, as by this time, it's too late and the sample is probably already oversaturated
//      adjustedMockingboard /= 8;

//WriteLog("WriteSampleToBuffer(): SDL_mutexP(mutex2)\n");
//      SDL_mutexP(mutex2);

        // This should almost never happen, but, if it does...
        while (soundBufferPos >= (SOUND_BUFFER_SIZE - 1))
        {
//WriteLog("WriteSampleToBuffer(): Waiting for sound thread. soundBufferPos=%i, SOUNDBUFFERSIZE-1=%i\n", soundBufferPos, SOUND_BUFFER_SIZE-1);
//              SDL_mutexV(mutex2);     // Release it so sound thread can get it,
//              SDL_mutexP(mutex);      // Must lock the mutex for the cond to work properly...
//              SDL_CondWait(conditional, mutex);       // Sleep/wait for the sound thread
//              SDL_mutexV(mutex);      // Must unlock the mutex for the cond to work properly...
//              SDL_mutexP(mutex2);     // Re-lock it until we're done with it...
                SDL_Delay(1);
        }

        SDL_LockAudioDevice(device);
        soundBuffer[soundBufferPos++] = sample + adjustedMockingboard;
        SDL_UnlockAudioDevice(device);

//      soundBuffer[soundBufferPos++] = sample;
//WriteLog("WriteSampleToBuffer(): SDL_mutexV(mutex2)\n");
//      SDL_mutexV(mutex2);
}


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

        speakerState = !speakerState;
        sample = (speakerState ? amplitude[ampPtr] : 0);//-amplitude[ampPtr]);
}


void VolumeUp(void)
{
        // Currently set for 16-bit samples
        if (ampPtr < 16)
                ampPtr++;
}


void VolumeDown(void)
{
        if (ampPtr > 0)
                ampPtr--;
}


uint8_t GetVolume(void)
{
        return ampPtr;
}