//
// 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 "log.h"

// Useful defines

//#define DEBUG
//#define WRITE_OUT_WAVE

//#define SAMPLE_RATE			(44100.0)
#define SAMPLE_RATE			(48000.0)
#define SAMPLES_PER_FRAME	(SAMPLE_RATE / 60.0)
#define CYCLES_PER_SAMPLE	(1024000.0 / SAMPLE_RATE)
//#define SOUND_BUFFER_SIZE	(8192)
#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 int16_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 int16_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 };
#ifdef WRITE_OUT_WAVE
static FILE * fp = NULL;
#endif

// Private function prototypes

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


//
// Initialize the SDL sound system
//
void SoundInit(void)
{
#if 0
// To weed out problems for now...
return;
#endif
	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_S16SYS;				// This uses the native endian (for portability)...
	desired.channels = 1;
	desired.samples = 512;						// Let's try a 1/2K buffer (can always go lower)
	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");

#ifdef WRITE_OUT_WAVE
	fp = fopen("./apple2.wav", "wb");
#endif
}


//
// 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");

#ifdef WRITE_OUT_WAVE
		fclose(fp);
#endif
	}
}


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


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


//
// Sound card callback handler
//
static void SDLSoundCallback(void * /*userdata*/, Uint8 * buffer8, int length8)
{
//WriteLog("SDLSoundCallback(): begin (soundBufferPos=%i)\n", soundBufferPos);
	// 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!)
	// (Should NOT starve now, now that we properly handle frame edges...)

	// 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...
	int16_t * buffer = (int16_t *)buffer8;
	uint32_t length = (uint32_t)length8 / 2;

//WriteLog("SDLSoundCallback(): filling buffer...\n");
	if (soundBufferPos < length)
	{
		// 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;
	}
	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 ~21.333 cycles.
void WriteSampleToBuffer(void)
{
//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...
	}

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


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

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