/*
 * Audio renderer
 */

#include <ultra64.h>
#include <PR/libaudio.h>
#include <PR/sched.h>
#include "viewer.h"
#include "audrenderer.h"

#define MAX_UPDATES             32
#define MAX_EVENTS              32
#define AUDIO_HEAP_SIZE         200000

#define MAX_VOICES              16
#define EXTRA_SAMPLES           80
#define NUM_OUTPUT_BUFFERS      3      /* Need three of these */
#define OUTPUT_RATE             44100
#define AUD_MAX_MESGS               8
#define QUIT_MSG                10

#define NUM_ACMD_LISTS          2      /* two lists used by this example                */
#define MAX_RSP_CMDS            4096   /* max number of commands in any command list.   */
                                       /* Mainly dependent on sequences used            */

#define FRAME_LAG               1      /* The number of frames to keep a dma buffer.    */
                                       /* Increasing this number causes buffers to not  */
                                       /* be deleted as quickly. This results in fewer  */
                                       /* DMA's but you need more buffers.              */

#define AUDIO_STACKSIZE         0x2000

typedef struct {
    u32       outputRate;
    u32       framesPerField;
    u32       maxACMDSize;
} AmConfig;


/****  type define's for structures unique to audiomgr ****/
typedef union {    
    struct {
        short     type;
    } gen;
    struct {
        short     type;
        struct    AudioInfo_s *info;
    } done;
    OSScMsg       app;
} AudioMsg;

typedef struct AudioInfo_s {
    short         *data;          /* Output data pointer */
    short         frameSamples;   /* # of samples synthesized in this frame */
    OSScTask      task;           /* scheduler structure */
    AudioMsg      msg;            /* completion message */
} AudioInfo;

typedef struct {
    Acmd          *ACMDList[NUM_ACMD_LISTS];
    AudioInfo     *audioInfo[NUM_OUTPUT_BUFFERS];
    OSThread      thread;
    OSMesgQueue   audioFrameMsgQ;
    OSMesg        audioFrameMsgBuf[AUD_MAX_MESGS];
    OSMesgQueue   audioReplyMsgQ;
    OSMesg        audioReplyMsgBuf[AUD_MAX_MESGS];
    ALGlobals     g;
} AMAudioMgr;

#include "sound_table.h"
#include "sound_bank.h"

/**** audio globals ****/
static u8 audioHeap[AUDIO_HEAP_SIZE];
static ALHeap hp;
static ALSndpConfig     SPConfig;
static OSScClient  client;
static OSMesgQueue *schedCmdQ;

static AMAudioMgr      am;
static u64      audioStack[AUDIO_STACKSIZE/sizeof(u64)];

static u32             audFrameCt = 0;
static u32             curAcmdList = 0;
static u32             minFrameSize;
static u32             frameSize;
static u32             maxFrameSize;
static u32             maxRSPCmds;

/**** private routines ****/
static void amMain(void *arg);
static s32  amDMA(s32 addr, s32 len, void *state);
static ALDMAproc amDmaNew(int **state);
static u32  amHandleFrameMsg(AudioInfo *, AudioInfo *);
static void amHandleDoneMsg(AudioInfo *);

static void    amCreateAudioMgr(ALSynConfig *c, OSPri priority, AmConfig *amc, OSSched *sc);

static ALSndPlayer *soundPlayer;
static ALSndId *sndId;
static int numSounds;

void clearAudio()
{
    int i;
    int j;
    short *data;

    for (i=0; i<NUM_OUTPUT_BUFFERS; i++) {
        data = (am.audioInfo[i])->data;
        for (j=0; j<(maxFrameSize<<1); j++) {
            data[j] = 0;
        }
    }
}

void playSound(int id, int waitFinish)
{
    /*
     * Always have to wait till its finished before starting again
     */
    while (isPlaying(sndId[id])) ;

    alSndpSetSound(soundPlayer, sndId[id]);
    alSndpSetPitch(soundPlayer, 1);
    alSndpSetVol(soundPlayer, 20000);
    alSndpPlay(soundPlayer);

    if (waitFinish == AUDIO_WAIT) {
        while (!isPlaying(id));
        while (isPlaying(id)) ;
    }
}

int isPlaying(int id)
{
    alSndpSetSound(soundPlayer, sndId[id]);
    if (alSndpGetState(soundPlayer) == AL_STOPPED) {
        return 0;
    }
    return 1;
}

void initAudio(OSSched *sc) 
{
    int i;
    ALSndPlayer *sndp;
    
    ALSound *snd;
    int numAllocated;

    ALBank *sfxBank;
    u8 *sfxBankPtr;

    ALInstrument *inst;

    u32           bankLen;
    ALSynConfig   c;

    AmConfig      amc;

    ALSndId *idPtr;

    alHeapInit(&hp, audioHeap, sizeof(audioHeap));    
    soundPlayer = alHeapAlloc(&hp, 1, sizeof(ALSndPlayer));
    sndp = soundPlayer;

    bankLen = sizeof(soundBank);
    sfxBankPtr = soundBank;

    /*
     * Create the Audio Manager
     */
    c.maxVVoices = MAX_VOICES;
    c.maxPVoices = MAX_VOICES;
    c.maxUpdates = MAX_UPDATES;
    c.dmaproc    = 0;                  /* audio mgr will fill this in */
    c.fxType	 = AL_FX_SMALLROOM;
    c.outputRate = 0;                  /* audio mgr will fill this in */
    c.heap       = &hp;
    
    amc.outputRate = OUTPUT_RATE;
    amc.framesPerField = NUM_FIELDS;
    amc.maxACMDSize = MAX_RSP_CMDS;

    amCreateAudioMgr(&c, AUDIO_PRIORITY, &amc, sc);

    /*
     * Initialize the soundplayer
     */
    SPConfig.maxSounds = MAX_VOICES;
    SPConfig.maxEvents = MAX_EVENTS;
    SPConfig.heap  = &hp;

    alSndpNew(sndp, &SPConfig);

    /*
     * Allocate all the sounds
     */
    alBnkfNew((ALBankFile *)sfxBankPtr, soundTable);
    sfxBank = ((ALBankFile *)sfxBankPtr)->bankArray[0];

    /*
     * Better make sure number of sounds is at least equal
     * to the number of effects
     */
    inst = sfxBank->instArray[0];
    idPtr = alHeapAlloc(&hp, 1, inst->soundCount*sizeof(ALSndId));
    for (i=0, numAllocated = 0; i<inst->soundCount; i++) {
        snd = inst->soundArray[i];
        if ((idPtr[i] = alSndpAllocate(sndp, snd)) != -1)
            numAllocated++;
    }
    sndId = idPtr;
    numSounds = numAllocated;

}


/******************************************************************************
 * Audio Manager API
 *****************************************************************************/
static void amCreateAudioMgr(ALSynConfig *c, OSPri pri, AmConfig *amc, OSSched *sc)
{
    u32     i;
    f32     fsize;

    c->dmaproc    = amDmaNew;    
    c->outputRate = osAiSetFrequency(amc->outputRate);

    /*
     * Calculate the frame sample parameters from the
     * video field rate and the output rate
     */
    fsize = (f32) amc->framesPerField * c->outputRate / (f32) 60;
    frameSize = (s32) fsize;
    if (frameSize < fsize)
        frameSize++;
    if (frameSize & 0xf)
        frameSize = (frameSize & ~0xf) + 0x10;
    minFrameSize = frameSize - 16;
    maxFrameSize = frameSize + EXTRA_SAMPLES + 16;

    alInit(&am.g, c);
    
    for(i=0;i<NUM_ACMD_LISTS;i++)
        am.ACMDList[i] = (Acmd*)alHeapAlloc(c->heap, 1, 
                            amc->maxACMDSize * sizeof(Acmd));

    maxRSPCmds = amc->maxACMDSize;

    /**** initialize the done messages ****/
    for (i = 0; i < NUM_OUTPUT_BUFFERS; i++) 
    {
        am.audioInfo[i] = (AudioInfo *)alHeapAlloc(c->heap, 1,
                                                     sizeof(AudioInfo));
        am.audioInfo[i]->msg.done.type = OS_SC_DONE_MSG;
        am.audioInfo[i]->msg.done.info = am.audioInfo[i];
        am.audioInfo[i]->data = alHeapAlloc(c->heap, 1, 4*maxFrameSize);
    }    
    
    osCreateMesgQueue(&am.audioReplyMsgQ, am.audioReplyMsgBuf, AUD_MAX_MESGS);
    osCreateMesgQueue(&am.audioFrameMsgQ, am.audioFrameMsgBuf, AUD_MAX_MESGS);

    osScAddClient(sc, &client, &am.audioFrameMsgQ);
    schedCmdQ = osScGetCmdQ(sc);

    osCreateThread(&am.thread, 3, amMain, 0,
                   (void *)(audioStack+AUDIO_STACKSIZE/sizeof(u64)), pri);
    osStartThread(&am.thread);

}

/******************************************************************************
 *
 * Audio Manager implementation. This thread wakes up at every retrace,
 * and builds an audio task, which it returns to the scheduler, who then
 * is responsible for its finally execution on the RSP. Once the task has
 * finished execution, the scheduler sends back a message saying the task
 * is complete. The audio is triple buffered because the switching to a new
 * audio buffer does not occur exactly at the gfx swapbuffer time.  With
 * 3 buffers you ensure that the program does not destroy data before it is
 * played.
 *
 *****************************************************************************/
static void amMain(void *arg) 
{
    u32         validTask;
    u32         done = 0;
    AudioMsg    *msg;
    AudioInfo   *lastInfo = 0;

    while (!done) 
    {
        (void) osRecvMesg(&am.audioFrameMsgQ, (OSMesg *)&msg, OS_MESG_BLOCK);

        switch (msg->gen.type) 
        {
            case (OS_SC_RETRACE_MSG):
                validTask = amHandleFrameMsg(am.audioInfo[audFrameCt % 3],
                                               lastInfo);
                if(validTask)
                {
                    /* wait for done message */
                    osRecvMesg(&am.audioReplyMsgQ, (OSMesg *)&msg, 
                               OS_MESG_BLOCK);
                    amHandleDoneMsg(msg->done.info);
                    lastInfo = msg->done.info;
                }
                break;

            case (OS_SC_PRE_NMI_MSG):
                /* what should we really do here? quit? ramp down volume? */
                break;

            case (QUIT_MSG):
                done = 1;
                break;

            default:
                break;
        }        
    }
    
    alClose(&am.g);
}

/******************************************************************************
 *
 * amHandleFrameMsg. Handles the video frame messages and schedules calculation
 * of a new set of samples.
 *
 *****************************************************************************/
static u32 amHandleFrameMsg(AudioInfo *info, AudioInfo *lastInfo)
{
    s16 *audioPtr;
    Acmd *cmdp;
    s32 cmdLen;
    int samplesLeft = 0;
    OSScTask *t;

    audFrameCt++;
    audioPtr = (s16 *) osVirtualToPhysical(info->data);
    
    if (lastInfo)
        osAiSetNextBuffer(lastInfo->data, lastInfo->frameSamples<<2);

    
    /* calculate how many samples needed for this frame to keep the DAC full */
    /* this will vary slightly frame to frame, must recalculate every frame */
    samplesLeft = osAiGetLength() >> 2; /* divide by four, to convert bytes */
                                        /* to stereo 16 bit samples */
    info->frameSamples = 16 + ((frameSize - samplesLeft + EXTRA_SAMPLES) & ~0xf);
    if(info->frameSamples < minFrameSize)
        info->frameSamples = minFrameSize;

    cmdp = alAudioFrame(am.ACMDList[curAcmdList], &cmdLen, audioPtr,
                        info->frameSamples);

#ifndef _FINALROM
    if (cmdLen > maxRSPCmds) {
        PRINTF("Command list to long!\n");
    } 
#endif
    
    if(cmdLen == 0)  /* no task produced, return zero to show no valid task */
        return 0;

    t = &info->task;
    
    t->next      = 0;                    /* paranoia */
    t->msgQ      = &am.audioReplyMsgQ; /* reply to when finished */
    t->msg       = (OSMesg)&info->msg;   /* reply with this message */
    t->flags     = OS_SC_NEEDS_RSP;
    
    t->list.t.data_ptr    = (u64 *) am.ACMDList[curAcmdList];
    t->list.t.data_size   = (cmdp - am.ACMDList[curAcmdList]) * sizeof(Acmd);
    t->list.t.type  = M_AUDTASK;
    t->list.t.ucode_boot = (u64 *)rspbootTextStart;
    t->list.t.ucode_boot_size =
        ((int) rspbootTextEnd - (int) rspbootTextStart);
    t->list.t.flags  = OS_TASK_DP_WAIT;
    t->list.t.ucode = (u64 *) aspMainTextStart;
    t->list.t.ucode_data = (u64 *) aspMainDataStart;
    t->list.t.ucode_data_size = SP_UCODE_DATA_SIZE;
    t->list.t.dram_stack = (u64 *) NULL;
    t->list.t.dram_stack_size = 0;
    t->list.t.output_buff = (u64 *) NULL;
    t->list.t.output_buff_size = 0;
    t->list.t.yield_data_ptr = NULL;
    t->list.t.yield_data_size = 0;

    osSendMesg(schedCmdQ, (OSMesg) t, OS_MESG_BLOCK);
    
    curAcmdList ^= 1; /* swap which acmd list you use each frame */    
    
    return 1;
}

/******************************************************************************
 *
 * amHandleDoneMsg. Really just debugging info in this frame. Checks
 * to make sure we completed before we were out of samples.
 *
 *****************************************************************************/
static void amHandleDoneMsg(AudioInfo *info) 
{
    s32    samplesLeft;
    static int firstTime = 1;

    samplesLeft = osAiGetLength()>>2;
    if (samplesLeft == 0 && !firstTime) 
    {
#ifndef _FINALROM
      PRINTF("audio: ai out of samples\n");    
#endif
        firstTime = 0;
    }
}

/******************************************************************************
 *
 * audioDMA. This routine usually handles DMA'ing of samples. In our case
 * they are already in memory.
 *
 *****************************************************************************/
s32 amDMA(s32 addr, s32 len, void *state)
{
    return (int) osVirtualToPhysical((u32 *) addr);
}

/******************************************************************************
 *
 * amDmaNew. Return the address of the DMA routine. For this case we don't do
 * DMA, but we need to follow the model.
 *
 *****************************************************************************/
ALDMAproc amDmaNew(int **state)
{
    return amDMA;
}