/******************************************************************************/
/*									      */
/*			    SAMPLE SUPPORT FOR RAINE			      */
/*									      */
/*			  modified by Hiromitsu Shioya			      */
/* Author : Emmanuel Anne						      */
/******************************************************************************/
//
// This is the main sound driver for raine right now.
// The seal driver is ported from this driver.
//
/* This is now the reference sound driver for raine.
   SDL allows an incredible flexibility.

   There is clearly an imprecision in the soundcards when playing a 22Khz sound for
   almost nothing, I tested it in streams.c a difference of 1 byte makes the sound
   late or in advance for base_len. And the precise base_len can't be guessed apparently.

   In allegro this problem is without solution, see the 2 versions of sasound.c in its
   directory (in windows, allegro works. Apparently its sampling functions lack precision
   in linux, but I didn't try to find out why).

   In SDL, a handler is called precisely when needed to update this sound buffer.
   Then when we update the streams buffers we check the difference between the point
   writen to by the streams and the point read by sdl. If the difference is changing
   too much, we just drop or add a sound frame (it can't be heared at all).

   This can't be adapted to allegro, the differences vary too much in allegro to be
   usable, and there is no callback.

   Notice : there are still some pieces of very old code from before version 0.28
   in this file. I should clean it up one day... maybe.
*/


#ifdef RAINE_DEBUG
/* Normally when you mix different voices, you are supposed to check if the result
   gets too loud. But we can avoid this by setting the volume of the different sound chips
   to reasonable values. That's exactly what mame is already doing. Nice idea, it's faster
   and since the sound has not to be bounded it probably sounds better too.
   So this setting is here only to know if our volume is still too loud when in debug
   mode (prints out on stderr) */
#define TEST_OVERFLOW
#endif

#include <time.h>
#include "raine.h"
#include "sasound.h"
#include "games.h"
#include "debug.h"
#include "timer.h"
#include "profile.h" // fps

// This define is required for the new glibc in linux to have access the usefull
// definitions in math.h (why doing something simple when it can be complicated ?)
#define __USE_ISOC99 1
#include <math.h> // double operations for resampling
#include "SDL.h"
#include "SDL_audio.h"
#ifndef NEO
#include "dxsmp.h"
#else
#include <SDL_mixer.h>
#include "neocd/neocd.h"
#include "neocd/cdda.h"
#endif

int GameSound;
UINT8 *PCMROM;
static UINT8 counter[MAX_STREAM_CHANNELS];

static char driver_name[40];

int RaineSoundCardTotal,
  RaineSoundCard;

/* Avoid to uncomment USE_8BITS unless you finish 8bit support... which is
    pretty useless nowdays ! */

//#define USE_8BITS 1


SoundRec      *SndMachine = NULL, snd_entry;

/* audio related stuff */

static INT16 *lpWave[NUMVOICES];

// The "normal" version of PlayStream does not use samples nor voicexs.
// It just uses "streams" ! What a mess !

static int	   playing[NUMVOICES];
static INT16	   *vout[NUMVOICES],*vend[NUMVOICES],*vout_stream[NUMVOICES];

int	    audio_sample_rate;

static int reserved_channel = 0,opened_audio = 0;

static int stream_buffer_max;

static int pause_sound;

void saCheckPlayStream( void );

void saSetVolume( int channel, int data )
{
  // For SDL, we mix the sound in software, see update_recording in streams.c
}

void saSetPan( int channel, int data )
{
}

#include "streams.c"

/*******************************************************************************************/
/*  sa???Sound										   */
/*******************************************************************************************/

/******************************************/
/*    update sound			  */
/******************************************/

void saUpdateSound( int nowclock )
{
   if( ! GameSound ) return;
   if( ! RaineSoundCard ) return;
   if( ! audio_sample_rate ) return;
   if( ! SndMachine ) return;

   if( nowclock ){
     //int i;
     // This part is called for each frame, which *should* be 60
  // times/sec, but it can be less (if the game slows down)
      streams_sh_update();
   }

}

int enh_stereo = 0;

extern int max_mixer_volume;

/******************************************/
/*    setup sound			  */
/******************************************/
BOOL saInitSoundCard( int soundcard, int sample_rate )
{

   int i,id;
   if (opened_audio)
     return TRUE;
   RaineSoundCard = 1;
   /* install a digital sound driver */
     // Normally, soundcard =0 means no sound in raine.
     // I will try not to break this to keep compatibility with the other
     // sources...
   id = sound_card_id(soundcard);

/*    if ( SDL_Init(SDL_INIT_AUDIO) < 0 ) { */
/*      fprintf(stderr, "Couldn't initialize SDL: %s\n",SDL_GetError()); */
/*      exit(1); */
/*    } */
/*    atexit(SDL_Quit); */

   for( i = 0; i < NUMVOICES; i++ ){

      lpWave[i]  = 0;
      playing[i] = 0;
   }

   stream_buffer_max = STREAM_BUFFER_MAXB;

   //reserved_channel = 0;

   pause_sound = 0;		/* pause flag off */
   if(!init_sound_emulators()) {
     return FALSE;  // Everything fine
   }

   return TRUE;
}

/******************************************/
/*    setup sound			  */
/******************************************/

void init_sound_list(void)
{
  const SOUND_INFO *sound_src; // games/games.h
  int ta;

  sound_src = current_game->sound_list;

  if(sound_src){

    saStopSoundEmulators();

    for( ta = 0; ta < SND_CONTROL_MAX; ta++ ){
      SndMachine->init[ta] = SOUND_NONE;
      SndMachine->intf[ta] = NULL;
    }

    SndMachine->first = 0;
    SndMachine->control_max = 0;

    ta = 0;

    while(sound_src[ta].interface){

      SndMachine->init[ta] = sound_src[ta].type;
      SndMachine->intf[ta] = sound_src[ta].interface;

      ta++;

    }

    SndMachine->control_max = ta;

    GameSound = 1;

  }
}

/******************************************/
/*    destroy sound			  */
/******************************************/

void saDestroyChannel( int chan )
{
  if( lpWave[chan] ){
    FreeMem( (UINT8*)lpWave[chan] );
    lpWave[chan] = 0;
    playing[chan] = 0;
  }
}

void saDestroySound( int remove_all_resources )
{
   int i;

#ifdef RAINE_DEBUG
   print_debug("saDestroySound: Removing SEAL\n");
#endif

   //pause_raine_ym3812();

   if(remove_all_resources){
     saStopSoundEmulators();
   }

   /* We *MUST* close the audio here because of implicit frequency conversion.
      If you load pacman first then the audio will be opened at 96 Khz, so it must
      be closed at the end in order to open it again at a more normal frequency later. */

   if (opened_audio) {
#ifdef NEO
     Mix_CloseAudio();
#else
     SDL_CloseAudio();
#endif
   }

   for( i = 0; i < NUMVOICES; i++ ){
      saDestroyChannel(i);

   }
#ifdef USE_COMPENS
   reset_streams();
#endif

#ifdef RAINE_DEBUG
   print_debug("saDestroySound: OK\n");
#endif
   opened_audio = 0;
}

void sa_pause_sound(void)
{
   if (!pause_sound) {
     pause_sound	    = 1;

     //pause_raine_ym3812();
     SDL_PauseAudio(1);
   }
}

void sa_unpause_sound(void)
{
  if (pause_sound) {
    pause_sound	   = 0;
    SDL_PauseAudio(0);
#ifdef NEO
    Mix_VolumeMusic(128*music_volume/100);
#endif
  }
}

/*******************************************************************************************/
/*******************************************************************************************/
/******************************************/
/*    play samples			  */
/******************************************/
#if 0
// This was supposed to resample the voice, but it does not work well at all !!!
INT16 get_average(signed short *din, int distance, double rapport) {
  int start = distance*rapport;
  int len = (distance+1)*rapport - start;
  int nb = 0;
  int i;
  return din[start];
  for (i=start; i<start+len; i++) {
    nb += din[i];
  }
  return nb/len;
}
#endif

// len of the buffer to update in samples
#define LEN_SAMPLES 2048

static void my_callback(void *userdata, Uint8 *stream, int len)
{
  int i,channel,diff;
  short *wstream = (short*) stream;
  // int nb=0;
#ifndef NEO
  memset(stream,0,len);
#endif
  len /= 2;

  for (channel=0; channel<NUMVOICES; channel++) {
    if( playing[channel] ){
      if (!vout_stream[channel])
	vout_stream[channel] = lpWave[channel];
      do {
	diff = vout[channel] - vout_stream[channel];
	if (diff < 0) {
	  diff += (vend[channel] - lpWave[channel]);
	}
	// printf("callback diff %d/%d\n",diff,len/2);
	if (diff > 3*len/2) {
	  vout_stream[channel] += len/2;
	  // printf("too much delay channel %d\n",channel);
	  if (vout_stream[channel] > vend[channel]) {
	    // printf("over buffer channel %d\n",channel);
	    vout_stream[channel] = lpWave[channel]+(vout_stream[channel] - vend[channel]);
	  }
	}
	if (diff < len/2) {
	  // printf("compens %d\n",nb++);
	  streams_sh_update();
	}
      } while (diff < len/2);
    }
  }
  /* Ideally in this case I would average the buffer.
     But normally this happens only when the sound becomes late because of the OS.
     Either heavy swapping or windows "multitasking" defieciency (saw it even in win2k
     when you change the focus of the window, the sound stops updating !!!). So in
     this case we just need to jump directly to the correct point of update */

  for (channel=0; channel<NUMVOICES; channel++) {
    if( playing[channel] ){
      int volume = SampleVol[channel];
      int vol_l = (255-SamplePan[channel])*volume/255;
      int vol_r = (SamplePan[channel])*volume/255;
#ifdef NEO
      vol_l = vol_l*sfx_volume/100;
      vol_r = vol_r*sfx_volume/100;
#endif
      signed short *din=((signed short*)vout_stream[channel]);
      diff = vout[channel] - vout_stream[channel];
      if (diff < 0) {
	diff += (vend[channel] - lpWave[channel]);
      }
      if (diff < len/2) {
	printf("diff %d for channel %d, should not happen\n",diff,channel);
	// exit(1);
      }
      // vol_l>>=1;
      // vol_r>>=1;
#if 0

      if (stream_sample_rate[channel] != audio_sample_rate) {
	/* Normally resampling should use SDL_ConvertAudio.
	   The problem is that when a stream is played at a different frequency, then
	   its buffer length is different too (of course), which makes the synchronization
	   code different (vout_stream must not be compared directly to vout, but adaptated
	   using the ratio of the frequencies.
	   So it would make the code notably more complex for something which is only an
	   exception. Until now, only the namco sound drivers requires a different playing
	   frequency. So the best for now is just to open the audio at the frequency of
	   stream 0, which makes the conversion automatic. */
      } else
#endif
	if (vout_stream[channel] + len/2 > vend[channel]) {
	  /* Overlap of the buffer */
	  int len2 = (vend[channel] - vout_stream[channel])*2;
	  for (i=0; i<len2; i+=2) {
	    *wstream++ += *(din)*vol_l/255;
	    *wstream++ += *(din++)*vol_r/255;
	  }
	  din = (signed short*)lpWave[channel];
	  len -= len2;
	  for (i=0; i<len; i+=2) {
	    *wstream++ += *(din)*vol_l/255;
	    *wstream++ += *(din++)*vol_r/255;
	  }
	  wstream = (INT16*) stream;
	  len += len2;
	} else {
	  /* normal buffer, no resample */
	  for (i=0; i<len; i+=2) {
	    INT16 left = *(din)*vol_l/255;
	    INT16 right = *(din++)*vol_r/255;
#ifdef TEST_OVERFLOW
	    INT32 sample = wstream[i]+left;
	    if (sample > 0x7fff) {
	      printf("overflow left %x\n",sample);
	      sample = 0x7fff;
	    } else if (sample < -0x8000) {
	      printf("underflow left %x\n",sample);
	      sample = -0x8000;
	    }
	    wstream[i] = sample;
	    sample = wstream[i+1] + right;
	    if (sample > 0x7fff) {
	      printf("overflow right %x\n",sample);
	      sample = 0x7fff;
	    } else if (sample < -0x8000) {
	      printf("underflow right %x\n",sample);
	      sample = -0x8000;
	    }
	    wstream[i+1] = sample;
#else
	    wstream[i] += left;
	    wstream[i+1] += right;
#endif
	  }
	}
      vout_stream[channel] = din;

    }
  }


  if (recording) {
    if (mixing_buff_len != len) {
      mixing_buff_len = len;
    }
    mixing_buff = (short*)stream;
    updated_recording++;
  }
}

SDL_AudioSpec gotspec;

extern int smallest_sound_buffer;

void saPlayBufferedStreamedSampleBase( int channel, signed char *data, int len, int freq, int volume, int bits , int pan ){
  /* This version works at low level, creating a sample, and following its
     advancement directly in the voice_position... */
  int i;
  INT16 *dout,*dfin;
  signed short *din;
  // fprintf(stderr,"saPlayBuffer %d freq %d bits %d pan %d len %d\n",channel,freq,bits,pan,len);
  if( audio_sample_rate == 0 || channel >= NUMVOICES )	return;
  if( SndMachine == NULL )  return;
  if( !playing[channel] ){
#ifdef USE_COMPENS
    int fin = stream_buffer_max * freq * 2 / fps;
#else
    int fin = stream_buffer_max * len;
#endif
    if( lpWave[channel] ){
      FreeMem( (UINT8*)lpWave[channel] );
      lpWave[channel] = 0;
    }

    if (!(lpWave[channel] = (INT16*)AllocateMem(fin*2))){
      lpWave[channel] = 0;
      return;
    }

    memset( lpWave[channel], 0, fin );
    dfin=lpWave[channel]+fin;
    vend[channel] = dfin;
    counter[channel] = 0;

    playing[channel] = 1;	/* use front surface */

    dout=lpWave[channel];
    din = (INT16*)data;
    //	      memcpy( dout, din, len );
    for (i=0; i<len; i+=2){
      *(dout++) = *(din++);
    }

    vout[channel] = dout;
    vout_stream[channel] = NULL;

    if (dout ==dfin){
      dout=lpWave[channel];
    }

    if (!opened_audio) {
      SDL_AudioSpec spec;
      int nb=0;
      spec.freq = freq; // audio_sample_rate
      spec.format = AUDIO_S16LSB;
      spec.channels = 2;
      spec.samples  = len/2;
      while (spec.samples) {
	spec.samples >>= 1;
	nb++;
      }
      spec.samples = 1 << nb;
      /* Notice : creative labs semms to make sound drivers which do not respect specs
	 in windows, since they are unable to handle small buffers for the updates.
	 For their drivers, spec.samples should be :
	 spec.samples = 512 * audio_sample_rate / 11025; // 1 << (nb);
	 But unfortunately this makes the correction of the sound delays much more
	 unprecise and you hear cracks if you play long enough (they don't last long, but
	 they come back regularly). So the best is to switch back windows to the allegro
	 sound driver, and use an optimum value for spec.samples here.

	 I NEVER heard about this bug in linux. They respect specs in linux usually.
	 So I just put this in an ifndef, and I should probably add some configuration
	 variable to force precise updates in win32 too ! */

#ifndef RAINE_UNIX
      /* This insanity never happens in linux afaik ! */
      if (spec.samples < LEN_SAMPLES && !smallest_sound_buffer) {
	// printf("forçage len_samples\n");
	spec.samples = LEN_SAMPLES;
      }
#endif
      spec.userdata = NULL;
      spec.callback = my_callback;
#ifdef NEO
      printf("open mix %d,%d,%d,%d\n",spec.freq,spec.format,spec.channels,spec.samples);
      if ( Mix_OpenAudio(spec.freq,spec.format,spec.channels,spec.samples) < 0){
#else
      if ( SDL_OpenAudio(&spec, &gotspec) < 0 ) {
#endif
	fprintf(stderr, "Couldn't open audio: %s\n", SDL_GetError());
	RaineSoundCard = 0;
	return;
      }
      opened_audio = 1;
#ifndef NEO
      read_dx_file();
#else
      Mix_SetPostMix(my_callback,NULL);
      Mix_Resume(-1);
      restore_cdda();
#endif
      strcpy(driver_name,"SDL ");
      SDL_AudioDriverName(&driver_name[4], 32);
      printf("driver name : %s\n",driver_name);
      // set_sound_variables(0);
      SDL_PauseAudio(0);
    }
    /* vout_stream is initalized at its first update.
       It's because the updates do not start at cpu frame 0, so it's better to wait */
  } else{

    /* At first I had the idea to compute the expected delay between the stream update
       point and the voice position and then asking more or less streams in
       streams_sh_update. But it's useless in SDL and produces bad quality !!!
       Since the callback is called reliably by hardware interrupts (normally) we can
       just synchronize the sound there : if the buffer produced by the streams is not
       long enough, then just ask for more in the callback, and if the buffer becomes
       too big, then just jump some frames to resynchronize (see the comments there
       for more details) */

    dout=vout[channel];
    if (!vout_stream[channel] && dout - lpWave[channel]>LEN_SAMPLES){
      // printf("waiting...\n");
      return;
    }
    din = ((signed short*)data);
    dfin = vend[channel];
    if (dout+len/2 > dfin) {
      memcpy(dout,din,(dfin-dout)*2);
      len -= (dfin - dout)*2;
      din += (dfin - dout);
      dout=(short*)lpWave[channel];
    }
    memcpy(dout,din,len);

    dout += len/2;
    if (dout >=dfin){
      dout=(short*)lpWave[channel];

    }

    vout[channel] = dout;

    // more than count frames of advance : more stream !
#if 0
    pos -= count*len/2;
    if (pos > th_pos && pos > 0) {
      more_stream = 1;
    } else
      more_stream = 0;
#endif
  }
}

#ifdef USE_8BITS
/******************************************/
/*    play samples			  */
/******************************************/
void saPlayStreamedSampleBase( int channel, signed char *data, int len, int freq, int volume, int bits , int pan ){
  // This one should leave most of the sync work to allegro
  int pos;
	void *buff; // position in the stream
  unsigned short *dout;
  signed short *din;
  int i;
  if (bits == 8) {
    fprintf(stderr,"error: Can't play 8 bits\n");
    // Just because I don't want to bother with this now.
    return;
  }
  if( audio_sample_rate == 0 || channel >= NUMVOICES )	return;
  if( SndMachine == NULL )  return;
  if( !playing[channel] ){
    if( stream[channel] ){
      stop_audio_stream(stream[channel]);
      free_audio_stream_buffer(stream[channel]);
      stream[channel] = NULL;
    }

    // printf("playing %d at freq %d\n",channel,freq);
    if (!(stream[channel] = play_audio_stream(len,bits,0,freq,volume,pan))){
      return;
    }
    playing[channel] = 1;	/* use front surface */

    // Wait for the buffer to be ready...
    while (!(buff = get_audio_stream_buffer(stream[channel])));
    //print_debug("first stream entry. [%d:%d:%d:%d]\n", channel, len, freq, volume );

  }

  if (!(buff = get_audio_stream_buffer(stream[channel]))) {
    fprintf(stderr,"init stream impossible : buffer NULL\n");
    return;
  }
  //	fprintf(stderr,"len memcpy : %d\n",len);
  dout=buff;
  din = ((signed short*)data);
  for (i=0; i<len; i+=2)
    *(dout++) = *(din++)^0x8000;

  //fprintf(stderr,"set chanel vol = %d\n",volume);
}
#endif

/******************************************************************************/
/*									      */
/*			  SOUND CHANNEL ALLOCATION			      */
/*									      */
/******************************************************************************/

int saGetPlayChannels( int request )
{
   int ret_value = reserved_channel;
   reserved_channel += request;
   return ret_value;
}

void saResetPlayChannels( void )
{
   reserved_channel = 0;
}

/******************************************************************************/
/*									      */
/*			    SOUND CARD INFORMATION			      */
/*									      */
/******************************************************************************/

char *sound_card_name( int num )
{
   int id = sound_card_id(num);

   RaineSoundCardTotal = 2;

   if (id == 0)
     return "Silence";
   if (!driver_name[0]) {
     strcpy(driver_name,"SDL <autodetect>");
   }
   return driver_name;
}

int sound_card_id( int i )
{
  return i; // for now no id in sdl
}

/******************************* END OF FILE **********************************/

