fix remapping behavior. Remapping is only necessary if we are rendering on the workbe...

[AROS-Contrib.git] / Games / Doom / i_sound.c

// Emacs style mode select   -*- C++ -*- 
//-----------------------------------------------------------------------------
//
// $Id$
//
// Copyright (C) 1993-1996 by id Software, Inc.
//
// This source is available for distribution and/or modification
// only under the terms of the DOOM Source Code License as
// published by id Software. All rights reserved.
//
// The source is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// FITNESS FOR A PARTICULAR PURPOSE. See the DOOM Source Code License
// for more details.
//
// $Log$
// Revision 1.1  2000/02/29 18:21:04  stegerg
// Doom port based on ADoomPPC. Read README.AROS!
//
//
// DESCRIPTION:
//      System interface for sound.
//
//-----------------------------------------------------------------------------

static const char
rcsid[] = "$Id$";

#include <stdio.h>
#include <stdlib.h>
#include <stdarg.h>

#include <math.h>

#include <sys/time.h>
#include <sys/types.h>

#ifndef LINUX
#include <sys/filio.h>
#endif

#include <fcntl.h>
#include <unistd.h>
#include <sys/ioctl.h>

// Linux voxware output.
#include <linux/soundcard.h>

// Timer stuff. Experimental.
#include <time.h>
#include <signal.h>

#include "z_zone.h"

#include "i_system.h"
#include "i_sound.h"
#include "m_argv.h"
#include "m_misc.h"
#include "w_wad.h"

#include "doomdef.h"

// UNIX hack, to be removed.
#ifdef SNDSERV
// Separate sound server process.
FILE*   sndserver=0;
char*   sndserver_filename = "./sndserver ";
#elif SNDINTR

// Update all 30 millisecs, approx. 30fps synchronized.
// Linux resolution is allegedly 10 millisecs,
//  scale is microseconds.
#define SOUND_INTERVAL     500

// Get the interrupt. Set duration in millisecs.
int I_SoundSetTimer( int duration_of_tick );
void I_SoundDelTimer( void );
#else
// None?
#endif


// A quick hack to establish a protocol between
// synchronous mix buffer updates and asynchronous
// audio writes. Probably redundant with gametic.
static int flag = 0;

// The number of internal mixing channels,
//  the samples calculated for each mixing step,
//  the size of the 16bit, 2 hardware channel (stereo)
//  mixing buffer, and the samplerate of the raw data.


// Needed for calling the actual sound output.
#define SAMPLECOUNT             512
#define NUM_CHANNELS            8
// It is 2 for 16bit, and 2 for two channels.
#define BUFMUL                  4
#define MIXBUFFERSIZE           (SAMPLECOUNT*BUFMUL)

#define SAMPLERATE              11025   // Hz
#define SAMPLESIZE              2       // 16bit

// The actual lengths of all sound effects.
int             lengths[NUMSFX];

// The actual output device.
int     audio_fd;

// The global mixing buffer.
// Basically, samples from all active internal channels
//  are modifed and added, and stored in the buffer
//  that is submitted to the audio device.
signed short    mixbuffer[MIXBUFFERSIZE];


// The channel step amount...
unsigned int    channelstep[NUM_CHANNELS];
// ... and a 0.16 bit remainder of last step.
unsigned int    channelstepremainder[NUM_CHANNELS];


// The channel data pointers, start and end.
unsigned char*  channels[NUM_CHANNELS];
unsigned char*  channelsend[NUM_CHANNELS];


// Time/gametic that the channel started playing,
//  used to determine oldest, which automatically
//  has lowest priority.
// In case number of active sounds exceeds
//  available channels.
int             channelstart[NUM_CHANNELS];

// The sound in channel handles,
//  determined on registration,
//  might be used to unregister/stop/modify,
//  currently unused.
int             channelhandles[NUM_CHANNELS];

// SFX id of the playing sound effect.
// Used to catch duplicates (like chainsaw).
int             channelids[NUM_CHANNELS];                       

// Pitch to stepping lookup, unused.
int             steptable[256];

// Volume lookups.
int             vol_lookup[128*256];

// Hardware left and right channel volume lookup.
int*            channelleftvol_lookup[NUM_CHANNELS];
int*            channelrightvol_lookup[NUM_CHANNELS];




//
// Safe ioctl, convenience.
//
void
myioctl
( int   fd,
  int   command,
  int*  arg )
{   
    int         rc;
    extern int  errno;
    
    rc = ioctl(fd, command, arg);  
    if (rc < 0)
    {
        fprintf(stderr, "ioctl(dsp,%d,arg) failed\n", command);
        fprintf(stderr, "errno=%d\n", errno);
        exit(-1);
    }
}





//
// This function loads the sound data from the WAD lump,
//  for single sound.
//
void*
getsfx
( char*         sfxname,
  int*          len )
{
    unsigned char*      sfx;
    unsigned char*      paddedsfx;
    int                 i;
    int                 size;
    int                 paddedsize;
    char                name[20];
    int                 sfxlump;

    
    // Get the sound data from the WAD, allocate lump
    //  in zone memory.
    sprintf(name, "ds%s", sfxname);

    // Now, there is a severe problem with the
    //  sound handling, in it is not (yet/anymore)
    //  gamemode aware. That means, sounds from
    //  DOOM II will be requested even with DOOM
    //  shareware.
    // The sound list is wired into sounds.c,
    //  which sets the external variable.
    // I do not do runtime patches to that
    //  variable. Instead, we will use a
    //  default sound for replacement.
    if ( W_CheckNumForName(name) == -1 )
      sfxlump = W_GetNumForName("dspistol");
    else
      sfxlump = W_GetNumForName(name);
    
    size = W_LumpLength( sfxlump );

    // Debug.
    // fprintf( stderr, "." );
    //fprintf( stderr, " -loading  %s (lump %d, %d bytes)\n",
    //       sfxname, sfxlump, size );
    //fflush( stderr );
    
    sfx = (unsigned char*)W_CacheLumpNum( sfxlump, PU_STATIC );

    // Pads the sound effect out to the mixing buffer size.
    // The original realloc would interfere with zone memory.
    paddedsize = ((size-8 + (SAMPLECOUNT-1)) / SAMPLECOUNT) * SAMPLECOUNT;

    // Allocate from zone memory.
    paddedsfx = (unsigned char*)Z_Malloc( paddedsize+8, PU_STATIC, 0 );
    // ddt: (unsigned char *) realloc(sfx, paddedsize+8);
    // This should interfere with zone memory handling,
    //  which does not kick in in the soundserver.

    // Now copy and pad.
    memcpy(  paddedsfx, sfx, size );
    for (i=size ; i<paddedsize+8 ; i++)
        paddedsfx[i] = 128;

    // Remove the cached lump.
    Z_Free( sfx );
    
    // Preserve padded length.
    *len = paddedsize;

    // Return allocated padded data.
    return (void *) (paddedsfx + 8);
}





//
// This function adds a sound to the
//  list of currently active sounds,
//  which is maintained as a given number
//  (eight, usually) of internal channels.
// Returns a handle.
//
int
addsfx
( int           sfxid,
  int           volume,
  int           step,
  int           seperation )
{
    static unsigned short       handlenums = 0;
 
    int         i;
    int         rc = -1;
    
    int         oldest = gametic;
    int         oldestnum = 0;
    int         slot;

    int         rightvol;
    int         leftvol;

    // Chainsaw troubles.
    // Play these sound effects only one at a time.
    if ( sfxid == sfx_sawup
         || sfxid == sfx_sawidl
         || sfxid == sfx_sawful
         || sfxid == sfx_sawhit
         || sfxid == sfx_stnmov
         || sfxid == sfx_pistol  )
    {
        // Loop all channels, check.
        for (i=0 ; i<NUM_CHANNELS ; i++)
        {
            // Active, and using the same SFX?
            if ( (channels[i])
                 && (channelids[i] == sfxid) )
            {
                // Reset.
                channels[i] = 0;
                // We are sure that iff,
                //  there will only be one.
                break;
            }
        }
    }

    // Loop all channels to find oldest SFX.
    for (i=0; (i<NUM_CHANNELS) && (channels[i]); i++)
    {
        if (channelstart[i] < oldest)
        {
            oldestnum = i;
            oldest = channelstart[i];
        }
    }

    // Tales from the cryptic.
    // If we found a channel, fine.
    // If not, we simply overwrite the first one, 0.
    // Probably only happens at startup.
    if (i == NUM_CHANNELS)
        slot = oldestnum;
    else
        slot = i;

    // Okay, in the less recent channel,
    //  we will handle the new SFX.
    // Set pointer to raw data.
    channels[slot] = (unsigned char *) S_sfx[sfxid].data;
    // Set pointer to end of raw data.
    channelsend[slot] = channels[slot] + lengths[sfxid];

    // Reset current handle number, limited to 0..100.
    if (!handlenums)
        handlenums = 100;

    // Assign current handle number.
    // Preserved so sounds could be stopped (unused).
    channelhandles[slot] = rc = handlenums++;

    // Set stepping???
    // Kinda getting the impression this is never used.
    channelstep[slot] = step;
    // ???
    channelstepremainder[slot] = 0;
    // Should be gametic, I presume.
    channelstart[slot] = gametic;

    // Separation, that is, orientation/stereo.
    //  range is: 1 - 256
    seperation += 1;

    // Per left/right channel.
    //  x^2 seperation,
    //  adjust volume properly.
    leftvol =
        volume - ((volume*seperation*seperation) >> 16); ///(256*256);
    seperation = seperation - 257;
    rightvol =
        volume - ((volume*seperation*seperation) >> 16);        

    // Sanity check, clamp volume.
    if (rightvol < 0 || rightvol > 127)
        I_Error("rightvol out of bounds");
    
    if (leftvol < 0 || leftvol > 127)
        I_Error("leftvol out of bounds");
    
    // Get the proper lookup table piece
    //  for this volume level???
    channelleftvol_lookup[slot] = &vol_lookup[leftvol*256];
    channelrightvol_lookup[slot] = &vol_lookup[rightvol*256];

    // Preserve sound SFX id,
    //  e.g. for avoiding duplicates of chainsaw.
    channelids[slot] = sfxid;

    // You tell me.
    return rc;
}





//
// SFX API
// Note: this was called by S_Init.
// However, whatever they did in the
// old DPMS based DOS version, this
// were simply dummies in the Linux
// version.
// See soundserver initdata().
//
void I_SetChannels()
{
  // Init internal lookups (raw data, mixing buffer, channels).
  // This function sets up internal lookups used during
  //  the mixing process. 
  int           i;
  int           j;
    
  int*  steptablemid = steptable + 128;
  
  // Okay, reset internal mixing channels to zero.
  /*for (i=0; i<NUM_CHANNELS; i++)
  {
    channels[i] = 0;
  }*/

  // This table provides step widths for pitch parameters.
  // I fail to see that this is currently used.
  for (i=-128 ; i<128 ; i++)
    steptablemid[i] = (int)(pow(2.0, (i/64.0))*65536.0);
  
  
  // Generates volume lookup tables
  //  which also turn the unsigned samples
  //  into signed samples.
  for (i=0 ; i<128 ; i++)
    for (j=0 ; j<256 ; j++)
      vol_lookup[i*256+j] = (i*(j-128)*256)/127;
}       

 
void I_SetSfxVolume(int volume)
{
  // Identical to DOS.
  // Basically, this should propagate
  //  the menu/config file setting
  //  to the state variable used in
  //  the mixing.
  snd_SfxVolume = volume;
}

// MUSIC API - dummy. Some code from DOS version.
void I_SetMusicVolume(int volume)
{
  // Internal state variable.
  snd_MusicVolume = volume;
  // Now set volume on output device.
  // Whatever( snd_MusciVolume );
}


//
// Retrieve the raw data lump index
//  for a given SFX name.
//
int I_GetSfxLumpNum(sfxinfo_t* sfx)
{
    char namebuf[9];
    sprintf(namebuf, "ds%s", sfx->name);
    return W_GetNumForName(namebuf);
}

//
// Starting a sound means adding it
//  to the current list of active sounds
//  in the internal channels.
// As the SFX info struct contains
//  e.g. a pointer to the raw data,
//  it is ignored.
// As our sound handling does not handle
//  priority, it is ignored.
// Pitching (that is, increased speed of playback)
//  is set, but currently not used by mixing.
//
int
I_StartSound
( int           id,
  int           vol,
  int           sep,
  int           pitch,
  int           priority )
{

  // UNUSED
  priority = 0;
  
#ifdef SNDSERV 
    if (sndserver)
    {
        fprintf(sndserver, "p%2.2x%2.2x%2.2x%2.2x\n", id, pitch, vol, sep);
        fflush(sndserver);
    }
    // warning: control reaches end of non-void function.
    return id;
#else
    // Debug.
    //fprintf( stderr, "starting sound %d", id );
    
    // Returns a handle (not used).
    id = addsfx( id, vol, steptable[pitch], sep );

    // fprintf( stderr, "/handle is %d\n", id );
    
    return id;
#endif
}



void I_StopSound (int handle)
{
  // You need the handle returned by StartSound.
  // Would be looping all channels,
  //  tracking down the handle,
  //  an setting the channel to zero.
  
  // UNUSED.
  handle = 0;
}


int I_SoundIsPlaying(int handle)
{
    // Ouch.
    return gametic < handle;
}




//
// This function loops all active (internal) sound
//  channels, retrieves a given number of samples
//  from the raw sound data, modifies it according
//  to the current (internal) channel parameters,
//  mixes the per channel samples into the global
//  mixbuffer, clamping it to the allowed range,
//  and sets up everything for transferring the
//  contents of the mixbuffer to the (two)
//  hardware channels (left and right, that is).
//
// This function currently supports only 16bit.
//
void I_UpdateSound( void )
{
#ifdef SNDINTR
  // Debug. Count buffer misses with interrupt.
  static int misses = 0;
#endif

  
  // Mix current sound data.
  // Data, from raw sound, for right and left.
  register unsigned int sample;
  register int          dl;
  register int          dr;
  
  // Pointers in global mixbuffer, left, right, end.
  signed short*         leftout;
  signed short*         rightout;
  signed short*         leftend;
  // Step in mixbuffer, left and right, thus two.
  int                           step;

  // Mixing channel index.
  int                           chan;
    
    // Left and right channel
    //  are in global mixbuffer, alternating.
    leftout = mixbuffer;
    rightout = mixbuffer+1;
    step = 2;

    // Determine end, for left channel only
    //  (right channel is implicit).
    leftend = mixbuffer + SAMPLECOUNT*step;

    // Mix sounds into the mixing buffer.
    // Loop over step*SAMPLECOUNT,
    //  that is 512 values for two channels.
    while (leftout != leftend)
    {
        // Reset left/right value. 
        dl = 0;
        dr = 0;

        // Love thy L2 chache - made this a loop.
        // Now more channels could be set at compile time
        //  as well. Thus loop those  channels.
        for ( chan = 0; chan < NUM_CHANNELS; chan++ )
        {
            // Check channel, if active.
            if (channels[ chan ])
            {
                // Get the raw data from the channel. 
                sample = *channels[ chan ];
                // Add left and right part
                //  for this channel (sound)
                //  to the current data.
                // Adjust volume accordingly.
                dl += channelleftvol_lookup[ chan ][sample];
                dr += channelrightvol_lookup[ chan ][sample];
                // Increment index ???
                channelstepremainder[ chan ] += channelstep[ chan ];
                // MSB is next sample???
                channels[ chan ] += channelstepremainder[ chan ] >> 16;
                // Limit to LSB???
                channelstepremainder[ chan ] &= 65536-1;

                // Check whether we are done.
                if (channels[ chan ] >= channelsend[ chan ])
                    channels[ chan ] = 0;
            }
        }
        
        // Clamp to range. Left hardware channel.
        // Has been char instead of short.
        // if (dl > 127) *leftout = 127;
        // else if (dl < -128) *leftout = -128;
        // else *leftout = dl;

        if (dl > 0x7fff)
            *leftout = 0x7fff;
        else if (dl < -0x8000)
            *leftout = -0x8000;
        else
            *leftout = dl;

        // Same for right hardware channel.
        if (dr > 0x7fff)
            *rightout = 0x7fff;
        else if (dr < -0x8000)
            *rightout = -0x8000;
        else
            *rightout = dr;

        // Increment current pointers in mixbuffer.
        leftout += step;
        rightout += step;
    }

#ifdef SNDINTR
    // Debug check.
    if ( flag )
    {
      misses += flag;
      flag = 0;
    }
    
    if ( misses > 10 )
    {
      fprintf( stderr, "I_SoundUpdate: missed 10 buffer writes\n");
      misses = 0;
    }
    
    // Increment flag for update.
    flag++;
#endif
}


// 
// This would be used to write out the mixbuffer
//  during each game loop update.
// Updates sound buffer and audio device at runtime. 
// It is called during Timer interrupt with SNDINTR.
// Mixing now done synchronous, and
//  only output be done asynchronous?
//
void
I_SubmitSound(void)
{
  // Write it to DSP device.
  write(audio_fd, mixbuffer, SAMPLECOUNT*BUFMUL);
}



void
I_UpdateSoundParams
( int   handle,
  int   vol,
  int   sep,
  int   pitch)
{
  // I fail too see that this is used.
  // Would be using the handle to identify
  //  on which channel the sound might be active,
  //  and resetting the channel parameters.

  // UNUSED.
  handle = vol = sep = pitch = 0;
}




void I_ShutdownSound(void)
{    
#ifdef SNDSERV
  if (sndserver)
  {
    // Send a "quit" command.
    fprintf(sndserver, "q\n");
    fflush(sndserver);
  }
#else
  // Wait till all pending sounds are finished.
  int done = 0;
  int i;
  

  // FIXME (below).
  fprintf( stderr, "I_ShutdownSound: NOT finishing pending sounds\n");
  fflush( stderr );
  
  while ( !done )
  {
    for( i=0 ; i<8 && !channels[i] ; i++);
    
    // FIXME. No proper channel output.
    //if (i==8)
    done=1;
  }
#ifdef SNDINTR
  I_SoundDelTimer();
#endif
  
  // Cleaning up -releasing the DSP device.
  close ( audio_fd );
#endif

  // Done.
  return;
}






void
I_InitSound()
{ 
#ifdef SNDSERV
  char buffer[256];
  
  if (getenv("DOOMWADDIR"))
    sprintf(buffer, "%s/%s",
            getenv("DOOMWADDIR"),
            sndserver_filename);
  else
    sprintf(buffer, "%s", sndserver_filename);
  
  // start sound process
  if ( !access(buffer, X_OK) )
  {
    strcat(buffer, " -quiet");
    sndserver = popen(buffer, "w");
  }
  else
    fprintf(stderr, "Could not start sound server [%s]\n", buffer);
#else
    
  int i;
  
#ifdef SNDINTR
  fprintf( stderr, "I_SoundSetTimer: %d microsecs\n", SOUND_INTERVAL );
  I_SoundSetTimer( SOUND_INTERVAL );
#endif
    
  // Secure and configure sound device first.
  fprintf( stderr, "I_InitSound: ");
  
  audio_fd = open("/dev/dsp", O_WRONLY);
  if (audio_fd<0)
    fprintf(stderr, "Could not open /dev/dsp\n");
  
                     
  i = 11 | (2<<16);                                           
  myioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &i);
  myioctl(audio_fd, SNDCTL_DSP_RESET, 0);
  
  i=SAMPLERATE;
  
  myioctl(audio_fd, SNDCTL_DSP_SPEED, &i);
  
  i=1;
  myioctl(audio_fd, SNDCTL_DSP_STEREO, &i);
  
  myioctl(audio_fd, SNDCTL_DSP_GETFMTS, &i);
  
  if (i&=AFMT_S16_LE)    
    myioctl(audio_fd, SNDCTL_DSP_SETFMT, &i);
  else
    fprintf(stderr, "Could not play signed 16 data\n");

  fprintf(stderr, " configured audio device\n" );

    
  // Initialize external data (all sounds) at start, keep static.
  fprintf( stderr, "I_InitSound: ");
  
  for (i=1 ; i<NUMSFX ; i++)
  { 
    // Alias? Example is the chaingun sound linked to pistol.
    if (!S_sfx[i].link)
    {
      // Load data from WAD file.
      S_sfx[i].data = getsfx( S_sfx[i].name, &lengths[i] );
    }   
    else
    {
      // Previously loaded already?
      S_sfx[i].data = S_sfx[i].link->data;
      lengths[i] = lengths[(S_sfx[i].link - S_sfx)/sizeof(sfxinfo_t)];
    }
  }

  fprintf( stderr, " pre-cached all sound data\n");
  
  // Now initialize mixbuffer with zero.
  for ( i = 0; i< MIXBUFFERSIZE; i++ )
    mixbuffer[i] = 0;
  
  // Finished initialization.
  fprintf(stderr, "I_InitSound: sound module ready\n");
    
#endif
}




//
// MUSIC API.
// Still no music done.
// Remains. Dummies.
//
void I_InitMusic(void)          { }
void I_ShutdownMusic(void)      { }

static int      looping=0;
static int      musicdies=-1;

void I_PlaySong(int handle, int looping)
{
  // UNUSED.
  handle = looping = 0;
  musicdies = gametic + TICRATE*30;
}

void I_PauseSong (int handle)
{
  // UNUSED.
  handle = 0;
}

void I_ResumeSong (int handle)
{
  // UNUSED.
  handle = 0;
}

void I_StopSong(int handle)
{
  // UNUSED.
  handle = 0;
  
  looping = 0;
  musicdies = 0;
}

void I_UnRegisterSong(int handle)
{
  // UNUSED.
  handle = 0;
}

int I_RegisterSong(void* data)
{
  // UNUSED.
  data = NULL;
  
  return 1;
}

// Is the song playing?
int I_QrySongPlaying(int handle)
{
  // UNUSED.
  handle = 0;
  return looping || musicdies > gametic;
}



//
// Experimental stuff.
// A Linux timer interrupt, for asynchronous
//  sound output.
// I ripped this out of the Timer class in
//  our Difference Engine, including a few
//  SUN remains...
//  
#ifdef sun
    typedef     sigset_t        tSigSet;
#else    
    typedef     int             tSigSet;
#endif


// We might use SIGVTALRM and ITIMER_VIRTUAL, if the process
//  time independend timer happens to get lost due to heavy load.
// SIGALRM and ITIMER_REAL doesn't really work well.
// There are issues with profiling as well.
static int /*__itimer_which*/  itimer = ITIMER_REAL;

static int sig = SIGALRM;

// Interrupt handler.
void I_HandleSoundTimer( int ignore )
{
  // Debug.
  //fprintf( stderr, "%c", '+' ); fflush( stderr );
  
  // Feed sound device if necesary.
  if ( flag )
  {
    // See I_SubmitSound().
    // Write it to DSP device.
    write(audio_fd, mixbuffer, SAMPLECOUNT*BUFMUL);

    // Reset flag counter.
    flag = 0;
  }
  else
    return;
  
  // UNUSED, but required.
  ignore = 0;
  return;
}

// Get the interrupt. Set duration in millisecs.
int I_SoundSetTimer( int duration_of_tick )
{
  // Needed for gametick clockwork.
  struct itimerval    value;
  struct itimerval    ovalue;
  struct sigaction    act;
  struct sigaction    oact;

  int res;
  
  // This sets to SA_ONESHOT and SA_NOMASK, thus we can not use it.
  //     signal( _sig, handle_SIG_TICK );
  
  // Now we have to change this attribute for repeated calls.
  act.sa_handler = I_HandleSoundTimer;
#ifndef sun    
  //ac  t.sa_mask = _sig;
#endif
  act.sa_flags = SA_RESTART;
  
  sigaction( sig, &act, &oact );

  value.it_interval.tv_sec    = 0;
  value.it_interval.tv_usec   = duration_of_tick;
  value.it_value.tv_sec       = 0;
  value.it_value.tv_usec      = duration_of_tick;

  // Error is -1.
  res = setitimer( itimer, &value, &ovalue );

  // Debug.
  if ( res == -1 )
    fprintf( stderr, "I_SoundSetTimer: interrupt n.a.\n");
  
  return res;
}


// Remove the interrupt. Set duration to zero.
void I_SoundDelTimer()
{
  // Debug.
  if ( I_SoundSetTimer( 0 ) == -1)
    fprintf( stderr, "I_SoundDelTimer: failed to remove interrupt. Doh!\n");
}