forked from KolibriOS/kolibrios
e15c3449f3
git-svn-id: svn://kolibrios.org@298 a494cfbc-eb01-0410-851d-a64ba20cac60
900 lines
21 KiB
C
900 lines
21 KiB
C
// Emacs style mode select -*- C++ -*-
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//-----------------------------------------------------------------------------
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//
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// $Id:$
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//
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// Copyright (C) 1993-1996 by id Software, Inc.
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//
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// This source is available for distribution and/or modification
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// only under the terms of the DOOM Source Code License as
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// published by id Software. All rights reserved.
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//
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// The source is distributed in the hope that it will be useful,
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// but WITHOUT ANY WARRANTY; without even the implied warranty of
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// FITNESS FOR A PARTICULAR PURPOSE. See the DOOM Source Code License
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// for more details.
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//
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// $Log:$
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//
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// DESCRIPTION:
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// System interface for sound.
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//
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//-----------------------------------------------------------------------------
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static const char
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rcsid[] = "$Id: i_unix.c,v 1.5 1997/02/03 22:45:10 b1 Exp $";
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#include <stdio.h>
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#include <stdlib.h>
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#include <stdarg.h>
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#include <math.h>
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#include <time.h>
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#include <sys/types.h>
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//#ifndef LINUX
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//#include <sys/filio.h>
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//#endif
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#include <fcntl.h>
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#include <io.h>
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//#include <unistd.h>
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//#include <sys/ioctl.h>
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// Linux voxware output.
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//#include <linux/soundcard.h>
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// Timer stuff. Experimental.
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#include <time.h>
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#include <signal.h>
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#include "z_zone.h"
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#include "i_system.h"
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#include "i_sound.h"
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#include "m_argv.h"
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#include "m_misc.h"
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#include "w_wad.h"
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#include "doomdef.h"
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typedef unsigned int DWORD;
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////////////////////////////////////////////////////////////////////////////
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// WinDoom - DirectSound
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////////////////////////////////////////////////////////////////////////////
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#define NUM_SOUND_FX 128
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#define SB_SIZE 20480
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void CreateSoundBuffer(int Channel, int length, unsigned char *data);
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void I_PlaySoundEffect(int sfxid, int Channel, int volume, int pan);
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#define NUM_DSBUFFERS 256
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typedef enum { dsb_perm, dsb_temp } dsb_type;
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typedef struct
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{
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void *origin;
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int dsb_type;
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int sfxid;
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}DSBControl_t;
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DSBControl_t DSBControl[NUM_DSBUFFERS];
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extern int swap_stereo;
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////////////////////////////////////////////////////////////////////////////
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// UNIX hack, to be removed.
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#ifdef SNDSERV
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// Separate sound server process.
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FILE* sndserver=0;
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char* sndserver_filename = "./sndserver ";
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#elif SNDINTR
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// Update all 30 millisecs, approx. 30fps synchronized.
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// Linux resolution is allegedly 10 millisecs,
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// scale is microseconds.
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#define SOUND_INTERVAL 500
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// Get the interrupt. Set duration in millisecs.
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int I_SoundSetTimer( int duration_of_tick );
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void I_SoundDelTimer( void );
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#else
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// None?
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#endif
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// A quick hack to establish a protocol between
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// synchronous mix buffer updates and asynchronous
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// audio writes. Probably redundant with gametic.
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static int flag = 0;
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// The number of internal mixing channels,
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// the samples calculated for each mixing step,
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// the size of the 16bit, 2 hardware channel (stereo)
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// mixing buffer, and the samplerate of the raw data.
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// Needed for calling the actual sound output.
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#define SAMPLECOUNT 512
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#define NUM_CHANNELS 16
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// It is 2 for 16bit, and 2 for two channels.
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#define BUFMUL 4
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#define MIXBUFFERSIZE (SAMPLECOUNT*BUFMUL)
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#define SAMPLERATE 11025 // Hz
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#define SAMPLESIZE 2 // 16bit
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// The actual lengths of all sound effects.
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int lengths[NUMSFX];
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// The actual output device.
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//int audio_fd;
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// The global mixing buffer.
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// Basically, samples from all active internal channels
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// are modifed and added, and stored in the buffer
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// that is submitted to the audio device.
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signed short mixbuffer[MIXBUFFERSIZE];
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// The channel step amount...
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unsigned int channelstep[NUM_CHANNELS];
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// ... and a 0.16 bit remainder of last step.
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unsigned int channelstepremainder[NUM_CHANNELS];
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// The channel data pointers, start and end.
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unsigned char* channels[NUM_CHANNELS];
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unsigned char* channelsend[NUM_CHANNELS];
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// Time/gametic that the channel started playing,
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// used to determine oldest, which automatically
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// has lowest priority.
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// In case number of active sounds exceeds
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// available channels.
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int channelstart[NUM_CHANNELS];
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// The sound in channel handles,
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// determined on registration,
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// might be used to unregister/stop/modify,
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// currently unused.
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int channelhandles[NUM_CHANNELS];
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// SFX id of the playing sound effect.
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// Used to catch duplicates (like chainsaw).
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int channelids[NUM_DSBUFFERS];
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// Pitch to stepping lookup, unused.
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int steptable[256];
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// Volume lookups.
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int vol_lookup[128*256];
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// Hardware left and right channel volume lookup.
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int* channelleftvol_lookup[NUM_CHANNELS];
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int* channelrightvol_lookup[NUM_CHANNELS];
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//
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// Safe ioctl, convenience.
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//
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void
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myioctl
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( int fd,
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int command,
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int* arg )
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{
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// FIXME
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/*
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int rc;
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extern int errno;
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rc = ioctl(fd, command, arg);
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if (rc < 0)
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{
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fprintf(stderr, "ioctl(dsp,%d,arg) failed\n", command);
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fprintf(stderr, "errno=%d\n", errno);
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exit(-1);
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}
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*/
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}
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//
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// This function loads the sound data from the WAD lump,
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// for single sound.
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//
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void *getsfx( char *sfxname, int *len )
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{
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unsigned char* sfx;
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unsigned char* paddedsfx;
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int i;
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int size;
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int paddedsize;
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char name[20];
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int sfxlump;
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sprintf(name, "ds%s", sfxname);
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// Get the sound data from the WAD, allocate lump
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// in zone memory.
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// Now, there is a severe problem with the
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// sound handling, in it is not (yet/anymore)
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// gamemode aware. That means, sounds from
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// DOOM II will be requested even with DOOM
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// shareware.
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// The sound list is wired into sounds.c,
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// which sets the external variable.
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// I do not do runtime patches to that
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// variable. Instead, we will use a
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// default sound for replacement.
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if ( W_CheckNumForName(name) == -1 )
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sfxlump = W_GetNumForName("dspistol");
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else
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sfxlump = W_GetNumForName(name);
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size = W_LumpLength( sfxlump );
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// sprintf(MsgText, "Getting sound effect : %s - %d\n", name, size);
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// WriteDebug(MsgText);
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// Debug.
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// fprintf( stderr, "." );
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//fprintf( stderr, " -loading %s (lump %d, %d bytes)\n",
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// sfxname, sfxlump, size );
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//fflush( stderr );
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sfx = (unsigned char*)W_CacheLumpNum( sfxlump, PU_STATIC );
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// Pads the sound effect out to the mixing buffer size.
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// The original realloc would interfere with zone memory.
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paddedsize = ((size-8 + (SAMPLECOUNT-1)) / SAMPLECOUNT) * SAMPLECOUNT;
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// Allocate from zone memory.
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paddedsfx = (unsigned char*)Z_Malloc( paddedsize+8, PU_STATIC, 0 );
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// ddt: (unsigned char *) realloc(sfx, paddedsize+8);
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// This should interfere with zone memory handling,
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// which does not kick in in the soundserver.
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// Now copy and pad.
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memcpy( paddedsfx, sfx, size );
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for (i = size; i < paddedsize+8; i++)
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paddedsfx[i] = 128;
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// Remove the cached lump.
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Z_Free( sfx );
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// Preserve padded length.
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*len = paddedsize;
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// Return allocated padded data.
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return (void *) (paddedsfx + 8);
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}
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/*
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//
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// This function adds a sound to the
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// list of currently active sounds,
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// which is maintained as a given number
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// (eight, usually) of internal channels.
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// Returns a handle.
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//
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int addsfx( int sfxid, int volume, int step, int seperation )
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{
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static unsigned short handlenums = 0;
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int i;
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int rc = -1;
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int oldest = gametic;
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int oldestnum = 0;
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int slot;
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int rightvol;
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int leftvol;
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int iVolume, iPan;
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// Chainsaw troubles.
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// Play these sound effects only one at a time.
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if ( sfxid == sfx_sawup || sfxid == sfx_sawidl || sfxid == sfx_sawful ||
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sfxid == sfx_sawhit || sfxid == sfx_stnmov || sfxid == sfx_pistol )
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{
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// Loop all channels, check.
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for (i = 0; i < NUM_CHANNELS; i++)
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{
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// Active, and using the same SFX?
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if ( (channels[i]) && (channelids[i] == sfxid) )
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{
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// Reset.
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channels[i] = 0;
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// We are sure that iff, there will only be one.
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break;
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}
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}
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}
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// Loop all channels to find oldest SFX.
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for (i = 0; (i<NUM_CHANNELS) && (channels[i]); i++)
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{
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if (channelstart[i] < oldest)
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{
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oldestnum = i;
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oldest = channelstart[i];
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}
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}
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// Tales from the cryptic.
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// If we found a channel, fine.
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// If not, we simply overwrite the first one, 0.
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// Probably only happens at startup.
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if (i == NUM_CHANNELS)
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slot = oldestnum;
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else
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slot = i;
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// Okay, in the less recent channel,
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// we will handle the new SFX.
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// Set pointer to raw data.
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channels[slot] = (unsigned char *) S_sfx[sfxid].data;
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// Set pointer to end of raw data.
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channelsend[slot] = channels[slot] + lengths[sfxid];
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// Reset current handle number, limited to 0..100.
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if (!handlenums)
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handlenums = 100;
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// Assign current handle number.
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// Preserved so sounds could be stopped (unused).
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channelhandles[slot] = rc = handlenums++;
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// Set stepping???
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// Kinda getting the impression this is never used.
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channelstep[slot] = step;
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// ???
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channelstepremainder[slot] = 0;
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// Should be gametic, I presume.
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channelstart[slot] = gametic;
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iVolume = 0-(128*(15-volume));
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if (iVolume < -10000)
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iVolume == -10000;
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iPan = (seperation-128)*20;
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if (iPan < -10000)
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iPan = -10000;
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if (iPan > 10000)
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iPan = 10000;
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if (swap_stereo == TRUE)
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iPan *= -1;
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// Separation, that is, orientation/stereo.
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// range is: 1 - 256
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seperation += 1;
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// Per left/right channel.
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// x^2 seperation,
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// adjust volume properly.
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leftvol = volume - ((volume*seperation*seperation) >> 16); //(256*256);
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seperation = seperation - 257;
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rightvol = volume - ((volume*seperation*seperation) >> 16);
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// Sanity check, clamp volume.
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if (rightvol < 0 || rightvol > 127)
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I_Error("rightvol out of bounds");
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if (leftvol < 0 || leftvol > 127)
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I_Error("leftvol out of bounds");
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// Get the proper lookup table piece
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// for this volume level???
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channelleftvol_lookup[slot] = &vol_lookup[leftvol*256];
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channelrightvol_lookup[slot] = &vol_lookup[rightvol*256];
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// Preserve sound SFX id,
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// e.g. for avoiding duplicates of chainsaw.
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channelids[slot] = sfxid;
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I_PlaySoundEffect(sfxid, iVolume, iPan);
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// You tell me.
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//return rc;
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return sfxid;
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}
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*/
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//
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// This function adds a sound to the
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// list of currently active sounds,
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// which is maintained as a given number
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// (eight, usually) of internal channels.
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// Returns a handle.
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//
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//
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// SFX API
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// Note: this was called by S_Init.
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// However, whatever they did in the
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// old DPMS based DOS version, this
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// were simply dummies in the Linux
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// version.
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// See soundserver initdata().
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//
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void I_SetChannels()
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{
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// Init internal lookups (raw data, mixing buffer, channels).
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// This function sets up internal lookups used during
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// the mixing process.
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int i;
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int j;
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int* steptablemid = steptable + 128;
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// Okay, reset internal mixing channels to zero.
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for (i=0; i<NUM_CHANNELS; i++)
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{
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channels[i] = 0;
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}
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// This table provides step widths for pitch parameters.
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// I fail to see that this is currently used.
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for (i=-128 ; i<128 ; i++)
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steptablemid[i] = (int)(pow(2.0, (i/64.0))*65536.0);
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// Generates volume lookup tables
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// which also turn the unsigned samples
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// into signed samples.
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for (i=0 ; i<128 ; i++)
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for (j=0 ; j<256 ; j++)
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vol_lookup[i*256+j] = (i*(j-128)*256)/127;
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}
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void I_SetSfxVolume(int volume)
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{
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// Identical to DOS.
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// Basically, this should propagate
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// the menu/config file setting
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// to the state variable used in
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// the mixing.
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snd_SfxVolume = volume;
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}
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// MUSIC API - dummy. Some code from DOS version.
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void I_SetMusicVolume(int volume)
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{
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// Internal state variable.
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snd_MusicVolume = volume;
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// Now set volume on output device.
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// Whatever( snd_MusciVolume );
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}
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//
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// Retrieve the raw data lump index
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// for a given SFX name.
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//
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int I_GetSfxLumpNum(sfxinfo_t* sfx)
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{
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char namebuf[9];
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sprintf(namebuf, "ds%s", sfx->name);
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return W_GetNumForName(namebuf);
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}
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//
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// Starting a sound means adding it
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// to the current list of active sounds
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// in the internal channels.
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// As the SFX info struct contains
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// e.g. a pointer to the raw data,
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// it is ignored.
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// As our sound handling does not handle
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// priority, it is ignored.
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// Pitching (that is, increased speed of playback)
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// is set, but currently not used by mixing.
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//
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int I_StartSound( int id, int vol, int sep, int pitch, int priority, void *origin )
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{
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// UNUSED
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priority = 0;
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// Debug.
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// sprintf(MsgText, "starting sound %d", id );
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// WriteDebug(MsgText);
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// Returns a handle (not used).
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id = addsfx( id, vol, steptable[pitch], sep, origin );
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// sprintf(MsgText, "/handle is %d\n", id );
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// WriteDebug(MsgText);
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return id;
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}
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//
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// This function loops all active (internal) sound
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// channels, retrieves a given number of samples
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// from the raw sound data, modifies it according
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// to the current (internal) channel parameters,
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// mixes the per channel samples into the global
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// mixbuffer, clamping it to the allowed range,
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// and sets up everything for transferring the
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// contents of the mixbuffer to the (two)
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// hardware channels (left and right, that is).
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//
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// This function currently supports only 16bit.
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//
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void I_UpdateSound( void )
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{
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#ifdef SNDINTR
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// Debug. Count buffer misses with interrupt.
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static int misses = 0;
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#endif
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// Mix current sound data.
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// Data, from raw sound, for right and left.
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register unsigned int sample;
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register int dl;
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register int dr;
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// Pointers in global mixbuffer, left, right, end.
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signed short* leftout;
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signed short* rightout;
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signed short* leftend;
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// Step in mixbuffer, left and right, thus two.
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int step;
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// Mixing channel index.
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int chan;
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// Left and right channel
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// are in global mixbuffer, alternating.
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leftout = mixbuffer;
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rightout = mixbuffer+1;
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step = 2;
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// Determine end, for left channel only
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// (right channel is implicit).
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leftend = mixbuffer + SAMPLECOUNT*step;
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// Mix sounds into the mixing buffer.
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// Loop over step*SAMPLECOUNT,
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// 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_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()
|
|
{
|
|
int i;
|
|
|
|
I_CreateSound();
|
|
|
|
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)];
|
|
}
|
|
//having to crack here cuz crashing out at sound file 86 chgun
|
|
// incidentally the only 'linked' sound in the whole pile
|
|
// 10.10.98 dlw tested good.
|
|
if(i==86)
|
|
CreateSoundBuffer(i, lengths[1], S_sfx[1].data);
|
|
else
|
|
CreateSoundBuffer(i, lengths[i], S_sfx[i].data);
|
|
|
|
|
|
DSBControl[i].origin = NULL;
|
|
DSBControl[i].sfxid = i;
|
|
DSBControl[i].dsb_type = dsb_perm;
|
|
}
|
|
for (; i < NUM_DSBUFFERS; i++)
|
|
{
|
|
DSBControl[i].origin = NULL;
|
|
DSBControl[i].sfxid = -1;
|
|
DSBControl[i].dsb_type = dsb_temp;
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
//
|
|
// MUSIC API.
|
|
// Still no music done.
|
|
// Remains. Dummies.
|
|
//
|
|
void I_InitMusic(void) { }
|
|
|
|
static int looping=0;
|
|
static int musicdies=-1;
|
|
|
|
|
|
|
|
// 10.12.98 dlw - cleanup the music files
|
|
void I_ShutdownMusic(void)
|
|
{
|
|
int trasherror; //self explained
|
|
|
|
trasherror=unlink("doomsong.mus");
|
|
trasherror=unlink("doomsong.mid");
|
|
printf("I_ShutdownMusic...\n");
|
|
}
|
|
|
|
|
|
|
|
|
|
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.
|
|
|
|
// FIXME
|
|
#define ITIMER_REAL 0
|
|
static int /*__itimer_which*/ itimer = ITIMER_REAL;
|
|
|
|
// FIXME
|
|
#define SIGALRM 0
|
|
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;
|
|
}
|
|
|
|
void I_SoundDelTimer()
|
|
{
|
|
// Debug.
|
|
//if ( I_SoundSetTimer( 0 ) == -1)
|
|
// fprintf( stderr, "I_SoundDelTimer: failed to remove interrupt. Doh!\n");
|
|
}
|
|
|
|
|