0345dadebf
Also now has working input (although not perfect) - Needs mouse or touch. Also now supports screen size doubling when the screen is significantly larger than doom's screen.
821 lines
18 KiB
C
821 lines
18 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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#include "gfx.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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// 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 8
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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_CHANNELS];
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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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// 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*
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getsfx
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( char* sfxname,
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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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// Get the sound data from the WAD, allocate lump
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// in zone memory.
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I_sprintf(name, "ds%s", sfxname);
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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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// Debug.
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// I_DBGprintf( "." );
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//I_DBGprintf( " -loading %s (lump %d, %d bytes)\n",
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// sfxname, sfxlump, size );
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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 *) I_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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// 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
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addsfx
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( int sfxid,
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int volume,
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int step,
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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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// Chainsaw troubles.
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// Play these sound effects only one at a time.
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if ( sfxid == sfx_sawup
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|| sfxid == sfx_sawidl
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|| sfxid == sfx_sawful
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|| sfxid == sfx_sawhit
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|| sfxid == sfx_stnmov
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|| 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])
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&& (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,
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// 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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// 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 =
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volume - ((volume*seperation*seperation) >> 16); ///(256*256);
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seperation = seperation - 257;
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rightvol =
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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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// You tell me.
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return rc;
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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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#if 0
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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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#endif
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}
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void I_SetSfxVolume(int volume)
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{
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#if 0
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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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#endif
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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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#if 0
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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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#endif
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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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I_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
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I_StartSound
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( int id,
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int vol,
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int sep,
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int pitch,
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int priority )
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{
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#if 0
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// UNUSED
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priority = 0;
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#ifdef SNDSERV
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if (sndserver)
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{
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fprintf(sndserver, "p%2.2x%2.2x%2.2x%2.2x\n", id, pitch, vol, sep);
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fflush(sndserver);
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}
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// warning: control reaches end of non-void function.
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return id;
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#else
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// Debug.
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//I_DBGprintf("starting sound %d", id );
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// Returns a handle (not used).
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id = addsfx( id, vol, steptable[pitch], sep );
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// I_DBGprintf("/handle is %d\n", id );
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return id;
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#endif
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#else
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return id;
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#endif
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}
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void I_StopSound (int handle)
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{
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// You need the handle returned by StartSound.
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// Would be looping all channels,
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// tracking down the handle,
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// an setting the channel to zero.
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// UNUSED.
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handle = 0;
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}
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int I_SoundIsPlaying(int handle)
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{
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#if 0
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// Ouch.
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return gametic < handle;
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#else
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return 0;
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#endif
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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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#if 0
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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.
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while (leftout != leftend)
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{
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// Reset left/right value.
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dl = 0;
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dr = 0;
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// Love thy L2 chache - made this a loop.
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// Now more channels could be set at compile time
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// as well. Thus loop those channels.
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for ( chan = 0; chan < NUM_CHANNELS; chan++ )
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{
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// Check channel, if active.
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if (channels[ chan ])
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{
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// Get the raw data from the channel.
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sample = *channels[ chan ];
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// Add left and right part
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// for this channel (sound)
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// to the current data.
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// Adjust volume accordingly.
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dl += channelleftvol_lookup[ chan ][sample];
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dr += channelrightvol_lookup[ chan ][sample];
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// Increment index ???
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channelstepremainder[ chan ] += channelstep[ chan ];
|
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// MSB is next sample???
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channels[ chan ] += channelstepremainder[ chan ] >> 16;
|
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// Limit to LSB???
|
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channelstepremainder[ chan ] &= 65536-1;
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|
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// Check whether we are done.
|
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if (channels[ chan ] >= channelsend[ chan ])
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channels[ chan ] = 0;
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}
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}
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// Clamp to range. Left hardware channel.
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// Has been char instead of short.
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// if (dl > 127) *leftout = 127;
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// else if (dl < -128) *leftout = -128;
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// else *leftout = dl;
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if (dl > 0x7fff)
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*leftout = 0x7fff;
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else if (dl < -0x8000)
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*leftout = -0x8000;
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else
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*leftout = dl;
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// Same for right hardware channel.
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if (dr > 0x7fff)
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*rightout = 0x7fff;
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else if (dr < -0x8000)
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*rightout = -0x8000;
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else
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*rightout = dr;
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// Increment current pointers in mixbuffer.
|
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leftout += step;
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rightout += step;
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}
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|
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#ifdef SNDINTR
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// Debug check.
|
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if ( flag )
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{
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misses += flag;
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flag = 0;
|
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}
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|
|
if ( misses > 10 )
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|
{
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|
I_DBGprintf("I_SoundUpdate: missed 10 buffer writes\n");
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misses = 0;
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}
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|
|
// Increment flag for update.
|
|
flag++;
|
|
#endif
|
|
|
|
#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)
|
|
{
|
|
#if 0
|
|
// Write it to DSP device.
|
|
write(audio_fd, mixbuffer, SAMPLECOUNT*BUFMUL);
|
|
#endif
|
|
}
|
|
|
|
|
|
|
|
void
|
|
I_UpdateSoundParams
|
|
( int handle,
|
|
int vol,
|
|
int sep,
|
|
int pitch)
|
|
{
|
|
#if 0
|
|
// 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;
|
|
#endif
|
|
}
|
|
|
|
|
|
|
|
|
|
void I_ShutdownSound(void)
|
|
{
|
|
}
|
|
|
|
void I_InitSound()
|
|
{
|
|
#if 0
|
|
|
|
#ifdef SNDSERV
|
|
char buffer[256];
|
|
|
|
if (getenv("DOOMWADDIR"))
|
|
I_sprintf(buffer, "%s/%s",
|
|
getenv("DOOMWADDIR"),
|
|
sndserver_filename);
|
|
else
|
|
I_sprintf(buffer, "%s", sndserver_filename);
|
|
|
|
// start sound process
|
|
if ( !access(buffer, X_OK) )
|
|
{
|
|
strcat(buffer, " -quiet");
|
|
sndserver = popen(buffer, "w");
|
|
}
|
|
else
|
|
I_DBGprintf("Could not start sound server [%s]\n", buffer);
|
|
#else
|
|
|
|
int i;
|
|
|
|
#ifdef SNDINTR
|
|
I_DBGprintf("I_SoundSetTimer: %d microsecs\n", SOUND_INTERVAL );
|
|
I_SoundSetTimer( SOUND_INTERVAL );
|
|
#endif
|
|
|
|
// Secure and configure sound device first.
|
|
I_DBGprintf( "I_InitSound: ");
|
|
|
|
audio_fd = open("/dev/dsp", O_WRONLY);
|
|
if (audio_fd<0)
|
|
I_DBGprintf( "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
|
|
I_DBGprintf("Could not play signed 16 data\n");
|
|
|
|
I_DBGprintf(" configured audio device\n" );
|
|
|
|
|
|
// Initialize external data (all sounds) at start, keep static.
|
|
I_DBGprintf("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)];
|
|
}
|
|
}
|
|
|
|
I_DBGprintf(" pre-cached all sound data\n");
|
|
|
|
// Now initialize mixbuffer with zero.
|
|
for ( i = 0; i< MIXBUFFERSIZE; i++ )
|
|
mixbuffer[i] = 0;
|
|
|
|
// Finished initialization.
|
|
I_DBGprintf("I_InitSound: sound module ready\n");
|
|
|
|
#endif
|
|
|
|
#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;
|
|
}
|
|
|
|
// Interrupt handler.
|
|
void I_HandleSoundTimer( int ignore )
|
|
{
|
|
#if 0
|
|
// Debug.
|
|
//I_DBGprintf("%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;
|
|
#endif
|
|
}
|