493 lines
10 KiB
C
493 lines
10 KiB
C
/*
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snd_dma.c
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main control for any streaming sound output device
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Copyright (C) 1996-1997 Id Software, Inc.
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This program is free software; you can redistribute it and/or
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modify it under the terms of the GNU General Public License
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as published by the Free Software Foundation; either version 2
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of the License, or (at your option) any later version.
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This program 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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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
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See the GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, write to:
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Free Software Foundation, Inc.
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59 Temple Place - Suite 330
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Boston, MA 02111-1307, USA
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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static __attribute__ ((used)) const char rcsid[] =
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"$Id$";
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#ifdef HAVE_STRING_H
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# include <string.h>
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#endif
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#ifdef HAVE_STRINGS_H
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# include <strings.h>
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#endif
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#include <stdlib.h>
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#include "winquake.h"
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#include "QF/cmd.h"
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#include "QF/cvar.h"
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#include "QF/dstring.h"
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#include "QF/model.h"
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#include "QF/qargs.h"
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#include "QF/sys.h"
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#include "QF/sound.h"
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#include "QF/plugin.h"
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#include "QF/va.h"
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#include "QF/quakefs.h"
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#include "snd_render.h"
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static qboolean snd_initialized = false;
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static int snd_blocked = 0;
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static unsigned soundtime; // sample PAIRS
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static int sound_started = 0;
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static cvar_t *nosound;
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static cvar_t *snd_mixahead;
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static cvar_t *snd_noextraupdate;
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static cvar_t *snd_show;
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static general_data_t plugin_info_general_data;
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static snd_output_funcs_t *snd_output_funcs;
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static int *snd_p, snd_linear_count, snd_vol;
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static short *snd_out;
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static void
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SND_WriteLinearBlastStereo16 (void)
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{
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int val, i;
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for (i = 0; i < snd_linear_count; i += 2) {
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val = (snd_p[i] * snd_vol) >> 8;
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if (val > 0x7fff)
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snd_out[i] = 0x7fff;
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else if (val < (short) 0x8000)
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snd_out[i] = (short) 0x8000;
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else
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snd_out[i] = val;
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val = (snd_p[i + 1] * snd_vol) >> 8;
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if (val > 0x7fff)
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snd_out[i + 1] = 0x7fff;
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else if (val < (short) 0x8000)
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snd_out[i + 1] = (short) 0x8000;
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else
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snd_out[i + 1] = val;
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}
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}
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static void
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s_xfer_stereo_16 (int endtime)
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{
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int lpaintedtime, lpos;
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unsigned int *pbuf;
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snd_vol = snd_volume->value * 256;
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snd_p = (int *) snd_paintbuffer;
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lpaintedtime = snd_paintedtime;
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pbuf = (unsigned int *) snd_shm->buffer;
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while (lpaintedtime < endtime) {
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// handle recirculating buffer issues
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lpos = lpaintedtime & ((snd_shm->samples >> 1) - 1);
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snd_out = (short *) pbuf + (lpos << 1);
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snd_linear_count = (snd_shm->samples >> 1) - lpos;
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if (lpaintedtime + snd_linear_count > endtime)
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snd_linear_count = endtime - lpaintedtime;
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snd_linear_count <<= 1;
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// write a linear blast of samples
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SND_WriteLinearBlastStereo16 ();
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snd_p += snd_linear_count;
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lpaintedtime += (snd_linear_count >> 1);
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}
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}
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static void
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s_xfer_paint_buffer (int endtime)
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{
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int count, out_idx, out_mask, snd_vol, step, val;
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int *p;
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unsigned int *pbuf;
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if (snd_shm->samplebits == 16 && snd_shm->channels == 2) {
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s_xfer_stereo_16 (endtime);
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return;
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}
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p = (int *) snd_paintbuffer;
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count = (endtime - snd_paintedtime) * snd_shm->channels;
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out_mask = snd_shm->samples - 1;
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out_idx = snd_paintedtime * snd_shm->channels & out_mask;
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step = 3 - snd_shm->channels;
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snd_vol = snd_volume->value * 256;
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pbuf = (unsigned int *) snd_shm->buffer;
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if (snd_shm->samplebits == 16) {
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short *out = (short *) pbuf;
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while (count--) {
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val = (*p * snd_vol) >> 8;
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p += step;
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if (val > 0x7fff)
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val = 0x7fff;
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else if (val < (short) 0x8000)
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val = (short) 0x8000;
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out[out_idx] = val;
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out_idx = (out_idx + 1) & out_mask;
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}
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} else if (snd_shm->samplebits == 8) {
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unsigned char *out = (unsigned char *) pbuf;
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while (count--) {
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val = (*p * snd_vol) >> 8;
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p += step;
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if (val > 0x7fff)
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val = 0x7fff;
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else if (val < (short) 0x8000)
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val = (short) 0x8000;
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out[out_idx] = (val >> 8) + 128;
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out_idx = (out_idx + 1) & out_mask;
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}
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}
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}
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static void
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s_clear_buffer (void)
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{
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int clear, i;
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if (!sound_started || !snd_shm || !snd_shm->buffer)
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return;
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if (snd_shm->samplebits == 8)
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clear = 0x80;
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else
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clear = 0;
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for (i = 0; i < snd_shm->samples * snd_shm->samplebits / 8; i++)
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snd_shm->buffer[i] = clear;
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}
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static void
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s_stop_all_sounds (void)
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{
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SND_StopAllSounds ();
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SND_ScanChannels (0);
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s_clear_buffer ();
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}
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//=============================================================================
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static void
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s_get_soundtime (void)
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{
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int fullsamples, samplepos;
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static int buffers, oldsamplepos;
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fullsamples = snd_shm->samples / snd_shm->channels;
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// it is possible to miscount buffers if it has wrapped twice between
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// calls to s_update. Oh well.
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if ((samplepos = snd_output_funcs->pS_O_GetDMAPos ()) == -1)
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return;
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if (samplepos < oldsamplepos) {
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buffers++; // buffer wrapped
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if (snd_paintedtime > 0x40000000) { // time to chop things off to avoid
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// 32 bit limits
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buffers = 0;
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snd_paintedtime = fullsamples;
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s_stop_all_sounds ();
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}
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}
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oldsamplepos = samplepos;
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soundtime = buffers * fullsamples + samplepos / snd_shm->channels;
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}
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static void
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s_update_ (void)
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{
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unsigned int endtime, samps;
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if (!sound_started || (snd_blocked > 0))
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return;
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// Updates DMA time
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s_get_soundtime ();
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// check to make sure that we haven't overshot
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if (snd_paintedtime < soundtime) {
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// Sys_Printf ("S_Update_ : overflow\n");
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snd_paintedtime = soundtime;
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}
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// mix ahead of current position
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endtime = soundtime + snd_mixahead->value * snd_shm->speed;
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samps = snd_shm->samples >> (snd_shm->channels - 1);
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if (endtime - soundtime > samps)
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endtime = soundtime + samps;
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SND_PaintChannels (endtime);
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snd_output_funcs->pS_O_Submit ();
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}
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/*
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s_update
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Called once each time through the main loop
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*/
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static void
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s_update (const vec3_t origin, const vec3_t forward, const vec3_t right,
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const vec3_t up)
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{
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if (!sound_started || (snd_blocked > 0))
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return;
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SND_SetListener (origin, forward, right, up);
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// mix some sound
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s_update_ ();
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SND_ScanChannels (0);
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}
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static void
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s_extra_update (void)
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{
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if (!sound_started || snd_noextraupdate->int_val)
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return; // don't pollute timings
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s_update_ ();
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}
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static void
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s_block_sound (void)
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{
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if (++snd_blocked == 1) {
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snd_output_funcs->pS_O_BlockSound ();
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s_clear_buffer ();
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}
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}
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static void
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s_unblock_sound (void)
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{
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if (!snd_blocked)
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return;
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if (!--snd_blocked) {
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s_clear_buffer ();
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snd_output_funcs->pS_O_UnblockSound ();
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}
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}
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/* console functions */
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static void
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s_soundinfo_f (void)
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{
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if (!sound_started || !snd_shm) {
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Sys_Printf ("sound system not started\n");
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return;
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}
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Sys_Printf ("%5d stereo\n", snd_shm->channels - 1);
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Sys_Printf ("%5d samples\n", snd_shm->samples);
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Sys_Printf ("%5d samplepos\n", snd_shm->samplepos);
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Sys_Printf ("%5d samplebits\n", snd_shm->samplebits);
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Sys_Printf ("%5d submission_chunk\n", snd_shm->submission_chunk);
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Sys_Printf ("%5d speed\n", snd_shm->speed);
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Sys_Printf ("0x%p dma buffer\n", snd_shm->buffer);
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Sys_Printf ("%5d total_channels\n", snd_total_channels);
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}
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static void
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s_stop_all_sounds_f (void)
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{
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s_stop_all_sounds ();
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}
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static void
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s_startup (void)
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{
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if (!snd_initialized)
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return;
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snd_shm = snd_output_funcs->pS_O_Init ();
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if (!snd_shm) {
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Sys_Printf ("S_Startup: S_O_Init failed.\n");
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sound_started = 0;
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return;
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}
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snd_shm->xfer = s_xfer_paint_buffer;
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if (snd_shm->speed > 44100) {
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Sys_Printf ("FIXME clamping Sps to 44100 until resampling is fixed\n");
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snd_shm->speed = 44100;
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}
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sound_started = 1;
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}
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static void
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s_snd_force_unblock (void)
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{
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snd_blocked = 1;
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s_unblock_sound ();
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}
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static void
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s_init (void)
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{
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snd_output_funcs = snd_render_data.output->functions->snd_output;
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snd_render_data.soundtime = &soundtime;
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Sys_Printf ("\nSound Initialization\n");
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Cmd_AddCommand ("stopsound", s_stop_all_sounds_f,
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"Stops all sounds currently being played");
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Cmd_AddCommand ("soundinfo", s_soundinfo_f,
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"Report information on the sound system");
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Cmd_AddCommand ("snd_force_unblock", s_snd_force_unblock,
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"fix permanently blocked sound");
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nosound = Cvar_Get ("nosound", "0", CVAR_NONE, NULL,
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"Set to turn sound off");
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snd_volume = Cvar_Get ("volume", "0.7", CVAR_ARCHIVE, NULL,
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"Set the volume for sound playback");
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snd_interp = Cvar_Get ("snd_interp", "1", CVAR_ARCHIVE, NULL,
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"control sample interpolation");
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snd_loadas8bit = Cvar_Get ("snd_loadas8bit", "0", CVAR_NONE, NULL,
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"Toggles loading sounds as 8-bit samples");
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snd_mixahead = Cvar_Get ("snd_mixahead", "0.1", CVAR_ARCHIVE, NULL,
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"Delay time for sounds");
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snd_noextraupdate = Cvar_Get ("snd_noextraupdate", "0", CVAR_NONE, NULL,
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"Toggles the correct value display in "
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"host_speeds. Usually messes up sound "
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"playback when in effect");
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snd_show = Cvar_Get ("snd_show", "0", CVAR_NONE, NULL,
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"Toggles display of sounds currently being played");
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// FIXME
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// if (host_parms.memsize < 0x800000) {
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// Cvar_Set (snd_loadas8bit, "1");
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// Sys_Printf ("loading all sounds as 8bit\n");
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// }
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snd_initialized = true;
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s_startup ();
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if (sound_started == 0) // sound startup failed? Bail out.
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return;
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SND_SFX_Init ();
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SND_Channels_Init ();
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SND_InitScaletable ();
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s_stop_all_sounds ();
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}
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static void
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s_shutdown (void)
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{
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if (!sound_started)
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return;
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sound_started = 0;
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snd_output_funcs->pS_O_Shutdown ();
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snd_shm = 0;
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}
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static general_funcs_t plugin_info_general_funcs = {
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s_init,
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s_shutdown,
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};
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static snd_render_funcs_t plugin_info_render_funcs = {
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SND_AmbientOff,
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SND_AmbientOn,
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SND_StaticSound,
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SND_StartSound,
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SND_StopSound,
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SND_PrecacheSound,
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s_update,
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s_stop_all_sounds,
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s_extra_update,
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SND_LocalSound,
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s_block_sound,
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s_unblock_sound,
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SND_LoadSound,
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SND_AllocChannel,
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SND_ChannelStop,
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};
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static plugin_funcs_t plugin_info_funcs = {
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&plugin_info_general_funcs,
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0,
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0,
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0,
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0,
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&plugin_info_render_funcs,
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};
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static plugin_data_t plugin_info_data = {
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&plugin_info_general_data,
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0,
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0,
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0,
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0,
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&snd_render_data,
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};
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static plugin_t plugin_info = {
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qfp_snd_render,
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0,
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QFPLUGIN_VERSION,
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"0.1",
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"Sound Renderer",
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"Copyright (C) 1996-1997 id Software, Inc.\n"
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"Copyright (C) 1999,2000,2001 contributors of the QuakeForge "
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"project\n"
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"Please see the file \"AUTHORS\" for a list of contributors",
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&plugin_info_funcs,
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&plugin_info_data,
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};
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PLUGIN_INFO(snd_render, default)
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{
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return &plugin_info;
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}
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