quakeforge/libs/audio/renderer/snd_dma.c

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