/* snd_sgi.c sound support for sgi 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 $Id$ */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include "qtypes.h" #include "qargs.h" #include "sound.h" #include "console.h" static int snd_inited = 0; static ALconfig alc; static ALport alp; static int tryrates[] = { 11025, 22050, 44100, 8000 }; static unsigned char *dma_buffer, *write_buffer; static int bufsize; static int wbufp; static int framecount; qboolean SNDDMA_Init (void) { ALpv alpv; int i; char *s; alc = alNewConfig (); if (!alc) { Con_Printf ("Could not make an new sound config: %s\n", alGetErrorString (oserror ())); return 0; } shm = &sn; shm->splitbuffer = 0; /* get & probe settings */ /* sample format */ if (alSetSampFmt (alc, AL_SAMPFMT_TWOSCOMP) < 0) { Con_Printf ("Could not sample format of default output to two's " "complement\n"); alFreeConfig (alc); return 0; } /* sample bits */ s = getenv ("QUAKE_SOUND_SAMPLEBITS"); if (s) shm->samplebits = atoi (s); else if ((i = COM_CheckParm ("-sndbits")) != 0) shm->samplebits = atoi (com_argv[i + 1]); if (shm->samplebits != 16 && shm->samplebits != 8) { alpv.param = AL_WORDSIZE; if (alGetParams (AL_DEFAULT_OUTPUT, &alpv, 1) < 0) { Con_Printf ("Could not get supported wordsize of default " "output: %s\n", alGetErrorString (oserror ())); return 0; } if (alpv.value.i >= 16) { shm->samplebits = 16; } else { if (alpv.value.i >= 8) shm->samplebits = 8; else { Con_Printf ("Sound disabled since interface " "doesn't even support 8 bit."); alFreeConfig (alc); return 0; } } } /* sample rate */ s = getenv ("QUAKE_SOUND_SPEED"); if (s) shm->speed = atoi (s); else if ((i = COM_CheckParm ("-sndspeed")) != 0) shm->speed = atoi (com_argv[i + 1]); else { alpv.param = AL_RATE; for (i = 0; i < sizeof (tryrates) / sizeof (int); i++) { alpv.value.ll = alDoubleToFixed (tryrates[i]); if (alSetParams (AL_DEFAULT_OUTPUT, &alpv, 1) >= 0) break; } if (i >= sizeof (tryrates) / sizeof (int)) { Con_Printf ("Sound disabled since interface doesn't even " "support a sample rate of %d\n", tryrates[i - 1]); alFreeConfig (alc); return 0; } shm->speed = tryrates[i]; } /* channels */ s = getenv ("QUAKE_SOUND_CHANNELS"); if (s) shm->channels = atoi (s); else if ((i = COM_CheckParm ("-sndmono")) != 0) shm->channels = 1; else if ((i = COM_CheckParm ("-sndstereo")) != 0) shm->channels = 2; else shm->channels = 2; /* set 'em */ /* channels */ while (shm->channels > 0) { if (alSetChannels (alc, shm->channels) < 0) { Con_Printf ("Unable to set number of channels to %d, trying half\n", shm->channels); shm->channels /= 2; } else break; } if (shm->channels <= 0) { Con_Printf ("Sound disabled since interface doesn't even support 1 " "channel\n"); alFreeConfig (alc); return 0; } /* sample rate */ alpv.param = AL_RATE; alpv.value.ll = alDoubleToFixed (shm->speed); if (alSetParams (AL_DEFAULT_OUTPUT, &alpv, 1) < 0) { Con_Printf ("Could not set samplerate of default output to %d: %s\n", shm->speed, alGetErrorString (oserror ())); alFreeConfig (alc); return 0; } /* set sizes of buffers relative to sizes of those for ** the 'standard' frequency of 11025 ** ** use *huge* buffers since at least my indigo2 has enough ** to do to get sound on the way anyway */ bufsize = 32768 * (int) ((double) shm->speed / 11025.0); dma_buffer = malloc (bufsize); if (dma_buffer == NULL) { Con_Printf ("Could not get %d bytes of memory for audio dma buffer\n", bufsize); alFreeConfig (alc); return 0; } write_buffer = malloc (bufsize); if (write_buffer == NULL) { Con_Printf ("Could not get %d bytes of memory for audio write buffer\n", bufsize); free (dma_buffer); alFreeConfig (alc); return 0; } /* sample bits */ switch (shm->samplebits) { case 24: i = AL_SAMPLE_24; break; case 16: i = AL_SAMPLE_16; break; default: i = AL_SAMPLE_8; break; } if (alSetWidth (alc, i) < 0) { Con_Printf ("Could not set wordsize of default output to %d: %s\n", shm->samplebits, alGetErrorString (oserror ())); free (write_buffer); free (dma_buffer); alFreeConfig (alc); return 0; } alp = alOpenPort ("quakeforge", "w", alc); if (!alp) { Con_Printf ("Could not open sound port: %s\n", alGetErrorString (oserror ())); free (write_buffer); free (dma_buffer); alFreeConfig (alc); return 0; } shm->soundalive = true; shm->samples = bufsize / (shm->samplebits / 8); shm->samplepos = 0; shm->submission_chunk = 1; shm->buffer = dma_buffer; framecount = 0; snd_inited = 1; return 1; } int SNDDMA_GetDMAPos (void) { /* Con_Printf("framecount: %d %d\n", (framecount * shm->channels) % shm->samples, alGetFilled(alp)); */ shm->samplepos = ((framecount - alGetFilled (alp)) * shm->channels) % shm->samples; return shm->samplepos; } void SNDDMA_Shutdown (void) { if (snd_inited) { free (write_buffer); free (dma_buffer); alClosePort (alp); alFreeConfig (alc); snd_inited = 0; } } /* ============== SNDDMA_Submit Send sound to device if buffer isn't really the dma buffer =============== */ void SNDDMA_Submit (void) { int bsize; int bytes, b; unsigned char *p; int idx; int stop = paintedtime; if (paintedtime < wbufp) wbufp = 0; // reset bsize = shm->channels * (shm->samplebits / 8); bytes = (paintedtime - wbufp) * bsize; if (!bytes) return; if (bytes > bufsize) { bytes = bufsize; stop = wbufp + bytes / bsize; } p = write_buffer; idx = (wbufp * bsize) & (bufsize - 1); for (b = bytes; b; b--) { *p++ = dma_buffer[idx]; idx = (idx + 1) & (bufsize - 1); } wbufp = stop; alWriteFrames (alp, write_buffer, bytes / bsize); framecount += bytes / bsize; } /* end of file */