Added alsa support
git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@1083 fc73d0e0-1445-4013-8a0c-d673dee63da5
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Spoike
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/*
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snd_alsa.c
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Support for the ALSA 1.0.1 sound driver
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Copyright (C) 1999,2000 contributors of the QuakeForge project
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Please see the file "AUTHORS" for a list of contributors
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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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//actually stolen from darkplaces.
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//I guess noone can be arsed to write it themselves. :/
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#include <alsa/asoundlib.h>
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#include "quakedef.h"
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static int snd_inited;
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static snd_pcm_uframes_t buffer_size;
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static const char *pcmname = NULL;
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static snd_pcm_t *pcm;
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soundcardinfo_t *sndcardinfo;
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qboolean snd_firsttime;
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int SNDDMA_Init (soundcardinfo_t *sc)
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{
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int err, i;
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int bps = -1, stereo = -1;
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unsigned int rate = 0;
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snd_pcm_hw_params_t *hw;
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snd_pcm_sw_params_t *sw;
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snd_pcm_uframes_t frag_size;
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snd_pcm_hw_params_alloca (&hw);
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snd_pcm_sw_params_alloca (&sw);
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// COMMANDLINEOPTION: Linux ALSA Sound: -sndpcm <devicename> selects which pcm device to us, default is "default"
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if ((i=COM_CheckParm("-sndpcm"))!=0)
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pcmname=com_argv[i+1];
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if (!pcmname)
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pcmname = "default";
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// COMMANDLINEOPTION: Linux ALSA Sound: -sndbits <number> sets sound precision to 8 or 16 bit (email me if you want others added)
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if ((i=COM_CheckParm("-sndbits")) != 0)
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{
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bps = atoi(com_argv[i+1]);
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if (bps != 16 && bps != 8)
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{
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Con_Printf("Error: invalid sample bits: %d\n", bps);
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return false;
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}
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}
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// COMMANDLINEOPTION: Linux ALSA Sound: -sndspeed <hz> chooses 44100 hz, 22100 hz, or 11025 hz sound output rate
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if ((i=COM_CheckParm("-sndspeed")) != 0)
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{
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rate = atoi(com_argv[i+1]);
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if (rate!=44100 && rate!=22050 && rate!=11025)
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{
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Con_Printf("Error: invalid sample rate: %d\n", rate);
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return false;
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}
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}
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// COMMANDLINEOPTION: Linux ALSA Sound: -sndmono sets sound output to mono
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if ((i=COM_CheckParm("-sndmono")) != 0)
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stereo=0;
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// COMMANDLINEOPTION: Linux ALSA Sound: -sndstereo sets sound output to stereo
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if ((i=COM_CheckParm("-sndstereo")) != 0)
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stereo=1;
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err = snd_pcm_open (&pcm, pcmname, SND_PCM_STREAM_PLAYBACK,
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SND_PCM_NONBLOCK);
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if (0 > err) {
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Con_Printf ("Error: audio open error: %s\n", snd_strerror (err));
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return 0;
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}
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Con_Printf ("ALSA: Using PCM %s.\n", pcmname);
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err = snd_pcm_hw_params_any (pcm, hw);
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if (0 > err) {
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Con_Printf ("ALSA: error setting hw_params_any. %s\n",
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snd_strerror (err));
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goto error;
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}
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err = snd_pcm_hw_params_set_access (pcm, hw,
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SND_PCM_ACCESS_MMAP_INTERLEAVED);
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if (0 > err) {
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Con_Printf ("ALSA: Failure to set noninterleaved PCM access. %s\n"
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"Note: Interleaved is not supported\n",
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snd_strerror (err));
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goto error;
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}
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switch (bps) {
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case -1:
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err = snd_pcm_hw_params_set_format (pcm, hw,
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SND_PCM_FORMAT_S16);
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if (0 <= err) {
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bps = 16;
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} else if (0 <= (err = snd_pcm_hw_params_set_format (pcm, hw,
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SND_PCM_FORMAT_U8))) {
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bps = 8;
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} else {
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Con_Printf ("ALSA: no useable formats. %s\n",
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snd_strerror (err));
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goto error;
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}
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break;
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case 8:
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case 16:
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err = snd_pcm_hw_params_set_format (pcm, hw, bps == 8 ?
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SND_PCM_FORMAT_U8 :
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SND_PCM_FORMAT_S16);
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if (0 > err) {
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Con_Printf ("ALSA: no usable formats. %s\n",
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snd_strerror (err));
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goto error;
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}
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break;
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default:
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Con_Printf ("ALSA: desired format not supported\n");
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goto error;
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}
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switch (stereo) {
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case -1:
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err = snd_pcm_hw_params_set_channels (pcm, hw, 2);
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if (0 <= err) {
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stereo = 1;
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} else if (0 <= (err = snd_pcm_hw_params_set_channels (pcm, hw,
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1))) {
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stereo = 0;
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} else {
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Con_Printf ("ALSA: no usable channels. %s\n",
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snd_strerror (err));
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goto error;
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}
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break;
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case 0:
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case 1:
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err = snd_pcm_hw_params_set_channels (pcm, hw, stereo ? 2 : 1);
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if (0 > err) {
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Con_Printf ("ALSA: no usable channels. %s\n",
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snd_strerror (err));
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goto error;
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}
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break;
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default:
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Con_Printf ("ALSA: desired channels not supported\n");
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goto error;
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}
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switch (rate) {
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case 0:
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rate = 44100;
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err = snd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
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if (0 <= err) {
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frag_size = 32 * bps;
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} else {
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rate = 22050;
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err = snd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
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if (0 <= err) {
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frag_size = 16 * bps;
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} else {
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rate = 11025;
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err = snd_pcm_hw_params_set_rate_near (pcm, hw, &rate,
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0);
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if (0 <= err) {
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frag_size = 8 * bps;
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} else {
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Con_Printf ("ALSA: no usable rates. %s\n",
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snd_strerror (err));
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goto error;
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}
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}
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}
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break;
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case 11025:
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case 22050:
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case 44100:
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err = snd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
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if (0 > err) {
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Con_Printf ("ALSA: desired rate %i not supported. %s\n", rate,
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snd_strerror (err));
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goto error;
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}
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frag_size = 8 * bps * rate / 11025;
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break;
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default:
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Con_Printf ("ALSA: desired rate %i not supported.\n", rate);
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goto error;
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}
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err = snd_pcm_hw_params_set_period_size_near (pcm, hw, &frag_size, 0);
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if (0 > err) {
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Con_Printf ("ALSA: unable to set period size near %i. %s\n",
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(int) frag_size, snd_strerror (err));
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goto error;
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}
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err = snd_pcm_hw_params (pcm, hw);
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if (0 > err) {
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Con_Printf ("ALSA: unable to install hw params: %s\n",
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snd_strerror (err));
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goto error;
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}
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err = snd_pcm_sw_params_current (pcm, sw);
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if (0 > err) {
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Con_Printf ("ALSA: unable to determine current sw params. %s\n",
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snd_strerror (err));
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goto error;
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}
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err = snd_pcm_sw_params_set_start_threshold (pcm, sw, ~0U);
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if (0 > err) {
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Con_Printf ("ALSA: unable to set playback threshold. %s\n",
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snd_strerror (err));
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goto error;
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}
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err = snd_pcm_sw_params_set_stop_threshold (pcm, sw, ~0U);
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if (0 > err) {
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Con_Printf ("ALSA: unable to set playback stop threshold. %s\n",
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snd_strerror (err));
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goto error;
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}
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err = snd_pcm_sw_params (pcm, sw);
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if (0 > err) {
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Con_Printf ("ALSA: unable to install sw params. %s\n",
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snd_strerror (err));
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goto error;
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}
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sc->sn.numchannels = stereo + 1;
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sc->sn.samplepos = 0;
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sc->sn.samplebits = bps;
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err = snd_pcm_hw_params_get_buffer_size (hw, &buffer_size);
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if (0 > err) {
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Con_Printf ("ALSA: unable to get buffer size. %s\n",
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snd_strerror (err));
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goto error;
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}
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sc->sn.samples = buffer_size * sc->sn.numchannels; // mono samples in buffer
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sc->sn.speed = rate;
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SNDDMA_GetDMAPos (sc); // sets shm->buffer
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snd_inited = 1;
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return true;
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error:
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snd_pcm_close (pcm);
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return false;
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}
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int SNDDMA_GetDMAPos (soundcardinfo_t *sc)
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{
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const snd_pcm_channel_area_t *areas;
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snd_pcm_uframes_t offset;
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snd_pcm_uframes_t nframes = sc->sn.samples / sc->sn.numchannels;
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if (!snd_inited)
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return 0;
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snd_pcm_avail_update (pcm);
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snd_pcm_mmap_begin (pcm, &areas, &offset, &nframes);
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offset *= sc->sn.numchannels;
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nframes *= sc->sn.numchannels;
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sc->sn.samplepos = offset;
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sc->sn.buffer = areas->addr;
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return sc->sn.samplepos;
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}
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void SNDDMA_Shutdown (soundcardinfo_t *sc)
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{
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if (snd_inited) {
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snd_pcm_close (pcm);
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snd_inited = 0;
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}
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}
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/*
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SNDDMA_Submit
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Send sound to device if buffer isn't really the dma buffer
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*/
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void SNDDMA_Submit (soundcardinfo_t *sc)
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{
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extern int soundtime;
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int state;
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int count = sc->paintedtime - soundtime;
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const snd_pcm_channel_area_t *areas;
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snd_pcm_uframes_t nframes;
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snd_pcm_uframes_t offset;
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nframes = count / sc->sn.numchannels;
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snd_pcm_avail_update (pcm);
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snd_pcm_mmap_begin (pcm, &areas, &offset, &nframes);
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state = snd_pcm_state (pcm);
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switch (state) {
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case SND_PCM_STATE_PREPARED:
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snd_pcm_mmap_commit (pcm, offset, nframes);
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snd_pcm_start (pcm);
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break;
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case SND_PCM_STATE_RUNNING:
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snd_pcm_mmap_commit (pcm, offset, nframes);
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break;
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default:
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break;
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}
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}
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void *S_LockBuffer(soundcardinfo_t *sc)
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{
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return sc->sn.buffer;
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}
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void S_UnlockBuffer(soundcardinfo_t *sc)
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{
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}
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void SNDDMA_SetUnderWater(qboolean underwater)
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{
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}
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void S_UpdateCapture(void)
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{
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}
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