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https://git.code.sf.net/p/quake/quakeforge
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0e6c56aa92
the alsa0_9 plugin to alsa.
397 lines
10 KiB
C
397 lines
10 KiB
C
/*
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snd_alsa.c
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Support for ALSA 0.9 sound driver (cvs development version)
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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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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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static __attribute__ ((unused)) const char rcsid[] =
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"$Id$";
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#include <stdio.h>
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#include <dlfcn.h>
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#include <alsa/asoundlib.h>
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#include "QF/cvar.h"
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#include "QF/plugin.h"
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#include "QF/qargs.h"
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#include "QF/sound.h"
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#include "QF/sys.h"
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static int snd_inited;
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static snd_pcm_t *pcm;
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static const char *pcmname = NULL;
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static size_t buffer_size;
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static int snd_blocked = 0;
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static volatile dma_t sn;
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static cvar_t *snd_stereo;
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static cvar_t *snd_rate;
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static cvar_t *snd_device;
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static cvar_t *snd_bits;
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static plugin_t plugin_info;
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static plugin_data_t plugin_info_data;
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static plugin_funcs_t plugin_info_funcs;
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static general_data_t plugin_info_general_data;
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static general_funcs_t plugin_info_general_funcs;
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static snd_output_data_t plugin_info_snd_output_data;
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static snd_output_funcs_t plugin_info_snd_output_funcs;
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static void *alsa_handle;
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#define QF_ALSA_NEED(ret, func, params) \
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static ret (*qf##func) params;
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#include "alsa_funcs_list.h"
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#undef QF_ALSA_NEED
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static qboolean
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load_libasound (void)
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{
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if (!(alsa_handle = dlopen ("libasound.so.2", RTLD_GLOBAL | RTLD_NOW))) {
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Sys_Printf ("couldn't load libasound.so.2: %s\n", dlerror ());
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return false;
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}
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#define QF_ALSA_NEED(ret, func, params) \
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if (!(qf##func = dlsym (alsa_handle, #func))) { \
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Sys_Printf ("Couldn't load ALSA function %s\n", #func); \
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dlclose (alsa_handle); \
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alsa_handle = 0; \
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return false; \
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}
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#include "alsa_funcs_list.h"
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#undef QF_ALSA_NEED
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return true;
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}
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#define snd_pcm_hw_params_sizeof qfsnd_pcm_hw_params_sizeof
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#define snd_pcm_sw_params_sizeof qfsnd_pcm_sw_params_sizeof
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static void
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SNDDMA_Init_Cvars (void)
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{
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snd_stereo = Cvar_Get ("snd_stereo", "1", CVAR_ROM, NULL,
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"sound stereo output");
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snd_rate = Cvar_Get ("snd_rate", "0", CVAR_ROM, NULL,
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"sound playback rate. 0 is system default");
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snd_device = Cvar_Get ("snd_device", "", CVAR_ROM, NULL,
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"sound device. \"\" is system default");
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snd_bits = Cvar_Get ("snd_bits", "0", CVAR_ROM, NULL,
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"sound sample depth. 0 is system default");
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}
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static int SNDDMA_GetDMAPos (void);
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static qboolean
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SNDDMA_Init (void)
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{
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int err, frag_size;
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int rate = -1, bps = -1, stereo = -1;
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snd_pcm_hw_params_t *hw;
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snd_pcm_sw_params_t *sw;
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if (!load_libasound ())
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return false;
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snd_pcm_hw_params_alloca (&hw);
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snd_pcm_sw_params_alloca (&sw);
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if (snd_device->string[0])
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pcmname = snd_device->string;
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if (snd_bits->int_val) {
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bps = snd_bits->int_val;
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if (bps != 16 && bps != 8) {
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Sys_Printf ("Error: invalid sample bits: %d\n", bps);
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return 0;
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}
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}
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if (snd_rate->int_val) {
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rate = snd_rate->int_val;
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if (rate != 44100 && rate != 22050 && rate != 11025) {
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Sys_Printf ("Error: invalid sample rate: %d\n", rate);
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return 0;
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}
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}
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stereo = snd_stereo->int_val;
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if (!pcmname)
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pcmname = "default";
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if ((err = qfsnd_pcm_open (&pcm, pcmname,
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SND_PCM_STREAM_PLAYBACK,
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SND_PCM_NONBLOCK)) < 0) {
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Sys_Printf ("Error: audio open error: %s\n", qfsnd_strerror (err));
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return 0;
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}
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Sys_Printf ("Using PCM %s.\n", pcmname);
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qfsnd_pcm_hw_params_any (pcm, hw);
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switch (rate) {
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case -1:
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if (qfsnd_pcm_hw_params_set_rate_near (pcm, hw, 44100, 0) >= 0) {
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frag_size = 256; /* assuming stereo 8 bit */
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rate = 44100;
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} else if (qfsnd_pcm_hw_params_set_rate_near (pcm, hw,
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22050, 0) >= 0) {
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frag_size = 128; /* assuming stereo 8 bit */
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rate = 22050;
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} else if (qfsnd_pcm_hw_params_set_rate_near (pcm, hw,
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11025, 0) >= 0) {
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frag_size = 64; /* assuming stereo 8 bit */
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rate = 11025;
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} else {
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Sys_Printf ("ALSA: no useable rates\n");
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goto error;
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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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if (qfsnd_pcm_hw_params_set_rate_near (pcm, hw, rate, 0) >= 0) {
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frag_size = 64 * rate / 11025; /* assuming stereo 8 bit */
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break;
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}
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/* Fall through */
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default:
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Sys_Printf ("ALSA: desired rate not supported\n");
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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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if (qfsnd_pcm_hw_params_set_format (pcm, hw,
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SND_PCM_FORMAT_S16_LE) >= 0) {
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bps = 16;
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} else if (qfsnd_pcm_hw_params_set_format (pcm, hw,
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SND_PCM_FORMAT_U8)
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>= 0) {
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bps = 8;
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} else {
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Sys_Printf ("ALSA: no useable formats\n");
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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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if (qfsnd_pcm_hw_params_set_format (pcm, hw,
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bps == 8 ? SND_PCM_FORMAT_U8 :
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SND_PCM_FORMAT_S16) >= 0) {
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break;
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}
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/* Fall through */
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default:
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Sys_Printf ("ALSA: desired format not supported\n");
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goto error;
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}
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if (qfsnd_pcm_hw_params_set_access (pcm, hw,
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SND_PCM_ACCESS_MMAP_INTERLEAVED) < 0) {
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Sys_Printf ("ALSA: interleaved is 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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if (qfsnd_pcm_hw_params_set_channels (pcm, hw, 2) >= 0) {
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stereo = 1;
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} else if (qfsnd_pcm_hw_params_set_channels (pcm, hw, 1) >= 0) {
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stereo = 0;
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} else {
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Sys_Printf ("ALSA: no useable channels\n");
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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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if (qfsnd_pcm_hw_params_set_channels (pcm, hw, stereo ? 2 : 1) >= 0)
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break;
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/* Fall through */
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default:
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Sys_Printf ("ALSA: desired channels not supported\n");
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goto error;
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}
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qfsnd_pcm_hw_params_set_period_size_near (pcm, hw, frag_size, 0);
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err = qfsnd_pcm_hw_params (pcm, hw);
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if (err < 0) {
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Sys_Printf ("ALSA: unable to install hw params\n");
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goto error;
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}
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qfsnd_pcm_sw_params_current (pcm, sw);
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qfsnd_pcm_sw_params_set_start_threshold (pcm, sw, ~0U);
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qfsnd_pcm_sw_params_set_stop_threshold (pcm, sw, ~0U);
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err = qfsnd_pcm_sw_params (pcm, sw);
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if (err < 0) {
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Sys_Printf ("ALSA: unable to install sw params\n");
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goto error;
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}
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shm = &sn;
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memset ((dma_t *) shm, 0, sizeof (*shm));
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shm->splitbuffer = 0;
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shm->channels = stereo + 1;
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shm->submission_chunk = qfsnd_pcm_hw_params_get_period_size (hw, 0);
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// don't
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// mix less than this #
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shm->samplepos = 0; // in mono samples
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shm->samplebits = bps;
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buffer_size = qfsnd_pcm_hw_params_get_buffer_size (hw);
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shm->samples = buffer_size * shm->channels; // mono samples in buffer
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shm->speed = rate;
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SNDDMA_GetDMAPos ();//XXX sets shm->buffer
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Sys_Printf ("%5d stereo\n", shm->channels - 1);
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Sys_Printf ("%5d samples\n", shm->samples);
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Sys_Printf ("%5d samplepos\n", shm->samplepos);
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Sys_Printf ("%5d samplebits\n", shm->samplebits);
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Sys_Printf ("%5d submission_chunk\n", shm->submission_chunk);
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Sys_Printf ("%5d speed\n", shm->speed);
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Sys_Printf ("0x%x dma buffer\n", (int) shm->buffer);
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snd_inited = 1;
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return 1;
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error:
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qfsnd_pcm_close (pcm);
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return 0;
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}
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static int
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SNDDMA_GetDMAPos (void)
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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 = shm->samples/shm->channels;
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if (!snd_inited)
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return 0;
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qfsnd_pcm_avail_update (pcm);
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qfsnd_pcm_mmap_begin (pcm, &areas, &offset, &nframes);
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offset *= shm->channels;
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nframes *= shm->channels;
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shm->samplepos = offset;
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shm->buffer = areas->addr;//XXX FIXME there's an area per channel
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return shm->samplepos;
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}
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static void
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SNDDMA_Shutdown (void)
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{
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if (snd_inited) {
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qfsnd_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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static void
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SNDDMA_Submit (void)
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{
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int state;
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int count = (*plugin_info_snd_output_data.paintedtime -
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*plugin_info_snd_output_data.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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if (snd_blocked)
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return;
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nframes = count / shm->channels;
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qfsnd_pcm_avail_update (pcm);
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qfsnd_pcm_mmap_begin (pcm, &areas, &offset, &nframes);
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state = qfsnd_pcm_state (pcm);
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switch (state) {
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case SND_PCM_STATE_PREPARED:
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qfsnd_pcm_mmap_commit (pcm, offset, nframes);
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qfsnd_pcm_start (pcm);
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break;
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case SND_PCM_STATE_RUNNING:
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qfsnd_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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static void
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SNDDMA_BlockSound (void)
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{
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if (snd_inited && ++snd_blocked == 1)
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qfsnd_pcm_pause (pcm, 1);
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}
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static void
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SNDDMA_UnblockSound (void)
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{
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if (!snd_inited || !snd_blocked)
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return;
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if (!--snd_blocked)
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qfsnd_pcm_pause (pcm, 0);
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}
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PLUGIN_INFO(snd_output, alsa)
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{
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plugin_info.type = qfp_snd_output;
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plugin_info.api_version = QFPLUGIN_VERSION;
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plugin_info.plugin_version = "0.1";
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plugin_info.description = "ALSA 0.9.x digital output";
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plugin_info.copyright = "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.functions = &plugin_info_funcs;
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plugin_info.data = &plugin_info_data;
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plugin_info_data.general = &plugin_info_general_data;
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plugin_info_data.input = NULL;
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plugin_info_data.snd_output = &plugin_info_snd_output_data;
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plugin_info_funcs.general = &plugin_info_general_funcs;
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plugin_info_funcs.input = NULL;
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plugin_info_funcs.snd_output = &plugin_info_snd_output_funcs;
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plugin_info_general_funcs.p_Init = SNDDMA_Init_Cvars;
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plugin_info_general_funcs.p_Shutdown = NULL;
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plugin_info_snd_output_funcs.pS_O_Init = SNDDMA_Init;
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plugin_info_snd_output_funcs.pS_O_Shutdown = SNDDMA_Shutdown;
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plugin_info_snd_output_funcs.pS_O_GetDMAPos = SNDDMA_GetDMAPos;
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plugin_info_snd_output_funcs.pS_O_Submit = SNDDMA_Submit;
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plugin_info_snd_output_funcs.pS_O_BlockSound = SNDDMA_BlockSound;
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plugin_info_snd_output_funcs.pS_O_UnblockSound = SNDDMA_UnblockSound;
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return &plugin_info;
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}
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