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quakeforge/libs/audio/targets/snd_alsa.c

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/*
snd_alsa.c
Support for the ALSA 1.0.1 sound driver
Copyright (C) 1999,2000 contributors of the QuakeForge project
Please see the file "AUTHORS" for a list of contributors
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[] =
"$Id$";
#include <stdio.h>
#include <dlfcn.h>
#include <alsa/asoundlib.h>
#include "QF/cvar.h"
#include "QF/plugin.h"
#include "QF/qargs.h"
#include "QF/sys.h"
#include "snd_render.h"
static int snd_inited;
static int snd_blocked = 0;
static volatile dma_t sn;
static snd_pcm_uframes_t buffer_size;
static void *alsa_handle;
static const char *pcmname = NULL;
static snd_pcm_t *pcm;
static plugin_t plugin_info;
static plugin_data_t plugin_info_data;
static plugin_funcs_t plugin_info_funcs;
static general_data_t plugin_info_general_data;
static general_funcs_t plugin_info_general_funcs;
static snd_output_data_t plugin_info_snd_output_data;
static snd_output_funcs_t plugin_info_snd_output_funcs;
static cvar_t *snd_bits;
static cvar_t *snd_device;
static cvar_t *snd_rate;
static cvar_t *snd_stereo;
#define QF_ALSA_NEED(ret, func, params) \
static ret (*qf##func) params;
#include "alsa_funcs_list.h"
#undef QF_ALSA_NEED
static qboolean
load_libasound (void)
{
if (!(alsa_handle = dlopen ("libasound.so.2", RTLD_GLOBAL | RTLD_NOW))) {
Sys_Printf ("Couldn't load libasound.so.2: %s\n", dlerror ());
return false;
}
#define QF_ALSA_NEED(ret, func, params) \
if (!(qf##func = dlsym (alsa_handle, #func))) { \
Sys_Printf ("Couldn't load ALSA function %s\n", #func); \
dlclose (alsa_handle); \
alsa_handle = 0; \
return false; \
}
#include "alsa_funcs_list.h"
#undef QF_ALSA_NEED
return true;
}
#define snd_pcm_hw_params_sizeof qfsnd_pcm_hw_params_sizeof
#define snd_pcm_sw_params_sizeof qfsnd_pcm_sw_params_sizeof
static void
SNDDMA_Init_Cvars (void)
{
snd_stereo = Cvar_Get ("snd_stereo", "1", CVAR_ROM, NULL,
"sound stereo output");
snd_rate = Cvar_Get ("snd_rate", "0", CVAR_ROM, NULL,
"sound playback rate. 0 is system default");
snd_device = Cvar_Get ("snd_device", "", CVAR_ROM, NULL,
"sound device. \"\" is system default");
snd_bits = Cvar_Get ("snd_bits", "0", CVAR_ROM, NULL,
"sound sample depth. 0 is system default");
}
static int SNDDMA_GetDMAPos (void);
static snd_pcm_uframes_t
round_buffer_size (snd_pcm_uframes_t sz)
{
snd_pcm_uframes_t mask = ~0;
while (sz & mask) {
sz &= mask;
mask <<= 1;
}
return sz;
}
static volatile dma_t *
SNDDMA_Init (void)
{
int err;
int bps = -1, stereo = -1;
unsigned int rate = 0;
snd_pcm_hw_params_t *hw;
snd_pcm_hw_params_t **_hw = &hw;
snd_pcm_sw_params_t *sw;
snd_pcm_sw_params_t **_sw = &sw;
snd_pcm_uframes_t frag_size;
if (!load_libasound ())
return false;
snd_pcm_hw_params_alloca (_hw);
snd_pcm_sw_params_alloca (_sw);
if (snd_device->string[0])
pcmname = snd_device->string;
if (snd_bits->int_val) {
bps = snd_bits->int_val;
if (bps != 16 && bps != 8) {
Sys_Printf ("Error: invalid sample bits: %d\n", bps);
return 0;
}
}
if (snd_rate->int_val) {
rate = snd_rate->int_val;
if (rate != 44100 && rate != 22050 && rate != 11025) {
Sys_Printf ("Error: invalid sample rate: %d\n", rate);
return 0;
}
}
stereo = snd_stereo->int_val;
if (!pcmname)
pcmname = "default";
err = qfsnd_pcm_open (&pcm, pcmname, SND_PCM_STREAM_PLAYBACK,
SND_PCM_NONBLOCK);
if (0 > err) {
Sys_Printf ("Error: audio open error: %s\n", qfsnd_strerror (err));
return 0;
}
Sys_Printf ("Using PCM %s.\n", pcmname);
err = qfsnd_pcm_hw_params_any (pcm, hw);
if (0 > err) {
Sys_Printf ("ALSA: error setting hw_params_any. %s\n",
qfsnd_strerror (err));
goto error;
}
err = qfsnd_pcm_hw_params_set_access (pcm, hw,
SND_PCM_ACCESS_MMAP_INTERLEAVED);
if (0 > err) {
Sys_Printf ("ALSA: Failure to set noninterleaved PCM access. %s\n"
"Note: Interleaved is not supported\n",
qfsnd_strerror (err));
goto error;
}
switch (bps) {
case -1:
err = qfsnd_pcm_hw_params_set_format (pcm, hw,
SND_PCM_FORMAT_S16_LE);
if (0 <= err) {
bps = 16;
} else if (0 <= (err = qfsnd_pcm_hw_params_set_format (pcm, hw,
SND_PCM_FORMAT_U8))) {
bps = 8;
} else {
Sys_Printf ("ALSA: no useable formats. %s\n",
qfsnd_strerror (err));
goto error;
}
break;
case 8:
case 16:
err = qfsnd_pcm_hw_params_set_format (pcm, hw, bps == 8 ?
SND_PCM_FORMAT_U8 :
SND_PCM_FORMAT_S16);
if (0 > err) {
Sys_Printf ("ALSA: no usable formats. %s\n",
qfsnd_strerror (err));
goto error;
}
break;
default:
Sys_Printf ("ALSA: desired format not supported\n");
goto error;
}
switch (stereo) {
case -1:
err = qfsnd_pcm_hw_params_set_channels (pcm, hw, 2);
if (0 <= err) {
stereo = 1;
} else if (0 <= (err = qfsnd_pcm_hw_params_set_channels (pcm, hw,
1))) {
stereo = 0;
} else {
Sys_Printf ("ALSA: no usable channels. %s\n",
qfsnd_strerror (err));
goto error;
}
break;
case 0:
case 1:
err = qfsnd_pcm_hw_params_set_channels (pcm, hw, stereo ? 2 : 1);
if (0 > err) {
Sys_Printf ("ALSA: no usable channels. %s\n",
qfsnd_strerror (err));
goto error;
}
break;
default:
Sys_Printf ("ALSA: desired channels not supported\n");
goto error;
}
switch (rate) {
case 0:
rate = 44100;
err = qfsnd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
if (0 <= err) {
frag_size = 32 * bps;
} else {
rate = 22050;
err = qfsnd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
if (0 <= err) {
frag_size = 16 * bps;
} else {
rate = 11025;
err = qfsnd_pcm_hw_params_set_rate_near (pcm, hw, &rate,
0);
if (0 <= err) {
frag_size = 8 * bps;
} else {
Sys_Printf ("ALSA: no usable rates. %s\n",
qfsnd_strerror (err));
goto error;
}
}
}
break;
case 11025:
case 22050:
case 44100:
err = qfsnd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
if (0 > err) {
Sys_Printf ("ALSA: desired rate %i not supported. %s\n", rate,
qfsnd_strerror (err));
goto error;
}
frag_size = 8 * bps * rate / 11025;
break;
default:
Sys_Printf ("ALSA: desired rate %i not supported.\n", rate);
goto error;
}
err = qfsnd_pcm_hw_params_set_period_size_near (pcm, hw, &frag_size, 0);
if (0 > err) {
Sys_Printf ("ALSA: unable to set period size near %i. %s\n",
(int) frag_size, qfsnd_strerror (err));
goto error;
}
err = qfsnd_pcm_hw_params (pcm, hw);
if (0 > err) {
Sys_Printf ("ALSA: unable to install hw params: %s\n",
qfsnd_strerror (err));
goto error;
}
err = qfsnd_pcm_sw_params_current (pcm, sw);
if (0 > err) {
Sys_Printf ("ALSA: unable to determine current sw params. %s\n",
qfsnd_strerror (err));
goto error;
}
err = qfsnd_pcm_sw_params_set_start_threshold (pcm, sw, ~0U);
if (0 > err) {
Sys_Printf ("ALSA: unable to set playback threshold. %s\n",
qfsnd_strerror (err));
goto error;
}
err = qfsnd_pcm_sw_params_set_stop_threshold (pcm, sw, ~0U);
if (0 > err) {
Sys_Printf ("ALSA: unable to set playback stop threshold. %s\n",
qfsnd_strerror (err));
goto error;
}
err = qfsnd_pcm_sw_params (pcm, sw);
if (0 > err) {
Sys_Printf ("ALSA: unable to install sw params. %s\n",
qfsnd_strerror (err));
goto error;
}
memset ((dma_t *) &sn, 0, sizeof (sn));
sn.channels = stereo + 1;
// don't mix less than this in mono samples:
err = qfsnd_pcm_hw_params_get_period_size (hw, (snd_pcm_uframes_t *)
(char *)
&sn.submission_chunk, 0);
if (0 > err) {
Sys_Printf ("ALSA: unable to get period size. %s\n",
qfsnd_strerror (err));
goto error;
}
sn.samplepos = 0;
sn.samplebits = bps;
err = qfsnd_pcm_hw_params_get_buffer_size (hw, &buffer_size);
if (0 > err) {
Sys_Printf ("ALSA: unable to get buffer size. %s\n",
qfsnd_strerror (err));
goto error;
}
if (buffer_size != round_buffer_size (buffer_size)) {
Sys_Printf ("ALSA: WARNING: non-power of 2 buffer size. sound may be unsatisfactory\n");
Sys_Printf ("recommend using either the plughw, or hw devices or adjusting dmix\n");
Sys_Printf ("to have a power of 2 buffer size\n");
}
sn.samples = buffer_size * sn.channels;
sn.speed = rate;
SNDDMA_GetDMAPos (); //XXX sets sn.buffer
Sys_Printf ("%5d stereo\n", sn.channels - 1);
Sys_Printf ("%5d samples\n", sn.samples);
Sys_Printf ("%5d samplepos\n", sn.samplepos);
Sys_Printf ("%5d samplebits\n", sn.samplebits);
Sys_Printf ("%5d submission_chunk\n", sn.submission_chunk);
Sys_Printf ("%5d speed\n", sn.speed);
Sys_Printf ("0x%lx dma buffer\n", (long) sn.buffer);
snd_inited = 1;
return &sn;
error:
qfsnd_pcm_close (pcm);
return 0;
}
static int
SNDDMA_GetDMAPos (void)
{
const snd_pcm_channel_area_t *areas;
snd_pcm_uframes_t offset;
snd_pcm_uframes_t nframes = sn.samples/sn.channels;
qfsnd_pcm_avail_update (pcm);
qfsnd_pcm_mmap_begin (pcm, &areas, &offset, &nframes);
offset *= sn.channels;
nframes *= sn.channels;
sn.samplepos = offset;
sn.buffer = areas->addr; //XXX FIXME there's an area per channel
return sn.samplepos;
}
static void
SNDDMA_Shutdown (void)
{
if (snd_inited) {
qfsnd_pcm_close (pcm);
snd_inited = 0;
}
}
/*
SNDDMA_Submit
Send sound to device if buffer isn't really the dma buffer
*/
static void
SNDDMA_Submit (void)
{
int state;
int count = (*plugin_info_snd_output_data.paintedtime -
*plugin_info_snd_output_data.soundtime);
const snd_pcm_channel_area_t *areas;
snd_pcm_uframes_t nframes;
snd_pcm_uframes_t offset;
if (snd_blocked)
return;
nframes = count / sn.channels;
qfsnd_pcm_avail_update (pcm);
qfsnd_pcm_mmap_begin (pcm, &areas, &offset, &nframes);
state = qfsnd_pcm_state (pcm);
switch (state) {
case SND_PCM_STATE_PREPARED:
qfsnd_pcm_mmap_commit (pcm, offset, nframes);
qfsnd_pcm_start (pcm);
break;
case SND_PCM_STATE_RUNNING:
qfsnd_pcm_mmap_commit (pcm, offset, nframes);
break;
default:
break;
}
}
static void
SNDDMA_BlockSound (void)
{
if (snd_inited && ++snd_blocked == 1)
qfsnd_pcm_pause (pcm, 1);
}
static void
SNDDMA_UnblockSound (void)
{
if (!snd_inited || !snd_blocked)
return;
if (!--snd_blocked)
qfsnd_pcm_pause (pcm, 0);
}
PLUGIN_INFO(snd_output, alsa)
{
plugin_info.type = qfp_snd_output;
plugin_info.api_version = QFPLUGIN_VERSION;
plugin_info.plugin_version = "0.1";
plugin_info.description = "ALSA digital output";
plugin_info.copyright = "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.functions = &plugin_info_funcs;
plugin_info.data = &plugin_info_data;
plugin_info_data.general = &plugin_info_general_data;
plugin_info_data.input = NULL;
plugin_info_data.snd_output = &plugin_info_snd_output_data;
plugin_info_funcs.general = &plugin_info_general_funcs;
plugin_info_funcs.input = NULL;
plugin_info_funcs.snd_output = &plugin_info_snd_output_funcs;
plugin_info_general_funcs.p_Init = SNDDMA_Init_Cvars;
plugin_info_general_funcs.p_Shutdown = NULL;
plugin_info_snd_output_funcs.pS_O_Init = SNDDMA_Init;
plugin_info_snd_output_funcs.pS_O_Shutdown = SNDDMA_Shutdown;
plugin_info_snd_output_funcs.pS_O_GetDMAPos = SNDDMA_GetDMAPos;
plugin_info_snd_output_funcs.pS_O_Submit = SNDDMA_Submit;
plugin_info_snd_output_funcs.pS_O_BlockSound = SNDDMA_BlockSound;
plugin_info_snd_output_funcs.pS_O_UnblockSound = SNDDMA_UnblockSound;
return &plugin_info;
}
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