mirror of
https://github.com/nzp-team/fteqw.git
synced 2024-11-23 20:32:43 +00:00
0f02f12b8b
git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@2717 fc73d0e0-1445-4013-8a0c-d673dee63da5
415 lines
13 KiB
C
Executable file
415 lines
13 KiB
C
Executable file
/*
|
|
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
|
|
|
|
*/
|
|
//actually stolen from darkplaces.
|
|
//I guess noone can be arsed to write it themselves. :/
|
|
//
|
|
//This file is otherwise known as 'will the linux jokers please stop fucking over the open sound system please'
|
|
|
|
#include <alsa/asoundlib.h>
|
|
|
|
#include "quakedef.h"
|
|
#include <dlfcn.h>
|
|
|
|
static void *alsasharedobject;
|
|
|
|
int (*psnd_pcm_open) (snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode);
|
|
int (*psnd_pcm_close) (snd_pcm_t *pcm);
|
|
const char *(*psnd_strerror) (int errnum);
|
|
int (*psnd_pcm_hw_params_any) (snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
|
|
int (*psnd_pcm_hw_params_set_access) (snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t _access);
|
|
int (*psnd_pcm_hw_params_set_format) (snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val);
|
|
int (*psnd_pcm_hw_params_set_channels) (snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val);
|
|
int (*psnd_pcm_hw_params_set_rate_near) (snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
|
|
int (*psnd_pcm_hw_params_set_period_size_near) (snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val, int *dir);
|
|
int (*psnd_pcm_hw_params) (snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
|
|
int (*psnd_pcm_sw_params_current) (snd_pcm_t *pcm, snd_pcm_sw_params_t *params);
|
|
int (*psnd_pcm_sw_params_set_start_threshold) (snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);
|
|
int (*psnd_pcm_sw_params_set_stop_threshold) (snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);
|
|
int (*psnd_pcm_sw_params) (snd_pcm_t *pcm, snd_pcm_sw_params_t *params);
|
|
int (*psnd_pcm_hw_params_get_buffer_size) (const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val);
|
|
int (*psnd_pcm_hw_params_set_buffer_size_near) (snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val);
|
|
snd_pcm_sframes_t (*psnd_pcm_avail_update) (snd_pcm_t *pcm);
|
|
int (*psnd_pcm_mmap_begin) (snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames);
|
|
snd_pcm_sframes_t (*psnd_pcm_mmap_commit) (snd_pcm_t *pcm, snd_pcm_uframes_t offset, snd_pcm_uframes_t frames);
|
|
snd_pcm_state_t (*psnd_pcm_state) (snd_pcm_t *pcm);
|
|
int (*psnd_pcm_start) (snd_pcm_t *pcm);
|
|
|
|
size_t (*psnd_pcm_hw_params_sizeof) (void);
|
|
size_t (*psnd_pcm_sw_params_sizeof) (void);
|
|
|
|
|
|
|
|
static unsigned int ALSA_GetDMAPos (soundcardinfo_t *sc)
|
|
{
|
|
const snd_pcm_channel_area_t *areas;
|
|
snd_pcm_uframes_t offset;
|
|
snd_pcm_uframes_t nframes = sc->sn.samples / sc->sn.numchannels;
|
|
|
|
psnd_pcm_avail_update (sc->handle);
|
|
psnd_pcm_mmap_begin (sc->handle, &areas, &offset, &nframes);
|
|
offset *= sc->sn.numchannels;
|
|
nframes *= sc->sn.numchannels;
|
|
sc->sn.samplepos = offset;
|
|
sc->sn.buffer = areas->addr;
|
|
return sc->sn.samplepos;
|
|
}
|
|
|
|
static void ALSA_Shutdown (soundcardinfo_t *sc)
|
|
{
|
|
psnd_pcm_close (sc->handle);
|
|
}
|
|
|
|
static void ALSA_Submit (soundcardinfo_t *sc)
|
|
{
|
|
extern int soundtime;
|
|
int state;
|
|
int count = sc->paintedtime - soundtime;
|
|
const snd_pcm_channel_area_t *areas;
|
|
snd_pcm_uframes_t nframes;
|
|
snd_pcm_uframes_t offset;
|
|
|
|
nframes = count / sc->sn.numchannels;
|
|
|
|
psnd_pcm_avail_update (sc->handle);
|
|
psnd_pcm_mmap_begin (sc->handle, &areas, &offset, &nframes);
|
|
|
|
state = psnd_pcm_state (sc->handle);
|
|
|
|
switch (state) {
|
|
case SND_PCM_STATE_PREPARED:
|
|
psnd_pcm_mmap_commit (sc->handle, offset, nframes);
|
|
psnd_pcm_start (sc->handle);
|
|
break;
|
|
case SND_PCM_STATE_RUNNING:
|
|
psnd_pcm_mmap_commit (sc->handle, offset, nframes);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void *ALSA_LockBuffer(soundcardinfo_t *sc)
|
|
{
|
|
return sc->sn.buffer;
|
|
}
|
|
|
|
static void ALSA_UnlockBuffer(soundcardinfo_t *sc, void *buffer)
|
|
{
|
|
}
|
|
|
|
static void ALSA_SetUnderWater(soundcardinfo_t *sc, qboolean underwater)
|
|
{
|
|
}
|
|
|
|
static qboolean Alsa_InitAlsa(void)
|
|
{
|
|
static qboolean tried;
|
|
static qboolean alsaworks;
|
|
if (tried)
|
|
return alsaworks;
|
|
tried = true;
|
|
|
|
alsasharedobject = dlopen("libasound.so", RTLD_LAZY|RTLD_LOCAL);
|
|
if (!alsasharedobject)
|
|
{
|
|
return false;
|
|
}
|
|
|
|
|
|
psnd_pcm_open = dlsym(alsasharedobject, "snd_pcm_open");
|
|
psnd_pcm_close = dlsym(alsasharedobject, "snd_pcm_close");
|
|
psnd_strerror = dlsym(alsasharedobject, "snd_strerror");
|
|
psnd_pcm_hw_params_any = dlsym(alsasharedobject, "snd_pcm_hw_params_any");
|
|
psnd_pcm_hw_params_set_access = dlsym(alsasharedobject, "snd_pcm_hw_params_set_access");
|
|
psnd_pcm_hw_params_set_format = dlsym(alsasharedobject, "snd_pcm_hw_params_set_format");
|
|
psnd_pcm_hw_params_set_channels = dlsym(alsasharedobject, "snd_pcm_hw_params_set_channels");
|
|
psnd_pcm_hw_params_set_rate_near = dlsym(alsasharedobject, "snd_pcm_hw_params_set_rate_near");
|
|
psnd_pcm_hw_params_set_period_size_near = dlsym(alsasharedobject, "snd_pcm_hw_params_set_period_size_near");
|
|
psnd_pcm_hw_params = dlsym(alsasharedobject, "snd_pcm_hw_params");
|
|
psnd_pcm_sw_params_current = dlsym(alsasharedobject, "snd_pcm_sw_params_current");
|
|
psnd_pcm_sw_params_set_start_threshold = dlsym(alsasharedobject, "snd_pcm_sw_params_set_start_threshold");
|
|
psnd_pcm_sw_params_set_stop_threshold = dlsym(alsasharedobject, "snd_pcm_sw_params_set_stop_threshold");
|
|
psnd_pcm_sw_params = dlsym(alsasharedobject, "snd_pcm_sw_params");
|
|
psnd_pcm_hw_params_get_buffer_size = dlsym(alsasharedobject, "snd_pcm_hw_params_get_buffer_size");
|
|
psnd_pcm_avail_update = dlsym(alsasharedobject, "snd_pcm_avail_update");
|
|
psnd_pcm_mmap_begin = dlsym(alsasharedobject, "snd_pcm_mmap_begin");
|
|
psnd_pcm_state = dlsym(alsasharedobject, "snd_pcm_state");
|
|
psnd_pcm_mmap_commit = dlsym(alsasharedobject, "snd_pcm_mmap_commit");
|
|
psnd_pcm_start = dlsym(alsasharedobject, "snd_pcm_start");
|
|
psnd_pcm_hw_params_sizeof = dlsym(alsasharedobject, "snd_pcm_hw_params_sizeof");
|
|
psnd_pcm_sw_params_sizeof = dlsym(alsasharedobject, "snd_pcm_sw_params_sizeof");
|
|
psnd_pcm_hw_params_set_buffer_size_near = dlsym(alsasharedobject, "snd_pcm_hw_params_set_buffer_size_near");
|
|
|
|
alsaworks = psnd_pcm_open
|
|
&& psnd_pcm_close
|
|
&& psnd_strerror
|
|
&& psnd_pcm_hw_params_any
|
|
&& psnd_pcm_hw_params_set_access
|
|
&& psnd_pcm_hw_params_set_format
|
|
&& psnd_pcm_hw_params_set_channels
|
|
&& psnd_pcm_hw_params_set_rate_near
|
|
&& psnd_pcm_hw_params_set_period_size_near
|
|
&& psnd_pcm_hw_params
|
|
&& psnd_pcm_sw_params_current
|
|
&& psnd_pcm_sw_params_set_start_threshold
|
|
&& psnd_pcm_sw_params_set_stop_threshold
|
|
&& psnd_pcm_sw_params
|
|
&& psnd_pcm_hw_params_get_buffer_size
|
|
&& psnd_pcm_avail_update
|
|
&& psnd_pcm_mmap_begin
|
|
&& psnd_pcm_state
|
|
&& psnd_pcm_mmap_commit
|
|
&& psnd_pcm_start
|
|
&& psnd_pcm_hw_params_sizeof
|
|
&& psnd_pcm_sw_params_sizeof
|
|
&& psnd_pcm_hw_params_set_buffer_size_near;
|
|
|
|
return alsaworks;
|
|
}
|
|
|
|
static int ALSA_InitCard (soundcardinfo_t *sc, int cardnum)
|
|
{
|
|
snd_pcm_t *pcm;
|
|
snd_pcm_uframes_t buffer_size;
|
|
|
|
soundcardinfo_t *ec; //existing card
|
|
char *pcmname;
|
|
cvar_t *devname;
|
|
|
|
int err;
|
|
int bps, stereo;
|
|
unsigned int rate;
|
|
snd_pcm_hw_params_t *hw;
|
|
snd_pcm_sw_params_t *sw;
|
|
snd_pcm_uframes_t frag_size;
|
|
|
|
if (!Alsa_InitAlsa())
|
|
{
|
|
Con_Printf(CON_ERROR "Alsa does not appear to be installed or compatible\n");
|
|
return 2;
|
|
}
|
|
|
|
hw = alloca(psnd_pcm_hw_params_sizeof());
|
|
sw = alloca(psnd_pcm_sw_params_sizeof());
|
|
memset(sw, 0, psnd_pcm_sw_params_sizeof());
|
|
memset(hw, 0, psnd_pcm_hw_params_sizeof());
|
|
|
|
//WARNING: 'default' as the default sucks arse. it adds about a second's worth of lag.
|
|
devname = Cvar_Get(va("snd_alsadevice%i", cardnum+1), cardnum==0?"hw":"", 0, "Sound controls");
|
|
pcmname = devname->string;
|
|
|
|
if (!*pcmname)
|
|
return 2;
|
|
|
|
for (ec = sndcardinfo; ec; ec = ec->next)
|
|
if (!strcmp(ec->name, pcmname))
|
|
break;
|
|
if (ec)
|
|
return 2; //no more
|
|
|
|
sc->inactive_sound = true; //linux sound devices always play sound, even when we're not the active app...
|
|
|
|
Con_Printf("Initing ALSA sound device %s\n", pcmname);
|
|
|
|
err = psnd_pcm_open (&pcm, pcmname, SND_PCM_STREAM_PLAYBACK,
|
|
SND_PCM_NONBLOCK);
|
|
if (0 > err)
|
|
{
|
|
Con_Printf (CON_ERROR "Error: audio open error: %s\n", psnd_strerror (err));
|
|
return 0;
|
|
}
|
|
Con_Printf ("ALSA: Using PCM %s.\n", pcmname);
|
|
|
|
err = psnd_pcm_hw_params_any (pcm, hw);
|
|
if (0 > err) {
|
|
Con_Printf (CON_ERROR "ALSA: error setting hw_params_any. %s\n",
|
|
psnd_strerror (err));
|
|
goto error;
|
|
}
|
|
|
|
err = psnd_pcm_hw_params_set_access (pcm, hw, SND_PCM_ACCESS_MMAP_INTERLEAVED);
|
|
if (0 > err)
|
|
{
|
|
Con_Printf (CON_ERROR "ALSA: Failure to set noninterleaved PCM access. %s\n"
|
|
"Note: Interleaved is not supported\n",
|
|
psnd_strerror (err));
|
|
goto error;
|
|
}
|
|
|
|
// get sample bit size
|
|
bps = sc->sn.samplebits;
|
|
{
|
|
snd_pcm_format_t spft;
|
|
if (bps == 16)
|
|
spft = SND_PCM_FORMAT_S16;
|
|
else
|
|
spft = SND_PCM_FORMAT_U8;
|
|
|
|
err = psnd_pcm_hw_params_set_format (pcm, hw, spft);
|
|
while (err < 0)
|
|
{
|
|
if (spft == SND_PCM_FORMAT_S16)
|
|
{
|
|
bps = 8;
|
|
spft = SND_PCM_FORMAT_U8;
|
|
}
|
|
else
|
|
{
|
|
Con_Printf (CON_ERROR "ALSA: no usable formats. %s\n", psnd_strerror (err));
|
|
goto error;
|
|
}
|
|
err = psnd_pcm_hw_params_set_format (pcm, hw, spft);
|
|
}
|
|
}
|
|
|
|
// get speaker channels
|
|
stereo = sc->sn.numchannels;
|
|
err = psnd_pcm_hw_params_set_channels (pcm, hw, stereo);
|
|
while (err < 0)
|
|
{
|
|
if (stereo > 2)
|
|
stereo = 2;
|
|
else if (stereo > 1)
|
|
stereo = 1;
|
|
else
|
|
{
|
|
Con_Printf (CON_ERROR "ALSA: no usable number of channels. %s\n", psnd_strerror (err));
|
|
goto error;
|
|
}
|
|
err = psnd_pcm_hw_params_set_channels (pcm, hw, stereo);
|
|
}
|
|
|
|
// get rate
|
|
rate = sc->sn.speed;
|
|
err = psnd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
|
|
while (err < 0)
|
|
{
|
|
if (rate > 48000)
|
|
rate = 48000;
|
|
else if (rate > 44100)
|
|
rate = 44100;
|
|
else if (rate > 22150)
|
|
rate = 22150;
|
|
else if (rate > 11025)
|
|
rate = 11025;
|
|
else if (rate > 800)
|
|
rate = 800;
|
|
else
|
|
{
|
|
Con_Printf (CON_ERROR "ALSA: no usable rates. %s\n", psnd_strerror (err));
|
|
goto error;
|
|
}
|
|
err = psnd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
|
|
}
|
|
|
|
if (rate > 11025)
|
|
frag_size = 8 * bps * rate / 11025;
|
|
else
|
|
frag_size = 8 * bps;
|
|
|
|
err = psnd_pcm_hw_params_set_period_size_near (pcm, hw, &frag_size, 0);
|
|
if (0 > err) {
|
|
Con_Printf (CON_ERROR "ALSA: unable to set period size near %i. %s\n",
|
|
(int) frag_size, psnd_strerror (err));
|
|
goto error;
|
|
}
|
|
err = psnd_pcm_hw_params (pcm, hw);
|
|
if (0 > err) {
|
|
Con_Printf (CON_ERROR "ALSA: unable to install hw params: %s\n",
|
|
psnd_strerror (err));
|
|
goto error;
|
|
}
|
|
err = psnd_pcm_sw_params_current (pcm, sw);
|
|
if (0 > err) {
|
|
Con_Printf (CON_ERROR "ALSA: unable to determine current sw params. %s\n",
|
|
psnd_strerror (err));
|
|
goto error;
|
|
}
|
|
err = psnd_pcm_sw_params_set_start_threshold (pcm, sw, ~0U);
|
|
if (0 > err) {
|
|
Con_Printf (CON_ERROR "ALSA: unable to set playback threshold. %s\n",
|
|
psnd_strerror (err));
|
|
goto error;
|
|
}
|
|
err = psnd_pcm_sw_params_set_stop_threshold (pcm, sw, ~0U);
|
|
if (0 > err) {
|
|
Con_Printf (CON_ERROR "ALSA: unable to set playback stop threshold. %s\n",
|
|
psnd_strerror (err));
|
|
goto error;
|
|
}
|
|
err = psnd_pcm_sw_params (pcm, sw);
|
|
if (0 > err) {
|
|
Con_Printf (CON_ERROR "ALSA: unable to install sw params. %s\n",
|
|
psnd_strerror (err));
|
|
goto error;
|
|
}
|
|
|
|
sc->sn.numchannels = stereo;
|
|
sc->sn.samplepos = 0;
|
|
sc->sn.samplebits = bps;
|
|
|
|
buffer_size = sc->sn.samples / stereo;
|
|
if (buffer_size)
|
|
{
|
|
err = psnd_pcm_hw_params_set_buffer_size_near(pcm, hw, &buffer_size);
|
|
if (err < 0)
|
|
{
|
|
Con_Printf (CON_ERROR "ALSA: unable to set buffer size. %s\n", psnd_strerror (err));
|
|
goto error;
|
|
}
|
|
}
|
|
|
|
err = psnd_pcm_hw_params_get_buffer_size (hw, &buffer_size);
|
|
if (0 > err) {
|
|
Con_Printf (CON_ERROR "ALSA: unable to get buffer size. %s\n",
|
|
psnd_strerror (err));
|
|
goto error;
|
|
}
|
|
|
|
sc->Lock = ALSA_LockBuffer;
|
|
sc->Unlock = ALSA_UnlockBuffer;
|
|
sc->SetWaterDistortion = ALSA_SetUnderWater;
|
|
sc->Submit = ALSA_Submit;
|
|
sc->Shutdown = ALSA_Shutdown;
|
|
sc->GetDMAPos = ALSA_GetDMAPos;
|
|
|
|
sc->sn.samples = buffer_size * sc->sn.numchannels; // mono samples in buffer
|
|
sc->sn.speed = rate;
|
|
sc->handle = pcm;
|
|
ALSA_GetDMAPos (sc); // sets shm->buffer
|
|
|
|
return true;
|
|
|
|
error:
|
|
psnd_pcm_close (pcm);
|
|
return false;
|
|
}
|
|
|
|
int (*pALSA_InitCard) (soundcardinfo_t *sc, int cardnum) = &ALSA_InitCard;
|
|
|
|
|