fteqw/engine/client/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
*/
//actually stolen from darkplaces.
//I guess noone can be arsed to write it themselves. :/
#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);
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");
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;
return alsaworks;
}
static int ALSA_InitCard (soundcardinfo_t *sc, int cardnum)
{
snd_pcm_t *pcm;
snd_pcm_uframes_t buffer_size;
extern cvar_t snd_speakers;
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(S_ERROR "Alsa does not appear to be installed or compatable\n");
return 2;
}
hw = alloca(psnd_pcm_hw_params_sizeof());
sw = alloca(psnd_pcm_sw_params_sizeof());
devname = Cvar_Get(va("snd_alsadevice%i", cardnum+1), cardnum==0?"default":"", 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 (S_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 (S_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 (S_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
sc->sn.samplebits = 16; // TODO: this should be changable by a cvar
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 (S_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 = (int)snd_speakers.value;
if (stereo > 6) // limit channels to 6 (engine limit)
stereo = 6;
if (!stereo)
stereo = 2;
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 (S_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 (S_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 (S_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 (S_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 (S_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 (S_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 (S_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 (S_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;
err = psnd_pcm_hw_params_get_buffer_size (hw, &buffer_size);
if (0 > err) {
Con_Printf ("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;