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. :/
//
//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);
int (*psnd_config_update_free_global)(void);
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);
int (*psnd_pcm_set_params) (snd_pcm_t *pcm, snd_pcm_format_t format, snd_pcm_access_t access, unsigned int channels, unsigned int rate, int soft_resample, unsigned int latency);
snd_pcm_sframes_t (*psnd_pcm_avail_update) (snd_pcm_t *pcm);
snd_pcm_state_t (*psnd_pcm_state) (snd_pcm_t *pcm);
int (*psnd_pcm_start) (snd_pcm_t *pcm);
int (*psnd_pcm_recover) (snd_pcm_t *pcm, int err, int silent);
size_t (*psnd_pcm_hw_params_sizeof) (void);
size_t (*psnd_pcm_sw_params_sizeof) (void);
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_sframes_t (*psnd_pcm_writei) (snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size);
int (*psnd_pcm_prepare) (snd_pcm_t *pcm);
int (*psnd_device_name_hint) (int card, const char *iface, void ***hints);
char * (*psnd_device_name_get_hint) (const void *hint, const char *id);
int (*psnd_device_name_free_hint) (void **hints);
static unsigned int ALSA_MMap_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_MMap_Submit (soundcardinfo_t *sc, int start, int end)
{
int state;
int count = end - start;
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 unsigned int ALSA_RW_GetDMAPos (soundcardinfo_t *sc)
{
int frames;
frames = psnd_pcm_avail_update(sc->handle);
if (frames < 0)
{
psnd_pcm_start (sc->handle);
psnd_pcm_recover(sc->handle, frames, true);
frames = psnd_pcm_avail_update(sc->handle);
}
if (frames >= 0)
{
sc->sn.samplepos = (sc->snd_sent + frames) * sc->sn.numchannels;
}
return sc->sn.samplepos;
}
static void ALSA_RW_Submit (soundcardinfo_t *sc, int start, int end)
{
// int state;
unsigned int frames, offset, ringsize;
unsigned chunk;
int result;
int stride = sc->sn.numchannels * sc->sn.samplebytes;
while(1)
{
/*we can't change the data that was already written*/
frames = end - sc->snd_sent;
if (frames <= 0)
return;
// state = psnd_pcm_state (sc->handle);
ringsize = sc->sn.samples / sc->sn.numchannels;
chunk = frames;
offset = sc->snd_sent % ringsize;
if (offset + chunk >= ringsize)
chunk = ringsize - offset;
result = psnd_pcm_writei(sc->handle, sc->sn.buffer + offset*stride, chunk);
if (result < chunk)
{
if (result < 0)
return;
}
sc->snd_sent += chunk;
chunk = frames - chunk;
if (chunk)
{
result = psnd_pcm_writei(sc->handle, sc->sn.buffer, chunk);
if (result > 0)
sc->snd_sent += result;
}
// if (state == SND_PCM_STATE_PREPARED)
// psnd_pcm_start (sc->handle);
};
}
static void ALSA_Shutdown (soundcardinfo_t *sc)
{
psnd_pcm_close (sc->handle);
psnd_config_update_free_global(); //and try to reduce leaks
if (sc->Submit == ALSA_RW_Submit)
free(sc->sn.buffer);
Con_DPrintf("Alsa closed\n");
}
static void *ALSA_LockBuffer(soundcardinfo_t *sc, unsigned int *sampidx)
{
return sc->sn.buffer;
}
static void ALSA_UnlockBuffer(soundcardinfo_t *sc, void *buffer)
{
}
static qboolean Alsa_InitAlsa(void)
{
static qboolean tried;
static qboolean alsaworks;
if (tried)
return alsaworks;
tried = true;
// Try alternative names of libasound, sometimes it is not linked correctly.
alsasharedobject = dlopen("libasound.so.2", RTLD_LAZY|RTLD_LOCAL);
if (!alsasharedobject)
{
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_config_update_free_global = dlsym(alsasharedobject, "snd_config_update_free_global");
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_state = dlsym(alsasharedobject, "snd_pcm_state");
psnd_pcm_start = dlsym(alsasharedobject, "snd_pcm_start");
psnd_pcm_recover = dlsym(alsasharedobject, "snd_pcm_recover");
psnd_pcm_set_params = dlsym(alsasharedobject, "snd_pcm_set_params");
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");
psnd_pcm_mmap_begin = dlsym(alsasharedobject, "snd_pcm_mmap_begin");
psnd_pcm_mmap_commit = dlsym(alsasharedobject, "snd_pcm_mmap_commit");
psnd_pcm_writei = dlsym(alsasharedobject, "snd_pcm_writei");
psnd_pcm_prepare = dlsym(alsasharedobject, "snd_pcm_prepare");
psnd_device_name_hint = dlsym(alsasharedobject, "snd_device_name_hint");
psnd_device_name_get_hint = dlsym(alsasharedobject, "snd_device_name_get_hint");
psnd_device_name_free_hint = dlsym(alsasharedobject, "snd_device_name_free_hint");
alsaworks = psnd_pcm_open
&& psnd_pcm_close
&& psnd_config_update_free_global
&& 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_state
&& psnd_pcm_start
&& psnd_pcm_hw_params_sizeof
&& psnd_pcm_sw_params_sizeof
&& psnd_pcm_hw_params_set_buffer_size_near
&& psnd_pcm_mmap_begin
&& psnd_pcm_mmap_commit
&& psnd_pcm_writei && psnd_pcm_prepare
&& psnd_device_name_hint && psnd_device_name_get_hint && psnd_device_name_free_hint
;
return alsaworks;
}
static qboolean QDECL ALSA_InitCard (soundcardinfo_t *sc, const char *pcmname)
{
snd_pcm_t *pcm;
snd_pcm_uframes_t buffer_size;
int err;
snd_pcm_hw_params_t *hw;
snd_pcm_sw_params_t *sw;
#if 0
int bps, stereo;
unsigned int rate;
snd_pcm_uframes_t frag_size;
#endif
qboolean mmap = false;
if (!Alsa_InitAlsa())
{
Con_Printf(CON_ERROR "Alsa does not appear to be installed or compatible\n");
return false;
}
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.
if (!pcmname)
pcmname = "default";
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 "ALSA Error: open error (%s): %s\n", pcmname, psnd_strerror (err));
return 0;
}
Con_Printf ("ALSA: Using PCM %s.\n", pcmname);
#if 1
if (!sc->sn.sampleformat)
sc->sn.sampleformat = (sc->sn.samplebytes==1)?QSF_U8:QSF_S16;
switch(sc->sn.sampleformat)
{
case QSF_U8: err = SND_PCM_FORMAT_U8; break;
case QSF_S8: err = SND_PCM_FORMAT_S8; break;
case QSF_S16: err = SND_PCM_FORMAT_S16; break;
case QSF_F32: err = SND_PCM_FORMAT_FLOAT; break;
default:
Con_Printf (CON_ERROR "ALSA: unsupported sample format %i\n", sc->sn.sampleformat);
goto error;
}
err = psnd_pcm_set_params(pcm, err, (mmap?SND_PCM_ACCESS_MMAP_INTERLEAVED:SND_PCM_ACCESS_RW_INTERLEAVED), sc->sn.numchannels, sc->sn.speed, true, 0.04*1000000);
if (0 > err)
{
Con_Printf (CON_ERROR "ALSA: error setting params. %s\n", psnd_strerror (err));
goto error;
}
// sc->sn.numchannels = stereo;
// sc->sn.samplepos = 0;
// sc->sn.samplebytes = bps/8;
sc->samplequeue = buffer_size = 2048;
#else
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, mmap?SND_PCM_ACCESS_MMAP_INTERLEAVED:SND_PCM_ACCESS_RW_INTERLEAVED);
if (0 > err)
{
Con_Printf (CON_ERROR "ALSA: Failure to set interleaved PCM access. %s\n", psnd_strerror (err));
goto error;
}
// get sample bit size
bps = sc->sn.samplebytes*8;
{
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.samplebytes = bps/8;
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->sn.speed = rate;
#endif
sc->sn.samples = buffer_size * sc->sn.numchannels; // mono samples in buffer
sc->handle = pcm;
sc->Lock = ALSA_LockBuffer;
sc->Unlock = ALSA_UnlockBuffer;
sc->Shutdown = ALSA_Shutdown;
if (mmap)
{
sc->GetDMAPos = ALSA_MMap_GetDMAPos;
sc->Submit = ALSA_MMap_Submit;
sc->GetDMAPos(sc); // sets shm->buffer
//alsa doesn't seem to like high mixahead values
//(maybe it tells us above somehow...)
//so force it lower
//quake's default of 0.2 was for 10fps rendering anyway
//so force it down to 0.1 which is the default for halflife at least, and should give better latency
{
extern cvar_t _snd_mixahead;
if (_snd_mixahead.value >= 0.2)
{
Con_Printf("Alsa Hack: _snd_mixahead forced lower\n");
_snd_mixahead.value = 0.1;
}
}
}
else
{
sc->GetDMAPos = ALSA_RW_GetDMAPos;
sc->Submit = ALSA_RW_Submit;
sc->samplequeue = sc->sn.samples;
sc->sn.buffer = malloc(sc->sn.samples * sc->sn.samplebytes);
err = psnd_pcm_prepare(pcm);
if (0 > err)
{
Con_Printf (CON_ERROR "ALSA: unable to prepare for use. %s\n", psnd_strerror (err));
goto error;
}
}
return true;
error:
psnd_pcm_close (pcm);
return false;
}
#define SDRVNAME "ALSA"
static qboolean QDECL ALSA_Enumerate(void (QDECL *cb) (const char *drivername, const char *devicecode, const char *readablename))
{
size_t i;
void **hints;
if (Alsa_InitAlsa())
{
if (!psnd_device_name_hint(-1, "pcm", &hints))
{
for (i = 0; hints[i]; i++)
{
char *n = psnd_device_name_get_hint(hints[i], "NAME");
if (n)
{
char *t = psnd_device_name_get_hint(hints[i], "IOID");
if (!t || strcasecmp(t, "Input"))
{
char *d = psnd_device_name_get_hint(hints[i], "DESC");
if (d)
cb(SDRVNAME, n, va("ALSA (%s)", d));
else
cb(SDRVNAME, n, n);
free(d);
}
free(t);
free(n); //dangerous to free things across boundaries.
}
}
psnd_device_name_free_hint(hints);
}
else
return false;
}
return true;
}
sounddriver_t ALSA_Output =
{
SDRVNAME,
ALSA_InitCard,
ALSA_Enumerate
};