faac72835c
git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@2686 fc73d0e0-1445-4013-8a0c-d673dee63da5
415 lines
13 KiB
C
Executable file
415 lines
13 KiB
C
Executable file
/*
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snd_alsa.c
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Support for the ALSA 1.0.1 sound driver
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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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//actually stolen from darkplaces.
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//I guess noone can be arsed to write it themselves. :/
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//
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//This file is otherwise known as 'will the linux jokers please stop fucking over the open sound system please'
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#include <alsa/asoundlib.h>
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#include "quakedef.h"
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#include <dlfcn.h>
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static void *alsasharedobject;
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int (*psnd_pcm_open) (snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode);
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int (*psnd_pcm_close) (snd_pcm_t *pcm);
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const char *(*psnd_strerror) (int errnum);
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int (*psnd_pcm_hw_params_any) (snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
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int (*psnd_pcm_hw_params_set_access) (snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t _access);
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int (*psnd_pcm_hw_params_set_format) (snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val);
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int (*psnd_pcm_hw_params_set_channels) (snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val);
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int (*psnd_pcm_hw_params_set_rate_near) (snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
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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);
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int (*psnd_pcm_hw_params) (snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
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int (*psnd_pcm_sw_params_current) (snd_pcm_t *pcm, snd_pcm_sw_params_t *params);
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int (*psnd_pcm_sw_params_set_start_threshold) (snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);
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int (*psnd_pcm_sw_params_set_stop_threshold) (snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);
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int (*psnd_pcm_sw_params) (snd_pcm_t *pcm, snd_pcm_sw_params_t *params);
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int (*psnd_pcm_hw_params_get_buffer_size) (const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val);
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int (*psnd_pcm_hw_params_set_buffer_size_near) (snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val);
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snd_pcm_sframes_t (*psnd_pcm_avail_update) (snd_pcm_t *pcm);
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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);
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snd_pcm_sframes_t (*psnd_pcm_mmap_commit) (snd_pcm_t *pcm, snd_pcm_uframes_t offset, snd_pcm_uframes_t frames);
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snd_pcm_state_t (*psnd_pcm_state) (snd_pcm_t *pcm);
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int (*psnd_pcm_start) (snd_pcm_t *pcm);
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size_t (*psnd_pcm_hw_params_sizeof) (void);
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size_t (*psnd_pcm_sw_params_sizeof) (void);
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static unsigned int ALSA_GetDMAPos (soundcardinfo_t *sc)
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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 = sc->sn.samples / sc->sn.numchannels;
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psnd_pcm_avail_update (sc->handle);
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psnd_pcm_mmap_begin (sc->handle, &areas, &offset, &nframes);
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offset *= sc->sn.numchannels;
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nframes *= sc->sn.numchannels;
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sc->sn.samplepos = offset;
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sc->sn.buffer = areas->addr;
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return sc->sn.samplepos;
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}
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static void ALSA_Shutdown (soundcardinfo_t *sc)
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{
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psnd_pcm_close (sc->handle);
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}
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static void ALSA_Submit (soundcardinfo_t *sc)
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{
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extern int soundtime;
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int state;
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int count = sc->paintedtime - 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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nframes = count / sc->sn.numchannels;
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psnd_pcm_avail_update (sc->handle);
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psnd_pcm_mmap_begin (sc->handle, &areas, &offset, &nframes);
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state = psnd_pcm_state (sc->handle);
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switch (state) {
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case SND_PCM_STATE_PREPARED:
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psnd_pcm_mmap_commit (sc->handle, offset, nframes);
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psnd_pcm_start (sc->handle);
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break;
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case SND_PCM_STATE_RUNNING:
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psnd_pcm_mmap_commit (sc->handle, 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 *ALSA_LockBuffer(soundcardinfo_t *sc)
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{
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return sc->sn.buffer;
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}
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static void ALSA_UnlockBuffer(soundcardinfo_t *sc, void *buffer)
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{
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}
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static void ALSA_SetUnderWater(soundcardinfo_t *sc, qboolean underwater)
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{
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}
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static qboolean Alsa_InitAlsa(void)
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{
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static qboolean tried;
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static qboolean alsaworks;
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if (tried)
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return alsaworks;
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tried = true;
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alsasharedobject = dlopen("libasound.so", RTLD_LAZY|RTLD_LOCAL);
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if (!alsasharedobject)
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{
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return false;
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}
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psnd_pcm_open = dlsym(alsasharedobject, "snd_pcm_open");
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psnd_pcm_close = dlsym(alsasharedobject, "snd_pcm_close");
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psnd_strerror = dlsym(alsasharedobject, "snd_strerror");
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psnd_pcm_hw_params_any = dlsym(alsasharedobject, "snd_pcm_hw_params_any");
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psnd_pcm_hw_params_set_access = dlsym(alsasharedobject, "snd_pcm_hw_params_set_access");
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psnd_pcm_hw_params_set_format = dlsym(alsasharedobject, "snd_pcm_hw_params_set_format");
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psnd_pcm_hw_params_set_channels = dlsym(alsasharedobject, "snd_pcm_hw_params_set_channels");
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psnd_pcm_hw_params_set_rate_near = dlsym(alsasharedobject, "snd_pcm_hw_params_set_rate_near");
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psnd_pcm_hw_params_set_period_size_near = dlsym(alsasharedobject, "snd_pcm_hw_params_set_period_size_near");
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psnd_pcm_hw_params = dlsym(alsasharedobject, "snd_pcm_hw_params");
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psnd_pcm_sw_params_current = dlsym(alsasharedobject, "snd_pcm_sw_params_current");
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psnd_pcm_sw_params_set_start_threshold = dlsym(alsasharedobject, "snd_pcm_sw_params_set_start_threshold");
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psnd_pcm_sw_params_set_stop_threshold = dlsym(alsasharedobject, "snd_pcm_sw_params_set_stop_threshold");
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psnd_pcm_sw_params = dlsym(alsasharedobject, "snd_pcm_sw_params");
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psnd_pcm_hw_params_get_buffer_size = dlsym(alsasharedobject, "snd_pcm_hw_params_get_buffer_size");
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psnd_pcm_avail_update = dlsym(alsasharedobject, "snd_pcm_avail_update");
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psnd_pcm_mmap_begin = dlsym(alsasharedobject, "snd_pcm_mmap_begin");
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psnd_pcm_state = dlsym(alsasharedobject, "snd_pcm_state");
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psnd_pcm_mmap_commit = dlsym(alsasharedobject, "snd_pcm_mmap_commit");
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psnd_pcm_start = dlsym(alsasharedobject, "snd_pcm_start");
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psnd_pcm_hw_params_sizeof = dlsym(alsasharedobject, "snd_pcm_hw_params_sizeof");
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psnd_pcm_sw_params_sizeof = dlsym(alsasharedobject, "snd_pcm_sw_params_sizeof");
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psnd_pcm_hw_params_set_buffer_size_near = dlsym(alsasharedobject, "snd_pcm_hw_params_set_buffer_size_near");
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alsaworks = psnd_pcm_open
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&& psnd_pcm_close
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&& psnd_strerror
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&& psnd_pcm_hw_params_any
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&& psnd_pcm_hw_params_set_access
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&& psnd_pcm_hw_params_set_format
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&& psnd_pcm_hw_params_set_channels
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&& psnd_pcm_hw_params_set_rate_near
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&& psnd_pcm_hw_params_set_period_size_near
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&& psnd_pcm_hw_params
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&& psnd_pcm_sw_params_current
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&& psnd_pcm_sw_params_set_start_threshold
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&& psnd_pcm_sw_params_set_stop_threshold
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&& psnd_pcm_sw_params
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&& psnd_pcm_hw_params_get_buffer_size
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&& psnd_pcm_avail_update
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&& psnd_pcm_mmap_begin
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&& psnd_pcm_state
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&& psnd_pcm_mmap_commit
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&& psnd_pcm_start
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&& psnd_pcm_hw_params_sizeof
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&& psnd_pcm_sw_params_sizeof
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&& psnd_pcm_hw_params_set_buffer_size_near;
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return alsaworks;
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}
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static int ALSA_InitCard (soundcardinfo_t *sc, int cardnum)
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{
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snd_pcm_t *pcm;
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snd_pcm_uframes_t buffer_size;
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soundcardinfo_t *ec; //existing card
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char *pcmname;
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cvar_t *devname;
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int err;
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int bps, stereo;
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unsigned int rate;
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snd_pcm_hw_params_t *hw;
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snd_pcm_sw_params_t *sw;
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snd_pcm_uframes_t frag_size;
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if (!Alsa_InitAlsa())
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{
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Con_Printf(CON_ERROR "Alsa does not appear to be installed or compatable\n");
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return 2;
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}
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hw = alloca(psnd_pcm_hw_params_sizeof());
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sw = alloca(psnd_pcm_sw_params_sizeof());
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memset(sw, 0, psnd_pcm_sw_params_sizeof());
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memset(hw, 0, psnd_pcm_hw_params_sizeof());
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//WARNING: 'default' as the default sucks arse. it adds about a second's worth of lag.
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devname = Cvar_Get(va("snd_alsadevice%i", cardnum+1), cardnum==0?"hw":"", 0, "Sound controls");
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pcmname = devname->string;
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if (!*pcmname)
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return 2;
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for (ec = sndcardinfo; ec; ec = ec->next)
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if (!strcmp(ec->name, pcmname))
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break;
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if (ec)
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return 2; //no more
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sc->inactive_sound = true; //linux sound devices always play sound, even when we're not the active app...
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Con_Printf("Initing ALSA sound device %s\n", pcmname);
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err = psnd_pcm_open (&pcm, pcmname, SND_PCM_STREAM_PLAYBACK,
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SND_PCM_NONBLOCK);
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if (0 > err)
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{
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Con_Printf (CON_ERROR "Error: audio open error: %s\n", psnd_strerror (err));
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return 0;
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}
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Con_Printf ("ALSA: Using PCM %s.\n", pcmname);
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err = psnd_pcm_hw_params_any (pcm, hw);
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if (0 > err) {
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Con_Printf (CON_ERROR "ALSA: error setting hw_params_any. %s\n",
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psnd_strerror (err));
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goto error;
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}
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err = psnd_pcm_hw_params_set_access (pcm, hw, SND_PCM_ACCESS_MMAP_INTERLEAVED);
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if (0 > err)
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{
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Con_Printf (CON_ERROR "ALSA: Failure to set noninterleaved PCM access. %s\n"
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"Note: Interleaved is not supported\n",
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psnd_strerror (err));
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goto error;
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}
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// get sample bit size
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bps = sc->sn.samplebits;
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{
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snd_pcm_format_t spft;
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if (bps == 16)
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spft = SND_PCM_FORMAT_S16;
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else
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spft = SND_PCM_FORMAT_U8;
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err = psnd_pcm_hw_params_set_format (pcm, hw, spft);
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while (err < 0)
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{
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if (spft == SND_PCM_FORMAT_S16)
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{
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bps = 8;
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spft = SND_PCM_FORMAT_U8;
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}
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else
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{
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Con_Printf (CON_ERROR "ALSA: no usable formats. %s\n", psnd_strerror (err));
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goto error;
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}
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err = psnd_pcm_hw_params_set_format (pcm, hw, spft);
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}
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}
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// get speaker channels
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stereo = sc->sn.numchannels;
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err = psnd_pcm_hw_params_set_channels (pcm, hw, stereo);
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while (err < 0)
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{
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if (stereo > 2)
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stereo = 2;
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else if (stereo > 1)
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stereo = 1;
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else
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{
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Con_Printf (CON_ERROR "ALSA: no usable number of channels. %s\n", psnd_strerror (err));
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goto error;
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}
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err = psnd_pcm_hw_params_set_channels (pcm, hw, stereo);
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}
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// get rate
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rate = sc->sn.speed;
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err = psnd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
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while (err < 0)
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{
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if (rate > 48000)
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rate = 48000;
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else if (rate > 44100)
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rate = 44100;
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else if (rate > 22150)
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rate = 22150;
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else if (rate > 11025)
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rate = 11025;
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else if (rate > 800)
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rate = 800;
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else
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{
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Con_Printf (CON_ERROR "ALSA: no usable rates. %s\n", psnd_strerror (err));
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goto error;
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}
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err = psnd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
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}
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if (rate > 11025)
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frag_size = 8 * bps * rate / 11025;
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else
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frag_size = 8 * bps;
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err = psnd_pcm_hw_params_set_period_size_near (pcm, hw, &frag_size, 0);
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if (0 > err) {
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Con_Printf (CON_ERROR "ALSA: unable to set period size near %i. %s\n",
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(int) frag_size, psnd_strerror (err));
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goto error;
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}
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err = psnd_pcm_hw_params (pcm, hw);
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if (0 > err) {
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Con_Printf (CON_ERROR "ALSA: unable to install hw params: %s\n",
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psnd_strerror (err));
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goto error;
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}
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err = psnd_pcm_sw_params_current (pcm, sw);
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if (0 > err) {
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Con_Printf (CON_ERROR "ALSA: unable to determine current sw params. %s\n",
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psnd_strerror (err));
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goto error;
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}
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err = psnd_pcm_sw_params_set_start_threshold (pcm, sw, ~0U);
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if (0 > err) {
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Con_Printf (CON_ERROR "ALSA: unable to set playback threshold. %s\n",
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psnd_strerror (err));
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goto error;
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}
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err = psnd_pcm_sw_params_set_stop_threshold (pcm, sw, ~0U);
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if (0 > err) {
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Con_Printf (CON_ERROR "ALSA: unable to set playback stop threshold. %s\n",
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psnd_strerror (err));
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goto error;
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}
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err = psnd_pcm_sw_params (pcm, sw);
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if (0 > err) {
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Con_Printf (CON_ERROR "ALSA: unable to install sw params. %s\n",
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psnd_strerror (err));
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goto error;
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}
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sc->sn.numchannels = stereo;
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sc->sn.samplepos = 0;
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sc->sn.samplebits = bps;
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buffer_size = sc->sn.samples / stereo;
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if (buffer_size)
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{
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err = psnd_pcm_hw_params_set_buffer_size_near(pcm, hw, &buffer_size);
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if (err < 0)
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{
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Con_Printf (CON_ERROR "ALSA: unable to set buffer size. %s\n", psnd_strerror (err));
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goto error;
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}
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}
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err = psnd_pcm_hw_params_get_buffer_size (hw, &buffer_size);
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if (0 > err) {
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Con_Printf (CON_ERROR "ALSA: unable to get buffer size. %s\n",
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psnd_strerror (err));
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goto error;
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}
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sc->Lock = ALSA_LockBuffer;
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sc->Unlock = ALSA_UnlockBuffer;
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sc->SetWaterDistortion = ALSA_SetUnderWater;
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sc->Submit = ALSA_Submit;
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sc->Shutdown = ALSA_Shutdown;
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sc->GetDMAPos = ALSA_GetDMAPos;
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sc->sn.samples = buffer_size * sc->sn.numchannels; // mono samples in buffer
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sc->sn.speed = rate;
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sc->handle = pcm;
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ALSA_GetDMAPos (sc); // sets shm->buffer
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return true;
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error:
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psnd_pcm_close (pcm);
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return false;
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}
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int (*pALSA_InitCard) (soundcardinfo_t *sc, int cardnum) = &ALSA_InitCard;
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