573 lines
17 KiB
C
Executable File
573 lines
17 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 "quakedef.h"
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#ifdef AUDIO_ALSA
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#include <alsa/asoundlib.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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int (*psnd_config_update_free_global)(void);
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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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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);
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snd_pcm_sframes_t (*psnd_pcm_avail_update) (snd_pcm_t *pcm);
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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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int (*psnd_pcm_recover) (snd_pcm_t *pcm, int err, int silent);
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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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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_sframes_t (*psnd_pcm_writei) (snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size);
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int (*psnd_pcm_prepare) (snd_pcm_t *pcm);
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int (*psnd_device_name_hint) (int card, const char *iface, void ***hints);
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char * (*psnd_device_name_get_hint) (const void *hint, const char *id);
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int (*psnd_device_name_free_hint) (void **hints);
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static unsigned int ALSA_MMap_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_MMap_Submit (soundcardinfo_t *sc, int start, int end)
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{
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int state;
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int count = end - start;
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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 unsigned int ALSA_RW_GetDMAPos (soundcardinfo_t *sc)
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{
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int frames;
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frames = psnd_pcm_avail_update(sc->handle);
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if (frames < 0)
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{
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psnd_pcm_start (sc->handle);
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psnd_pcm_recover(sc->handle, frames, true);
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frames = psnd_pcm_avail_update(sc->handle);
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}
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if (frames >= 0)
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{
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sc->sn.samplepos = (sc->snd_sent + frames) * sc->sn.numchannels;
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}
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return sc->sn.samplepos;
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}
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static void ALSA_RW_Submit (soundcardinfo_t *sc, int start, int end)
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{
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// int state;
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unsigned int frames, offset, ringsize;
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unsigned chunk;
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int result;
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int stride = sc->sn.numchannels * sc->sn.samplebytes;
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while(1)
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{
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/*we can't change the data that was already written*/
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frames = end - sc->snd_sent;
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if (frames <= 0)
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return;
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// state = psnd_pcm_state (sc->handle);
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ringsize = sc->sn.samples / sc->sn.numchannels;
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chunk = frames;
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offset = sc->snd_sent % ringsize;
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if (offset + chunk >= ringsize)
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chunk = ringsize - offset;
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result = psnd_pcm_writei(sc->handle, sc->sn.buffer + offset*stride, chunk);
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if (result < chunk)
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{
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if (result < 0)
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return;
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}
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sc->snd_sent += chunk;
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chunk = frames - chunk;
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if (chunk)
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{
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result = psnd_pcm_writei(sc->handle, sc->sn.buffer, chunk);
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if (result > 0)
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sc->snd_sent += result;
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}
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// if (state == SND_PCM_STATE_PREPARED)
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// psnd_pcm_start (sc->handle);
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};
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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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psnd_config_update_free_global(); //and try to reduce leaks
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if (sc->Submit == ALSA_RW_Submit)
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free(sc->sn.buffer);
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Con_DPrintf("Alsa closed\n");
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}
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static void *ALSA_LockBuffer(soundcardinfo_t *sc, unsigned int *sampidx)
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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 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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static dllfunction_t funcs[] =
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{
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{(void**)&psnd_pcm_open, "snd_pcm_open"},
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{(void**)&psnd_pcm_close, "snd_pcm_close"},
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{(void**)&psnd_config_update_free_global, "snd_config_update_free_global"},
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{(void**)&psnd_strerror, "snd_strerror"},
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{(void**)&psnd_pcm_hw_params_any, "snd_pcm_hw_params_any"},
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{(void**)&psnd_pcm_hw_params_set_access, "snd_pcm_hw_params_set_access"},
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{(void**)&psnd_pcm_hw_params_set_format, "snd_pcm_hw_params_set_format"},
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{(void**)&psnd_pcm_hw_params_set_channels, "snd_pcm_hw_params_set_channels"},
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{(void**)&psnd_pcm_hw_params_set_rate_near, "snd_pcm_hw_params_set_rate_near"},
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{(void**)&psnd_pcm_hw_params_set_period_size_near, "snd_pcm_hw_params_set_period_size_near"},
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{(void**)&psnd_pcm_hw_params, "snd_pcm_hw_params"},
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{(void**)&psnd_pcm_sw_params_current, "snd_pcm_sw_params_current"},
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{(void**)&psnd_pcm_sw_params_set_start_threshold, "snd_pcm_sw_params_set_start_threshold"},
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{(void**)&psnd_pcm_sw_params_set_stop_threshold, "snd_pcm_sw_params_set_stop_threshold"},
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{(void**)&psnd_pcm_sw_params, "snd_pcm_sw_params"},
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{(void**)&psnd_pcm_hw_params_get_buffer_size, "snd_pcm_hw_params_get_buffer_size"},
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{(void**)&psnd_pcm_avail_update, "snd_pcm_avail_update"},
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{(void**)&psnd_pcm_state, "snd_pcm_state"},
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{(void**)&psnd_pcm_start, "snd_pcm_start"},
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{(void**)&psnd_pcm_recover, "snd_pcm_recover"},
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{(void**)&psnd_pcm_set_params, "snd_pcm_set_params"},
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{(void**)&psnd_pcm_hw_params_sizeof, "snd_pcm_hw_params_sizeof"},
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{(void**)&psnd_pcm_sw_params_sizeof, "snd_pcm_sw_params_sizeof"},
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{(void**)&psnd_pcm_hw_params_set_buffer_size_near, "snd_pcm_hw_params_set_buffer_size_near"},
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{(void**)&psnd_pcm_mmap_begin, "snd_pcm_mmap_begin"},
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{(void**)&psnd_pcm_mmap_commit, "snd_pcm_mmap_commit"},
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{(void**)&psnd_pcm_writei, "snd_pcm_writei"},
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{(void**)&psnd_pcm_prepare, "snd_pcm_prepare"},
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{(void**)&psnd_device_name_hint, "snd_device_name_hint"},
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{(void**)&psnd_device_name_get_hint, "snd_device_name_get_hint"},
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{(void**)&psnd_device_name_free_hint, "snd_device_name_free_hint"},
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{NULL,NULL}
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};
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if (tried)
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return alsaworks;
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tried = true;
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//pulseaudio's wrapper library fucks with alsa in bad ways, making it unusable on some systems.
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if (COM_CheckParm("-noalsa"))
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return false;
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// Try alternative names of libasound, sometimes it is not linked correctly.
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alsasharedobject = Sys_LoadLibrary("libasound.so.2", funcs);
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if (!alsasharedobject)
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alsasharedobject = Sys_LoadLibrary("libasound.so", funcs);
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if (!alsasharedobject)
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return false;
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alsaworks = true;
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return alsaworks;
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}
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static qboolean QDECL ALSA_InitCard (soundcardinfo_t *sc, const char *pcmname)
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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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int err;
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snd_pcm_hw_params_t *hw;
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snd_pcm_sw_params_t *sw;
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#if 0
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int bps, stereo;
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unsigned int rate;
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snd_pcm_uframes_t frag_size;
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#endif
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qboolean mmap = false;
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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 compatible\n");
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return false;
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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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if (!pcmname)
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pcmname = "default";
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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 "ALSA Error: open error (%s): %s\n", pcmname, 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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#if 1
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if (!sc->sn.sampleformat)
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{
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if (sc->sn.samplebytes >= 4)
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sc->sn.sampleformat = QSF_F32;
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else if (sc->sn.samplebytes != 1)
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sc->sn.sampleformat = QSF_S16;
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else
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sc->sn.sampleformat = QSF_U8;
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}
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switch(sc->sn.sampleformat)
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{
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case QSF_U8: err = SND_PCM_FORMAT_U8; sc->sn.samplebytes=1; break;
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case QSF_S8: err = SND_PCM_FORMAT_S8; sc->sn.samplebytes=1; break;
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case QSF_S16: err = SND_PCM_FORMAT_S16; sc->sn.samplebytes=2; break;
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case QSF_F32: err = SND_PCM_FORMAT_FLOAT; sc->sn.samplebytes=4; break;
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default:
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Con_Printf (CON_ERROR "ALSA: unsupported sample format %i\n", sc->sn.sampleformat);
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goto error;
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}
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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);
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if (0 > err)
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{
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Con_Printf (CON_ERROR "ALSA: error setting params. %s\n", 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.samplebytes = bps/8;
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sc->samplequeue = buffer_size = 2048;
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#else
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err = psnd_pcm_hw_params_any (pcm, hw);
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if (0 > err)
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{
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Con_Printf (CON_ERROR "ALSA: error setting hw_params_any. %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_access (pcm, hw, mmap?SND_PCM_ACCESS_MMAP_INTERLEAVED:SND_PCM_ACCESS_RW_INTERLEAVED);
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if (0 > err)
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{
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Con_Printf (CON_ERROR "ALSA: Failure to set interleaved PCM access. %s\n", 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.samplebytes*8;
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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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{
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Con_Printf (CON_ERROR "ALSA: unable to set period size near %i. %s\n", (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));
|
|
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
|
|
};
|
|
#endif
|