This repository has been archived by the owner on Feb 11, 2021. It is now read-only.
SDL_alsa_audio.c
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/*
SDL - Simple DirectMedia Layer
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Copyright (C) 1997-2004 Sam Lantinga
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This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Library General Public
License as published by the Free Software Foundation; either
version 2 of the License, or (at your option) any later version.
This library 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
Library General Public License for more details.
You should have received a copy of the GNU Library General Public
License along with this library; if not, write to the Free
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
Sam Lantinga
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slouken@libsdl.org
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*/
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#include "SDL_config.h"
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/* Allow access to a raw mixing buffer */
#include <sys/types.h>
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#include <signal.h> /* For kill() */
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#include <dlfcn.h>
#include <errno.h>
#include <string.h>
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#include "SDL_timer.h"
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#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
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#include "SDL_alsa_audio.h"
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/* The tag name used by ALSA audio */
#define DRIVER_NAME "alsa"
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/* The default ALSA audio driver */
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#define DEFAULT_DEVICE "default"
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static int (*ALSA_snd_pcm_open)
(snd_pcm_t **, const char *, snd_pcm_stream_t, int);
static int (*ALSA_snd_pcm_close)(snd_pcm_t * pcm);
static snd_pcm_sframes_t(*ALSA_snd_pcm_writei)
(snd_pcm_t *,const void *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_resume)(snd_pcm_t *);
static int (*ALSA_snd_pcm_prepare)(snd_pcm_t *);
static int (*ALSA_snd_pcm_drain)(snd_pcm_t *);
static const char *(*ALSA_snd_strerror)(int);
static size_t(*ALSA_snd_pcm_hw_params_sizeof)(void);
static size_t(*ALSA_snd_pcm_sw_params_sizeof)(void);
static int (*ALSA_snd_pcm_hw_params_any)(snd_pcm_t *, snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_hw_params_set_access)
(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_access_t);
static int (*ALSA_snd_pcm_hw_params_set_format)
(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_format_t);
static int (*ALSA_snd_pcm_hw_params_set_channels)
(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int);
static int (*ALSA_snd_pcm_hw_params_get_channels)(const snd_pcm_hw_params_t *);
static unsigned int (*ALSA_snd_pcm_hw_params_set_rate_near)
(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int, int *);
static snd_pcm_uframes_t (*ALSA_snd_pcm_hw_params_set_period_size_near)
(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_uframes_t, int *);
static snd_pcm_sframes_t (*ALSA_snd_pcm_hw_params_get_period_size)
(const snd_pcm_hw_params_t *);
static unsigned int (*ALSA_snd_pcm_hw_params_set_periods_near)
(snd_pcm_t *,snd_pcm_hw_params_t *, unsigned int, int *);
static int (*ALSA_snd_pcm_hw_params_get_periods)(snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_hw_params)(snd_pcm_t *, snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_sw_params_current)(snd_pcm_t*, snd_pcm_sw_params_t*);
static int (*ALSA_snd_pcm_sw_params_set_start_threshold)
(snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_sw_params_set_avail_min)
(snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_sw_params)(snd_pcm_t *, snd_pcm_sw_params_t *);
static int (*ALSA_snd_pcm_nonblock)(snd_pcm_t *, int);
#define snd_pcm_hw_params_sizeof ALSA_snd_pcm_hw_params_sizeof
#define snd_pcm_sw_params_sizeof ALSA_snd_pcm_sw_params_sizeof
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#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
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static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC;
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static void *alsa_handle = NULL;
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static int
load_alsa_sym(const char *fn, void **addr)
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#if HAVE_DLVSYM
*addr = dlvsym(alsa_handle, fn, "ALSA_0.9");
if (*addr == NULL)
#endif
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*addr = dlsym(alsa_handle, fn);
if (*addr == NULL) {
return 0;
}
}
return 1;
}
/* cast funcs to char* first, to please GCC's strict aliasing rules. */
#define SDL_ALSA_SYM(x) \
if (!load_alsa_sym(#x, (void **) (char *) &ALSA_##x)) return -1
#else
#define SDL_ALSA_SYM(x) ALSA_##x = x
#endif
static int load_alsa_syms(void)
{
SDL_ALSA_SYM(snd_pcm_open);
SDL_ALSA_SYM(snd_pcm_close);
SDL_ALSA_SYM(snd_pcm_writei);
SDL_ALSA_SYM(snd_pcm_resume);
SDL_ALSA_SYM(snd_pcm_prepare);
SDL_ALSA_SYM(snd_pcm_drain);
SDL_ALSA_SYM(snd_strerror);
SDL_ALSA_SYM(snd_pcm_hw_params_sizeof);
SDL_ALSA_SYM(snd_pcm_sw_params_sizeof);
SDL_ALSA_SYM(snd_pcm_hw_params_any);
SDL_ALSA_SYM(snd_pcm_hw_params_set_access);
SDL_ALSA_SYM(snd_pcm_hw_params_set_format);
SDL_ALSA_SYM(snd_pcm_hw_params_set_channels);
SDL_ALSA_SYM(snd_pcm_hw_params_get_channels);
SDL_ALSA_SYM(snd_pcm_hw_params_set_rate_near);
SDL_ALSA_SYM(snd_pcm_hw_params_set_period_size_near);
SDL_ALSA_SYM(snd_pcm_hw_params_get_period_size);
SDL_ALSA_SYM(snd_pcm_hw_params_set_periods_near);
SDL_ALSA_SYM(snd_pcm_hw_params_get_periods);
SDL_ALSA_SYM(snd_pcm_hw_params);
SDL_ALSA_SYM(snd_pcm_sw_params_current);
SDL_ALSA_SYM(snd_pcm_sw_params_set_start_threshold);
SDL_ALSA_SYM(snd_pcm_sw_params_set_avail_min);
SDL_ALSA_SYM(snd_pcm_sw_params);
SDL_ALSA_SYM(snd_pcm_nonblock);
return 0;
}
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#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
static int library_load_count = 0;
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static void
UnloadALSALibrary(void)
{
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if ((alsa_handle != NULL) && (--library_load_count == 0)) {
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dlclose(alsa_handle);
alsa_handle = NULL;
}
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static int
LoadALSALibrary(void)
{
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int retval = 0;
if (library_load_count++ == 0) {
alsa_handle = dlopen(alsa_library, RTLD_NOW);
if (alsa_handle == NULL) {
library_load_count--;
retval = -1;
SDL_SetError("ALSA: dlopen('%s') failed: %s\n",
alsa_library, strerror(errno));
} else {
retval = load_alsa_syms();
if (retval < 0) {
UnloadALSALibrary();
}
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}
}
return retval;
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}
#else
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static void
UnloadALSALibrary(void)
{
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static int
LoadALSALibrary(void)
{
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load_alsa_syms();
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#endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */
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static const char *
get_audio_device(int channels)
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{
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const char *device;
device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */
if (device == NULL) {
if (channels == 6)
device = "surround51";
else if (channels == 4)
device = "surround40";
else
device = DEFAULT_DEVICE;
}
return device;
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}
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ALSA_Available(void)
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{
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int available = 0;
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int status;
snd_pcm_t *handle;
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if (LoadALSALibrary() >= 0) {
int status = ALSA_snd_pcm_open(&handle, get_audio_device(2),
SND_PCM_STREAM_PLAYBACK,
SND_PCM_NONBLOCK);
if (status >= 0) {
available = 1;
ALSA_snd_pcm_close(handle);
}
UnloadALSALibrary();
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}
return (available);
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}
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/* This function waits until it is possible to write a full sound buffer */
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ALSA_WaitDevice(_THIS)
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{
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/* Check to see if the thread-parent process is still alive */
{
static int cnt = 0;
/* Note that this only works with thread implementations
that use a different process id for each thread.
*/
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/* Check every 10 loops */
if (this->hidden->parent && (((++cnt) % 10) == 0)) {
if (kill(this->hidden->parent, 0) < 0) {
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this->enabled = 0;
}
}
}
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}
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/* !!! FIXME: is there a channel swizzler in alsalib instead? */
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/*
* http://bugzilla.libsdl.org/show_bug.cgi?id=110
* "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
* and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
*/
#define SWIZ6(T) \
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T *ptr = (T *) this->hidden->mixbuf; \
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const Uint32 count = (this->spec.samples / 6); \
Uint32 i; \
for (i = 0; i < count; i++, ptr += 6) { \
T tmp; \
tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \
tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \
}
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static __inline__ void
swizzle_alsa_channels_6_64bit(_THIS)
{
SWIZ6(Uint64);
}
static __inline__ void
swizzle_alsa_channels_6_32bit(_THIS)
{
SWIZ6(Uint32);
}
static __inline__ void
swizzle_alsa_channels_6_16bit(_THIS)
{
SWIZ6(Uint16);
}
static __inline__ void
swizzle_alsa_channels_6_8bit(_THIS)
{
SWIZ6(Uint8);
}
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#undef SWIZ6
/*
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* Called right before feeding this->hidden->mixbuf to the hardware. Swizzle
* channels from Windows/Mac order to the format alsalib will want.
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static __inline__ void
swizzle_alsa_channels(_THIS)
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{
if (this->spec.channels == 6) {
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const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */
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if (fmtsize == 16)
swizzle_alsa_channels_6_16bit(this);
else if (fmtsize == 8)
swizzle_alsa_channels_6_8bit(this);
else if (fmtsize == 32)
swizzle_alsa_channels_6_32bit(this);
else if (fmtsize == 64)
swizzle_alsa_channels_6_64bit(this);
}
/* !!! FIXME: update this for 7.1 if needed, later. */
}
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ALSA_PlayDevice(_THIS)
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{
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int status;
int sample_len;
signed short *sample_buf;
swizzle_alsa_channels(this);
sample_len = this->spec.samples;
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sample_buf = (signed short *) this->hidden->mixbuf;
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while (sample_len > 0) {
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status = ALSA_snd_pcm_writei(this->hidden->pcm_handle,
sample_buf, sample_len);
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if (status < 0) {
if (status == -EAGAIN) {
SDL_Delay(1);
continue;
}
if (status == -ESTRPIPE) {
do {
SDL_Delay(1);
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status = ALSA_snd_pcm_resume(this->hidden->pcm_handle);
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} while (status == -EAGAIN);
}
if (status < 0) {
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status = ALSA_snd_pcm_prepare(this->hidden->pcm_handle);
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}
if (status < 0) {
/* Hmm, not much we can do - abort */
this->enabled = 0;
return;
}
continue;
}
sample_buf += status * this->spec.channels;
sample_len -= status;
}
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}
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static Uint8 *
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ALSA_GetDeviceBuf(_THIS)
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{
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return (this->hidden->mixbuf);
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}
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ALSA_CloseDevice(_THIS)
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{
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if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
if (this->hidden->pcm_handle) {
ALSA_snd_pcm_drain(this->hidden->pcm_handle);
ALSA_snd_pcm_close(this->hidden->pcm_handle);
this->hidden->pcm_handle = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
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}
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ALSA_OpenDevice(_THIS, const char *devname, int iscapture)
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{
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int status = 0;
snd_pcm_t *pcm_handle = NULL;
snd_pcm_hw_params_t *hwparams = NULL;
snd_pcm_sw_params_t *swparams = NULL;
snd_pcm_format_t format = 0;
snd_pcm_uframes_t frames = 0;
SDL_AudioFormat test_format = 0;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
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/* Open the audio device */
/* Name of device should depend on # channels in spec */
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status = ALSA_snd_pcm_open(&pcm_handle,
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get_audio_device(this->spec.channels),
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SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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if (status < 0) {
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ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't open audio device: %s",
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ALSA_snd_strerror(status));
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this->hidden->pcm_handle = pcm_handle;
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/* Figure out what the hardware is capable of */
snd_pcm_hw_params_alloca(&hwparams);
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status = ALSA_snd_pcm_hw_params_any(pcm_handle, hwparams);
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if (status < 0) {
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ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't get hardware config: %s",
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ALSA_snd_strerror(status));
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}
/* SDL only uses interleaved sample output */
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status = ALSA_snd_pcm_hw_params_set_access(pcm_handle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
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if (status < 0) {
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ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't set interleaved access: %s",
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ALSA_snd_strerror(status));
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}
/* Try for a closest match on audio format */
status = -1;
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for (test_format = SDL_FirstAudioFormat(this->spec.format);
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test_format && (status < 0);) {
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status = 0; /* if we can't support a format, it'll become -1. */
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switch (test_format) {
case AUDIO_U8:
format = SND_PCM_FORMAT_U8;
break;
case AUDIO_S8:
format = SND_PCM_FORMAT_S8;
break;
case AUDIO_S16LSB:
format = SND_PCM_FORMAT_S16_LE;
break;
case AUDIO_S16MSB:
format = SND_PCM_FORMAT_S16_BE;
break;
case AUDIO_U16LSB:
format = SND_PCM_FORMAT_U16_LE;
break;
case AUDIO_U16MSB:
format = SND_PCM_FORMAT_U16_BE;
break;
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case AUDIO_S32LSB:
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format = SND_PCM_FORMAT_S32_LE;
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break;
case AUDIO_S32MSB:
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format = SND_PCM_FORMAT_S32_BE;
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break;
case AUDIO_F32LSB:
format = SND_PCM_FORMAT_FLOAT_LE;
break;
case AUDIO_F32MSB:
format = SND_PCM_FORMAT_FLOAT_BE;
break;
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if (status >= 0) {
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status = ALSA_snd_pcm_hw_params_set_format(pcm_handle,
hwparams, format);
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}
if (status < 0) {
test_format = SDL_NextAudioFormat();
}
}
if (status < 0) {
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ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't find any hardware audio formats");
return 0;
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this->spec.format = test_format;
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/* Set the number of channels */
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status = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams,
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this->spec.channels);
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if (status < 0) {
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status = ALSA_snd_pcm_hw_params_get_channels(hwparams);
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if ((status <= 0) || (status > 2)) {
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ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't set audio channels");
return 0;
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this->spec.channels = status;
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}
/* Set the audio rate */
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status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams,
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this->spec.freq, NULL);
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if (status < 0) {
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ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't set audio frequency: %s",
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ALSA_snd_strerror(status));
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this->spec.freq = status;
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/* Set the buffer size, in samples */
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frames = this->spec.samples;
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frames = ALSA_snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams,
frames, NULL);
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this->spec.samples = frames;
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ALSA_snd_pcm_hw_params_set_periods_near(pcm_handle, hwparams, 2, NULL);
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/* "set" the hardware with the desired parameters */
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status = ALSA_snd_pcm_hw_params(pcm_handle, hwparams);
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if (status < 0) {
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ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't set hardware audio parameters: %s",
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ALSA_snd_strerror(status));
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#if AUDIO_DEBUG
{
snd_pcm_sframes_t bufsize;
int fragments;
bufsize = ALSA_snd_pcm_hw_params_get_period_size(hwparams);
fragments = ALSA_snd_pcm_hw_params_get_periods(hwparams);
fprintf(stderr,"ALSA: bufsize = %ld, fragments = %d\n",bufsize,fragments);
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}
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/* Set the software parameters */
snd_pcm_sw_params_alloca(&swparams);
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status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams);
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if (status < 0) {
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ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't get software config: %s",
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ALSA_snd_strerror(status));
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status = ALSA_snd_pcm_sw_params_set_start_threshold(pcm_handle,swparams,0);
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if (status < 0) {
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ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't set start threshold: %s",
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ALSA_snd_strerror(status));
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status = ALSA_snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, frames);
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if (status < 0) {
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ALSA_CloseDevice(this);
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SDL_SetError("Couldn't set avail min: %s", ALSA_snd_strerror(status));
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status = ALSA_snd_pcm_sw_params(pcm_handle, swparams);
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if (status < 0) {
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ALSA_CloseDevice(this);
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SDL_SetError("Couldn't set software audio parameters: %s",
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ALSA_snd_strerror(status));
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}
/* Calculate the final parameters for this audio specification */
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SDL_CalculateAudioSpec(&this->spec);
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/* Allocate mixing buffer */
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this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
ALSA_CloseDevice(this);
SDL_OutOfMemory();
return 0;
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SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
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/* Get the parent process id (we're the parent of the audio thread) */
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this->hidden->parent = getpid();
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/* Switch to blocking mode for playback */
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ALSA_snd_pcm_nonblock(pcm_handle, 0);
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/* We're ready to rock and roll. :-) */
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}
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static int
ALSA_Init(SDL_AudioDriverImpl *impl)
{
/* Set the function pointers */
impl->OpenDevice = ALSA_OpenDevice;
impl->WaitDevice = ALSA_WaitDevice;
impl->GetDeviceBuf = ALSA_GetDeviceBuf;
impl->PlayDevice = ALSA_PlayDevice;
impl->CloseDevice = ALSA_CloseDevice;
impl->OnlyHasDefaultOutputDevice = 1; /* !!! FIXME: Add device enum! */
return 1;
}
AudioBootStrap ALSA_bootstrap = {
DRIVER_NAME, "ALSA 0.9 PCM audio",
ALSA_Available, ALSA_Init, 0
};
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/* vi: set ts=4 sw=4 expandtab: */