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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*/
/* 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 "SDL_audio.h"
#include "SDL_audiomem.h"
#include "SDL_audio_c.h"
#include "SDL_timer.h"
#include "SDL_alsa_audio.h"
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#ifdef ALSA_DYNAMIC
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#ifdef USE_DLVSYM
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#ifndef __USE_GNU
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#define __USE_GNU
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#include <dlfcn.h>
#include "SDL_name.h"
#include "SDL_loadso.h"
#else
#define SDL_NAME(X) X
#endif
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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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/* Audio driver functions */
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static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec);
static void ALSA_WaitAudio(_THIS);
static void ALSA_PlayAudio(_THIS);
static Uint8 *ALSA_GetAudioBuf(_THIS);
static void ALSA_CloseAudio(_THIS);
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#ifdef ALSA_DYNAMIC
static const char *alsa_library = ALSA_DYNAMIC;
static void *alsa_handle = NULL;
static int alsa_loaded = 0;
static int (*SDL_snd_pcm_open)(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode);
static int (*SDL_NAME(snd_pcm_open))(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode);
static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm);
static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_writei))(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size);
static int (*SDL_NAME(snd_pcm_resume))(snd_pcm_t *pcm);
static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm);
static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm);
static const char *(*SDL_NAME(snd_strerror))(int errnum);
static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void);
static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
static int (*SDL_NAME(snd_pcm_hw_params_set_access))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t access);
static int (*SDL_NAME(snd_pcm_hw_params_set_format))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val);
static int (*SDL_NAME(snd_pcm_hw_params_set_channels))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val);
static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params);
static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_rate_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir);
static snd_pcm_uframes_t (*SDL_NAME(snd_pcm_hw_params_set_period_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t val, int *dir);
static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_periods_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir);
static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock);
#define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof)
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/* cast funcs to char* first, to please GCC's strict aliasing rules. */
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static struct {
const char *name;
void **func;
} alsa_functions[] = {
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{ "snd_pcm_open", (void**)(char*)&SDL_NAME(snd_pcm_open) },
{ "snd_pcm_close", (void**)(char*)&SDL_NAME(snd_pcm_close) },
{ "snd_pcm_writei", (void**)(char*)&SDL_NAME(snd_pcm_writei) },
{ "snd_pcm_resume", (void**)(char*)&SDL_NAME(snd_pcm_resume) },
{ "snd_pcm_prepare", (void**)(char*)&SDL_NAME(snd_pcm_prepare) },
{ "snd_pcm_drain", (void**)(char*)&SDL_NAME(snd_pcm_drain) },
{ "snd_strerror", (void**)(char*)&SDL_NAME(snd_strerror) },
{ "snd_pcm_hw_params_sizeof", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_sizeof) },
{ "snd_pcm_hw_params_any", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_any) },
{ "snd_pcm_hw_params_set_access", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_access) },
{ "snd_pcm_hw_params_set_format", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_format) },
{ "snd_pcm_hw_params_set_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_channels) },
{ "snd_pcm_hw_params_get_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_channels) },
{ "snd_pcm_hw_params_set_rate_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_rate_near) },
{ "snd_pcm_hw_params_set_period_size_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_period_size_near) },
{ "snd_pcm_hw_params_set_periods_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_periods_near) },
{ "snd_pcm_hw_params", (void**)(char*)&SDL_NAME(snd_pcm_hw_params) },
{ "snd_pcm_nonblock", (void**)(char*)&SDL_NAME(snd_pcm_nonblock) },
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};
static void UnloadALSALibrary(void) {
if (alsa_loaded) {
/* SDL_UnloadObject(alsa_handle);*/
dlclose(alsa_handle);
alsa_handle = NULL;
alsa_loaded = 0;
}
}
static int LoadALSALibrary(void) {
int i, retval = -1;
/* alsa_handle = SDL_LoadObject(alsa_library);*/
alsa_handle = dlopen(alsa_library,RTLD_NOW);
if (alsa_handle) {
alsa_loaded = 1;
retval = 0;
for (i = 0; i < SDL_TABLESIZE(alsa_functions); i++) {
/* *alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);*/
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#ifdef USE_DLVSYM
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*alsa_functions[i].func = dlvsym(alsa_handle,alsa_functions[i].name,"ALSA_0.9");
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if (!*alsa_functions[i].func)
#endif
*alsa_functions[i].func = dlsym(alsa_handle,alsa_functions[i].name);
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if (!*alsa_functions[i].func) {
retval = -1;
UnloadALSALibrary();
break;
}
}
}
return retval;
}
#else
static void UnloadALSALibrary(void) {
return;
}
static int LoadALSALibrary(void) {
return 0;
}
#endif /* 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;
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device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */
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if ( device == NULL ) {
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if (channels == 6) device = "surround51";
else if (channels == 4) device = "surround40";
else device = DEFAULT_DEVICE;
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}
return device;
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}
/* Audio driver bootstrap functions */
static int Audio_Available(void)
{
int available;
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int status;
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snd_pcm_t *handle;
available = 0;
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if (LoadALSALibrary() < 0) {
return available;
}
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status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(2), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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if ( status >= 0 ) {
available = 1;
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SDL_NAME(snd_pcm_close)(handle);
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}
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UnloadALSALibrary();
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return(available);
}
static void Audio_DeleteDevice(SDL_AudioDevice *device)
{
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SDL_free(device->hidden);
SDL_free(device);
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UnloadALSALibrary();
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}
static SDL_AudioDevice *Audio_CreateDevice(int devindex)
{
SDL_AudioDevice *this;
/* Initialize all variables that we clean on shutdown */
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LoadALSALibrary();
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this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
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if ( this ) {
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SDL_memset(this, 0, (sizeof *this));
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this->hidden = (struct SDL_PrivateAudioData *)
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SDL_malloc((sizeof *this->hidden));
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}
if ( (this == NULL) || (this->hidden == NULL) ) {
SDL_OutOfMemory();
if ( this ) {
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SDL_free(this);
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}
return(0);
}
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SDL_memset(this->hidden, 0, (sizeof *this->hidden));
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/* Set the function pointers */
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this->OpenAudio = ALSA_OpenAudio;
this->WaitAudio = ALSA_WaitAudio;
this->PlayAudio = ALSA_PlayAudio;
this->GetAudioBuf = ALSA_GetAudioBuf;
this->CloseAudio = ALSA_CloseAudio;
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this->free = Audio_DeleteDevice;
return this;
}
AudioBootStrap ALSA_bootstrap = {
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DRIVER_NAME, "ALSA 0.9 PCM audio",
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Audio_Available, Audio_CreateDevice
};
/* This function waits until it is possible to write a full sound buffer */
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static void ALSA_WaitAudio(_THIS)
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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.
*/
if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */
if ( kill(parent, 0) < 0 ) {
this->enabled = 0;
}
}
}
}
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static void ALSA_PlayAudio(_THIS)
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{
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int status;
int sample_len;
signed short *sample_buf;
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sample_len = this->spec.samples;
sample_buf = (signed short *)mixbuf;
while ( sample_len > 0 ) {
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status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len);
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if ( status < 0 ) {
if ( status == -EAGAIN ) {
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continue;
}
if ( status == -ESTRPIPE ) {
do {
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status = SDL_NAME(snd_pcm_resume)(pcm_handle);
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} while ( status == -EAGAIN );
}
if ( status < 0 ) {
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status = SDL_NAME(snd_pcm_prepare)(pcm_handle);
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}
if ( status < 0 ) {
/* Hmm, not much we can do - abort */
this->enabled = 0;
return;
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}
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continue;
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}
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sample_buf += status * this->spec.channels;
sample_len -= status;
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}
}
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static Uint8 *ALSA_GetAudioBuf(_THIS)
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{
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return(mixbuf);
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}
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static void ALSA_CloseAudio(_THIS)
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{
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if ( mixbuf != NULL ) {
SDL_FreeAudioMem(mixbuf);
mixbuf = NULL;
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}
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if ( pcm_handle ) {
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SDL_NAME(snd_pcm_drain)(pcm_handle);
SDL_NAME(snd_pcm_close)(pcm_handle);
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pcm_handle = NULL;
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}
}
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static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
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{
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int status;
snd_pcm_hw_params_t *params;
snd_pcm_format_t format;
snd_pcm_uframes_t frames;
Uint16 test_format;
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/* Open the audio device */
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/* Name of device should depend on # channels in spec */
status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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if ( status < 0 ) {
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SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status));
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return(-1);
}
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/* Figure out what the hardware is capable of */
snd_pcm_hw_params_alloca(¶ms);
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status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, params);
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if ( status < 0 ) {
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SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status));
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ALSA_CloseAudio(this);
return(-1);
}
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/* SDL only uses interleaved sample output */
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status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
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if ( status < 0 ) {
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SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status));
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ALSA_CloseAudio(this);
return(-1);
}
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/* Try for a closest match on audio format */
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status = -1;
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for ( test_format = SDL_FirstAudioFormat(spec->format);
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test_format && (status < 0); ) {
switch ( test_format ) {
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case AUDIO_U8:
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format = SND_PCM_FORMAT_U8;
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break;
case AUDIO_S8:
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format = SND_PCM_FORMAT_S8;
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break;
case AUDIO_S16LSB:
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format = SND_PCM_FORMAT_S16_LE;
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break;
case AUDIO_S16MSB:
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format = SND_PCM_FORMAT_S16_BE;
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break;
case AUDIO_U16LSB:
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format = SND_PCM_FORMAT_U16_LE;
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break;
case AUDIO_U16MSB:
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format = SND_PCM_FORMAT_U16_BE;
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break;
default:
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format = 0;
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break;
}
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if ( format != 0 ) {
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status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, params, format);
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}
if ( status < 0 ) {
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test_format = SDL_NextAudioFormat();
}
}
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if ( status < 0 ) {
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SDL_SetError("Couldn't find any hardware audio formats");
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ALSA_CloseAudio(this);
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return(-1);
}
spec->format = test_format;
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/* Set the number of channels */
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status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, params, spec->channels);
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if ( status < 0 ) {
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status = SDL_NAME(snd_pcm_hw_params_get_channels)(params);
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if ( (status <= 0) || (status > 2) ) {
SDL_SetError("Couldn't set audio channels");
ALSA_CloseAudio(this);
return(-1);
}
spec->channels = status;
}
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/* Set the audio rate */
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status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, params, spec->freq, NULL);
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if ( status < 0 ) {
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SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status));
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ALSA_CloseAudio(this);
return(-1);
}
spec->freq = status;
/* Set the buffer size, in samples */
frames = spec->samples;
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frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, params, frames, NULL);
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spec->samples = frames;
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SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, params, 2, NULL);
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/* "set" the hardware with the desired parameters */
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status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, params);
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if ( status < 0 ) {
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SDL_SetError("Couldn't set audio parameters: %s", SDL_NAME(snd_strerror)(status));
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ALSA_CloseAudio(this);
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return(-1);
}
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/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(spec);
/* Allocate mixing buffer */
mixlen = spec->size;
mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
if ( mixbuf == NULL ) {
ALSA_CloseAudio(this);
return(-1);
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}
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SDL_memset(mixbuf, spec->silence, spec->size);
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/* Get the parent process id (we're the parent of the audio thread) */
parent = getpid();
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/* Switch to blocking mode for playback */
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SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0);
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/* We're ready to rock and roll. :-) */
return(0);
}