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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 <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <errno.h>
#include <unistd.h>
#include <fcntl.h>
#include <signal.h>
#include <sys/types.h>
#include <sys/time.h>
#include "SDL_audio.h"
#include "SDL_error.h"
#include "SDL_audiomem.h"
#include "SDL_audio_c.h"
#include "SDL_timer.h"
#include "SDL_alsa_audio.h"
/* 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);
static const char *get_audio_device()
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{
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const char *device;
device = getenv("AUDIODEV"); /* Is there a standard variable name? */
if ( device == NULL ) {
device = DEFAULT_DEVICE;
}
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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status = snd_pcm_open(&handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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if ( status >= 0 ) {
available = 1;
snd_pcm_close(handle);
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}
return(available);
}
static void Audio_DeleteDevice(SDL_AudioDevice *device)
{
free(device->hidden);
free(device);
}
static SDL_AudioDevice *Audio_CreateDevice(int devindex)
{
SDL_AudioDevice *this;
/* Initialize all variables that we clean on shutdown */
this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice));
if ( this ) {
memset(this, 0, (sizeof *this));
this->hidden = (struct SDL_PrivateAudioData *)
malloc((sizeof *this->hidden));
}
if ( (this == NULL) || (this->hidden == NULL) ) {
SDL_OutOfMemory();
if ( this ) {
free(this);
}
return(0);
}
memset(this->hidden, 0, (sizeof *this->hidden));
/* 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 ) {
status = snd_pcm_writei(pcm_handle, sample_buf, sample_len);
if ( status < 0 ) {
if ( status == -EAGAIN ) {
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SDL_Delay(1);
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continue;
}
if ( status == -ESTRPIPE ) {
do {
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SDL_Delay(1);
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status = snd_pcm_resume(pcm_handle);
} while ( status == -EAGAIN );
}
if ( status < 0 ) {
status = snd_pcm_prepare(pcm_handle);
}
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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snd_pcm_drain(pcm_handle);
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snd_pcm_close(pcm_handle);
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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status = snd_pcm_open(&pcm_handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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if ( status < 0 ) {
SDL_SetError("Couldn't open audio device: %s", 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);
status = snd_pcm_hw_params_any(pcm_handle, params);
if ( status < 0 ) {
SDL_SetError("Couldn't get hardware config: %s", snd_strerror(status));
ALSA_CloseAudio(this);
return(-1);
}
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/* SDL only uses interleaved sample output */
status = snd_pcm_hw_params_set_access(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
if ( status < 0 ) {
SDL_SetError("Couldn't set interleaved access: %s", snd_strerror(status));
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 ) {
status = snd_pcm_hw_params_set_format(pcm_handle, params, format);
}
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 */
status = snd_pcm_hw_params_set_channels(pcm_handle, params, spec->channels);
if ( status < 0 ) {
status = snd_pcm_hw_params_get_channels(params);
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 */
status = snd_pcm_hw_params_set_rate_near(pcm_handle, params, spec->freq, NULL);
if ( status < 0 ) {
SDL_SetError("Couldn't set audio frequency: %s", snd_strerror(status));
ALSA_CloseAudio(this);
return(-1);
}
spec->freq = status;
/* Set the buffer size, in samples */
frames = spec->samples;
frames = snd_pcm_hw_params_set_period_size_near(pcm_handle, params, frames, NULL);
spec->samples = frames;
snd_pcm_hw_params_set_periods_near(pcm_handle, params, 2, NULL);
/* "set" the hardware with the desired parameters */
status = snd_pcm_hw_params(pcm_handle, params);
if ( status < 0 ) {
SDL_SetError("Couldn't set audio parameters: %s", snd_strerror(status));
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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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 */
snd_pcm_nonblock(pcm_handle, 0);
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/* We're ready to rock and roll. :-) */
return(0);
}