/*

SDL - Simple DirectMedia Layer

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

*/

/* 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"

/* The default ALSA audio driver */

#define DEFAULT_DEVICE "plughw:0,0"

/* Audio driver functions */

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()

const char *device;

device = getenv("AUDIODEV"); /* Is there a standard variable name? */

if ( device == NULL ) {

device = DEFAULT_DEVICE;

}

return device;

}

/* Audio driver bootstrap functions */

static int Audio_Available(void)

{

int available;

snd_pcm_t *handle;

available = 0;

status = snd_pcm_open(&handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, 0);

if ( status >= 0 ) {

available = 1;

snd_pcm_close(handle);

}

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 */

this->OpenAudio = ALSA_OpenAudio;

this->WaitAudio = ALSA_WaitAudio;

this->PlayAudio = ALSA_PlayAudio;

this->GetAudioBuf = ALSA_GetAudioBuf;

this->CloseAudio = ALSA_CloseAudio;

this->free = Audio_DeleteDevice;

return this;

}

AudioBootStrap ALSA_bootstrap = {

Audio_Available, Audio_CreateDevice

};

/* This function waits until it is possible to write a full sound buffer */

{

/* 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;

}

}

}

}

int status;

int sample_len;

signed short *sample_buf;

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 ) {

continue;

}

if ( status == -ESTRPIPE ) {

do {

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;

sample_buf += status * this->spec.channels;

sample_len -= status;

}

}

}

if ( mixbuf != NULL ) {

SDL_FreeAudioMem(mixbuf);

mixbuf = NULL;

if ( pcm_handle ) {

snd_pcm_close(pcm_handle);

pcm_handle = NULL;

}

}

int status;

snd_pcm_hw_params_t *params;

snd_pcm_format_t format;

snd_pcm_uframes_t frames;

Uint16 test_format;

/* Open the audio device */

status = snd_pcm_open(&pcm_handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, 0);

if ( status < 0 ) {

SDL_SetError("Couldn't open audio device: %s", snd_strerror(status));

return(-1);

}

/* Figure out what the hardware is capable of */

snd_pcm_hw_params_alloca(&params);

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);

}

/* 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);

}

/* Try for a closest match on audio format */

test_format && (status < 0); ) {

switch ( test_format ) {

break;

case AUDIO_S8:

break;

case AUDIO_S16LSB:

break;

case AUDIO_S16MSB:

break;

case AUDIO_U16LSB:

break;

case AUDIO_U16MSB:

break;

default:

break;

}

if ( format != 0 ) {

status = snd_pcm_hw_params_set_format(pcm_handle, params, format);

}

if ( status < 0 ) {

test_format = SDL_NextAudioFormat();

}

}

return(-1);

}

spec->format = test_format;

/* 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;

}

/* 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);

return(-1);

}

/* 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);

/* Get the parent process id (we're the parent of the audio thread) */

parent = getpid();

/* We're ready to rock and roll. :-) */

return(0);

}