/*

SDL - Simple DirectMedia Layer

Copyright (C) 1997-2004 Sam Lantinga

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

slouken@libsdl.org

*/

/* 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"

#ifdef ALSA_DYNAMIC

#define __USE_GNU

#include <dlfcn.h>

#include "SDL_name.h"

#include "SDL_loadso.h"

#else

#define SDL_NAME(X) X

#endif

/* The tag name used by ALSA audio */

#define DRIVER_NAME "alsa"

/* The default ALSA audio driver */

#define DEFAULT_DEVICE "default"

/* 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);

#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)

static struct {

const char *name;

void **func;

} alsa_functions[] = {

{ "snd_pcm_open", (void**)&SDL_NAME(snd_pcm_open) },

{ "snd_pcm_close", (void**)&SDL_NAME(snd_pcm_close) },

{ "snd_pcm_writei", (void**)&SDL_NAME(snd_pcm_writei) },

{ "snd_pcm_resume", (void**)&SDL_NAME(snd_pcm_resume) },

{ "snd_pcm_prepare", (void**)&SDL_NAME(snd_pcm_prepare) },

{ "snd_pcm_drain", (void**)&SDL_NAME(snd_pcm_drain) },

{ "snd_strerror", (void**)&SDL_NAME(snd_strerror) },

{ "snd_pcm_hw_params_sizeof", (void**)&SDL_NAME(snd_pcm_hw_params_sizeof) },

{ "snd_pcm_hw_params_any", (void**)&SDL_NAME(snd_pcm_hw_params_any) },

{ "snd_pcm_hw_params_set_access", (void**)&SDL_NAME(snd_pcm_hw_params_set_access) },

{ "snd_pcm_hw_params_set_format", (void**)&SDL_NAME(snd_pcm_hw_params_set_format) },

{ "snd_pcm_hw_params_set_channels", (void**)&SDL_NAME(snd_pcm_hw_params_set_channels) },

{ "snd_pcm_hw_params_get_channels", (void**)&SDL_NAME(snd_pcm_hw_params_get_channels) },

{ "snd_pcm_hw_params_set_rate_near", (void**)&SDL_NAME(snd_pcm_hw_params_set_rate_near) },

{ "snd_pcm_hw_params_set_period_size_near", (void**)&SDL_NAME(snd_pcm_hw_params_set_period_size_near) },

{ "snd_pcm_hw_params_set_periods_near", (void**)&SDL_NAME(snd_pcm_hw_params_set_periods_near) },

{ "snd_pcm_hw_params", (void**)&SDL_NAME(snd_pcm_hw_params) },

{ "snd_pcm_nonblock", (void**)&SDL_NAME(snd_pcm_nonblock) },

};

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);*/

*alsa_functions[i].func = dlvsym(alsa_handle,alsa_functions[i].name,"ALSA_0.9");

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 */

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;

int status;

snd_pcm_t *handle;

available = 0;

if (LoadALSALibrary() < 0) {

return available;

}

status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);

if ( status >= 0 ) {

available = 1;

SDL_NAME(snd_pcm_close)(handle);

}

UnloadALSALibrary();

return(available);

}

static void Audio_DeleteDevice(SDL_AudioDevice *device)

{

free(device->hidden);

free(device);

UnloadALSALibrary();

}

static SDL_AudioDevice *Audio_CreateDevice(int devindex)

{

SDL_AudioDevice *this;

/* Initialize all variables that we clean on shutdown */

LoadALSALibrary();

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 = {

DRIVER_NAME, "ALSA 0.9 PCM audio",

Audio_Available, Audio_CreateDevice

};

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

static void ALSA_WaitAudio(_THIS)

{

/* 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;

}

}

}

}

static void ALSA_PlayAudio(_THIS)

{

int status;

int sample_len;

signed short *sample_buf;

sample_len = this->spec.samples;

sample_buf = (signed short *)mixbuf;

while ( sample_len > 0 ) {

status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len);

if ( status < 0 ) {

if ( status == -EAGAIN ) {

SDL_Delay(1);

continue;

}

if ( status == -ESTRPIPE ) {

do {

SDL_Delay(1);

status = SDL_NAME(snd_pcm_resume)(pcm_handle);

} while ( status == -EAGAIN );

}

if ( status < 0 ) {

status = SDL_NAME(snd_pcm_prepare)(pcm_handle);

}

if ( status < 0 ) {

/* Hmm, not much we can do - abort */

this->enabled = 0;

return;

}

continue;

}

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

sample_len -= status;

}

}

static Uint8 *ALSA_GetAudioBuf(_THIS)

{

return(mixbuf);

}

static void ALSA_CloseAudio(_THIS)

{

if ( mixbuf != NULL ) {

SDL_FreeAudioMem(mixbuf);

mixbuf = NULL;

}

if ( pcm_handle ) {

SDL_NAME(snd_pcm_drain)(pcm_handle);

SDL_NAME(snd_pcm_close)(pcm_handle);

pcm_handle = NULL;

}

}

static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)

{

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 = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);

if ( status < 0 ) {

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

return(-1);

}

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

snd_pcm_hw_params_alloca(&params);

status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, params);

if ( status < 0 ) {

SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status));

ALSA_CloseAudio(this);

return(-1);

}

/* SDL only uses interleaved sample output */

status = SDL_NAME(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", SDL_NAME(snd_strerror)(status));

ALSA_CloseAudio(this);

return(-1);

}

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

status = -1;

for ( test_format = SDL_FirstAudioFormat(spec->format);

test_format && (status < 0); ) {

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;

default:

format = 0;

break;

}

if ( format != 0 ) {

status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, params, format);

}

if ( status < 0 ) {

test_format = SDL_NextAudioFormat();

}

}

if ( status < 0 ) {

SDL_SetError("Couldn't find any hardware audio formats");

ALSA_CloseAudio(this);

return(-1);

}

spec->format = test_format;

/* Set the number of channels */

status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, params, spec->channels);

if ( status < 0 ) {

status = SDL_NAME(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 = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, params, spec->freq, NULL);

if ( status < 0 ) {

SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status));

ALSA_CloseAudio(this);

return(-1);

}

spec->freq = status;

/* Set the buffer size, in samples */

frames = spec->samples;

frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, params, frames, NULL);

spec->samples = frames;

SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, params, 2, NULL);

/* "set" the hardware with the desired parameters */

status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, params);

if ( status < 0 ) {

SDL_SetError("Couldn't set audio parameters: %s", SDL_NAME(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);

}

memset(mixbuf, spec->silence, spec->size);

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

parent = getpid();

/* Switch to blocking mode for playback */

SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0);

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

return(0);

}