/*

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 <sys/types.h>

#include "SDL_timer.h"

#include "SDL_audio.h"

#include "../SDL_audiomem.h"

#include "../SDL_audio_c.h"

#include "SDL_alsa_audio.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"

/* 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 void *alsa_handle = NULL;

static int alsa_loaded = 0;

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_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_set_rate_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir);

static int (*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 int (*SDL_NAME(snd_pcm_sw_params_current))(snd_pcm_t *pcm, snd_pcm_sw_params_t *swparams);

static int (*SDL_NAME(snd_pcm_sw_params_set_start_threshold))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);

static int (*SDL_NAME(snd_pcm_sw_params))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params);

static struct {

const char *name;

void **func;

} alsa_functions[] = {

{ "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_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_sw_params_current", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_current) },

{ "snd_pcm_sw_params_set_start_threshold", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_set_start_threshold) },

{ "snd_pcm_sw_params", (void**)(char*)&SDL_NAME(snd_pcm_sw_params) },

};

static void UnloadALSALibrary(void) {

if (alsa_loaded) {

alsa_handle = NULL;

alsa_loaded = 0;

}

}

static int LoadALSALibrary(void) {

int i, retval = -1;

if (alsa_handle) {

alsa_loaded = 1;

retval = 0;

if (!*alsa_functions[i].func) {

retval = -1;

UnloadALSALibrary();

break;

}

}

}

return retval;

}

#else

static void UnloadALSALibrary(void) {

return;

}

static int LoadALSALibrary(void) {

return 0;

}

const char *device;

switch (channels) {

case 6:

device = "plug:surround51";

break;

case 4:

device = "plug:surround40";

break;

default:

device = "default";

break;

}

}

return device;

}

/* Audio driver bootstrap functions */

static int Audio_Available(void)

{

int available;

snd_pcm_t *handle;

available = 0;

if (LoadALSALibrary() < 0) {

return available;

}

if ( status >= 0 ) {

available = 1;

return(available);

}

static void Audio_DeleteDevice(SDL_AudioDevice *device)

{

SDL_free(device->hidden);

SDL_free(device);

}

static SDL_AudioDevice *Audio_CreateDevice(int devindex)

{

SDL_AudioDevice *this;

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

}

if ( (this == NULL) || (this->hidden == NULL) ) {

SDL_OutOfMemory();

if ( this ) {

}

return(0);

}

/* 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;

}

}

}

}

/*

* 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) \

T *ptr = (T *) mixbuf; \

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; \

}

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); }

#undef SWIZ6

/*

* Called right before feeding this->mixbuf to the hardware. Swizzle channels

* from Windows/Mac order to the format alsalib will want.

*/

static __inline__ void swizzle_alsa_channels(_THIS)

{

if (this->spec.channels == 6) {

const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */

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. */

}

swizzle_alsa_channels(this);

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

this->enabled = 0;

return;

}

}

}

if ( mixbuf != NULL ) {

SDL_FreeAudioMem(mixbuf);

mixbuf = NULL;

SDL_NAME(snd_pcm_drain)(pcm_handle);

SDL_NAME(snd_pcm_close)(pcm_handle);

}

}

static int ALSA_finalize_hardware(_THIS, SDL_AudioSpec *spec, snd_pcm_hw_params_t *hwparams, int override)

{

int status;

snd_pcm_uframes_t bufsize;

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

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

if ( status < 0 ) {

return(-1);

}

/* Get samples for the actual buffer size */

status = SDL_NAME(snd_pcm_hw_params_get_buffer_size)(hwparams, &bufsize);

if ( status < 0 ) {

return(-1);

}

if ( !override && bufsize != spec->samples * 2 ) {

return(-1);

}

/* FIXME: Is this safe to do? */

spec->samples = bufsize / 2;

/* This is useful for debugging */

if ( getenv("SDL_AUDIO_ALSA_DEBUG") ) {

snd_pcm_sframes_t persize = 0;

unsigned int periods = 0;

SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams, &persize, NULL);

SDL_NAME(snd_pcm_hw_params_get_periods)(hwparams, &periods, NULL);

fprintf(stderr, "ALSA: period size = %ld, periods = %u, buffer size = %lu\n", persize, periods, bufsize);

}

return(0);

}

static int ALSA_set_period_size(_THIS, SDL_AudioSpec *spec, snd_pcm_hw_params_t *params)

{

const char *env;

int status;

int override = 0;

snd_pcm_hw_params_t *hwparams;

snd_pcm_uframes_t frames;

unsigned int periods;

/* Copy the hardware parameters for this setup */

snd_pcm_hw_params_alloca(&hwparams);

SDL_NAME(snd_pcm_hw_params_copy)(hwparams, params);

env = getenv("SDL_AUDIO_ALSA_SET_PERIOD_SIZE");

if ( env ) {

if ( override == 0 ) {

return(-1);

}

}

frames = spec->samples;

status = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, &frames, NULL);

if ( status < 0 ) {

return(-1);

}

periods = 2;

status = SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, &periods, NULL);

if ( status < 0 ) {

return(-1);

}

return ALSA_finalize_hardware(this, spec, hwparams, override);

}

static int ALSA_set_buffer_size(_THIS, SDL_AudioSpec *spec, snd_pcm_hw_params_t *params)

{

const char *env;

int status;

int override = 0;

snd_pcm_hw_params_t *hwparams;

snd_pcm_uframes_t frames;

/* Copy the hardware parameters for this setup */

snd_pcm_hw_params_alloca(&hwparams);

SDL_NAME(snd_pcm_hw_params_copy)(hwparams, params);

env = getenv("SDL_AUDIO_ALSA_SET_BUFFER_SIZE");

if ( env ) {

if ( override == 0 ) {

return(-1);

}

}

frames = spec->samples * 2;

status = SDL_NAME(snd_pcm_hw_params_set_buffer_size_near)(pcm_handle, hwparams, &frames);

if ( status < 0 ) {

return(-1);

}

return ALSA_finalize_hardware(this, spec, hwparams, override);

}

snd_pcm_hw_params_t *hwparams;

snd_pcm_sw_params_t *swparams;

/* Open the audio device */

/* 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);

return(-1);

}

snd_pcm_hw_params_alloca(&hwparams);

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

ALSA_CloseAudio(this);

return(-1);

}

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 ( status < 0 ) {

test_format = SDL_NextAudioFormat();

}

}

return(-1);

}

spec->format = test_format;

status = SDL_NAME(snd_pcm_hw_params_get_channels)(hwparams, &channels);

if ( status < 0 ) {

SDL_SetError("Couldn't set audio channels");

ALSA_CloseAudio(this);

return(-1);

}

rate = spec->freq;

status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, &rate, NULL);

ALSA_CloseAudio(this);

return(-1);

}

/* Set the buffer size, in samples */

if ( ALSA_set_period_size(this, spec, hwparams) < 0 &&

ALSA_set_buffer_size(this, spec, hwparams) < 0 ) {

/* Failed to set buffer size, try to just use the defaults */

if ( ALSA_finalize_hardware(this, spec, hwparams, 1) < 0 ) {

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

ALSA_CloseAudio(this);

return(-1);

}

/* Set the software parameters */

snd_pcm_sw_params_alloca(&swparams);

status = SDL_NAME(snd_pcm_sw_params_current)(pcm_handle, swparams);

if ( status < 0 ) {

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

ALSA_CloseAudio(this);

return(-1);

}

if ( status < 0 ) {

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

ALSA_CloseAudio(this);

return(-1);

}

status = SDL_NAME(snd_pcm_sw_params)(pcm_handle, swparams);

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);

}