/*

SDL - Simple DirectMedia Layer

Copyright (C) 1997-2010 Sam Lantinga

This library is free software; you can redistribute it and/or

modify it under the terms of the GNU Lesser General Public

License as published by the Free Software Foundation; either

version 2.1 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

Lesser General Public License for more details.

You should have received a copy of the GNU Lesser General Public

License along with this library; if not, write to the Free Software

Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA

Sam Lantinga

slouken@libsdl.org

*/

#include "SDL_config.h"

/* Allow access to a raw mixing buffer */

#include <sys/types.h>

#include <signal.h> /* For kill() */

#include <errno.h>

#include <string.h>

#include "SDL_timer.h"

#include "SDL_audio.h"

#include "../SDL_audiomem.h"

#include "../SDL_audio_c.h"

#include "SDL_alsa_audio.h"

#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC

#include "SDL_loadso.h"

#endif

/* The tag name used by ALSA audio */

#define DRIVER_NAME "alsa"

static int (*ALSA_snd_pcm_open)

(snd_pcm_t **, const char *, snd_pcm_stream_t, int);

static int (*ALSA_snd_pcm_close) (snd_pcm_t * pcm);

static snd_pcm_sframes_t(*ALSA_snd_pcm_writei)

(snd_pcm_t *, const void *, snd_pcm_uframes_t);

static int (*ALSA_snd_pcm_recover) (snd_pcm_t *, int, int);

static int (*ALSA_snd_pcm_prepare) (snd_pcm_t *);

static int (*ALSA_snd_pcm_drain) (snd_pcm_t *);

static const char *(*ALSA_snd_strerror) (int);

static size_t(*ALSA_snd_pcm_hw_params_sizeof) (void);

static size_t(*ALSA_snd_pcm_sw_params_sizeof) (void);

static void (*ALSA_snd_pcm_hw_params_copy)

(snd_pcm_hw_params_t *, const snd_pcm_hw_params_t *);

static int (*ALSA_snd_pcm_hw_params_any) (snd_pcm_t *, snd_pcm_hw_params_t *);

static int (*ALSA_snd_pcm_hw_params_set_access)

(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_access_t);

static int (*ALSA_snd_pcm_hw_params_set_format)

(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_format_t);

static int (*ALSA_snd_pcm_hw_params_set_channels)

(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int);

static int (*ALSA_snd_pcm_hw_params_get_channels)

(const snd_pcm_hw_params_t *, unsigned int *);

static int (*ALSA_snd_pcm_hw_params_set_rate_near)

(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);

static int (*ALSA_snd_pcm_hw_params_set_period_size_near)

(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *);

static int (*ALSA_snd_pcm_hw_params_get_period_size)

(const snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *);

static int (*ALSA_snd_pcm_hw_params_set_periods_near)

(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);

static int (*ALSA_snd_pcm_hw_params_get_periods)

(const snd_pcm_hw_params_t *, unsigned int *, int *);

static int (*ALSA_snd_pcm_hw_params_set_buffer_size_near)

(snd_pcm_t *pcm, snd_pcm_hw_params_t *, snd_pcm_uframes_t *);

static int (*ALSA_snd_pcm_hw_params_get_buffer_size)

(const snd_pcm_hw_params_t *, snd_pcm_uframes_t *);

static int (*ALSA_snd_pcm_hw_params) (snd_pcm_t *, snd_pcm_hw_params_t *);

static int (*ALSA_snd_pcm_sw_params_current) (snd_pcm_t *,

snd_pcm_sw_params_t *);

static int (*ALSA_snd_pcm_sw_params_set_start_threshold)

(snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);

static int (*ALSA_snd_pcm_sw_params) (snd_pcm_t *, snd_pcm_sw_params_t *);

static int (*ALSA_snd_pcm_nonblock) (snd_pcm_t *, int);

static int (*ALSA_snd_pcm_wait)(snd_pcm_t *, int);

#define snd_pcm_hw_params_sizeof ALSA_snd_pcm_hw_params_sizeof

#define snd_pcm_sw_params_sizeof ALSA_snd_pcm_sw_params_sizeof

#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC

static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC;

static void *alsa_handle = NULL;

static int

load_alsa_sym(const char *fn, void **addr)

{

*addr = SDL_LoadFunction(alsa_handle, fn);

if (*addr == NULL) {

/* Don't call SDL_SetError(): SDL_LoadFunction already did. */

return 0;

}

return 1;

}

/* cast funcs to char* first, to please GCC's strict aliasing rules. */

#define SDL_ALSA_SYM(x) \

if (!load_alsa_sym(#x, (void **) (char *) &ALSA_##x)) return -1

#else

#define SDL_ALSA_SYM(x) ALSA_##x = x

#endif

static int

load_alsa_syms(void)

{

SDL_ALSA_SYM(snd_pcm_open);

SDL_ALSA_SYM(snd_pcm_close);

SDL_ALSA_SYM(snd_pcm_writei);

SDL_ALSA_SYM(snd_pcm_recover);

SDL_ALSA_SYM(snd_pcm_prepare);

SDL_ALSA_SYM(snd_pcm_drain);

SDL_ALSA_SYM(snd_strerror);

SDL_ALSA_SYM(snd_pcm_hw_params_sizeof);

SDL_ALSA_SYM(snd_pcm_sw_params_sizeof);

SDL_ALSA_SYM(snd_pcm_hw_params_copy);

SDL_ALSA_SYM(snd_pcm_hw_params_any);

SDL_ALSA_SYM(snd_pcm_hw_params_set_access);

SDL_ALSA_SYM(snd_pcm_hw_params_set_format);

SDL_ALSA_SYM(snd_pcm_hw_params_set_channels);

SDL_ALSA_SYM(snd_pcm_hw_params_get_channels);

SDL_ALSA_SYM(snd_pcm_hw_params_set_rate_near);

SDL_ALSA_SYM(snd_pcm_hw_params_set_period_size_near);

SDL_ALSA_SYM(snd_pcm_hw_params_get_period_size);

SDL_ALSA_SYM(snd_pcm_hw_params_set_periods_near);

SDL_ALSA_SYM(snd_pcm_hw_params_get_periods);

SDL_ALSA_SYM(snd_pcm_hw_params_set_buffer_size_near);

SDL_ALSA_SYM(snd_pcm_hw_params_get_buffer_size);

SDL_ALSA_SYM(snd_pcm_hw_params);

SDL_ALSA_SYM(snd_pcm_sw_params_current);

SDL_ALSA_SYM(snd_pcm_sw_params_set_start_threshold);

SDL_ALSA_SYM(snd_pcm_sw_params);

SDL_ALSA_SYM(snd_pcm_nonblock);

SDL_ALSA_SYM(snd_pcm_wait);

return 0;

}

#undef SDL_ALSA_SYM

#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC

static void

UnloadALSALibrary(void)

{

if (alsa_handle != NULL) {

SDL_UnloadObject(alsa_handle);

alsa_handle = NULL;

}

}

static int

LoadALSALibrary(void)

{

int retval = 0;

if (alsa_handle == NULL) {

alsa_handle = SDL_LoadObject(alsa_library);

if (alsa_handle == NULL) {

retval = -1;

/* Don't call SDL_SetError(): SDL_LoadObject already did. */

} else {

retval = load_alsa_syms();

if (retval < 0) {

UnloadALSALibrary();

}

}

}

return retval;

}

#else

static void

UnloadALSALibrary(void)

{

}

static int

LoadALSALibrary(void)

{

load_alsa_syms();

return 0;

}

#endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */

static const char *

get_audio_device(int channels)

{

const char *device;

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

if (device == NULL) {

switch (channels) {

case 6:

device = "plug:surround51";

break;

case 4:

device = "plug:surround40";

break;

default:

device = "default";

break;

}

}

return device;

}

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

static void

ALSA_WaitDevice(_THIS)

{

/* We're in blocking mode, so there's nothing to do here */

}

/* !!! FIXME: is there a channel swizzler in alsalib instead? */

/*

* 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 *) this->hidden->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->hidden->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. */

}

static void

ALSA_PlayDevice(_THIS)

{

int status;

const Uint8 *sample_buf = (const Uint8 *) this->hidden->mixbuf;

const int frame_size = (((int) (this->spec.format & 0xFF)) / 8) *

this->spec.channels;

snd_pcm_uframes_t frames_left = ((snd_pcm_uframes_t) this->spec.samples);

swizzle_alsa_channels(this);

while ( frames_left > 0 && this->enabled ) {

/* !!! FIXME: This works, but needs more testing before going live */

/*ALSA_snd_pcm_wait(this->hidden->pcm_handle, -1);*/

status = ALSA_snd_pcm_writei(this->hidden->pcm_handle,

sample_buf, frames_left);

if (status < 0) {

if (status == -EAGAIN) {

/* Apparently snd_pcm_recover() doesn't handle this case -

does it assume snd_pcm_wait() above? */

SDL_Delay(1);

continue;

}

status = ALSA_snd_pcm_recover(this->hidden->pcm_handle, status, 0);

if (status < 0) {

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

fprintf(stderr, "ALSA write failed (unrecoverable): %s\n",

ALSA_snd_strerror(status));

this->enabled = 0;

return;

}

continue;

}

sample_buf += status * frame_size;

frames_left -= status;

}

}

static Uint8 *

ALSA_GetDeviceBuf(_THIS)

{

return (this->hidden->mixbuf);

}

static void

ALSA_CloseDevice(_THIS)

{

if (this->hidden != NULL) {

if (this->hidden->mixbuf != NULL) {

SDL_FreeAudioMem(this->hidden->mixbuf);

this->hidden->mixbuf = NULL;

}

if (this->hidden->pcm_handle) {

ALSA_snd_pcm_drain(this->hidden->pcm_handle);

ALSA_snd_pcm_close(this->hidden->pcm_handle);

this->hidden->pcm_handle = NULL;

}

SDL_free(this->hidden);

this->hidden = NULL;

}

}

static int

ALSA_finalize_hardware(_THIS, snd_pcm_hw_params_t *hwparams, int override)

{

int status;

snd_pcm_uframes_t bufsize;

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

status = ALSA_snd_pcm_hw_params(this->hidden->pcm_handle, hwparams);

if ( status < 0 ) {

return(-1);

}

/* Get samples for the actual buffer size */

status = ALSA_snd_pcm_hw_params_get_buffer_size(hwparams, &bufsize);

if ( status < 0 ) {

return(-1);

}

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

return(-1);

}

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

this->spec.samples = bufsize / 2;

/* This is useful for debugging */

if ( SDL_getenv("SDL_AUDIO_ALSA_DEBUG") ) {

snd_pcm_uframes_t persize = 0;

unsigned int periods = 0;

ALSA_snd_pcm_hw_params_get_period_size(hwparams, &persize, NULL);

ALSA_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, snd_pcm_hw_params_t *params, int override)

{

const char *env;

int status;

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);

ALSA_snd_pcm_hw_params_copy(hwparams, params);

if ( !override ) {

env = SDL_getenv("SDL_AUDIO_ALSA_SET_PERIOD_SIZE");

if ( env ) {

override = SDL_atoi(env);

if ( override == 0 ) {

return(-1);

}

}

}

frames = this->spec.samples;

status = ALSA_snd_pcm_hw_params_set_period_size_near(

this->hidden->pcm_handle, hwparams, &frames, NULL);

if ( status < 0 ) {

return(-1);

}

periods = 2;

status = ALSA_snd_pcm_hw_params_set_periods_near(

this->hidden->pcm_handle, hwparams, &periods, NULL);

if ( status < 0 ) {

return(-1);

}

return ALSA_finalize_hardware(this, hwparams, override);

}

static int

ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params, int override)

{

const char *env;

int status;

snd_pcm_hw_params_t *hwparams;

snd_pcm_uframes_t frames;

/* Copy the hardware parameters for this setup */

snd_pcm_hw_params_alloca(&hwparams);

ALSA_snd_pcm_hw_params_copy(hwparams, params);

if ( !override ) {

env = SDL_getenv("SDL_AUDIO_ALSA_SET_BUFFER_SIZE");

if ( env ) {

override = SDL_atoi(env);

if ( override == 0 ) {

return(-1);

}

}

}

frames = this->spec.samples * 2;

status = ALSA_snd_pcm_hw_params_set_buffer_size_near(

this->hidden->pcm_handle, hwparams, &frames);

if ( status < 0 ) {

return(-1);

}

return ALSA_finalize_hardware(this, hwparams, override);

}

static int

ALSA_OpenDevice(_THIS, const char *devname, int iscapture)

{

int status = 0;

snd_pcm_t *pcm_handle = NULL;

snd_pcm_hw_params_t *hwparams = NULL;

snd_pcm_sw_params_t *swparams = NULL;

snd_pcm_format_t format = 0;

SDL_AudioFormat test_format = 0;

unsigned int rate = 0;

unsigned int channels = 0;

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

this->hidden = (struct SDL_PrivateAudioData *)

SDL_malloc((sizeof *this->hidden));

if (this->hidden == NULL) {

SDL_OutOfMemory();

return 0;

}

SDL_memset(this->hidden, 0, (sizeof *this->hidden));

/* Open the audio device */

/* Name of device should depend on # channels in spec */

status = ALSA_snd_pcm_open(&pcm_handle,

get_audio_device(this->spec.channels),

SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);

if (status < 0) {

ALSA_CloseDevice(this);

SDL_SetError("ALSA: Couldn't open audio device: %s",

ALSA_snd_strerror(status));

return 0;

}

this->hidden->pcm_handle = pcm_handle;

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

snd_pcm_hw_params_alloca(&hwparams);

status = ALSA_snd_pcm_hw_params_any(pcm_handle, hwparams);

if (status < 0) {

ALSA_CloseDevice(this);

SDL_SetError("ALSA: Couldn't get hardware config: %s",

ALSA_snd_strerror(status));

return 0;

}

/* SDL only uses interleaved sample output */

status = ALSA_snd_pcm_hw_params_set_access(pcm_handle, hwparams,

SND_PCM_ACCESS_RW_INTERLEAVED);

if (status < 0) {

ALSA_CloseDevice(this);

SDL_SetError("ALSA: Couldn't set interleaved access: %s",

ALSA_snd_strerror(status));

return 0;

}

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

status = -1;

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

test_format && (status < 0);) {

status = 0; /* if we can't support a format, it'll become -1. */

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;

case AUDIO_S32LSB:

format = SND_PCM_FORMAT_S32_LE;

break;

case AUDIO_S32MSB:

format = SND_PCM_FORMAT_S32_BE;

break;

case AUDIO_F32LSB:

format = SND_PCM_FORMAT_FLOAT_LE;

break;

case AUDIO_F32MSB:

format = SND_PCM_FORMAT_FLOAT_BE;

break;

default:

status = -1;

break;

}

if (status >= 0) {

status = ALSA_snd_pcm_hw_params_set_format(pcm_handle,

hwparams, format);

}

if (status < 0) {

test_format = SDL_NextAudioFormat();

}

}

if (status < 0) {

ALSA_CloseDevice(this);

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

return 0;

}

this->spec.format = test_format;

/* Set the number of channels */

status = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams,

this->spec.channels);

channels = this->spec.channels;

if (status < 0) {

status = ALSA_snd_pcm_hw_params_get_channels(hwparams, &channels);

if (status < 0) {

ALSA_CloseDevice(this);

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

return 0;

}

this->spec.channels = channels;

}

/* Set the audio rate */

rate = this->spec.freq;

status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams,

&rate, NULL);

if (status < 0) {

ALSA_CloseDevice(this);

SDL_SetError("ALSA: Couldn't set audio frequency: %s",

ALSA_snd_strerror(status));

return 0;

}

this->spec.freq = rate;

/* Set the buffer size, in samples */

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

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

/* Failed to set desired buffer size, do the best you can... */

if ( ALSA_set_period_size(this, hwparams, 1) < 0 ) {

ALSA_CloseDevice(this);

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

return(-1);

}

}

/* Set the software parameters */

snd_pcm_sw_params_alloca(&swparams);

status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams);

if (status < 0) {

ALSA_CloseDevice(this);

SDL_SetError("ALSA: Couldn't get software config: %s",

ALSA_snd_strerror(status));

return 0;

}

status =

ALSA_snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 1);

if (status < 0) {

ALSA_CloseDevice(this);

SDL_SetError("ALSA: Couldn't set start threshold: %s",

ALSA_snd_strerror(status));

return 0;

}

status = ALSA_snd_pcm_sw_params(pcm_handle, swparams);

if (status < 0) {

ALSA_CloseDevice(this);

SDL_SetError("Couldn't set software audio parameters: %s",

ALSA_snd_strerror(status));

return 0;

}

/* Calculate the final parameters for this audio specification */

SDL_CalculateAudioSpec(&this->spec);

/* Allocate mixing buffer */

this->hidden->mixlen = this->spec.size;

this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);

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

ALSA_CloseDevice(this);

SDL_OutOfMemory();

return 0;

}

SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);

/* Switch to blocking mode for playback */

ALSA_snd_pcm_nonblock(pcm_handle, 0);

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

return 1;

}

static void

ALSA_Deinitialize(void)

{

UnloadALSALibrary();

}

static int

ALSA_Init(SDL_AudioDriverImpl * impl)

{

if (LoadALSALibrary() < 0) {

return 0;

}

/* Set the function pointers */

impl->OpenDevice = ALSA_OpenDevice;

impl->WaitDevice = ALSA_WaitDevice;

impl->GetDeviceBuf = ALSA_GetDeviceBuf;

impl->PlayDevice = ALSA_PlayDevice;

impl->CloseDevice = ALSA_CloseDevice;

impl->Deinitialize = ALSA_Deinitialize;

impl->OnlyHasDefaultOutputDevice = 1; /* !!! FIXME: Add device enum! */

return 1; /* this audio target is available. */

}

AudioBootStrap ALSA_bootstrap = {

DRIVER_NAME, "ALSA PCM audio", ALSA_Init, 0

};

/* vi: set ts=4 sw=4 expandtab: */