/*

Simple DirectMedia Layer

This software is provided 'as-is', without any express or implied

warranty. In no event will the authors be held liable for any damages

arising from the use of this software.

Permission is granted to anyone to use this software for any purpose,

including commercial applications, and to alter it and redistribute it

freely, subject to the following restrictions:

1. The origin of this software must not be misrepresented; you must not

claim that you wrote the original software. If you use this software

in a product, an acknowledgment in the product documentation would be

appreciated but is not required.

2. Altered source versions must be plainly marked as such, and must not be

misrepresented as being the original software.

3. This notice may not be removed or altered from any source distribution.

*/

#include "../../SDL_internal.h"

#if SDL_AUDIO_DRIVER_ALSA

/* 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_audio_c.h"

#include "SDL_alsa_audio.h"

#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC

#include "SDL_loadso.h"

#endif

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_readi)

(snd_pcm_t *, 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);

static int (*ALSA_snd_pcm_sw_params_set_avail_min)

(snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);

static int (*ALSA_snd_device_name_hint) (int, const char *, void ***);

static char* (*ALSA_snd_device_name_get_hint) (const void *, const char *);

static int (*ALSA_snd_device_name_free_hint) (void **);

#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC

#define snd_pcm_hw_params_sizeof ALSA_snd_pcm_hw_params_sizeof

#define snd_pcm_sw_params_sizeof ALSA_snd_pcm_sw_params_sizeof

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);

SDL_ALSA_SYM(snd_pcm_sw_params_set_avail_min);

SDL_ALSA_SYM(snd_device_name_hint);

SDL_ALSA_SYM(snd_device_name_get_hint);

SDL_ALSA_SYM(snd_device_name_free_hint);

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 *

{

const char *device;

if (handle != NULL) {

return (const char *) handle;

}

/* !!! FIXME: we also check "SDL_AUDIO_DEVICE_NAME" at the higher level. */

if (device != NULL) {

return device;

}

if (channels == 6) {

return "plug:surround51";

} else if (channels == 4) {

return "plug:surround40";

return "default";

}

/* 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, buf, numframes) \

T *ptr = (T *) buf; \

T tmp; \

tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \

tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \

}

static SDL_INLINE void

}

static SDL_INLINE void

}

static SDL_INLINE void

}

static SDL_INLINE void

}

#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 SDL_INLINE void

{

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

switch (SDL_AUDIO_BITSIZE(this->spec.format)) {

case 8: swizzle_alsa_channels_6_8bit(buffer, bufferlen); break;

case 16: swizzle_alsa_channels_6_16bit(buffer, bufferlen); break;

case 32: swizzle_alsa_channels_6_32bit(buffer, bufferlen); break;

case 64: swizzle_alsa_channels_6_64bit(buffer, bufferlen); break;

default: SDL_assert(!"unhandled bitsize"); break;

}

}

/* !!! FIXME: update this for 7.1 if needed, later. */

}

static void

ALSA_PlayDevice(_THIS)

{

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

this->spec.channels;

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

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

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

int 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));

SDL_OpenedAudioDeviceDisconnected(this);

return;

}

continue;

}

sample_buf += status * frame_size;

frames_left -= status;

}

}

static Uint8 *

ALSA_GetDeviceBuf(_THIS)

{

return (this->hidden->mixbuf);

}

static int

ALSA_CaptureFromDevice(_THIS, void *buffer, int buflen)

{

Uint8 *sample_buf = (Uint8 *) buffer;

const int frame_size = (((int) SDL_AUDIO_BITSIZE(this->spec.format)) / 8) *

this->spec.channels;

const int total_frames = buflen / frame_size;

snd_pcm_uframes_t frames_left = total_frames;

SDL_assert((buflen % frame_size) == 0);

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

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

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

int status = ALSA_snd_pcm_readi(this->hidden->pcm_handle,

sample_buf, frames_left);

if (status < 0) {

/*printf("ALSA: capture error %d\n", status);*/

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 read failed (unrecoverable): %s\n",

ALSA_snd_strerror(status));

return -1;

}

continue;

}

/*printf("ALSA: captured %d bytes\n", status * frame_size);*/

sample_buf += status * frame_size;

frames_left -= status;

}

swizzle_alsa_channels(this, buffer, total_frames - frames_left);

return (total_frames - frames_left) * frame_size;

}

static void

ALSA_FlushCapture(_THIS)

{

ALSA_snd_pcm_reset(this->hidden->pcm_handle);

}

static void

ALSA_CloseDevice(_THIS)

{

/* Wait for the submitted audio to drain

ALSA_snd_pcm_drop() can hang, so don't use that.

*/

Uint32 delay = ((this->spec.samples * 1000) / this->spec.freq) * 2;

SDL_Delay(delay);

}

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, void *handle, 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) {

return SDL_OutOfMemory();

}

/* Open the audio device */

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

status = ALSA_snd_pcm_open(&pcm_handle,

get_audio_device(handle, this->spec.channels),

iscapture ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,

SND_PCM_NONBLOCK);

if (status < 0) {

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

ALSA_snd_strerror(status));

}

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) {

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

ALSA_snd_strerror(status));

}

/* 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) {

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

ALSA_snd_strerror(status));

}

/* 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) {

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

}

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) {

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

}

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) {

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

ALSA_snd_strerror(status));

}

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... */

status = ALSA_set_period_size(this, hwparams, 1);

if (status < 0) {

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

}

}

/* Set the software parameters */

snd_pcm_sw_params_alloca(&swparams);

status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams);

if (status < 0) {

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

ALSA_snd_strerror(status));

}

status = ALSA_snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, this->spec.samples);

if (status < 0) {

return SDL_SetError("Couldn't set minimum available samples: %s",

ALSA_snd_strerror(status));

}

status =

ALSA_snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 1);

if (status < 0) {

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

ALSA_snd_strerror(status));

}

status = ALSA_snd_pcm_sw_params(pcm_handle, swparams);

if (status < 0) {

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

ALSA_snd_strerror(status));

}

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

SDL_CalculateAudioSpec(&this->spec);

/* Allocate mixing buffer */

if (!iscapture) {

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

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

return SDL_OutOfMemory();

}

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

}

/* Switch to blocking mode for playback */

ALSA_snd_pcm_nonblock(pcm_handle, 0);

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

return 0;

}

char *name;

SDL_bool iscapture;

struct ALSA_Device *next;

} ALSA_Device;

ALSA_Device *dev = SDL_malloc(sizeof (ALSA_Device));

char *desc = ALSA_snd_device_name_get_hint(hint, "DESC");

SDL_free(dev);

return;

} else if (!dev) {

free(desc);

return;

}

SDL_assert(name != NULL);

/* some strings have newlines, like "HDA NVidia, HDMI 0\nHDMI Audio Output".

just chop the extra lines off, this seems to get a reasonable device

name without extra details. */

if ((ptr = strchr(desc, '\n')) != NULL) {

*ptr = '\0';

}

/*printf("ALSA: adding %s device '%s' (%s)\n", iscapture ? "capture" : "output", name, desc);*/

if (!handle) {

free(desc);

SDL_free(dev);

return;

}

SDL_AddAudioDevice(iscapture, desc, handle);

free(desc);

dev->name = handle;

dev->iscapture = iscapture;

dev->next = *pSeen;

*pSeen = dev;

}

static SDL_atomic_t ALSA_hotplug_shutdown;

static SDL_Thread *ALSA_hotplug_thread;

static int SDLCALL

ALSA_HotplugThread(void *arg)

{

SDL_sem *first_run_semaphore = (SDL_sem *) arg;

ALSA_Device *devices = NULL;

ALSA_Device *next;

ALSA_Device *dev;

Uint32 ticks;

while (!SDL_AtomicGet(&ALSA_hotplug_shutdown)) {

void **hints = NULL;

if (ALSA_snd_device_name_hint(-1, "pcm", &hints) != -1) {

ALSA_Device *unseen = devices;

ALSA_Device *seen = NULL;

ALSA_Device *prev;

int i, j;

const char *match = NULL;

int bestmatch = 0xFFFF;

size_t match_len = 0;

int defaultdev = -1;

static const char * const prefixes[] = {

"hw:", "sysdefault:", "default:", NULL

};

/* Apparently there are several different ways that ALSA lists

actual hardware. It could be prefixed with "hw:" or "default:"

or "sysdefault:" and maybe others. Go through the list and see

if we can find a preferred prefix for the system. */

for (i = 0; hints[i]; i++) {

char *name = ALSA_snd_device_name_get_hint(hints[i], "NAME");

if (!name) {

continue;

}

/* full name, not a prefix */

if ((defaultdev == -1) && (SDL_strcmp(name, "default") == 0)) {

defaultdev = i;

}

for (j = 0; prefixes[j]; j++) {

const char *prefix = prefixes[j];

if (SDL_strncmp(name, prefix, prefixlen) == 0) {

if (j < bestmatch) {

bestmatch = j;

match = prefix;

match_len = prefixlen;

}

}

}

free(name);

}

/* look through the list of device names to find matches */

char *name;

/* if we didn't find a device name prefix we like at all... */

if ((!match) && (defaultdev != i)) {

continue; /* ...skip anything that isn't the default device. */

}

name = ALSA_snd_device_name_get_hint(hints[i], "NAME");

if (!name) {

continue;

}

/* only want physical hardware interfaces */

char *ioid = ALSA_snd_device_name_get_hint(hints[i], "IOID");

const SDL_bool isoutput = (ioid == NULL) || (SDL_strcmp(ioid, "Output") == 0);

const SDL_bool isinput = (ioid == NULL) || (SDL_strcmp(ioid, "Input") == 0);

SDL_bool have_output = SDL_FALSE;

SDL_bool have_input = SDL_FALSE;

free(ioid);

if (!isoutput && !isinput) {

free(name);

continue;

}

prev = NULL;

for (dev = unseen; dev; dev = next) {

next = dev->next;

if ( (SDL_strcmp(dev->name, name) == 0) && (((isinput) && dev->iscapture) || ((isoutput) && !dev->iscapture)) ) {

if (prev) {

prev->next = next;

} else {

unseen = next;

}

dev->next = seen;

seen = dev;

if (isinput) have_input = SDL_TRUE;

if (isoutput) have_output = SDL_TRUE;

} else {

prev = dev;

}

}

if (isinput && !have_input) {

add_device(SDL_TRUE, name, hints[i], &seen);

}

if (isoutput && !have_output) {

add_device(SDL_FALSE, name, hints[i], &seen);

}

}

free(name);

ALSA_snd_device_name_free_hint(hints);

devices = seen; /* now we have a known-good list of attached devices. */

/* report anything still in unseen as removed. */

for (dev = unseen; dev; dev = next) {

/*printf("ALSA: removing %s device '%s'\n", dev->iscapture ? "capture" : "output", dev->name);*/

next = dev->next;

SDL_RemoveAudioDevice(dev->iscapture, dev->name);

SDL_free(dev->name);

SDL_free(dev);

}

/* On first run, tell ALSA_DetectDevices() that we have a complete device list so it can return. */

if (first_run_semaphore) {

SDL_SemPost(first_run_semaphore);

first_run_semaphore = NULL; /* let other thread clean it up. */

}

/* Block awhile before checking again, unless we're told to stop. */

ticks = SDL_GetTicks() + 5000;

SDL_Delay(100);

}

}

/* Shutting down! Clean up any data we've gathered. */

for (dev = devices; dev; dev = next) {

/*printf("ALSA: at shutdown, removing %s device '%s'\n", dev->iscapture ? "capture" : "output", dev->name);*/

next = dev->next;

SDL_free(dev->name);

SDL_free(dev);

}

return 0;

}

static void

/* Start the device detection thread here, wait for an initial iteration to complete. */

SDL_sem *semaphore = SDL_CreateSemaphore(0);

if (!semaphore) {

return; /* oh well. */

}

SDL_AtomicSet(&ALSA_hotplug_shutdown, 0);

ALSA_hotplug_thread = SDL_CreateThread(ALSA_HotplugThread, "SDLHotplugALSA", semaphore);

if (ALSA_hotplug_thread) {

SDL_SemWait(semaphore); /* wait for the first iteration to finish. */

}

SDL_DestroySemaphore(semaphore);

}

static void

ALSA_Deinitialize(void)

{

if (ALSA_hotplug_thread != NULL) {

SDL_AtomicSet(&ALSA_hotplug_shutdown, 1);

SDL_WaitThread(ALSA_hotplug_thread, NULL);

ALSA_hotplug_thread = NULL;

}

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->CaptureFromDevice = ALSA_CaptureFromDevice;

impl->FlushCapture = ALSA_FlushCapture;

impl->HasCaptureSupport = SDL_TRUE;

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

}

AudioBootStrap ALSA_bootstrap = {

"alsa", "ALSA PCM audio", ALSA_Init, 0

};

#endif /* SDL_AUDIO_DRIVER_ALSA */

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