audio/sdlaudio.c - qemu - Git at Google

 /*
  * QEMU SDL audio driver
  *
  * Copyright (c) 2004-2005 Vassili Karpov (malc)
  *
  * Permission is hereby granted, free of charge, to any person obtaining a copy
  * of this software and associated documentation files (the "Software"), to deal
  * in the Software without restriction, including without limitation the rights
  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
  * copies of the Software, and to permit persons to whom the Software is
  * furnished to do so, subject to the following conditions:
  *
  * The above copyright notice and this permission notice shall be included in
  * all copies or substantial portions of the Software.
  *
  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
  * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  * THE SOFTWARE.
  */
 #include <SDL.h>
 #include <SDL_thread.h>
 #include "qemu-common.h"
 #include "audio.h"

 #ifndef _WIN32
 #ifdef __sun__
 #define _POSIX_PTHREAD_SEMANTICS 1
 #elif defined(__OpenBSD__) || defined(__FreeBSD__) || defined(__DragonFly__)
 #include <pthread.h>
 #endif
 #include <signal.h>
 #endif

 #define AUDIO_CAP "sdl"
 #include "audio_int.h"

 typedef struct SDLVoiceOut {
     HWVoiceOut hw;
     int live;
     int rpos;
     int decr;
 } SDLVoiceOut;

 static struct {
     int nb_samples;
 } conf = {
     .nb_samples = 1024
 };

 static struct SDLAudioState {
     int exit;
     SDL_mutex *mutex;
     SDL_sem *sem;
     int initialized;
 } glob_sdl;
 typedef struct SDLAudioState SDLAudioState;

 static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
 {
     va_list ap;

     va_start (ap, fmt);
     AUD_vlog (AUDIO_CAP, fmt, ap);
     va_end (ap);

     AUD_log (AUDIO_CAP, "Reason: %s\n", SDL_GetError ());
 }

 static int sdl_lock (SDLAudioState *s, const char *forfn)
 {
     if (SDL_LockMutex (s->mutex)) {
         sdl_logerr ("SDL_LockMutex for %s failed\n", forfn);
         return -1;
     }
     return 0;
 }

 static int sdl_unlock (SDLAudioState *s, const char *forfn)
 {
     if (SDL_UnlockMutex (s->mutex)) {
         sdl_logerr ("SDL_UnlockMutex for %s failed\n", forfn);
         return -1;
     }
     return 0;
 }

 static int sdl_post (SDLAudioState *s, const char *forfn)
 {
     if (SDL_SemPost (s->sem)) {
         sdl_logerr ("SDL_SemPost for %s failed\n", forfn);
         return -1;
     }
     return 0;
 }

 static int sdl_wait (SDLAudioState *s, const char *forfn)
 {
     if (SDL_SemWait (s->sem)) {
         sdl_logerr ("SDL_SemWait for %s failed\n", forfn);
         return -1;
     }
     return 0;
 }

 static int sdl_unlock_and_post (SDLAudioState *s, const char *forfn)
 {
     if (sdl_unlock (s, forfn)) {
         return -1;
     }

     return sdl_post (s, forfn);
 }

 static int aud_to_sdlfmt (audfmt_e fmt, int *shift)
 {
     switch (fmt) {
     case AUD_FMT_S8:
         *shift = 0;
         return AUDIO_S8;

     case AUD_FMT_U8:
         *shift = 0;
         return AUDIO_U8;

     case AUD_FMT_S16:
         *shift = 1;
         return AUDIO_S16LSB;

     case AUD_FMT_U16:
         *shift = 1;
         return AUDIO_U16LSB;

     default:
         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 #ifdef DEBUG_AUDIO
         abort ();
 #endif
         return AUDIO_U8;
     }
 }

 static int sdl_to_audfmt (int sdlfmt, audfmt_e *fmt, int *endianess)
 {
     switch (sdlfmt) {
     case AUDIO_S8:
         *endianess = 0;
         *fmt = AUD_FMT_S8;
         break;

     case AUDIO_U8:
         *endianess = 0;
         *fmt = AUD_FMT_U8;
         break;

     case AUDIO_S16LSB:
         *endianess = 0;
         *fmt = AUD_FMT_S16;
         break;

     case AUDIO_U16LSB:
         *endianess = 0;
         *fmt = AUD_FMT_U16;
         break;

     case AUDIO_S16MSB:
         *endianess = 1;
         *fmt = AUD_FMT_S16;
         break;

     case AUDIO_U16MSB:
         *endianess = 1;
         *fmt = AUD_FMT_U16;
         break;

     default:
         dolog ("Unrecognized SDL audio format %d\n", sdlfmt);
         return -1;
     }

     return 0;
 }

 static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
 {
     int status;
 #ifndef _WIN32
     sigset_t new, old;

     /* Make sure potential threads created by SDL don't hog signals.  */
     sigfillset (&new);
     pthread_sigmask (SIG_BLOCK, &new, &old);
 #endif

     status = SDL_OpenAudio (req, obt);
     if (status) {
         sdl_logerr ("SDL_OpenAudio failed\n");
     }

 #ifndef _WIN32
     pthread_sigmask (SIG_SETMASK, &old, NULL);
 #endif
     return status;
 }

 static void sdl_close (SDLAudioState *s)
 {
     if (s->initialized) {
         sdl_lock (s, "sdl_close");
         s->exit = 1;
         sdl_unlock_and_post (s, "sdl_close");
         SDL_PauseAudio (1);
         SDL_CloseAudio ();
         s->initialized = 0;
     }
 }

 static void sdl_callback (void *opaque, Uint8 *buf, int len)
 {
     SDLVoiceOut *sdl = opaque;
     SDLAudioState *s = &glob_sdl;
     HWVoiceOut *hw = &sdl->hw;
     int samples = len >> hw->info.shift;

     if (s->exit) {
         return;
     }

     while (samples) {
         int to_mix, decr;

         /* dolog ("in callback samples=%d\n", samples); */
         sdl_wait (s, "sdl_callback");
         if (s->exit) {
             return;
         }

         if (sdl_lock (s, "sdl_callback")) {
             return;
         }

         if (audio_bug (AUDIO_FUNC, sdl->live < 0 || sdl->live > hw->samples)) {
             dolog ("sdl->live=%d hw->samples=%d\n",
                    sdl->live, hw->samples);
             return;
         }

         if (!sdl->live) {
             goto again;
         }

         /* dolog ("in callback live=%d\n", live); */
         to_mix = audio_MIN (samples, sdl->live);
         decr = to_mix;
         while (to_mix) {
             int chunk = audio_MIN (to_mix, hw->samples - hw->rpos);
             struct st_sample *src = hw->mix_buf + hw->rpos;

             /* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */
             hw->clip (buf, src, chunk);
             sdl->rpos = (sdl->rpos + chunk) % hw->samples;
             to_mix -= chunk;
             buf += chunk << hw->info.shift;
         }
         samples -= decr;
         sdl->live -= decr;
         sdl->decr += decr;

     again:
         if (sdl_unlock (s, "sdl_callback")) {
             return;
         }
     }
     /* dolog ("done len=%d\n", len); */
 }

 static int sdl_write_out (SWVoiceOut *sw, void *buf, int len)
 {
     return audio_pcm_sw_write (sw, buf, len);
 }

 static int sdl_run_out (HWVoiceOut *hw)
 {
     int decr, live;
     SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
     SDLAudioState *s = &glob_sdl;

     if (sdl_lock (s, "sdl_callback")) {
         return 0;
     }

     live = audio_pcm_hw_get_live_out (hw);

     if (sdl->decr > live) {
         ldebug ("sdl->decr %d live %d sdl->live %d\n",
                 sdl->decr,
                 live,
                 sdl->live);
     }

     decr = audio_MIN (sdl->decr, live);
     sdl->decr -= decr;

     sdl->live = live - decr;
     hw->rpos = sdl->rpos;

     if (sdl->live > 0) {
         sdl_unlock_and_post (s, "sdl_callback");
     }
     else {
         sdl_unlock (s, "sdl_callback");
     }
     return decr;
 }

 static void sdl_fini_out (HWVoiceOut *hw)
 {
     (void) hw;

     sdl_close (&glob_sdl);
 }

 static int sdl_init_out (HWVoiceOut *hw, struct audsettings *as)
 {
     SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
     SDLAudioState *s = &glob_sdl;
     SDL_AudioSpec req, obt;
     int shift;
     int endianess;
     int err;
     audfmt_e effective_fmt;
     struct audsettings obt_as;

     shift <<= as->nchannels == 2;

     req.freq = as->freq;
     req.format = aud_to_sdlfmt (as->fmt, &shift);
     req.channels = as->nchannels;
     req.samples = conf.nb_samples;
     req.callback = sdl_callback;
     req.userdata = sdl;

     if (sdl_open (&req, &obt)) {
         return -1;
     }

     err = sdl_to_audfmt (obt.format, &effective_fmt, &endianess);
     if (err) {
         sdl_close (s);
         return -1;
     }

     obt_as.freq = obt.freq;
     obt_as.nchannels = obt.channels;
     obt_as.fmt = effective_fmt;
     obt_as.endianness = endianess;

     audio_pcm_init_info (&hw->info, &obt_as);
     hw->samples = obt.samples;

     s->initialized = 1;
     s->exit = 0;
     SDL_PauseAudio (0);
     return 0;
 }

 static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
 {
     (void) hw;

     switch (cmd) {
     case VOICE_ENABLE:
         SDL_PauseAudio (0);
         break;

     case VOICE_DISABLE:
         SDL_PauseAudio (1);
         break;
     }
     return 0;
 }

 static void *sdl_audio_init (void)
 {
     SDLAudioState *s = &glob_sdl;

     if (SDL_InitSubSystem (SDL_INIT_AUDIO)) {
         sdl_logerr ("SDL failed to initialize audio subsystem\n");
         return NULL;
     }

     s->mutex = SDL_CreateMutex ();
     if (!s->mutex) {
         sdl_logerr ("Failed to create SDL mutex\n");
         SDL_QuitSubSystem (SDL_INIT_AUDIO);
         return NULL;
     }

     s->sem = SDL_CreateSemaphore (0);
     if (!s->sem) {
         sdl_logerr ("Failed to create SDL semaphore\n");
         SDL_DestroyMutex (s->mutex);
         SDL_QuitSubSystem (SDL_INIT_AUDIO);
         return NULL;
     }

     return s;
 }

 static void sdl_audio_fini (void *opaque)
 {
     SDLAudioState *s = opaque;
     sdl_close (s);
     SDL_DestroySemaphore (s->sem);
     SDL_DestroyMutex (s->mutex);
     SDL_QuitSubSystem (SDL_INIT_AUDIO);
 }

 static struct audio_option sdl_options[] = {
     {
         .name  = "SAMPLES",
         .tag   = AUD_OPT_INT,
         .valp  = &conf.nb_samples,
         .descr = "Size of SDL buffer in samples"
     },
     { /* End of list */ }
 };

 static struct audio_pcm_ops sdl_pcm_ops = {
     .init_out = sdl_init_out,
     .fini_out = sdl_fini_out,
     .run_out  = sdl_run_out,
     .write    = sdl_write_out,
     .ctl_out  = sdl_ctl_out,
 };

 struct audio_driver sdl_audio_driver = {
     .name           = "sdl",
     .descr          = "SDL http://www.libsdl.org",
     .options        = sdl_options,
     .init           = sdl_audio_init,
     .fini           = sdl_audio_fini,
     .pcm_ops        = &sdl_pcm_ops,
     .can_be_default = 1,
     .max_voices_out = 1,
     .max_voices_in  = 0,
     .voice_size_out = sizeof (SDLVoiceOut),
     .voice_size_in  = 0
 };
	/*
	* QEMU SDL audio driver
	*
	* Copyright (c) 2004-2005 Vassili Karpov (malc)
	*
	* Permission is hereby granted, free of charge, to any person obtaining a copy
	* of this software and associated documentation files (the "Software"), to deal
	* in the Software without restriction, including without limitation the rights
	* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
	* copies of the Software, and to permit persons to whom the Software is
	* furnished to do so, subject to the following conditions:
	*
	* The above copyright notice and this permission notice shall be included in
	* all copies or substantial portions of the Software.
	*
	* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
	* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
	* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
	* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
	* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
	* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
	* THE SOFTWARE.
	*/
	#include <SDL.h>
	#include <SDL_thread.h>
	#include "qemu-common.h"
	#include "audio.h"

	#ifndef _WIN32
	#ifdef __sun__
	#define _POSIX_PTHREAD_SEMANTICS 1
	#elif defined(__OpenBSD__) \|\| defined(__FreeBSD__) \|\| defined(__DragonFly__)
	#include <pthread.h>
	#endif
	#include <signal.h>
	#endif

	#define AUDIO_CAP "sdl"
	#include "audio_int.h"

	typedef struct SDLVoiceOut {
	HWVoiceOut hw;
	int live;
	int rpos;
	int decr;
	} SDLVoiceOut;

	static struct {
	int nb_samples;
	} conf = {
	.nb_samples = 1024
	};

	static struct SDLAudioState {
	int exit;
	SDL_mutex *mutex;
	SDL_sem *sem;
	int initialized;
	} glob_sdl;
	typedef struct SDLAudioState SDLAudioState;

	static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
	{
	va_list ap;

	va_start (ap, fmt);
	AUD_vlog (AUDIO_CAP, fmt, ap);
	va_end (ap);

	AUD_log (AUDIO_CAP, "Reason: %s\n", SDL_GetError ());
	}

	static int sdl_lock (SDLAudioState s, const char forfn)
	{
	if (SDL_LockMutex (s->mutex)) {
	sdl_logerr ("SDL_LockMutex for %s failed\n", forfn);
	return -1;
	}
	return 0;
	}

	static int sdl_unlock (SDLAudioState s, const char forfn)
	{
	if (SDL_UnlockMutex (s->mutex)) {
	sdl_logerr ("SDL_UnlockMutex for %s failed\n", forfn);
	return -1;
	}
	return 0;
	}

	static int sdl_post (SDLAudioState s, const char forfn)
	{
	if (SDL_SemPost (s->sem)) {
	sdl_logerr ("SDL_SemPost for %s failed\n", forfn);
	return -1;
	}
	return 0;
	}

	static int sdl_wait (SDLAudioState s, const char forfn)
	{
	if (SDL_SemWait (s->sem)) {
	sdl_logerr ("SDL_SemWait for %s failed\n", forfn);
	return -1;
	}
	return 0;
	}

	static int sdl_unlock_and_post (SDLAudioState s, const char forfn)
	{
	if (sdl_unlock (s, forfn)) {
	return -1;
	}

	return sdl_post (s, forfn);
	}

	static int aud_to_sdlfmt (audfmt_e fmt, int *shift)
	{
	switch (fmt) {
	case AUD_FMT_S8:
	*shift = 0;
	return AUDIO_S8;

	case AUD_FMT_U8:
	*shift = 0;
	return AUDIO_U8;

	case AUD_FMT_S16:
	*shift = 1;
	return AUDIO_S16LSB;

	case AUD_FMT_U16:
	*shift = 1;
	return AUDIO_U16LSB;

	default:
	dolog ("Internal logic error: Bad audio format %d\n", fmt);
	#ifdef DEBUG_AUDIO
	abort ();
	#endif
	return AUDIO_U8;
	}
	}

	static int sdl_to_audfmt (int sdlfmt, audfmt_e fmt, int endianess)
	{
	switch (sdlfmt) {
	case AUDIO_S8:
	*endianess = 0;
	*fmt = AUD_FMT_S8;
	break;

	case AUDIO_U8:
	*endianess = 0;
	*fmt = AUD_FMT_U8;
	break;

	case AUDIO_S16LSB:
	*endianess = 0;
	*fmt = AUD_FMT_S16;
	break;

	case AUDIO_U16LSB:
	*endianess = 0;
	*fmt = AUD_FMT_U16;
	break;

	case AUDIO_S16MSB:
	*endianess = 1;
	*fmt = AUD_FMT_S16;
	break;

	case AUDIO_U16MSB:
	*endianess = 1;
	*fmt = AUD_FMT_U16;
	break;

	default:
	dolog ("Unrecognized SDL audio format %d\n", sdlfmt);
	return -1;
	}

	return 0;
	}

	static int sdl_open (SDL_AudioSpec req, SDL_AudioSpec obt)
	{
	int status;
	#ifndef _WIN32
	sigset_t new, old;

	/* Make sure potential threads created by SDL don't hog signals. */
	sigfillset (&new);
	pthread_sigmask (SIG_BLOCK, &new, &old);
	#endif

	status = SDL_OpenAudio (req, obt);
	if (status) {
	sdl_logerr ("SDL_OpenAudio failed\n");
	}

	#ifndef _WIN32
	pthread_sigmask (SIG_SETMASK, &old, NULL);
	#endif
	return status;
	}

	static void sdl_close (SDLAudioState *s)
	{
	if (s->initialized) {
	sdl_lock (s, "sdl_close");
	s->exit = 1;
	sdl_unlock_and_post (s, "sdl_close");
	SDL_PauseAudio (1);
	SDL_CloseAudio ();
	s->initialized = 0;
	}
	}

	static void sdl_callback (void opaque, Uint8 buf, int len)
	{
	SDLVoiceOut *sdl = opaque;
	SDLAudioState *s = &glob_sdl;
	HWVoiceOut *hw = &sdl->hw;
	int samples = len >> hw->info.shift;

	if (s->exit) {
	return;
	}

	while (samples) {
	int to_mix, decr;

	/* dolog ("in callback samples=%d\n", samples); */
	sdl_wait (s, "sdl_callback");
	if (s->exit) {
	return;
	}

	if (sdl_lock (s, "sdl_callback")) {
	return;
	}

	if (audio_bug (AUDIO_FUNC, sdl->live < 0 \|\| sdl->live > hw->samples)) {
	dolog ("sdl->live=%d hw->samples=%d\n",
	sdl->live, hw->samples);
	return;
	}

	if (!sdl->live) {
	goto again;
	}

	/* dolog ("in callback live=%d\n", live); */
	to_mix = audio_MIN (samples, sdl->live);
	decr = to_mix;
	while (to_mix) {
	int chunk = audio_MIN (to_mix, hw->samples - hw->rpos);
	struct st_sample *src = hw->mix_buf + hw->rpos;

	/* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */
	hw->clip (buf, src, chunk);
	sdl->rpos = (sdl->rpos + chunk) % hw->samples;
	to_mix -= chunk;
	buf += chunk << hw->info.shift;
	}
	samples -= decr;
	sdl->live -= decr;
	sdl->decr += decr;

	again:
	if (sdl_unlock (s, "sdl_callback")) {
	return;
	}
	}
	/* dolog ("done len=%d\n", len); */
	}

	static int sdl_write_out (SWVoiceOut sw, void buf, int len)
	{
	return audio_pcm_sw_write (sw, buf, len);
	}

	static int sdl_run_out (HWVoiceOut *hw)
	{
	int decr, live;
	SDLVoiceOut sdl = (SDLVoiceOut ) hw;
	SDLAudioState *s = &glob_sdl;

	if (sdl_lock (s, "sdl_callback")) {
	return 0;
	}

	live = audio_pcm_hw_get_live_out (hw);

	if (sdl->decr > live) {
	ldebug ("sdl->decr %d live %d sdl->live %d\n",
	sdl->decr,
	live,
	sdl->live);
	}

	decr = audio_MIN (sdl->decr, live);
	sdl->decr -= decr;

	sdl->live = live - decr;
	hw->rpos = sdl->rpos;

	if (sdl->live > 0) {
	sdl_unlock_and_post (s, "sdl_callback");
	}
	else {
	sdl_unlock (s, "sdl_callback");
	}
	return decr;
	}

	static void sdl_fini_out (HWVoiceOut *hw)
	{
	(void) hw;

	sdl_close (&glob_sdl);
	}

	static int sdl_init_out (HWVoiceOut hw, struct audsettings as)
	{
	SDLVoiceOut sdl = (SDLVoiceOut ) hw;
	SDLAudioState *s = &glob_sdl;
	SDL_AudioSpec req, obt;
	int shift;
	int endianess;
	int err;
	audfmt_e effective_fmt;
	struct audsettings obt_as;

	shift <<= as->nchannels == 2;

	req.freq = as->freq;
	req.format = aud_to_sdlfmt (as->fmt, &shift);
	req.channels = as->nchannels;
	req.samples = conf.nb_samples;
	req.callback = sdl_callback;
	req.userdata = sdl;

	if (sdl_open (&req, &obt)) {
	return -1;
	}

	err = sdl_to_audfmt (obt.format, &effective_fmt, &endianess);
	if (err) {
	sdl_close (s);
	return -1;
	}

	obt_as.freq = obt.freq;
	obt_as.nchannels = obt.channels;
	obt_as.fmt = effective_fmt;
	obt_as.endianness = endianess;

	audio_pcm_init_info (&hw->info, &obt_as);
	hw->samples = obt.samples;

	s->initialized = 1;
	s->exit = 0;
	SDL_PauseAudio (0);
	return 0;
	}

	static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
	{
	(void) hw;

	switch (cmd) {
	case VOICE_ENABLE:
	SDL_PauseAudio (0);
	break;

	case VOICE_DISABLE:
	SDL_PauseAudio (1);
	break;
	}
	return 0;
	}

	static void *sdl_audio_init (void)
	{
	SDLAudioState *s = &glob_sdl;

	if (SDL_InitSubSystem (SDL_INIT_AUDIO)) {
	sdl_logerr ("SDL failed to initialize audio subsystem\n");
	return NULL;
	}

	s->mutex = SDL_CreateMutex ();
	if (!s->mutex) {
	sdl_logerr ("Failed to create SDL mutex\n");
	SDL_QuitSubSystem (SDL_INIT_AUDIO);
	return NULL;
	}

	s->sem = SDL_CreateSemaphore (0);
	if (!s->sem) {
	sdl_logerr ("Failed to create SDL semaphore\n");
	SDL_DestroyMutex (s->mutex);
	SDL_QuitSubSystem (SDL_INIT_AUDIO);
	return NULL;
	}

	return s;
	}

	static void sdl_audio_fini (void *opaque)
	{
	SDLAudioState *s = opaque;
	sdl_close (s);
	SDL_DestroySemaphore (s->sem);
	SDL_DestroyMutex (s->mutex);
	SDL_QuitSubSystem (SDL_INIT_AUDIO);
	}

	static struct audio_option sdl_options[] = {
	{
	.name = "SAMPLES",
	.tag = AUD_OPT_INT,
	.valp = &conf.nb_samples,
	.descr = "Size of SDL buffer in samples"
	},
	{ /* End of list */ }
	};

	static struct audio_pcm_ops sdl_pcm_ops = {
	.init_out = sdl_init_out,
	.fini_out = sdl_fini_out,
	.run_out = sdl_run_out,
	.write = sdl_write_out,
	.ctl_out = sdl_ctl_out,
	};

	struct audio_driver sdl_audio_driver = {
	.name = "sdl",
	.descr = "SDL http://www.libsdl.org",
	.options = sdl_options,
	.init = sdl_audio_init,
	.fini = sdl_audio_fini,
	.pcm_ops = &sdl_pcm_ops,
	.can_be_default = 1,
	.max_voices_out = 1,
	.max_voices_in = 0,
	.voice_size_out = sizeof (SDLVoiceOut),
	.voice_size_in = 0
	};