 ed2a4a7941
			
		
	
	
		ed2a4a7941
		
	
	
	
	
		
			
			This adds proper support for float samples in mixeng by adding a new audio format for it. Limitations: only native endianness is supported. None of the virtual sound cards support float samples (it looks like most of them only support 8 and 16 bit, only hda supports 32 bit), it is only used for the audio backends (i.e. host side). Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Acked-by: Markus Armbruster <armbru@redhat.com> Message-id: 8a8b0b5698401b78d3c4c8ec90aef83b95babb06.1580672076.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
		
			
				
	
	
		
			378 lines
		
	
	
		
			9.6 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			378 lines
		
	
	
		
			9.6 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
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|  * QEMU SDL audio driver
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|  *
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|  * Copyright (c) 2004-2005 Vassili Karpov (malc)
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|  *
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|  * Permission is hereby granted, free of charge, to any person obtaining a copy
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|  * of this software and associated documentation files (the "Software"), to deal
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|  * in the Software without restriction, including without limitation the rights
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|  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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|  * copies of the Software, and to permit persons to whom the Software is
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|  * furnished to do so, subject to the following conditions:
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|  *
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|  * The above copyright notice and this permission notice shall be included in
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|  * all copies or substantial portions of the Software.
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|  *
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|  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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|  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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|  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
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|  * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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|  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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|  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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|  * THE SOFTWARE.
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|  */
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| 
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| #include "qemu/osdep.h"
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| #include <SDL.h>
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| #include <SDL_thread.h>
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| #include "qemu/module.h"
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| #include "audio.h"
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| 
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| #ifndef _WIN32
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| #ifdef __sun__
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| #define _POSIX_PTHREAD_SEMANTICS 1
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| #elif defined(__OpenBSD__) || defined(__FreeBSD__) || defined(__DragonFly__)
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| #include <pthread.h>
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| #endif
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| #endif
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| 
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| #define AUDIO_CAP "sdl"
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| #include "audio_int.h"
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| 
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| typedef struct SDLVoiceOut {
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|     HWVoiceOut hw;
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| } SDLVoiceOut;
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| 
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| static struct SDLAudioState {
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|     int exit;
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|     int initialized;
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|     bool driver_created;
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|     Audiodev *dev;
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| } glob_sdl;
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| typedef struct SDLAudioState SDLAudioState;
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| 
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| static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
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| {
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|     va_list ap;
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| 
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|     va_start (ap, fmt);
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|     AUD_vlog (AUDIO_CAP, fmt, ap);
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|     va_end (ap);
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| 
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|     AUD_log (AUDIO_CAP, "Reason: %s\n", SDL_GetError ());
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| }
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| 
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| static int aud_to_sdlfmt (AudioFormat fmt)
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| {
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|     switch (fmt) {
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|     case AUDIO_FORMAT_S8:
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|         return AUDIO_S8;
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| 
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|     case AUDIO_FORMAT_U8:
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|         return AUDIO_U8;
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| 
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|     case AUDIO_FORMAT_S16:
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|         return AUDIO_S16LSB;
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| 
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|     case AUDIO_FORMAT_U16:
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|         return AUDIO_U16LSB;
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| 
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|     case AUDIO_FORMAT_S32:
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|         return AUDIO_S32LSB;
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| 
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|     /* no unsigned 32-bit support in SDL */
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| 
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|     case AUDIO_FORMAT_F32:
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|         return AUDIO_F32LSB;
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| 
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|     default:
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|         dolog ("Internal logic error: Bad audio format %d\n", fmt);
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| #ifdef DEBUG_AUDIO
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|         abort ();
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| #endif
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|         return AUDIO_U8;
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|     }
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| }
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| 
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| static int sdl_to_audfmt(int sdlfmt, AudioFormat *fmt, int *endianness)
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| {
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|     switch (sdlfmt) {
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|     case AUDIO_S8:
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|         *endianness = 0;
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|         *fmt = AUDIO_FORMAT_S8;
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|         break;
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| 
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|     case AUDIO_U8:
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|         *endianness = 0;
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|         *fmt = AUDIO_FORMAT_U8;
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|         break;
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| 
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|     case AUDIO_S16LSB:
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|         *endianness = 0;
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|         *fmt = AUDIO_FORMAT_S16;
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|         break;
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| 
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|     case AUDIO_U16LSB:
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|         *endianness = 0;
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|         *fmt = AUDIO_FORMAT_U16;
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|         break;
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| 
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|     case AUDIO_S16MSB:
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|         *endianness = 1;
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|         *fmt = AUDIO_FORMAT_S16;
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|         break;
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| 
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|     case AUDIO_U16MSB:
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|         *endianness = 1;
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|         *fmt = AUDIO_FORMAT_U16;
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|         break;
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| 
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|     case AUDIO_S32LSB:
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|         *endianness = 0;
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|         *fmt = AUDIO_FORMAT_S32;
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|         break;
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| 
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|     case AUDIO_S32MSB:
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|         *endianness = 1;
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|         *fmt = AUDIO_FORMAT_S32;
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|         break;
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| 
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|     case AUDIO_F32LSB:
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|         *endianness = 0;
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|         *fmt = AUDIO_FORMAT_F32;
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|         break;
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| 
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|     case AUDIO_F32MSB:
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|         *endianness = 1;
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|         *fmt = AUDIO_FORMAT_F32;
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|         break;
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| 
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|     default:
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|         dolog ("Unrecognized SDL audio format %d\n", sdlfmt);
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|         return -1;
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|     }
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| 
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|     return 0;
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| }
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| 
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| static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
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| {
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|     int status;
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| #ifndef _WIN32
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|     int err;
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|     sigset_t new, old;
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| 
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|     /* Make sure potential threads created by SDL don't hog signals.  */
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|     err = sigfillset (&new);
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|     if (err) {
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|         dolog ("sdl_open: sigfillset failed: %s\n", strerror (errno));
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|         return -1;
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|     }
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|     err = pthread_sigmask (SIG_BLOCK, &new, &old);
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|     if (err) {
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|         dolog ("sdl_open: pthread_sigmask failed: %s\n", strerror (err));
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|         return -1;
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|     }
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| #endif
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| 
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|     status = SDL_OpenAudio (req, obt);
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|     if (status) {
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|         sdl_logerr ("SDL_OpenAudio failed\n");
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|     }
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| 
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| #ifndef _WIN32
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|     err = pthread_sigmask (SIG_SETMASK, &old, NULL);
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|     if (err) {
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|         dolog ("sdl_open: pthread_sigmask (restore) failed: %s\n",
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|                strerror (errno));
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|         /* We have failed to restore original signal mask, all bets are off,
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|            so exit the process */
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|         exit (EXIT_FAILURE);
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|     }
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| #endif
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|     return status;
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| }
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| 
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| static void sdl_close (SDLAudioState *s)
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| {
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|     if (s->initialized) {
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|         SDL_LockAudio();
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|         s->exit = 1;
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|         SDL_UnlockAudio();
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|         SDL_PauseAudio (1);
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|         SDL_CloseAudio ();
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|         s->initialized = 0;
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|     }
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| }
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| 
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| static void sdl_callback (void *opaque, Uint8 *buf, int len)
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| {
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|     SDLVoiceOut *sdl = opaque;
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|     SDLAudioState *s = &glob_sdl;
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|     HWVoiceOut *hw = &sdl->hw;
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| 
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|     if (s->exit) {
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|         return;
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|     }
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| 
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|     /* dolog ("in callback samples=%zu live=%zu\n", samples, sdl->live); */
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| 
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|     while (hw->pending_emul && len) {
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|         size_t write_len;
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|         ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
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|         if (start < 0) {
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|             start += hw->size_emul;
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|         }
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|         assert(start >= 0 && start < hw->size_emul);
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| 
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|         write_len = MIN(MIN(hw->pending_emul, len),
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|                         hw->size_emul - start);
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| 
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|         memcpy(buf, hw->buf_emul + start, write_len);
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|         hw->pending_emul -= write_len;
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|         len -= write_len;
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|         buf += write_len;
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|     }
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| 
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|     /* clear remaining buffer that we couldn't fill with data */
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|     if (len) {
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|         memset(buf, 0, len);
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|     }
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| }
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| 
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| #define SDL_WRAPPER_FUNC(name, ret_type, args_decl, args, fail, unlock) \
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|     static ret_type glue(sdl_, name)args_decl                           \
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|     {                                                                   \
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|         ret_type ret;                                                   \
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|                                                                         \
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|         SDL_LockAudio();                                                \
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|                                                                         \
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|         ret = glue(audio_generic_, name)args;                           \
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|                                                                         \
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|         SDL_UnlockAudio();                                              \
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|         return ret;                                                     \
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|     }
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| 
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| SDL_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size),
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|                  (hw, size), *size = 0, sdl_unlock)
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| SDL_WRAPPER_FUNC(put_buffer_out, size_t,
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|                  (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size),
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|                  /*nothing*/, sdl_unlock_and_post)
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| SDL_WRAPPER_FUNC(write, size_t,
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|                  (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size),
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|                  /*nothing*/, sdl_unlock_and_post)
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| 
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| #undef SDL_WRAPPER_FUNC
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| 
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| static void sdl_fini_out (HWVoiceOut *hw)
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| {
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|     (void) hw;
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| 
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|     sdl_close (&glob_sdl);
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| }
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| 
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| static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
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|                         void *drv_opaque)
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| {
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|     SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
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|     SDLAudioState *s = &glob_sdl;
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|     SDL_AudioSpec req, obt;
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|     int endianness;
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|     int err;
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|     AudioFormat effective_fmt;
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|     struct audsettings obt_as;
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| 
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|     req.freq = as->freq;
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|     req.format = aud_to_sdlfmt (as->fmt);
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|     req.channels = as->nchannels;
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|     req.samples = audio_buffer_samples(s->dev->u.sdl.out, as, 11610);
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|     req.callback = sdl_callback;
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|     req.userdata = sdl;
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| 
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|     if (sdl_open (&req, &obt)) {
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|         return -1;
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|     }
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| 
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|     err = sdl_to_audfmt(obt.format, &effective_fmt, &endianness);
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|     if (err) {
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|         sdl_close (s);
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|         return -1;
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|     }
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| 
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|     obt_as.freq = obt.freq;
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|     obt_as.nchannels = obt.channels;
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|     obt_as.fmt = effective_fmt;
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|     obt_as.endianness = endianness;
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| 
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|     audio_pcm_init_info (&hw->info, &obt_as);
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|     hw->samples = obt.samples;
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| 
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|     s->initialized = 1;
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|     s->exit = 0;
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|     SDL_PauseAudio (0);
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|     return 0;
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| }
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| 
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| static void sdl_enable_out(HWVoiceOut *hw, bool enable)
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| {
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|     SDL_PauseAudio(!enable);
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| }
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| 
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| static void *sdl_audio_init(Audiodev *dev)
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| {
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|     SDLAudioState *s = &glob_sdl;
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|     if (s->driver_created) {
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|         sdl_logerr("Can't create multiple sdl backends\n");
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|         return NULL;
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|     }
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| 
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|     if (SDL_InitSubSystem (SDL_INIT_AUDIO)) {
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|         sdl_logerr ("SDL failed to initialize audio subsystem\n");
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|         return NULL;
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|     }
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| 
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|     s->driver_created = true;
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|     s->dev = dev;
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|     return s;
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| }
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| 
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| static void sdl_audio_fini (void *opaque)
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| {
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|     SDLAudioState *s = opaque;
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|     sdl_close (s);
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|     SDL_QuitSubSystem (SDL_INIT_AUDIO);
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|     s->driver_created = false;
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|     s->dev = NULL;
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| }
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| 
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| static struct audio_pcm_ops sdl_pcm_ops = {
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|     .init_out = sdl_init_out,
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|     .fini_out = sdl_fini_out,
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|   /* wrapper for audio_generic_write */
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|     .write    = sdl_write,
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|   /* wrapper for audio_generic_get_buffer_out */
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|     .get_buffer_out = sdl_get_buffer_out,
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|   /* wrapper for audio_generic_put_buffer_out */
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|     .put_buffer_out = sdl_put_buffer_out,
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|     .enable_out = sdl_enable_out,
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| };
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| 
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| static struct audio_driver sdl_audio_driver = {
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|     .name           = "sdl",
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|     .descr          = "SDL http://www.libsdl.org",
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|     .init           = sdl_audio_init,
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|     .fini           = sdl_audio_fini,
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|     .pcm_ops        = &sdl_pcm_ops,
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|     .can_be_default = 1,
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|     .max_voices_out = 1,
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|     .max_voices_in  = 0,
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|     .voice_size_out = sizeof (SDLVoiceOut),
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|     .voice_size_in  = 0
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| };
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| 
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| static void register_audio_sdl(void)
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| {
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|     audio_driver_register(&sdl_audio_driver);
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| }
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| type_init(register_audio_sdl);
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