symbian-qemu-0.9.1-12/libsdl-trunk/src/audio/alsa/SDL_alsa_audio.c
changeset 1 2fb8b9db1c86
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/symbian-qemu-0.9.1-12/libsdl-trunk/src/audio/alsa/SDL_alsa_audio.c	Fri Jul 31 15:01:17 2009 +0100
@@ -0,0 +1,538 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2004 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Library General Public
+    License as published by the Free Software Foundation; either
+    version 2 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Library General Public License for more details.
+
+    You should have received a copy of the GNU Library General Public
+    License along with this library; if not, write to the Free
+    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Allow access to a raw mixing buffer */
+
+#include <sys/types.h>
+#include <signal.h>	/* For kill() */
+
+#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 <dlfcn.h>
+#include "SDL_name.h"
+#include "SDL_loadso.h"
+#else
+#define SDL_NAME(X)	X
+#endif
+
+
+/* The tag name used by ALSA audio */
+#define DRIVER_NAME         "alsa"
+
+/* The default ALSA audio driver */
+#define DEFAULT_DEVICE	"default"
+
+/* Audio driver functions */
+static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec);
+static void ALSA_WaitAudio(_THIS);
+static void ALSA_PlayAudio(_THIS);
+static Uint8 *ALSA_GetAudioBuf(_THIS);
+static void ALSA_CloseAudio(_THIS);
+
+#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
+
+static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC;
+static void *alsa_handle = NULL;
+static int alsa_loaded = 0;
+
+static int (*SDL_snd_pcm_open)(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode);
+static int (*SDL_NAME(snd_pcm_open))(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode);
+static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm);
+static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_writei))(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size);
+static int (*SDL_NAME(snd_pcm_resume))(snd_pcm_t *pcm);
+static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm);
+static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm);
+static const char *(*SDL_NAME(snd_strerror))(int errnum);
+static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void);
+static size_t (*SDL_NAME(snd_pcm_sw_params_sizeof))(void);
+static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
+static int (*SDL_NAME(snd_pcm_hw_params_set_access))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t access);
+static int (*SDL_NAME(snd_pcm_hw_params_set_format))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val);
+static int (*SDL_NAME(snd_pcm_hw_params_set_channels))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val);
+static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params);
+static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_rate_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir);
+static snd_pcm_uframes_t (*SDL_NAME(snd_pcm_hw_params_set_period_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t val, int *dir);
+static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_hw_params_get_period_size))(const snd_pcm_hw_params_t *params);
+static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_periods_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir);
+static int (*SDL_NAME(snd_pcm_hw_params_get_periods))(snd_pcm_hw_params_t *params);
+static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
+/*
+*/
+static int (*SDL_NAME(snd_pcm_sw_params_current))(snd_pcm_t *pcm, snd_pcm_sw_params_t *swparams);
+static int (*SDL_NAME(snd_pcm_sw_params_set_start_threshold))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);
+static int (*SDL_NAME(snd_pcm_sw_params_set_avail_min))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);
+static int (*SDL_NAME(snd_pcm_sw_params))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params);
+static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock);
+#define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof)
+#define snd_pcm_sw_params_sizeof SDL_NAME(snd_pcm_sw_params_sizeof)
+
+/* cast funcs to char* first, to please GCC's strict aliasing rules. */
+static struct {
+	const char *name;
+	void **func;
+} alsa_functions[] = {
+	{ "snd_pcm_open",	(void**)(char*)&SDL_NAME(snd_pcm_open)		},
+	{ "snd_pcm_close",	(void**)(char*)&SDL_NAME(snd_pcm_close)	},
+	{ "snd_pcm_writei",	(void**)(char*)&SDL_NAME(snd_pcm_writei)	},
+	{ "snd_pcm_resume",	(void**)(char*)&SDL_NAME(snd_pcm_resume)	},
+	{ "snd_pcm_prepare",	(void**)(char*)&SDL_NAME(snd_pcm_prepare)	},
+	{ "snd_pcm_drain",	(void**)(char*)&SDL_NAME(snd_pcm_drain)	},
+	{ "snd_strerror",	(void**)(char*)&SDL_NAME(snd_strerror)		},
+	{ "snd_pcm_hw_params_sizeof",		(void**)(char*)&SDL_NAME(snd_pcm_hw_params_sizeof)		},
+	{ "snd_pcm_sw_params_sizeof",		(void**)(char*)&SDL_NAME(snd_pcm_sw_params_sizeof)		},
+	{ "snd_pcm_hw_params_any",		(void**)(char*)&SDL_NAME(snd_pcm_hw_params_any)		},
+	{ "snd_pcm_hw_params_set_access",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_access)		},
+	{ "snd_pcm_hw_params_set_format",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_format)		},
+	{ "snd_pcm_hw_params_set_channels",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_channels)	},
+	{ "snd_pcm_hw_params_get_channels",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_channels)	},
+	{ "snd_pcm_hw_params_set_rate_near",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_rate_near)	},
+	{ "snd_pcm_hw_params_set_period_size_near",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_period_size_near)	},
+	{ "snd_pcm_hw_params_get_period_size",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_period_size)	},
+	{ "snd_pcm_hw_params_set_periods_near",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_periods_near)	},
+	{ "snd_pcm_hw_params_get_periods",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_periods)	},
+	{ "snd_pcm_hw_params",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params)	},
+	{ "snd_pcm_sw_params_current",	(void**)(char*)&SDL_NAME(snd_pcm_sw_params_current)	},
+	{ "snd_pcm_sw_params_set_start_threshold",	(void**)(char*)&SDL_NAME(snd_pcm_sw_params_set_start_threshold)	},
+	{ "snd_pcm_sw_params_set_avail_min",	(void**)(char*)&SDL_NAME(snd_pcm_sw_params_set_avail_min)	},
+	{ "snd_pcm_sw_params",	(void**)(char*)&SDL_NAME(snd_pcm_sw_params)	},
+	{ "snd_pcm_nonblock",	(void**)(char*)&SDL_NAME(snd_pcm_nonblock)	},
+};
+
+static void UnloadALSALibrary(void) {
+	if (alsa_loaded) {
+/*		SDL_UnloadObject(alsa_handle);*/
+		dlclose(alsa_handle);
+		alsa_handle = NULL;
+		alsa_loaded = 0;
+	}
+}
+
+static int LoadALSALibrary(void) {
+	int i, retval = -1;
+
+/*	alsa_handle = SDL_LoadObject(alsa_library);*/
+	alsa_handle = dlopen(alsa_library,RTLD_NOW);
+	if (alsa_handle) {
+		alsa_loaded = 1;
+		retval = 0;
+		for (i = 0; i < SDL_arraysize(alsa_functions); i++) {
+/*			*alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);*/
+#if HAVE_DLVSYM
+			*alsa_functions[i].func = dlvsym(alsa_handle,alsa_functions[i].name,"ALSA_0.9");
+			if (!*alsa_functions[i].func)
+#endif
+				*alsa_functions[i].func = dlsym(alsa_handle,alsa_functions[i].name);
+			if (!*alsa_functions[i].func) {
+				retval = -1;
+				UnloadALSALibrary();
+				break;
+			}
+		}
+	}
+	return retval;
+}
+
+#else
+
+static void UnloadALSALibrary(void) {
+	return;
+}
+
+static int LoadALSALibrary(void) {
+	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 ) {
+		if (channels == 6) device = "surround51";
+		else if (channels == 4) device = "surround40";
+		else device = DEFAULT_DEVICE;
+	}
+	return device;
+}
+
+/* Audio driver bootstrap functions */
+
+static int Audio_Available(void)
+{
+	int available;
+	int status;
+	snd_pcm_t *handle;
+
+	available = 0;
+	if (LoadALSALibrary() < 0) {
+		return available;
+	}
+	status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(2), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+	if ( status >= 0 ) {
+		available = 1;
+        	SDL_NAME(snd_pcm_close)(handle);
+	}
+	UnloadALSALibrary();
+	return(available);
+}
+
+static void Audio_DeleteDevice(SDL_AudioDevice *device)
+{
+	SDL_free(device->hidden);
+	SDL_free(device);
+	UnloadALSALibrary();
+}
+
+static SDL_AudioDevice *Audio_CreateDevice(int devindex)
+{
+	SDL_AudioDevice *this;
+
+	/* Initialize all variables that we clean on shutdown */
+	LoadALSALibrary();
+	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
+	if ( this ) {
+		SDL_memset(this, 0, (sizeof *this));
+		this->hidden = (struct SDL_PrivateAudioData *)
+				SDL_malloc((sizeof *this->hidden));
+	}
+	if ( (this == NULL) || (this->hidden == NULL) ) {
+		SDL_OutOfMemory();
+		if ( this ) {
+			SDL_free(this);
+		}
+		return(0);
+	}
+	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
+
+	/* Set the function pointers */
+	this->OpenAudio = ALSA_OpenAudio;
+	this->WaitAudio = ALSA_WaitAudio;
+	this->PlayAudio = ALSA_PlayAudio;
+	this->GetAudioBuf = ALSA_GetAudioBuf;
+	this->CloseAudio = ALSA_CloseAudio;
+
+	this->free = Audio_DeleteDevice;
+
+	return this;
+}
+
+AudioBootStrap ALSA_bootstrap = {
+	DRIVER_NAME, "ALSA 0.9 PCM audio",
+	Audio_Available, Audio_CreateDevice
+};
+
+/* This function waits until it is possible to write a full sound buffer */
+static void ALSA_WaitAudio(_THIS)
+{
+	/* Check to see if the thread-parent process is still alive */
+	{ static int cnt = 0;
+		/* Note that this only works with thread implementations 
+		   that use a different process id for each thread.
+		*/
+		if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */
+			if ( kill(parent, 0) < 0 ) {
+				this->enabled = 0;
+			}
+		}
+	}
+}
+
+
+/*
+ * http://bugzilla.libsdl.org/show_bug.cgi?id=110
+ * "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
+ *  and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
+ */
+#define SWIZ6(T) \
+    T *ptr = (T *) mixbuf; \
+    const Uint32 count = (this->spec.samples / 6); \
+    Uint32 i; \
+    for (i = 0; i < count; i++, ptr += 6) { \
+        T tmp; \
+        tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \
+        tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \
+    }
+
+static __inline__ void swizzle_alsa_channels_6_64bit(_THIS) { SWIZ6(Uint64); }
+static __inline__ void swizzle_alsa_channels_6_32bit(_THIS) { SWIZ6(Uint32); }
+static __inline__ void swizzle_alsa_channels_6_16bit(_THIS) { SWIZ6(Uint16); }
+static __inline__ void swizzle_alsa_channels_6_8bit(_THIS) { SWIZ6(Uint8); }
+
+#undef SWIZ6
+
+
+/*
+ * Called right before feeding this->mixbuf to the hardware. Swizzle channels
+ *  from Windows/Mac order to the format alsalib will want.
+ */
+static __inline__ void swizzle_alsa_channels(_THIS)
+{
+    if (this->spec.channels == 6) {
+        const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */
+        if (fmtsize == 16)
+            swizzle_alsa_channels_6_16bit(this);
+        else if (fmtsize == 8)
+            swizzle_alsa_channels_6_8bit(this);
+        else if (fmtsize == 32)
+            swizzle_alsa_channels_6_32bit(this);
+        else if (fmtsize == 64)
+            swizzle_alsa_channels_6_64bit(this);
+    }
+
+    /* !!! FIXME: update this for 7.1 if needed, later. */
+}
+
+
+static void ALSA_PlayAudio(_THIS)
+{
+	int           status;
+	int           sample_len;
+	signed short *sample_buf;
+
+	swizzle_alsa_channels(this);
+
+	sample_len = this->spec.samples;
+	sample_buf = (signed short *)mixbuf;
+
+	while ( sample_len > 0 ) {
+		status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len);
+		if ( status < 0 ) {
+			if ( status == -EAGAIN ) {
+				SDL_Delay(1);
+				continue;
+			}
+			if ( status == -ESTRPIPE ) {
+				do {
+					SDL_Delay(1);
+					status = SDL_NAME(snd_pcm_resume)(pcm_handle);
+				} while ( status == -EAGAIN );
+			}
+			if ( status < 0 ) {
+				status = SDL_NAME(snd_pcm_prepare)(pcm_handle);
+			}
+			if ( status < 0 ) {
+				/* Hmm, not much we can do - abort */
+				this->enabled = 0;
+				return;
+			}
+			continue;
+		}
+		sample_buf += status * this->spec.channels;
+		sample_len -= status;
+	}
+}
+
+static Uint8 *ALSA_GetAudioBuf(_THIS)
+{
+	return(mixbuf);
+}
+
+static void ALSA_CloseAudio(_THIS)
+{
+	if ( mixbuf != NULL ) {
+		SDL_FreeAudioMem(mixbuf);
+		mixbuf = NULL;
+	}
+	if ( pcm_handle ) {
+		SDL_NAME(snd_pcm_drain)(pcm_handle);
+		SDL_NAME(snd_pcm_close)(pcm_handle);
+		pcm_handle = NULL;
+	}
+}
+
+static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
+{
+	int                  status;
+	snd_pcm_hw_params_t *hwparams;
+	snd_pcm_sw_params_t *swparams;
+	snd_pcm_format_t     format;
+	snd_pcm_uframes_t    frames;
+	Uint16               test_format;
+
+	/* Open the audio device */
+	/* Name of device should depend on # channels in spec */
+	status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+
+	if ( status < 0 ) {
+		SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status));
+		return(-1);
+	}
+
+	/* Figure out what the hardware is capable of */
+	snd_pcm_hw_params_alloca(&hwparams);
+	status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, hwparams);
+	if ( status < 0 ) {
+		SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status));
+		ALSA_CloseAudio(this);
+		return(-1);
+	}
+
+	/* SDL only uses interleaved sample output */
+	status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
+	if ( status < 0 ) {
+		SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status));
+		ALSA_CloseAudio(this);
+		return(-1);
+	}
+
+	/* Try for a closest match on audio format */
+	status = -1;
+	for ( test_format = SDL_FirstAudioFormat(spec->format);
+	      test_format && (status < 0); ) {
+		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;
+			default:
+				format = 0;
+				break;
+		}
+		if ( format != 0 ) {
+			status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, hwparams, format);
+		}
+		if ( status < 0 ) {
+			test_format = SDL_NextAudioFormat();
+		}
+	}
+	if ( status < 0 ) {
+		SDL_SetError("Couldn't find any hardware audio formats");
+		ALSA_CloseAudio(this);
+		return(-1);
+	}
+	spec->format = test_format;
+
+	/* Set the number of channels */
+	status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, hwparams, spec->channels);
+	if ( status < 0 ) {
+		status = SDL_NAME(snd_pcm_hw_params_get_channels)(hwparams);
+		if ( (status <= 0) || (status > 2) ) {
+			SDL_SetError("Couldn't set audio channels");
+			ALSA_CloseAudio(this);
+			return(-1);
+		}
+		spec->channels = status;
+	}
+
+	/* Set the audio rate */
+	status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, spec->freq, NULL);
+	if ( status < 0 ) {
+		SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status));
+		ALSA_CloseAudio(this);
+		return(-1);
+	}
+	spec->freq = status;
+
+	/* Set the buffer size, in samples */
+	frames = spec->samples;
+	frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, frames, NULL);
+	spec->samples = frames;
+	SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, 2, NULL);
+
+	/* "set" the hardware with the desired parameters */
+	status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, hwparams);
+	if ( status < 0 ) {
+		SDL_SetError("Couldn't set hardware audio parameters: %s", SDL_NAME(snd_strerror)(status));
+		ALSA_CloseAudio(this);
+		return(-1);
+	}
+
+/* This is useful for debugging... */
+/*
+{ snd_pcm_sframes_t bufsize; int fragments;
+   bufsize = SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams);
+   fragments = SDL_NAME(snd_pcm_hw_params_get_periods)(hwparams);
+
+   fprintf(stderr, "ALSA: bufsize = %ld, fragments = %d\n", bufsize, fragments);
+}
+*/
+
+	/* Set the software parameters */
+	snd_pcm_sw_params_alloca(&swparams);
+	status = SDL_NAME(snd_pcm_sw_params_current)(pcm_handle, swparams);
+	if ( status < 0 ) {
+		SDL_SetError("Couldn't get software config: %s", SDL_NAME(snd_strerror)(status));
+		ALSA_CloseAudio(this);
+		return(-1);
+	}
+	status = SDL_NAME(snd_pcm_sw_params_set_start_threshold)(pcm_handle, swparams, 0);
+	if ( status < 0 ) {
+		SDL_SetError("Couldn't set start threshold: %s", SDL_NAME(snd_strerror)(status));
+		ALSA_CloseAudio(this);
+		return(-1);
+	}
+	status = SDL_NAME(snd_pcm_sw_params_set_avail_min)(pcm_handle, swparams, frames);
+	if ( status < 0 ) {
+		SDL_SetError("Couldn't set avail min: %s", SDL_NAME(snd_strerror)(status));
+		ALSA_CloseAudio(this);
+		return(-1);
+	}
+	status = SDL_NAME(snd_pcm_sw_params)(pcm_handle, swparams);
+	if ( status < 0 ) {
+		SDL_SetError("Couldn't set software audio parameters: %s", SDL_NAME(snd_strerror)(status));
+		ALSA_CloseAudio(this);
+		return(-1);
+	}
+
+	/* Calculate the final parameters for this audio specification */
+	SDL_CalculateAudioSpec(spec);
+
+	/* Allocate mixing buffer */
+	mixlen = spec->size;
+	mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
+	if ( mixbuf == NULL ) {
+		ALSA_CloseAudio(this);
+		return(-1);
+	}
+	SDL_memset(mixbuf, spec->silence, spec->size);
+
+	/* Get the parent process id (we're the parent of the audio thread) */
+	parent = getpid();
+
+	/* Switch to blocking mode for playback */
+	SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0);
+
+	/* We're ready to rock and roll. :-) */
+	return(0);
+}