ext/sdl/sdlaudiosink.c - mtk-gst-plugins-bad-debian - Git at Google

 /* GStreamer
  * Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
  *
  * 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
  */

 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif

 #include "sdlaudiosink.h"

 #include <SDL_byteorder.h>
 #include <string.h>

 #include <unistd.h>

 #include <gst/glib-compat-private.h>

 GST_DEBUG_CATEGORY_EXTERN (sdl_debug);
 #define GST_CAT_DEFAULT sdl_debug

 static void gst_sdlaudio_sink_dispose (GObject * object);

 static GstCaps *gst_sdlaudio_sink_getcaps (GstBaseSink * bsink);

 static gboolean gst_sdlaudio_sink_open (GstAudioSink * asink);
 static gboolean gst_sdlaudio_sink_close (GstAudioSink * asink);
 static gboolean gst_sdlaudio_sink_prepare (GstAudioSink * asink,
     GstRingBufferSpec * spec);
 static gboolean gst_sdlaudio_sink_unprepare (GstAudioSink * asink);
 static guint gst_sdlaudio_sink_write (GstAudioSink * asink, gpointer data,
     guint length);

 #if 0
 static guint gst_sdlaudio_sink_delay (GstAudioSink * asink);
 static void gst_sdlaudio_sink_reset (GstAudioSink * asink);
 #endif


 /* SdlaudioSink signals and args */
 enum
 {
   LAST_SIGNAL
 };

 #define  SEMAPHORE_INIT(s,f)			\
   do {						\
     s.cond = g_cond_new();			\
     s.mutex = g_mutex_new();			\
     s.mutexflag = f;				\
   } while(0)

 #define  SEMAPHORE_CLOSE(s)			\
   do {						\
     if ( s.cond ) {				\
       g_cond_free(s.cond);			\
       s.cond = NULL;				\
     }						\
     if ( s.mutex ) {				\
       g_mutex_free(s.mutex);			\
       s.mutex = NULL;				\
     }						\
   } while(0)

 #define SEMAPHORE_UP(s)					\
   do							\
     {							\
       g_mutex_lock(s.mutex);				\
       s.mutexflag = TRUE;				\
       g_mutex_unlock(s.mutex);				\
       g_cond_signal(s.cond);				\
     } while(0)

 #define SEMAPHORE_DOWN(s, e)				\
   do							\
     {							\
       while (1) {					\
 	g_mutex_lock(s.mutex);				\
 	if (!s.mutexflag) {				\
 	  if ( e ) {					\
 	    g_mutex_unlock(s.mutex);			\
 	    break;					\
 	  }						\
 	  g_cond_wait(s.cond,s.mutex);			\
 	}						\
 	else {						\
 	  s.mutexflag = FALSE;				\
 	  g_mutex_unlock(s.mutex);			\
 	  break;					\
 	}						\
 	g_mutex_unlock(s.mutex);			\
       }							\
     } while(0)


 static GstStaticPadTemplate sdlaudiosink_sink_factory =
     GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw-int, "
         "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
         "signed = (boolean) { TRUE, FALSE }, "
         "width = (int) 16, "
         "depth = (int) 16, "
         "rate = (int) [ 1, MAX ], "
         "channels = (int) [ 1, 2 ]; "
         "audio/x-raw-int, "
         "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
         "signed = (boolean) { TRUE, FALSE }, "
         "width = (int) 8, "
         "depth = (int) 8, "
         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
     );

 GST_BOILERPLATE (GstSDLAudioSink, gst_sdlaudio_sink, GstAudioSink,
     GST_TYPE_AUDIO_SINK);

 static void
 gst_sdlaudio_sink_dispose (GObject * object)
 {
   GstSDLAudioSink *sdlaudiosink = GST_SDLAUDIOSINK (object);

   SEMAPHORE_CLOSE (sdlaudiosink->semB);

   SEMAPHORE_CLOSE (sdlaudiosink->semA);

   if (sdlaudiosink->buffer) {
     g_free (sdlaudiosink->buffer);
   }

   G_OBJECT_CLASS (parent_class)->dispose (object);

 }

 static void
 gst_sdlaudio_sink_base_init (gpointer g_class)
 {
   GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);

   gst_element_class_set_static_metadata (element_class, "SDL audio sink",
       "Sink/Audio",
       "Output to a sound card via SDLAUDIO",
       "Edgard Lima <edgard.lima@indt.org.br>");

   gst_element_class_add_static_pad_template (element_class,
       &sdlaudiosink_sink_factory);
 }

 static void
 gst_sdlaudio_sink_class_init (GstSDLAudioSinkClass * klass)
 {
   GObjectClass *gobject_class;
   GstBaseSinkClass *gstbasesink_class;
   GstAudioSinkClass *gstaudiosink_class;

   gobject_class = (GObjectClass *) klass;
   gstbasesink_class = (GstBaseSinkClass *) klass;
   gstaudiosink_class = (GstAudioSinkClass *) klass;

   gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_dispose);

   gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_getcaps);

   gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_open);
   gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_close);
   gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_prepare);
   gstaudiosink_class->unprepare =
       GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_unprepare);
   gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_write);

 #if 0
   gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_delay);
   gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_reset);
 #endif

 }

 static void
 gst_sdlaudio_sink_init (GstSDLAudioSink * sdlaudiosink,
     GstSDLAudioSinkClass * g_class)
 {
   GST_DEBUG ("initializing sdlaudiosink");

   memset (&sdlaudiosink->fmt, 0, sizeof (SDL_AudioSpec));

   sdlaudiosink->buffer = NULL;
   sdlaudiosink->eos = FALSE;

   SEMAPHORE_INIT (sdlaudiosink->semA, TRUE);

   SEMAPHORE_INIT (sdlaudiosink->semB, FALSE);

 }

 static GstCaps *
 gst_sdlaudio_sink_getcaps (GstBaseSink * bsink)
 {
   return gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD
           (bsink)));
 }

 static gint
 gst_sdlaudio_sink_get_format (GstBufferFormat fmt)
 {
   gint result = GST_UNKNOWN;

   switch (fmt) {
     case GST_U8:
       result = AUDIO_U8;
       break;
     case GST_S8:
       result = AUDIO_S8;
       break;
     case GST_S16_LE:
       result = AUDIO_S16LSB;
       break;
     case GST_S16_BE:
       result = AUDIO_S16MSB;
       break;
     case GST_U16_LE:
       result = AUDIO_U16LSB;
       break;
     case GST_U16_BE:
       result = AUDIO_U16MSB;
       break;
     default:
       break;
   }
   return result;
 }

 static gboolean
 gst_sdlaudio_sink_open (GstAudioSink * asink)
 {
   GstSDLAudioSink *sdlaudio;

   sdlaudio = GST_SDLAUDIOSINK (asink);

   if (SDL_Init (SDL_INIT_AUDIO) < 0) {
     goto open_failed;
   }

   return TRUE;

 open_failed:
   {
     GST_ELEMENT_ERROR (sdlaudio, LIBRARY, INIT,
         ("Unable to init SDL: %s\n", SDL_GetError ()), (NULL));
     return FALSE;
   }
 }

 static gboolean
 gst_sdlaudio_sink_close (GstAudioSink * asink)
 {
   GstSDLAudioSink *sdlaudio = GST_SDLAUDIOSINK (asink);

   sdlaudio->eos = TRUE;
   SEMAPHORE_UP (sdlaudio->semA);
   SEMAPHORE_UP (sdlaudio->semB);
   SDL_QuitSubSystem (SDL_INIT_AUDIO);
   return TRUE;
 }

 static guint
 gst_sdlaudio_sink_write (GstAudioSink * asink, gpointer data, guint length)
 {
   GstSDLAudioSink *sdlaudio = GST_SDLAUDIOSINK (asink);

   if (sdlaudio->fmt.size != length) {
     GST_ERROR ("ring buffer segment length (%u) != sdl buffer len (%u)", length,
         sdlaudio->fmt.size);
   }

   SEMAPHORE_DOWN (sdlaudio->semA, sdlaudio->eos);

   if (!sdlaudio->eos)
     memcpy (sdlaudio->buffer, data, length);

   SEMAPHORE_UP (sdlaudio->semB);

   return sdlaudio->fmt.size;
 }


 static void
 mixaudio (void *unused, Uint8 * stream, int len)
 {
   GstSDLAudioSink *sdlaudio;

   sdlaudio = GST_SDLAUDIOSINK (unused);

   if (sdlaudio->fmt.size != len) {
     GST_ERROR ("fmt buffer len (%u) != sdl callback len (%d)",
         sdlaudio->fmt.size, len);
   }

   SEMAPHORE_DOWN (sdlaudio->semB, sdlaudio->eos);

   if (!sdlaudio->eos)
     SDL_MixAudio (stream, sdlaudio->buffer, sdlaudio->fmt.size,
         SDL_MIX_MAXVOLUME);

   SEMAPHORE_UP (sdlaudio->semA);

 }

 static gboolean
 gst_sdlaudio_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
 {
   GstSDLAudioSink *sdlaudio;
   gint power2 = -1;

   sdlaudio = GST_SDLAUDIOSINK (asink);

   sdlaudio->fmt.format = gst_sdlaudio_sink_get_format (spec->format);
   if (sdlaudio->fmt.format == 0)
     goto wrong_format;

   if (spec->width != 16 && spec->width != 8)
     goto dodgy_width;

   sdlaudio->fmt.freq = spec->rate;
   sdlaudio->fmt.channels = spec->channels;
   sdlaudio->fmt.samples =
       spec->segsize / (spec->channels * ((sdlaudio->fmt.format & 0xFF) >> 3));
   sdlaudio->fmt.callback = mixaudio;
   sdlaudio->fmt.userdata = sdlaudio;

   GST_DEBUG ("set segsize: %d, segtotal: %d, samples: %d", spec->segsize,
       spec->segtotal, sdlaudio->fmt.samples);

   while (sdlaudio->fmt.samples) {
     sdlaudio->fmt.samples >>= 1;
     ++power2;
   }

   sdlaudio->fmt.samples = 1;
   sdlaudio->fmt.samples <<= power2;

   GST_DEBUG ("set segsize: %d, segtotal: %d, samples: %d", spec->segsize,
       spec->segtotal, sdlaudio->fmt.samples);

   if (SDL_OpenAudio (&sdlaudio->fmt, NULL) < 0) {
     goto unable_open;
   }

   spec->segsize = sdlaudio->fmt.size;

   sdlaudio->buffer = g_malloc (sdlaudio->fmt.size);
   memset (sdlaudio->buffer, sdlaudio->fmt.silence, sdlaudio->fmt.size);

   GST_DEBUG ("set segsize: %d, segtotal: %d, samples: %d", spec->segsize,
       spec->segtotal, sdlaudio->fmt.samples);

   spec->bytes_per_sample =
       spec->channels * ((sdlaudio->fmt.format & 0xFF) >> 3);
   memset (spec->silence_sample, sdlaudio->fmt.silence, spec->bytes_per_sample);

   SDL_PauseAudio (0);

   return TRUE;

 unable_open:
   {
     GST_ELEMENT_ERROR (sdlaudio, RESOURCE, OPEN_READ,
         ("Unable to open audio: %s", SDL_GetError ()), (NULL));
     return FALSE;
   }
 wrong_format:
   {
     GST_ELEMENT_ERROR (sdlaudio, RESOURCE, OPEN_READ,
         ("Unable to get format %d", spec->format), (NULL));
     return FALSE;
   }
 dodgy_width:
   {
     GST_ELEMENT_ERROR (sdlaudio, RESOURCE, OPEN_READ,
         ("unexpected width %d", spec->width), (NULL));
     return FALSE;
   }
 }

 static gboolean
 gst_sdlaudio_sink_unprepare (GstAudioSink * asink)
 {

   SDL_CloseAudio ();

   return TRUE;

 #if 0
   if (!gst_sdlaudio_sink_close (asink))
     goto couldnt_close;

   if (!gst_sdlaudio_sink_open (asink))
     goto couldnt_reopen;

   return TRUE;

 couldnt_close:
   {
     GST_DEBUG ("Could not close the audio device");
     return FALSE;
   }
 couldnt_reopen:
   {
     GST_DEBUG ("Could not reopen the audio device");
     return FALSE;
   }
 #endif

 }

 #if 0
 static guint
 gst_sdlaudio_sink_delay (GstAudioSink * asink)
 {
   GstSDLAudioSink *sdlaudio;

   sdlaudio = GST_SDLAUDIOSINK (asink);

   return 0;
 }

 static void
 gst_sdlaudio_sink_reset (GstAudioSink * asink)
 {
 }
 #endif
	/* GStreamer
	* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
	*
	* 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
	*/

	#ifdef HAVE_CONFIG_H
	#include "config.h"
	#endif

	#include "sdlaudiosink.h"

	#include <SDL_byteorder.h>
	#include <string.h>

	#include <unistd.h>

	#include <gst/glib-compat-private.h>

	GST_DEBUG_CATEGORY_EXTERN (sdl_debug);
	#define GST_CAT_DEFAULT sdl_debug

	static void gst_sdlaudio_sink_dispose (GObject * object);

	static GstCaps gst_sdlaudio_sink_getcaps (GstBaseSink bsink);

	static gboolean gst_sdlaudio_sink_open (GstAudioSink * asink);
	static gboolean gst_sdlaudio_sink_close (GstAudioSink * asink);
	static gboolean gst_sdlaudio_sink_prepare (GstAudioSink * asink,
	GstRingBufferSpec * spec);
	static gboolean gst_sdlaudio_sink_unprepare (GstAudioSink * asink);
	static guint gst_sdlaudio_sink_write (GstAudioSink * asink, gpointer data,
	guint length);

	#if 0
	static guint gst_sdlaudio_sink_delay (GstAudioSink * asink);
	static void gst_sdlaudio_sink_reset (GstAudioSink * asink);
	#endif


	/* SdlaudioSink signals and args */
	enum
	{
	LAST_SIGNAL
	};

	#define SEMAPHORE_INIT(s,f) \
	do { \
	s.cond = g_cond_new(); \
	s.mutex = g_mutex_new(); \
	s.mutexflag = f; \
	} while(0)

	#define SEMAPHORE_CLOSE(s) \
	do { \
	if ( s.cond ) { \
	g_cond_free(s.cond); \
	s.cond = NULL; \
	} \
	if ( s.mutex ) { \
	g_mutex_free(s.mutex); \
	s.mutex = NULL; \
	} \
	} while(0)

	#define SEMAPHORE_UP(s) \
	do \
	{ \
	g_mutex_lock(s.mutex); \
	s.mutexflag = TRUE; \
	g_mutex_unlock(s.mutex); \
	g_cond_signal(s.cond); \
	} while(0)

	#define SEMAPHORE_DOWN(s, e) \
	do \
	{ \
	while (1) { \
	g_mutex_lock(s.mutex); \
	if (!s.mutexflag) { \
	if ( e ) { \
	g_mutex_unlock(s.mutex); \
	break; \
	} \
	g_cond_wait(s.cond,s.mutex); \
	} \
	else { \
	s.mutexflag = FALSE; \
	g_mutex_unlock(s.mutex); \
	break; \
	} \
	g_mutex_unlock(s.mutex); \
	} \
	} while(0)


	static GstStaticPadTemplate sdlaudiosink_sink_factory =
	GST_STATIC_PAD_TEMPLATE ("sink",
	GST_PAD_SINK,
	GST_PAD_ALWAYS,
	GST_STATIC_CAPS ("audio/x-raw-int, "
	"endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
	"signed = (boolean) { TRUE, FALSE }, "
	"width = (int) 16, "
	"depth = (int) 16, "
	"rate = (int) [ 1, MAX ], "
	"channels = (int) [ 1, 2 ]; "
	"audio/x-raw-int, "
	"endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
	"signed = (boolean) { TRUE, FALSE }, "
	"width = (int) 8, "
	"depth = (int) 8, "
	"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
	);

	GST_BOILERPLATE (GstSDLAudioSink, gst_sdlaudio_sink, GstAudioSink,
	GST_TYPE_AUDIO_SINK);

	static void
	gst_sdlaudio_sink_dispose (GObject * object)
	{
	GstSDLAudioSink *sdlaudiosink = GST_SDLAUDIOSINK (object);

	SEMAPHORE_CLOSE (sdlaudiosink->semB);

	SEMAPHORE_CLOSE (sdlaudiosink->semA);

	if (sdlaudiosink->buffer) {
	g_free (sdlaudiosink->buffer);
	}

	G_OBJECT_CLASS (parent_class)->dispose (object);

	}

	static void
	gst_sdlaudio_sink_base_init (gpointer g_class)
	{
	GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);

	gst_element_class_set_static_metadata (element_class, "SDL audio sink",
	"Sink/Audio",
	"Output to a sound card via SDLAUDIO",
	"Edgard Lima <edgard.lima@indt.org.br>");

	gst_element_class_add_static_pad_template (element_class,
	&sdlaudiosink_sink_factory);
	}

	static void
	gst_sdlaudio_sink_class_init (GstSDLAudioSinkClass * klass)
	{
	GObjectClass *gobject_class;
	GstBaseSinkClass *gstbasesink_class;
	GstAudioSinkClass *gstaudiosink_class;

	gobject_class = (GObjectClass *) klass;
	gstbasesink_class = (GstBaseSinkClass *) klass;
	gstaudiosink_class = (GstAudioSinkClass *) klass;

	gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_dispose);

	gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_getcaps);

	gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_open);
	gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_close);
	gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_prepare);
	gstaudiosink_class->unprepare =
	GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_unprepare);
	gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_write);

	#if 0
	gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_delay);
	gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_sdlaudio_sink_reset);
	#endif

	}

	static void
	gst_sdlaudio_sink_init (GstSDLAudioSink * sdlaudiosink,
	GstSDLAudioSinkClass * g_class)
	{
	GST_DEBUG ("initializing sdlaudiosink");

	memset (&sdlaudiosink->fmt, 0, sizeof (SDL_AudioSpec));

	sdlaudiosink->buffer = NULL;
	sdlaudiosink->eos = FALSE;

	SEMAPHORE_INIT (sdlaudiosink->semA, TRUE);

	SEMAPHORE_INIT (sdlaudiosink->semB, FALSE);

	}

	static GstCaps *
	gst_sdlaudio_sink_getcaps (GstBaseSink * bsink)
	{
	return gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD
	(bsink)));
	}

	static gint
	gst_sdlaudio_sink_get_format (GstBufferFormat fmt)
	{
	gint result = GST_UNKNOWN;

	switch (fmt) {
	case GST_U8:
	result = AUDIO_U8;
	break;
	case GST_S8:
	result = AUDIO_S8;
	break;
	case GST_S16_LE:
	result = AUDIO_S16LSB;
	break;
	case GST_S16_BE:
	result = AUDIO_S16MSB;
	break;
	case GST_U16_LE:
	result = AUDIO_U16LSB;
	break;
	case GST_U16_BE:
	result = AUDIO_U16MSB;
	break;
	default:
	break;
	}
	return result;
	}

	static gboolean
	gst_sdlaudio_sink_open (GstAudioSink * asink)
	{
	GstSDLAudioSink *sdlaudio;

	sdlaudio = GST_SDLAUDIOSINK (asink);

	if (SDL_Init (SDL_INIT_AUDIO) < 0) {
	goto open_failed;
	}

	return TRUE;

	open_failed:
	{
	GST_ELEMENT_ERROR (sdlaudio, LIBRARY, INIT,
	("Unable to init SDL: %s\n", SDL_GetError ()), (NULL));
	return FALSE;
	}
	}

	static gboolean
	gst_sdlaudio_sink_close (GstAudioSink * asink)
	{
	GstSDLAudioSink *sdlaudio = GST_SDLAUDIOSINK (asink);

	sdlaudio->eos = TRUE;
	SEMAPHORE_UP (sdlaudio->semA);
	SEMAPHORE_UP (sdlaudio->semB);
	SDL_QuitSubSystem (SDL_INIT_AUDIO);
	return TRUE;
	}

	static guint
	gst_sdlaudio_sink_write (GstAudioSink * asink, gpointer data, guint length)
	{
	GstSDLAudioSink *sdlaudio = GST_SDLAUDIOSINK (asink);

	if (sdlaudio->fmt.size != length) {
	GST_ERROR ("ring buffer segment length (%u) != sdl buffer len (%u)", length,
	sdlaudio->fmt.size);
	}

	SEMAPHORE_DOWN (sdlaudio->semA, sdlaudio->eos);

	if (!sdlaudio->eos)
	memcpy (sdlaudio->buffer, data, length);

	SEMAPHORE_UP (sdlaudio->semB);

	return sdlaudio->fmt.size;
	}


	static void
	mixaudio (void unused, Uint8 stream, int len)
	{
	GstSDLAudioSink *sdlaudio;

	sdlaudio = GST_SDLAUDIOSINK (unused);

	if (sdlaudio->fmt.size != len) {
	GST_ERROR ("fmt buffer len (%u) != sdl callback len (%d)",
	sdlaudio->fmt.size, len);
	}

	SEMAPHORE_DOWN (sdlaudio->semB, sdlaudio->eos);

	if (!sdlaudio->eos)
	SDL_MixAudio (stream, sdlaudio->buffer, sdlaudio->fmt.size,
	SDL_MIX_MAXVOLUME);

	SEMAPHORE_UP (sdlaudio->semA);

	}

	static gboolean
	gst_sdlaudio_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
	{
	GstSDLAudioSink *sdlaudio;
	gint power2 = -1;

	sdlaudio = GST_SDLAUDIOSINK (asink);

	sdlaudio->fmt.format = gst_sdlaudio_sink_get_format (spec->format);
	if (sdlaudio->fmt.format == 0)
	goto wrong_format;

	if (spec->width != 16 && spec->width != 8)
	goto dodgy_width;

	sdlaudio->fmt.freq = spec->rate;
	sdlaudio->fmt.channels = spec->channels;
	sdlaudio->fmt.samples =
	spec->segsize / (spec->channels * ((sdlaudio->fmt.format & 0xFF) >> 3));
	sdlaudio->fmt.callback = mixaudio;
	sdlaudio->fmt.userdata = sdlaudio;

	GST_DEBUG ("set segsize: %d, segtotal: %d, samples: %d", spec->segsize,
	spec->segtotal, sdlaudio->fmt.samples);

	while (sdlaudio->fmt.samples) {
	sdlaudio->fmt.samples >>= 1;
	++power2;
	}

	sdlaudio->fmt.samples = 1;
	sdlaudio->fmt.samples <<= power2;

	GST_DEBUG ("set segsize: %d, segtotal: %d, samples: %d", spec->segsize,
	spec->segtotal, sdlaudio->fmt.samples);

	if (SDL_OpenAudio (&sdlaudio->fmt, NULL) < 0) {
	goto unable_open;
	}

	spec->segsize = sdlaudio->fmt.size;

	sdlaudio->buffer = g_malloc (sdlaudio->fmt.size);
	memset (sdlaudio->buffer, sdlaudio->fmt.silence, sdlaudio->fmt.size);

	GST_DEBUG ("set segsize: %d, segtotal: %d, samples: %d", spec->segsize,
	spec->segtotal, sdlaudio->fmt.samples);

	spec->bytes_per_sample =
	spec->channels * ((sdlaudio->fmt.format & 0xFF) >> 3);
	memset (spec->silence_sample, sdlaudio->fmt.silence, spec->bytes_per_sample);

	SDL_PauseAudio (0);

	return TRUE;

	unable_open:
	{
	GST_ELEMENT_ERROR (sdlaudio, RESOURCE, OPEN_READ,
	("Unable to open audio: %s", SDL_GetError ()), (NULL));
	return FALSE;
	}
	wrong_format:
	{
	GST_ELEMENT_ERROR (sdlaudio, RESOURCE, OPEN_READ,
	("Unable to get format %d", spec->format), (NULL));
	return FALSE;
	}
	dodgy_width:
	{
	GST_ELEMENT_ERROR (sdlaudio, RESOURCE, OPEN_READ,
	("unexpected width %d", spec->width), (NULL));
	return FALSE;
	}
	}

	static gboolean
	gst_sdlaudio_sink_unprepare (GstAudioSink * asink)
	{

	SDL_CloseAudio ();

	return TRUE;

	#if 0
	if (!gst_sdlaudio_sink_close (asink))
	goto couldnt_close;

	if (!gst_sdlaudio_sink_open (asink))
	goto couldnt_reopen;

	return TRUE;

	couldnt_close:
	{
	GST_DEBUG ("Could not close the audio device");
	return FALSE;
	}
	couldnt_reopen:
	{
	GST_DEBUG ("Could not reopen the audio device");
	return FALSE;
	}
	#endif

	}

	#if 0
	static guint
	gst_sdlaudio_sink_delay (GstAudioSink * asink)
	{
	GstSDLAudioSink *sdlaudio;

	sdlaudio = GST_SDLAUDIOSINK (asink);

	return 0;
	}

	static void
	gst_sdlaudio_sink_reset (GstAudioSink * asink)
	{
	}
	#endif