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authorEdgard Lima <edgard.lima@indt.org.br>2006-01-09 18:20:56 +0000
committerEdgard Lima <edgard.lima@indt.org.br>2006-01-09 18:20:56 +0000
commita438034608c14b446943fcc0fbb0fa04df93aa47 (patch)
tree765d96dbb8f9f3711cae0183c945612f09a84071 /ext/sdl
parentce77e3b44bc291c54039918f2136dff8b44a4c9f (diff)
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Created new element, sdlaudiosink.
Original commit message from CVS: Created new element, sdlaudiosink.
Diffstat (limited to 'ext/sdl')
-rw-r--r--ext/sdl/Makefile.am19
-rw-r--r--ext/sdl/gstsdl.c45
-rw-r--r--ext/sdl/sdlaudiosink.c484
-rw-r--r--ext/sdl/sdlaudiosink.h65
-rw-r--r--ext/sdl/sdlvideosink.c22
5 files changed, 611 insertions, 24 deletions
diff --git a/ext/sdl/Makefile.am b/ext/sdl/Makefile.am
index c183ee80..24e6d985 100644
--- a/ext/sdl/Makefile.am
+++ b/ext/sdl/Makefile.am
@@ -1,8 +1,19 @@
plugin_LTLIBRARIES = libgstsdlvideosink.la
-libgstsdlvideosink_la_SOURCES = sdlvideosink.c
-libgstsdlvideosink_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(SDL_CFLAGS)
-libgstsdlvideosink_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstvideo-$(GST_MAJORMINOR) -lgstinterfaces-$(GST_MAJORMINOR) $(SDL_LIBS)
+libgstsdlvideosink_la_SOURCES = gstsdl.c \
+ sdlvideosink.c \
+ sdlaudiosink.c
+
+libgstsdlvideosink_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) \
+ $(SDL_CFLAGS)
+
+libgstsdlvideosink_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) \
+ -lgstvideo-$(GST_MAJORMINOR) \
+ -lgstaudio-@GST_MAJORMINOR@ \
+ -lgstinterfaces-$(GST_MAJORMINOR) \
+ $(SDL_LIBS)
+
libgstsdlvideosink_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
-noinst_HEADERS = sdlvideosink.h
+noinst_HEADERS = sdlvideosink.h \
+ sdlaudiosink.h
diff --git a/ext/sdl/gstsdl.c b/ext/sdl/gstsdl.c
new file mode 100644
index 00000000..bf5d7da9
--- /dev/null
+++ b/ext/sdl/gstsdl.c
@@ -0,0 +1,45 @@
+/* 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 "sdlvideosink.h"
+#include "sdlaudiosink.h"
+
+
+GST_DEBUG_CATEGORY (sdl_debug);
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+
+ if (!gst_element_register (plugin, "sdlvideosink", GST_RANK_NONE,
+ GST_TYPE_SDLVIDEOSINK) ||
+ !gst_element_register (plugin, "sdlaudiosink", GST_RANK_NONE,
+ GST_TYPE_SDLAUDIOSINK)) {
+ return FALSE;
+ }
+
+ GST_DEBUG_CATEGORY_INIT (sdl_debug, "sdl", 0, "SDL elements");
+
+ return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ "sdl",
+ "SDL (Simple DirectMedia Layer) support for GStreamer",
+ plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)
diff --git a/ext/sdl/sdlaudiosink.c b/ext/sdl/sdlaudiosink.c
new file mode 100644
index 00000000..4465a529
--- /dev/null
+++ b/ext/sdl/sdlaudiosink.c
@@ -0,0 +1,484 @@
+/* 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>
+
+GST_DEBUG_CATEGORY_EXTERN (sdl_debug);
+#define GST_CAT_DEFAULT sdl_debug
+
+/* elementfactory information */
+static GstElementDetails gst_sdlaudio_sink_details =
+GST_ELEMENT_DETAILS ("Audio Sink (SDLAUDIO)",
+ "Sink/Audio",
+ "Output to a sound card via SDLAUDIO",
+ "Edgard Lima <edgard.lima@indt.org.br>");
+
+static void gst_sdlaudio_sink_base_init (gpointer g_class);
+static void gst_sdlaudio_sink_class_init (GstSDLAudioSinkClass * klass);
+static void gst_sdlaudio_sink_init (GstSDLAudioSink * sdlaudiosink);
+static void gst_sdlaudio_sink_dispose (GObject * object);
+
+static void gst_sdlaudio_sink_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static void gst_sdlaudio_sink_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+
+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 ]")
+ );
+
+static GstElementClass *parent_class = NULL;
+
+GType
+gst_sdlaudio_sink_get_type (void)
+{
+ static GType sdlaudiosink_type = 0;
+
+ if (!sdlaudiosink_type) {
+ static const GTypeInfo sdlaudiosink_info = {
+ sizeof (GstSDLAudioSinkClass),
+ gst_sdlaudio_sink_base_init,
+ NULL,
+ (GClassInitFunc) gst_sdlaudio_sink_class_init,
+ NULL,
+ NULL,
+ sizeof (GstSDLAudioSink),
+ 0,
+ (GInstanceInitFunc) gst_sdlaudio_sink_init,
+ };
+
+ sdlaudiosink_type =
+ g_type_register_static (GST_TYPE_AUDIO_SINK, "GstSDLAudioSink",
+ &sdlaudiosink_info, 0);
+ }
+
+ return sdlaudiosink_type;
+}
+
+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_details (element_class, &gst_sdlaudio_sink_details);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&sdlaudiosink_sink_factory));
+}
+static void
+gst_sdlaudio_sink_class_init (GstSDLAudioSinkClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBaseSinkClass *gstbasesink_class;
+ GstBaseAudioSinkClass *gstbaseaudiosink_class;
+ GstAudioSinkClass *gstaudiosink_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ gstbasesink_class = (GstBaseSinkClass *) klass;
+ gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
+ gstaudiosink_class = (GstAudioSinkClass *) klass;
+
+ parent_class = g_type_class_ref (GST_TYPE_BASE_AUDIO_SINK);
+
+ 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)
+{
+ 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, TRUE);
+
+}
+
+static GstCaps *
+gst_sdlaudio_sink_getcaps (GstBaseSink * bsink)
+{
+ GstSDLAudioSink *sdlaudiosink;
+ GstCaps *caps = NULL;
+
+ sdlaudiosink = GST_SDLAUDIOSINK (bsink);
+
+ caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD
+ (bsink)));
+ return caps;
+}
+
+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;
+ int mode;
+
+ 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_Quit ();
+ 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 lenght (%u) != sdl buffer len", 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;
+}
+
+
+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
diff --git a/ext/sdl/sdlaudiosink.h b/ext/sdl/sdlaudiosink.h
new file mode 100644
index 00000000..646a7747
--- /dev/null
+++ b/ext/sdl/sdlaudiosink.h
@@ -0,0 +1,65 @@
+/* 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
+ */
+
+#ifndef __GST_SDLAUDIOSINK_H__
+#define __GST_SDLAUDIOSINK_H__
+
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiosink.h>
+
+#include <SDL.h>
+#include <SDL_audio.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_SDLAUDIOSINK (gst_sdlaudio_sink_get_type())
+#define GST_SDLAUDIOSINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_SDLAUDIOSINK,GstSDLAudioSink))
+#define GST_SDLAUDIOSINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_SDLAUDIOSINK,GstSDLAudioSinkClass))
+#define GST_IS_SDLAUDIOSINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_SDLAUDIOSINK))
+#define GST_IS_SDLAUDIOSINK_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SDLAUDIOSINK))
+
+typedef struct _GstSDLAudioSink GstSDLAudioSink;
+typedef struct _GstSDLAudioSinkClass GstSDLAudioSinkClass;
+
+typedef struct _gstsdl_semaphore {
+ GCond *cond;
+ GMutex *mutex;
+ gboolean mutexflag;
+
+} gstsdl_semaphore;
+
+struct _GstSDLAudioSink {
+ GstAudioSink sink;
+
+ SDL_AudioSpec fmt;
+ guint8 *buffer;
+
+ gstsdl_semaphore semA;
+
+ gstsdl_semaphore semB;
+
+ gboolean eos;
+
+};
+
+struct _GstSDLAudioSinkClass {
+ GstAudioSinkClass parent_class;
+};
+
+GType gst_sdlaudio_sink_get_type(void);
+
+G_END_DECLS
+
+#endif /* __GST_SDLAUDIOSINK_H__ */
diff --git a/ext/sdl/sdlvideosink.c b/ext/sdl/sdlvideosink.c
index 4f99835d..6d4688fb 100644
--- a/ext/sdl/sdlvideosink.c
+++ b/ext/sdl/sdlvideosink.c
@@ -32,8 +32,8 @@
#include "sdlvideosink.h"
-GST_DEBUG_CATEGORY_STATIC (sdlvideosink_debug);
-#define GST_CAT_DEFAULT sdlvideosink_debug
+GST_DEBUG_CATEGORY_EXTERN (sdl_debug);
+#define GST_CAT_DEFAULT sdl_debug
/* These macros are adapted from videotestsrc.c
* and/or gst-plugins/gst/games/gstvideoimage.c */
@@ -173,8 +173,6 @@ gst_sdlvideosink_base_init (gpointer g_class)
gst_element_class_add_pad_template (element_class, sink_template);
gst_element_class_set_details (element_class, &gst_sdlvideosink_details);
- GST_DEBUG_CATEGORY_INIT (sdlvideosink_debug, "sdlvideosink", 0,
- "SDL video sink element");
}
static void
@@ -844,19 +842,3 @@ init_failed:
return GST_STATE_CHANGE_FAILURE;
}
}
-
-
-static gboolean
-plugin_init (GstPlugin * plugin)
-{
- if (!gst_element_register (plugin, "sdlvideosink", GST_RANK_NONE,
- GST_TYPE_SDLVIDEOSINK))
- return FALSE;
-
- return TRUE;
-}
-
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
- GST_VERSION_MINOR,
- "sdlvideosink",
- "SDL Video Sink", plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)