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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstbaseaudiosink.c:
*
* 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 Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include "gstbaseaudiosink.h"
GST_DEBUG_CATEGORY_STATIC (gst_baseaudiosink_debug);
#define GST_CAT_DEFAULT gst_baseaudiosink_debug
/* BaseAudioSink signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
};
#define _do_init(bla) \
GST_DEBUG_CATEGORY_INIT (gst_baseaudiosink_debug, "baseaudiosink", 0, "baseaudiosink element");
GST_BOILERPLATE_FULL (GstBaseAudioSink, gst_baseaudiosink, GstBaseSink,
GST_TYPE_BASESINK, _do_init);
static void gst_baseaudiosink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_baseaudiosink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstElementStateReturn gst_baseaudiosink_change_state (GstElement *
element);
static GstFlowReturn gst_baseaudiosink_preroll (GstBaseSink * bsink,
GstBuffer * buffer);
static GstFlowReturn gst_baseaudiosink_render (GstBaseSink * bsink,
GstBuffer * buffer);
static void gst_baseaudiosink_event (GstBaseSink * bsink, GstEvent * event);
static void gst_baseaudiosink_get_times (GstBaseSink * bsink,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
static gboolean gst_baseaudiosink_setcaps (GstBaseSink * bsink, GstCaps * caps);
//static guint gst_baseaudiosink_signals[LAST_SIGNAL] = { 0 };
static void
gst_baseaudiosink_base_init (gpointer g_class)
{
}
static void
gst_baseaudiosink_class_init (GstBaseAudioSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_baseaudiosink_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_baseaudiosink_get_property);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_baseaudiosink_change_state);
gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_baseaudiosink_event);
gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_baseaudiosink_preroll);
gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_baseaudiosink_render);
gstbasesink_class->get_times =
GST_DEBUG_FUNCPTR (gst_baseaudiosink_get_times);
gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_baseaudiosink_setcaps);
}
static void
gst_baseaudiosink_init (GstBaseAudioSink * baseaudiosink)
{
}
static void
gst_baseaudiosink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstBaseAudioSink *sink;
sink = GST_BASEAUDIOSINK (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_baseaudiosink_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstBaseAudioSink *sink;
sink = GST_BASEAUDIOSINK (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_baseaudiosink_setcaps (GstBaseSink * bsink, GstCaps * caps)
{
GstBaseAudioSink *sink = GST_BASEAUDIOSINK (bsink);
GstRingBufferSpec spec;
spec.caps = caps;
spec.segsize = 64;
spec.segtotal = 64;
gst_ringbuffer_release (sink->ringbuffer);
gst_ringbuffer_acquire (sink->ringbuffer, &spec);
return TRUE;
}
static void
gst_baseaudiosink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
*start = GST_CLOCK_TIME_NONE;
*end = GST_CLOCK_TIME_NONE;
}
static void
gst_baseaudiosink_event (GstBaseSink * bsink, GstEvent * event)
{
}
static GstFlowReturn
gst_baseaudiosink_preroll (GstBaseSink * bsink, GstBuffer * buffer)
{
return GST_FLOW_OK;
}
static GstFlowReturn
gst_baseaudiosink_render (GstBaseSink * bsink, GstBuffer * buf)
{
GstBaseAudioSink *sink = GST_BASEAUDIOSINK (bsink);
gst_ringbuffer_write (sink->ringbuffer, GST_CLOCK_TIME_NONE,
GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
return GST_FLOW_OK;
}
GstRingBuffer *
gst_baseaudiosink_create_ringbuffer (GstBaseAudioSink * sink)
{
GstBaseAudioSinkClass *bclass;
GstRingBuffer *buffer = NULL;
bclass = GST_BASEAUDIOSINK_GET_CLASS (sink);
if (bclass->create_ringbuffer)
buffer = bclass->create_ringbuffer (sink);
return buffer;
}
void
gst_baseaudiosink_callback (GstRingBuffer * rbuf, guint advance, gpointer data)
{
//GstBaseAudioSink *sink = GST_BASEAUDIOSINK (data);
}
static GstElementStateReturn
gst_baseaudiosink_change_state (GstElement * element)
{
GstElementStateReturn ret = GST_STATE_SUCCESS;
GstBaseAudioSink *sink = GST_BASEAUDIOSINK (element);
GstElementState transition = GST_STATE_TRANSITION (element);
switch (transition) {
case GST_STATE_NULL_TO_READY:
break;
case GST_STATE_READY_TO_PAUSED:
sink->ringbuffer = gst_baseaudiosink_create_ringbuffer (sink);
gst_ringbuffer_set_callback (sink->ringbuffer, gst_baseaudiosink_callback,
sink);
break;
case GST_STATE_PAUSED_TO_PLAYING:
gst_ringbuffer_play (sink->ringbuffer);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element);
switch (transition) {
case GST_STATE_PLAYING_TO_PAUSED:
gst_ringbuffer_stop (sink->ringbuffer);
break;
case GST_STATE_PAUSED_TO_READY:
gst_object_unref (GST_OBJECT (sink->ringbuffer));
break;
case GST_STATE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
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