gstreamer/gst/inter/gstinteraudiosink.c
Sebastian Dröge a2a4300241 Merge branch 'master' into 0.11
Conflicts:
	configure.ac
	ext/kate/gstkateenc.c
	gst/colorspace/colorspace.c
	gst/mpegvideoparse/mpegvideoparse.c
2012-01-25 13:22:43 +01:00

342 lines
9.7 KiB
C

/* GStreamer
* Copyright (C) 2011 David A. Schleef <ds@schleef.org>
*
* 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., 51 Franklin Street, Suite 500,
* Boston, MA 02110-1335, USA.
*/
/**
* SECTION:element-gstinteraudiosink
*
* The interaudiosink element does FIXME stuff.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch -v fakesrc ! interaudiosink ! FIXME ! fakesink
* ]|
* FIXME Describe what the pipeline does.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/base/gstbasesink.h>
#include <gst/audio/audio.h>
#include "gstinteraudiosink.h"
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (gst_inter_audio_sink_debug_category);
#define GST_CAT_DEFAULT gst_inter_audio_sink_debug_category
/* prototypes */
static void gst_inter_audio_sink_set_property (GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec);
static void gst_inter_audio_sink_get_property (GObject * object,
guint property_id, GValue * value, GParamSpec * pspec);
static void gst_inter_audio_sink_dispose (GObject * object);
static void gst_inter_audio_sink_finalize (GObject * object);
static GstCaps *gst_inter_audio_sink_get_caps (GstBaseSink * sink);
static gboolean gst_inter_audio_sink_set_caps (GstBaseSink * sink,
GstCaps * caps);
static GstFlowReturn gst_inter_audio_sink_buffer_alloc (GstBaseSink * sink,
guint64 offset, guint size, GstCaps * caps, GstBuffer ** buf);
static void gst_inter_audio_sink_get_times (GstBaseSink * sink,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
static gboolean gst_inter_audio_sink_start (GstBaseSink * sink);
static gboolean gst_inter_audio_sink_stop (GstBaseSink * sink);
static gboolean gst_inter_audio_sink_unlock (GstBaseSink * sink);
static gboolean gst_inter_audio_sink_event (GstBaseSink * sink,
GstEvent * event);
static GstFlowReturn gst_inter_audio_sink_preroll (GstBaseSink * sink,
GstBuffer * buffer);
static GstFlowReturn gst_inter_audio_sink_render (GstBaseSink * sink,
GstBuffer * buffer);
static GstStateChangeReturn gst_inter_audio_sink_async_play (GstBaseSink *
sink);
static gboolean gst_inter_audio_sink_activate_pull (GstBaseSink * sink,
gboolean active);
static gboolean gst_inter_audio_sink_unlock_stop (GstBaseSink * sink);
enum
{
PROP_0,
PROP_CHANNEL
};
/* pad templates */
static GstStaticPadTemplate gst_inter_audio_sink_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) true, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
);
/* class initialization */
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_inter_audio_sink_debug_category, "interaudiosink", 0, \
"debug category for interaudiosink element");
GST_BOILERPLATE_FULL (GstInterAudioSink, gst_inter_audio_sink, GstBaseSink,
GST_TYPE_BASE_SINK, DEBUG_INIT);
static void
gst_inter_audio_sink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_inter_audio_sink_sink_template));
gst_element_class_set_details_simple (element_class, "FIXME Long name",
"Generic", "FIXME Description", "FIXME <fixme@example.com>");
}
static void
gst_inter_audio_sink_class_init (GstInterAudioSinkClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstBaseSinkClass *base_sink_class = GST_BASE_SINK_CLASS (klass);
gobject_class->set_property = gst_inter_audio_sink_set_property;
gobject_class->get_property = gst_inter_audio_sink_get_property;
gobject_class->dispose = gst_inter_audio_sink_dispose;
gobject_class->finalize = gst_inter_audio_sink_finalize;
base_sink_class->get_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_get_caps);
base_sink_class->set_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_set_caps);
if (0)
base_sink_class->buffer_alloc =
GST_DEBUG_FUNCPTR (gst_inter_audio_sink_buffer_alloc);
base_sink_class->get_times =
GST_DEBUG_FUNCPTR (gst_inter_audio_sink_get_times);
base_sink_class->start = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_start);
base_sink_class->stop = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_stop);
base_sink_class->unlock = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_unlock);
if (0)
base_sink_class->event = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_event);
//if (0)
base_sink_class->preroll = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_preroll);
base_sink_class->render = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_render);
if (0)
base_sink_class->async_play =
GST_DEBUG_FUNCPTR (gst_inter_audio_sink_async_play);
if (0)
base_sink_class->activate_pull =
GST_DEBUG_FUNCPTR (gst_inter_audio_sink_activate_pull);
base_sink_class->unlock_stop =
GST_DEBUG_FUNCPTR (gst_inter_audio_sink_unlock_stop);
g_object_class_install_property (gobject_class, PROP_CHANNEL,
g_param_spec_string ("channel", "Channel",
"Channel name to match inter src and sink elements",
"default", G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_inter_audio_sink_init (GstInterAudioSink * interaudiosink,
GstInterAudioSinkClass * interaudiosink_class)
{
interaudiosink->surface = gst_inter_surface_get ("default");
}
void
gst_inter_audio_sink_set_property (GObject * object, guint property_id,
const GValue * value, GParamSpec * pspec)
{
/* GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object); */
switch (property_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
void
gst_inter_audio_sink_get_property (GObject * object, guint property_id,
GValue * value, GParamSpec * pspec)
{
/* GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object); */
switch (property_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
void
gst_inter_audio_sink_dispose (GObject * object)
{
/* GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object); */
/* clean up as possible. may be called multiple times */
G_OBJECT_CLASS (parent_class)->dispose (object);
}
void
gst_inter_audio_sink_finalize (GObject * object)
{
/* GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object); */
/* clean up object here */
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstCaps *
gst_inter_audio_sink_get_caps (GstBaseSink * sink)
{
return NULL;
}
static gboolean
gst_inter_audio_sink_set_caps (GstBaseSink * sink, GstCaps * caps)
{
return TRUE;
}
static GstFlowReturn
gst_inter_audio_sink_buffer_alloc (GstBaseSink * sink, guint64 offset,
guint size, GstCaps * caps, GstBuffer ** buf)
{
return GST_FLOW_ERROR;
}
static void
gst_inter_audio_sink_get_times (GstBaseSink * sink, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
*start = GST_BUFFER_TIMESTAMP (buffer);
if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
*end = *start + GST_BUFFER_DURATION (buffer);
} else {
if (interaudiosink->fps_n > 0) {
*end = *start +
gst_util_uint64_scale_int (GST_SECOND, interaudiosink->fps_d,
interaudiosink->fps_n);
}
}
}
}
static gboolean
gst_inter_audio_sink_start (GstBaseSink * sink)
{
return TRUE;
}
static gboolean
gst_inter_audio_sink_stop (GstBaseSink * sink)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
GST_DEBUG ("stop");
g_mutex_lock (interaudiosink->surface->mutex);
gst_adapter_clear (interaudiosink->surface->audio_adapter);
g_mutex_unlock (interaudiosink->surface->mutex);
return TRUE;
}
static gboolean
gst_inter_audio_sink_unlock (GstBaseSink * sink)
{
return TRUE;
}
static gboolean
gst_inter_audio_sink_event (GstBaseSink * sink, GstEvent * event)
{
return TRUE;
}
static GstFlowReturn
gst_inter_audio_sink_preroll (GstBaseSink * sink, GstBuffer * buffer)
{
return GST_FLOW_OK;
}
static GstFlowReturn
gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
int n;
GST_DEBUG ("render %d", GST_BUFFER_SIZE (buffer));
g_mutex_lock (interaudiosink->surface->mutex);
n = gst_adapter_available (interaudiosink->surface->audio_adapter) / 4;
if (n > (800 * 2 * 2)) {
GST_INFO ("flushing 800 samples");
gst_adapter_flush (interaudiosink->surface->audio_adapter, 800 * 4);
n -= 800;
}
gst_adapter_push (interaudiosink->surface->audio_adapter,
gst_buffer_ref (buffer));
g_mutex_unlock (interaudiosink->surface->mutex);
return GST_FLOW_OK;
}
static GstStateChangeReturn
gst_inter_audio_sink_async_play (GstBaseSink * sink)
{
return GST_STATE_CHANGE_SUCCESS;
}
static gboolean
gst_inter_audio_sink_activate_pull (GstBaseSink * sink, gboolean active)
{
return TRUE;
}
static gboolean
gst_inter_audio_sink_unlock_stop (GstBaseSink * sink)
{
return TRUE;
}