gstreamer/gst/inter/gstinteraudiosink.c

247 lines
7.4 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 is an audio sink element. It is used
* in connection with a interaudiosrc element in a different pipeline,
* similar to intervideosink and intervideosrc.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch -v audiotestsrc ! queue ! interaudiosink
* ]|
*
* The interaudiosink element cannot be used effectively with gst-launch,
* as it requires a second pipeline in the application to receive the
* audio.
* See the gstintertest.c example in the gst-plugins-bad source code for
* more details.
* </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_finalize (GObject * object);
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 GstFlowReturn gst_inter_audio_sink_render (GstBaseSink * sink,
GstBuffer * buffer);
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, format = (string) " GST_AUDIO_NE (S16) ", "
"rate = (int) 48000, channels = (int) 2")
);
/* class initialization */
G_DEFINE_TYPE (GstInterAudioSink, gst_inter_audio_sink, GST_TYPE_BASE_SINK);
static void
gst_inter_audio_sink_class_init (GstInterAudioSinkClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseSinkClass *base_sink_class = GST_BASE_SINK_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (gst_inter_audio_sink_debug_category,
"interaudiosink", 0, "debug category for interaudiosink element");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_inter_audio_sink_sink_template));
gst_element_class_set_metadata (element_class,
"Internal audio sink",
"Sink/Audio",
"Virtual audio sink for internal process communication",
"David Schleef <ds@schleef.org>");
gobject_class->set_property = gst_inter_audio_sink_set_property;
gobject_class->get_property = gst_inter_audio_sink_get_property;
gobject_class->finalize = gst_inter_audio_sink_finalize;
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->render = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_render);
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)
{
interaudiosink->channel = g_strdup ("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) {
case PROP_CHANNEL:
g_free (interaudiosink->channel);
interaudiosink->channel = g_value_dup_string (value);
break;
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) {
case PROP_CHANNEL:
g_value_set_string (value, interaudiosink->channel);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
void
gst_inter_audio_sink_finalize (GObject * object)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object);
/* clean up object here */
g_free (interaudiosink->channel);
G_OBJECT_CLASS (gst_inter_audio_sink_parent_class)->finalize (object);
}
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)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
GST_DEBUG ("start");
interaudiosink->surface = gst_inter_surface_get (interaudiosink->channel);
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);
gst_inter_surface_unref (interaudiosink->surface);
interaudiosink->surface = NULL;
return TRUE;
}
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_get_size (buffer));
g_mutex_lock (interaudiosink->surface->mutex);
n = gst_adapter_available (interaudiosink->surface->audio_adapter) / 4;
#define SIZE 1600
if (n > (SIZE * 3)) {
GST_WARNING ("flushing 800 samples");
gst_adapter_flush (interaudiosink->surface->audio_adapter, (SIZE / 2) * 4);
n -= (SIZE / 2);
}
gst_adapter_push (interaudiosink->surface->audio_adapter,
gst_buffer_ref (buffer));
g_mutex_unlock (interaudiosink->surface->mutex);
return GST_FLOW_OK;
}