gstreamer/ext/artsd/gstartsdsink.c

366 lines
10 KiB
C
Raw Normal View History

/* GStreamer
* Copyright (C) <2001> Richard Boulton <richard-gst@tartarus.org>
*
* Based on example.c:
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstartsdsink.h"
/* elementfactory information */
static GstElementDetails artsdsink_details = {
"aRtsd audio sink",
"Sink/Audio",
"Plays audio to an aRts server",
"Richard Boulton <richard-gst@tartarus.org>",
};
/* Signals and args */
enum {
/* FILL ME */
LAST_SIGNAL
};
enum {
ARG_0,
ARG_MUTE,
ARG_NAME,
};
GST_PAD_TEMPLATE_FACTORY (sink_factory,
"sink", /* the name of the pads */
GST_PAD_SINK, /* type of the pad */
GST_PAD_ALWAYS, /* ALWAYS/SOMETIMES */
GST_CAPS_NEW (
"artsdsink_sink", /* the name of the caps */
"audio/x-raw-int", /* the mime type of the caps */
"format", GST_PROPS_STRING ("int"),
"law", GST_PROPS_INT (0),
"endianness", GST_PROPS_INT (G_BYTE_ORDER),
"signed", GST_PROPS_BOOLEAN (FALSE),
"width", GST_PROPS_LIST (
GST_PROPS_INT (8),
GST_PROPS_INT (16)
),
"depth", GST_PROPS_LIST (
GST_PROPS_INT (8),
GST_PROPS_INT (16)
),
"rate", GST_PROPS_INT_RANGE (8000, 96000),
"channels", GST_PROPS_LIST (
GST_PROPS_INT (1),
GST_PROPS_INT (2)
)
)
);
static void gst_artsdsink_base_init (gpointer g_class);
static void gst_artsdsink_class_init (GstArtsdsinkClass *klass);
static void gst_artsdsink_init (GstArtsdsink *artsdsink);
static gboolean gst_artsdsink_open_audio (GstArtsdsink *sink);
static void gst_artsdsink_close_audio (GstArtsdsink *sink);
static GstElementStateReturn gst_artsdsink_change_state (GstElement *element);
static gboolean gst_artsdsink_sync_parms (GstArtsdsink *artsdsink);
static GstPadLinkReturn gst_artsdsink_link (GstPad *pad, GstCaps *caps);
static void gst_artsdsink_chain (GstPad *pad, GstData *_data);
static void gst_artsdsink_set_property (GObject *object, guint prop_id,
const GValue *value, GParamSpec *pspec);
static void gst_artsdsink_get_property (GObject *object, guint prop_id,
GValue *value, GParamSpec *pspec);
static GstElementClass *parent_class = NULL;
/*static guint gst_artsdsink_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_artsdsink_get_type (void)
{
static GType artsdsink_type = 0;
if (!artsdsink_type) {
static const GTypeInfo artsdsink_info = {
sizeof(GstArtsdsinkClass),
gst_artsdsink_base_init,
NULL,
(GClassInitFunc)gst_artsdsink_class_init,
NULL,
NULL,
sizeof(GstArtsdsink),
0,
(GInstanceInitFunc)gst_artsdsink_init,
};
artsdsink_type = g_type_register_static(GST_TYPE_ELEMENT, "GstArtsdsink", &artsdsink_info, 0);
}
return artsdsink_type;
}
static void
gst_artsdsink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class, GST_PAD_TEMPLATE_GET (sink_factory));
gst_element_class_set_details (element_class, &artsdsink_details);
}
static void
gst_artsdsink_class_init (GstArtsdsinkClass *klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass*)klass;
gstelement_class = (GstElementClass*)klass;
parent_class = g_type_class_ref(GST_TYPE_ELEMENT);
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_MUTE,
g_param_spec_boolean("mute","mute","mute",
TRUE,G_PARAM_READWRITE)); /* CHECKME */
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_NAME,
g_param_spec_string("name","name","name",
NULL, G_PARAM_READWRITE)); /* CHECKME */
gobject_class->set_property = gst_artsdsink_set_property;
gobject_class->get_property = gst_artsdsink_get_property;
gstelement_class->change_state = gst_artsdsink_change_state;
}
static void
gst_artsdsink_init(GstArtsdsink *artsdsink)
{
artsdsink->sinkpad = gst_pad_new_from_template (
GST_PAD_TEMPLATE_GET (sink_factory), "sink");
gst_element_add_pad(GST_ELEMENT(artsdsink), artsdsink->sinkpad);
gst_pad_set_chain_function(artsdsink->sinkpad, gst_artsdsink_chain);
gst_pad_set_link_function(artsdsink->sinkpad, gst_artsdsink_link);
artsdsink->connected = FALSE;
artsdsink->mute = FALSE;
artsdsink->connect_name = NULL;
}
static gboolean
gst_artsdsink_sync_parms (GstArtsdsink *artsdsink)
{
g_return_val_if_fail (artsdsink != NULL, FALSE);
g_return_val_if_fail (GST_IS_ARTSDSINK (artsdsink), FALSE);
if (!artsdsink->connected) return TRUE;
/* Need to set stream to use new parameters: only way to do this is to reopen. */
gst_artsdsink_close_audio (artsdsink);
return gst_artsdsink_open_audio (artsdsink);
}
static GstPadLinkReturn
gst_artsdsink_link (GstPad *pad, GstCaps *caps)
{
GstArtsdsink *artsdsink = GST_ARTSDSINK (gst_pad_get_parent (pad));
if (!GST_CAPS_IS_FIXED (caps))
return GST_PAD_LINK_DELAYED;
gst_caps_get (caps,
"rate", &artsdsink->frequency,
"depth", &artsdsink->depth,
"signed", &artsdsink->signd,
"channels", &artsdsink->channels,
NULL);
if (gst_artsdsink_sync_parms (artsdsink))
return GST_PAD_LINK_OK;
return GST_PAD_LINK_REFUSED;
}
static void
gst_artsdsink_chain (GstPad *pad, GstData *_data)
{
GstBuffer *buf = GST_BUFFER (_data);
GstArtsdsink *artsdsink;
g_return_if_fail(pad != NULL);
g_return_if_fail(GST_IS_PAD(pad));
g_return_if_fail(buf != NULL);
artsdsink = GST_ARTSDSINK (gst_pad_get_parent (pad));
if (GST_BUFFER_DATA (buf) != NULL) {
gst_trace_add_entry(NULL, 0, GPOINTER_TO_INT(buf), "artsdsink: writing to server");
if (!artsdsink->mute && artsdsink->connected) {
int bytes;
void * bufptr = GST_BUFFER_DATA (buf);
int bufsize = GST_BUFFER_SIZE (buf);
GST_DEBUG ("artsdsink: stream=%p data=%p size=%d",
artsdsink->stream, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
do {
bytes = arts_write (artsdsink->stream, bufptr, bufsize);
if(bytes < 0) {
fprintf(stderr,"arts_write error: %s\n", arts_error_text(bytes));
gst_buffer_unref (buf);
return;
}
bufptr += bytes;
bufsize -= bytes;
} while (bufsize > 0);
}
}
gst_buffer_unref (buf);
}
static void
gst_artsdsink_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
{
GstArtsdsink *artsdsink;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail(GST_IS_ARTSDSINK(object));
artsdsink = GST_ARTSDSINK(object);
switch (prop_id) {
case ARG_MUTE:
artsdsink->mute = g_value_get_boolean (value);
break;
case ARG_NAME:
if (artsdsink->connect_name != NULL) g_free(artsdsink->connect_name);
if (g_value_get_string (value) == NULL)
artsdsink->connect_name = NULL;
else
artsdsink->connect_name = g_strdup (g_value_get_string (value));
break;
default:
break;
}
}
static void
gst_artsdsink_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
{
GstArtsdsink *artsdsink;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail(GST_IS_ARTSDSINK(object));
artsdsink = GST_ARTSDSINK(object);
switch (prop_id) {
case ARG_MUTE:
g_value_set_boolean (value, artsdsink->mute);
break;
case ARG_NAME:
g_value_set_string (value, artsdsink->connect_name);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
plugin_init (GstPlugin *plugin)
{
if (!gst_element_register (plugin, "artsdsink", GST_RANK_NONE, GST_TYPE_ARTSDSINK))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (
GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"artsdsink",
"Plays audio to an aRts server",
plugin_init,
VERSION,
"LGPL",
GST_PACKAGE,
GST_ORIGIN)
static gboolean
gst_artsdsink_open_audio (GstArtsdsink *sink)
{
const char * connname = "gstreamer";
int errcode;
/* Name used by aRtsd for this connection. */
if (sink->connect_name != NULL) connname = sink->connect_name;
/* FIXME: this should only ever happen once per process. */
/* Really, artsc needs to be made thread safe to fix this (and other related */
/* problems). */
errcode = arts_init();
if(errcode < 0) {
fprintf(stderr,"arts_init error: %s\n", arts_error_text(errcode));
return FALSE;
}
GST_DEBUG ("artsdsink: attempting to open connection to aRtsd server");
sink->stream = arts_play_stream(sink->frequency, sink->depth,
sink->channels, connname);
/* FIXME: check connection */
/* GST_DEBUG ("artsdsink: can't open connection to aRtsd server"); */
GST_FLAG_SET (sink, GST_ARTSDSINK_OPEN);
sink->connected = TRUE;
return TRUE;
}
static void
gst_artsdsink_close_audio (GstArtsdsink *sink)
{
if (!sink->connected) return;
arts_close_stream(sink->stream);
arts_free();
GST_FLAG_UNSET (sink, GST_ARTSDSINK_OPEN);
sink->connected = FALSE;
g_print("artsdsink: closed connection\n");
}
static GstElementStateReturn
gst_artsdsink_change_state (GstElement *element)
{
g_return_val_if_fail (GST_IS_ARTSDSINK (element), FALSE);
/* if going down into NULL state, close the stream if it's open */
if (GST_STATE_PENDING (element) == GST_STATE_NULL) {
if (GST_FLAG_IS_SET (element, GST_ARTSDSINK_OPEN))
gst_artsdsink_close_audio (GST_ARTSDSINK (element));
/* otherwise (READY or higher) we need to open the stream */
} else {
if (!GST_FLAG_IS_SET (element, GST_ARTSDSINK_OPEN)) {
if (!gst_artsdsink_open_audio (GST_ARTSDSINK (element)))
return GST_STATE_FAILURE;
}
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}