mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-24 02:31:03 +00:00
6863dd9240
Original commit message from CVS: * a hack to work around intltool's brokenness * a current check for mpeg2dec * details->klass reorganizations * an element browser that uses details->klass * separated cdxa parse out from the avi directory
408 lines
12 KiB
C
408 lines
12 KiB
C
/* 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.
|
|
*/
|
|
|
|
#include "gstartsdsink.h"
|
|
|
|
/* elementfactory information */
|
|
static GstElementDetails artsdsink_details = {
|
|
"aRtsd audio sink",
|
|
"Sink/Audio",
|
|
"Plays audio to an aRts server",
|
|
VERSION,
|
|
"Richard Boulton <richard-gst@tartarus.org>",
|
|
"(C) 2001",
|
|
};
|
|
|
|
/* Signals and args */
|
|
enum {
|
|
/* FILL ME */
|
|
LAST_SIGNAL
|
|
};
|
|
|
|
enum {
|
|
ARG_0,
|
|
ARG_MUTE,
|
|
ARG_DEPTH,
|
|
ARG_CHANNELS,
|
|
ARG_RATE,
|
|
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/raw", /* 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_INT (8),
|
|
"depth", GST_PROPS_INT (8),
|
|
"rate", GST_PROPS_INT_RANGE (8000, 96000),
|
|
"channels", GST_PROPS_LIST (GST_PROPS_INT (1), GST_PROPS_INT (2))
|
|
),
|
|
GST_CAPS_NEW (
|
|
"artsdsink_sink", /* the name of the caps */
|
|
"audio/raw", /* 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 (TRUE),
|
|
"width", GST_PROPS_INT (16),
|
|
"depth", 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_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 void gst_artsdsink_chain (GstPad *pad, GstBuffer *buf);
|
|
|
|
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);
|
|
|
|
#define GST_TYPE_ARTSDSINK_DEPTHS (gst_artsdsink_depths_get_type())
|
|
static GType
|
|
gst_artsdsink_depths_get_type (void)
|
|
{
|
|
static GType artsdsink_depths_type = 0;
|
|
static GEnumValue artsdsink_depths[] = {
|
|
{8, "8", "8 Bits"},
|
|
{16, "16", "16 Bits"},
|
|
{0, NULL, NULL},
|
|
};
|
|
if (!artsdsink_depths_type) {
|
|
artsdsink_depths_type = g_enum_register_static("GstArtsdsinkDepths", artsdsink_depths);
|
|
}
|
|
return artsdsink_depths_type;
|
|
}
|
|
|
|
#define GST_TYPE_ARTSDSINK_CHANNELS (gst_artsdsink_channels_get_type())
|
|
static GType
|
|
gst_artsdsink_channels_get_type (void)
|
|
{
|
|
static GType artsdsink_channels_type = 0;
|
|
static GEnumValue artsdsink_channels[] = {
|
|
{1, "1", "Mono"},
|
|
{2, "2", "Stereo"},
|
|
{0, NULL, NULL},
|
|
};
|
|
if (!artsdsink_channels_type) {
|
|
artsdsink_channels_type = g_enum_register_static("GstArtsdsinkChannels", artsdsink_channels);
|
|
}
|
|
return artsdsink_channels_type;
|
|
}
|
|
|
|
|
|
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), NULL,
|
|
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_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_DEPTH,
|
|
g_param_spec_enum("depth","depth","depth",
|
|
GST_TYPE_ARTSDSINK_DEPTHS,16,G_PARAM_READWRITE)); /* CHECKME! */
|
|
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_CHANNELS,
|
|
g_param_spec_enum("channels","channels","channels",
|
|
GST_TYPE_ARTSDSINK_CHANNELS,2,G_PARAM_READWRITE)); /* CHECKME! */
|
|
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_RATE,
|
|
g_param_spec_int("frequency","frequency","frequency",
|
|
G_MININT,G_MAXINT,0,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);
|
|
|
|
artsdsink->connected = FALSE;
|
|
artsdsink->mute = FALSE;
|
|
|
|
/* FIXME: get default from somewhere better than just putting them inline. */
|
|
artsdsink->signd = TRUE;
|
|
artsdsink->depth = 16;
|
|
artsdsink->channels = 2;
|
|
artsdsink->frequency = 44100;
|
|
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 void
|
|
gst_artsdsink_chain (GstPad *pad, GstBuffer *buf)
|
|
{
|
|
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, 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 (0, "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_DEPTH:
|
|
artsdsink->depth = g_value_get_enum (value);
|
|
gst_artsdsink_sync_parms (artsdsink);
|
|
break;
|
|
case ARG_CHANNELS:
|
|
artsdsink->channels = g_value_get_enum (value);
|
|
gst_artsdsink_sync_parms (artsdsink);
|
|
break;
|
|
case ARG_RATE:
|
|
artsdsink->frequency = g_value_get_int (value);
|
|
gst_artsdsink_sync_parms (artsdsink);
|
|
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_DEPTH:
|
|
g_value_set_enum (value, artsdsink->depth);
|
|
break;
|
|
case ARG_CHANNELS:
|
|
g_value_set_enum (value, artsdsink->channels);
|
|
break;
|
|
case ARG_RATE:
|
|
g_value_set_int (value, artsdsink->frequency);
|
|
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 (GModule *module, GstPlugin *plugin)
|
|
{
|
|
GstElementFactory *factory;
|
|
|
|
factory = gst_element_factory_new("artsdsink", GST_TYPE_ARTSDSINK,
|
|
&artsdsink_details);
|
|
g_return_val_if_fail(factory != NULL, FALSE);
|
|
|
|
gst_element_factory_add_pad_template(factory, GST_PAD_TEMPLATE_GET (sink_factory));
|
|
|
|
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
GstPluginDesc plugin_desc = {
|
|
GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
"artsdsink",
|
|
plugin_init
|
|
};
|
|
|
|
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 (0, "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 (0, "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;
|
|
}
|
|
|