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e69ece9d7e
Original commit message from CVS: remove copyright field from plugins
365 lines
10 KiB
C
365 lines
10 KiB
C
/* GStreamer
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* Copyright (C) <2001> Richard Boulton <richard-gst@tartarus.org>
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*
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* Based on example.c:
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstartsdsink.h"
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/* elementfactory information */
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static GstElementDetails artsdsink_details = {
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"aRtsd audio sink",
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"Sink/Audio",
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"Plays audio to an aRts server",
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"Richard Boulton <richard-gst@tartarus.org>",
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};
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/* Signals and args */
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enum {
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/* FILL ME */
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LAST_SIGNAL
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};
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enum {
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ARG_0,
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ARG_MUTE,
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ARG_NAME,
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};
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GST_PAD_TEMPLATE_FACTORY (sink_factory,
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"sink", /* the name of the pads */
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GST_PAD_SINK, /* type of the pad */
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GST_PAD_ALWAYS, /* ALWAYS/SOMETIMES */
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GST_CAPS_NEW (
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"artsdsink_sink", /* the name of the caps */
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"audio/x-raw-int", /* the mime type of the caps */
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"format", GST_PROPS_STRING ("int"),
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"law", GST_PROPS_INT (0),
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"endianness", GST_PROPS_INT (G_BYTE_ORDER),
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"signed", GST_PROPS_BOOLEAN (FALSE),
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"width", GST_PROPS_LIST (
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GST_PROPS_INT (8),
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GST_PROPS_INT (16)
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),
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"depth", GST_PROPS_LIST (
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GST_PROPS_INT (8),
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GST_PROPS_INT (16)
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),
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"rate", GST_PROPS_INT_RANGE (8000, 96000),
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"channels", GST_PROPS_LIST (
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GST_PROPS_INT (1),
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GST_PROPS_INT (2)
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)
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)
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);
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static void gst_artsdsink_base_init (gpointer g_class);
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static void gst_artsdsink_class_init (GstArtsdsinkClass *klass);
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static void gst_artsdsink_init (GstArtsdsink *artsdsink);
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static gboolean gst_artsdsink_open_audio (GstArtsdsink *sink);
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static void gst_artsdsink_close_audio (GstArtsdsink *sink);
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static GstElementStateReturn gst_artsdsink_change_state (GstElement *element);
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static gboolean gst_artsdsink_sync_parms (GstArtsdsink *artsdsink);
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static GstPadLinkReturn gst_artsdsink_link (GstPad *pad, GstCaps *caps);
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static void gst_artsdsink_chain (GstPad *pad, GstData *_data);
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static void gst_artsdsink_set_property (GObject *object, guint prop_id,
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const GValue *value, GParamSpec *pspec);
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static void gst_artsdsink_get_property (GObject *object, guint prop_id,
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GValue *value, GParamSpec *pspec);
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static GstElementClass *parent_class = NULL;
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/*static guint gst_artsdsink_signals[LAST_SIGNAL] = { 0 }; */
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GType
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gst_artsdsink_get_type (void)
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{
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static GType artsdsink_type = 0;
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if (!artsdsink_type) {
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static const GTypeInfo artsdsink_info = {
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sizeof(GstArtsdsinkClass),
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gst_artsdsink_base_init,
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NULL,
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(GClassInitFunc)gst_artsdsink_class_init,
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NULL,
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NULL,
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sizeof(GstArtsdsink),
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0,
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(GInstanceInitFunc)gst_artsdsink_init,
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};
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artsdsink_type = g_type_register_static(GST_TYPE_ELEMENT, "GstArtsdsink", &artsdsink_info, 0);
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}
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return artsdsink_type;
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}
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static void
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gst_artsdsink_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class, GST_PAD_TEMPLATE_GET (sink_factory));
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gst_element_class_set_details (element_class, &artsdsink_details);
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}
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static void
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gst_artsdsink_class_init (GstArtsdsinkClass *klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass*)klass;
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gstelement_class = (GstElementClass*)klass;
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parent_class = g_type_class_ref(GST_TYPE_ELEMENT);
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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_MUTE,
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g_param_spec_boolean("mute","mute","mute",
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TRUE,G_PARAM_READWRITE)); /* CHECKME */
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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_NAME,
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g_param_spec_string("name","name","name",
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NULL, G_PARAM_READWRITE)); /* CHECKME */
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gobject_class->set_property = gst_artsdsink_set_property;
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gobject_class->get_property = gst_artsdsink_get_property;
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gstelement_class->change_state = gst_artsdsink_change_state;
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}
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static void
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gst_artsdsink_init(GstArtsdsink *artsdsink)
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{
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artsdsink->sinkpad = gst_pad_new_from_template (
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GST_PAD_TEMPLATE_GET (sink_factory), "sink");
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gst_element_add_pad(GST_ELEMENT(artsdsink), artsdsink->sinkpad);
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gst_pad_set_chain_function(artsdsink->sinkpad, gst_artsdsink_chain);
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gst_pad_set_link_function(artsdsink->sinkpad, gst_artsdsink_link);
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artsdsink->connected = FALSE;
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artsdsink->mute = FALSE;
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artsdsink->connect_name = NULL;
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}
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static gboolean
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gst_artsdsink_sync_parms (GstArtsdsink *artsdsink)
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{
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g_return_val_if_fail (artsdsink != NULL, FALSE);
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g_return_val_if_fail (GST_IS_ARTSDSINK (artsdsink), FALSE);
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if (!artsdsink->connected) return TRUE;
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/* Need to set stream to use new parameters: only way to do this is to reopen. */
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gst_artsdsink_close_audio (artsdsink);
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return gst_artsdsink_open_audio (artsdsink);
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}
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static GstPadLinkReturn
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gst_artsdsink_link (GstPad *pad, GstCaps *caps)
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{
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GstArtsdsink *artsdsink = GST_ARTSDSINK (gst_pad_get_parent (pad));
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if (!GST_CAPS_IS_FIXED (caps))
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return GST_PAD_LINK_DELAYED;
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gst_caps_get (caps,
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"rate", &artsdsink->frequency,
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"depth", &artsdsink->depth,
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"signed", &artsdsink->signd,
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"channels", &artsdsink->channels,
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NULL);
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if (gst_artsdsink_sync_parms (artsdsink))
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return GST_PAD_LINK_OK;
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return GST_PAD_LINK_REFUSED;
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}
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static void
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gst_artsdsink_chain (GstPad *pad, GstData *_data)
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{
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GstBuffer *buf = GST_BUFFER (_data);
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GstArtsdsink *artsdsink;
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g_return_if_fail(pad != NULL);
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g_return_if_fail(GST_IS_PAD(pad));
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g_return_if_fail(buf != NULL);
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artsdsink = GST_ARTSDSINK (gst_pad_get_parent (pad));
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if (GST_BUFFER_DATA (buf) != NULL) {
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gst_trace_add_entry(NULL, 0, GPOINTER_TO_INT(buf), "artsdsink: writing to server");
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if (!artsdsink->mute && artsdsink->connected) {
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int bytes;
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void * bufptr = GST_BUFFER_DATA (buf);
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int bufsize = GST_BUFFER_SIZE (buf);
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GST_DEBUG ("artsdsink: stream=%p data=%p size=%d",
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artsdsink->stream, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
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do {
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bytes = arts_write (artsdsink->stream, bufptr, bufsize);
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if(bytes < 0) {
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fprintf(stderr,"arts_write error: %s\n", arts_error_text(bytes));
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gst_buffer_unref (buf);
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return;
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}
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bufptr += bytes;
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bufsize -= bytes;
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} while (bufsize > 0);
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}
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}
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gst_buffer_unref (buf);
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}
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static void
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gst_artsdsink_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
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{
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GstArtsdsink *artsdsink;
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/* it's not null if we got it, but it might not be ours */
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g_return_if_fail(GST_IS_ARTSDSINK(object));
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artsdsink = GST_ARTSDSINK(object);
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switch (prop_id) {
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case ARG_MUTE:
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artsdsink->mute = g_value_get_boolean (value);
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break;
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case ARG_NAME:
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if (artsdsink->connect_name != NULL) g_free(artsdsink->connect_name);
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if (g_value_get_string (value) == NULL)
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artsdsink->connect_name = NULL;
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else
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artsdsink->connect_name = g_strdup (g_value_get_string (value));
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break;
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default:
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break;
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}
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}
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static void
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gst_artsdsink_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
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{
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GstArtsdsink *artsdsink;
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/* it's not null if we got it, but it might not be ours */
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g_return_if_fail(GST_IS_ARTSDSINK(object));
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artsdsink = GST_ARTSDSINK(object);
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switch (prop_id) {
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case ARG_MUTE:
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g_value_set_boolean (value, artsdsink->mute);
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break;
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case ARG_NAME:
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g_value_set_string (value, artsdsink->connect_name);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static gboolean
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plugin_init (GstPlugin *plugin)
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{
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if (!gst_element_register (plugin, "artsdsink", GST_RANK_NONE, GST_TYPE_ARTSDSINK))
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return FALSE;
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return TRUE;
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}
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GST_PLUGIN_DEFINE (
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GST_VERSION_MAJOR,
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GST_VERSION_MINOR,
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"artsdsink",
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"Plays audio to an aRts server",
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plugin_init,
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VERSION,
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"LGPL",
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GST_PACKAGE,
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GST_ORIGIN)
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static gboolean
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gst_artsdsink_open_audio (GstArtsdsink *sink)
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{
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const char * connname = "gstreamer";
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int errcode;
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/* Name used by aRtsd for this connection. */
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if (sink->connect_name != NULL) connname = sink->connect_name;
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/* FIXME: this should only ever happen once per process. */
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/* Really, artsc needs to be made thread safe to fix this (and other related */
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/* problems). */
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errcode = arts_init();
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if(errcode < 0) {
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fprintf(stderr,"arts_init error: %s\n", arts_error_text(errcode));
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return FALSE;
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}
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GST_DEBUG ("artsdsink: attempting to open connection to aRtsd server");
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sink->stream = arts_play_stream(sink->frequency, sink->depth,
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sink->channels, connname);
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/* FIXME: check connection */
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/* GST_DEBUG ("artsdsink: can't open connection to aRtsd server"); */
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GST_FLAG_SET (sink, GST_ARTSDSINK_OPEN);
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sink->connected = TRUE;
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return TRUE;
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}
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static void
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gst_artsdsink_close_audio (GstArtsdsink *sink)
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{
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if (!sink->connected) return;
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arts_close_stream(sink->stream);
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arts_free();
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GST_FLAG_UNSET (sink, GST_ARTSDSINK_OPEN);
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sink->connected = FALSE;
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g_print("artsdsink: closed connection\n");
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}
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static GstElementStateReturn
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gst_artsdsink_change_state (GstElement *element)
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{
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g_return_val_if_fail (GST_IS_ARTSDSINK (element), FALSE);
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/* if going down into NULL state, close the stream if it's open */
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if (GST_STATE_PENDING (element) == GST_STATE_NULL) {
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if (GST_FLAG_IS_SET (element, GST_ARTSDSINK_OPEN))
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gst_artsdsink_close_audio (GST_ARTSDSINK (element));
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/* otherwise (READY or higher) we need to open the stream */
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} else {
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if (!GST_FLAG_IS_SET (element, GST_ARTSDSINK_OPEN)) {
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if (!gst_artsdsink_open_audio (GST_ARTSDSINK (element)))
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return GST_STATE_FAILURE;
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}
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}
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if (GST_ELEMENT_CLASS (parent_class)->change_state)
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return GST_ELEMENT_CLASS (parent_class)->change_state (element);
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return GST_STATE_SUCCESS;
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}
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