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348 lines
10 KiB
C
348 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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#include <gst/audio/audio.h>
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/* Signals and args */
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enum
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{
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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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{
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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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static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS)
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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 GstStateChangeReturn gst_artsdsink_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_artsdsink_sync_parms (GstArtsdsink * artsdsink);
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static GstPadLinkReturn gst_artsdsink_link (GstPad * pad, const 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 =
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g_type_register_static (GST_TYPE_ELEMENT, "GstArtsdsink",
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&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,
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gst_static_pad_template_get (&sink_factory));
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gst_element_class_set_details_simple (element_class, "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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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_peek_parent (klass);
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/* FIXME: add long property descriptions */
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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", TRUE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/* FIXME: rename to e.g. "client-name" */
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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", NULL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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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 =
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gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT
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(artsdsink), "sink"), "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)
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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, const GstCaps * caps)
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{
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GstArtsdsink *artsdsink = GST_ARTSDSINK (gst_pad_get_parent (pad));
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GstStructure *structure;
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_get_int (structure, "rate", &artsdsink->frequency);
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gst_structure_get_int (structure, "depth", &artsdsink->depth);
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gst_structure_get_int (structure, "signed", &artsdsink->signd);
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gst_structure_get_int (structure, "channels", &artsdsink->channels);
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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),
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"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,
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const GValue * value, GParamSpec * pspec)
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{
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GstArtsdsink *artsdsink;
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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)
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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,
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GParamSpec * pspec)
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{
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GstArtsdsink *artsdsink;
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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,
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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 (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, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
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static gboolean 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)
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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_OBJECT_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)
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return;
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arts_close_stream (sink->stream);
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arts_free ();
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GST_OBJECT_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 GstStateChangeReturn
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gst_artsdsink_change_state (GstElement * element, GstStateChange transition)
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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_OBJECT_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_OBJECT_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_CHANGE_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, transition);
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return GST_STATE_CHANGE_SUCCESS;
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}
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