/* 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" #include <gst/audio/audio.h> /* Signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { ARG_0, ARG_MUTE, ARG_NAME }; static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS) ); 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 GstStateChangeReturn gst_artsdsink_change_state (GstElement * element, GstStateChange transition); static gboolean gst_artsdsink_sync_parms (GstArtsdsink * artsdsink); static GstPadLinkReturn gst_artsdsink_link (GstPad * pad, const 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_static_pad_template (element_class, &sink_factory); gst_element_class_set_details_simple (element_class, "aRtsd audio sink", "Sink/Audio", "Plays audio to an aRts server", "Richard Boulton <richard-gst@tartarus.org>"); } 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_peek_parent (klass); /* FIXME: add long property descriptions */ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MUTE, g_param_spec_boolean ("mute", "mute", "mute", TRUE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /* FIXME: rename to e.g. "client-name" */ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_NAME, g_param_spec_string ("name", "name", "name", NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); 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_element_get_pad_template (GST_ELEMENT (artsdsink), "sink"), "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, const GstCaps * caps) { GstArtsdsink *artsdsink = GST_ARTSDSINK (gst_pad_get_parent (pad)); GstStructure *structure; structure = gst_caps_get_structure (caps, 0); gst_structure_get_int (structure, "rate", &artsdsink->frequency); gst_structure_get_int (structure, "depth", &artsdsink->depth); gst_structure_get_int (structure, "signed", &artsdsink->signd); gst_structure_get_int (structure, "channels", &artsdsink->channels); 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; 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; 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_NAME, GST_PACKAGE_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_OBJECT_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_OBJECT_FLAG_UNSET (sink, GST_ARTSDSINK_OPEN); sink->connected = FALSE; g_print ("artsdsink: closed connection\n"); } static GstStateChangeReturn gst_artsdsink_change_state (GstElement * element, GstStateChange transition) { 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_OBJECT_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_OBJECT_FLAG_IS_SET (element, GST_ARTSDSINK_OPEN)) { if (!gst_artsdsink_open_audio (GST_ARTSDSINK (element))) return GST_STATE_CHANGE_FAILURE; } } if (GST_ELEMENT_CLASS (parent_class)->change_state) return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); return GST_STATE_CHANGE_SUCCESS; }