/* GStreamer * Copyright (C) <2001> Richard Boulton * * Based on example.c: * Copyright (C) <1999> Erik Walthinsen * * 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", "LGPL", "Plays audio to an aRts server", VERSION, "Richard Boulton ", "(C) 2001", }; /* 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_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, 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); 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_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, 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, 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 (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 ("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; }