mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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cda239c8b4
Original commit message from CVS: added two api precondition guards use g_strdup with getenv to fix crash when using ENVVAR
527 lines
14 KiB
C
527 lines
14 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 "esdsink.h"
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#include <esd.h>
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#include <unistd.h>
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#include <errno.h>
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GST_DEBUG_CATEGORY_EXTERN (esd_debug);
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#define GST_CAT_DEFAULT esd_debug
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/* elementfactory information */
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static GstElementDetails esdsink_details = {
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"Esound audio sink",
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"Sink/Audio",
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"Plays audio to an esound 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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{
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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_HOST,
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ARG_SYNC,
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ARG_FALLBACK
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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 ("audio/x-raw-int, "
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"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", "
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"signed = (boolean) TRUE, "
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"width = (int) 16, "
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"depth = (int) 16, " "rate = 44100, " "channels = 2")
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);
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static void gst_esdsink_base_init (gpointer g_class);
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static void gst_esdsink_class_init (gpointer g_class, gpointer class_data);
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static void gst_esdsink_init (GTypeInstance * instance, gpointer g_class);
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static void gst_esdsink_finalize (GObject * object);
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static gboolean gst_esdsink_open_audio (GstEsdsink * sink);
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static void gst_esdsink_close_audio (GstEsdsink * sink);
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static GstElementStateReturn gst_esdsink_change_state (GstElement * element);
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static GstClockTime gst_esdsink_get_time (GstClock * clock, gpointer data);
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static GstClock *gst_esdsink_get_clock (GstElement * element);
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static void gst_esdsink_set_clock (GstElement * element, GstClock * clock);
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static void gst_esdsink_chain (GstPad * pad, GstData * _data);
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static void gst_esdsink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_esdsink_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_esdsink_signals[LAST_SIGNAL] = { 0 }; */
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GType
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gst_esdsink_get_type (void)
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{
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static GType esdsink_type = 0;
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if (!esdsink_type) {
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static const GTypeInfo esdsink_info = {
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sizeof (GstEsdsinkClass),
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gst_esdsink_base_init,
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NULL,
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gst_esdsink_class_init,
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NULL,
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NULL,
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sizeof (GstEsdsink),
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0,
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gst_esdsink_init,
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};
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esdsink_type =
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g_type_register_static (GST_TYPE_ELEMENT, "GstEsdsink", &esdsink_info,
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0);
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}
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return esdsink_type;
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}
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static void
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gst_esdsink_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 (element_class, &esdsink_details);
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}
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static void
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gst_esdsink_class_init (gpointer g_class, gpointer class_data)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (g_class);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
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parent_class = g_type_class_peek_parent (g_class);
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g_object_class_install_property (gobject_class, ARG_MUTE, g_param_spec_boolean ("mute", "mute", "mute", TRUE, G_PARAM_READWRITE)); /* CHECKME */
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g_object_class_install_property (gobject_class, ARG_HOST, g_param_spec_string ("host", "host", "host", NULL, G_PARAM_READWRITE)); /* CHECKME */
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g_object_class_install_property (gobject_class, ARG_SYNC,
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g_param_spec_boolean ("sync", "sync", "Synchronize output to clock",
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TRUE, G_PARAM_READWRITE));
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#if 0
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/* This option is disabled because it is dumb in GStreamer's architecture. */
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g_object_class_install_property (gobject_class, ARG_FALLBACK,
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g_param_spec_boolean ("fallback", "fallback",
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"Fall back to using OSS if Esound daemon is not present", FALSE,
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G_PARAM_READWRITE));
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#endif
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gobject_class->set_property = gst_esdsink_set_property;
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gobject_class->get_property = gst_esdsink_get_property;
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gobject_class->finalize = gst_esdsink_finalize;
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gstelement_class->change_state = gst_esdsink_change_state;
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gstelement_class->set_clock = gst_esdsink_set_clock;
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gstelement_class->get_clock = gst_esdsink_get_clock;
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}
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static void
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gst_esdsink_init (GTypeInstance * instance, gpointer g_class)
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{
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GstEsdsink *esdsink = GST_ESDSINK (instance);
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esdsink->sinkpad =
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gst_pad_new_from_template (gst_element_class_get_pad_template
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(GST_ELEMENT_GET_CLASS (instance), "sink"), "sink");
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gst_element_add_pad (GST_ELEMENT (esdsink), esdsink->sinkpad);
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gst_pad_set_chain_function (esdsink->sinkpad,
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GST_DEBUG_FUNCPTR (gst_esdsink_chain));
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GST_FLAG_SET (esdsink, GST_ELEMENT_EVENT_AWARE);
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esdsink->mute = FALSE;
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esdsink->fd = -1;
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/* FIXME: get default from somewhere better than just putting them inline. */
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/*esdsink->negotiated = FALSE; */
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/* we have static caps on our template, so it always is negotiated */
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esdsink->negotiated = TRUE;
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esdsink->format = 16;
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esdsink->depth = 16;
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esdsink->channels = 2;
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esdsink->frequency = 44100;
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esdsink->bytes_per_sample = esdsink->channels * (esdsink->depth / 8);
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esdsink->host = g_strdup (getenv ("ESPEAKER"));
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esdsink->provided_clock =
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gst_audio_clock_new ("esdclock", gst_esdsink_get_time, esdsink);
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gst_object_set_parent (GST_OBJECT (esdsink->provided_clock),
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GST_OBJECT (esdsink));
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esdsink->sync = TRUE;
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esdsink->fallback = FALSE;
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esdsink->link_open = FALSE;
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}
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static void
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gst_esdsink_finalize (GObject * object)
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{
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GstEsdsink *esdsink = GST_ESDSINK (object);
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g_free (esdsink->host);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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#ifdef unused
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static GstPadLinkReturn
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gst_esdsink_link (GstPad * pad, const GstCaps * caps)
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{
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GstEsdsink *esdsink;
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GstStructure *structure;
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g_return_val_if_fail (caps != NULL, GST_PAD_LINK_REFUSED);
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g_return_val_if_fail (pad != NULL, GST_PAD_LINK_REFUSED);
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esdsink = GST_ESDSINK (gst_pad_get_parent (pad));
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_get_int (structure, "depth", &esdsink->depth);
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gst_structure_get_int (structure, "channels", &esdsink->channels);
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gst_structure_get_int (structure, "rate", &esdsink->frequency);
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esdsink->bytes_per_sample = esdsink->channels * (esdsink->depth / 8);
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if (esdsink->link_open) {
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gst_esdsink_close_audio (esdsink);
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if (gst_esdsink_open_audio (esdsink)) {
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esdsink->negotiated = TRUE;
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esdsink->link_open = TRUE;
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return GST_PAD_LINK_OK;
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}
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}
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/* FIXME: is it supposed to be correct to have closed audio when caps nego
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failed? */
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GST_DEBUG ("esd link function could not negotiate, returning delayed");
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return GST_PAD_LINK_REFUSED;
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}
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#endif
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static GstClockTime
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gst_esdsink_get_time (GstClock * clock, gpointer data)
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{
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GstEsdsink *esdsink = GST_ESDSINK (data);
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GstClockTime res;
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res = (esdsink->handled * GST_SECOND) / esdsink->frequency;
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//- GST_SECOND * 2;
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return res;
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}
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static GstClock *
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gst_esdsink_get_clock (GstElement * element)
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{
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GstEsdsink *esdsink;
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esdsink = GST_ESDSINK (element);
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return GST_CLOCK (esdsink->provided_clock);
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}
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static void
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gst_esdsink_set_clock (GstElement * element, GstClock * clock)
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{
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GstEsdsink *esdsink;
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esdsink = GST_ESDSINK (element);
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esdsink->clock = clock;
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}
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static void
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gst_esdsink_chain (GstPad * pad, GstData * _data)
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{
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GstBuffer *buf = GST_BUFFER (_data);
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GstEsdsink *esdsink;
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esdsink = GST_ESDSINK (gst_pad_get_parent (pad));
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if (!esdsink->negotiated) {
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GST_ELEMENT_ERROR (esdsink, CORE, NEGOTIATION, (NULL),
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("element wasn't negotiated before chain function"));
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goto done;
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}
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if (GST_IS_EVENT (buf)) {
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GstEvent *event = GST_EVENT (buf);
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:
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gst_audio_clock_set_active (GST_AUDIO_CLOCK (esdsink->provided_clock),
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FALSE);
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gst_pad_event_default (pad, event);
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return;
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default:
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gst_pad_event_default (pad, event);
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return;
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}
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gst_event_unref (event);
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return;
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}
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if (GST_BUFFER_DATA (buf) != NULL) {
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if (!esdsink->mute && esdsink->fd >= 0) {
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guchar *data = GST_BUFFER_DATA (buf);
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gint size = GST_BUFFER_SIZE (buf);
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gint to_write = 0;
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to_write = size;
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GST_LOG ("fd=%d data=%p size=%d",
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esdsink->fd, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
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while (to_write > 0) {
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int done;
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done = write (esdsink->fd, data, to_write);
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if (done < 0) {
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if (errno == EINTR) {
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goto done;
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}
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/* connection closed? */
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GST_ELEMENT_ERROR (esdsink, RESOURCE, WRITE, (NULL),
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("communication with ESD failed"));
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return;
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}
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to_write -= done;
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data += done;
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esdsink->handled += done / esdsink->bytes_per_sample;
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}
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}
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}
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gst_audio_clock_update_time ((GstAudioClock *) esdsink->provided_clock,
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gst_esdsink_get_time (esdsink->provided_clock, esdsink));
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done:
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gst_buffer_unref (buf);
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}
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static void
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gst_esdsink_set_property (GObject * object, guint prop_id, const GValue * value,
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GParamSpec * pspec)
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{
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GstEsdsink *esdsink;
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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_ESDSINK (object));
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esdsink = GST_ESDSINK (object);
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switch (prop_id) {
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case ARG_MUTE:
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esdsink->mute = g_value_get_boolean (value);
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break;
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case ARG_HOST:
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g_free (esdsink->host);
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if (g_value_get_string (value) == NULL)
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esdsink->host = NULL;
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else
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esdsink->host = g_strdup (g_value_get_string (value));
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break;
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case ARG_SYNC:
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esdsink->sync = g_value_get_boolean (value);
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break;
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case ARG_FALLBACK:
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esdsink->fallback = g_value_get_boolean (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_esdsink_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec)
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{
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GstEsdsink *esdsink;
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esdsink = GST_ESDSINK (object);
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switch (prop_id) {
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case ARG_MUTE:
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g_value_set_boolean (value, esdsink->mute);
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break;
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case ARG_HOST:
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g_value_set_string (value, esdsink->host);
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break;
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case ARG_SYNC:
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g_value_set_boolean (value, esdsink->sync);
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break;
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case ARG_FALLBACK:
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g_value_set_boolean (value, esdsink->fallback);
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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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gboolean
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gst_esdsink_factory_init (GstPlugin * plugin)
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{
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if (!gst_element_register (plugin, "esdsink", GST_RANK_NONE,
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GST_TYPE_ESDSINK))
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return FALSE;
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return TRUE;
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}
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static gboolean
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gst_esdsink_open_audio (GstEsdsink * sink)
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{
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/* Name used by esound for this connection. */
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const char *connname = "GStreamer";
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/* Bitmap describing audio format. */
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esd_format_t esdformat = ESD_STREAM | ESD_PLAY;
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g_return_val_if_fail (sink->fd == -1, FALSE);
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if (sink->depth == 16)
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esdformat |= ESD_BITS16;
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else if (sink->depth == 8)
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esdformat |= ESD_BITS8;
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else {
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GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL),
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("invalid bit depth (%d)", sink->depth));
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return FALSE;
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}
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if (sink->channels == 2)
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esdformat |= ESD_STEREO;
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else if (sink->channels == 1)
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esdformat |= ESD_MONO;
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else {
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GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL),
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("invalid number of channels (%d)", sink->channels));
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return FALSE;
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}
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GST_INFO ("attempting to open connection to esound server");
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if (sink->fallback) {
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sink->fd =
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esd_play_stream_fallback (esdformat, sink->frequency, sink->host,
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connname);
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} else {
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sink->fd =
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esd_play_stream (esdformat, sink->frequency, sink->host, connname);
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}
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if (sink->fd < 0) {
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GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, (NULL),
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("can't open connection to esound server"));
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return FALSE;
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}
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GST_INFO ("successfully opened connection to esound server");
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return TRUE;
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}
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static void
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gst_esdsink_close_audio (GstEsdsink * sink)
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{
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if (sink->fd < 0)
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return;
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close (sink->fd);
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sink->fd = -1;
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GST_INFO ("esdsink: closed sound device");
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}
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static GstElementStateReturn
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gst_esdsink_change_state (GstElement * element)
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{
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GstEsdsink *esdsink;
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esdsink = GST_ESDSINK (element);
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switch (GST_STATE_TRANSITION (element)) {
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case GST_STATE_NULL_TO_READY:
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break;
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case GST_STATE_READY_TO_PAUSED:
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/* Open and close the link to test it's available */
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if (!esdsink->link_open) {
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if (!gst_esdsink_open_audio (GST_ESDSINK (element))) {
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return GST_STATE_FAILURE;
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}
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gst_esdsink_close_audio (GST_ESDSINK (element));
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}
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break;
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case GST_STATE_PAUSED_TO_PLAYING:
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if (!esdsink->link_open) {
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if (!gst_esdsink_open_audio (GST_ESDSINK (element))) {
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return GST_STATE_FAILURE;
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}
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esdsink->link_open = TRUE;
|
|
}
|
|
gst_audio_clock_set_active (GST_AUDIO_CLOCK (esdsink->provided_clock),
|
|
TRUE);
|
|
break;
|
|
case GST_STATE_PLAYING_TO_PAUSED:
|
|
gst_audio_clock_set_active (GST_AUDIO_CLOCK (esdsink->provided_clock),
|
|
FALSE);
|
|
if (esdsink->link_open) {
|
|
gst_esdsink_close_audio (GST_ESDSINK (element));
|
|
esdsink->link_open = FALSE;
|
|
}
|
|
esdsink->resync = TRUE;
|
|
break;
|
|
case GST_STATE_PAUSED_TO_READY:
|
|
/* Make doubly sure we don't leave our esd connection open
|
|
* or we'll block the other users
|
|
*/
|
|
if (esdsink->link_open) {
|
|
gst_esdsink_close_audio (GST_ESDSINK (element));
|
|
esdsink->link_open = FALSE;
|
|
}
|
|
break;
|
|
case GST_STATE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (GST_ELEMENT_CLASS (parent_class)->change_state)
|
|
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
|
|
|
|
return GST_STATE_SUCCESS;
|
|
}
|