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541 lines
16 KiB
C
541 lines
16 KiB
C
/* GStreamer
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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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/*************************************************************************/
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/* */
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/* Centre for Speech Technology Research */
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/* University of Edinburgh, UK */
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/* Copyright (c) 1999 */
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/* All Rights Reserved. */
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/* */
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/* Permission is hereby granted, free of charge, to use and distribute */
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/* this software and its documentation without restriction, including */
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/* without limitation the rights to use, copy, modify, merge, publish, */
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/* distribute, sublicense, and/or sell copies of this work, and to */
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/* permit persons to whom this work is furnished to do so, subject to */
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/* the following conditions: */
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/* 1. The code must retain the above copyright notice, this list of */
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/* conditions and the following disclaimer. */
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/* 2. Any modifications must be clearly marked as such. */
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/* 3. Original authors' names are not deleted. */
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/* 4. The authors' names are not used to endorse or promote products */
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/* derived from this software without specific prior written */
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/* permission. */
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/* */
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/* THE UNIVERSITY OF EDINBURGH AND THE CONTRIBUTORS TO THIS WORK */
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/* DISCLAIM ALL WARRANTIES WITH REGARD TO THIS SOFTWARE, INCLUDING */
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/* ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS, IN NO EVENT */
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/* SHALL THE UNIVERSITY OF EDINBURGH NOR THE CONTRIBUTORS BE LIABLE */
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/* FOR ANY SPECIAL, INDIRECT OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES */
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/* WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN */
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/* AN ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, */
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/* ARISING OUT OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF */
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/* THIS SOFTWARE. */
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/* */
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/*************************************************************************/
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/* Author : Alan W Black (awb@cstr.ed.ac.uk) */
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/* Date : March 1999 */
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/*-----------------------------------------------------------------------*/
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/* */
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/* Client end of Festival server API in C designed specifically for */
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/* Galaxy Communicator use though might be of use for other things */
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/* */
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/* This is a modified version of the standalone client as provided in */
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/* festival example code: festival_client.c */
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/* */
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/*=======================================================================*/
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/**
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* SECTION:element-festival
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* @title: festival
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*
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* This element connects to a
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* [festival](http://www.festvox.org/festival/index.html) server process and
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* uses it to synthesize speech. Festival need to run already in server mode,
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* started as `festival --server`
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*
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* ## Example pipeline
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* |[
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* echo 'Hello G-Streamer!' | gst-launch-1.0 fdsrc fd=0 ! festival ! wavparse ! audioconvert ! alsasink
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* ]|
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*
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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 <glib.h> /* Needed for G_OS_XXXX macros */
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#include <stdio.h>
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#include <stdlib.h>
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#ifdef HAVE_UNISTD_H
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#include <unistd.h>
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#endif
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#include <ctype.h>
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#include <string.h>
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#include <sys/types.h>
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#ifdef G_OS_WIN32
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#include <winsock2.h>
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#include <ws2tcpip.h>
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#else
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#include <sys/socket.h>
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#include <netdb.h>
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#include <netinet/in.h>
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#include <arpa/inet.h>
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#endif
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#include "gstfestival.h"
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#include <gst/audio/audio.h>
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GST_DEBUG_CATEGORY_STATIC (festival_debug);
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#define GST_CAT_DEFAULT festival_debug
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static void gst_festival_finalize (GObject * object);
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static GstFlowReturn gst_festival_chain (GstPad * pad, GstObject * parent,
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GstBuffer * buf);
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static gboolean gst_festival_src_query (GstPad * pad, GstObject * parent,
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GstQuery * query);
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static GstStateChangeReturn gst_festival_change_state (GstElement * element,
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GstStateChange transition);
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static FT_Info *festival_default_info (void);
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static char *socket_receive_file_to_buff (int fd, int *size);
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static char *client_accept_s_expr (int fd);
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static GstStaticPadTemplate sink_template_factory =
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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 ("text/x-raw, format=(string)utf8")
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);
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static GstStaticPadTemplate src_template_factory =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-wav")
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);
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/* Festival 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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PROP_0
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/* FILL ME */
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};
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/*static guint gst_festival_signals[LAST_SIGNAL] = { 0 }; */
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G_DEFINE_TYPE (GstFestival, gst_festival, GST_TYPE_ELEMENT)
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static void gst_festival_class_init (GstFestivalClass * 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 = G_OBJECT_CLASS (klass);
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gstelement_class = GST_ELEMENT_CLASS (klass);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_festival_finalize);
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_festival_change_state);
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/* register pads */
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gst_element_class_add_static_pad_template (gstelement_class,
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&sink_template_factory);
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gst_element_class_add_static_pad_template (gstelement_class,
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&src_template_factory);
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gst_element_class_set_static_metadata (gstelement_class,
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"Festival Text-to-Speech synthesizer", "Filter/Effect/Audio",
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"Synthesizes plain text into audio",
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"Wim Taymans <wim.taymans@gmail.com>");
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}
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static void
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gst_festival_init (GstFestival * festival)
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{
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festival->sinkpad =
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gst_pad_new_from_static_template (&sink_template_factory, "sink");
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gst_pad_set_chain_function (festival->sinkpad, gst_festival_chain);
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gst_element_add_pad (GST_ELEMENT (festival), festival->sinkpad);
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festival->srcpad =
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gst_pad_new_from_static_template (&src_template_factory, "src");
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gst_pad_set_query_function (festival->srcpad, gst_festival_src_query);
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gst_element_add_pad (GST_ELEMENT (festival), festival->srcpad);
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festival->info = festival_default_info ();
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}
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static void
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gst_festival_finalize (GObject * object)
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{
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GstFestival *festival = GST_FESTIVAL (object);
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g_free (festival->info);
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G_OBJECT_CLASS (gst_festival_parent_class)->finalize (object);
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}
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static gboolean
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read_response (GstFestival * festival)
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{
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char ack[4];
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char *data;
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int filesize;
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int fd;
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int n;
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gboolean ret = TRUE;
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fd = festival->info->server_fd;
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do {
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for (n = 0; n < 3;)
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n += read (fd, ack + n, 3 - n);
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ack[3] = '\0';
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GST_DEBUG_OBJECT (festival, "got response %s", ack);
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if (strcmp (ack, "WV\n") == 0) {
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GstBuffer *buffer;
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/* receive a waveform */
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data = socket_receive_file_to_buff (fd, &filesize);
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GST_DEBUG_OBJECT (festival, "received %d bytes of waveform data",
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filesize);
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/* push contents as a buffer */
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buffer = gst_buffer_new_wrapped (data, filesize);
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GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
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gst_pad_push (festival->srcpad, buffer);
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} else if (strcmp (ack, "LP\n") == 0) {
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/* receive an s-expr */
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data = client_accept_s_expr (fd);
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GST_DEBUG_OBJECT (festival, "received s-expression: %s", data);
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g_free (data);
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} else if (strcmp (ack, "ER\n") == 0) {
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/* server got an error */
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GST_ELEMENT_ERROR (festival,
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LIBRARY,
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FAILED,
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("Festival speech server returned an error"),
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("Make sure you have voices/languages installed"));
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ret = FALSE;
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break;
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}
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} while (strcmp (ack, "OK\n") != 0);
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return ret;
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}
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static GstFlowReturn
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gst_festival_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
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{
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GstFlowReturn ret = GST_FLOW_OK;
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GstFestival *festival;
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GstMapInfo info;
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guint8 *p, *ep;
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gint f;
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FILE *fd;
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festival = GST_FESTIVAL (parent);
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GST_LOG_OBJECT (festival, "Got text buffer, %" G_GSIZE_FORMAT " bytes",
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gst_buffer_get_size (buf));
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f = dup (festival->info->server_fd);
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if (f < 0)
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goto fail_open;
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fd = fdopen (f, "wb");
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if (fd == NULL) {
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close (f);
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goto fail_open;
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}
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/* Copy text over to server, escaping any quotes */
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fprintf (fd, "(Parameter.set 'Audio_Required_Rate 16000)\n");
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fflush (fd);
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GST_DEBUG_OBJECT (festival, "issued Parameter.set command");
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if (read_response (festival) == FALSE) {
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fclose (fd);
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goto fail_read;
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}
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fprintf (fd, "(tts_textall \"");
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gst_buffer_map (buf, &info, GST_MAP_READ);
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p = info.data;
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ep = p + info.size;
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for (; p < ep && (*p != '\0'); p++) {
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if ((*p == '"') || (*p == '\\')) {
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putc ('\\', fd);
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}
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putc (*p, fd);
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}
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fprintf (fd, "\" \"%s\")\n", festival->info->text_mode);
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fclose (fd);
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gst_buffer_unmap (buf, &info);
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GST_DEBUG_OBJECT (festival, "issued tts_textall command");
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/* Read back info from server */
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if (read_response (festival) == FALSE)
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goto fail_read;
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out:
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gst_buffer_unref (buf);
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return ret;
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/* ERRORS */
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fail_open:
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{
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GST_ELEMENT_ERROR (festival, RESOURCE, OPEN_WRITE, (NULL), (NULL));
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ret = GST_FLOW_ERROR;
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goto out;
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}
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fail_read:
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{
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GST_ELEMENT_ERROR (festival, RESOURCE, READ, (NULL), (NULL));
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ret = GST_FLOW_ERROR;
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goto out;
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}
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}
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static FT_Info *
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festival_default_info (void)
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{
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FT_Info *info;
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info = (FT_Info *) malloc (1 * sizeof (FT_Info));
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info->server_host = FESTIVAL_DEFAULT_SERVER_HOST;
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info->server_port = FESTIVAL_DEFAULT_SERVER_PORT;
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info->text_mode = FESTIVAL_DEFAULT_TEXT_MODE;
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info->server_fd = -1;
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return info;
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}
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static int
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festival_socket_open (const char *host, int port)
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{
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/* Return an FD to a remote server */
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struct sockaddr_in serv_addr;
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struct hostent *serverhost;
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int fd;
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fd = socket (AF_INET, SOCK_STREAM, IPPROTO_TCP);
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if (fd < 0) {
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fprintf (stderr, "festival_client: can't get socket\n");
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return -1;
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}
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memset (&serv_addr, 0, sizeof (serv_addr));
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if ((serv_addr.sin_addr.s_addr = inet_addr (host)) == -1) {
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/* its a name rather than an ipnum */
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serverhost = gethostbyname (host);
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if (serverhost == (struct hostent *) 0) {
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fprintf (stderr, "festival_client: gethostbyname failed\n");
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close (fd);
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return -1;
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}
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memmove (&serv_addr.sin_addr, serverhost->h_addr, serverhost->h_length);
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}
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serv_addr.sin_family = AF_INET;
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serv_addr.sin_port = htons (port);
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if (connect (fd, (struct sockaddr *) &serv_addr, sizeof (serv_addr)) != 0) {
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fprintf (stderr, "festival_client: connect to server failed\n");
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close (fd);
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return -1;
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}
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return fd;
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}
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static char *
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client_accept_s_expr (int fd)
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{
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/* Read s-expression from server, as a char * */
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char *expr;
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int filesize;
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expr = socket_receive_file_to_buff (fd, &filesize);
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expr[filesize] = '\0';
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return expr;
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}
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static char *
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socket_receive_file_to_buff (int fd, int *size)
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{
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/* Receive file (probably a waveform file) from socket using */
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/* Festival key stuff technique, but long winded I know, sorry */
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/* but will receive any file without closeing the stream or */
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/* using OOB data */
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static const char file_stuff_key[] = "ft_StUfF_key"; /* must == Festival's key */
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char *buff;
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int bufflen;
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int n, k, i;
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char c;
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bufflen = 1024;
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buff = (char *) g_malloc (bufflen);
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*size = 0;
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for (k = 0; file_stuff_key[k] != '\0';) {
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n = read (fd, &c, 1);
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if (n == 0)
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break; /* hit stream eof before end of file */
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if ((*size) + k + 1 >= bufflen) {
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/* +1 so you can add a NULL if you want */
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bufflen += bufflen / 4;
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buff = (char *) g_realloc (buff, bufflen);
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}
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if (file_stuff_key[k] == c)
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k++;
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else if ((c == 'X') && (file_stuff_key[k + 1] == '\0')) {
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/* It looked like the key but wasn't */
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for (i = 0; i < k; i++, (*size)++)
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buff[*size] = file_stuff_key[i];
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k = 0;
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/* omit the stuffed 'X' */
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} else {
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for (i = 0; i < k; i++, (*size)++)
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buff[*size] = file_stuff_key[i];
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k = 0;
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buff[*size] = c;
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(*size)++;
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}
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}
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return buff;
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}
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|
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/***********************************************************************/
|
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/* Public Functions to this API */
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/***********************************************************************/
|
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|
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static gboolean
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gst_festival_open (GstFestival * festival)
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{
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/* Open socket to server */
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if (festival->info == NULL)
|
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festival->info = festival_default_info ();
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|
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festival->info->server_fd =
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festival_socket_open (festival->info->server_host,
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festival->info->server_port);
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if (festival->info->server_fd == -1) {
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GST_ERROR
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("Could not talk to festival server (no server running or wrong host/port?)");
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return FALSE;
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}
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GST_OBJECT_FLAG_SET (festival, GST_FESTIVAL_OPEN);
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return TRUE;
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}
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|
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static void
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gst_festival_close (GstFestival * festival)
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{
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if (festival->info == NULL)
|
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return;
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|
|
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if (festival->info->server_fd != -1)
|
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close (festival->info->server_fd);
|
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GST_OBJECT_FLAG_UNSET (festival, GST_FESTIVAL_OPEN);
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return;
|
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}
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|
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static GstStateChangeReturn
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gst_festival_change_state (GstElement * element, GstStateChange transition)
|
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{
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g_return_val_if_fail (GST_IS_FESTIVAL (element), GST_STATE_CHANGE_FAILURE);
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|
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if (GST_STATE_PENDING (element) == GST_STATE_NULL) {
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if (GST_OBJECT_FLAG_IS_SET (element, GST_FESTIVAL_OPEN)) {
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GST_DEBUG ("Closing connection ");
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gst_festival_close (GST_FESTIVAL (element));
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}
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} else {
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if (!GST_OBJECT_FLAG_IS_SET (element, GST_FESTIVAL_OPEN)) {
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GST_DEBUG ("Opening connection ");
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if (!gst_festival_open (GST_FESTIVAL (element)))
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return GST_STATE_CHANGE_FAILURE;
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}
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}
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|
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if (GST_ELEMENT_CLASS (gst_festival_parent_class)->change_state)
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return GST_ELEMENT_CLASS (gst_festival_parent_class)->change_state (element,
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transition);
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|
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return GST_STATE_CHANGE_SUCCESS;
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}
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|
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static gboolean
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gst_festival_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
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{
|
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_POSITION:
|
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/* Not supported */
|
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return FALSE;
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case GST_QUERY_DURATION:
|
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gst_query_set_duration (query, GST_FORMAT_BYTES, -1);
|
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return TRUE;
|
|
case GST_QUERY_SEEKING:
|
|
gst_query_set_seeking (query, GST_FORMAT_BYTES, FALSE, 0, -1);
|
|
return TRUE;
|
|
case GST_QUERY_FORMATS:
|
|
gst_query_set_formats (query, 1, GST_FORMAT_BYTES);
|
|
return TRUE;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return gst_pad_query_default (pad, parent, query);
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
GST_DEBUG_CATEGORY_INIT (festival_debug, "festival",
|
|
0, "Festival text-to-speech synthesizer");
|
|
|
|
if (!gst_element_register (plugin, "festival", GST_RANK_NONE,
|
|
GST_TYPE_FESTIVAL))
|
|
return FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
festival,
|
|
"Synthesizes plain text into audio",
|
|
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
|