mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-20 07:16:55 +00:00
191c8742d3
This fixes issues where wavparse would query the file size upstream and assert because the file size is way smaller then what the WAVE header says. This patch disable or cane a handful of queries that make no sense to forward. https://bugzilla.gnome.org/show_bug.cgi?id=791811
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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* <ulink url="http://www.festvox.org/festival/index.html">festival</ulink>
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* server process and uses it to synthesize speech. Festival need to run already
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* in server mode, started as <screen>festival --server</screen>
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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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|
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return buff;
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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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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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|
|
|
if (festival->info->server_fd != -1)
|
|
close (festival->info->server_fd);
|
|
GST_OBJECT_FLAG_UNSET (festival, GST_FESTIVAL_OPEN);
|
|
return;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
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gst_festival_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
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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;
|
|
}
|
|
}
|
|
|
|
if (GST_ELEMENT_CLASS (gst_festival_parent_class)->change_state)
|
|
return GST_ELEMENT_CLASS (gst_festival_parent_class)->change_state (element,
|
|
transition);
|
|
|
|
return GST_STATE_CHANGE_SUCCESS;
|
|
}
|
|
|
|
static gboolean
|
|
gst_festival_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
|
|
{
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_POSITION:
|
|
/* Not supported */
|
|
return FALSE;
|
|
case GST_QUERY_DURATION:
|
|
gst_query_set_duration (query, GST_FORMAT_BYTES, -1);
|
|
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);
|