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gstreamer/gst/festival/gstfestival.c
Vineeth TM 7c42ba97d7 plugins-bad: Fix example pipelines 
rename gst-launch --> gst-launch-1.0
replace old elements with new elements(ffmpegcolorspace -> videoconvert, ffenc_** -> avenc_**)
fix caps in examples

https://bugzilla.gnome.org/show_bug.cgi?id=759432
2015-12-15 10:30:49 +00:00

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/*************************************************************************/
/* */
/* Centre for Speech Technology Research */
/* University of Edinburgh, UK */
/* Copyright (c) 1999 */
/* All Rights Reserved. */
/* */
/* Permission is hereby granted, free of charge, to use and distribute */
/* this software and its documentation without restriction, including */
/* without limitation the rights to use, copy, modify, merge, publish, */
/* distribute, sublicense, and/or sell copies of this work, and to */
/* permit persons to whom this work is furnished to do so, subject to */
/* the following conditions: */
/* 1. The code must retain the above copyright notice, this list of */
/* conditions and the following disclaimer. */
/* 2. Any modifications must be clearly marked as such. */
/* 3. Original authors' names are not deleted. */
/* 4. The authors' names are not used to endorse or promote products */
/* derived from this software without specific prior written */
/* permission. */
/* */
/* THE UNIVERSITY OF EDINBURGH AND THE CONTRIBUTORS TO THIS WORK */
/* DISCLAIM ALL WARRANTIES WITH REGARD TO THIS SOFTWARE, INCLUDING */
/* ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS, IN NO EVENT */
/* SHALL THE UNIVERSITY OF EDINBURGH NOR THE CONTRIBUTORS BE LIABLE */
/* FOR ANY SPECIAL, INDIRECT OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES */
/* WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN */
/* AN ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, */
/* ARISING OUT OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF */
/* THIS SOFTWARE. */
/* */
/*************************************************************************/
/* Author : Alan W Black (awb@cstr.ed.ac.uk) */
/* Date : March 1999 */
/*-----------------------------------------------------------------------*/
/* */
/* Client end of Festival server API in C designed specifically for */
/* Galaxy Communicator use though might be of use for other things */
/* */
/* This is a modified version of the standalone client as provided in */
/* festival example code: festival_client.c */
/* */
/*=======================================================================*/
/**
* SECTION:element-festival
*
* This element connects to a
* <ulink url="http://www.festvox.org/festival/index.html">festival</ulink>
* server process and uses it to synthesize speech. Festival need to run already
* in server mode, started as <screen>festival --server</screen>
*
* <refsect2>
* <title>Example pipeline</title>
* |[
* echo 'Hello G-Streamer!' | gst-launch-1.0 fdsrc fd=0 ! festival ! wavparse ! audioconvert ! alsasink
* ]|
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <glib.h> /* Needed for G_OS_XXXX macros */
#include <stdio.h>
#include <stdlib.h>
#ifdef HAVE_UNISTD_H
#include <unistd.h>
#endif
#include <ctype.h>
#include <string.h>
#include <sys/types.h>
#ifdef G_OS_WIN32
#include <winsock2.h>
#include <ws2tcpip.h>
#else
#include <sys/socket.h>
#include <netdb.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#endif
#include "gstfestival.h"
#include <gst/audio/audio.h>
GST_DEBUG_CATEGORY_STATIC (festival_debug);
#define GST_CAT_DEFAULT festival_debug
static void gst_festival_finalize (GObject * object);
static GstFlowReturn gst_festival_chain (GstPad * pad, GstObject * parent,
GstBuffer * buf);
static GstStateChangeReturn gst_festival_change_state (GstElement * element,
GstStateChange transition);
static FT_Info *festival_default_info (void);
static char *socket_receive_file_to_buff (int fd, int *size);
static char *client_accept_s_expr (int fd);
static GstStaticPadTemplate sink_template_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("text/x-raw, format=(string)utf8")
);
static GstStaticPadTemplate src_template_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wav")
);
/* Festival signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0
/* FILL ME */
};
/*static guint gst_festival_signals[LAST_SIGNAL] = { 0 }; */
G_DEFINE_TYPE (GstFestival, gst_festival, GST_TYPE_ELEMENT)
static void gst_festival_class_init (GstFestivalClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = G_OBJECT_CLASS (klass);
gstelement_class = GST_ELEMENT_CLASS (klass);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_festival_finalize);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_festival_change_state);
/* register pads */
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sink_template_factory));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_template_factory));
gst_element_class_set_static_metadata (gstelement_class,
"Festival Text-to-Speech synthesizer", "Filter/Effect/Audio",
"Synthesizes plain text into audio",
"Wim Taymans <wim.taymans@gmail.com>");
}
static void
gst_festival_init (GstFestival * festival)
{
festival->sinkpad =
gst_pad_new_from_static_template (&sink_template_factory, "sink");
gst_pad_set_chain_function (festival->sinkpad, gst_festival_chain);
gst_element_add_pad (GST_ELEMENT (festival), festival->sinkpad);
festival->srcpad =
gst_pad_new_from_static_template (&src_template_factory, "src");
gst_element_add_pad (GST_ELEMENT (festival), festival->srcpad);
festival->info = festival_default_info ();
}
static void
gst_festival_finalize (GObject * object)
{
GstFestival *festival = GST_FESTIVAL (object);
g_free (festival->info);
G_OBJECT_CLASS (gst_festival_parent_class)->finalize (object);
}
static gboolean
read_response (GstFestival * festival)
{
char ack[4];
char *data;
int filesize;
int fd;
int n;
gboolean ret = TRUE;
fd = festival->info->server_fd;
do {
for (n = 0; n < 3;)
n += read (fd, ack + n, 3 - n);
ack[3] = '\0';
GST_DEBUG_OBJECT (festival, "got response %s", ack);
if (strcmp (ack, "WV\n") == 0) {
GstBuffer *buffer;
/* receive a waveform */
data = socket_receive_file_to_buff (fd, &filesize);
GST_DEBUG_OBJECT (festival, "received %d bytes of waveform data",
filesize);
/* push contents as a buffer */
buffer = gst_buffer_new_wrapped (data, filesize);
GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
gst_pad_push (festival->srcpad, buffer);
} else if (strcmp (ack, "LP\n") == 0) {
/* receive an s-expr */
data = client_accept_s_expr (fd);
GST_DEBUG_OBJECT (festival, "received s-expression: %s", data);
g_free (data);
} else if (strcmp (ack, "ER\n") == 0) {
/* server got an error */
GST_ELEMENT_ERROR (festival,
LIBRARY,
FAILED,
("Festival speech server returned an error"),
("Make sure you have voices/languages installed"));
ret = FALSE;
break;
}
} while (strcmp (ack, "OK\n") != 0);
return ret;
}
static GstFlowReturn
gst_festival_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstFestival *festival;
GstMapInfo info;
guint8 *p, *ep;
gint f;
FILE *fd;
festival = GST_FESTIVAL (parent);
GST_LOG_OBJECT (festival, "Got text buffer, %" G_GSIZE_FORMAT " bytes",
gst_buffer_get_size (buf));
f = dup (festival->info->server_fd);
if (f < 0)
goto fail_open;
fd = fdopen (f, "wb");
if (fd == NULL) {
close (f);
goto fail_open;
}
/* Copy text over to server, escaping any quotes */
fprintf (fd, "(Parameter.set 'Audio_Required_Rate 16000)\n");
fflush (fd);
GST_DEBUG_OBJECT (festival, "issued Parameter.set command");
if (read_response (festival) == FALSE) {
fclose (fd);
goto fail_read;
}
fprintf (fd, "(tts_textall \"");
gst_buffer_map (buf, &info, GST_MAP_READ);
p = info.data;
ep = p + info.size;
for (; p < ep && (*p != '\0'); p++) {
if ((*p == '"') || (*p == '\\')) {
putc ('\\', fd);
}
putc (*p, fd);
}
fprintf (fd, "\" \"%s\")\n", festival->info->text_mode);
fclose (fd);
gst_buffer_unmap (buf, &info);
GST_DEBUG_OBJECT (festival, "issued tts_textall command");
/* Read back info from server */
if (read_response (festival) == FALSE)
goto fail_read;
out:
gst_buffer_unref (buf);
return ret;
/* ERRORS */
fail_open:
{
GST_ELEMENT_ERROR (festival, RESOURCE, OPEN_WRITE, (NULL), (NULL));
ret = GST_FLOW_ERROR;
goto out;
}
fail_read:
{
GST_ELEMENT_ERROR (festival, RESOURCE, READ, (NULL), (NULL));
ret = GST_FLOW_ERROR;
goto out;
}
}
static FT_Info *
festival_default_info (void)
{
FT_Info *info;
info = (FT_Info *) malloc (1 * sizeof (FT_Info));
info->server_host = FESTIVAL_DEFAULT_SERVER_HOST;
info->server_port = FESTIVAL_DEFAULT_SERVER_PORT;
info->text_mode = FESTIVAL_DEFAULT_TEXT_MODE;
info->server_fd = -1;
return info;
}
static int
festival_socket_open (const char *host, int port)
{
/* Return an FD to a remote server */
struct sockaddr_in serv_addr;
struct hostent *serverhost;
int fd;
fd = socket (AF_INET, SOCK_STREAM, IPPROTO_TCP);
if (fd < 0) {
fprintf (stderr, "festival_client: can't get socket\n");
return -1;
}
memset (&serv_addr, 0, sizeof (serv_addr));
if ((serv_addr.sin_addr.s_addr = inet_addr (host)) == -1) {
/* its a name rather than an ipnum */
serverhost = gethostbyname (host);
if (serverhost == (struct hostent *) 0) {
fprintf (stderr, "festival_client: gethostbyname failed\n");
close (fd);
return -1;
}
memmove (&serv_addr.sin_addr, serverhost->h_addr, serverhost->h_length);
}
serv_addr.sin_family = AF_INET;
serv_addr.sin_port = htons (port);
if (connect (fd, (struct sockaddr *) &serv_addr, sizeof (serv_addr)) != 0) {
fprintf (stderr, "festival_client: connect to server failed\n");
close (fd);
return -1;
}
return fd;
}
static char *
client_accept_s_expr (int fd)
{
/* Read s-expression from server, as a char * */
char *expr;
int filesize;
expr = socket_receive_file_to_buff (fd, &filesize);
expr[filesize] = '\0';
return expr;
}
static char *
socket_receive_file_to_buff (int fd, int *size)
{
/* Receive file (probably a waveform file) from socket using */
/* Festival key stuff technique, but long winded I know, sorry */
/* but will receive any file without closeing the stream or */
/* using OOB data */
static const char file_stuff_key[] = "ft_StUfF_key"; /* must == Festival's key */
char *buff;
int bufflen;
int n, k, i;
char c;
bufflen = 1024;
buff = (char *) g_malloc (bufflen);
*size = 0;
for (k = 0; file_stuff_key[k] != '\0';) {
n = read (fd, &c, 1);
if (n == 0)
break; /* hit stream eof before end of file */
if ((*size) + k + 1 >= bufflen) {
/* +1 so you can add a NULL if you want */
bufflen += bufflen / 4;
buff = (char *) g_realloc (buff, bufflen);
}
if (file_stuff_key[k] == c)
k++;
else if ((c == 'X') && (file_stuff_key[k + 1] == '\0')) {
/* It looked like the key but wasn't */
for (i = 0; i < k; i++, (*size)++)
buff[*size] = file_stuff_key[i];
k = 0;
/* omit the stuffed 'X' */
} else {
for (i = 0; i < k; i++, (*size)++)
buff[*size] = file_stuff_key[i];
k = 0;
buff[*size] = c;
(*size)++;
}
}
return buff;
}
/***********************************************************************/
/* Public Functions to this API */
/***********************************************************************/
static gboolean
gst_festival_open (GstFestival * festival)
{
/* Open socket to server */
if (festival->info == NULL)
festival->info = festival_default_info ();
festival->info->server_fd =
festival_socket_open (festival->info->server_host,
festival->info->server_port);
if (festival->info->server_fd == -1) {
GST_ERROR
("Could not talk to festival server (no server running or wrong host/port?)");
return FALSE;
}
GST_OBJECT_FLAG_SET (festival, GST_FESTIVAL_OPEN);
return TRUE;
}
static void
gst_festival_close (GstFestival * festival)
{
if (festival->info == NULL)
return;
if (festival->info->server_fd != -1)
close (festival->info->server_fd);
GST_OBJECT_FLAG_UNSET (festival, GST_FESTIVAL_OPEN);
return;
}
static GstStateChangeReturn
gst_festival_change_state (GstElement * element, GstStateChange transition)
{
g_return_val_if_fail (GST_IS_FESTIVAL (element), GST_STATE_CHANGE_FAILURE);
if (GST_STATE_PENDING (element) == GST_STATE_NULL) {
if (GST_OBJECT_FLAG_IS_SET (element, GST_FESTIVAL_OPEN)) {
GST_DEBUG ("Closing connection ");
gst_festival_close (GST_FESTIVAL (element));
}
} else {
if (!GST_OBJECT_FLAG_IS_SET (element, GST_FESTIVAL_OPEN)) {
GST_DEBUG ("Opening connection ");
if (!gst_festival_open (GST_FESTIVAL (element)))
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
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);
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