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339 COPYING
@@ -0,0 +1,339 @@
+ GNU GENERAL PUBLIC LICENSE
+ Version 2, June 1991
+
+ Copyright (C) 1989, 1991 Free Software Foundation, Inc.,
+ 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
+
+ Preamble
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+ The licenses for most software are designed to take away your
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+ TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
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+ Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
+ This is free software, and you are welcome to redistribute it
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+
+ <signature of Ty Coon>, 1 April 1989
+ Ty Coon, President of Vice
+
+This General Public License does not permit incorporating your program into
+proprietary programs. If your program is a subroutine library, you may
+consider it more useful to permit linking proprietary applications with the
+library. If this is what you want to do, use the GNU Lesser General
+Public License instead of this License.
2  ChangeLog
@@ -0,0 +1,2 @@
+2009-02-09 Lefteris Zafiris <zaf.000@gmail.com> - 1.6-0.1-beta
+ Initial release 1.6-0.1-beta
68 Makefile
@@ -0,0 +1,68 @@
+#
+# Makefile for Asterisk espeak application
+#
+
+INSTALL=install
+ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk
+MODULES_DIR=$(ASTLIBDIR)/modules
+ASTETCDIR=$(INSTALL_PREFIX)/etc/asterisk
+
+CC=gcc
+OPTIMIZE=-O2
+DEBUG=-g
+
+LIBS+=-lm -lespeak -lsndfile -lresample
+CFLAGS+=-pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE
+
+all: _all
+ @echo " +-------- app_espeak Build Complete --------+"
+ @echo " + app_espeak has successfully been built, +"
+ @echo " + and can be installed by running: +"
+ @echo " + +"
+ @echo " + make install +"
+ @echo " +-------------------------------------------+"
+
+_all: app_espeak.so
+
+app_espeak.o: app_espeak.c
+ $(CC) $(CFLAGS) $(DEBUG) $(OPTIMIZE) -c -o app_espeak.o app_espeak.c
+
+app_espeak.so: app_espeak.o
+ $(CC) -shared -Xlinker -x -o $@ $< $(LIBS)
+
+clean:
+ rm -f app_espeak.o app_espeak.so .*.d
+
+install: _all
+ $(INSTALL) -m 755 -d $(DESTDIR)$(MODULES_DIR)
+ $(INSTALL) -m 755 app_espeak.so $(DESTDIR)$(MODULES_DIR)
+ @echo " +---- app_espeak Installation Complete -----+"
+ @echo " + +"
+ @echo " + app_espeak has successfully been installed+"
+ @echo " + If you would like to install the sample +"
+ @echo " + configuration file run: +"
+ @echo " + +"
+ @echo " + make samples +"
+ @echo " +-------------------------------------------+"
+
+samples:
+ mkdir -p $(DESTDIR)$(ASTETCDIR)
+ for x in *.sample; do \
+ if [ -f $(DESTDIR)$(ASTETCDIR)/`basename $$x .sample` ]; then \
+ if [ "$(OVERWRITE)" = "y" ]; then \
+ if cmp -s $(DESTDIR)$(ASTETCDIR)/`basename $$x .sample` $$x ; then \
+ echo "Config file $$x is unchanged"; \
+ continue; \
+ fi ; \
+ mv -f $(DESTDIR)$(ASTETCDIR)/`basename $$x .sample` $(DESTDIR)$(ASTETCDIR)/`basename $$x .sample`.old ; \
+ else \
+ echo "Skipping config file $$x"; \
+ continue; \
+ fi ;\
+ fi ; \
+ $(INSTALL) -m 644 $$x $(DESTDIR)$(ASTETCDIR)/`basename $$x .sample` ;\
+ done
+
+ifneq ($(wildcard .*.d),)
+ include .*.d
+endif
59 README
@@ -0,0 +1,59 @@
+===========================================
+ Espeak For Asterisk 1.6
+===========================================
+
+This provides the "Espeak" dialplan application, which allows you to use
+the Espeak speech synthesizer with Asterisk. This module invokes the Espeak TTS engine
+locally, and uses it to render text to speech.
+
+------------
+Requirements
+------------
+Asterisk 1.6 header files
+Espeak libraries and header files
+libsndfile libraries and header files
+libresample libraries and header files
+
+------------
+Installation
+------------
+$ make
+$ make install
+
+To install the sample configuration file, issue the following command after
+the 'make install' command:
+
+$ make samples
+
+-----
+Usage
+-----
+Espeak(text[,intkeys,language]): This will invoke the eSpeak TTS engine,
+send a text string, get back the resulting waveform and play it to
+the user, allowing any given interrupt keys to immediately terminate
+and return.
+
+--------
+Examples
+--------
+dialplan sample code for your extensions.conf
+
+;Espeak Demo
+ exten => 1234,1,Answer()
+ ;;Play mesage using default language as set in espeak.conf
+ exten => 1234,n,Espeak("This is a simple espeak test in english.",any,)
+ ;;Play message in Spanish
+ exten => 1234,n,Espeak("Esta es una simple prueba espeak en español.",any,es)
+ ;;Play message in Grrek
+ exten => 1234,n,Espeak("Αυτό είναι ένα απλό τέστ του espeak στα ελληνικά.",any,el)
+ ;;Read a text file from disk (relative to the channel language)
+ ;;and play it with espeak using the asterisk channel language.
+ exten => 1234,n,ReadFile(MYTEXT=/path/${LANGUAGE}/myfile,200)
+ exten => 1234,n,Espeak("${MYTEXY}",any,${LANGUAGE})
+ exten => 1234,n,Hangup()
+
+-------
+License
+-------
+The Espeak module for asterisk is distributed under the GNU General Public
+License v2. See COPYING for details.
328 app_espeak.c
@@ -0,0 +1,328 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2009, Lefteris Zafiris
+ *
+ * Lefteris Zafiris <zaf.000@gmail.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Say text to the user, using eSpeak TTS engine.
+ *
+ * \author\verbatim Lefteris Zafiris <zaf.000@gmail.com> \endverbatim
+ *
+ * \extref eSpeak text to speech Synthesis System - http://espeak.sourceforge.net/
+ *
+ * \ingroup applications
+ */
+
+/*** MODULEINFO
+ <defaultenabled>no</defaultenabled>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision: 00 $")
+
+#include <stdio.h>
+#include <espeak/speak_lib.h>
+#include <sndfile.h>
+#include <libresample.h>
+#include "asterisk/file.h"
+#include "asterisk/channel.h"
+#include "asterisk/module.h"
+#include "asterisk/config.h"
+#include "asterisk/app.h"
+#include "asterisk/utils.h"
+
+#define AST_MODULE "eSpeak"
+#define ESPEAK_CONFIG "espeak.conf"
+#define MAXLEN 2048
+
+static char *app = "eSpeak";
+
+static char *synopsis = "Say text to the user, using eSpeak speech synthesizer.";
+
+static char *descrip =
+" eSpeak(text[,intkeys,language]): This will invoke the eSpeak TTS engine,\n"
+"send a text string, get back the resulting waveform and play it to\n"
+"the user, allowing any given interrupt keys to immediately terminate\n"
+"and return.\n";
+
+static int synth_callback(short *wav, int numsamples, espeak_EVENT * events) {
+ if (wav) {
+ fwrite(wav, sizeof(short), numsamples, events[0].user_data);
+ }
+ return 0;
+}
+
+static int app_exec(struct ast_channel *chan, void *data)
+{
+ int res = 0;
+ SNDFILE *src_file, *dst_file;
+ SF_INFO src_info, dst_info;
+ FILE *fl;
+ int src_len, dst_len;
+ float *src, *dst;
+ void *handle;
+ double ratio;
+ const char *mydata;
+ const char *cachedir = "";
+ const char *temp;
+ int usecache = 0;
+ int writecache = 0;
+ char MD5_name[33] = "";
+ char cachefile[MAXLEN] = "";
+ char tmp_name[23];
+ char wav_or_name[30];
+ char wav_name[27];
+ int bits = 16;
+ int *offset, sample_rate, pos, bufferpos, outcount;
+ float file_size;
+ int speed = 150;
+ int volume = 100;
+ int wordgap = 1;
+ int pitch = 50;
+ int capind = 0;
+ const char *voice = "default";
+ static unsigned char wave_hdr[44] = {
+ 'R', 'I', 'F', 'F', 0, 0, 0, 0, 'W', 'A', 'V', 'E', 'f', 'm', 't', ' ',
+ 0x10, 0, 0, 0, 1, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 2, 0, 0x10, 0, 'd', 'a', 't', 'a', 0, 0, 0, 0};
+ struct ast_config *cfg;
+ struct ast_flags config_flags = { 0 };
+ AST_DECLARE_APP_ARGS(args,
+ AST_APP_ARG(text);
+ AST_APP_ARG(interrupt);
+ AST_APP_ARG(language);
+ );
+
+ if (ast_strlen_zero(data)) {
+ ast_log(AST_LOG_ERROR, "eSpeak requires an argument (text)\n");
+ return -1;
+ }
+
+ cfg = ast_config_load(ESPEAK_CONFIG, config_flags);
+
+ if (!cfg) {
+ ast_log(LOG_WARNING,
+ "eSpeak: No such configuration file %s using default settings\n",
+ ESPEAK_CONFIG);
+ } else {
+ if ((temp = ast_variable_retrieve(cfg, "general", "usecache")))
+ usecache = ast_true(temp);
+
+ if (!(cachedir = ast_variable_retrieve(cfg, "general", "cachedir")))
+ cachedir = "/tmp";
+
+ if ((temp = ast_variable_retrieve(cfg, "voice", "speed")))
+ speed = atoi(temp);
+
+ if ((temp = ast_variable_retrieve(cfg, "voice", "wordgap")))
+ wordgap = atoi(temp);
+
+ if ((temp = ast_variable_retrieve(cfg, "voice", "volume")))
+ volume = atoi(temp);
+
+ if ((temp = ast_variable_retrieve(cfg, "voice", "pitch")))
+ pitch = atoi(temp);
+
+ if ((temp = ast_variable_retrieve(cfg, "voice", "capind")))
+ capind = atoi(temp);
+
+ if ((temp = ast_variable_retrieve(cfg, "voice", "voice")))
+ voice = temp;
+ }
+
+ mydata = ast_strdupa(data);
+ AST_STANDARD_APP_ARGS(args, mydata);
+
+ if (args.interrupt && !strcasecmp(args.interrupt, "any"))
+ args.interrupt = AST_DIGIT_ANY;
+
+ if (strcasecmp(args.language, ""))
+ voice = args.language;
+
+ ast_debug(1, "eSpeak: Text passed: %s\nInterrupt key(s): %s\nLanguage: %s\n", args.text,
+ args.interrupt, voice);
+
+ /*Cache mechanism */
+ if (usecache) {
+ ast_md5_hash(MD5_name, args.text);
+ if (strlen(cachedir) + strlen(MD5_name) + 5 <= MAXLEN) {
+ ast_debug(1, "eSpeak: Activating cache mechanism...\n");
+ snprintf(cachefile, sizeof(cachefile), "%s/%s", cachedir, MD5_name);
+ if (ast_fileexists(cachefile, NULL, NULL) <= 0) {
+ ast_debug(1, "eSpeak: Cache file does not yet exist.\n");
+ writecache = 1;
+ } else {
+ ast_debug(1, "eSpeak: Cache file exists.\n");
+ if (chan->_state != AST_STATE_UP)
+ ast_answer(chan);
+ res = ast_streamfile(chan, cachefile, chan->language);
+ if (res) {
+ ast_log(AST_LOG_ERROR, "eSpeak: ast_streamfile failed on %s\n",
+ chan->name);
+ } else {
+ res = ast_waitstream(chan, args.interrupt);
+ ast_stopstream(chan);
+ ast_config_destroy(cfg);
+ return res;
+ }
+ }
+ }
+ }
+
+ /* Create temp filenames */
+ snprintf(tmp_name, sizeof(tmp_name), "/tmp/eSpeak_%li", ast_random());
+ snprintf(wav_name, sizeof(wav_name), "%s.wav", tmp_name);
+ snprintf(wav_or_name, sizeof(wav_or_name), "%s_or.wav", tmp_name);
+
+ /* Invoke eSpeak */
+ sample_rate = espeak_Initialize(AUDIO_OUTPUT_SYNCHRONOUS, 2000, NULL, 0);
+ espeak_SetSynthCallback(synth_callback);
+ espeak_SetVoiceByName(voice);
+ espeak_SetParameter(espeakRATE, speed, 0);
+ espeak_SetParameter(espeakVOLUME, volume, 0);
+ espeak_SetParameter(espeakWORDGAP, wordgap, 0);
+ espeak_SetParameter(espeakPITCH, pitch, 0);
+ espeak_SetParameter(espeakCAPITALS, capind, 0);
+
+ fl = fopen(wav_or_name, "w+");
+ if (fl = NULL) {
+ ast_log(LOG_ERROR, "eSpeak: Failed to create audio buffer file '%s'\n", wav_or_name);
+ ast_config_destroy(cfg);
+ return -1;
+ } else {
+ /* Copy wav header */
+ offset = (int *) (&wave_hdr[24]);
+ offset[0] = sample_rate;
+ offset[1] = sample_rate * (bits/8);
+ fwrite(wave_hdr, 1, sizeof(wave_hdr), fl);
+ }
+
+ espeak_Synth(args.text, strlen(args.text) + 1, 0, POS_CHARACTER, 0, espeakCHARS_AUTO, NULL, fl);
+ espeak_Terminate();
+
+ /* Fix file header */
+ file_size = ftell(fl);
+ offset = (int *) (&wave_hdr[4]);
+ offset[0] = (int) file_size - 8;
+ offset = (int *) (&wave_hdr[40]);
+ offset[0] = (int) file_size - 44;
+ fseek(fl, 0, SEEK_SET);
+ fwrite(wave_hdr, 1, sizeof(wave_hdr), fl);
+ fclose(fl);
+
+ /* Resample file */
+ ratio = (double) ((double) 8000 / (double) sample_rate);
+ memset(&src_info, 0, sizeof(src_info));
+ memset(&dst_info, 0, sizeof(dst_info));
+ src_file = sf_open(wav_or_name, SFM_READ, &src_info);
+ if (!src_file) {
+ ast_log(LOG_ERROR, "eSpeak: Failed to read audio file '%s'\n", wav_or_name);
+ ast_config_destroy(cfg);
+ remove(wav_or_name);
+ return -1;
+ }
+
+ memcpy(&dst_info, &src_info, sizeof(SF_INFO));
+ dst_info.samplerate = 8000;
+ dst_file = sf_open(wav_name, SFM_WRITE, &dst_info);
+ if (!dst_file) {
+ ast_log(LOG_ERROR, "eSpeak: Failed to create audio file '%s'\n", wav_name);
+ ast_config_destroy(cfg);
+ sf_close(src_file);
+ remove(wav_or_name);
+ return -1;
+ }
+ dst_info.frames = src_info.frames * (sf_count_t)8000 / (sf_count_t)sample_rate;
+
+ src_len = 16384;
+ dst_len = (src_len * ratio + 1000);
+ src = (float *) malloc(src_len * sizeof(float));
+ dst = (float *) malloc(dst_len * sizeof(float));
+
+ handle = (void *) malloc(sizeof(void *));
+ handle = resample_open(1, ratio, ratio);
+ if (!handle) {
+ ast_log(LOG_ERROR, "eSpeak: Failed open resampler\n");
+ ast_config_destroy(cfg);
+ sf_close(src_file);
+ sf_close(dst_file);
+ free(src);
+ free(dst);
+ remove(wav_or_name);
+ return -1;
+ }
+
+ pos = 0;
+ bufferpos = 0;
+ outcount = 0;
+
+ while (pos < src_info.frames) {
+ int block = MIN(src_len - bufferpos, src_info.frames - pos);
+ int last_flag = (pos + block == src_info.frames);
+ int in_used, out = 0;
+ sf_readf_float(src_file, &src[bufferpos], block);
+ block += bufferpos;
+ in_used = 0;
+ out = resample_process(handle, ratio, src, block, last_flag, &in_used, dst, dst_len);
+ sf_writef_float(dst_file, dst, out);
+ bufferpos = block - in_used;
+ pos += in_used;
+ outcount += out;
+ }
+
+ resample_close(handle);
+ sf_write_sync(dst_file);
+ sf_close(src_file);
+ sf_close(dst_file);
+ free(src);
+ free(dst);
+ remove(wav_or_name);
+
+ /* Save file to cache if set */
+ if (writecache) {
+ ast_debug(1, "eSpeak: Saving cache file %s\n", cachefile);
+ ast_filecopy(tmp_name, cachefile, NULL);
+ }
+
+ if (chan->_state != AST_STATE_UP)
+ ast_answer(chan);
+ res = ast_streamfile(chan, tmp_name, chan->language);
+ if (res) {
+ ast_log(LOG_ERROR, "eSpeak: ast_streamfile failed on %s\n", chan->name);
+ } else {
+ res = ast_waitstream(chan, args.interrupt);
+ ast_stopstream(chan);
+ }
+
+ ast_filedelete(tmp_name, NULL);
+ ast_config_destroy(cfg);
+ return res;
+}
+
+static int unload_module(void)
+{
+ return ast_unregister_application(app);
+}
+
+static int load_module(void)
+{
+ return ast_register_application(app, app_exec, synopsis, descrip) ?
+ AST_MODULE_LOAD_DECLINE : AST_MODULE_LOAD_SUCCESS;
+}
+
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "eSpeak TTS Interface");
BIN  app_espeak.o
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BIN  app_espeak.so
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1  authors
@@ -0,0 +1 @@
+Lefteris Zafiris <zaf.000@gmail.com>
48 espeak.conf.sample
@@ -0,0 +1,48 @@
+;
+; eSpeak Configuration
+;
+[general]
+;
+; Use cache (yes, no - defaults to no)
+;
+;usecache=yes
+;
+; If usecache=yes, a directory to store waveform cache files.
+; The cache is never cleared by the application, so you must take care of
+; cleaning it yourself (just delete any or all files from the cache).
+; THIS DIRECTORY *MUST* EXIST and must be writable from the asterisk process.
+; Defaults to /tmp, so it will be cleared periodically by Linux
+;
+;cachedir=/var/lib/asterisk/espeakcache/
+;
+
+[voice]
+;
+; The voice to use, i.e.
+; the name of the voice file to use from from espeak-data/voices
+; Use "espeak --voices" for a list of available voices
+; (default is "default")
+;
+;voice=af
+;
+; The playback speed in words per minute (default is 150)
+;
+;speed=130
+;
+; Amplitude, 0 to 200, default is 100
+;
+;volume=100
+;
+; Word gap. Pause between words, units of 10mS at the default speed
+;
+;wordgap=1
+;
+; Pitch adjustment, 0 to 99, default is 50
+;
+;pitch=40
+;
+; Indicate capital letters with: 1=sound, 2=the word "capitals",
+; higher values = a pitch increase
+;
+;capind=15
+
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