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Pierre Hubert
2016-11-19 12:08:12 +01:00
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<!DOCTYPE html>
<html>
<head>
<title>Speaker.js - Comunic</title>
<!-- Styles de la page -->
<style type="text/css">
body {
text-align: justify;
}
#speaker_area {
text-align: center;
margin-top: 17%;
background-color: rgba(27, 161, 226, 0.44);
padding: 20px;
border-radius: 5px;
}
.texte_parole {
width: 55% !important;
}
</style>
<!-- Inclusion de Speak.js -->
<script type="text/javascript" src="js/mespeak.js"></script>
<!-- Initialisation de Speak.js -->
<script type="text/javascript">
//Chargement de la voix française
meSpeak.loadVoice("voices/fr.json");
//Chargement de la configuration
meSpeak.loadConfig("js/mespeak_config.json");
//Permet de forcer le telechargement d'un fichier en JS
function forceDownload(fileURL, fileName) {
// non-IE
if (!window.ActiveXObject) {
var save = document.createElement('a');
save.href = fileURL;
save.target = '_blank';
save.download = fileName || 'unknown';
var event = document.createEvent('Event');
event.initEvent('click', true, true);
save.dispatchEvent(event);
(window.URL || window.webkitURL).revokeObjectURL(save.href);
}
// for IE
else if ( !! window.ActiveXObject && document.execCommand) {
var _window = window.open(fileURL, '_blank');
_window.document.close();
_window.document.execCommand('SaveAs', true, fileName || fileURL)
_window.close();
}
}
function gospeak() {
//Récupération du texte de parole
texte = document.getElementById('text_speak').value;
//On vérifie si il faut utiliser une voix de femme ou non
use_female_voice = document.getElementById('enable_female_voice').checked;
//On vérifie si il faut télécharger le résultat ou non
download_result = document.getElementById('download_result').checked;
if(use_female_voice == false)
{
//Utilisation d'une voix d'homme
var variante = "m3";
}
else
{
//Utilisation d'une voix de femme
var variante = "f5";
}
if(download_result == false)
{
//On parle
meSpeak.speak(texte, {voice: "fr", variant: variante});
}
else
{
//Préparer le fichier pour le téléchargement puis le télécharger
var myDataUrl = meSpeak.speak(texte, {voice: "fr", variant: variante, 'rawdata': 'data-url' });
window.open(myDataUrl);
}
}
</script>
</head>
<body>
<?php $sub_folder = true; include('../menu.php'); ?>
<div class="container">
<h1><a href="../" class="nav-button transform"><span></span></a>&nbsp;Speaker.js</h1>
<p>Ce logiciel web vous permet d'effectuer la synth&egrave;se vocale de ce que vous saisissez dans le champ de texte. Note: Gr&acirc;ce &agrave; la puissance de Javascript, les paroles sont transform&eacute;es en fichier vocal localement sur votre navigateur. Votre confidentialit&eacute; est ainsi respect&eacute;e. Ce syst&egrave;me est bas&eacute; sur le projet meSpeak.js, dont le site est <a href="http://www.masswerk.at/mespeak/" target="_blank">accessible ici</a>.</p>
<!-- Zone de gestion de la parole -->
<div id="speaker_area">
<!-- Saisie la parole -->
<div class="input-control text texte_parole">
<input type="text" placeholder="Saissez ici votre texte" id="text_speak" />
</div>
&nbsp;
<!-- Activer ou non la voix de femme -->
<label class="switch">
<input type="checkbox" id="enable_female_voice"/>
<span class="check"></span>
Voix de femme
</label>
<!-- Télécharger ou non -->
<label class="switch">
<input type="checkbox" id="download_result"/>
<span class="check"></span>
T&eacute;l&eacute;charger
</label>
<!-- Parler! -->
<button class="button" onClick="gospeak();">
<span class="mif-volume-high"></span>
</button>
</div>
</div>
</body>
</html>

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/*
meSpeakFallback.as
An experimental fallback option for meSpeak to play wav files
Wav-files are sent as a plain array of uint 8-bit data via ExternalInterface
JS interface:
meSpeakFallback.play( <wav-array> );
meSpeak.setVolume( value ) // 0 >= value <= 1
Handshake: calls JS function 'meSpeakFallbackHandshake()', when initialized and ready
Norbert Landsteiner, www.masswerk.at, July 2013
*/
package {
import flash.display.Sprite;
import flash.events.Event;
import flash.external.ExternalInterface;
import flash.media.Sound;
import flash.media.SoundTransform;
import flash.utils.ByteArray;
import flash.utils.setTimeout;
import org.as3wavsound.WavSound;
import org.as3wavsound.WavSoundChannel;
public class meSpeakFallback extends Sprite
{
private var sndTransform:SoundTransform=new SoundTransform(1, 0);
public function meSpeakFallback()
{
initExtIF();
}
private function initExtIF():void {
var available:Boolean=false;
try {
if (ExternalInterface.available) {
ExternalInterface.addCallback("play", play);
ExternalInterface.addCallback("setVolume", setVolume);
ExternalInterface.call("meSpeakFallbackHandshake");
available=true;
}
}
catch (e:Error) {}
if (!available) setTimeout(initExtIF, 100);
}
public function setVolume(v:Number=0):void {
if (v>=0 && v<=1) sndTransform.volume=v;
}
public function play(data:Array=null):void {
if (!data) return;
var l:uint=data.length;
if (!l) return;
// copy data to a ByteArray (oops: time, memory!)
var ba:ByteArray=new ByteArray();
ba.length=l;
for (var i:uint=0; i<l; i++) ba.writeByte(data[i]);
// play sound with global volume
var snd:WavSound = new WavSound(ba);
var chnl:WavSoundChannel=snd.play(0, 0, sndTransform);
}
}
}

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/*
* --------------------------------------
* Benny Bottema -- WavSound Sound adaption
* http://blog.projectnibble.org/
* --------------------------------------
* sazameki -- audio manipulating library
* http://sazameki.org/
* --------------------------------------
*
* - developed by:
* Benny Bottema
* blog.projectnibble.org
* hosted by:
* Google Code (code.google.com)
* code.google.com/p/as3wavsound/
*
* - audio library in its original state developed by:
* Takaaki Yamazaki
* www.zkdesign.jp
* hosted by:
* Spark project (www.libspark.org)
* www.libspark.org/svn/as3/sazameki/branches/fp10/
*/
/*
* Licensed under the MIT License
*
* Copyright (c) 2008 Takaaki Yamazaki
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
package org.as3wavsound {
import flash.media.SoundTransform;
import flash.utils.ByteArray;
import org.as3wavsound.sazameki.core.AudioSamples;
import org.as3wavsound.sazameki.core.AudioSetting;
import org.as3wavsound.sazameki.format.wav.Wav;
/**
* Sound extension that directly plays WAVE data. Also backwards compatible with
* MP3's played through the load() function. This class acts as facade to loading,
* extracting, decoding and playing wav sound data and represents a single sound.
*
* This class is analog to Adobe's Sound class and is designed to function the same
* way.
*
* Usage:
* Simply embed .wav files as you would mp3's and play with this Sound class.
* Make sure you provide mimetype 'application/octet-stream' when embedding to
* ensure Flash embeds the data as ByteArray.
*
* Example:
* [Embed(source = "drumloop.wav", mimeType = "application/octet-stream")]
* public const DrumLoop:Class;
* public const rain:WavSound = new WavSound(new DrumLoop() as ByteArray);
*
*
* @author Benny Bottema
*/
public class WavSound {
// the master Sound player, which mixes all playing WavSound samples on any given moment
private static const player:WavSoundPlayer = new WavSoundPlayer();
// length of the original encoded wav data
private var _bytesTotal:Number;
// extracted sound data for mixing
private var _samples:AudioSamples;
// each sound can be configured to be played mono/stereo using AudioSetting
private var _playbackSettings:AudioSetting;
// calculated length of the entire sound in milliseconds, made global to avoid recalculating all the time
private var _length:Number;
/**
* Constructor: loads wavdata using load().
*
* loads WAVE data and decodes it into playable samples. Finally calculates
* the length of the sound in milliseconds.
*
* @param wavData A ByteArray containing uncmopressed wav data.
* @param audioSettings An optional playback configuration (mono/stereo,
* sample rate and bit rate).
*/
public function WavSound(wavData:ByteArray, audioSettings:AudioSetting = null) {
load(wavData, audioSettings);
}
/**
* Key function: loads WAVE data and decodes it into playable samples.
* Finally calculates the length of the sound in milliseconds.
*
* @param wavData The byte array that is the embedded .was file (octet-stream).
* @param audioSettings Optional settings for playback (samplerate will enforced
* if it differs from the .wav header data or header is missing).
* @see Wav#decode(ByteArray)
*/
internal function load(wavData:ByteArray, audioSettings:AudioSetting = null): void {
this._bytesTotal = wavData.length;
this._samples = new Wav().decode(wavData, audioSettings);
this._playbackSettings = (audioSettings != null) ? audioSettings : new AudioSetting();
this._length = samples.length / samples.setting.sampleRate * 1000;
}
/**
* See Adobe's Sound.play(): http://help.adobe.com/en_US/FlashPlatform/reference/actionscript/3/flash/media/Sound.html#play().
*
* Playback function that performs the following tasks:
*
* - calculates the startingPhase, bases on startTime in ms.
* - initializes loopsLeft variable
* - adds the playing channel in combination with its originating WavSound to the playingWavSounds
*
* @param startTime The starting time in milliseconds, applies to each loop (as with regular MP3 Sounds).
* @param loops The number of loops to take in *addition* to the default playback (loops == 2 means 3 playthroughs).
* @param sndTransform An optional soundtransform to apply for playback that controls volume and panning.
* @return The SoundChannel used for playing back the sound (and stopping the sound).
*/
public function play(startTime:Number = 0, loops:int = 0, sndTransform:SoundTransform = null): WavSoundChannel {
return player.play(this, startTime, loops, sndTransform);
}
/**
* No idea if this works. Alpha state. Read up on Sound.extract():
* http://help.adobe.com/en_US/FlashPlatform/reference/actionscript/3/flash/media/Sound.html#extract()
*
* Apparently, some people have used this succesfully, see comment 1 on Issue 11:
* http://code.google.com/p/as3wavsound/issues/detail?id=11#c1
*/
public function extract(target:ByteArray, length:Number, startPosition:Number = -1): Number {
var start:Number = Math.max(startPosition, 0);
var end:Number = Math.min(length, samples.length);
for (var i:Number = start; i < end; i++) {
target.writeFloat(samples.left[i]);
if (samples.setting.channels == 2) {
target.writeFloat(samples.right[i]);
} else {
target.writeFloat(samples.left[i]);
}
}
return samples.length;
}
/**
* Returns the total bytes of the wavData a WavSound was created with.
*
* Note:
* This function is probably legacy, since we're not extending Adobe's
* Sound anymore (backwards compatibility was dropped in v0.7.
*/
public function get bytesLoaded () : uint {
return _bytesTotal;
}
/**
* Returns the total bytes of the wavData a WavSound was created with.
*
* Note:
* This function is probably legacy, since we're not extending Adobe's
* Sound anymore (backwards compatibility was dropped in v0.7.
*/
public function get bytesTotal () : int {
return _bytesTotal;
}
/**
* Returns the total length of the sound in milliseconds.
*/
public function get length() : Number {
return _length;
}
internal function get samples():AudioSamples {
return _samples;
}
/**
* _playbackSettings is set when the load() function is called.
*/
internal function get playbackSettings():AudioSetting {
return _playbackSettings;
}
}
}

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/*
* --------------------------------------
* Benny Bottema -- WavSound Sound adaption
* http://blog.projectnibble.org/
* --------------------------------------
* sazameki -- audio manipulating library
* http://sazameki.org/
* --------------------------------------
*
* - developed by:
* Benny Bottema
* blog.projectnibble.org
* hosted by:
* Google Code (code.google.com)
* code.google.com/p/as3wavsound/
*
* - audio library in its original state developed by:
* Takaaki Yamazaki
* www.zkdesign.jp
* hosted by:
* Spark project (www.libspark.org)
* www.libspark.org/svn/as3/sazameki/branches/fp10/
*/
/*
* Licensed under the MIT License
*
* Copyright (c) 2008 Takaaki Yamazaki
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
package org.as3wavsound {
import flash.events.EventDispatcher;
import flash.media.SoundChannel;
import flash.events.Event;
import flash.media.SoundTransform;
import org.as3wavsound.sazameki.core.AudioSamples;
import org.as3wavsound.WavSound;
/**
* Used to keep track of open channels during playback. Each channel represents
* an 'instance' of a sound and so each channel is responsible for its own mixing.
*
* The WavSound class uses the WavSoundPlayer to play instances of itself and
* returns the WavSoundChannel returned by the player.
*
* Also, the WavSoundPlayer uses the buffer() function to make the playing WavSoundChannel
* mix its own samples into the master buffer.
*
* Dispatches the Event.SOUND_COMPLETE event when the last sample has been mixed
* into the master buffer.
*
* Also see buffer().
*
* @author Benny Bottema
*/
public class WavSoundChannel extends EventDispatcher {
/*
* creation-time information
*/
// The player to delegate play() stop() requests to.
private var player:WavSoundPlayer;
// a WavSound currently playing back on this channel instance
// (there can be mutliple instances with the same WavSound).
private var _wavSound:WavSound;
// works the same as Adobe's SoundChannel.soundTransform
private var _soundTransform:SoundTransform = new SoundTransform();
/*
* play-time information *per WavSound instance*
*/
// starting phase if not at the beginning, made global to avoid recalculating all the time
private var startPhase:Number;
// current phase of the sound, basically matches a single current sample frame for each WavSound
private var phase:Number = 0;
// the current avarage volume of samples buffered to the left audiochannel
private var _leftPeak:Number = 0;
// the current avarage volume of samples buffered to the right audiochannel
private var _rightPeak:Number = 0;
// how many loops we need to buffer
private var loopsLeft:Number;
// indicates if the phase has reached total sample count and no loops are left
private var finished:Boolean;
/**
* Constructor: pre-calculates starting phase (and performs some validation for this), see init().
*/
public function WavSoundChannel(player:WavSoundPlayer, wavSound:WavSound, startTime:Number, loops:int, soundTransform:SoundTransform) {
this.player = player;
this._wavSound = wavSound;
if (soundTransform != null) {
this._soundTransform = soundTransform;
}
init(startTime, loops);
}
/**
* Calculates and validates the starting time. Starting time in milliseconds is converted into
* sample position and then marked as starting phase.
*
* Also resets finished state and sets 'loopsLeft' equal to the given 'loops' value.
*/
internal function init(startTime:Number, loops:int):void {
var startPositionInMillis:Number = Math.floor(startTime);
var maxPositionInMillis:Number = Math.floor(_wavSound.length);
if (startPositionInMillis > maxPositionInMillis) {
throw new Error("startTime greater than sound's length, max startTime is " + maxPositionInMillis);
}
phase = startPhase = Math.floor(startPositionInMillis * _wavSound.samples.length / _wavSound.length);
finished = false;
loopsLeft = loops;
}
/**
* Tells the WavsoundPlayer to stop this specific SoundWavChannel instance.
*/
public function stop():void {
player.stop(this);
}
/**
* Called from WavSoundPlayer when the player is ready to mix new samples into the master
* sample buffer.
*
* Fills a target samplebuffer with (optionally transformed) samples from the current
* WavSoundChannel instance.
*
* Keeps filling the buffer until the last samples are buffered or until the buffersize is
* reached. When the buffer is full, phase and loopsLeft keep track of how which samples
* still need to be buffered in the next buffering cycle (when this method is called again).
*
* @param sampleBuffer The target buffer to mix in the current (transformed) samples.
* @param soundTransform The soundtransform that belongs to a single channel being played
* (containing volume, panning etc.).
*/
internal function buffer(sampleBuffer:AudioSamples):void {
// calculate volume and panning
var volume: Number = (_soundTransform.volume / 1);
var volumeLeft: Number = volume * (1 - _soundTransform.pan) / 2;
var volumeRight: Number = volume * (1 + _soundTransform.pan) / 2;
// channel settings
var needRightChannel:Boolean = _wavSound.playbackSettings.channels == 2;
var hasRightChannel:Boolean = _wavSound.samples.setting.channels == 2;
// extra references to avoid excessive getter calls in the following
// for-loop (it appeares CPU is being hogged otherwise)
var samplesLength:Number = _wavSound.samples.length;
var samplesLeft:Vector.<Number> = _wavSound.samples.left;
var samplesRight:Vector.<Number> = _wavSound.samples.right;
var sampleBufferLength:Number = sampleBuffer.length;
var sampleBufferLeft:Vector.<Number> = sampleBuffer.left;
var sampleBufferRight:Vector.<Number> = sampleBuffer.right;
var leftPeakRecord:Number = 0;
var rightPeakRecord:Number = 0;
// finally, mix the samples in the master sample buffer
if (!finished) {
for (var i:int = 0; i < sampleBufferLength; i++) {
if (!finished) {
// write (transformed) samples to buffer
var sampleLeft:Number = samplesLeft[phase] * volumeLeft;
sampleBufferLeft[i] += sampleLeft;
leftPeakRecord += sampleLeft;
var channelValue:Number = ((needRightChannel && hasRightChannel) ? samplesRight[phase] : samplesLeft[phase]);
var sampleRight:Number = channelValue * volumeRight;
sampleBufferRight[i] += sampleRight;
rightPeakRecord += sampleRight;
// check playing and looping state
if (++phase >= samplesLength) {
phase = startPhase;
finished = loopsLeft-- == 0;
}
}
}
if (finished) {
dispatchEvent(new Event(Event.SOUND_COMPLETE));
}
}
_leftPeak = leftPeakRecord / sampleBufferLength;
_rightPeak = rightPeakRecord / sampleBufferLength
}
public function get leftPeak(): Number {
return _leftPeak;
}
public function get rightPeak(): Number {
return _rightPeak;
}
/**
* Returns the current position in milliseconds:
*
* phase * wavSound.length / wavSound.samples.length
*/
public function get position(): Number {
return phase * _wavSound.length / _wavSound.samples.length;
}
public function get soundTransform():SoundTransform {
return _soundTransform;
}
}
}

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/*
* --------------------------------------
* Benny Bottema -- WavSound Sound adaption
* http://blog.projectnibble.org/
* --------------------------------------
* sazameki -- audio manipulating library
* http://sazameki.org/
* --------------------------------------
*
* - developed by:
* Benny Bottema
* blog.projectnibble.org
* hosted by:
* Google Code (code.google.com)
* code.google.com/p/as3wavsound/
*
* - audio library in its original state developed by:
* Takaaki Yamazaki
* www.zkdesign.jp
* hosted by:
* Spark project (www.libspark.org)
* www.libspark.org/svn/as3/sazameki/branches/fp10/
*/
/*
* Licensed under the MIT License
*
* Copyright (c) 2008 Takaaki Yamazaki
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
package org.as3wavsound {
import flash.events.SampleDataEvent;
import flash.media.Sound;
import flash.media.SoundChannel;
import flash.media.SoundTransform;
import flash.net.URLRequest;
import flash.utils.ByteArray;
import org.as3wavsound.sazameki.core.AudioSamples;
import org.as3wavsound.sazameki.core.AudioSetting;
import org.as3wavsound.WavSoundChannel;
/**
* This player is used by WavSound instances to relay play() calls to and
* return the resulting WavSoundChannel instances.
*
* This player is used by WavSoundChannel instances to relay stop() calls to.
*
* This player contains a single Sound object which acts as the master buffer in which
* all playing sounds are mixed to. This is done to reduce cpu / memory footprint. The
* player will loop through all playing WavSoundChannel instances and call
* buffer(masterSampleBuffer) function on each, before writing the end result to the
* sound card's outputstream.
*
* @author Benny Bottema
*/
internal class WavSoundPlayer {
// The size of the master sample buffer used for playback.
// Too small: the sound will have a jittery playback.
// Too big: the sound will have high latencies (loading, stopping, playing, etc.).
public static var MAX_BUFFERSIZE:Number = 8192;
// the master samples buffer in which all seperate Wavsounds are mixed into, always stereo at 44100Hz and bitrate 16
private const sampleBuffer:AudioSamples = new AudioSamples(new AudioSetting(), MAX_BUFFERSIZE);
// a list of all WavSound currenctly in playing mode
private const playingWavSounds:Vector.<WavSoundChannel> = new Vector.<WavSoundChannel>();
// the singular playback Sound with which all other WavSounds are played back
private const player:Sound = configurePlayer();
/**
* Static initializer: creates, configures and a sound player using the 'sample
* data event technique'. Until play() has been called on a WavSound, nothing is
* audible, because playingWavSounds will still be empty.
*
* Also see: http://help.adobe.com/en_US/FlashPlatform/reference/actionscript/3/flash/events/SampleDataEvent.html
*/
private function configurePlayer():Sound {
var player:Sound = new Sound();
player.addEventListener(SampleDataEvent.SAMPLE_DATA, onSamplesCallback);
player.play();
return player;
}
/**
* Creates WavSoundChannel and adds it to the list of currently playing channels
* (which are mixed into the master sample buffer).
*
* This function is called by WavSound instances which returns the new WavSoundChannel
* instance to the user.
*/
internal function play(sound:WavSound, startTime:Number, loops:int, sndTransform:SoundTransform):WavSoundChannel {
var channel:WavSoundChannel = new WavSoundChannel(this, sound, startTime, loops, sndTransform);
playingWavSounds.push(channel);
return channel;
}
/**
* Remove a specific currently playing channel, so that its samples won't be
* mixed to the master sample buffer anymore and therefor playback will stop.
*/
internal function stop(channel:WavSoundChannel):void {
for each (var playingWavSound:WavSoundChannel in playingWavSounds) {
if (playingWavSound == channel) {
playingWavSounds.splice(playingWavSounds.lastIndexOf(playingWavSound), 1);
}
}
}
/**
* The heartbeat of the WavSound approach, invoked by the master Sound object.
*
* This function handles the SampleDataEvent to mix all playing sounds in playingWavSounds
* into the Sound's buffer. For each playing WavSoundChannel instance, the player will call
* the channel's buffer() function to have it mix itself into the master sample buffer.
* Finally, the resulting master buffer is written to the event's output stream.
*
* Also see: http://help.adobe.com/en_US/FlashPlatform/reference/actionscript/3/flash/events/SampleDataEvent.html
*
* @param event Contains the soundcard outputstream to mix sound samples into.
*/
private function onSamplesCallback(event:SampleDataEvent):void {
// clear the buffer
sampleBuffer.clearSamples();
// have all channels mix their into the master sample buffer
for each (var playingWavSound:WavSoundChannel in playingWavSounds) {
playingWavSound.buffer(sampleBuffer);
}
// extra references to avoid excessive getter calls in the following
// for-loop (it appeares CPU is being hogged otherwise)
var outputStream:ByteArray = event.data;
var samplesLength:Number = sampleBuffer.length;
var samplesLeft:Vector.<Number> = sampleBuffer.left;
var samplesRight:Vector.<Number> = sampleBuffer.right;
// write all mixed samples to the sound's outputstream
for (var i:int = 0; i < samplesLength; i++) {
outputStream.writeFloat(samplesLeft[i]);
outputStream.writeFloat(samplesRight[i]);
}
}
}
}

57
tools/speaker/msie_flashFallback/as3/meSpeakFallback/src/org/as3wavsound/sazameki/core/AudioSamples.as Executable file
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package org.as3wavsound.sazameki.core {
/**
* Contains lists of samples -left and optionally right- decoded from a
* WAVE ByteArray or manually mixed samples.
*
* Also contains a reference to an AudioSetting instance associated by
* this samples container.
*
* @author Takaaki Yamazaki(zk design),
* @author Benny Bottema (modified, optimized and cleaned up code)
*/
public class AudioSamples {
public var _left:Vector.<Number>;
public var _right:Vector.<Number>;
private var _setting:AudioSetting;
/**
* @param length Can be zero when decoding WAVE data, or a fixed buffer
* size when mixing to a Sound's outputstream.
*/
public function AudioSamples(setting:AudioSetting, length:Number = 0) {
this._setting = setting;
this._left = new Vector.<Number>(length, length > 0);
if (setting.channels == 2) {
this._right = new Vector.<Number>(length, length > 0);
}
}
/**
* Always resets length to its former state. Don't call this after creating
* an instance of AudioSamples, or its length is always zero.
*/
public function clearSamples():void {
_left = new Vector.<Number>(length, true);
if (setting.channels == 2) {
_right = new Vector.<Number>(length, true);
}
}
public function get length():int {
return left.length;
}
public function get setting():AudioSetting {
return _setting;
}
public function get left():Vector.<Number> {
return _left;
}
public function get right():Vector.<Number> {
return _right;
}
}
}

50
tools/speaker/msie_flashFallback/as3/meSpeakFallback/src/org/as3wavsound/sazameki/core/AudioSetting.as Executable file
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package org.as3wavsound.sazameki.core {
/**
* Contains a sound's playback configuration, such as mono / stereo,
* sample rate and bit rate.
*
* @author Takaaki Yamazaki(zk design),
* @author Benny Bottema (modified, optimized and cleaned up code)
*/
public class AudioSetting {
// 1 or 2
private var _channels:uint;
// 11025, 22050 or 44100
private var _sampleRate:uint;
// 8 or 16
private var _bitRate:uint;
/**
* Constructor: performs some validations on the values being passed in.
*/
public function AudioSetting(channels:uint = 2, sampleRate:uint = 44100, bitRate:uint = 16) {
if (sampleRate != 44100 && sampleRate != 22050 && sampleRate != 11025) {
throw new Error("bad sample rate. sample rate must be 44100, 22050 or 11025");
}
if (channels != 1 && channels != 2) {
throw new Error("channels must be 1 or 2");
}
if (bitRate != 16 && bitRate != 8) {
throw new Error("bitRate must be 8 or 16");
}
_channels=channels;
_sampleRate=sampleRate;
_bitRate=bitRate;
}
public function get channels():uint{
return _channels;
}
public function get sampleRate():uint{
return _sampleRate;
}
public function get bitRate():uint{
return _bitRate;
}
}
}

48
tools/speaker/msie_flashFallback/as3/meSpeakFallback/src/org/as3wavsound/sazameki/format/riff/Chunk.as Executable file
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package org.as3wavsound.sazameki.format.riff {
import flash.utils.ByteArray;
import flash.utils.Endian;
/**
* RIFF Chunk class
*
* @author Takaaki Yamazaki(zk design),
* @author Benny Bottema (modified, optimized and cleaned up code)
*/
public class Chunk {
protected const ENDIAN:String = Endian.LITTLE_ENDIAN;
protected var _id:String;
public function Chunk(id:String) {
this.id = id;
}
public function set id(value:String):void {
if (value.length > 4) {
value = value.substr(0, 4);
} else if (value.length < 4) {
while (value.length < 4) {
value += " ";
}
}
_id = value;
}
public function get id():String {
return _id;
}
public function toByteArray():ByteArray {
var result:ByteArray = new ByteArray();
result.endian = ENDIAN;
result.writeUTFBytes(_id);
var data:ByteArray = encodeData();
result.writeUnsignedInt(data.length);
result.writeBytes(data);
return result;
}
protected function encodeData():ByteArray {
throw new Error("'encodeData()' method must be overriden");
}
}
}

75
tools/speaker/msie_flashFallback/as3/meSpeakFallback/src/org/as3wavsound/sazameki/format/riff/LIST.as Executable file
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package org.as3wavsound.sazameki.format.riff {
import flash.utils.ByteArray;
/**
* ...
*
* @author Takaaki Yamazaki(zk design),
* @author Benny Bottema (modified, optimized and cleaned up code)
*/
public class LIST extends Chunk {
protected var _type:String;
protected var _chunks:Vector.<Chunk>;
public function LIST(type:String) {
this.type = type;
super("LIST");
}
public function set type(value:String):void {
if (value.length > 4) {
value = value.substr(0, 4);
} else if (value.length < 4) {
while (value.length < 4) {
value += " ";
}
}
_type = value;
}
public function get type():String {
return _type;
}
override protected function encodeData():ByteArray {
var result:ByteArray = new ByteArray();
result.writeUTFBytes(_type);
for (var i:int = 0; i < _chunks.length; i++) {
result.writeBytes(_chunks[i].toByteArray());
}
return result;
}
protected function splitList(bytes:ByteArray):Object {
var obj:Object = new Object();
bytes.position = 0;
bytes.endian = ENDIAN;
if (bytes.readUTFBytes(4) == 'RIFF') {
bytes.readInt();
bytes.readUTFBytes(4);//type
} else {
bytes.position = 0;
}
while (bytes.position < bytes.length) {
var currentName:String = bytes.readUTFBytes(4);
var current:int = bytes.readInt();
if (currentName == 'LIST') {
currentName = bytes.readUTFBytes(4);
current -= 4;
}
var tmpByte:ByteArray = new ByteArray();
bytes.readBytes(tmpByte, 0, current);
if (current % 2 == 1) {
bytes.readByte();
}
obj[currentName] = tmpByte;
}
return obj;
}
}
}

15
tools/speaker/msie_flashFallback/as3/meSpeakFallback/src/org/as3wavsound/sazameki/format/riff/RIFF.as Executable file
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package org.as3wavsound.sazameki.format.riff {
/**
* ...
*
* @author Takaaki Yamazaki(zk design),
* @author Benny Bottema (modified, optimized and cleaned up code)
*/
public class RIFF extends LIST {
public function RIFF(type:String) {
super(type);
id = 'RIFF';
}
}
}

106
tools/speaker/msie_flashFallback/as3/meSpeakFallback/src/org/as3wavsound/sazameki/format/wav/Wav.as Executable file
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package org.as3wavsound.sazameki.format.wav {
import org.as3wavsound.sazameki.core.AudioSamples;
import org.as3wavsound.sazameki.core.AudioSetting;
import org.as3wavsound.sazameki.format.riff.Chunk;
import org.as3wavsound.sazameki.format.riff.RIFF;
import org.as3wavsound.sazameki.format.wav.chunk.WavdataChunk;
import org.as3wavsound.sazameki.format.wav.chunk.WavfmtChunk;
import flash.utils.ByteArray;
/**
* The WAVE decoder used for playing back wav files.
*
* @author Takaaki Yamazaki(zk design),
* @author Benny Bottema (modified, optimized and cleaned up code)
*/
public class Wav extends RIFF {
public function Wav() {
super('WAVE');
}
public function encode(samples:AudioSamples):ByteArray {
var fmt:WavfmtChunk = new WavfmtChunk();
var data:WavdataChunk = new WavdataChunk();
_chunks = new Vector.<Chunk>;
_chunks.push(fmt);
_chunks.push(data);
data.setAudioData(samples);
fmt.setSetting(samples.setting);
return toByteArray();
}
public function decode(wavData:ByteArray, setting:AudioSetting):AudioSamples {
var obj:Object = splitList(wavData);
var data:AudioSamples;
var relevantSetting:AudioSetting = setting;
if (relevantSetting == null && obj['fmt ']) {
relevantSetting = new WavfmtChunk().decodeData(obj['fmt '] as ByteArray);
}
if (obj['fmt '] && obj['data']) {
data = new WavdataChunk().decodeData(obj['data'] as ByteArray, relevantSetting);
} else {
data = new WavdataChunk().decodeData(wavData, relevantSetting);
}
var needsResampling:Boolean = relevantSetting != null && relevantSetting.sampleRate != 44100;
return (needsResampling) ? resampleAudioSamples(data, relevantSetting.sampleRate) : data;
}
/**
* Resamples the given audio samples from a given sample rate to a target sample rate (or default 44100).
*
* @author Simion Medvedi (medvedisimion@gmail.com)
* @author Benny Bottema (sanitized code and added support for stereo resampling)
*/
private function resampleAudioSamples(data:AudioSamples, sourceRate:int, targetRate:int = 44100):AudioSamples {
var newSize:int = data.length * targetRate / sourceRate;
var newData:AudioSamples = new AudioSamples(new AudioSetting(data.setting.channels, targetRate, 16), newSize);
resampleSamples(data.left, newData.left, newSize, sourceRate, targetRate);
// playback buffering in WavSoundChannel will take care of a possibly missing right channel
if (data.setting.channels == 2) {
resampleSamples(data.right, newData.right, newSize, sourceRate, targetRate);
}
return newData;
}
/**
* Resamples the given audio samples from a given sample rate to a target sample rate (or default 44100).
*
* @author Simion Medvedi (medvedisimion@gmail.com)
* @author Benny Bottema (sanitized code)
*/
private function resampleSamples(sourceSamples:Vector.<Number>, targetSamples:Vector.<Number>, newSize:int, sourceRate:int, targetRate:int = 44100):void {
// we need to expand the sample rate from whatever it is to targetRate Khz. This code
// is assuming that the sample rate will be < targetRate Khz.
var multiplier:Number = targetRate / sourceRate;
// convert the data
var measure:int = targetRate;
var sourceIndex:int = 0;
var targetIndex:int = 0;
while (targetIndex < newSize) {
if (measure >= sourceRate) {
var increment:Number = 0;
if (targetIndex > 0 && sourceIndex < sourceSamples.length - 1) {
increment = (sourceSamples[sourceIndex + 1] - sourceSamples[sourceIndex]) / multiplier;
}
targetSamples[targetIndex++] = sourceSamples[sourceIndex] + increment;
measure -= sourceRate;
} else {
sourceIndex++;
measure += targetRate;
}
}
}
}
}

126
tools/speaker/msie_flashFallback/as3/meSpeakFallback/src/org/as3wavsound/sazameki/format/wav/chunk/WavdataChunk.as Executable file
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package org.as3wavsound.sazameki.format.wav.chunk {
import flash.utils.ByteArray;
import org.as3wavsound.sazameki.core.AudioSamples;
import org.as3wavsound.sazameki.core.AudioSetting;
import org.as3wavsound.sazameki.format.riff.Chunk;
/**
* ...
*
* @author Takaaki Yamazaki(zk design),
* @author Benny Bottema (modified, optimized and cleaned up code)
*/
public class WavdataChunk extends Chunk {
private var _samples:AudioSamples;
public function WavdataChunk() {
super('data');
}
public function setAudioData(samples:AudioSamples):void {
_samples = samples;
}
override protected function encodeData():ByteArray {
var bytes:ByteArray = new ByteArray();
bytes.endian = ENDIAN;
var setting:AudioSetting = _samples.setting;
var i:int;
var sig:Number;
var len:int = _samples.left.length;
var left:Vector.<Number>;
if (setting.channels == 2) {
left=_samples.left;
var right:Vector.<Number>=_samples.right;
if (setting.bitRate == 16) {
for (i = 0; i < len; i++) {
sig = left[i];
if (sig < -1) bytes.writeShort( -32767);
else if (sig > 1) bytes.writeShort( 32767);
else bytes.writeShort(sig * 32767);
sig = right[i];
if (sig < -1) bytes.writeShort(-32767);
else if (sig > 1) bytes.writeShort(32767);
else bytes.writeShort(sig * 32767);
}
} else {
for (i = 0; i < len; i++) {
sig = left[i];
if (sig<-1) bytes.writeByte(0);
else if (sig>1) bytes.writeByte(255);
else bytes.writeByte(sig*127+128);
sig = right[i];
if (sig<-1) bytes.writeByte(0);
else if (sig>1) bytes.writeByte(255);
else bytes.writeByte(sig*127+128);
}
}
} else {
left = _samples.left;
if (setting.bitRate == 16) {
for (i = 0; i < len; i++) {
sig = left[i];
if (sig < -1) bytes.writeShort(-32767);
else if (sig > 1) bytes.writeShort(32767);
else bytes.writeShort(sig * 32768);
}
} else {
for (i = 0; i < len; i++) {
sig = left[i];
if (sig<-1) bytes.writeByte(0);
else if (sig>1) bytes.writeByte(255);
else bytes.writeByte(sig*127+128);
}
}
}
return bytes;
}
public function decodeData(bytes:ByteArray, setting:AudioSetting):AudioSamples {
bytes.position = 0;
bytes.endian = ENDIAN;
var samples:AudioSamples = new AudioSamples(setting);
var length:int = bytes.length / (setting.bitRate / 8) / setting.channels;
var i:int;
var left:Vector.<Number>;
if (setting.channels == 2) {
left = samples.left;
var right:Vector.<Number> = samples.right;
if (setting.bitRate == 16) {
for (i = 0; i < length; ++i) {
left[i] = bytes.readShort() / 32767;
right[i] = bytes.readShort() / 32767;
}
} else {
for (i = 0; i < length; i++)
{
left[i] = bytes.readByte() / 255;
right[i] = bytes.readByte() / 255;
}
}
} else {
left = samples.left;
if (setting.bitRate == 16) {
for (i = 0; i < length; i++) {
left[i] = bytes.readShort() / 32767;
}
} else {
for (i = 0; i < length; i++) {
left[i] = bytes.readByte() / 255;
}
}
}
return samples;
}
}
}

56
tools/speaker/msie_flashFallback/as3/meSpeakFallback/src/org/as3wavsound/sazameki/format/wav/chunk/WavfmtChunk.as Executable file
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package org.as3wavsound.sazameki.format.wav.chunk {
import flash.utils.ByteArray;
import org.as3wavsound.sazameki.core.AudioSetting;
import org.as3wavsound.sazameki.format.riff.Chunk;
/**
* ...
*
* @author Takaaki Yamazaki(zk design),
* @author Benny Bottema (modified, optimized and cleaned up code)
*/
public class WavfmtChunk extends Chunk {
private var _setting:AudioSetting;
public function WavfmtChunk() {
super('fmt ');
}
public function setSetting(setting:AudioSetting):void {
_setting = setting;
}
override protected function encodeData():ByteArray {
var result:ByteArray = new ByteArray();
result.endian = ENDIAN;
//fmt ID(2)
result.writeShort(1);
//channels(2)
result.writeShort(_setting.channels);
//sampling rate(4)
result.writeInt(_setting.sampleRate);
//data rate(4)
result.writeInt(_setting.sampleRate * _setting.channels * (_setting.bitRate / 8));
//block size(2)
result.writeShort((_setting.bitRate / 8) * _setting.channels);
//bit rate(2)
result.writeShort(_setting.bitRate);
return result;
}
public function decodeData(bytes:ByteArray):AudioSetting {
bytes.position = 0;
bytes.endian = ENDIAN;
bytes.readShort();
var channels:int = bytes.readShort();
var smplRate:int = bytes.readInt();
bytes.readInt();
bytes.readShort();
var bit:int = bytes.readShort();
_setting = new AudioSetting(channels, smplRate, bit);
return _setting;
}
}
}

228
tools/speaker/msie_flashFallback/flashFallback.js Executable file
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/*
A fallback to flash for wav-output (for IE 10)
Please mind that wav data has to be copied to an ArrayBuffer object internally,
since we may not send binary data to the swf.
This may take some time and memory for longer utterances.
*/
var meSpeakFlashFallback = new function() {
var swfDefaultId='meSpeakFallback',
swfDefaultUrl='meSpeakFallback.swf',
swfElementId='', swfViaAX=false, swfInstalled=false, swfHasLoaded=false, swfVol=1;
// public
function swfInstallFallback(swfUrl, swfId, parentElementOrId) {
var parentEl, url;
if (swfInstalled) return true;
if (!swfIsAvailable(10)) return false;
swfInstalled=true;
// set defaults
swfElementId = (swfId && typeof swfId == 'string')? swfId:swfDefaultId;
url = (swfUrl && typeof swfUrl == 'string')? swfUrl:swfDefaultUrl;
if (parentElementOrId) {
if (typeof parentElementOrId == 'string') {
parentEl=document.getElementById(parentElementOrId);
}
else if (typeof parentElementOrId == 'object') {
parentEl=parentElementOrId=null;
}
}
if (!parentEl) parentEl=document.getElementsByTagName('body')[0];
if (!parentEl) return false;
// inject
var obj = swfCreate(
{
'data': url,
'width': '2',
'height': '2',
'id': swfElementId,
'name': swfElementId,
'align': 'top'
},
{
'quality': 'low',
'bgcolor': 'transparent',
'allowscriptaccess': 'sameDomain',
'allowfullscreen': 'false'
}
);
parentEl.appendChild(obj);
swfRegisterUnloadHandler();
return true;
}
function swfReady() {
return swfHasLoaded;
}
function swfSetVolume(v) {
if (wfHasLoaded) {
var obj=document.getElementById(swfElementId);
if (obj) el.setVolume(v);
}
swfVol=v;
}
function swfSpeak(txt, options) {
if (swfHasLoaded && window.meSpeak) {
var obj=document.getElementById(swfElementId);
if (obj) {
if (!typeof options != 'object') options={};
options.rawdata='array';
obj.play( meSpeak.speak(txt, options) );
}
}
}
function swf10Available() {
return swfIsAvailable(10);
}
function swfFallbackHandshake() {
swfHasLoaded=true;
if (swfVol!=1) swfSetVolume(swfVol);
if (window.console) console.log('meSpeak-SWF-fallback available.');
}
// private: a stripped-down version of swfobject.js
function swfIsAvailable(leastMajorVersion) {
// returns Boolean: flashplayer and version at least 10.x
var sf='Shockwave Flash', sfm='application/x-shockwave-flash';
if (navigator.plugins !== undefined && typeof navigator.plugins[sf] == 'object') {
var d=navigator.plugins[sf].description;
if (d && !(typeof navigator.mimeTypes !==undefined && navigator.mimeTypes[sfm] && !navigator.mimeTypes[sfm].enabledPlugin)) {
d=d.replace(/^.*\s+(\S+\s+\S+$)/, '$1');
if (leastMajorVersion<= parseInt(d.replace(/^(.*)\..*$/, '$1'), 10)) return true;
}
}
else if (window.ActiveXObject) {
try {
var a=new ActiveXObject('ShockwaveFlash.ShockwaveFlash');
if (a) {
swfViaAX=true;
d=a.GetVariable('$version');
if (d) {
d=d.split(' ')[1].split(',');
if (leastMajorVersion<= parseInt(d[0], 10)) return true;
}
}
}
catch(e) {}
}
return false;
}
function swfCreate(attributes, params) {
if (swfViaAX) {
var att='', par='', i;
for (i in attributes) {
var a=i.toLowerCase;
if (a=='data') {
params.movie=attributes[i];
}
else if (a=='styleclass') {
att+=' class="'+attributes[i]+'"';
}
else if (a!='classid') {
att+=' '+i+'="'+attributes[i]+'"';
}
}
for (i in params) {
if (params[i] != Object.prototype[i]) par+=' <param name="'+i+'" value="'+params[i]+'" />';
}
var el=document.createElement('div');
el.outerHTML='<object classid="clsid:D27CDB6E-AE6D-11cf-96B8-444553540000"'+att+'>'+par+'</object>';
return el;
}
else {
var o=document.createElement('object');
o.setAttribute('type', 'application/x-shockwave-flash');
for (var i in attributes) {
if (attributes[i] != Object.prototype[i]) {
var a=i.toLowerCase();
if (a=='styleclass') {
o.setAttribute('class', attributes[i]);
}
else if (a!='styleclass') {
o.setAttribute(i, attributes[i]);
}
}
}
for (i in params) {
if (attributes[i] != Object.prototype[i] && i.toLowerCase() != 'movie') {
var p=document.createElement('param');
p.setAttribute('name', i);
p.setAttribute('value', attributes[i]);
o.appendChild(p);
}
}
return o;
}
}
function swfRemove(obj) {
try {
if (typeof obj =='string') obj=document.getElementById(obj);
if (!obj || typeof obj !='object') return;
if (swfViaAX) {
obj.style.display='none';
swfRemoveObjectInIE(obj.id);
}
else if (obj.parentNode) {
obj.parentNode.removeChild(obj);
}
swfInstalled=false;
}
catch(e) {}
}
function swfRemoveObjectInIE(id) {
var obj=document.getElementById(obj);
if (obj) {
if (obj.readyState==4) {
for (var i in obj) {
if (typeof obj[i] =='function') obj[i] = null;
}
if (obj.parentNode) obj.parentNode.removeChild(obj);
}
else {
setTimeout(function() {swfRemoveObjectInIE(id)}, 10);
}
}
}
function swfUnloadHandler() {
if (swfElementId) swfRemove(swfElementId);
if (!window.addEventListener && window.detachEvent) window.detachEvent('onunload', swfUnloadHandler);
}
function swfRegisterUnloadHandler() {
if (window.addEventListener) {
window.addEventListener('unload', swfUnloadHandler, false);
}
else if (window.attachEvent) {
window.attachEvent('onunload', swfUnloadHandler);
}
}
return {
'install': swfInstallFallback,
'isAvailable': swf10Available,
'ready': swfReady,
'speak': swfSpeak,
'setVolume': swfSetVolume,
'swfFallbackHandshake': swfFallbackHandshake
}
};
function meSpeakFallbackHandshake() {
// handshake handler with swf external interface
meSpeakFlashFallback.swfFallbackHandshake();
}

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tools/speaker/msie_flashFallback/index.html Executable file
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<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="utf-8" />
<title>meSpeak Flash Fallback</title>
<link href="http://fonts.googleapis.com/css?family=Open+Sans&amp;subset=latin" rel="stylesheet" type="text/css" />
<link href="http://fonts.googleapis.com/css?family=Lato:300&amp;subset=latin" rel="stylesheet" type="text/css" />
<script type="text/javascript" src="../mespeak.js"></script>
<script type="text/javascript" src="flashFallback.js"></script>
<script type="text/javascript">
meSpeak.loadConfig("../mespeak_config.json");
meSpeak.loadVoice("../voices/en/en.json");
var fallbackInstalled = false;
var hasTypedArrays = Boolean(window.Int32Array && window.Float32Array
&& (window.Int32Array.prototype.subarray || window.Int32Array.subarray)
),
canSpeak = (hasTypedArrays && window.meSpeak && meSpeak.canPlay());
function checkAndInstall() {
// check typed array support (we need this) and wav-support (we fallback, if not present)
if (hasTypedArrays && window.meSpeak && !meSpeak.canPlay()) {
canSpeak = installFallback();
}
}
function installFallback() {
// install with explicit default values
// swf will be 2px x 2px (h x w), transparent
return fallbackInstalled = meSpeakFlashFallback.install(
'meSpeakFallback.swf', // url
'meSpeakFallback', // id of swf object
null // parent element to inject into
// (default: append to body)
);
}
function fallbackSpeak() {
meSpeakFlashFallback.speak('Hello world.', { 'speed': 180 } );
}
// some functions for abstracting
function speakAny( txt, options, vol ) {
if (fallbackInstalled) {
meSpeakFlashFallback.speak( txt, options );
}
else if (canSpeak) {
meSpeak.speak( txt, options, vol );
}
}
function setVolume( vol) {
if (fallbackInstalled) {
meSpeakFlashFallback.setVolume( vol );
}
else if (canSpeak) {
meSpeak.setVolume( vol );
}
}
function fallbackReady() {
// check if the swf has loaded and performed a handshake, indicating that it's available
return meSpeakFlashFallback.ready();
}
// install only, if needed
// if (document.addEventListener) document.addEventListener('DOMContentLoaded', checkAndInstall, false);
// install unconditionally (for testing purpose)
if (document.addEventListener) document.addEventListener('DOMContentLoaded', installFallback, false);
</script>
<style type="text/css">
html
{
margin: 0;
padding: 2em 1.5em 4.5em 1.5em;
background-color: #e2e3e4;
}
body
{
max-width: 900px;
padding: 2px 40px 60px 40px;
margin: 0 auto 0 auto;
background-color: #fafafb;
color: #111;
font-family: 'Open Sans',sans-serif;
font-size: 13px;
line-height: 19px;
}
h1,h2,h3,h4
{
font-family: 'Lato',sans-serif;
font-weight: 300;
}
h1 {
font-size: 46px;
line-height: 46px;
color: #2681a7;
margin-top: 0.5em;
margin-bottom: 0.5em;
padding: 0;
}
h2
{
font-size: 36px;
color: #111;
margin-top: 0;
margin-bottom: 1.5em;
clear: both;
}
h1 span.pict { font-size: 38px; color: #ccc; margin-left: 0.5em; letter-spacing: -2px; }
p.codesample,xmp
{
margin: 1em 0;
padding: 1em 0 1em 2em;
white-space: pre;
font-family: monospace;
line-height: 18px;
background-color: #f2f3f5;
color: #111;
}
a { color: #006f9e; }
a:hover,a:focus { color: #2681a7; }
a:active { color: #cd360e; }
</style>
</head>
<body>
<h1>meSpeak.js <span class="pict">(( &bull; ))</span></h1>
<h2>Flash Fallback <small>(for IE 10)</small></h2>
<p>A fallback to play meSpeak's wav-files via Flash.<br />Since typed arrays are still a requirement, the only use-case is MS Internet Explorer 10.<br />
Please note that, since we may not send binary data to a swf-object, the data has to be copied internally to a binary object, which may take some time and memory.<br />Because of this, the status of this solution is at best &quot;experimental&quot; and not for everyday use.</p>
<p>The fallback &quot;meSpeakFallback.swf&quot; uses the <a href="http://code.google.com/p/as3wavsound/">AS3WavSound</a> library (v0.9) for wav-playback inside the SWF. (Ironically Flash doesn't support native wav-playback either.)<br />The swf-file is compiled for network use, meaning it will work only, if loaded over a network (web-server).</p>
<p>For testing, the fallback is installed for all browsers on this page:
<input type="button" value="speak: Hello world." onclick="fallbackSpeak();" /></p>
<p>See page source for API &amp; details.</p>
<p>Download: <a href="http://www.masswerk.at/mespeak/msie_flashFallback.zip">msie_flashFallback.zip</a><br />
<em>(Installation: Put the unzipped directory inside your mespeak-directory.)</em></p>
<hr style="margin-top: 2em; margin-bottom: 2em;">
<p><strong>A Note on Usage:</strong><br />While the fallback-script strips down the coded needed to inject a swf-object to a minimum, best practice would be to include the script only, if applicable (i.e.: for MS IE 10 only), using MS-specific conditional comments:</p>
<xmp style="margin-left: 1em;">
<!--[if IE 10]>
<script type="text/javascript" src="flashFallback.js"></script>
<![endif]-->
</xmp>
<p>If doing so, you would have to check for the existence of <code>meSpeakFlashFallback</code> first, before calling it:</p>
<xmp style="margin-left: 1em;">
if (window.meSpeakFlashFallback) callbackInstalled = meSpeakFlashFallback.install();
// later, see page source for API details and usage
if (callbackInstalled) meSpeakFlashFallback.speak('Hello world.');
</xmp>
<p>This way, page- and memory-loads are reduced to a minimum for all browsers.</p>
<p>Please note that there is a delay between calling <code>install()</code> and the fallback actually being available, since the swf-file has to load and initialize first.</p>
<p>&nbsp;</p>
<p>Norbert Landsteiner, mass:werk &ndash; media environments, <a href="http://www.masswerk.at/" target="_top">www.masswerk.at</a><br />
Vienna, July 2013</p>
</body>
</html>

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{
"voice_id":"zh-yue",
"dict_id":"zhy_dict",
"dict":"AAQAAAgEAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAABmEABQEDFQAFAgMjAGEFAgNwAGFpBQIDcQBhdQUCA3IAYW5nBQIDdABpBQIDdQB1BQIDdgAHBmIABQEDFQAFAgNFAAcGYwAFAQMVAGgFAgNKAAUCA04ABwZkAAUBAxUABQIDRgAHBmUABQEDFQAFAgMkAGkFAgN3AHUFAgN5AG8FAgOEAG9pBQIDhQAHBmYABQEDFQAFAgNRAAcGZwAFAQMVAAUCA08ABwZoAAUBAxUABQIDaQAHBmkABQEDFQAFAgMlAHUFAgN7AAcGagAFAQMVAAUCAzkABwZrAAUBAxUABQIDMQAHBmwABQEDFQAFAgM3AAcGbQAFAQMVAAUCAz8ABwZuAAUBAxUABQIDMgBnBQIDQgAHBm8ABQEDFQAFAgMnAGkFAgN+AHUFAgN/AG5nBQIDgABlBQIDgwAHBnAABQEDFQAFAgMwAAcGcQAFAQMVAAUCAzEAdQUCAzE6AAcGcgAFAQMVAAUCAyIABwZzAAUBAxUABQIDVwAHBnQABQEDFQAFAgMvAAcGdQAFAQMVAAUCAygAaQUCA4IABwZ2AAUBAxUABQIDUgAHBncABQEDFQAFAgM6AAcGeAAFAQMVAAUCAzFXAAcGeQAFAQMVAAUCAzkAdQUCA4YABwZ6AAUBAxUABQIDVgAHBgAxAxEAMgNqADMDawA0A2wANQNtADYDbgA3A28ABwA=",
"voice":"bmFtZSBjYW50b25lc2UKbGFuZ3VhZ2UgemgteXVlCmxhbmd1YWdlIHl1ZQpsYW5ndWFnZSB6aHkKCnRyYW5zbGF0b3Igemh5CnBob25lbWVzIHpoeQpkaWN0aW9uYXJ5IHpoeQpnZW5kZXIgbWFsZQoKLy8gaW50ZXJwcmV0IEVuZ2xpc2ggbGV0dGVycyBhcyAxPUVuZ2xpc2ggd29yZHMsIDI9anl1dHBpbmcKZGljdHJ1bGVzIDEKCndvcmRzIDEK"
}

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