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  • The Best Text to Speech (TTS) Software Programs and Online Tools

    - by Lori Kaufman
    Text to Speech (TTS) software allows you to have text read aloud to you. This is useful for struggling readers and for writers, when editing and revising their work. You can also convert eBooks to audiobooks so you can listen to them on long drives. We’ve posted some websites here where you can find some good TTS software programs and online tools that are free or at least have free versions available. 8 Deadly Commands You Should Never Run on Linux 14 Special Google Searches That Show Instant Answers How To Create a Customized Windows 7 Installation Disc With Integrated Updates

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  • Microsoft TTS (Text to Speech) Dat File Locations

    - by neddy
    Ok, so I've downloaded some TTS engines to replace the default microsoft TTS engine, and make my program sound a little more 'human' -- basically i am wondering where abouts the TTS engine files are stored on the local pc (windows 7) -- the files i have are in .Dat format, does anyone have any idea where abouts the should go to be registered as a voice for Text-to-Speech? Cheers.

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  • Festival TTS showing SIOD:ran out off storage message

    - by Peeyush
    Hello i am designing a front end for Festival TTS using it's C++ API Everything is working fine in my programme but i have a problem that i am giving a drop down option to user to select other languages when user select a language from drop down then festival tts shows a message on console saying: SIOD:ran out of storage This message only shows if the text given to festival TTS is greater then 5 or 6 lines. i think this message comes because SIOD's heap is not free when i am going to call a new language. So please tell me the solution of this problem. Thanks

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  • How to get the default audio format of a TTS Engine

    - by Itslava
    In Microsoft TTS 5.1 or newer. The SpVoice.AudioOutputStream property says: The AudioOutputStream property gets and sets the current audio stream object used by the voice. Setting the voice's AudioOutputStream property may cause its audio output format to be automatically changed to match the text-to-speech (TTS) engine's preferred audio output format. If the voice's AllowAudioOutputFormatChangesOnNextSet property is True, the format change takes place; if False, the format remains unchanged. In order to set the AudioOutputStream property of a voice to a specific format, its AllowOutputFormatChangesOnNextSet should be False. It means a engine's always has a preferred audio output format. So, how can i get it.. i have not found any interface to get that attribute.

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  • Track status of Microsoft TTS output to wav file

    - by user325478
    I'm trying to track the status of my applications TTS output to a wav file. When speaking the text (to the speaker) the expected events (StartStream, Word, EndStream) are fired, however, no events are fired when outputing to a wave file. SpVoice vox = new SpVoice(); vox.Word += VoxWord; // Handle word processed event SpFileStream voxStream = new SpFileStream(); voxStream.Open(@"c:\test.wav", SpeechStreamFileMode.SSFMCreateForWrite, false); vox.AudioOutputStream = voxStream; vox.Speak("Hello World. Please track my status!", SpeechVoiceSpeakFlags.SVSFlagsAsync); Is it possible to asynchronously know the status of TTS output to wav?

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  • TTS on App Engine

    - by yati sagade
    I have written a small front-end to the Festival TTS system using Python/Django. I wish to deploy it on the Google App Engine cloud. A few questions: My application uses the Festival app 'text2wave'. Will is work on the cloud? I have used Python primitives like subprocess.call() to invoke the aforementioned program. Will that work? If your answer to any or both of (1) and (2) is no, is there a free api on the web that I can use (from the appengine)? I read somewhere about placing calls from Phono to a Voxeo backend, but I'm not sure what that means. I am aware of the Google Translate extension that allows translation using an HTTP GET (REST) request, but here the text is limited to 100 chars. Bad. Plus, they may take it down any point of time.

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  • What the performance of Festival TTS vs FreeTTS on windows platforms?

    - by Steel Plume
    9 hours ago I had the wonderful idea to build up from scratch Festival TTS on Windows. In the meanwhile, and in less than 30 minutes, I build up the same source code on Linux, but only because tons of development tools were already installed :D Also this morning I downloaded FreeTTS. I started a simple testing project in less of 10 minutes, thanks to the Java power :D. Unfortunately FreeTTS does not provide immediate support for international mbrola languages, so this morning I thinked "it needs too much work"... Backing to Festival TTS on Windows, after many obstacles, and tons of wrong downloads, CYGWIN before, then "make" and other exotic C++ tools "forgotten" in the installation of CYGWIN, then Visual Studio 2008, then Windows Platform SDK 2008, but it lacks of System.h, so just now I am downloading Windows Server Platform SDK 2003, etc... So now, as I am yet here without having compiled a single line of damned C-- code, my question is: at least Festival TTS is a good choice on Windows for Java development?

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  • how to lengthen the pause between the words with text-to-speech (pyTTS or SAPI5)

    - by Berry Tsakala
    Is it possible to extend the gap between spoken words when using text to speech with SAPI5 ? The problem is that esp. with some voices, the words are almost connected to each other, which makes the speech more difficult to understand. I'm using python and pyTTS module (on windows, since it's using SAPI) I tried to hook to the OnWord event and add a time.sleep() or tts.Pause(), but apparently even though all the events are caught, they are being processed only at the end of the spoken text, whether i'm using the sync or async flag. In this NON WORKING example, the sleep() method is executed only after the sentence is spoken: tts = pyTTS.Create() def f(x): tts.Pause() sleep(0.5) tts.Resume() tts.OnWord = f tts.Speak(text)

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  • High-Quality Text-To-Speech engine for personal use

    - by phihag
    I'm looking for a high-quality TTS engine that I can afford (let's say less than 1000$). So far, I've tried flite and festival. However, while the results are certainly understandable, technical texts are hard to follow. Commercial TTS solutions from Loquendo and Readspeaker sound way better. However, these companies don't seem to be willing to sell their product to mere mortals - I can't find a price on either's homepage. So, what are good TTS solutions for personal use?

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  • Text to speech(TTS)-Android

    - by bharathi
    Hi, I am new to this android platform.now I am Working on TTS(Text to Speech).what I am trying to do is that when i enter the text in the Text area it has to be converted to speech,when i click the speak button.can anyone help me out.Thanks in Advance.

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  • Text To Speech using c++

    - by Peeyush
    hi i want to develop a text to speech(TTS) programme using c++ so please suggest me where from where i should start because i have no knowledge about TTS so pease explain a little more.

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  • [Android SDK] Text-To-Speech addSpeech not working properly

    - by arcoraven
    Hi, I'm trying to get my Android app to play a .wav file recording of the word "Spinach Salad" whenever it sees that phrase being spoken by TTS. Here's the relevant code: spinach_salad.wav is located in /res/raw prodName = "Spinach Salad" mTts.addSpeech(prodName, "com.example.textextractor", R.raw.spinach_salad); ...and later in the code: mTts.speak("blah blah blah " + prodName, TextToSpeech.QUEUE_ADD, null); I've also tried: mTts.speak("blah blah blah Spinach Salad", TextToSpeech.QUEUE_ADD, null); and mTts.speak("blah blah blah", TextToSpeech.QUEUE_ADD, null); mTts.speak(productName_str, TextToSpeech.QUEUE_ADD, null); In both cases, I'm just hearing the TTS synthesized audio, rather than my custom .wav file. (On a related note, the last chunk of code sometimes speaks out of order, saying the second line before the first).

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  • How can I do a fastAgi to use with Asterisk in Perl

    - by Edwin Jaws
    Hi! I am trying to do an IVR using FAstAGI to say information from my database to my clients. I done AGI doing this but I need now run this from another server, windows server, but I dont know how can I do this. I checked the Asterisk::fastagi module but it is so confused and I dont understand anything. I did a few AGI perl scripts without problem but now I get. If somebody can give me an example to simple script to put me on rigth direction to do that will be so appreciated. Just I need one exmaple and to know if all AGI normal commands from AGI like stream_file are available in fastagi Thks In Advance Edwin Jaws

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  • SpeechSynthesizer in C# creates wav that has 22kHz... needs to be 16kHz

    - by Adrian
    My C# application needs to covert text to wav file and inject it into a Skype call. The code that creates the wav file is below. The problem is that the file has 22kHz sample rate and Skype accepts only 16kHz. Is there any way to adjust this setting? using (System.IO.FileStream stream = System.IO.File.Create("message.wav")) { System.Speech.Synthesis.SpeechSynthesizer speechEngine = new System.Speech.Synthesis.SpeechSynthesizer(); speechEngine.SetOutputToWaveStream(stream); speechEngine.Speak(number); stream.Flush(); }

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  • Getting started with Festival on iphone

    - by Moshe
    I'd like to use Festival in an iPhone app. Is this possible, considering the way Festival generates and then plays back audio files? Can I pass Festival an audio file directly from itself to play back? or is there a different way to read things? Is there another Text-to-speech engine for C/C++ to use with iphone?

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  • Credentials work for SSMS but not (ODBC) LogParser script

    - by justSteve
    Via SSMS I'm able to connect and navigate the server/db in question. but trying to connect via a logparser script the same credentials fail. I'm trying to execute this from the same box on which the server's running. the username is owner/dbo of the db. The db has mixed mode authentication. [linebreaks for clarity] C:\TTS\tools\LogParserc:\tts\tools\logparser\logparser file:c:\tts\tools\logparser\errors2SQL.sql?source="C:\inetpub\logs\LogFiles\W3SVC8\u_ex100521.log" -i:IISW3C -o:SQL -createTable:ON -oConnString:"Driver={SQL Server Native Client 10.0};Server=servername\SQLEXPRESS;db=Tter;uid=logger2;pwd=foo" -stats:OFF Task aborted. Error connecting to ODBC Server SQL State: 28000 Native Error: 18456 Error Message: [Microsoft][SQL Server Native Client 10.0][SQL Server]Login failed for user 'logger2'. C:\TTS\tools\LogParser

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  • New Source Database Added for EBS 12 + 11gR2 Transportable Tablespaces

    - by John Abraham
    The Transportable Tablespaces (TTS) process was originally certified for the migration of E-Business Suite R12 databases going from a source database of 11gR1 or 11gR2 to a target of 11gR2. This requirement has now been expanded to include a source database of 10gR2 (10.2.0.5) - this will potentially save time for existing 10gR2 customers as they can remove on a crucial upgrade step prior to performing the platform migration. The migration process requires an updated Controlled patch delivered by the Oracle E-Business Suite Platform Engineering team, i.e. it requires a password obtainable from Oracle Support. We released the patch in this manner to gauge uptake, and help identify and monitor any customer issues due to the nature of this technology. This patch has been updated to now include supporting 10gR2 as a source database. Does it meet your requirements?Note that for migration across platforms of the same "endian" format, users are advised to use the Transportable Database (TDB) migration process instead for large databases. The "endian-ness" target platforms can be verified by querying the view V$DB_TRANSPORTABLE_PLATFORM using SQL*Plus (connected as sysdba) on the source platform:SQL>select platform_name from v$db_transportable_platform;If the intended target platform does not appear in the output, it means that it is of a different endian format from the source. Consequently. database migration will need to be performed via Transportable Tablespaces (for large databases) or export/import.The use of Transportable Tablespaces can greatly speed up the migration of the data portion of the database. However, it does not affect metadata, which must still be migrated using export/import. We recommend that users initially perform a test migration on their database, using export/import with the 'metrics=y' parameter. This will help identify the relative amounts of data and metadata, and provide a basis for assessing likely gains in timing. In general, the larger the amount of data (compared to metadata), the greater the reduction in downtime that can be expected from using TTS as a migration process. For smaller databases or for those that have relatively small data compared to metadata, TTS will not be as beneficial for cross endian migration and the use of export/import (datapump) for the whole database is recommended. Where can I find more information? Using Transportable Tablespaces to Migrate Oracle E-Business Suite Release 12 Using Oracle Database 11g Release 2 Enterprise Edition (My Oracle Support Document 1311487.1) Oracle Database Administrator's Guide 11g Release 2 (11.2) Related Articles Database Migration using 11gR2 Transportable Tablespaces Now Certified for EBS 12 New Source Databases Added for Transportable Tablespaces + EBS 11i 10gR2 Transportable Tablespaces Certified for EBS 11i Migrating E-Business Suite Release 11i Databases Between Platforms Migrating E-Business Suite Release 12 Databases Between Platforms

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  • Random pauses with "Key dispatching timed out sending to <null>" when closing Android SurfaceView

    - by kenyee
    When I close an Android SurfaceView Activity, the app sometimes pauses and almost gets an ANR (Application Not Responding) message. Looking at LogCat, it appears to be timing out trying to send keys to it. I tried modifying the code to setFocusable(false) when in the onKeyDown handler for the SurfaceView, but that doesn't seem to affect this. Any other ideas on what might be causing this? Or what these messages in Logcat even mean? ============================= 05-31 19:35:35.285: INFO/WindowManager(586): focus null mToken is null at event dispatch! 05-31 19:35:35.295: WARN/WindowManager(586): Current state: {{KeyEvent{action=1 code=4 repeat=0 meta=0 scancode=158 mFlags=8} to null @ 1275334535292 lw=null lb=null fin=true gfw=true ed=true tts=0 wf=false fp=false mcf=null}} 05-31 19:35:35.305: WARN/WindowManager(586): Continuing to wait for key to be dispatched 05-31 19:35:40.306: WARN/WindowManager(586): Key dispatching timed out sending to 05-31 19:35:40.316: WARN/WindowManager(586): Dispatch state: {{KeyEvent{action=0 code=4 repeat=0 meta=0 scancode=158 mFlags=8} to Window{43763540 com.myapp/com.myapp.DiagramEdit paused=false} @ 1275334499512 lw=Window{43763540 com.myapp/com.myapp.DiagramEdit paused=false} lb=android.os.BinderProxy@43702190 fin=false gfw=true ed=true tts=0 wf=false fp=false mcf=null}} 05-31 19:35:40.326: INFO/WindowManager(586): focus null mToken is null at event dispatch! 05-31 19:35:40.326: WARN/WindowManager(586): Current state: {{KeyEvent{action=1 code=4 repeat=0 meta=0 scancode=158 mFlags=8} to null @ 1275334540327 lw=null lb=null fin=true gfw=true ed=true tts=0 wf=false fp=false mcf=null}} 05-31 19:35:40.326: WARN/WindowManager(586): Continuing to wait for key to be dispatched

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  • Making a Text-To-Speech Wrapper in Android

    - by John Montgomery
    I am attempting to create a wrapper class for Google Android's Text-To-Speech functionality. However, I'm having trouble finding a way to have the system pause until after the onInit function has finished. Attached at the bottom is something of a solution I created based on what I found here: http://stackoverflow.com/questions/1160876/android-speech-how-can-you-read-text-in-android However, this solution does not seem to work. Any thoughts on why this might not be working, or what would be a good idea in order to make sure that any Speak() calls happen after my onInit() call? public class SpeechSynth implements OnInitListener { private TextToSpeech tts; static final int TTS_CHECK_CODE = 0; private int ready = 0; private ReentrantLock waitForInitLock = new ReentrantLock(); SpeechSynth( Activity screen ) { ready = 0; tts = new TextToSpeech( screen, this ); waitForInitLock.lock(); } public void onInit(int status) { if (status == TextToSpeech.SUCCESS) { ready = 1; } waitForInitLock.unlock(); } public int Speak( String text ) { if( ready == 1 ) { tts.speak(text, TextToSpeech.QUEUE_ADD, null); return 1; } else { return 0; } } } I have been able to make it so that I can pass a string of text through the constructor, then have it played in the onInit() function. However, I would really like to avoid having to destroy and re-create the whole text-to-speech engine every time I need to have my program say something different.

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  • app not working

    - by pranay
    hi, i have written a simple app which would speak out to the user any incoming message. Both programmes seem to work perfectly when i lauched them as two separate pgms , but on keeping them in the same project/package only the speaker programme screen is seen and the receiver pgm doesn't seem to work . Can someone please help me out on it? the speaker pgm is: package com.example.TextSpeaker; import java.util.Locale; import android.app.Activity; import android.content.Intent; import android.os.Bundle; import android.speech.tts.TextToSpeech; import android.speech.tts.TextToSpeech.OnInitListener; import android.view.View; import android.view.View.OnClickListener; import android.widget.Button; import android.widget.Toast; // the following programme converts the msg user to speech public class TextSpeaker extends Activity implements OnInitListener { /** Called when the activity is first created. */ int MY_DATA_CHECK_CODE = 0; public TextToSpeech mtts; public Button button; //public EditText edittext; @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.main); button = (Button)findViewById(R.id.button); //edit text=(EditText)findViewById(R.id.edittext); button.setOnClickListener(new OnClickListener(){ @Override public void onClick(View v) { //mtts.speak(edittext.getText().toString(), TextToSpeech.QUEUE_FLUSH, null); Toast.makeText(getApplicationContext(), "The service has been started\n Every new message will now be read out", Toast.LENGTH_LONG).show(); } }); Intent myintent = new Intent(); myintent.setAction(TextToSpeech.Engine.ACTION_CHECK_TTS_DATA); startActivityForResult(myintent, MY_DATA_CHECK_CODE); } protected void onActivityResult(int requestcode,int resultcode,Intent data) { if(requestcode == MY_DATA_CHECK_CODE) { if(resultcode==TextToSpeech.Engine.CHECK_VOICE_DATA_PASS) { // success so create the TTS engine mtts = new TextToSpeech(this,this); mtts.setLanguage(Locale.ENGLISH); } else { //install the Engine Intent install = new Intent(); install.setAction(TextToSpeech.Engine.ACTION_INSTALL_TTS_DATA); startActivity(install); } } } public void onDestroy(Bundle savedInstanceStatBundle) { mtts.shutdown(); } public void onPause() { super.onPause(); // if our app has no focus if(mtts!=null) mtts.stop(); } @Override public void onInit(int status) { if(status==TextToSpeech.SUCCESS) button.setEnabled(true); } } and the Receiver programme is: package com.example.TextSpeaker; import android.content.BroadcastReceiver; import android.content.Context; import android.content.Intent; import android.os.Bundle; import android.speech.tts.TextToSpeech; import android.telephony.SmsMessage; // supports both gsm and cdma import android.widget.Toast; public class Receiver extends BroadcastReceiver{ @Override public void onReceive(Context context, Intent intent) { Bundle bundle = intent.getExtras(); SmsMessage[] msgs = null; String str=""; if(bundle!=null) { // retrive the sms received Object[] pdus = (Object[])bundle.get("pdus"); msgs = new SmsMessage[pdus.length]; for(int i=0;i } } }

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  • Group traffic shaping with traffic control?

    - by mmcbro
    I'm trying to limit the output bandwidth generated by an application with linux tc. This application sends me the source port of the request that I use has a filter to limit each user at a given downloadspeed. I feel that my setup could be managed way better if I had a better knowledge of linux tc. At the application level users are categorized as members of a group, each group have a limited bandwidth. Example : Members of group A : 512kbit/s Members of group B : 1Mbit/s Members of group C : 2Mbit/s When a user connects to the application, it retrieves the source port to the origin of the request from the user and sends me the source port and the bandwidth at which the user must be limited depending on group to which it belongs. With these informations I must add the appropriate rules so that the user (the source port in reality) is limited to the right bandwidth. If the user that connect isn't a member of any group it should be limited at a default bandwidth speed. I'm actually managing this by using a self made daemon that add or remove rules from when it receive a request from the application. With my little knowledge of tc I'm not able to limit other users (ones that aren't in a group, all others in fact) at a default speed and my configuration seems awful to me. Here is the base of my tc qdisc and classes : tc qdisc add dev eth0 root handle 1: htb tc class add dev eth0 parent 1: classid 1:1 htb rate 100mbps ceil 125mbps To classify a user at a given speed I have to add one subclass and then associate one filter to it : # a member of group A tc class add dev eth0 parent 1:1 classid 1:11 htb rate 512kbps ceil 512kbps # tts associated filter to match his source port tc filter add dev eth0 protocol ip parent 1:0 prio 1 u32 match ip sport 50001 flowid 1:11 # a member of group A again tc class add dev eth0 parent 1:1 classid 1:12 htb rate 512kbps ceil 512kbps # tts associated filter to match his source port tc filter add dev eth0 protocol ip parent 1:0 prio 1 u32 match ip sport 61524 flowid 1:12 # a member of group B again tc class add dev eth0 parent 1:1 classid 1:13 htb rate 1000kbps ceil 1000kbps # tts associated filter to match his source port tc filter add dev eth0 protocol ip parent 1:0 prio 1 u32 match ip sport 57200 flowid 1:13 I already know that a source port could be the same if its coming from a different IP address the thing is the application is behind a proxy so I don't have to manage any IP address in that situation. I would like to know how to manage the fact that for all other users (request/source port, whatever you name it) could be limited at a given speed each. I mean that each connection should be able to use at max 100kbit/s for example, not a shared 100kbit/s. I also would like to know if there is a way to simplify my rules. I don't know if it is possible to use only one class per group and associate multiple filters to the same class so each users could be handled by one class and not one class per user. I appreciate any advice, thanks.

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  • How can I use the voice recognition used by Android on Ubuntu?

    - by aking1012
    If I'm developing an Android app that uses TTS and Voice recognition, which libraries are used for the same voice recognition and speech on Ubuntu? I'm assuming espeak for text to speech, but I'm unsure which voice recognition library and dictionary/learning/calibration system is used for voice recognition. I'ld like to make the app available on Ubuntu Desktop. as well as test it outside an emulator

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