voice

Today I'm very exited about the speech synthesis function is available in iOS7. I want to select the male voice(default in OSX, called alex). I don't know what's the BCP-47 code for him, and BTW how to get the full list of all voice code Here's how to get the BCP-47 codes of the available voices: for (AVSpeechSynthesisVoice *voice in [AVSpeechSynthesisVoice speechVoices]) { NSLog(@"%@", voice.language); } Alex's locale is "English - United States" (en-US), as you can see in the Dictation & Speech control panel on OS X. (Click "Customize..." in the "System Voice" drop down.) iOS 8 added Hebrew,

I'm looking for a simple C# real-time voice detection library. The input should be an audio stream, and the output should be "human voice" or "not a human voice". I have no knowledge in speech recognition or signal processing, and I'll appreciate any kind of assistance. Kevin Junghans Take a look at the answer for " Detecting audio silence in WAV files using C# ". I am assuming the input is a WAV file. If not please provide the format of the audio stream, or if you are intending on taking input from the microphone directly. If you can measure the amount of silence in an audio stream and you

Closed . This question needs to be more focused. It is not currently accepting answers. Learn more . Want to improve this question? Update the question so it focuses on one problem only by editing this post . Closed 5 years ago . We are looking to implement a group - voice - chat in the browser using node.js on the server. Is this possible? If you want your solution to be purely JavaScript based on both, server and client side, then you would probably have to wait for the WebRTC , Device API or media capture API and then implement node.js backend for it. Other than that it is, I would say, a

I want to implement an app by which voice & text chat between two iOS devcices under a network become possible. I dont need voice call facility to any phone like LinPhone or SiPhone. I have studied over them & found too complex for me. Is there any simple SDK to make this possible??? By the way, user identification may be done by e-mail verification.... I think one best way to do it using XMPP Framework. Using XMPP you can send files & text to other persons. Using it you can record voice message & send it across. There are some links for it: GitHub Project Link for XMPP Chat Building a Jabber

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: My company is currently using Grasshopper as our voice system for receiving and routing inbound customer support calls. It's working pretty well. Except that it's 2014 and people expect to be able to text issues to our toll free customer support number and get responses. Grasshopper doesn't support receiving/forwarding SMS. So I want to use Twilio just for receiving inbound text and Grasshopper for voice on the same number . It seems like a number has to be registered either with Twilio or Grasshopper and can't be shared. Does anyone know if

I am trying to record the voice in android But it will create the .mp3 file on the path (sdcard/filename) But when i run this file it doesen't play because it doesn't record the voice. Here is My code public void onClick(View v) { // TODO Auto-generated method stub switch(v.getId()){ case(R.id.Button01): try { //audio.start(); startRecord(); } catch (IOException e) { // TODO Auto-generated catch block e.printStackTrace(); } case(R.id.Button02): //audio.stop(); stopRecord(); } } private void startRecord() throws IllegalStateException, IOException{ // recorder = new MediaRecorder(); recorder

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 由 翻译 强力驱动 问题: I've coded an app that streams audio over TCP from client to server but it does not work, i.e. no audible output. Could you check my code tell me whats wrong with it? Client: public void startStreaming () { Thread streamThread = new Thread ( new Runnable () { @Override public void run () { try { int minBufSize = AudioRecord . getMinBufferSize ( sampleRate , channelConfig , audioFormat ); Log . d ( "VD " , "Bufer intioalised " + minBufSize ); short [] buffer = new short [ minBufSize ]; Log . d ( "VS" , "Buffer created of size .c" +

I'm trying to detect voice throught input from the microphone in real-time. I allready receive the input, execute FFT algorithm and have the result in dB. I have a frequency domain, a time domain and a spectogram. How can I get the fundamental frequency? If I get the fundamental frequency can I specify that if the frequency is between certain values, then it is voice that we are talking? Is there any other way to do this with the things that I allready have? Tks in advance There are many different algorithms for frequency estimation , and the right one to use depends on what you're doing. What