I'm developing a web chat app using webrtc I want to know that can we change the voice of the user live calling in webrtc and is there any API for live call voice changer so let me know thanks
What you looking for is insertable streams api. It allows you to access the media stream and apply transformations to the stream.
Check out this example which applies low pass filter on the audio track. There's a link to code at the bottom of the page.
Related
I am trying to build an application which works this way: I as a user want to start a call with another user. The way I want the connection be made is by random. So it will connect to one of the many clients out there by random. Also when other clients try to make a call, it should connect to another random client and so on. I want those phone calls be made via application(such as WhatsApp) not as a phone call.
Now, the question is; is Twilio a good approach for this purpose?
If yes can you tell me which of their feature would fit my app the best?
Thanks for any suggestions!
Twilio developer evangelist here.
I can answer that Twilio would be a good approach for you to do this within your own application. I'd recommend using Twilio Video to build this as it allows cross platform communication via audio or video (in your case, you may not need the video, but this will give you the best audio quality).
As an example, my colleague Dominik built a video roulette application. It is the case that the interface was built in JavaScript for the web, but the idea would be the same for a native app. The code for the server side part of the application should give some insight into how to connect random pairings.
It's also possible to integrate Twilio Video with CallKit and Connection Services so that you can make outbound calls to other devices that ring like a real incoming call.
I am using this library https://github.com/muaz-khan/RTCMultiConnection to create web based audio broadcasting application like a radio but now my customer needs to take his phone and mobile calls audio stream and broadcast them along with his own audio stream
So I wonder if there is any way to do that without depending on other services
Any ideas would be highly appreciated
Thanks in advance
I built a conversation bot with text-to-speech, but no matter how well I tune it, the voice sounds robotic.
I think it would be simpler to have the conversation bot pick a pre-recorded audio and stream it back to the user.
Does anyone see issues with this?
Is there already an example of this so I don't reinvent the wheel?
This functionality needs to be implemented on the client side of the application. Watson Conversation Service can return a text answer and for example an index of the audio record you want to play.
This index then needs to be picked up by the client application communicating with Watson Conversation Service (e.g. a web page in node.js) and the audio record can be played to the user.
As for some examples...in Conversation Service docs there are links to github projects that integrate Watson Conversation Service with node.js web applications - these can be extended by adding the audio records and functionality that will play those records to the user.
Does anyone know how technically to send videos (i.e. Youtube Videos) to a Roku player? There is a "Twonky Beam" app that allows streaming and what it appears to do is to send .mp4 files to Roku for playback. See the demo here: http://gigaom.com/video/youtube-on-roku-twonky-airplay/
This is done without a "Twonky Beam" Roku app. Looks like something that Roku supports natively, although I cannot find anything documented.
I want to know how they were able to accomplish this without Roku being a UPNP or DLNA device.
Any insights here would be great!
There are discussions on how to extract the mp4 URL from YouTube here and here
In terms of how to do airplay style video playback on Roku, you would use the External Control Protocol to launch a channel with the URLs of the video you wish to play back, or once your channel is launched, us the ECP in combination with the roInput component to send the URL's to your channel. Your channel would then send the URLs to a video playback compoenent which would initiate playback from Youtube or whatever source you send it. If you want to play URL's from your device (android/IOS) you would need to run a web server on the device to serve videos to the device.
here is an Open Source YouTube project referenced in that second thread.
Any unofficial project that plays video's from YouTube is subject to DMCA takedown by YouTube should they decide your project does not fit with their goals.
roInput is not really well documented, here is an example that demonstrates both roInput and launch parameters (launch parameters are keywords you include in an http POST):
function main(params as object)
if params.parameter <> invalid then
print "This channnel was launched with Launch Parameters!"
print params
else
print "launched without input parameters"
end if
port=CreateObject("roMessagePort")
input=createobject("roInput")
input.setmessageport(port)
while true
msg=wait(100,port)
if type(msg)="roInputEvent" then
params=msg.getinfo()
print params
end if
end while
end function
so your parameters might be "vidurl=http://myserver.com/video300k.mp4&vidurl=http://myserver.com/video600k.mp4" if you wanted to send multiple bit rate videos.
there are plenty of examples of how to play video on a Roku in the RokuSDK, the simplest being the simplevideoplayer exmaple.
As to the last part of the question re UPNP, you can find a roku on your lan either via brute force telnet on port 8060 to every ip or by using SSDP, also documented in the ECP guide linked above
I want to create audio, video and text messagtes chat. Is it possible using WebRTC? Or it only allow audio and video chats?
One side of my app will be implemented using browser. An other one - using C++ native API.
Does anyone have examples in native C++ API and/or javascript?
The WebRTC specification is still very much in flux, but there's a DataChannel API in the spec that is implemented in an early form in both Firefox and Chrome. DataChannels are intended to allow you to send arbitrary bytes between peers, and the spec provides for both reliable (TCP-like) and unreliable (UDP-like) channels.
I am not sure if WebRTC allows for text chatting. I was able to successfully create an Android Application that performed all of this, but only with the combination of Google's Libjingle and WebRTC libraries. Within the Libjingle library there are several example programs/pieces of code that demonstrate the library's functionality. The call example in Libjingle sounds very similar to what you are wanting to do, and is what I built my Android application out of. The only thing is I have not yet ported it to an web browser, so I am not sure if Libjingle will work with that.
I have begun looking into it, and I have found some people on the WebRTC discussion group that have developed a very nice Multi-user video chat application for a web browser that is built using WebRTC. It is capable of video (along with voice) communications as well as text chatting. I do not know if this matters, but it all occurs within a single interface (meaning that it does not seem to allow for separated/singular form communications -- text only, voice only, video only). I am sure that it would not be too difficult to separate them all out if you wanted/needed. They have posted all of their code onto GitHub and seem to be actively updating and improving it.