does somebody have instructions, how do to make a RTSP client with Qt? I have already heard of live555, but I don't know how to link it with Qt.
Is there another way?
I would like to do it with Qt, so that it also runs under Linux and other platformens.
To have a RTSP client, you need to process the RTSP protocol one way or another.
Live555 is one way to do that, it is just a C++ library that can be linked with other applications, including Qt. It is certainly possible to link Live555 with Qt.
Another way would be to write your own RTSP client based off the RFC spec .
The last option would be to just use the Phonon framework in Qt. http://doc.trolltech.com/4.6/phonon-overview.html (provided your Phonon backend will support RTSP). This is the easiest way as Qt and the system handle all the backend decoding of the media, integrates seamlessly with Qt and does not require extra libraries to be linked in with your app.
Related
I'm making a call software using Qt and Qml, and I need to get the microphone feed from Qml running as webgl to the C++ side, if not straight to GStreamer using server.
I already have a Qt program as the client using GStreamer to push audio stream to the server. GStreamer, of course, doesn't go to the webgl client side though. I've found, that you can get permissions to use mic/camera from Qml, but I haven't found any example actually grabbing the stream from there. I've also checked out the usage of WebRTC. It seems like it could work with Qml and I have found some examples using it with GStreamer, but I haven't been able to get the combination of WebRTC and GStreamer working even with the examples.
So the full question:
How can I get the audio from the Qml running as webgl? Is there a way within Qt or do I have to go through WebRTC? If so, is there some simpler or more beginner friendly example than the Nirbheek's gstwebrtc demos for connecting WebRTC to GStreamer?
Not the answer I wanted, but this ended up working in my case:
As the C++ side is also running another Qt GUI, Qml to be specific, I
can use WebEngineView with html and javascript to never bother the C++
implementation with GStreamer for WebRTC. So currently I'm running
PeerJS on both sides of the connection with PeerJS' signaling
server in between.
I'd have preferred to use C++ with GStreamer to connect to WebRTC, but
I can't find other easy way for me to connect browser's audio to Qt.
edit: I apologise, this answer doesn't work in the end. I have been testing the programs on a single computer, thus I didn't realize WebGL hosted Qml doesn't run WebEngineView's Javascript on the frontend, but backend instead.
I would like, from a native Windows application using C++, to receive video/audio data sent from a browser located in a remote location. It seems like WebRTC is the way to go for this.
Most information I find is about how to interact with the browser to write WebRTC apps, but it may case the data would be received by my C++ app. Is it correct that I would need to use the WebRTC Native Code package for this, which is described as being 'for browser developers'? Document is located here: http://www.webrtc.org/webrtc-native-code-package
And what if I want to send video/audio data that I generate (ie not directly coming from a webcam and microphone), would I be able to send it to the remote location browser?
Any sample code out there which does something like I'm trying to accomplish?
The wording in that link is a bit misleading. They intend people that are developing browsers to use the native code, and advise those that are developing "applications" in a browser to use the WebRTC API.
I have worked with their native code for over a year to develop an Android application that is capable of performing audio and / or video calls between other Android devices and to browsers. So, I a pretty sure that it is completely possible to to take their native code and create a Windows application (especially since they have example code that does that for Linux and Mac -- look at peerconnection client and peerconnection server for this). You might have to write and re-write code to get it to work on Windows.
As for as data that you generate. In the Android project that I worked with, we didn't rely on the Android device / system to provide us with video, we captured and sent that out our selves using the "LibJingle" / WebRTC libraries. So, I know that that is possible, as long as you provide the libraries with video data in the correct format. I would imagine that one would be able to do the same with audio, but we never fiddled with that, so I cannot say for sure.
And as for example code, I can only suggest Luke Weber's gitbug repositories. Although it is for Android, it might be of some help to look at how he interfaces with the two libraries. Probably the better code to look at is the peerconnection client stuff that comes in the "LibJingle" second of the native code. [edit]: That is located in /talk/examples/peerconection/client/ .
If you get lost from my use of "LibJingle", that will show you when I started working with all of this code. Sometime around July of 2013 they migrated "LibJingle" into the WebRTC "talk" folder. From everything that I have seen, they are the same thing, just with the location and named changed.
The project I'm currently working on, requires an addition to the already existing VoIP capabilities. The core for speech processing is in C, the remainder is in C++ with Qt - the audio is handled via portaudio. The connection between users is currently established via UDP, which I think has to be changed for the planned video connection. Developing platform is Windows on VS2012 - however, the system is cross-platform.
In a nutshell, what I want to do is: Grab the video signal from a webcam, synchronize audio coming from C core and video from webcam and use a library and codecs for (de-)coding/muxing the signals on the respective sides and sending via RTP. The system should be capable of multicast transmission.
I did some research for possible libraries and stumbled upon ffmpeg and libVLC. For the codec I thought about using x264. And if I'm correct, ffmpeg and libVLC should both be capable of what I'm looking for?
However I'm not sure which one to pick, and from their documentations I really can't extract, which library is the better fit. Has anybody had similar problems and can help me out - I'm quite a newbie, when it comes to video processing and encoding.
Extra question: Do you have any hints or approaches on syncing the video and audio signals?
If anyone is interested, this is what I ended up doing:
I am using the WebM container format, VP8 with Vorbis currently (but going to change to VP9 with Opus soon if out of beta), handled by ffmpeg/libav libraries for encoding/decoding/muxing etc. and SDL for displaying and threading. ffmpeg/libav was cross-compiled on Unix with LGPL support to keep our project closed source.
i am electronics and communication engineering student. i am working on a project in which a microcontroller send data to serial port of computer (asynchronously).
i currently learning wxWidgets for my GUI work.
my question-
is it possible to do using wxWidgets to get data and display. is it very difficult to implement this.
i search for it but not get direct answer that how to implement this. so suggest me what i need to read for this. and what alternative available to do this. i use windows 7.
i also need to upload this data to web server (it is needed later)
wxWidgets, to the best of my knowledge, does not implement serial I/O facilities. There are other libraries available that provide this. One of the best choices is probably boost::asio (see http://www.boost.org/doc/libs/1_40_0/doc/html/boost_asio/overview/serial_ports.html)
wxWidgets is a cross-platform GUI framework.
Reading a serial port is nothing to do with the GUI, and not cross platform.
So wxWidgets has nothing to do with reading serial ports.
You can read/write from/to a serial port using the windows API.
CreateFile opens the port
ReadFile reads data from the port
Difficult? Kinda!
I've never used it myself but there is wxCTB.
I want to write a program to create VOIP application for my final year project. For this time being, I spend my time to understand how SIP works and later I need to implement it into my project. As far as i know, the protocol requirement to build this project are:
UDP
RTP
As for my project, it doesn't requires me to write a complex coding. I just need to provide a server that can be used as a bridge for clients to communicate. So, is there any resources that I can used for my reference to start programming my project?
Maybe these will be helpful to look at:
Open source RTP library:
jrtplib:
Open source RTSP, RTP with a SIP client AFAIR:
liveMedia
Open source SIP library: reSIProcate
Your best bet is pjsip.
http://www.pjsip.org/
It runs on all kinds of OSes
OpenSipStack is quite nice, it's 100% C++.
The project also contains a library for creating clients using ATL.
Take a look at OPAL. It allows you to write applications that will support SIP, H.323 and IAX2.