I'm using c++ and poco libraries. I'm trying to implement a video streaming httpserver.
Initially i used Poco::StreamCopier.
But client failed to stream.
Instead client is downloading the video.
How can i make the server to send a streamresponse so that client can stream the video in browser instead of downloading?
While not within POCO, you could use ffmpeg. It has streaming servers for a number of video protocols and is written in C (which you could write POCO-like adapters for).
http://ffmpeg.org/ffmpeg.html#rtp
http://ffmpeg.org/ffmpeg.html#toc-Protocols
http://git.videolan.org/?p=ffmpeg.git;a=tree
And it has a pretty liberal license:
http://ffmpeg.org/legal.html
You need to research which video encoding and container that is right for streaming -- not all video files can stream
Without using something to decode the video at the other end but simply over HTTP, you can use The mime encoding "content-type:multipart/x-mixed-replace; boundary=..." and send a series of jpeg images.
This is actually called M-JPEG over HTTP. See: http://en.wikipedia.org/wiki/Motion_JPEG
The browser will replace each image as it receives it which makes it look like it's video. It's probably the easiest way to stream video to a browser and many IP webcameras support this natively.
However, it's not bandwidth friendly by any means since it has to send a whole jpeg file for each frame. So if you're going to be using this over the internet it'll work but will use more bandwidth than other method.
However, It is naively supported in most browsers now and it sounds like that is what you're after.
Related
I am actually working on a server-client multimedia player. This player can be a server to stream a MP3 file (or wma, wav, ogg, flac ...) over the network to another player (client).
I worked first on a basic network communication (client-server), that send and receive bits. But I have a problem : the audio encoding. I need a tool to encode the audio data to be able to send a little part of it through the network and let the client play it before the next part is coming.
I saw a few tools on internet such as BASS library, Live555 ... I used to work with PortAudio for student's projects but I hate it.
So basically, I need a tool to encode audio data (server side), (I can send it over lan), and decode data to play it (client-side).
Do you guys have some ideas about how to do it ? Which tool could be useful for me in that case?
PS : I am trying to use Qt library for the network interface (it is efficient, and it works on windows, linux, mac) ... Is there any audio streaming tool included in the Qt library ?
You can try FFMPEG. It can convert almost anything to anything (so it claims) and it is a widely used open source library.
We use it in our application mainly for decoding video/audio streams.
Is it posible to have 2 sources in the same mount point?
Example:
Source 1 (from S1 IP adress) sends music to IP Icecast2 server.
Source 2 (from S2 IP adress) sends voice to Ip Icecast2 server.
Listener run: mplayer ip_icecast2:8000/example.ogg..
The listener listen the music and sound at the send time..
Liquidsoap should be able to handle the mixing and setting the proper metadata.. The web site is at http://savonet.sourceforge.net/
Yes, but not with Icecast alone.
What you need to do is mix the two streams. Icecast doesn't have any features for doing anything like this. There are many ways to do this. I would probably look at mixing the streams together with FFMPEG. You can use the amerge and amix filters.
Now, you need to get the output of FFMPEG to your Icecast server. With some scripting, you should be able to pipe the STDOUT from FFMPEG to a TCP connection to Icecast. Prior to sending data, you will need to send the appropriate headers and what not.
you can actually do it without anything else, you need to specify 3 streams
stream
live
autodj
the trick relies in a tag called fallback, you directly configure it on icecast xml file, and it does something like - if the live audio is not available, fall back to autodj, and any1 of it will directly play on stream, giving preference to live
I'm assuming you mean one source of audio and one of speaking, from different URLs. If you don't know how to use Soap, you could grab both the audio and music streams using a 3rd party application like SAM broadcaster.
This will decode the streams and mix them like a conventional audio mixer before re-encoding and sending out to a single Icecast server as one stream.
Keep in mind, if you are doing voice overs, there will be latency to deal with. i.e. speaking will be heard by the final listener slightly after the part of audio you will be speaking to. This depends on the buffer lengths involved, and is because SAM broadcaster will be 'listening' to the audio at the same place you are (assuming you are speaking to the source audio stream). Then you need to add to that, the playing buffer SAM needs to process, playing your voice's stream to be mixed and passed on.
I have a Windows native desktop app (C++/Delphi), and I'm successfully using Directshow to display live video in it from a 'local' video capture device.
The next thing I want to do is display video from a 'remote' capture device, streamed over the LAN.
To stream the video, I guess I can use something like Expression Encoder or VLC, but I'm not sure what's the easiest way to receive/decode the streamed video. Inserting an ActiveX VLC or Flash player might be one option (although the licensing may be an issue then), but I was wondering if there's any way to achieve this with Directshow...
Application needs to run on XP, and the video decoding should ideally be royalty free.
Suggestions, please!
Using Directshow for receiving and displaying your video can work, but the simplicity, "openness" and performances will depend on the video format and streaming method you'll be using.
A lot of open/free source filters exist for RTSP (e.g. based on live555), but you may also find that creating your own source filter is a better fit.
The best solution won't be the same for H264 diffusion through RTP/RTSP and for MJPEG diffusion through simple UDP.
In my application we will present the video stream from a traffic camera to a client viewer. (And eventually several client viewers.) The client should have the ability to watch the live video or rewind the video and watch earlier footage including video that occurred prior to connecting with the video stream. We intend to use wxWidgets to view the video and within that we will probably use the wxMediaCtrl.
Now, from the above statements some of you might be thinking "Hey, he doesn't know what he's talking about." And you would be correct! I'm new to these concepts and I confused by the surplus of information. Are the statements above reasonable? Can anyone recommend a basic server/client architecture for this? We will definitely be using C++ wxWidgets for the GUI, but perhaps wxMediaCtrl is not what I want... should I be directly using something like the ffmpeg libraries?
Our current method seems less than optimal. The server extracts a bitmap from each video frame and then waits for the single client to send a "next frame" message, at which point the server sends the bitmap. Effectively we've recreated our own awkward, non-standard, inefficient, and low-functionality video streaming protocol and viewer. There has to be something better!
You should check out this C++ RTMP Server: http://www.rtmpd.com/. I quickly downloaded, compiled and successfully tested it without any real problems (on Ubuntu Maverick). The documentation is pretty good if a little all over the place. I suspect that once you have a streaming media server capable of supporting the typical protocols (which rtmpd seems to do), then writing a client should fall into place naturally, especially if you're using wxWidgets as the interface api. Of course, it's easy to write that here, from the comfort of my living room, it'll be a different story when you're knee deep in code :)
you can modify your software such that:
The server connect, server grabs an image, passes it to ffmpeg establishing stream, then copy the encoded data from ffmpeg stream and send to client via network, if the connection drops, close the ffmpeg stream.
Maybe you can use the following to your own advantage:
http://www.kirsle.net/blog.html?u=kirsle&id=63
There is a player called VLC. It has a library for c++ and you can use it to embed the player in your GUI application. It supports a very wide range of protocols. So you should leave connecting, retrieving and playing jobs to VLC and take care of the starting and stopping jobs only. It would be easy and probably a better solution than doing it yourself.
For media playing facility, both music and audio, you can a look on GStream. And talking about the server, I think Twisted (A network library in Python) should be good option. The famous live video social website justin.tv is based on Twisted. Here you can read the story from here. Also, I built a group of server for streaming audio on Twisted, too. They can serve thousands of listener on line in same time.
I have an application that grabs video from multiple webcams, does some image processing, and displays the result on the screen. I'd like to be able to stream the video output on to the web - preferably to some kind of distribution service rather than connecting to clients directly myself.
So my questions are:
Do such streaming distribution services exist? I'm thinking of something like ShoutCAST relays, but for video. I'm aware of ustream.tv, but I think they just take a direct webcam connection rather than allow you to send any stream.
If so, is there a standard protocol for doing this?
If so, is there a free library implementation of this protocol for Win32?
Ideally I'd just like to throw a frame of video in DIB format at a SendToServer(bitmap) function, and have it compress, send, and distribute it for me ;)
Take a look at video LAN client (or VLC for short) as a means for streaming video.
As for distribution sites, I don't know how well it works with ustream.tv and similar new services.
ustream.tv works by using Adobe Flash's support for reading input from a webcam. To fake it out, you need a fake webcam driver. Looking on the ustream.tv site, they point to an application called WebCamMax that allows effects and splicing in video. It works by creating a pseudo-webcam that mixes video from one or more cameras along with other sources. Since that app can do it, your own code could do that too, although you'll probably need to write a Windows driver to get it all working correctly.