Is any way to capture video from usb camera with javafx or other way with java except jmf?
Right now, javafx.scene.media provides playback support only.
From the documentation -
public Media(java.lang.String source)
Constructs a Media instance. This is the only way to specify the media
source. The source must represent a valid URI and is immutable. Only
HTTP, FILE, and JAR URLs are supported. If the provided URL is invalid
then an exception will be thrown.
http://docs.oracle.com/javafx/2/media/simpleplayer.htm
http://docs.oracle.com/javafx/2/api/javafx/scene/media/package-summary.html
Related
I have a transfer of a live video stream from a server to the javascript function of a client browser:
server: gstreamer x264enc-hardware ! whatever-I-want ! appsink
=== transfer of data stream with a proprietary protocol ===>
HTML5 browser client: javascript function receives data sent by the appsink
In other words, I'm trying to display a h264 live stream created on a server with a proprietary transfer protocol, the data re-appearing in a javascript function inside an HTML5 client browser.
I was thinking of using MediaSource MSE in the browser to decode h264 and display the image.
Note that the video stream settings (video only, resolution, bandwidth) are fixed and known on both sides. So, everything can be hard-coded and the purpose is not to implement a generic solution.
What could I use on the server side (replacement of the "whatever-I-want" gstreamer plugin) so that the work in the HTML5 browser is not too complicated?
One solution would be to do nothing on the server side and use the broadway.js library to decode NALU h264 in javascript but it obviously doesn't leverage video MediaSource and the decoding capability of the browser.
Could I use Gstreamer avmux_dash and hope that MediaSource can input the transmitted data?
Alternatively, how could I create "MP4 fragments" and could MediaSource read them "easily"?
One approach, that has been used by some major players in the past to translate from one streaming protocol to another, is to receive in your proprietary transfer protocol and then re-package into HLS or DASH on a local server on the device.
You can then stream from that local host to a regular HLS or DASh player on the device.
It sounds inefficient (it is inefficient) but it works, even on mobile devices which their lower processing and power capabilities.
I have a product that can analyze video after inputting an rtsp url.
I would like to use a webcam to stream and feed my product the webcam rtsp.
How can I do that?
It will depend on the webcam you are using - most support RTSP but many do not publish the interface to access the stream as they are designed to be used with the webcam's own companion app.
There are some web resources which provide the RTSP urls for common web cams - you may find it hard to find a match as new versions of webcams roll out but it should give you a feel how to try accessing a vendors camera if you have a specific web cam you are testing against. Some examples (at the time of writing):
https://www.getscw.com/decoding/rtsp
https://soleratec.com/get-support/rtsp/
If you can't find the info for the camera you are using, and you have the companion app, you can also use a network sniffer tool like Wireshark (https://www.wireshark.org) and try to search the traffic for 'rtsp://' pattern.
If you just need to test your app and have access to a raspberry pi with a camera module you can also use this to generate an RTSP stream - there are several approaches for this but one I have found reliable is the v4l2rtspserver server:
https://github.com/mpromonet/v4l2rtspserver
There are specific instructions for setting it up on PI (https://github.com/mpromonet/v4l2rtspserver/wiki/Setup-on-Pi) and you can also verify it is working using VLC player on a laptop etc before testing in your specific application.
There are also a small number of test RTSP urls available on the web - the most reliable seem to be the one at this link provided by Wowza (again, link valid at time of writing):
https://www.wowza.com/html/mobile.html
I want to let WebRTC encoded and play h264(NAL) stream(local file).
In the WebRTC tutorial, getUserMedia is use for get local camera connecting to the system, I don`t know if the getUserMedia function support
capture the local stream file like h264 stream.
If it doesn't work that way, may be I should modify WebRTC source code(I'm studying it).
Here is the question, If i change WebRTC code, how can i integration the new code into browser? Made it a plugin?
Firefox supports an extension to the <video> element that you can use to do this.
First, set the source of a video element:
v1.src = "file:///...";
Then you can call the (currently prefixed) mozCaptureStream or mozCaptureStreamUntilEnded function to get a MediaStream.
stream = v1.mozCaptureStream();
The proposed specification.
Note however that you need to ensure that the file is same origin with respect to the page. The same origin rules for file:/// are probably going to cause issues. Otherwise your MediaStream isn't going to be accessible to you. One way to ensure that is not to set the location directly, but to load the file using an <input type="file"> element.
As noted in other answers, Firefox currently only supports the baseline profile of H.264.
First, you are right getusermedia will not work for you. However, there are a couple of options.
Hack a stream together using RTCDataChannel. Breaking up the media stream and delivering each packet and then handling it on the client side.
Take a look at this demo for precorded media streams. I do not believe that H264 is addressed but it could help you on your way(probably for Firefox only)
Use some sort of webrtc breaker/endpoint that is native to stream the file. I know specifically that others(including myself) have streamed H264 to Firefox through the Janus-Gateway
Couple of asides:
Firefox only supports Baseline profiles in streaming h264 for a webrtc peerconnection
Chrome does not support h264 for webrtc at all
Are you trying to have getUserMedia return a h.264 encoded stream?
In which case, today it will only be possible with Firefox today, under some specific environment (cisco 264 plugin installed) and only for the base profile.
Chrome promised in november to add this capacity, but there is no timeline that I know of Expect at least Q2 2015.
Using our (temasys) commercial plugin you will soon be able to do that in IE and Safari.
Those are the only options on client side I can think of. On server side you can use whatever you want to transcode, including janus, kurento, powermedia, licode/lynkia, ....
Note: using other means like Datachannel or WebSocket are ok to transfer files, but would greatly reduce the user experience as you would not have all the added recovery (and security) mechanisms included in SRTP, DTLS, and would also not have specific mistreated media enhancements that are in webRTC like jitter, buffers, netQ, ect ...
I am new to WebRTC and trying to figure out how to create a program outside a browser which receives a WebRTC audio stream and outputs it on speakers.
Are there any WebRTC libraries for Java or C#?
That receiver will be running on a linux machine.
--
I've been thinking about using getUserMedia() to access the microphone. But then:
In what format will such a stream be transmitted?
Let's say I use WebRTC2SIP and build a Java endpoint using JSIP;
or I just use a socket and send the stream over http.
What audio format will I get on the receiver side? So far I have read WebRTC does compress the stream somehow.
I guess there are two ways for you:
build the whole WebRTC voice engine for android/iOS or Mac etc., and just use the API provide by VOE.
build standalone NS/VAD/AECM/AGC modules and using it in your project. for example, you build standalone NS module for android mobile, you use AudioRecord(java layer, android things) to record sound from MIC, and do the noise suppression process on these data(jni layer, WebRTC things), and finally playback the processed data by using AudioTrack(java layer, android things).
EDIT:
for the 2nd situation, the format is PCM raw data.
Check out the working Audio demo and code at demo.easyrtc.com
The code is all open source and can be checked out at https://github.com/priologic/easyrtc
You can look for any known issues around easyRTC at our forum at
https://groups.google.com/forum/#!forum/easyrtc
Also check out our main site at easyrtc.com
I'm trying used mediaplay play http or rtsp protocol uri from server, when I play the
address as http://**.wma or *.mp3 ,it can working,but I tryed played the address as
"http://qr.fm.qq.com/qqradio?qqradio",it didn't working.
and also I'trying used VideoView play rstp protocol uri from server,when I play the adress as
“rstp：//*.sdp”,it can working ,but I tryed play the adress as "rtsp://vs1.thmz.com/radio31"
,it didn't working.
Anybody help me and tell me how
These are live streams, not static files, so, while it may play back some .wma and .mp3 content - these live streams are not defined like that.
Are you sure the first stream link is valid? After a quick scan with nmap, it seems you may need to be in china to connect to this feed (qq.com Registrant Country Code - CN) I get 1000 scanned ports all filtered, usually means a firewall blocking specific geographic regions.
rtsp://vs1.thmz.com/radio31 -> This is a Windows Media Audio stream, using WMA2 codec, delivered via RTSP, which according to the Android Supported Media Formats: http://developer.android.com/guide/appendix/media-formats.html - is NOT supported.