So far, I have found red5, but I can't get it to run (no video arrives at the server side), so I was looking for a flash-based getUserMedia and found: https://github.com/addyosmani/getUserMedia.js . But how do I get the video on the server? IE10 doesn't support webRTC, which http://lynckia.com/licode/ is built on.
getUserMedia is a WebRTC API. It doesn't exist on IE10.
Your alternatives are
Go with a Flash based solution (red5, Wowza, etc)
Use a plugin for WebRTC on IE (check out this one: https://bloggeek.me/temasys-free-webrtc-plugin/
Use Ziggeo (they should be able to use WebRTC or Flash automatically for you, taking care of all relevant transcoding and format changes necessary to playback the recorded stream). CameraTag (virtually the same at first glance) was also suggested in another answer, and likely, there are more.
Related
I've doing Javascript programming for some time but it's always been related to data updating, saving, manipulating, etc.
I have no idea how something like an in-browser audio player gets audio (especially live, streaming audio) from the internet and plays it out of my computer speakers.
How does this happen in Javascript?
For example, how does a website deliver live audio to my speakers using Javascript? http://player.streamtheworld.com/liveplayer.php?callsign=WVIEAM
The live audio is not much different from pre-recorded audio... it's just played back as it's received, and when live it's encoded as it's recorded.
In browsers these days, the most basic form of streaming audio is a simple <audio> tag. By changing the src attribute from a file to a stream, you're up and running:
<audio src="http://cdn.audiopump.co/waug/main_mp3_256k" />
The browser doesn't know or care in this case that the audio is a live stream. All it knows is that there's some media data that it's fetching via HTTP, and playing back while it comes in.
If your browser compatibility is good, it would be preferable to use the MediaSource API, giving you more control (such as switching to a different quality stream mid-stream, like in HLS) and ensuring that the browser doesn't try to cache what is effectively an inifinitely sized file.
For example, how does a website deliver live audio to my speakers using Javascript? http://player.streamtheworld.com/liveplayer.php?callsign=WVIEAM
This particular site is ran by Triton Digital, and they still use Flash. Many sites still do this as a holdover from a time when HTML5 audio was not widely supported. There is little reason to do this today.
Other reasons to use Flash include incompatible server protocols. If your streaming server is using RTMP, you're stuck with Flash as browsers don't speak RTMP.
There used to be an issue with streaming AAC in-browser due to browsers not properly handling AAC wrapped in ADTS. (This encapsulation is required for streaming AAC in most situations.) Most browsers have resolved this, but I suspect that this is the reason Triton Digital is still using their Flash solution. By using Flash, they can play AAC/ADTS streams.
I'm a bit at my wits end here.
I want to stream a live video broadcast to a web browser.
Currently I use ffmpeg to stream a directshow live source as a webm stream to node.js which then forwards the stream to the http request from the <video> element. So far everything works.
live source -> ffmpeg -> POST [webm] -> node.js -> GET [webm] -> video tag
My problem is that the source clock and the web clients clock doesn't exactly match each other (not that surprising). For video this is not a problem, dropping or duplicating a frame every now and then is not noticeable. However, with audio it is another issue. From what I've been able to figure out so far Chrome (or any other browser) does not perform any form of audio resampling compensation (e.g. swr_set_compensation from ffmpeg) to compensate this mismatch. Instead I get quite audible audio distortions (a loud beep) when the playback buffer runs out of samples.
My question is whether it is possible to achieve proper playback (with audio) of a live source in a web browser?
I haven't tried using silverlight or flash for playback yet. Would that possibly work better?
Live media (audio or/and video) streaming to a web browser has been possible for a couple of years though it is still making progress as of today. It is the next big thing for media on the web and many platforms like Youtube are already on board.
A typical live media streaming scenario is:
audio/video feed > transcoding > streaming > player
At each step you have several technological possibilities available. However I should already mention here that the road to live media streaming is paved with proprietary technologies.
audio/video feed: either raw or very lightly compressed media format and cannot be uploaded as such to the Internet. You need to transcode it. You may have to use a grabbing device like a PCI Express card or USB/thunderbolt device to get your cam onto a computer.
transcoding: you have software (ffmpeg, Flash media live encoder, Wirecast) or hardware solutions (streamingmedia.com has a wealth of information on the subject like here). H264/AAC is the current media professional standard and streams are often transcoded to multiple renditions (bitrate) to suit different network conditions.
streaming: you most likely need to target multiple devices to deliver your live stream. Not all devices support the same streaming protocol. HLS works on Apple devices and Android > 4.1. HDS or RTMP works in Flash, Smooth streaming in Silverlight. You cannot reach all devices with one protocol so in this case you would need a streaming server like Wowza or Red5. A streaming server take as an input a transcoded live stream and prepare it for cross device delivery while sustaining a massive number of simultaneous connections (over a thousand is not uncommon nowadays). It can also add functionalities like DVR or DRM. As of today the effort is around HTTP adaptive bitrate delivery. Large companies add CDN support for global delivery.
player: to display your live stream with various options like custom layout, closed captions, ads, chat module and more. Flash has been leading the market up until now for live media streaming on desktop. You can use HTML5 video for iOS and Android where HLS is supported.
Coming in fast is MPEG DASH and it works live with HTML5 video. There is a JS lib that supports live. I have tested it and it works though I may not use it for a production case scenario just yet as it is still a bit clunky (on demand support is better) and browser support is narrow at the moment (As of 8/30/13, Desktop Chrome, Desktop Internet Explorer 11, and Mobile Chrome Beta for Android are the only browsers supported).
I cannot comment much on your solution because I have not used node.js for streaming but it sounds like an interesting effort. A typical solution I would use relating to your case:
Device > ffmpeg (H264/AAC) > Wowza > Hybrid player (Flash + HTML5).
Instead of Wowza you could use Red5 (free/open source - but not much activity as of late). You can also look into Nginx RTMP module which supports HLS and MPEG DASH on top of RTMP.
For flash I use Strobe from Adobe which support live streaming and is easy to set up and a fallback to HTML5 where flash is not supported. I use SWFObject lib to detect flash support and feed a HLS URL to an HTML5 video tag for mobile devices. You can use RTSP for Android < 4.1 and other mobile devices.
Another thing I should mention is real time communications. For video/audio conferencing you could have a look at WebRTC. Those 2 articles should get you on the right track. Here and here. WebRTC will work great for one to few, one to one, few to few. If you need to support more concurrent connections you can have a look at Licode or tokbox.
I've been experimenting with WebRTC and its ability to stream video/audio using p2p, however I've found that it's not able to stream any resolutions > 720p. Using getUserMedia() I'm able to capture the local webcam in 1080p, but it seems to fail at encoding/decoding in 1080p (I'm not sure which it is) and the client-side displays the video in 1280x720.
There are a few bug reports regarding this issue: Here and here.
Is there a way I can capture the webcam using getUserMedia(), encode the video myself in 1080p and still use WebRTC for its peer-to-peer capabilities?
Basically I'm trying to achieve the same sort of thing shown in this video. That is, live streaming a video and sharing it with other viewers so there's significantly less server/broadcaster bandwidth used. WebRTC seemed the obvious choice for this.
Are there any alternative solutions, besides waiting for the WebRTC bug to be fixed? Thanks.
This issue was resolved on the tickets you posted:
https://code.google.com/p/webrtc/issues/detail?id=1750
Which points to this url for verification.
I want to be able to record videos with audio using HTML and Javascript.
After some research i can get video streaming with getUserMedia. Also There is WebRTC for recording but as far as i understood its not yet implemented in desktop browsers (only mobile browsers support it). So now i can just capture video, but i cant save it to server or record it.
What other options do i have ?. Does anyone knows a good flash alternative or HTML5 alternative that allows me to capture and save video to server with audio and also has maximum time of recording
Full disclosure: I work for Ziggeo.
When it comes to WebRTC, here is the rundown for browsers supporting it:
on Chrome and Opera, you have to record audio and video separately and encode them yourself in JS; then, send them to your servers and transcode them using e.g. ffmpeg to mp4s and other target formats
on Firefox, you can get a webm object for video and audio combined and send it to your servers.
For all other browsers and older versions of the ones mentioned you'd need to fall back to Flash recording which usually is based on RTMP and flv.
I would like to make a web application where people can add recorded sounds and samples to a timeline.
I want it to output 1 soundfile (approximately 3min long) which will be send to the server.
Now I would like to do this with the HTML5 Audio API and found out that I could do this with the AudioContext But Audiocontext is only supported in chrome.
Now I do dislike flash and I wanted to ask if there is any way to do this with HTML5 with decent browser support (so newest Chrome, IE and Firefox).
Some thoughts:
I could record audio using the HTML5 audio API and the user can add it to the timeline. When finished the client will upload all the audio files to the server including the location on the timetable. The server can combine the audio files to make it one file. Now this is an solution, but I would prefer to do this kind of work on the client side
At the moment in my opinion it is not possible (if it should be html5 and supported by IE and Firefox), see the List of Browser that Support the Audio API
Browsers with Audio API Support. But this Information could be outdated already.(these Browsers update so frequently)
You could wait, first serve only Chrome and hope the other Browser, catch up(IE might be a problem). Or you use Java (if you dont like flash). The other technologie out there is Silverlight, but it is "dead", so i wouldn't recommend it.
I hope my input helps a bit.