I'm interested in uploading a series of files to my web server and directing viewers to page which will autoplay the videos from a specific point dependent on the current time. My intention is to create the illusion of a live stream or actual TV channel, where they are unable to control the playback, but will return to the same point if they refresh the page.
I'm having difficulty finding answers, since it's descriptively so close to an actual webcast.
Here's my thought process on a solution.
Use the JavaScript Date Object API to capture the current time
Include your video with preload set to true, and controls false
Then use onLoad() and setup your video in JS
$(video).get(0).currentTime = XX; //You need an algorithm based on the time & length of video
$(video).get(0).play();
Related
I am newly working with DotNetBrowser. wanted to record video using the webcam. is there any way to do that using Dotnetbrwser?
IMediaDevices mediaDevices = engine.MediaDevices;
IEnumerable<MediaDevice> videoDevices = mediaDevices.VideoCaptureDevices;
IEnumerable<MediaDevice> audioDevices = mediaDevices.AudioCaptureDevices;
browser = engine.CreateBrowser ();
In fact, DotNetBrowser does not perform video capturing directly. However, you can consider creating a web page that uses WebRTC MediaStream Recording API and displaying it in DotNetBrowser.
https://developer.mozilla.org/en-US/docs/Web/API/MediaStream_Recording_API
Here is an example of the web page that captures the video from the web camera:
https://webrtc.github.io/samples/src/content/getusermedia/record/
https://github.com/webrtc/samples/tree/gh-pages/src/content/getusermedia/record
In case of any further questions, feel free to contact us at customer-care#teamdev.com.
I have a app where I play different code-generated sounds. I place these sounds in a AudioBufferSourceNode.
I allow the the user to choose what output device to play the sound through, so I use a MediaStreamAudioDestinationNode with its stream used as the source for an Audio Element. This way when the user chooses an audio output to play the sound to, I set the Sink Id of the Audio element to the requested audio output.
So I have AudioBufferSourceNode -> some Audio Graph (gain nodes, etc) -> MediaStreamAudioDestinationNode -> Audio element.
When I Play the first sound, it sound fine. But when I create a new source and connect it to the same MediaStreamAudioDestinationNode, the sound is played with the wrong pitch.
I created a Fiddle that shows the problem.
Is this a bug, or am I doing something wrong?
The problem was identified based on the OP Chrome Ticket.
It seems to come from the lack of sync between AudioElement and its source AudioNode (AudioBufferSourceNode, OscillatorNode, etc.) when you pause the source and play it back again.
The solution is to always call AudioElement.pause() and AudioElement.start() alongside your source stop and start.
https://jsfiddle.net/k1r7o0xj/3/
It's possible to dynamically change your graph layout by using .connect() and .disconnect(), even when audio is playing or sent through a stream (which could even be streamed over WebRTC).
I couldn't find a reference in the spec, so I'm pretty sure this is taken for granted.
For example, if you have two AudioBufferSourceNodes bufferSource1 and bufferSource2, and a MediaStreamAudioDestinationNode streamDestination:
bufferSource1.connect(streamDestination);
//do some other things here, and after some time, switch to bufferSource2:
//(streamDestination doesn't need to be explicitly specified here)
bufferSource1.disconnect(streamDestination);
bufferSource2.connect(streamDestination);
Example in action.
Edit 1:
Proper implementation:
According to the Editors Draft on the Audio Output API, it is planned/will be possible to choose a custom audio output device for the AudioContext as well (by means of new AudioContext({ sinkId: requestedSinkId });). I couldn't find any info on the progress, and even found a related discussion which the asker apparently read already. According to this and (many) other references, it doesn't seem te be an easy task, but it's planned for WA V1.
Edit:
That section has been removed from the API Draft, but you can still find it in an older version.
Current workaround:
I played around with your workaround (using a MediaStreamAudioDestinationNode and Audio object), and it seems to be related to nothing being connected. I modified my example to toggle a single buffer (similar to your example but with an AudioBufferSourceNode), and observed a similar frequency drop. However, when using a GainNode inbetween and setting it's gain.value to either 0 or 1, the frequency drops disappeared (this isn't gonna be the solution if you want to create and connect new AudioBuffers dynamically).
I was watching a youtube video and I decided to investigate some parts of its video player. I noticed that unlike most HTML5 video I have seen, Youtube's video player does not do a normal video source and instead utilizes a blob url as the source.
Previously I have tested HTML5 videos and I found that the server starts streaming the whole video from the start and buffers in the background the complete rest of the video. This means that if your video is 300 megs, all 300 megs will be downloaded. If you seek to the middle, it will start downloading from the seek position all the way to the end.
Youtube does not work this way (at least in chrome). Instead it manages to control buffering so it only buffers a certain amount while paused. It also seems to only buffer the relevant pieces, so if you skip around it will make sure not to buffer pieces that are unlikely to be watched.
In my attempts to investigate how this worked, I noticed the video src tag has a value of blob:http%3A//www.youtube.com/ee625eee-2802-49b2-a13f-eb374d551d54, which pointed me to blobs, which then led me to typed arrays. Using those two resources I am able to load a mp4 video into a blob and display it in a HTML5 video tag.
However, what I am now stuck on is how Youtube deals with the pieces. Looking at the network traffic it appears to sends requests to http://r6---sn-p5q7ynee.c.youtube.com/videoplayback which returns binary video data back in chunks of 1.1mb. It also seems worth noting that most normal requests due to HTML5 video requests seem to receive a 206 response code back while it streams, yet youtube's playvideo calls get a 200 back.
I tried to attempt to only load a range of bytes (via setting the Range http header) which unfortunately failed (I'm assuming because there was no meta-data for the video coming with the video).
At this point I'm stuck on figuring out how Youtube accomplishes this. I came up with several ideas though none of which I am completely sold on:
1) Youtube is sending down self contained video and audio chunks with each /videoplayback call. This seems like a pretty heavy burden on the upload side and it seems like it would be difficult to stitch these together to make it appear like it's one seemless video. Also, the video tag seems to think it's one full video, judging from calling $('video').duration and $('video').currentTime, which leads me to believe that the video tag thinks it's a single video file. Finally, the vidoe src tag never changes which makes me believe it is working with a singular blob and not switching out blobs.
2) Youtube constructs an empty blob pre-sized to the full video array and updates the blob with pieces as it downloads it. It would then make sure the user has not gotten too close to the last downloaded piece (to prevent the user from entering an undownloaded section of the blob). The problem that I see with this that I don't see any way to dynamically update a blob through javascript (although maybe I'm just having trouble googling for it)
3) Youtube downloads the meta data and then starts constructing the blob in order by appending the video pieces as it downloads them. The problem I see with this method is I don't understand how it would handle seeks in post-buffered territory.
Maybe I"m just missing an obvious answer that's right in front of me. Anyone have any ideas?
edit: I just thought of a fourth option. Another idea is they might use the file API to write the binary chunks to a file and use that file to stream off of. The file API seems to have the ability to seek to specific positions, therefore allowing you to fill a video with empty bytes and fill them in as they are received. This would definitely accommodate video seeking as well.
Okay, so few things you need to know is that YouTube is based on this great open source Project. It behaves different for every browser and if your browser supports more intensive decoding like WEBM it will use that to save Google's bandwidth. Also if you look at this Demo
Then you will find a section which downloads the entire video into a thing called "offline storage". I know chrome has it and some other browsers not every in some cases they do have to use the entire video source instead of a blob. So that blob is streaming depending on the user interaction with the video. Yes the video is just 1 file and they have metadata for that video like a little database that tells the time of the video and the points at which chunks can be divided in.
You can find out more by reading the Project's documentation. I really recommend you have a look at the demo.
When you look at the AppData of GoogleChrome, while playing a youtube video, you will see that it buffers in segmented files. The videos uploaded to youtube are segmented, which is why you can't perfectly pinpoint a timeframe in the first click on the bar if that timeframe is outside of the current segment.
The amount of segments depends on the length of the video, and the time from which you start and stop playing back the video.
When you are linked to a timeframe of a video, it will simply skip the buffering of the segments that come before that timeframe.
Unfortunately I don't know much about the coding for video playback, but I hope this points you in the right direction.
there is a canvas element in the page ,Maybe This Will Help
http://html5doctor.com/video-canvas-magic/
we knew the video is been segmented,the question is how to stitch them together.i think the real video element doesn't do the play work,it support the datasource,and draw the seagments each frame to the canvas element。
var v = document.getElementById('v');
var canvas = document.getElementById('c');
v.addEventListener('play', function(){
if(v.paused || v.ended) return false;
c.drawImage(v,0,0,w,h);
setTimeout(draw,20,v,c,w,h);
},false);
Youtube is using this feature only in browsers that support Media Source Extensions so it is up to the browser decide about all the rest because of this feature.
I am new to actionScript programming. I know some html and I am currently learning html5. I need to do an interactive video by putting html content in a specific time of the video. I'll be more concise:
For example, I have a video that is 5 minutes long, let's suppose that from the second 3:50 to 4:00 I need to display two boxes over the video, each one representing one choice. If at 3:50 the video shows the possibility to the viewer to select among two paths (the video told the user to select among those paths for instance) the viewer will have the possibility to select one of the paths by clicking on one of the two boxes that will appear in that time interval. I know this needs to be made with the tag and with hyperlinks.
My question is How do I tell the html5 video player to display a canvas from the minute 3:50 to the minute 4:00 in which two hyperlinks will display??
Thanks for your attention I will appreciate very much your help. I need some kind of guidance because I have been looking for many days.
For your use case it seems you want to be able to control the video flow of the user through interactions that jump to different times in the video.
Using html5 video player to seek to a different time in a video (using currentTime) you could create a click event on a box that you lay on top of the video and set the time when you click that box, using:
// Jump 30 seconds into the video
var time = '30';
var video = document.createElement('video');
video.src = "video.mp4";
// Set the time
video.currentTime = time;
video.play();
You can check out how we created an interactive video authoring tool(open source) using html5 and JS and use that.
If you don't want to spend time coding an interactive video you should check out H5Ps authoring tool through this simple example. You can test out creating your own at H5P.org as well. The tool is completely free.
I may be wrong, but I believe that you mean javascript instead of actionscript. If that is the case then I would definitely check this out Video.JS.
When you reach the current time you trigger your method/function which adds whatever you want on top of the video.
var whereYouAt = myPlayer.currentTime();
However, if you DO mean actionscript then you are working with a flash player. Therefore I suggest you take a look at this Vimeo Player
currentTime:Number [read-only] Returns the current playback time of the video.
I'm on an AS3 project, playing a video (H264). I want, for some special reasons, to go to a certain position.
a) I try it with NetStream.seek(). There it only goes to keyframes. In my current setting, this means, i can find a position every 1 second. (for a better resolution, i'd have to encode the movie with as many keyframes as possible, aka every frame a keyframe)
this is definetly not my favourite way, because I don't want to reencode all the vids.
b) I try it with NetStream.step(). This should give me the opportunity to step slowly from frame to frame. But in the documentation it says:
This method is available only when data is streaming from Flash Media Server 3.5.3 or higher and when NetStream.inBufferSeek is true.
http://help.adobe.com/en_US/FlashPlatform/reference/actionscript/3/flash/net/NetStream.html#step()
Does this mean, it is not possible with Air for Desktop? When I try it, nothing works.
Any suggestions, how to solve this problem?
Greetings & Thank you!
Nicolas
Flash video can only be advanced by seconds unless you have Flash Media Server hosting your video. Technically, that means that you can have it working as intended in Air, however, the video would have to be streaming (silly adobe...).
You have two options:
1) Import the footage as a movieclip. The Flash IDE has a wizard for this, and if you're developing exclusively in non-FlashIDE environment, you can convert and export as an external asset such as an SWF or SWC. This would then be embedded or runtime loaded into your app giving you access to the per-frame steppable methods of MovieClip. This, however, does come with some audio syncing issues (iirc). Also, scrubbing backwards is not an MC's forté.
2) Write your own video object that loads an image sequence and displays each frame in order. You'd have to setup your own audio syncing abilities, but it might be the most direct solution apart from FLVComponent or NetStream.
I've noticed that flash player 9 scrubs nice and smooth but in players 10+ I get this no scrub problem.
My fix, was to limit frequency the calls to the seek function to <= 200ms. This fixed scrubbing but is much less smooth as player 9. Perhaps because of the "Flash video can only be advanced by seconds" limitation? I used a timer to tigger the function that calls seek() for the video.
private var scrubInterval:Timer = new Timer(200);
private function videoScrubberTouch():void {
_ns.pause();
var bounds:Rectangle = new Rectangle(0,0,340,0);
scrubInterval.addEventListener(TimerEvent.TIMER, scrubTimeline);
scrubInterval.start();
videoThumb.startDrag(false, bounds);
}
private function scrubTimeline(e:TimerEvent):void {
var amt:Number = Math.floor((videoThumb.x / 340) * duration);
trace("SCRUB duration: "+duration+" videoThumb.x: "+videoThumb.x+" amt "+amt);
_ns.seek(amt);
}
Please check this Demo link (or get the SWF file to test outside of browser via desktop Flash Player).
Note: Demo requires FLV with H.264 video codec and AAC or MP3 audio codec.
The source code for that is here: Github link
In the above demo there is (bytes-based) seeking and frame by frame stepping. The functions you want to study mainly are:
Append_SEEK ( position amount ) - This will got to the specified position in bytes and search for the nearest available keyframe.
get_frame_TAG - This will extract a tag holding one frame of data. Audio can be in frames too but lets assume you have video-only. That function is your opportunity to adjust timestamps. When it's run it will also append the tag (so each "get_frame_TAG" is also a "frame step").
For example : You have a 25fps video, you want the third-frame at 4 seconds into playback...
1000 milisecs / 25 fps = 40 units for each timestamp. So 4000 ms == 4 secs + add the 40 x 3rd frame == an expected timestamp of 4120.
So getting that frame means... First find a keyframe. Then step through each frame checking the timestamps that represent a frame you want. If it isnt then change it to the same as most recent keyframe timestamp (this forces Flash to fast-forward through the frames to keep things in sync as it assumes the frame [with smaller than expected timestamp] should have been played by that time). You can "hide" the video object during this process if you don't like the look of fast-forwarding.