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.
Related
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 am making an all-client-side audio/music editor. I have created a few tones mathematically that are stored as base64 in the <audio> src-attribute. I can play DIFFERENT tones at the same time BUT, I can only play ONE instance of ONE specific tone at the same time.
For example clicking the key to play C like crazy will sound very awkward since the C that was playing gets stopped and the new C starts. I would like there to be possibility to play several C tones at the same time!
Now I guess this could be made by having by simple copying the audio element (one or more times) and make the keypress, sort of, cycle through them. For example if the first C tone is playing and the key to play C is clicked, then play the second C audio element, and so on and so forth.
That would work... but since I am using base64 in the source I would also have to have that copied.
<audio id="C1"><source src="data:audio/wav;base64,audio_data"></source></audio>
<audio id="C2"><source src="data:audio/wav;base64,audio_data"></source></audio>
If "audio_data" would be really long then the html would become humongous and also I think that the browser would not understand that both are actually the exactly same data, so it would be come very unoptimized.
So to the concrete question: Is there a way to play the same base64 data several times at the same time without the need of copying the whole src-attribute with the base64 string in it? (Since my application is all-client so I have not the ability to save the data to a sound-file on a server)
See a simple example. It might not work in other browsers than Firefox because I have not tested:
https://jsfiddle.net/tx3hpptL/
I'm currently working on an HTML5/WebGL based project that requires the use of stereo audio for locating objects.
In order to handle this I need to vary the Left/Right Stereo Pan based on the location and distance the audio source is from the user.
However, when trying to use the following code as suggested by the SoundManager2 Documentation (http://www.schillmania.com/projects/soundmanager2/doc/#sm-config) nothing happens, the audio remains balance between left/right
soundManager.setPan('soundObjectId', -100);
I've tried other forms, such as:
soundManager.createSound({
url: '/path/to/some.mp3',
autoPlay: false,
pan: -75
});
I don't know how common SoundManager2 is as Googling around doesn't seem to show that many answers, I'm happy to try other libraries if they can be recommended for playing browser audio (must support IE11)
I ended up doing a very nasty workaround for this - I created two copies of my audio file in Audacity, one with the audio panned all the way left, and one panned all the way right. And then I 'mixed' these in SM2 by reducing/increasing the volume of the appropriate 'side' audio file. It works, but isn't a particularly elegant solution
I am using Vimeo's Flash API so that I can embed and read the timecode of a video using the playProgressHandler, pause it at certain times, pop a menu, and use buttons that trigger seekTo calls. Although everything works, the timecode is inaccurate to varying degrees. Anywhere from 1-2 seconds. I can tell this because:
1) If I play my video on Vimeo and pause it at 6:03 and do the same with it embedded in Flash the visuals do not match up. Flash is lagging behind a tad.
2) I did a test using the JavaScript API. My seekTo calls were consistently accurate. To seek to the same spot using the AS3 API I had to add 1.5 seconds. But even this isn't foolproof. Sometimes it works, but sometimes it's still off.
Any ideas what would account for this inaccuracy and how I might fix this problem? Yes, I can ditch the AS3 and use the JS version, but I'd prefer to just fix what I've already built.
(I also posted this on Vimeo's forum, but I'm following their "Limited support in API Forum" post which suggests to post here)
Unfortunately, there's not much we can do to fix this other than to recommend that you use our iframe embed.
It has to do with the way that we retrieve files from our CDN. Because Flash doesn't support byterange requests, we pass a parameter that returns part of the file starting at that position. The nature of how that works means it's always going to be imprecise.
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();