I use media element to play a video online. It's okie when I played it on emulator. But when I test on lumia 520, It only plays 5-10 minutes then it has a error and back to the home screen.
How should I go about fixing it? I think that it is not enough ram memory.
Narrow down the root cause.
Try other videos and note what happens with them. Use differing lenths.
Try videos which are not streamed, but local to the phone with greater lengths.
If possible, put a try catch block around the player or within its interactions in code behind., verify its not throwing an error. If it is promote the error (at least for now) to a bound text block so you can seen the error.
Subscribe to MediaElement.MediaFailed and Media Ended event to see if that gives any indication of what is happening.
Can you fully stream the file to the phone then play it locally, then remove the file? Maybe separating that code will give a better experience.
Related
I'm using an API to get some data, but one of the JSON values, contains HTML inside of it. Is there a way for me to use the HTML tags that are inside the JSON value exactly like they are shown, instead of having to make the exact same HTML tags in my MVC view, and editing these ones out? If this wasnt descriptive enough heres an example of what i mean:
This is just one of the JSON values, I didnt wanna paste the whole thing:
"detail": "<h1>Overwatch Patch Notes – October 19, 2016</h1>\r\n\r\n<p>A new patch is now live on Windows PC. Read below to learn more about the latest changes.</p>\r\n\r\n<p>To share your feedback, please post in the General Discussion forum.<br />\r\nFor a list of known issues, visit our Bug Report forum.<br />\r\nFor troubleshooting assistance, visit our Technical Support forum.</p>\r\n\r\n<p>Please note that these changes will be rolled into a larger patch for PlayStation 4 and Xbox One at a later date.</p>\r\n\r\n<h2>BUG FIXES</h2>\r\n\r\n<p><strong>General</strong></p>\r\n\r\n<ul>\r\n\t<li>Fixed an issue causing the default Overwatch spray to override a player’s chosen spray when watching a Play of the Game or Highlight</li>\r\n\t<li>Fixed an issue causing players to frequently disconnect while viewing Highlights</li>\r\n\t<li>Fixed a bug preventing the appropriate music from playing after a loss on the Junkenstein's Revenge Brawl</li>\r\n</ul>\r\n\r\n<p><strong>Gameplay</strong></p>\r\n\r\n<ul>\r\n\t<li>Fixed a bug causing multiple issues when displaying data on the leaderboards</li>\r\n</ul>\r\n\r\n<p><strong>Heroes</strong></p>\r\n\r\n<ul>\r\n\t<li>Fixed an issue preventing Ana’s Nano Boost callouts from being heard by the enemy team</li>\r\n\t<li>Fixed a bug preventing players from receiving credit toward the Healing Done commendation when healing D.Va’s mech</li>\r\n\t<li>Fixed a graphical issue that was preventing the liquid in Mei’s Endothermic Blaster from appearing</li>\r\n\t<li>Fixed a bug causing Reinhardt’s Charge to unexpectedly stop when crossing certain thresholds (e.g. when exiting a dropship)</li>\r\n\t<li>Increased the volume of Roadhog’s “Want some candy” voice line</li>\r\n</ul>\r\n\r\n<p><strong>Map</strong></p>\r\n\r\n<ul>\r\n\t<li>Fixed a bug on Eichenwalde that caused some textures to stretch across the map for some players</li>\r\n</ul>\r\n".
You can use Html.Raw inside your view to use the raw string value:
#Html.Raw(Model.detail);
https://msdn.microsoft.com/en-us/library/gg480740(v=vs.118).aspx
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 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 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.
Here's my problem:
I have a Flash swf that uploads files from local machine and if they are images it resizes them if needed. This involves creating a JPGEncoded bytearray from a bitmapData object. After im finished with the bitmapData I dispose() of it.
I am noticing that flash will get stuck while resizing an image sometimes and have tracked this down to an "invalid bitmapData" error message. I tried last night before leaving work and it was throwing this message after 2 images!! This morning it all seemed to be fine so I decided to push it and tried uploading 20 images of 5616x3744 pixels and 5.32MB (the same images I tried previously).
I switched on performance monitor in windows and started the upload in the SWF running in mozilla + firebug for good measure. Things worked great for about 12 images then on the 13th (!!) it froze again. Ive attached a screenshot of the graph from performance mon.
Im guessing I need to do more tests like this to see where there are any problems. Can anyone shed some light on what Im seeing here that might cause problems - the yellow line looks suspicious!(?)
Large green spike corresponds to redrawing the bitmapData I think and the smaller green spike is drawing a thumbnail version from the same Bitmap object after the large version has been successfully loaded. What other counters should I use to monitor memory usage etc.
Any advice is appreciated.
many thanks
You've got too many pixels, that's all.
In Flash Player 10, the maximum number of pixels a bitmap can have si 16,777,215 (or 0xFFFFFF). Also, the maximum width or height is 8,191, as long as the total pixel count is under the maximum value.
Your test bitmap has 21,026,304, which is way over the top.