I recently tried out YT SHORTS. And the first ever short video I posted it would start rapidly growing then just die off in a way... wondering if there is any analytics that might indicate why this is so or maybe other things I might not be counting?
Falling off
First hour
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Hello Friends,
I am working on OTT platform app, I need to play video very smoothly without any delay like Snapchat and instagram as reference. I am using Cloudinary for uploading videos and everything is working good but at first time, AVPlayer takes time of 1-2 second to start video, which is bad thing for Me. Once video play, next time I come on same video it plays smoothly with less delay of max Half second.
As far as I tried to learn through different blogs and stack over flow answers, I get rid this is default AVPlayer Buffering time and it depends of video durations and its fetching video information like title, metadata etc. But I don't have to use these information anywhere.
I tried to set false this property of AVPlayer .automaticallyWaitsToMinimizeStalling = false, but still no luck.
I tried few solutions from StackOverflow posts, but didn't get success
How to reduce iOS AVPlayer start delay
This is demo video Link Which you can try http://res.cloudinary.com/dtzhnffrp/video/upload/v1621013678/1on1/bgasthklqvixukocv6xy.mov
If you can suggest, what I can use for OTT platforms to play video smoothly really grateful to everyone...
Thanks In Advance
Most streaming services use ABR, which creates multiple resolution copies of the video and beaks each into 2-10 second, typicaLLY, chunks.
One benefit of ABR is that to speed up playback start up, the video can start on a lower resolution bit rate and then 'step up' to higher bit rates as it proceeds.
You can often see this on popular streaming services where you will see the video quality is lower when the video starts and improves after a short time.
See here for more on ABRs: https://stackoverflow.com/a/42365034/334402
This requires you to do work on the server side to prepare the video for HLS and DASH streaming, the two most common ABR streaming protocols.
Typically dedicated streaming servers, or a combination of encoders and packagers, are use to prepare and serve the ABR streams. There are also cloud services, for example AWS Media Services or Azure Media Services, which allow on demand streaming models.
You can make the videos smaller either by reducing the dimensions or by compressing it more. Both of these have the effect of lowering startup time - but will sacrifice quality in exchange.
Cloudinary will create ABR versions for you, but the last I checked, you pay for each version created.
I'm trying to develop a rhythm game in Monogame. The notes are frame rate independent and work consistently over different computers, but the song playing does not. Right now I have it hooked up so the notes spawn and MediaPlayer.Play() do not start until I hit a key (so it's outside of loadcontent/initialize like I've seen other solutions suggest). The song does not start playing until I hit the "I" key (gets checked in Update). However, the song seems to start playing at different times across different devices. I pull down the project on both my laptop (on battery) and desktop, and hit the "I" key at the same time on both, and I get different results on my laptop. Sometimes it'll be seconds ahead of the desktop, other times it'll be a second or two behind (this is usually the case). It seems pretty consistent on desktop. I'd try more than a dozen times and everything is still synced, but not on my laptop. I'm worried this is something that works only on my PC but not on others.
I've tried enabling fixedTimeStep and setting FPS to 60 (and setting the TargetElapsedTime to the window title ensuring that they are indeed both hitting 60) and I still get inconsistent results. I've read from other answers about using PlayPosition from the MediaPlayer class, but that property is readonly so I'm not quite how I could use that.
At first, I thought my notes were simply dependent on frame rate, but after setting FPS and fixedTimeStep, and also holding my laptop up to my desktop monitor they match up perfectly. It's just the song not syncing up. I'm still a newbie to XNA but I couldn't find anyone else reporting different song starting timings like this. Any help would be greatly appreciated.
I thought of a solution the second I got out of bed the next morning. I figured there must be some overhead on MediaPlayer.Play(), but I wondered if I could start it, pause it, then resume instead. Turns out, there is a Pause() and Resume() method which is what what missing from making this work.
So! If anyone else out there is having similar issues, here's what I did. I kept a key to activate MediaPlayer.Play(), but immediately added MediaPlayer.Pause() after. Then I got another key to resume the song, and the notes spawned when the song was resumed. So pressing "I" played the song and paused it immediately. Pressing "R" resumed the song and started spawning the notes.
My iOS app uses AVPlayer to decode H.264 videos with AAC audio tracks out of local device storage. Content with bit rate spikes cause audio to drop shortly (less than a second) after the spike is played, yet video playback continues normally. Playing the videos through Safari seems to work fine, and this behavior is repeatable on several models of iPhones ranging from 6s through 8 plus.
I've been looking for any messages generated, delegates called with error information, or interesting KVOs, but there's been no helpful information so far. What might I do to get some sort of more detailed information that can point me in the right direction?
Turned out that the AVPlayer was configured to utilize methods for loading data in a custom way. The implementation of these methods failed to follow the pattern of satisfying the requests completely. (Apple docs are a vague about this.) The video portion of the AVPlayer asked for more data repeatedly, so eventually all its data got pulled. However, the audio portion patiently waited for the data to come in because there were neither an error state reported nor was all the data provided -- the presumption being that it was pending.
So, in short, sounds like there's provisions in the video handling code to treat missing data as a stall of some form and to plow onward, whereas audio doesn't have that feature. Not a bad design -- if audio cuts out it's very noticeable, and it's also by far the smaller stream so it's much less likely.
Despite spending quite a few days on the problem before posting, the lack of any useful signals made it hard to chase down the problem. I eventually reasoned that if there's no error in producing output from the stream, the problem must be in the delivery of the stream, and the problem revealed itself once I started tweaking the data loading code.
I need to playback short audio samples with precise timing, including up to 4 sounds starting simultaneously.
These sound samples are triggered with NSTimers (alternatively, I've also tried dispatch_after).
I've tried with AVPlayer and AVAudioPlayer but they are just not precise enough in timing.
Multiple sounds played at once will be all over the place, especially on the real device.
I've read about NSTimer allowing up to a few 100 milliseconds deviation, which is just too much for me.
As a test I've setup a few AVAudioPlayers with one audio sample each and triggered them all at the same time in didSelectRow...() but they will not sound in exactly the same moment, even with no NSTimer involved.
It seems that it's just not possible to playback 2 sounds starting exactly at the same time with AVAudioPlayer. Is this confirmed?
From what I've gathered there are not many alternatives, Audio Queue Service being one that allows precise timing and multiple sounds at once.
However, it's written in C, which I've never worked with, and it is hard to find any examples showing how to integrate this for simple audio playback of a sound (I'm using Swift). I'd basically just need to know how to integrate Audio Queue Services to playback a simple sound.
If someone can point me in the right direction (or knows a better solution to what I'm looking for), that would be much appreciated.
I have access to a proxy server and I can find out the time a video was requested. The log has the form (time, IP, URL). I want somehow figure out for how many seconds did a particular user using IP address A watched a YouTube video. Any suggestions?
If you only have access to requests, you obviously can't tell the difference if someone just loaded a video or watched it.
So, the best you can do is to come up with a set of heuristics that tries to 'guess' it by observing certain actions of the user. Here are a few ideas:
Does you log count the requests for the video buffer itself? If it does, you can see how much of the video was actually loaded, and the watched time can't be more than that.
If you (quite naively, I guess) assume that they're finished watching when they request another video URL, you can use this as your trigger for ending a 'video session'.
Install Wireshark or similar and start watching activity from YouTube during the video. Can you identify if there's a request when advertising is shown, or the related videos are displayed when the video finishes?
In all honesty, though, I think it will be virtually impossible trying to derive such an specific metric like seconds watched from such limited data as the point in time a video was requested. Just think of what could mess up any strategy you come up with: the user could load several videos in different tabs in a burst, or he could load a video page, pause it and forget it for several minutes or hours before he does watch it.
In short: I don't think you'll get a reliable guess using only the data you have, but if you absolutely must at least try, observing network activity between client and YouTube that only happens when a video is in the 'playing state' (pulling advertisings, related videos, some sort of internal YouTube logging, etc) is probably your best bet. Even that probably won't have a granularity nearly close to seconds, though.