I have a very fast connection and it takes about 2-3 seconds before the song actually starts playing. It's a relatively average 128kbps MP3 size (3mb-4mb). I have set preload="auto" but that didn't help much. Is there a way to just start playing the audio right away and continue to buffer it (sort of like YouTube does)?
Here is an example that I am currently working on. It's going to play an audio simultaneously on all connected clients. So if you have 2+ laptops, you can try it out. All computers must be connected before you start playing the audio. (double click on a song to start playing).
Running video and audio without complete buffering is called smooth / adaptive streaming. It can be achieved in players like silverlight and flash.
What it actually does is to create chunks of files and let the user play file chunk by chunk. Since you are downloading chunks, it will not require whole file to download.
Well, I am not giving you the full fledged answer since I haven't studied much but I am giving you the exact idea of how it works.
I had same issue but with HTML5 Video.. I overcame it with using Smooth streaming media Azure..
Here is a tutorial of the same : http://www.wrapcode.com/featured/windows-azure-media-services-mp4-to-smooth-streaming/
I will keep you updated once I find something useful :-)
If you use preload=none, then you have no buffer at the beginning but it will buffer your content "on the fly"
I have an Icecast server which streamsmy contet, and when I use pause and play, it buffers my content, even with preload=none.
Do not use preload=auto. It will take some time to start.
Related
I'm trying to ensure (to the extent possible) that an HTML5 video begins playing only only when it is able to play through completely without buffering. For context, the MediaStream of the video is then used to mix with another audio source and sent over peer WebRTC connections. The videos are typically 5-10MB and a few minutes long (i.e. a decent broadband connection should have no trouble loading the entire video well before it's done playing).
To achieve this, my code currently waits for the canplaythrough event on the video element to begin and calls play() when it fires.
This "works" in the sense that the video begins playing and, in most cases, buffering is sufficient for the video to play through uninterrupted. But, in a few cases (specifically for two people so far that happen to both have been running Chrome on MacBook Airs and with apparently not incredible but decent broadband Internet connections) the video plays staggered and choppy---which I believe to mean the video has not sufficiently buffered.
Are there better techniques for either ensuring that video is sufficiently buffered on most browsers?
Would using fetch() to buffer the entire video in memory probably do the trick? Or is a resulting blob() also actually lazily buffered behind the scenes?
Are there good practices for testing and debugging these sorts of issues given that I can't really replicate this locally?
I have built a tool called Stream or Not that might help on the network side. It will tell you how long the video takes to start, how many stalls, etc. You can use your browser's devTools to throttle the network (and in Chrome, you can emulate the CPU).
Honestly - to see if the network is the issue, as long as the bitrate (use FFprobe https://www.streamclarity.com/probe?url=) is lower than the network speed, you are not network constrained.
There is another possibility. What are the dimensions of your video? How what are the dimensions of the viewport on the browser? If you are asking the device to chop down a lot of pixels - the playback limit might move away from bandwidth to CPU processing speed. I have seen this happen on mobile devices and on older Macs trying to play 4k videos - there just isn't enough CPU to process that many pixels.
I'd test the network speeds, just to be sure.
Make sure you are not sending more pixels than you need. Underpowered devices will have issues.
I have videos on my website with duration of almost 2h (=> large file size).
I have managed to convert them to h265 to reduce server load. To further reduce server load I also want the video to only load e.g. the next 10min from the point the user currently is at in the video (and not the entire video). Youtube is doing it this way.
The HTML preload attribute does not have this option. Is there such a feature in ffmpeg (or anywhere else)?
Thanks for a hint
You need to work with "Dynamic Adaptive Streaming over HTTP". It could be implemented via javascript.
I suggest looking into Dash.js github page
I'd like to know how YouTube plays long-form videos so quickly, with seeking, on mobile.
This is an example video: https://www.youtube.com/watch?v=eyU3bRy2x44
I can load it just fine on mobile within 5-15 seconds and I can even seek through it.
Are they using HLS? Or are they using any other streaming technology? Are they using MP4 with highly optimized MOOV Atoms placed at the front of the files?
I'd like to know because I want to serve up long-form videos on one of my websites, and they take forever to load even if served from a CDN.
Thank you in advance!
Your videos should not really take a long time to load even with 'normal' HTTP streaming if the CDN is doing its job properly.
One possible problem might be the quality/bit rate of your videos - if they are only available in high quality or high bit rate then this will definitely cause a delay in initial playback.
Many (most?) YouTube videos now will support multiple bit rates, which allows the client device select the bit rate that is most appropriate for the current network conditions. This technique is called adaptive bit rate streaming, as you likely are aware given the reference to HLS above.
MPEG DASH, as Aquary mentions, is an adaptive bit rate streaming format. It is designed to be an open standard - Apple's HLS, Microsofts's Smooth streaming and Adobe Dynamic Streaming are the other main adaptive bit rate formats.
For videos that support adaptive bit rate streaming the client will usually start up at a low or medium bit rate to ensure quick start up and then 'step up' to the highest bit rate the network will support once the video is playing. This helps fast startup. When you jump to the middle of a video the same approach is used to 'start' again from the point you have selected.
You can quite often see this if you look closely at a video when it starts up - the playback quality will improve after a short while as the video steps up through the bit rates to higher quality streams.
YouTube uses MPEG-DASH in HTML5 on the devices that are capable of that. This allows seeking through the media and start from the moment which you select.
Traditional progressive download (AKA pseudo-streaming) is not a good option in case of long videos because by default, media players try to download entire video even though you may stop the playback. Seeking is also supported in PD but your video should be prepared for that and your media server needs to be able to process seek requests properly.
I am currently using the HTML 5 audio tag in one of my projects. I load an MP3 file from an external source and then initiate it like that:
$("#audioPlayer").attr("src",audioStreamURL);
document.getElementById("audioPlayer").play()
This works fine, but I noticed that the Audio-Tag takes a long time to buffer before playing the file (Chrome and Safari), which causes a delay of a couple of seconds and thus reduces the user experience significantly.
Checking the console in Chrome, I noticed that the audio most times starts once ca. 5 MB have been "transferred".
I also checked if it is due to the server's latency and loaded the audio file in VLC-Player. However, here it started right away, with no delay at all.
Does anybody know, why Chrome does that? And even more important: Does anybody know a solution for that problem? A workaround or audio-tag alternative?
I would really appreciate your help!
Have a look at the "canplaythrough"-event of HTML5 Audio. This event fires when the Audio is able to play without buffering. You could bind a listener to that event, that will play the Audio.
Maybe this will be faster than just playing the Audio, which will wait for the complete file to be downloaded.
There I demonstrated how to preload an HTML5 Audio: preloading the next song in a playlist a bit before the current one ends
Check out this space shooter demo.
The HTML5 audio is perfect on Chrome 18 and Firefox 10. There is no lag in playing sounds and each sample plays perfectly. The last time I tried to play sounds using HTML5 audio and JavaScript I couldn't get a sound to play more than once.
What sorcery is Scirra doing to make this so perfect?
I'm the developer of Construct 2, so I hope I'm sufficiently qualified to answer your question :)
HTML5 audio is indeed a mess, so I've gone to considerable lengths to try and make it bulletproof in Construct 2. Here's an outline of what I've done:
Use the Web Audio API
HTML5 audio appears designed for streaming music, so a HTML5 Audio object is kind of a heavyweight object. Playing 10 sounds a second with it like Space Blaster does can easily seize up the browser. On the other hand, the Web Audio API is a high-performance audio engine with routing, effects, and lightweight sound playback. It's perfect for games. Audio buffers and audio playback are separated, so you can have one data buffer and efficiently play it many times simultaneously, whereas some browsers are so buggy if you play a HTML5 sound a few times it re-downloads it each time! Since it was actually designed for games and such, you can happily play back tonnes of sound for ages and it will still hum along nicely. It can also use HTML5 audio as a sound source, although I only use HTML5 audio for things the user has designated as music tracks (since that's where you'd prefer to have streaming - typically everything else in the Web Audio API is fully downloaded before playing).
The Web Audio API is supported in Chrome, has also made it in to iOS 6+ (although it's muted until you try to play some audio in a touch event), Firefox are working on support, and it should be coming soon to Chrome for Android. So on these platforms audio will be significantly more reliable.
More info on HTML5Rocks and the proposed spec - you'll have to use the spec as the documentation for now, there's not much else out there.
Other browsers: implement an audio recycling system
The Web Audio API isn't yet supported everywhere, notably IE, which means you still need to crowbar HTML5 audio in to something that might work for games for backwards compatibility. The way to do this is to recycle audio objects.
The player's laser in Space Blaster fires 10 times a second - and that's not including any other sound effects! As I mentioned earlier, Audio is kind of a heavyweight object, so if you're doing new Audio() 10+ times a second, lo and behold, the browser eventually dies and audio starts glitching up. However, you can drastically reduce the number of Audio objects created by recycling them.
Basically, for each sound effect, keep a cache of every Audio object you've created with that sound as a source. Then, when playing a new sound, search the cache for any sound effects which have finished playing (the ended property will be true). If you find one, rewind it back to the beginning (currentTime = 0) and play() again. Otherwise, create a new Audio() object in the cache.
Since the player's laser sound effect is short, instead of creating 600 Audio objects a minute, there will just be 3 or 4 that it keeps cycling round. Some browsers unfortunately will still download it 4 times (Safari did this last I checked!) or have high latency the first time each sound buffer is played, but eventually the browser catches up since the same buffers are always being reused. So basically sound might be a bit weird for a few moments, then it clears up. We also use the HTML5 app cache so next time you play everything loads from disk, so subsequent plays should perform well immediately.
That's basically it. It's still a little dodgy on the first play with HTML5 audio, but every time after that should be fairly solid providing the browser has a sane audio implementation. There are a number of ways to try to clone Audio objects, but I've found that rewinding existing Audios works best.
There's no SoundManager or any Flash/plugin-based fallbacks at all since we make a point of being pure HTML5.
We also support audio APIs provided by PhoneGap and appMobi for mobile, since HTML5 audio on mobile isn't even worth trying. That makes a total of four audio APIs our audio engine wraps, and yes, it does look like a frankenstein mess, but it works.
That's it. I suppose our competitors will read this, but who cares when there's SO rep to be had???!!!1111