问题
I have a server application which renders a 30 FPS video stream then encodes and muxes it in real-time into a WebM Byte Stream.
On the client side, an HTML5 page opens a WebSocket to the server, which starts generating the stream when connection is accepted. After the header is delivered, each subsequent WebSocket frame consists of a single WebM SimpleBlock. A keyframe occurs every 15 frames and when this happens a new Cluster is started.
The client also creates a MediaSource, and on receiving a frame from the WS, appends the content to its active buffer. The <video> starts playback immediately after the first frame is appended.
Everything works reasonably well. My only issue is that the network jitter causes the playback position to drift from the actual time after a while. My current solution is to hook into the updateend event, check the difference between the video.currentTime and the timecode on the incoming Cluster and manually update the currentTime if it falls outside an acceptable range. Unfortunately, this causes a noticeable pause and jump in the playback which is rather unpleasant.
The solution also feels a bit odd: I know exactly where the latest keyframe is, yet I have to convert it into a whole second (as per the W3C spec) before I can pass it into currentTime, where the browser presumably has to then go around and find the nearest keyframe.
My question is this: is there a way to tell the Media Element to always seek to the latest keyframe available, or keep the playback time synchronised with the system clock time?
回答1:
network jitter causes the playback position to drift
That's not your problem. If you are experiencing drop-outs in the stream, you aren't buffering enough before playback to begin with, and playback just has an appropriately sized buffer, even if a few seconds behind realtime (which is normal).
My current solution is to hook into the updateend event, check the difference between the video.currentTime and the timecode on the incoming Cluster
That's close to the correct method. I suggest you ignore the timecode of incoming cluster and instead inspect your buffered time ranges. What you've received on the WebM cluster, and what's been decoded are two different things.
Unfortunately, this causes a noticeable pause and jump in the playback which is rather unpleasant.
How else would you do it? You can either jump to realtime, or you can increase playback speed to catch up to realtime. Either way, if you want to catch up to realtime, you have to skip in time to do that.
The solution also feels a bit odd: I know exactly where the latest keyframe is
You may, but the player doesn't until that media is decoded. In any case, keyframe is irrelevant... you can seek to non-keyframe locations. The browser will decode ahead of P/B-frames as required.
I have to convert it into a whole second (as per the W3C spec) before I can pass it into currentTime
That's totally false. The currentTime is specified as a double. https://www.w3.org/TR/2011/WD-html5-20110113/video.html#dom-media-currenttime
My question is this: is there a way to tell the Media Element to always seek to the latest keyframe available, or keep the playback time synchronised with the system clock time?
It's going to play the last buffer automatically. You don't need to do anything. You're doing your job by ensuring media data lands in the buffer and setting playback as close to that as reasonable. You can always advance it forward if a network condition changes that allows you to do this, but frankly it sounds as if you just have broken code and a broken buffering strategy. Otherwise, playback would be simply smooth.
Catching up if fallen behind is not going to happen automatically, and nor should it. If the player pauses due to the buffer being drained, a buffer needs to be built back up again before playback can resume. That's the whole point of the buffer.
Furthermore, your expectation of keeping anything in-time with the system clock is not a good idea and is unreasonable. Different devices have different refresh rates, will handle video at different rates. Just hit play and let it play. If you end up being several seconds off, go ahead and set currentTime, but be very confident of what you've buffered before doing so.
来源：https://stackoverflow.com/questions/23301496/how-to-keep-a-live-mediasource-video-stream-in-sync