Buffering video for playback on poor connections

[sanek11591]

I have a device on openipc, it works via a cellular network, where the connection is extremely bad, I would like to accumulate a buffer for about a minute (selected by the user), then while this buffer is being read, load the video further (all this in the browser since go2rtc is installed on the device itself), so that the user can watch the video, albeit with a delay, but smoothly.

Is it possible to implement this for the MSE stream? For the HLS stream , I think this should be possible by increasing the number of segments, but why doesn't it work (h264 + aac)

[AlexxIT]

You can do this by writing custom MSE viewer. Current viewer has 5 seconds buffer and 1 second delay: https://github.com/AlexxIT/go2rtc/blob/f45fef29d841e052e5625792b49581ec67b59f3e/www/video-rtc.js#L452-L466

[sanek11591]

You can do this by writing custom MSE viewer. Current viewer has 5 seconds buffer and 1 second delay:

go2rtc/www/video-rtc.js

Lines 452 to 466 in f45fef2

if (!sb.updating && sb.buffered && sb.buffered.length) { const end = sb.buffered.end(sb.buffered.length - 1); const start = end - 5; const start0 = sb.buffered.start(0); if (start > start0) { sb.remove(start0, start); ms.setLiveSeekableRange(start, end); } if (this.video.currentTime < start) { this.video.currentTime = start; } const gap = end - this.video.currentTime; this.video.playbackRate = gap > 0.1 ? gap : 0.1; // console.debug('VideoRTC.buffered', gap, this.video.playbackRate, this.video.readyState); }

thank you i will try )

[b0ch3nski]

In case anyone stumbles across this issue, here is what has made my cameras behind a weak LTE connection much more pleasant to view:

                if (!sb.updating && sb.buffered && sb.buffered.length) {
                    const end = sb.buffered.end(sb.buffered.length - 1);
-                    const start = end - 5;
+                    const start = end - 20;
                    const start0 = sb.buffered.start(0);
                    if (start > start0) {
                        sb.remove(start0, start);
                        ms.setLiveSeekableRange(start, end);
                    }
                    if (this.video.currentTime < start) {
                        this.video.currentTime = start;
                    }
                    const gap = end - this.video.currentTime;
-                    this.video.playbackRate = gap > 0.1 ? gap : 0.1;
+                    this.video.playbackRate = gap > 2.0 ? 1.2 : 1.0;
                    // console.debug('VideoRTC.buffered', gap, this.video.playbackRate, this.video.readyState);
                }

[AlexxIT]

I think it needs to remove this logic or completely rewrite it.