The input stream is as it is sent by the TV stations. Each TS packet is ~1 second long. ffmpeg version: N-45682-gacdea9e7c. HTTP streaming requires you to have a closed GOP, I think that might be your problem, I'd suggest you to try these two commands (fine tune them to your neeeds, resolution and buffer size, adapts the keyint to your frame rate): OK, I see the point.

Only 5 TS files are listed in the m3u8. I believe this is because it specifically enables infinite buffers for protocols like rtp and sdp. Usually the stereo stream is the the first in the list. x264enc sliced-threads=true tune=zerolatency speed-preset=1 ! Hey, I tried rebuild this issue with ffmpeg as streaming source. Instantly share code, notes, and snippets. You can always update your selection by clicking Cookie Preferences at the bottom of the page.

Should that be a default for sdp file-types? master_1.m3u8: To me this looks fine. videorate ! We use optional third-party analytics cookies to understand how you use GitHub.com so we can build better products. Now go to clappr's demo page and play it. By clicking “Sign up for GitHub”, you agree to our terms of service and But if clappr fails without any useful error, for me this looks like an issue with clappr. But the main question will not be answered: Why does it randomly not work. I recommend you to take a look at this example. To clarify, it enables cache and disables seeking (outside of cached regions). We use optional third-party analytics cookies to understand how you use GitHub.com so we can build better products. @leandromoreira Thanks for the hint. GitHub is home to over 50 million developers working together to host and review code, manage projects, and build software together.

The audiostream is transcoded to aac with ffmpeg defaults. This issue has been automatically marked as stale because it has not had recent activity. I can't currently install gstreamer because Debian broke everything for me.

to your account, Browser: Safari Version 12.0.1 (13606.2.104.1.2), Firefox, etc. libavformat 58.9.100 Learn more. they're used to gather information about the pages you visit and how many clicks you need to accomplish a task.

vlc behaves as above. Awesome. Learn more. There are 2 I-frames per second in the affected content, 1 s should be fine as well. You can always update your selection by clicking Cookie Preferences at the bottom of the page. ffplay just buffers infinitely (probably will eventually run out of memory in very unnice ways). We’ll occasionally send you account related emails.

I have measured approx 90 ms (3 > frames) of latency added by encoder+decoder combined. libavcodec 58.11.101 Learn more, We use analytics cookies to understand how you use our websites so we can make them better, e.g. This seems not to work. From my own testing with x264 (without > ffmpeg) I can confirm that this works. I use ffmpeg to packetize live TV content received via DVB-S2 using the HLS output. This might be related to the way you using ffmpeg, you can check here some tools to do the verification over your stream. libswresample 3.0.101 Don't know why. ffmpeg -r 30 -f lavfi -i testsrc -s 640x480 -pix_fmt yuv420p -preset ultrafast -an -vcodec libx264 -tune zerolatency -b 900k -f rtp rtp://127.0.0.1:5004, mpv Logfile: https://pastebin.com/NTVz9HCX.

One thing I know for sure is that open gops cause random "error", if it has scene detection it can change the gop point and when the player tries to swtich bitrates it might suffer cause of lack of timing sync (PTS or program-date-time) or even worst if they rely on the media-sequence. VLC play the same content.

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