), ffmpeg -r 3 -i -preset ultrafast -an -vcodec libx264 -tune zerolatency -b 900k -f rtp rtp://127.0.0.1:5004. But if clappr fails without any useful error, for me this looks like an issue with clappr. Randomly the stream is not played and hls:4 error is raised. @REELcoder this might point to a problem at your ffmpeg command line to generate hls. You can always update your selection by clicking Cookie Preferences at the bottom of the page. Thanks for looking into that. ... v baseline -pix_fmt yuv420p -preset ultrafast -tune zerolatency -crf 28 \-acodec aac \ rtmp://streaming-server/live/key: Sign up for free to join this conversation on GitHub. https://hackersgarden.dyn.ch/media/master_1.m3u8, https://gist.github.com/use-sparingly/7041ee993adb5c911f90#file-iframe-probe-py. So I can't reproduce anything mpv does specifically wrong. VLC play the same content. they're used to log you in. video/x-raw,framerate=3/1 ! Can I provide more information to get this fixed? It behaves quite similar although ffplay stream playback is not always flawless anymore... You can try this ffmpeg line to create the stream: The manifests created by ffmpeg with above command are as follows: Video is h264, Audio is ac3 and transcoded to aac. did you double check to see if you are reliable for, can you increase your hls_time? VLC plays it fine. This is violated in the affected example. This might be related to the way you using ffmpeg, you can check here some tools to do the verification over your stream. It seems like it has something todo with timing. Other framerates behave quiet similar (<25 fps). The TV stations send multiple audio streams. I use ffmpeg to packetize live TV content received via DVB-S2 using the HLS output. Learn more, hls:4 error from ffmpeg generated content. (hls_time, like for a 2s chunk you need to have either 1s or 2s gop). To clarify, it enables cache and disables seeking (outside of cached regions). does your audio framerate (frequency) is equal across all the renditions? We use optional third-party analytics cookies to understand how you use GitHub.com so we can build better products. You can always update your selection by clicking Cookie Preferences at the bottom of the page. libswresample 3.0.101 ffmpeg library versions: they're used to log you in. Millions of developers and companies build, ship, and maintain their software on GitHub — the largest and most advanced development platform in the world. From my own testing with x264 (without > ffmpeg) I can confirm that this works. they're used to log you in. Didn't work well in any of mpv/vlc/ffplay. For mpv playback the behavior is like with the gstreamer src: Playback breaks after a few seconds for both ffmpeg stream examples and then playback stop completely. they're used to gather information about the pages you visit and how many clicks you need to accomplish a task. ffmpeg -f avfoundation -i "1" -vcodec libx264 -r 10 -pix_fmt uyvy422 -tune zerolatency -b:v 500k -bufsize 300k -f mpegts udp://192.168.88.38:1234 Windows ffmpeg -f dshow -i video="screen-capture-recorder" -r 10 -vcodec libx264 -preset ultrafast -tune zerolatency -crf 18 -b:v 500k -bufsize 300k -f mpegts udp://172.31.66.20:1234 But anyway, your comment about stream and demuxer cache got me thinking, I tried earlier disabling the cache but now I got the feeling mpv isn't using a cache at all. We use optional third-party analytics cookies to understand how you use GitHub.com so we can build better products. I had issues both with vlc and ffplay when using the testsrc. It worked here =) joking aside, this command I gave you do not transcode audio to aac maybe that might be your problem, the audio transcoding not the video. Don't know why. For more information, see our Privacy Statement. (like 2s for segment), does your gop is a multiplier of the chunk size? privacy statement. libavutil 56.7.101 You signed in with another tab or window. ffplay just buffers infinitely (probably will eventually run out of memory in very unnice ways). Have a question about this project? The input stream is as it is sent by the TV stations. For more information, see our Privacy Statement. We use essential cookies to perform essential website functions, e.g. So I think the main issue is, playing a stream via sdp file does not enable the cache. So I added --cache=yes to my mpv cmd line and voila mpv plays flawlessly, even my initial gstreamer stream with low framerate. x264enc sliced-threads=true tune=zerolatency speed-preset=1 ! libswscale 5.0.101 I can tell you that I use ffmpeg for my tests and it works just fine! But a real movie src worked for both (First ffmpeg cmdline). Just tested your commit 1420013 and I can confirm the cache is now enabled and the playback of the stream works well. I have measured approx 90 ms (3 > frames) of latency added by encoder+decoder combined. videorate ! There is only 1 quality level as the videostream is not transcoded. I had issues both with vlc and ffplay when using the testsrc. We use optional third-party analytics cookies to understand how you use GitHub.com so we can build better products. ffplay plays it fine: The audiostream is transcoded to aac with ffmpeg defaults. Have a question about this project? libavcodec 58.11.101 I believe this is because it specifically enables infinite buffers for protocols like rtp and sdp. > > Very low latency can be achieved when using the x264 encoder by using > settings like tune zerolatency. April 20th, 2018, FFmpeg 4.0 "Wu" FFmpeg 4.0 "Wu", a new major release, is now available! In mpv/ffplay it drops packets and doesn't get to play it properly. For my application transcoding is not an option. Your example is for VOD content which is statically held on the server. Successfully merging a pull request may close this issue. rtph264pay ! Logfile: https://pastebin.com/FQiZMYYK, And I did a test with a regular framerate of 30 fps and mpv playback breaks there too: ;/. Or ffmpeg does not properly transcode the 5.1 stream. libavfilter 7.12.100 Still the same error on Safari and Firefox. You can always update your selection by clicking Cookie Preferences at the bottom of the page. By clicking “Sign up for GitHub”, you agree to our terms of service and ffmpeg -r 3 -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. libavformat 58.9.100 ffmpeg version: N-45682-gacdea9e7c. For more information, see our Privacy Statement. Learn more. By clicking “Sign up for GitHub”, you agree to our terms of service and If we know what "something" is, I'm sure this would be fixed fast in ffmpeg. The following command is used. We use optional third-party analytics cookies to understand how you use GitHub.com so we can build better products. Learn more, FFMPEG command for generating a "bars and tone" RTMP live stream. This issue has been automatically marked as stale because it has not had recent activity. We use optional third-party analytics cookies to understand how you use GitHub.com so we can build better products. Learn more, We use analytics cookies to understand how you use our websites so we can make them better, e.g. ffmpeg -i http://localhost:8008/1 -map 0:v -map 0:a:0 -c:v copy -c:a aac -f hls -hls_time 1 -hls_start_number_source datetime -hls_allow_cache 0 -hls_flags delete_segments -master_pl_name master_1.m3u8 ffmpeg_1_data.m3u8. Sign up for a free GitHub account to open an issue and contact its maintainers and the community. udpsink host=127.0.0.1 port=5004, Play the stream using a sdp file: The question is still why for the same TV station it works in 9 of 10 cases without trouble and a few minutes later the player does not start... @REELcoder it's hard to wonder about that, those tools I pointed to you could help but even them can't guarantee that you what's the problem. Don't know why. Only 5 TS files are listed in the m3u8. Hey, I tried rebuild this issue with ffmpeg as streaming source. The video stream is h264 720p and is only copied by ffmpeg. I checked GOP using https://gist.github.com/use-sparingly/7041ee993adb5c911f90#file-iframe-probe-py. I retested this with mpv 0.27.2 and ffmpeg 3.4.2 and it also breaks. GitHub is home to over 50 million developers working together to host and review code, manage projects, and build software together. Awesome. Usually the stereo stream is the the first in the list. We use essential cookies to perform essential website functions, e.g. But the log looks the same as without --rtsp-transport=lavf. Millions of developers and companies build, ship, and maintain their software on GitHub — the largest and most advanced development platform in the world. mpv --no-config --log-file=output.txt --demuxer-seekable-cache=no --cache=no test.sdp, Plays the stream with the given framerate. Detecting whether some FFmpeg thing (avio stream or demuxer) works over network isn't easy. rather simple sdp file: http://sprunge.us/HLHh, mpv --no-config --log-file=output.txt test.sdp, I tried disabling caching but the behaviour is the same: Other players, e.g. There are 2 I-frames per second in the affected content, 1 s should be fine as well. Sign in Some of the highlights: Bitstream filters for editing metadata in H.264, HEVC and MPEG-2 streams; Experimental MagicYUV encoder; TiVo ty/ty+ demuxer; Intel QSV-accelerated MJPEG encoding As per Apple HLS requirement 2.3 one stereo AAC stream must be provided. But a real movie src worked for both (First ffmpeg cmdline). ffplay -loglevel debug -i -protocol_whitelist rtp,file,udp test.sdp, Decoding breaks after a few seconds with decoding errors, artefacts and choppy playback. 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. Sign up for a free GitHub account to open an issue and contact its maintainers and the community. Playing a stream with 30fps is fine. Learn more, Decoding errors when playing rtp streams with low framerates. ffmpeg -y -rtbufsize 1500M -f gdigrab -framerate 30000/1001 -draw_mouse 0 -offset_x 224 -offset_y 232 -video_size 1280x720 -i desktop -c:v libx264 -r 30000/1001 -an -tune zerolatency -preset ultrafast -crf 0 Capture.mp4 and then In rare cases the 5.1 stream is the first and selected by the ffmpeg option -map 0:a:0.
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