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    Par défaut trois menus sont créés automatiquement à l’initialisation du site : Le menu principal ; Identifiant : barrenav ; Ce menu s’insère en général en haut de la page après le bloc d’entête, son identifiant le rend compatible avec les squelettes basés sur Zpip ; (...)

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    Un logiciel n’est malheureusement jamais parfait...
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Sur d’autres sites (8883)

  • FFMPEG stream video to Youtube Live

    13 juin 2022, par BlessedHIT

    I have a mov file and I'm using ffmpeg to stream it to youtube live using the following command,

    


    ffmpeg -re -i "episode.mov" -pix_fmt yuvj420p -x264-params keyint=48:min-keyint=48:scenecut=-1 -b:v 4500k -b:a 128k -ar 44100 -acodec aac -vcodec libx264 -preset medium -crf 28 -threads 4 -f flv "rtmp://a.rtmp.youtube.com/live2/YOUTUBE.LIVESTREAM.KEY"


    


    But im getting the following message on youtube,

    


    YouTube is not receiving enough video to maintain smooth streaming. As such, viewers will experience buffering


    


    My ffmpeg output showed my bitrate being between 800 - 1000 mbps, way lower than what i have specified in my ffmpeg command.

    


    I am using a not so powerful virtual machine, and so i thought this might be why i am not getting the desired bitrate.

    


    To overcome my hardware limitations, I then decided to encode the file for streaming using this command :

    


    ffmpeg -i episode.mov -c:v libx264 -preset medium -b:v 4500k -maxrate 4500k -bufsize 6000k -vf "scale=1280:-1,format=yuv420p" -g 50 -c:a aac -b:a 128k -ac 2 -ar 44100 episode.flv


    


    Then I stream copy the file using :

    


    ffmpeg -re -i episode.flv -c copy -f flv "rtmp://a.rtmp.youtube.com/live2/YOUTUBE.LIVESTREAM.KEY"


    


    And that seems to give me a stream that youtube is happy with.

    


    My question is, is there a way I can rewrite my ffmpeg command to livestream with the desired bitrate without needing to first encode my mov to another file or is adding more memory the only way forward here ?

    


  • Ffmpeg - How can I create HLS multiple language streams, in multiple qualities ?

    28 avril 2022, par Daniel Ellis

    Preface

    


    I'm working on converting videos from 4k to multiple qualities with multiple languages but am having issues with the multiple languages overlaying, sometimes losing quality and sometimes being out of sync. (this is less of a problem in the German audio, as this is voice over anyhow)

    


    We as a team are complete noobs in terms of Video / Audio + HLS — I'm a front end developer who has no experience of this so apologies if my question is poorly phrased

    



    


    Videos

    


    I have the video in a 4k format and have removed the original sound as I have English and German audio files that need to be overlayed. I am then taking these files and throwing them together into a .ts file like this :

    


    $ ffmpeg -i ep03-ns-4k.mp4 -i nkit-ep3-de-output.m4a -i nkit-ep3-en-output.m4a \
> -thread 0 -muxdelay 0 -y \
> -map 0:v -map 1 -map 2  -movflags +faststart -refs 1 \
> -vcodec libx264 -acodec aac -profile:v baseline -level 30 -ar 44100 -ab 64k -f mpegts out.ts 


    


    This outputs a 4k out.ts video, with both audio tracks playing.

    


    The hard part

    


    This is where I'm finding it tricky, I now need to convert this single file into multiple quality levels (480, 720, 1080, 1920) and I attempt this with the following command :

    


    ffmpeg -hide_banner -y -i out.ts \
-crf 20 -sc_threshold 0 -g 48 -keyint_min 48 -ar 48000 \
-map 0:v:0 -map 0:v:0 -map 0:v:0 -map 0:v:0 \
-c:v:0 h264 -profile:v:0 main -filter:v:0 "scale=w=848:h=480:force_original_aspect_ratio=decrease" -b:v:0 1400k -maxrate:v:0 1498k -bufsize:v:0 2100k \
-c:v:1 h264 -profile:v:1 main -filter:v:1 "scale=w=1280:h=720:force_original_aspect_ratio=decrease" -b:v:1 2800k -maxrate:v:1 2996k -bufsize:v:1 4200k \
-c:v:2 h264 -profile:v:2 main -filter:v:2 "scale=w=1920:h=1080:force_original_aspect_ratio=decrease" -b:v:2 5600k -maxrate:v:2 5992k -bufsize:v:2 8400k \
-c:v:3 h264 -profile:v:3 main -filter:v:3 "scale=w=3840:h=1920:force_original_aspect_ratio=decrease" -b:v:3 11200k -maxrate:v:3 11984k -bufsize:v:3 16800k \
-var_stream_map "v:0 v:1 v:2 v:3" \
-master_pl_name master.m3u8 \
-f hls -hls_time 4 -hls_playlist_type vod -hls_list_size 0 \
-hls_segment_filename "%v/episode-%03d.ts" "%v/episode.m3u8"


    


    This creates the required qualities, but I'm now at a loss of how this might work with the audio

    


    Audio

    


    For the audio I run this command :

    


    ffmpeg -i out.ts -threads 0 -muxdelay 0 -y -map 0:a:0 -codec copy -f segment -segment_time 4 -segment_list_size 0 -segment_list audio-de/audio-de.m3u8 -segment_format mpegts audio-de/audio-de_%d.aac
ffmpeg -i out.ts -threads 0 -muxdelay 0 -y -map 0:a:1 -codec copy -f segment -segment_time 4 -segment_list_size 0 -segment_list audio-en/audio-en.m3u8 -segment_format mpegts audio-en/audio-en_%d.aac



    


    This creates the required audio segments.

    


    The question

    


    I realise this is quite an ask, but is there anything wrong with our inputs ? Is there a way that this can be done a bit more streamlined ?

    


    Any answers are greatly appreciated.

    


  • Convert DTS to AC3 but only if there is no AC3 track already present in container

    24 mars, par Domagoj

    I have sound system that does not support DTS only AC3. I'm automating the process using bash that detects when movie was added to folder, downloads subtitles and converts audio track to AC3 using this command (one part of it) :

    


    ffmpeg -i "{{episode}}" -map 0:v -map 0:a:0 -map 0:a -map 0:s -c:v copy -c:a copy -c:s copy -c:a:0 ac3 -b:a:0 640k "{{directory}}"/{{episode_name}}temp2.mkv

    


    This works without issue and I end up with a .mkv file that contains original DTS audio track and newly created AC3 audio track. The issue is that some files already contain both AC3 and DTS tracks and in those cases I end up with two AC3 tracks and one DTS track. Another issue is that this command is triggered every time there is update to subtitles. So it's possible that the command will execute multiple times in a period of a few days and the container will have X number of the AC3 tracks.

    


    I need a way to detect if file already contains AC3 track before I initiate command from above, but I'm not sure what the command would be. Any help is appreciated !