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<Muimi> I want to shorten two commands into one.
<Muimi> ffmpeg -f concat -safe 0 -i mylist.txt -c copy output.mp4 AND ffmpeg -i output.mp4 output.mp3
<Muimi> anyone knows
<Muimi> i tried just changing the output filename to .mp3 but I think since I only want the audio from the inputs I need to isolate the audio stream
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<Muimi> [mp3 @ 0000025e4f3fe380] Invalid audio stream. Exactly one MP3 audio stream is required.
<Muimi> Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
<Muimi> Error initializing output stream 0:0 --
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<galad> if the audio in the .mp4 is not mp3 -c copy won't work
<galad> you will have to encode it to mp3
<Muimi> yeah well
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<Muimi> hey yyou know if i type the two commands it's super fast if i try other commands it takes forever or fails. would you help me with a command and break it down with me? :(
<Muimi> This was my attempt to encoude it: ffmpeg -f concat -safe 0 -i mylist.txt -i pipe:0 -f mp3 output.mp3
<Muimi> If I do it this way, it takes a long time: ffmpeg -f concat -safe 0 -i mylist.txt output.mp3
<Muimi> And I just thought it's strange because if I just type ffmpeg -i output.mp4 aoutput.mp3 then it's SUPER fast
<Muimi> so if i could figure out what's going on in the background by default and simulate it during the concatenation process i could skip a command
<Muimi> thank you for your help by the way. i understand if these questions go past you because they are going past me. ^^
<Muimi> or if you don't have time whatever
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<zsolt> if i want to use hardware acceleration for decoding in C, is it enough to set the `AVCodecContext`'s `hw_device_ctx` and a custom `get_format`?
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<JEEB> zsolt: doc/examples/hw_decode.c might give you an answer for that in the code base
<zsolt> i see, thank you
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<stevew> ffplay supports all that.
<stevew> or it does with a little tweaking, anyhow.
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<petaflot> hello! I want to record a documentary, wondering what encoder is best
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<JEEB> lol
<JEEB> how does that one look on AreWeCompressedYet?
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<JEEB> seems like AWCY only has VVC reference encoder (VTM) in there
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<JEEB> and the sets of clips utilized make it not really comparable with most modern runs
<JEEB> anyways, I would be very suspicious of any encoder recommendations without some AWCY-like test results
<JEEB> :P
<JEEB> also the VVC decoder is not yet merged into FFmpeg completely so even if you encode VVC you won't be able to decode it :P
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<petaflot> k. I have another issue though, can't record any audio. I'm trying to figure out what source (or sink)? I should use
<JEEB> anyways, if you are recording digital tv, just record the transmitted data?
<JEEB> for web streams, check if yt-dlp can save the content
<petaflot> JEEB: yt-dlt says 404, wget says 404, my tab is still open (and I can play the video)
<petaflot> also, I'm pretty sure they add DRM (although it was a public conference, broadcast on public TV)
<JEEB> you could see whether it has drm by finding the m3u8 or mpd file in the network view
<JEEB> network tab of browser's developer tools
<JEEB> if there's no protection related info there, there's no DRM
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<petaflot> afaik there is. but all I see is related to a json
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<petaflot> `pacmd list-sink-inputs` shows the list of applications that are playing sound on the server ; how can I request ffmpeg to record audio from a specific sink-input ?
<petaflot> ^server^pulseaudio server
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<petaflot> I tried following the doc at https://trac.ffmpeg.org/wiki/Capture/PulseAudio#Recordaudiofromanapplication but this fails with "cannot open audio device alsa_output.pci-0000_00_1b.0.analog-stereo.monitor (No such file or directory)" then I/O error on the same device. Using the number that `pactl list short sources` spits out yields the same result
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<iconoclast_hero> so if i use -acodec libopus -ar 2400 what sample rate will I *actually* get?
<iconoclast_hero> because if I ffprobe that I see: "Stream #0:0: Audio: opus, 48000 Hz, mono, fltp"
<JEEB> opus internally is 48000kHz I think
<JEEB> *48kHz
<iconoclast_hero> well, when I omitted the -ar 24000, the resultant file is 4.5% smaller.
<iconoclast_hero> i don't exactly understand how sample rate translates into file size, but it's clearly not 50% the size.
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<iconoclast_hero> so if it should be a 1:1 correlation, then the 24 kHz file should be half...and it isn't, so where'd the 4.5% go?
<iconoclast_hero> all the settings: -acodec libopus -b:a 17k -ar 24000
<iconoclast_hero> source file: 30 kb/s .opus from fre:ac.
<iconoclast_hero> i mean, I'll take 4.5% if there's not something else weird going on.
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<iconoclast_hero> that would be 15 GB across my library.
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<iconoclast_hero> oh and mediainfo reports the file from fre:ac as 24 kHz (VLC reports 48 kHz), the one without -ar 24000 as 48 kHz (VLC doesn't display codec info), and the one with it as 24 kHz (VLC doesn't show anything)
<iconoclast_hero> ffprobe -show_format says they're all 48 kHz
<iconoclast_hero> so what's happening to that 4.55%?
<another|> internally opus is always 48 kHz
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<iconoclast_hero> so how does issuing -ar 24000 cause it to lose 4.55% of its size? Just another transcode during the non-conversion?
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<Kelllman5> Client: HexChat 2.16.1 • OS: Microsoft Windows 10 Pro (x64) • CPU: Intel(R) Core(TM) i7-3770K CPU @ 3.50GHz (3.90GHz) • Memory: 8.0 GiB Total (2.8 GiB Free) • Storage: 358.0 GiB / 699.9 GiB (342.0 GiB Free) • VGA: NVIDIA GeForce GTX 1050 • Uptime: 1d 7h 14m 25s
<petaflot> erm.. still trying to capture and encode my screen. it was using a lot of CPU, so following some advice google gave me I tried using h264 codec : I recompiled ffmpeg (h264 was not in the list), tried with libx264,libx264rgb,libopenh264 and even h264_v4l2m2m for what it was worth, each time I get "Unable to find a suitable output format for 'insert_codec_name_here'"
<petaflot> ^a lot of^too much
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<petaflot> oh got it. ther was a stlae '-vf' laying around
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<tm512> petaflot: you might consider leveraging a hardware encoder if CPU use is an issue, though H.264 isn't too expensive in software, compared to newer codecs
<tm512> though x264 is apparently almost always better than hardware encoders, with regards to quality, unless you compare against x264's fastest presets
<tm512> so it's a tradeoff
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<tm512> with newer codecs like VP9, HEVC, or AV1, with their better compression, it's possible that just using a hardware encoder for one of those would still produce better results at a given bitrate than even x264's slower presets, maybe worth checking
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