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<EveHQ> any idea in my command? :)
<durandal_1707> looks like issue with input file
<EveHQ> its a video
<EveHQ> command is ok?
<durandal_1707> i mean yoou get those messages because you decode original file and than you transcode it
<durandal_1707> and original file when decoded gives those logs
<durandal_1707> you get those errors when transcoding? or when decoding on receiver side?
<EveHQ> i guess transconding but i dont know, sorry im not expert, im trying to stream that video
<durandal_1707> so you get those messages when streaming from server or when receiving from client?
<EveHQ> when I stream from the server
<durandal_1707> because in first case, you probably cant do anything about it - input file is broken in some way
<durandal_1707> also give full logs, by uploading them via pastebin.com or similar
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<ShaedS> how can I take this string: -af silenceremove=window=0:detection=peak:stop_mode=all:start_mode=all:stop_periods=-1:stop_threshold=0 and mark all the silent portions to a variable/list/array?
<durandal_1707> silenceremove filters modifies stream
<durandal_1707> it efectively trims silence
<durandal_1707> there is only silencedetect that outputs text with info when silence starts/stops...
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<ShaedS> durandal_1707> there is only silencedetect that outputs text with info when silence starts/stops. => yes I'm trying ot achiee the lossless verisno of this by identifying those start stops, and then passing that array/varibles into -ss -to
<durandal_1707> that only works with uncompressed PCM
<durandal_1707> you cant losslessy trim already compressed mp3/aac/whatever
<durandal_1707> you can only do some approximations of trim/cut
<durandal_1707> at the best
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<durandal_1707> and anything using filters and lossless makes no sense unless you use lossless codecs through all process chain
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<znf> Is it possible to do Super-Resolution (ie: upscaling) live?
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<Eoan> Hi, quick question, if you are playing something with ffplay on Windows, with current time showing in the terminal window, does pressing Enter move the output to a new line with the old line frozen?
<Eoan> This is how it works in gnome-shell. I use this to refine audio time measurements and would like to make sure it works the same in Windows
<Eoan> *gnome-terminal running bash shell
<Eoan> Clarification, focusing the terminal window and pressing Enter
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<aaabbb> znf: you mean in real time, >1x speed?
<znf> aaabbb, ye
<aaabbb> that depends on the upscaling algorithm you use, how much spare cpu cycles you have, the source resolution, and the target resolution
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<aaabbb> the biggest bottleneck will be reencoding the larger resolution video unless you're literally just upscaling and displaying directly to a monitor imediately
<znf> Eoan, that's not something that is specific to ffmpeg itself (or, ffplay in this case), but the terminal type that you're _currently_ using
<aaabbb> znf: ffmpeg -i input -vf "scale=1920x1080:flags=lanczos" -f null -
<aaabbb> for example, to test it
<znf> neither cmd.exe nor Windows Terminal allow you to hit enter when using ffplay
<znf> aaabbb, that's not the filter I'm talking about
<aaabbb> that will run only the decoder and upscaling using the lanczos algorithm btw
<znf> think sr=dnn_backend :)
<aaabbb> oh i see, the neural network upscaling thing
<znf> yup
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<aaabbb> do you have a fast cuda-enabled gpu?
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<znf> I can't test now, as I'm currently on an Intel ARC on my desktop, my 3090 is in a box somewhere
<aaabbb> are you using a pre trained model? i haven't used the sr filter before, but i've used ml for image procesing and i've found that the model can drstically influence performance
<znf> I haven't used that either, hence why I was wondering
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<Eoan> Thanks, znf!
<znf> Eoan, still, you _can_ actually use git's Bash to do that
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<Eoan> I see. Thank you again, znf
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<aaabbb> are there any good intermediate (intra-only, generation loss resistant) video formats supported by ffmpeg that are not patented encumbered and are open?
<aaabbb> supported either natively or via a well-maintained patch or fork
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<Eoan> aaabbb, would ffmpeg's own ffv1 work? Wiki says some people use it. Just use '-g 1' for intra-only?
<aaabbb> Eoan: isn't it lossless? i want a mezzanine codec, so lossy, but resistant to generation loss
<aaabbb> like prores
<kepstin> i haven't heard of anything that's explicitly resistant to generation loss. you're pretty much left with intra-only mpeg2 or vp9 or av1, none of which have that as a goal.
<kepstin> (mpeg2 counts since all of the patents on it have expired)
<Eoan> Hopefully someone else can give a better response. My understanding is that generation loss has not been a priority for most lossy codecs because they assume a short propagation length: things being created directly from the high-quality master file. But just having high bitrate prevents most generation loss, so why not just high-bitrate VP8, as a
<Eoan> codec that's reasonably fast and nearly as good as H.264?
<Eoan> I'm not really sure on patents status of it, though
<aaabbb> kepstin: all mezzanine codecs are resistant to generation loss, even tho they are lossy
<aaabbb> because they are intra-only so edits only affect individual frames and one edit doesn't require recompression of the whole gop
<kepstin> sure, but most lossy codecs can be used in intra-only mode
<aaabbb> kepstin: yeah but mezzanine codecs are optimized for that purpose, they are extremely fast to encode and decode, have accurate seeking etc
<kepstin> accurate seeking is a property of the container, not the codec, and many codecs are tunable to have extremely fast encode and decode.
<kepstin> I honestly normally use a fast preset on x264 for that sort of thing.
<aaabbb> that's not very good for editing
<aaabbb> prores is like 20x faster than i-frame x264 at the fastest preset and a low crf
<aaabbb> and the iframes in prores can be compressed more times before generation loss occurs (at the expense of compression ratio)
<aaabbb> but prores is proprietary apple...
<kepstin> well, when you make a new non-patent-encumbered codec which does what you want, please send patches to support it in ffmpeg.
<aaabbb> kepstin: my question was whether one already exists and is supported by ffmpeg either natively, or via a well-maintained 3rd party patch :)
<kepstin> if there was, everybody would already be using it.
<aaabbb> i doubt it, most people have no need for using an open intermediate codec since it's never used for distrubtion
<Eoan> Did a test. On my 14-year-old laptop with the CPUs limited to 800 MHz, I decoded 1 sec of 1188p ffv1 in 7.9 user seconds, all intra frames at 144 Mb/s. 1 sec of '-crf 10' H264 decoded in 3.6 user seconds, also all intra frames at 56 Mb/s. So if speed is your reason for not using ffv1, I don't think you need to worry about that. Just test it to see
<Eoan> if your system is fast enough
<aaabbb> Eoan: i'll have to test 8k 10-bit yuv444p at 120 fps with ffv1 and see if it can do anything
<aaabbb> intermediate codecs are often used at obscene resolutions, in order to later downscale them for releases
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<Eoan> A quick search suggests downscaling footage during editing: "Finally, if you want a small file size and ease of editing, consider a lower-resolution ProRes or DNxHD file. You might try transcoding your 4K footage to 720p ProRes proxies. In a proxy workflow, you convert your original camera media to a more edit-friendly codec while you’re working
<Eoan> in your NLE. Then, when you’re done, you relink the original full-resolution media for color grading and delivery."
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<aaabbb> yeah i do that when making simple edits where i can determine the correctness on 720p
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<Eoan> I was going to test generation loss, but '-b:v' apparently does not work for ffmpeg's Prores encoders. So my results for the same input, using ffmpeg with default codec options: codecs 'prores' and 'prores_aw', output size 21004kB at 15 fps; decode time 1.06 sec. Codec 'prores_kw', 33822kB at 3.9 fps; decode time 1.5 sec. Codec 'ffv1', 17633kB at
<Eoan> 15 fps; decode time 2.5 sec.
<kepstin> prores is _bigger_ than ffv1? that's kinda weird. But note that by default ffv1 is _not_ an intra-only codec; it maintains some state (in its entropy encoder? not sure) between frames.
<aaabbb> maybe it's using yuv444 and ffv1 is using 420 or something?
<Eoan> Except that the Prores was automatically converted to yuv422p10le (from yuv420). With that conversion, the ffv1 is 22070kB, encoded at 8.7 fps, decode time 5.3 sec. So it does decode a lot slower than Prores, but the file size is about the same.
<aaabbb> ah
<aaabbb> how big are ffv1's gops?
<kepstin> i don't recall what the default is, but you're free to set it down to 1 if you prefer
<Eoan> Thanks for reminding me, haha. I remembered '-g 1' on some tests but not this one. Add that, and it goes up to 22324kB, still 8.7 fps, still 5.3 sec to decode.
<aaabbb> ah!
<furq> do you need nle support for this intermediate codec
<furq> i guess you do if you want -g 1
<kepstin> ffv1 can also do sliced encoding which should improve multithread usage
<furq> because hardly anything supports ffv1
<Eoan> Yep, all my tests are on a 2-core processor
<aaabbb> furq: i do 90% of my editing with ffmpeg :)
<aaabbb> and things that use the various libav*
<furq> if you're sticking with oss stuff then that should be broadly fine
<aaabbb> i'll give ffv1 a try!
<Eoan> But those are the outputs from '  time ffmpeg -i ffv1.mkv -f null -an -' etc.
<furq> ffv1 is really impressive for an intra only codec
<furq> just a bit slow and impractical if you want to use a popular nle
<aaabbb> Eoan: oh i see, is the encoding speed similar to decoding?
<kepstin> yeah, it's pretty rare that I'm actually encoding stuff to a temporary file for re-encoding later; most of the editing stuff i do is setting up pipelines which do all of the processing from original input files to final output file in one step (helps that i basically never need to use 2-pass encodes)
<aaabbb> the files i work with are the kind that i have to do deep inspections. not uncommon for me to consider it finished, and a week later find a problem
<aaabbb> in some instances i literally step frame by frame
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<kepstin> oh, huh, ffv1 supports alpha. that's handy.
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<Eoan> Report: compared prores with standard options to x264 with ' -tune fastdecode -preset ultrafast -crf 4 -g 1'. Subtracting the decode time, Prores encoded in 2.97 sec, x264 in 2.39. Decode time is about the same, 1 sec. H264 is 121 Mb/s, prores is 172 Mb/s but Prores 30 dB from the ssim filter while H264 is just 28 dB. But I might be doing something
<Eoan> wrong, because even with '-crf 0' I only get 32 dB for H264; the plane checksums differ. With 420p input and output, the frames are identical with -crf 0, so maybe something is going wrong with the 10-bit processing.
<kepstin> note that crf 0 is not lossless in 10bit ; the reference way to enable lossless is "-qp 0" in both 8-bit and 10-bit
<Eoan> Ah~
<kepstin> that applies to x264 specifically (other encoders work differently)
<aaabbb> interesting
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<Admin__> hello
<aaabbb> hello Admin__
<Admin__> can some one help me fix this?
<Admin__> ffmpeg version N-112688-gfd1712b6fb-20231104 Copyright (c) 2000-2023 the FFmpeg developers
<Admin__> built with gcc 13.2.0 (crosstool-NG 1.25.0.232_c175b21)
<Admin__> configuration: --prefix=/ffbuild/prefix --pkg-config-flags=--static --pkg-config=pkg-config --cross-prefix=x86_64-w64-mingw32- --arch=x86_64 --target-os=mingw32 --enable-gpl --enable-version3 --disable-debug --disable-w32threads --enable-pthreads --enable-iconv --enable-libxml2 --enable-zlib --enable-libfreetype --enable-libfribidi --enable-gmp --enable-lzma --enable-fontconfig --enable-libharfbuzz --enable-libvorbis --enable-opencl --disable-libpulse --
<Admin__> enable-libvmaf --disable-libxcb --disable-xlib --enable-amf --enable-libaom --enable-libaribb24 --enable-avisynth --enable-chromaprint --enable-libdav1d --enable-libdavs2 --disable-libfdk-aac --enable-ffnvcodec --enable-cuda-llvm --enable-frei0r --enable-libgme --enable-libkvazaar --enable-libass --enable-libbluray --enable-libjxl --enable-libmp3lame --enable-libopus --enable-librist --enable-libssh --enable-libtheora --enable-libvpx --enable-libwebp --
<Admin__> enable-lv2 --enable-libvpl --enable-openal --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopenmpt --enable-librav1e --enable-librubberband --enable-schannel --enable-sdl2 --enable-libsoxr --enable-libsrt --enable-libsvtav1 --enable-libtwolame --enable-libuavs3d --disable-libdrm --enable-vaapi --enable-libvidstab --enable-vulkan --enable-libshaderc --enable-libplacebo --enable-libx264 --enable-
<Admin__> libx265 --enable-libxavs2 --enable-libxvid --enable-libzimg --enable-libzvbi --extra-cflags=-DLIBTWOLAME_STATIC --extra-cxxflags= --extra-ldflags=-pthread --extra-ldexeflags= --extra-libs=-lgomp --extra-version=20231104
<Admin__> libavutil 58. 31.100 / 58. 31.100
<Admin__> libavcodec 60. 32.102 / 60. 32.102
<aaabbb> please don't dump everything to irc
<Admin__> libavformat 60. 17.100 / 60. 17.100
<Admin__> libavdevice 60. 4.100 / 60. 4.100
<Admin__> libavfilter 9. 13.100 / 9. 13.100
<Admin__> libswscale 7. 6.100 / 7. 6.100
<Admin__> libswresample 4. 13.100 / 4. 13.100
<Admin__> libpostproc 57. 4.100 / 57. 4.100
<Admin__> [youtube] Extracting URL: https://www.youtube.com/watch?v=SItIaWAjI_4
<aaabbb> use a paste site
<Admin__> [youtube] SItIaWAjI_4: Downloading webpage
<Admin__> [youtube] SItIaWAjI_4: Downloading ios player API JSON
<Admin__> [youtube] SItIaWAjI_4: Downloading android player API JSON
<Admin__> [youtube] SItIaWAjI_4: Downloading m3u8 information
<Admin__> [info] SItIaWAjI_4: Downloading 1 format(s): 251
<Admin__> [download] Destination: -
<Admin__> [download] 0.4% of 3.71MiB at 1.72MiB/s ETA 00:02Input #0, matroska,webm, from 'fd:':
<Admin__> Metadata:
<Admin__> encoder : google/video-file
<Admin__> Duration: 00:03:54.50, start: -0.007000, bitrate: N/A
<Admin__> Stream #0:0(eng): Audio: opus, 48000 Hz, stereo, fltp (default)
<Admin__> Stream mapping:
<Admin__> Stream #0:0 -> #0:0 (opus (native) -> pcm_s16le (native))
<Admin__> Output #0, s16le, to 'pipe:':
<Admin__> Metadata:
<Admin__> encoder : Lavf60.17.100
<Admin__> Stream #0:0(eng): Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s (default)
<Admin__> Metadata:
<Admin__> encoder : Lavc60.32.102 pcm_s16le
<Admin__> read 11520 bytestime=00:00:00.00 bitrate= 0.0kbits/s speed=N/A
<Admin__> read 11520 bytes
<Admin__> read 3356 bytes
<Admin__> [aost#0:0/pcm_s16le @ 000001d5140f07c0] Error submitting a packet to the muxer: Broken pipe
<Admin__> [out#0/s16le @ 000001d5140ab300] Error muxing a packet
<Admin__> [out#0/s16le @ 000001d5140ab300] Error writing trailer: Broken pipe
<Admin__> [out#0/s16le @ 000001d5140ab300] Error closing file: Broken pipe
<Admin__> [out#0/s16le @ 000001d5140ab300] video:0kB audio:33kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
<Admin__> size= 29kB time=00:00:00.34 bitrate= 692.4kbits/s speed=26.9x
<Admin__> Conversion failed!
<Admin__> ERROR: unable to write data: [Errno 32] Broken pipe
<Admin__> Exception ignored in: <_io.TextIOWrapper name='<stdout>' mode='w' encoding='utf-8'>
<Admin__> OSError: [Errno 22] Invalid argument
<Admin__> sorry
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<aaabbb> next time use a paste site lol
<Eoan> My results: '-tune fastdecode -preset veryfast -qp 0 -g 1' (the fastdecode actually does nothing measurable here) is lossless, encodes in 3.6 sec compared to the 2.97 for Prores, and is 305 MB/s compared to the 182 MB/s for ffv1, but decodes in 1.9 sec compared to 5.3 for ffv1 and 1.0 for Prores. The higher decode time is just because of the higher
<Eoan> bitrate to make it lossless. So it's at least comparable to Prores's performance in both encoding and decoding. But if patents are still an issue here, VP8 is probably similar; even though x264 won the Moscow codec competition for a bunch of years for quality at a given speed and bitrate, the quality difference is mostly at lower bitrates.
<Admin__> here
<aaabbb> Eoan: ffv1 isn't patent encumbered is it?
<Admin__> i used yt-dlp.cmd -f 251 -o - https://www.youtube.com/watch?v=SItIaWAjI_4 | ffmpeg -acodec opus -i - -f s16le -ar 48000 -ac 2 -
<Eoan> The only thing I see on https://en.wikipedia.org/wiki/FFV1 is "Early experimental versions of FFV1 used the CABAC Arithmetic coder from H.264, but due to the uncertain patent/royalty situation, as well as its slightly worse performance, CABAC was replaced by Range encoding."
<Admin__> how can i fix this?
<aaabbb> Admin__: can you replicate it with just an ffmpeg command?
<Admin__> sure
<Eoan> Admin__, this outputs it to standard output, right? Is that what you're trying to do?
<aaabbb> it looks to me like a problem with wherever you're feeding the output
<aaabbb> because ffmpeg isn't the one that broke the pipe, the output pipe was broken, and ffmpeg can't do anything about it
<Admin__> it gives me this
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<Admin__> and freeze
<aaabbb> Admin__: i mean try to do it with just ffmpeg, no piping
<furq> Admin__: not a solution but you can just do ffmpeg -i "$(yt-dlp -f 251 -g https://www.youtube.com/watch?v=SItIaWAjI_4)"
<furq> well maybe not on windows but you can just copy the url from yt-dlp -g
<aaabbb> furq: it looks like the broken pipe is on the output of ffmpeg not the input
<furq> yeah that's why i said not a solution
<furq> if i had to guess whatever the pipe is connected to isn't reading from it
<Eoan> If you're trying to make your system play it, you might be able to do something like "-f pulse -", but ffplay may be better for this as it's interactive. And beyond that, 1) you can sometimes cache input 2) YouTube sometimes acts weird, or used to in the past, if you tried to seek using the URLs that yt-dl provides, like closing connections
<Eoan> unexpectedly, so I've always found it better to download things, even if just to a temporary location
<furq> well you wouldn't be able to seek that anyway
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<Admin__> i think i put that command in my code so it return error
<furq> i assume from the io.TextIOWrapper exception that this is a python script calling ffmpeg and reading in pcm
<furq> so there's probably a bug in that
<aaabbb> yeah, i think ffmpeg i working just fine
<Admin__> i tried my command and it work
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<Admin__> but where the error frm
<Admin__> *from
<aaabbb> in your script probably
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<Eoan> If it's still not working, maybe use a local file as the input so you can eliminate one source of error. The 'utf-8' isn't causing an error, is it? From an invalid bit sequence?
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<Admin__> let me try it again
<Admin__> i found the problem
<aaabbb> what was it?
<Admin__> my code need 16 bit streo raw
<aaabbb> and it closed the pipe on ffmpeg when it didn't getthat?
<Admin__> and it check that by divisible by 4
<Admin__> and its stop my read loop
<aaabbb> you don't need to force opus
<aaabbb> it will detect automatically
<Admin__> but does it detect correctly
<aaabbb> yes
<aaabbb> and unambiguously
<Admin__> so i dont need -f s16le right
<aaabbb> for output you do
<aaabbb> well input as well
<aaabbb> but for codecs like opus you don't
<aaabbb> any "raw" codec like pcm, you have to specify the format but if it's in a container like ogg opus, or mp4 opus then it's detected unambiguously
<Admin__> but i need audio raw
<aaabbb> that's the output
<aaabbb> you don't need -acodec opus
<Admin__> let me try again
<Admin__> what does this error mean
<aaabbb> not sure, try doing just a plain ffmpeg command
<Eoan> Did you remove the '-f s16le'? "Unable to choose an output format"
<aaabbb> you need to keep the -f, i just meant you don't need -acodec opus
<aaabbb> without -f, it has to detect output type based on file extension such as .mp3, it knows what codec to use. if you output to stdout with - you need to be explicit about the output
<Admin__> it still can't divisible by 4
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<furq> what's supposed to be divisible by 4
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<Admin__> i think its legth of data
<Admin__> *length
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<Nintendo> I am looking at a decrypted DVD folder with 5 VOB files for the main video. If I use mpv to watch VTS_01_1.VOB, the video has subtitles available, but ffprobe says there are none. What might be happening?
<Nintendo> Ffprobe output: http://ix.io/4Lhd
<galad> instead of opening the VOB files it would be better to use something that understands the VIDEO_TS structure, some info is only in the .ifo files
<furq> probably increasing -analyzeduration and/or -probesize would find the subtitle streams
<furq> but do what galad said
<galad> there was someone working on adding some video_ts reading support to ffmpeg, but I don't think it's ready yet
<furq> amazing that this still sucks all these years later
<furq> not in ffmpeg just in general
<galad> handbrake can read a video_ts folder, maybe it has all the things you need
<furq> that works if you commit to doing everything with handbrake
<Nintendo> Very true. But from what I can tell, the DVD format isn't a very fun one to work with. Thank you both for the suggestions. Since ffmpeg does not seem to be able to process IFO, I will give -probesize and whatnot a shot before resorting to other tools.
<galad> furq: the issue is that almost no one cares about dvd-video anymore
<furq> last i checked it wouldn't just remux it
<Admin__> can someone help me?
<galad> well, not really an issue, just a fact
<furq> you should always remux with something that can actually process dvd titlesets before encoding
<furq> makemkv or dvdbackup -t or pgcdemux or something
<furq> unless you're 100% sure that the vobs exactly correspond to the titles
<aaabbb> Admin__: try replicating it with a pure ffmpeg command
<aaabbb> outside of your script
<Eoan> Admin__, try outputting from ffmpeg to a file (same format as current), and then using that file as input to your script.
<furq> i think he said the ffmpeg command by itself works
<furq> in which case we can't guess what the issue is without seeing the script
<Admin__> ok
<furq> maybe more of a question for #python or however many #s they have
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<admin_____> how can i read it to stdout?
<aaabbb> admin_____: that's a general linux (or perhaps python) question. ffmpeg will write to stdout with setting - for the output file
<admin_____> how can i read my out audio file?
<Eoan> No idea how to do it from a script; can you use 'cat'? Is this a shell script where you can use an operator like "<"?
<Eoan> I know nothing about python
<aaabbb> admin_____: ffplay -f s16le -i yourfile
<admin_____> it work
<admin_____> but how can i put that into my code?
<aaabbb> i don't know. if it works, that means ffmpeg is working fine
<admin_____> no one know how to fix that?
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<aaabbb> this is the ffmpeg channel so we're not gonna be experts in python
<aaabbb> try joining #python
<admin_____> i thing that error from ffmpeg
<aaabbb> the error is from ffmpeg because the pipe is being closed by your script
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<Eoan> Like I said, output to a file, and then use the file as input to the script. If you just do 'cat file -' it should provide the same input to the script: if it doesn't, then you can work from there. It eliminates ffmpeg as a potential source of the problem, simplifyig your troubleshooting
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<Eoan> Or just "cat file", automatically goes to stdout
<aaabbb> he's gone already
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<robbi_blechdose> Hey everyone, I have a question about MPEG2 (TS) encoding:
<robbi_blechdose> I recorded video with a Sony HRV S270 and the MRC1 recorder. Said recorder splits the HDV/MPEG2 files once they hit 4GiB
<robbi_blechdose> And apparently this means incomplete GOPs, which translates to dropped/frozen frames at the end of one file, in my case 9 full frames.
<robbi_blechdose> Sony provided a tool to fix this, but it's old and windows-only so I have no way to know how this worked.
<robbi_blechdose> Can ffmpeg repair these files somehow?
<Eoan> If it's splitting them at the bit level, you may be able to just concat them
<Eoan> I'm thinking the concat protocol, not the format
<robbi_blechdose> I did try that (using "ffmpeg -f concat -i concatlist.txt -g 12 -c copy out.m2t"), but still get those dropped frames
<Eoan> But assuming you have a modern filesystem that supports files above 4 GB, try copying the first file and concatting the next onto it
<robbi_blechdose> With or without the "-g 12" makes no difference
<Eoan> So that's the concat format. I'm looking at the manual about the concat protocol, which is different
<Eoan> Search in the manual for 'Physical concatenation protocol.'
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<Eoan> Anyway, if it works, it's just like actual joining the files together, which depend son how your camera does it
<furq> the protocol does the same thing as cat foo.ts bar.ts > baz.ts
<Eoan> If it only splits at frame boundaries ... actually then your command should have worked, because I've split files at non-GOP points (using -copyinkf) and it worked fine
<furq> it should always work with mpegts but maybe you'll end up with undecodable frames at the joins
<Eoan> If there was that tool, it was probably just joining them (it's like multi-part zip archives) so the camera isn't throwing away data
<robbi_blechdose> I'm amazed, I *think* this worked
<furq> yeah it would make sense that it just splits exactly at 4GB
<furq> in which case you can just cat
<robbi_blechdose> I wasn't quite sure about that, since it almost looked like it started a new GOP in the new file
<furq> well that should still be fine as long as the rest of the gop is in the new file
<furq> i'm surprised the demuxer didn't deal with that though seeing as it doesn't decode the stream
<robbi_blechdose> I just checked again in kdenlive to have a look at the individual frames
<robbi_blechdose> and this has indeed worked! Thank you very much guys!
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<robbi_blechdose> One more thing: Is ffmpeg capable of decoding MPEG2 multichannel audio?
<robbi_blechdose> I had the camera set to 4-channel for this recording
<Eoan> Fastest way to check is probably to use ffplay
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<robbi_blechdose> I've found no documentation on it, so that's why I'm asking
<robbi_blechdose> This is the only thing google turns up: https://trac.ffmpeg.org/ticket/1258
<Eoan> ffmpeg can repan audio or whatever
<furq> if you mean mp2 multichannel then i don't think so
<robbi_blechdose> Yeah it's mp2 multichannel
<robbi_blechdose> From what I understand it's 4 channels somehow mixed into 2 in a backwards-compatible way
<Eoan> But the ticket is 12 years old. So just open it with ffplay, and press W to switch to audio waveform display and see if it shows 4 channels
<durandal_1707> no that is not supported
<furq> ffprobe will show the channel layout
<durandal_1707> why you use 4ch mp2?
<furq> i assume it'll just report stereo
<durandal_1707> it ignores >2 channels
<durandal_1707> in mp2
<robbi_blechdose> Yep, this is ffprobe: "Stream #0:1[0x101]: Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, fltp, 384 kb/s"
<robbi_blechdose> I used 4ch mp2 because I was stupid, I only needed 2 channels but didn't think about it in the moment
<robbi_blechdose> And the camera only does HDV, which does 4 channel audio this way
<durandal_1707> adding >2 channel support in mp2 is non-trivial work
<robbi_blechdose> I bet, yeah. Just thought I'd ask in case anything had happened
<robbi_blechdose> I did try mctoolamed which is linked in that ticket and it can extract the channels into individual wav files
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<Admin__> can i opus data for using this command?
<Admin__> can i opus data for using this command?
<Admin__> yt-dlp.cmd -f 251 -o - https://www.youtube.com/watch?v=j-2DGYNXRx0 | ffmpeg -i - -f s16le -ar 48000 -ac 2 -
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<Nintendo> Admin__, I think the output of yt-dlp's --list-formats will tell you.
<ecapi> Sega
<another|> Admin__: What's the goal here?
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<Admin__> yt-dlp --list-formats dont work for me
<Admin__> i want it give me opus packets
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<another|> Why doesn't it work? What doas it say?
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<raini_ok> yt-dlp https://www.youtube.com/watch?v=j-2DGYNXRx0 -x --extract-audio opus -o -
<raini_ok> idk
<another|> not how it works
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<ecapi> does windows have a zlib like macos or linux?
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<BtbN> zlib is plain C, it works literally anywhere that has a C compiler
<ecapi> i know but is there system library for it or does it have to be supplied bu the application
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<BtbN> You will have to ship it yourself. It's not part of the WinAPI
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<ecapi> ok thats what i was thinking thanks
<Swedaniel> i get this error from web browser when using the kiwi irc iframe plugin thing: Error Connecting (Error: xhr poll error)
<Swedaniel> any clues?
<Swedaniel> must poll get back thru the router via the selected port?
<Swedaniel> oh wrong chat maybe i tought i was in Kiwi IRC chat but it is FFMpeg :D
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