<znf> should try that, thanks
<znf> if that doesn't pan out, patching ffmpeg.c is always an option for me, I guess
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<BtbN> Just symlink to the ffmpeg binary a bunch of times, with descriptive names of what it's used for
<znf> or that
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<znf> I'm trying to fix one of my custom patches to ffmpeg, but I'm having some trouble figuring out what I'm doing wrong. It used to apply and work fine on the 6.0 branch. It's very sloppy code, but this worked: https://pastie.dev/V9jCMQ.diff
<znf> I refactored it for 6.1, but something is wrong
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<Marth64> znf: is git apply just not working? or a bug after you apply the patch
<znf> The scope of the patch is for ffmpeg to POST it's progress to a local URL, but instead of it's usual behavior of doing a chunked transfer over the same connection, POST on every ~45s (more often if it's in the first few seconds of the video)
<znf> Marth64, getting to it, didn't want o post a novel :)
<Marth64> np
<Marth64> i am guilty of posting novels :D
<znf> The patch *does* apply now, properly, it's just that my check @ line 33 always returns true, due to `hack_cur_frame` always being -1
<znf> before 6.1 (I'm not sure the commit that changed that behaviour), I was just setting `hack_cur_frame = frame_number;` at line 50
<znf> (of the patch)
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<znf> now my logging shows that `hack_cur_frame` is always -1, thus spamming the URL
<znf> my C knowledge is pretty poor, so I'm not sure what I'm doing wrong here
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<Marth64> znf: I see the issue.
<znf> do tell, please
<Marth64> I think. What happens if you log out the value of frame_number before you set it to hack_cur_frame ?
<Marth64> I noticed hack_cur_frame is int64_t (signed int), frame_number is uint64_t (unsigned int)
<Marth64> I am not familiar with this part of the code, but that's possibly supsicous
<znf> used to be the same, I just recently changed it, but let me see
<znf> that makes it not run at all
<znf> assuming it's because it's unsigned and I can't use -1?
<Marth64> You won't be able to set hack_cur_frame to -1 from that point
<Marth64> Yes, correct. sorry, my internet is bad so a little delayed.
<Marth64> Try 0?
<znf> yeah, that makes it work again, but it still remains 0 forever
<Marth64> hmm. okay, then that's not the issue
<Marth64> sorry. wondering if it's related to some of the threading changes that were ramping up between 6.1->7.0
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<znf> it does kinda seem that the stats code doesn't execute
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<znf> as I don't see the stats output on stdout like I see in the previous version, ie: https://pastie.dev/8vkRuo.yaml
<Marth64> hmm
<znf> wonder if I broke it
<znf> ...perhaps I broke it, indeed
<Marth64> find a fix? :D
<znf> perhaps around line 42 of the patch
<Marth64> ah
<Marth64> yeah, looks like you're adding something around an existing return
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<znf> ...but so is the 6.1
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<znf> how the hell do you git diff between the release/6.0 release/6.1 branches
<znf> keeps telling me fatal: ambiguous argument 'release/6.0': unknown revision or path not in the working tree.
<znf> meh, just gonna use the tags
<Marth64> znf: got an idea
<znf> yes?
<Marth64> bear with me, trying to go one year back in commit history. found a commit that might be worth testing before/after
<Marth64> then you can at leas isolate the problem to a change
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<znf> I think my patch was applying correctly on 9f0077cc0397cb4ab24f33d38248beb604eb0ff5
<znf> which is what where I think I wrote it
<znf> or not
<Marth64> does it work on 1e91a3950232e391ac49a5dfdc958154a212fa4a
<znf> let me see
<Marth64> shortly after this, saw some refactoring go in to that neck of the woods
<monokrome> Does anyone know how to get ffmpeg to stop "Guessing channel layout"? I provide -ac and -channel_layout before and after -i and it still decides to guess
<znf> my _current_ patch or the _old_ patch?
<Marth64> znf: the old 6.0 patch should work, since 1e91a3950232e391ac49a5dfdc958154a212fa4a is relatively close just after 6.0
<znf> doesn't apply
<Marth64> hm
<znf> I'm very unsure on _which_ commit I wrote it around
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<znf> my old patch says:
<znf> diff --git a/fftools/ffmpeg.c b/fftools/ffmpeg.c
<znf> index 46a85b41a8..189b0e9e4e 100644
<Marth64> much has changed in there, but I am not so familiar with fftools. I apologize, wish I could pinpoint exactly what to do.
<znf> so, I'd assume it works on around commit number 100644
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<znf> mkay, think I found a way to fix it
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<znf> I've moved: hack_cur_frame = atomic_load(&ost->packets_written);
<znf> right under const float q = ost->enc ? atomic_load(&ost->quality) / (float) FF_QP2LAMBDA : -1;
<znf> as it "enters" the loop
<znf> which makes sense, because I don't have a god damn filter, what was I even thinking
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<Marth64> znf: fingers crossed
<znf> wonder WHY did it work before
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<Marth64> had one of those couple months back. hours spent debugging. was simply running in the wrong directory
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<aaabbb> can someone reproduce this bug? removing the sei from h264 prevents seeking past the first gop if using open gops
<aaabbb> ffmpeg -f lavfi -i testsrc=d=20 -c:v libx264 -x264opts open-gop -bsf:v filter_units=remove_types=6 test.mp4
<aaabbb> ffmpeg -ss 15 -i test.mp4 -f null -
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<JEEB> aaabbb: welcome to H.264; open GOP bases on normal intra frames and the random access point SEI
<JEEB> they moved that into their own NAL units in HEVC
<JEEB> in theory the packet should have the random access point flag, which the muxer should follow, but not sure what happens with the BSF in the middle (it's after the encoder and before the muxer)
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<aaabbb> JEEB: even with a closed gop, if i remove sei i get the error "mmco: unref short failure" when seeking to a gop past the first one (although it decodes correctly)
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<aaabbb> JEEB: so if i wanted to remove the sei user data, would i use sei_user_data from h264_metadata? then just specify the right uuid according to uuid_iso_iec_11578 with a blank string?
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<destmaster> Hi, on encoding audio using libtwolame and mpegts as muxer I'm getting this error after about 1:30 hours
<destmaster> ideo:0kB audio:109227kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 59.999962%
<destmaster> someone know what can be the reason?
<destmaster> this is the ffmpeg command
<destmaster> ffmpeg -y -hide_banner -nostats -thread_queue_size 1024 -i /tmp/inpipe -osr 48000 -c:a libtwolame -b:a 160k -minrate 160k -maxrate 160k -bufsize 160k -mode stereo -mpegts_transport_stream_id 1 -mpegts_original_network_id 1 -mpegts_service_id 1 -mpegts_service_type digital_radio -pcr_period 20 -metadata service_name="-" -metadata service_provider="-" -mpegts_start_pid 769 -mpegts_pmt_start_pid
<destmaster> -muxrate 256k -mpegts_flags 'system_b' -mpegts_flags 'nit' -f mpegts pipe:1 | cat > /dev/spidev4.0
<JEEB> what you posted is not the error
<JEEB> use a pastebin and possibly add `-v verbose` to your command line
<JEEB> then link the pastebin here
<destmaster> ok, thank you
<destmaster> here the pastebin https://pastebin.com/n3m6UX6f
<destmaster> meanwhile I'll do a -v verbose log
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<SnakesAndStuff> ffmpeg -f v4l2 -video_size 3840x2160 -input_format mjpeg -framerate 24 -i /dev/video0 -f alsa -i hw:2 -map 0:0 -f rawvideo video.raw -map 1:a -c:a copy out.wav -f rawvideo - | ffplay -f rawvideo -video_size 3840x2160 -pixel_format yuvj422p -framerate 24 -framedrop -
<SnakesAndStuff> is ther any sane reason why this would create an out.wav that lasts only a few seconds, but continues to stream and record video?
<SnakesAndStuff> I'm strongly guessing it is something on my end. I'm guessing I need to increase the verbosity and start digging unless there is something obvious?
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<destmaster> Hi, someone can help me to identify the issue on live encoding... here the verbose log
<JEEB> there's no errors so it seems like the pipe got closed? > No more output streams to write to, finishing.
<JEEB> which might be that the input finished
<kepstin> yeah, i'd assume that it has EOF on the input pipe
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<destmaster> @JEEB You're right. It's strange, my pipe is filled by arecord process
<destmaster> arecord -c 2 -r 48000 -f S32_LE -D hw:0,1 > /tmp/inpipe
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<destmaster> is the right way to make a pipe?
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<JEEB> for files you may utilize follow + rw_timeout to attempt to still read a file even if for a while there are no updates
<JEEB> but... if your input dies it sounds like you have a larger problem ^^;
<JEEB> you probably want to debug for how long it dies
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<devinheitmueller> I would suggest noting the PID of the arecord process, and see if it changes after the EOF. I’ve seen people who automatically respawn stuff like arecord. After the EOF they see it’s still running not noticing that it’s a different instance of the process (and the first instance terminating caused the EOF to be sent).
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<devinheitmueller> Also, it might be worthwhile to use the alsa libavdevice to talk to the capture source directly, rather than separately running arecord and piping its output to ffmpeg.
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<devinheitmueller> See https://trac.ffmpeg.org/wiki/Capture/ALSA for more info
<destmaster> @devinheitmueller thanks for suggestions. I've tried to use alsa in ffmpeg but I'm got buffer errors
<destmaster> I'will try again to use alsa directly in ffmpeg
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<devinheitmueller> I’ve dealt with plenty of finicky ALSA drivers over the years. Many of them have subtle bugs that only crop up after minutes/hours of operation.
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<destmaster> @devinheitmueller Many thanks. I've, just now, launched a new test using alsa. It seems to work well. I will check for error in the next 24 hours.
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<devinheitmueller> destmaster: Ok, good luck.
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<rudds> i'm having a weird issue streaming to Twitch with ffmpeg and using hardware h.264 encoding (both h264_qsv and h264_nvenc) that doesn't happen with libx264
<rudds> which is that after starting the stream, the Twitch player doesn't load in and start playing, although the channel page does flip over to saying i'm live
<rudds> audio comes through, but no video until 60-90 seconds after starting the stream
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<rudds> and you have to hard refresh the page to get the player to load in and display anything
<rudds> i've A/B tested this a ton, the stream starts instantly with libx264
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<rudds> but both Quick Sync and nvenc have the same issue
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<rudds> any ideas? the video output parameters look identical to me between libx264 and the hardware encoders (other than the pixel format being listed as 'qsv' for Quick Sync, but it's still yuv420p for nvenc)
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<rodeo> not enough information I'd say, we don't even know which muxer(s) you're using or how you're sending the video to Twitch
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<rudds> the maximal command i'm using for qsv is this, although i've stripped it down as much as possible (removed hardware decoding, tried different h.264 presets etc.) and it's the same behavior regardless
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<rudds> fmpeg -re -init_hw_device qsv=hw -filter_hw_device hw -hwaccel qsv -hwaccel_output_format qsv -i $INPUT -map:v 0:0 -map:a 0:1 -c:v h264_qsv -profile:v main -preset:v veryslow -vf hwupload=extra_hw_frames=64,format=qsv -g 120 -b:v 8M -maxrate 8M -bufsize 4M -pix_fmt yuv420p -c:a aac -b:a 320k -f flv rtmp://$TWITCH/$STREAMKEY
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<rodeo> My guess is x264 might output SPS/PPS with every IDR frame by default whereas the other two might not
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<rodeo> I’m not sure how FLV works in terms of storing the extradata, but I’m guessing it possibly/probably doesn’t have a dedicated way of having track-level extradata (unlike, say, MP4, MOV or Matroska)
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<rudds> rodeo: can that be corrected for in ffmpeg, or is that hard coded into each encoder?
<rodeo> There should be options controlling placement of the parameter sets (SPS, PPS) specific to each encoder but I do not know what they are for QSV nor NVENC
<rodeo> They must be documented somewhere though
<rudds> ok thanks, i'll start digging around in that direction and see what i can find
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<rodeo> worst case scenario some of the qsv and nvenc files in libavcodec will contain the supported options, but I cannot easily check on my phone
<rudds> time to put my very limited ability to read C to the test
<rudds> assuming i can find the appropriate file
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<FlorianBad> What are your thoughts on `-O3 -mtune=native -march=native` for ffmpeg? Is it likely to still work between a i9-14900HX and a next generation of intel?
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<furq> is there even a next generation of intel
<furq> it's a waste of time in general and -O3 isn't guaranteed to work at all
<JEEB> not to mention that autovectorization is disabled in most configurations in FFmpeg's configure
<JEEB> and that a lot of the commonly utilized optimizations are hand-written and don't require extra optimizations from the compiler
<JEEB> but of course, feel free to benchmark use cases useful to you :P
<furq> and most of your cpu time is most likely spent in an external library
<FlorianBad> ok, what about -mtune=native -march=native options?
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<furq> those are what we were just talking about
<JEEB> feel free to benchmark is all I'm going to say
<JEEB> and if you go into such stuff, while at it disable the disabling of auto-vectorization, and hope FATE still passes for your binary :)
<furq> at least enable lto if you're going to go down this road
<furq> but probably just don't
<JEEB> also probably profile guided optimization? :D
<JEEB> x264 has an option for that and I wonder how many people still use it.
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<xx> I need to blur an entire video from start to finish, the whole picture not just a part of it (like some box), how should I do that?
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<furq> -vf boxblur
<furq> or whatever blur filter you want
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<xx> is there a neat list somewhere of the different blur filters? I can read the manpage but would prefer a place where it's presented a little more comprehensibly
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<xx> -filters and I'll test them out
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