December 2019 - Ffmpeg-devel-irc

ffmpeg-devel.log.20191225
by burek 26 Dec '19

26 Dec '19

[00:00:00 CET] --- Wed Dec 25 2019

1 0

ffmpeg.log.20191225
by burek 26 Dec '19

26 Dec '19

[00:00:04 CET] <kepstin> no, 6 bits per second isn't high enough to do anything resembling video :) [00:00:24 CET] <kepstin> (maybe you mean -b:v 6M for 6 megabits/s?) [00:00:28 CET] <a8_> oh [00:00:29 CET] <a8_> ues [00:00:31 CET] <a8_> sorry [00:01:08 CET] <a8_> how to convert an entire folder and have the output be in another folder [00:01:44 CET] <kepstin> for best results (closest to the desired target bitrate, and consistent quality over the length of the video), you should run that command twice - first with "-pass 1" and again with "-pass 2" [00:01:55 CET] <kepstin> if you have audio, you probably want to add "-c:a copy" as well [00:02:13 CET] <kepstin> to convert an entire folder, write a shell or batch script loop [00:03:10 CET] <a8_> thanks! i was googleing about passes and it keeps the first pass too? how would i tell it to not keep the first pass [00:03:51 CET] <kepstin> when you run the second pass, you normally overwrite the output file from the first pass [00:04:04 CET] <a8_> ok, thanks [00:05:03 CET] <kepstin> (ffmpeg writes an extra log file that stores the information from the first pass for the second pass to read, you might want to delete it when you're done) [00:05:30 CET] <a8_> ok, whats the name of the log file going to be? [00:06:15 CET] <kepstin> "ffmpeg2pass-N.log" where N is the stream number within the file that the log is for [00:06:33 CET] <kepstin> you can override the prefix with the -passlogfile option, look for it in the docs [00:06:49 CET] <a8_> thanks [00:07:03 CET] <kepstin> (don't run two ffmpegs at the same time using the default log filename, they'll write to the same file and mess everything up) [00:07:19 CET] <kepstin> at the same time in the same directory* [00:13:31 CET] <a8_> ffmpeg -i /h264/h264Input.mp4 -c:v libx265 -b:v 6M -c:a copy -output /h265/h265output.mp4 -pass 1 [00:13:43 CET] <a8_> can i tell ffmpeg to output to folders like that? [00:13:53 CET] <k4m3nk0> is there a way to convert videos to 1080p or 720p to be h265 .mkv i'm looking for good ffmpeg command that wont take a long time on my specifications https://i.imgur.com/MokBs0V.png [00:14:01 CET] <a8_> and replace /h264/ with the full path* [00:14:20 CET] <a8_> h265 is mp4 i think @k4m3nk0 [00:21:55 CET] <kepstin> a8_: output options go before the output filename, and there is no option named "-output" [00:22:49 CET] <k4m3nk0> also can i do -pass 2 [00:22:52 CET] <k4m3nk0> ?? [00:22:57 CET] <kepstin> k4m3nk0: there is no way to encode h265 quickly on your computer [00:23:00 CET] <k4m3nk0> for two pass encoding [00:23:21 CET] <a8_> k4m3nk0 you need to run command with -pass 1 first [00:23:37 CET] <Hello71> >h265 is mp4 [00:23:40 CET] <Hello71> yeesh [00:23:44 CET] <a8_> is it not? [00:23:50 CET] <a8_> can it not be* [00:23:53 CET] <kepstin> mp4 is one container that you can put h265 video into [00:24:00 CET] <kepstin> but that's not the only codec you can put into mp4 files [00:24:38 CET] <a8_> ffmpeg -i h264Input.mp4 -c:v libx265 -b:v 6M -c:a copy h265output.mp4 -pass 1 -threads 2 -preset veryslow [00:24:43 CET] <a8_> does that work? i think i fixed it [00:24:46 CET] <k4m3nk0> why ni -pass 2 [00:24:53 CET] <kepstin> a8_: output options go before the output filename [00:25:04 CET] <kepstin> a8_: move the output filename to the end after all the other options [00:25:14 CET] <a8_> ffmpeg -i h264Input.mp4 -c:v libx265 -b:v 6M -c:a copy -pass 1 -threads 2 -preset veryslow h265output.mp4 [00:25:21 CET] <a8_> like this? [00:25:34 CET] <kepstin> a8_: should work now. [00:25:37 CET] <a8_> thanks [00:27:09 CET] <a8_> i get this [00:27:10 CET] <a8_> [NULL @ 0x7ffc3802bc00] Unable to find a suitable output format for '6M' [00:27:10 CET] <a8_> 6M: Invalid argument [00:28:08 CET] <kepstin> k4m3nk0: actually, hmm, the celeron N3350 might have the intel quicksync h265 encoder. [00:28:40 CET] <kepstin> k4m3nk0: run "vainfo" and see if there's a line that says "VAProfileHEVCMain : VAEntrypointEncSlice" [00:30:04 CET] <kepstin> k4m3nk0: note that the results using the hardware encoder won't be particularly great, depending on what your input files are it might be better to leave them as-is rather than re-encode. [00:30:16 CET] <kepstin> k4m3nk0: why do you want them in hevc anyways? [00:31:04 CET] <a8_> can i just do -pass 2 instead of -pass 1 and then do -pass 2 [00:31:37 CET] <kepstin> a8_: no [00:31:47 CET] <kepstin> a8_: pass 2 works by reading the log file generated by pass 1 [00:31:52 CET] <a8_> ok [00:31:56 CET] <k4m3nk0> i just did -pass 2 at the end [00:31:58 CET] <k4m3nk0> and it works [00:32:38 CET] <a8_> probs didnt work right [00:32:55 CET] <kepstin> if you do -pass 2 at the end (after the output filename) it's ignored and does nothing. [00:33:23 CET] <kepstin> (it will print a helpful warning "Trailing options were found on the commandline." if you do that) [00:33:59 CET] <a8_> ok thanks [00:34:03 CET] <a8_> im a noob with ffmpeg sorry [00:34:05 CET] <a8_> but i get this [00:34:09 CET] <a8_> [NULL @ 0x7ffc3802bc00] Unable to find a suitable output format for '6M' [00:34:09 CET] <a8_> 6M: Invalid argument [00:34:27 CET] <a8_> i think i did something wrong with -b:v 6M [00:35:03 CET] <kepstin> you have a typo somewhere in your command line, yes [00:35:45 CET] <kepstin> that usually means that you have an option before the -b:v which is missing its value, so it eats the -b:v as a value [00:35:58 CET] <kepstin> then the 6M on its own is treated as an output filename [00:36:11 CET] <a8_> all i have before -b:v is [00:36:13 CET] <a8_> ffmpeg -i h264Input.mp4 -c:v libx265 [00:36:37 CET] <kepstin> that can't be true, because if it was it would work :) [00:36:40 CET] <kepstin> are you missing a space? [00:36:41 CET] <a8_> my entire command is [00:36:42 CET] <a8_> ffmpeg -i h264Input.mp4 -c:v libx265 -b:v 6M -c:a copy -pass 1 -threads 2 -preset veryslow h265output.mp4 [00:37:25 CET] <kepstin> the command as you pasted it is correct. [00:37:38 CET] <kepstin> try re-copying it from icr back to your terminal? :) [00:37:41 CET] <kepstin> from irc* [00:40:14 CET] <a8_> it calls my output an invalid argument [00:40:26 CET] <a8_> not -output [00:40:33 CET] <a8_> ok [00:41:10 CET] <kepstin> i just copy/pasted your command into a terminal and ran it (after making a file named h264Input.mp4) and it worked perfectly. [00:41:36 CET] <a8_> https://pastebin.com/eiVGxL7U [00:41:39 CET] <kepstin> so if you have a problem, it's in the stuff that you're not sharing with us [00:41:47 CET] <a8_> here is my entire thing [00:42:08 CET] <kepstin> your output filename is missing the .mp4 extension [00:42:20 CET] <a8_> oh [00:42:22 CET] <a8_> thanks [00:42:36 CET] <a8_> heyyy its working [00:42:39 CET] <a8_> thanks so mutch [00:43:55 CET] <a8_> im gonna change it to 4 threds as i have 4 core cpu [00:45:11 CET] <kepstin> libx265 defaults to using all cpu cores if you don't pass the "-threads" option. [00:45:25 CET] <a8_> i will put it there just to be safe [00:46:27 CET] <kepstin> (actually, hmm, it doesn't actually manage to use all the cores/threads on my 16 thread ryzen box, but that's probably because my test video isn't high enough resolution) [00:46:53 CET] <a8_> is there a way to make ffmpeg print a ETA of when it will be done? btw thanks for all the help so far [00:47:45 CET] <kepstin> no, ffmpeg doesn't do etas (a mostly technical limitation, since filters and things can change the length of the video in ways that can't be determined in advance) [00:48:09 CET] <kepstin> if you're doing a simple transcode you can figure it out yourself using the speed value if you know the length of the original file [00:48:54 CET] <kepstin> hint: encoding h265 is very slow, so the eta will be far in the future :) [00:57:40 CET] <a8_> it is 0.7% speed [00:57:52 CET] <a8_> video is10min [00:57:58 CET] <a8_> so 70min encode? [01:09:37 CET] <Hello71> I don't think that's how math works [01:15:53 CET] <a8_> 0.7 * 10 = 70 [01:16:01 CET] <a8_> oh [01:16:08 CET] <a8_> so its 7 min encode? [01:16:28 CET] <a8_> but i spent 22min on the encode [01:18:00 CET] <a8_> and its still goin [01:21:55 CET] <k4m3nk0> hi [01:22:12 CET] <k4m3nk0> i made a script for converting all movies from folder using ffmpeg [01:22:17 CET] <k4m3nk0> what you guys think [01:22:23 CET] <k4m3nk0> #!/bin/bash for name in *.mkv; do ffmpeg -i "$name" -c:v libx265 -b:v 6M -c:a copy "${name%.*}.mp4" -pass 1 -threads 2 -preset veryslow ; ffmpeg -i "$name" -c:v libx265 -b:v 6M -c:a copy "${name%.*}.mp4" -pass 2 -threads 2 -preset veryslow ; done [02:27:25 CET] <Hello71> > 0.7 * 10 = 70 [02:27:33 CET] <Hello71> we need that in the topic or something [12:14:30 CET] <celebrate> Hey everyone [12:14:49 CET] <celebrate> If anyone can help me translate this to fluent ffmpeg syntax i'd be forever greatful! [12:14:50 CET] <celebrate> let command = ffmpeg() .input(tempFilePath) .size('640x?') .input(watermarkFilePath) .complexFilter([ "[0:v][1:v]overlay=W-w-20:H-h-20" ]) .videoBitrate(2000) .videoCodec('libx264') .format('mp4') .output(targetTempFilePath) [12:15:26 CET] <celebrate> When trying to run it this way getting an error [12:15:27 CET] <celebrate> ffmpeg exited with code 1: Filtergraph 'scale=w=640:h=trunc(ow/a/2)*2' was specified through the -vf/-af/-filter option for output stream 0:0, which is fed from a complex filtergraph.-vf/-af/-filter and -filter_complex cannot be used together for the same stream [12:16:09 CET] <JEEB> you do understand that any JS wrapper over the cli (or API) has nothing to do with FFmpeg itself? [12:16:33 CET] <JEEB> anyways, that error just says that something in your thing is adding -vf or -af into the command line, and then you have a complex filter chain :P [12:16:59 CET] <JEEB> which cannot be utilized together since vf is a simplified version of the complex filter chain, pretty much [12:24:53 CET] <celebrate> yeah i get that [12:25:15 CET] <celebrate> but what do you mean has nothing to do with the ffmpeg itself? [12:25:35 CET] <celebrate> you're saying if i do the exact same "process" through the command string it won't work either? [12:28:56 CET] <JEEB> celebrate: simple: the Javascript thingy you're seemingly using has *nothing* to do with FFmpeg itself [12:29:05 CET] <JEEB> so nobody here can help you with that syntax [12:29:37 CET] <JEEB> but yes if you just take the result of whatever wrapper that is outputting as arguments you will end up in the same thing :P since what that wrapper now has let you do is invalid [12:30:00 CET] <JEEB> probably the scale part ends up as -vf (because in 99% of cases that's an easy way to scale all outputs to that size) [12:30:29 CET] <JEEB> yet then you have the complex filter chain, and at that point the ffmpeg.c command line app goes "nope, that's not how you do it" [12:31:21 CET] <JEEB> in other words, whatever you're trying to do within a vf/af should end up in the complex filter chain if you're doing a complex filter chain [12:36:44 CET] <celebrate> ah i see [12:38:26 CET] <celebrate> thanks jeeb :) [12:51:21 CET] <celebrate> This is getting me an error [12:51:22 CET] <celebrate> .complexFilter([ "[0:v]scale=320:-1[1:v]overlay=W-w-20:H-h-20" ]) [12:51:35 CET] <celebrate> Invalid argument [12:51:36 CET] <celebrate> ? [12:53:19 CET] <JEEB> you should look up some examples of complex filter chains :P https://trac.ffmpeg.org/wiki/FilteringGuide f.ex. [14:24:55 CET] <HickHoward> so [14:25:12 CET] <HickHoward> i got into a emulation thread on 4chan's /vg/ board by accident [14:26:42 CET] <HickHoward> fuck [14:26:43 CET] <HickHoward> wrong thread [19:50:42 CET] <bencc1> I'm capturing screen and pulseaudio with ffmpeg [19:51:21 CET] <bencc1> mediainfo tells me that video has 30 constant fps and audio has 36.875 FPS [19:51:22 CET] <bencc1> https://pastebin.com/raw/njgT1pRd [19:51:31 CET] <bencc1> what does audio FPS means? [19:51:54 CET] <JEEB> whatever mediainfo means with that :D [19:52:01 CET] <bencc1> can it cause audio/video sync issues when editing in Adobe Premiere? [19:52:21 CET] <bencc1> so how can I check with ffprobe? [19:52:46 CET] <bencc1> I don't have Adobe Premiere and I don't know how to use it but a video editor told me that there is a audio/video sync issue [19:52:59 CET] <bencc1> although I'm using constant framerate [19:53:35 CET] <bencc1> ffmpeg -f pulse -ac 2 -ar 48000 -i default -f x11grab -framerate 30 -i :0.0 -acodec aac -b:a 96k -vcodec libx264 -pix_fmt yuv420p -preset:v veryfast -crf 23 -vf fps=30 cap.nut [19:53:48 CET] <bencc1> am I missing something for constant framerate or audio/video sync? [19:54:06 CET] <JEEB> also that frame rate looking at that seems like it's just audio frames (decoder units) per second :P it says 1024 SPF there which means that the packets are 1024 samples [19:55:14 CET] <JEEB> bencc1: it can be a whole lot of stuff. at the end of the day my intermediates are generally some lossless format in AVI (Ut Video generally). it gets big, but you can mitigate it a bit in transfer. [19:55:37 CET] <JEEB> bencc1: like it could just dislike edit lists which are used to handle b-frames or encoder delay [19:56:04 CET] <JEEB> and that's not "edit lists are evil" but rather "ugh why do these people let these borked implementations out here" [19:56:18 CET] <JEEB> it could of course be something like "it doesn't like b-frames pronto" [19:57:06 CET] <bencc1> most of my users don't need to edit so I don't serve raw [19:57:24 CET] <JEEB> yea, I wonder people conflate lossless with raw :P [19:57:38 CET] <JEEB> last time I asked for a lossless copy for a thing I got rar'd raw RGB in mov [19:57:44 CET] <bencc1> sorry, lossless [19:58:25 CET] <JEEB> anyways, the main thing would be able to iterate on it :P [19:58:30 CET] <JEEB> and you can't unless you have the editor at hand [19:58:42 CET] <JEEB> otherwise you can at most start guesstimating and have no idea if it helps or not [19:58:56 CET] <bencc1> I'll try installing a trial and make simple edits [19:59:16 CET] <bencc1> sure but there are probably common parameters to check [19:59:28 CET] <JEEB> you wish [19:59:42 CET] <bencc1> when I view the mp4 in vlc the a/v sync is fine [19:59:46 CET] <JEEB> nobody in that side seems to actually want to know the "common requirements" of this crap [19:59:53 CET] <JEEB> instead the whack at buttons and see if something works [19:59:58 CET] <JEEB> which I understand, but it's unfortunate [19:59:59 CET] <JEEB> :P [20:00:20 CET] <JEEB> and it probably even depends on the version of the editor in the worst case [20:00:34 CET] <JEEB> which corporation's licensed demux/decoding stuff it uses etc [20:04:53 CET] <bencc1> ffmpeg -i input.mp4 -vf vfrdet -f null - [20:05:02 CET] <bencc1> VFR:0.000000 (0/257532) [20:05:25 CET] <bencc1> that means that I have perfect constant framerate? [20:05:40 CET] <JEEB> you can check that with any MP4 thing, like L-SMASH's boxdumper with its timestamp dumping option [20:05:45 CET] <JEEB> or ffprobe's show_packets [20:05:57 CET] <JEEB> but I will guesstimate that your timestamps are OK :P [20:05:58 CET] <bencc1> check what? how? [20:06:23 CET] <bencc1> check each packet with "ffprobe show_packets"? [20:06:28 CET] <bencc1> manually? :) [20:06:41 CET] <JEEB> last time I checked it was with -of json and a short python script [20:06:53 CET] <JEEB> which looked at the timestamps and checked how they raised [20:07:19 CET] <bencc1> I'll search [20:07:34 CET] <bencc1> but the problem is probably somewhere else [20:07:42 CET] <JEEB> I would guesstimate that your timestamps are OK [20:07:58 CET] <JEEB> unless there indeed are some random differences [20:08:13 CET] <JEEB> like a large diff between initial timestamps or something [20:11:38 CET] <bencc1> if I have random differences it will make the video editor angry [20:11:50 CET] <JEEB> sure [20:12:22 CET] <JEEB> but at most I think you would have different start times of the time line, and possibly inability the piece of shit of handling edit lists [20:12:49 CET] <JEEB> but yea, better check if you have the editor available what sort of desync it is first of all [20:12:58 CET] <JEEB> might give hints on wtf it's not handling [20:13:28 CET] <bencc1> all answers I can find are just about how to get the fps metadata [20:13:44 CET] <bencc1> I thought that this "ffmpeg -i input.mp4 -vf vfrdet -f null -" is a good way to verify [20:13:58 CET] <bencc1> it actually check all frames [20:14:50 CET] <JEEB> vfrdet I think it was to detect VFR content in a CFR stream or so [20:15:15 CET] <JEEB> you want the actual timestamps in the container, which is either L-SMASH's boxdumper, or ffprobe's -show_packets -of json [20:15:55 CET] <bencc1> ok [20:16:03 CET] <bencc1> maybe I need asyncts during capture [20:16:31 CET] <JEEB> you're thinking too much [20:16:42 CET] <JEEB> stop, and wait until you have the means to repro [20:17:02 CET] <bencc1> I still not sure how to repro [20:17:14 CET] <JEEB> having the editor and seeing how it ends up desync? [20:17:39 CET] <JEEB> then just validate that the timestamps in the mp4 are OK, and after that pretty much *any* source with similar output parameters should be possible to get to desync [20:18:19 CET] <JEEB> but I see no reason to help you further until you can easily pass things through some test which gives you green or red light on the A/V sync [20:18:37 CET] <JEEB> you have been given the tools to check the timestamps, to validate them with two tools at least [20:18:41 CET] <JEEB> boxdumper and ffprobe [20:19:22 CET] <bencc1> ffprobe gives me the frame timestamps. I still need to find a script that verify the framerate [20:19:46 CET] <JEEB> ffprobe -i BLAH.mp4 -of json -show_packets > BLAH.json [20:20:05 CET] <JEEB> then `from json import load` (I think load is "load file" and loads is "load string") [20:20:13 CET] <JEEB> then you have a dictionary [20:21:04 CET] <_Fremen_> hello everyone [20:21:39 CET] <JEEB> it will have packets for all streams there so you can have like a dictionary or whatever for each stream_index there and just make the script yell when the diff for a specific stream index is not static [20:21:44 CET] <JEEB> also print out the initial PTS or so [20:22:24 CET] <_Fremen_> I am sorry if this is on-topic for this channel but I wanted to ask this to a person with experience in mp3 or media files in general. [20:23:04 CET] <bencc1> JEEB: thanks. I'll try it [20:23:52 CET] <_Fremen_> I have two different mp3 files of the same music and their properties seems to same on Smplayer information tab, but their sizes is different. What could be the cause of this and is it usually the better choice to keep larger one? [20:25:06 CET] <_Fremen_> This question can be applied to every media format, same properties and etc. but different sizes. [20:25:17 CET] <_Fremen_> will be back in 20 minutes [20:36:24 CET] <_Fremen_> anyway, after better searching, I find mediainfo software and will try it out. Have a nice day [23:15:46 CET] <mindlight> Hi all. Downloaded nightly for Windows. Trying to deinterlae with "-vf yadif" but the resulting video file runs nicely but after a while (less than a minute) it just freezes the video. The audo continues. I have tried to google but can't find anything about yadif problems like this. I have tried with only "-vf yadif" and it still freezes. [23:16:36 CET] <mindlight> any known problems or things to think about when doing deinterlace / yadif? [23:20:24 CET] <JEEB> mindlight: nope [23:22:05 CET] <mindlight> thanks JEEB.... and of course... 2 hours of googling and 3 minutes after I wrote here I found a clue to what's causing this. Testing parameters now. Most likely I will be looking like an idiot in a short while... :-) [23:23:41 CET] <mindlight> Yeah... the "-t 00:00:30" parameter was specified before the -i. The resulting file was running perfectly for 30 secs and then with frozen video and ok audio for another 30 secs...Solution: move the "-t" parameter to after the "-i" :-) [00:00:00 CET] --- Thu Dec 26 2019

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ffmpeg-devel.log.20191224
by burek 25 Dec '19

25 Dec '19

[00:00:00 CET] --- Tue Dec 24 2019

1 0

ffmpeg.log.20191224
by burek 25 Dec '19

25 Dec '19

[00:02:21 CET] <cehoyos> A good starting point is likely to remove "-ar 192k" [00:07:15 CET] <LFSVeteran> but then it won't sample at 192k [00:07:26 CET] <LFSVeteran> 192kHz is a must [00:37:09 CET] <cehoyos> I wonder if your hardware (that doesn't encode mp3 in realtime) is really able to record 192k, more likely is that you request alsa to resample to 192k (which takes cpu cycles), later it has to be resampled again because mp3 does not support 192k [00:38:02 CET] <cehoyos> maximum is 48k [00:46:56 CET] <Hello71> but the sound is so much warmer [00:51:31 CET] <LFSVeteran> mp3 doesn't support 192kHz....ok mp3 isn't option then [00:52:15 CET] <LFSVeteran> I guess it's resampling [01:46:05 CET] <cehoyos> Hello71: You know that the human ear is limited to 48kHz 16bit? [01:46:31 CET] <Hello71> but, like, the sound is crisper [01:46:41 CET] <LFSVeteran> not using it for audio... [01:46:43 CET] <Hello71> obviously you don't have the requisite diamond directional ethernet cables [01:47:14 CET] <LFSVeteran> :D [01:47:29 CET] <LFSVeteran> the usual audio nonsens [02:22:41 CET] <kepstin> if you're not using it for audio, you shouldn't be using a lossy audio codec [02:23:21 CET] <kepstin> lossy audio codecs remove stuff that humans are unlikely to perceive, but if you're not working with audio, that "stuff" might be important signal [02:24:08 CET] <LFSVeteran> My goal is to sample at 192kHz and stream the signal to the world [02:24:37 CET] <LFSVeteran> icecast/darkice wasn't the option [02:24:48 CET] <LFSVeteran> maybe icecast/ffmpeg can do the job [02:26:44 CET] <LFSVeteran> don't know if ogg/vorbis would be an option [02:41:12 CET] <Hello71> but I would still be skeptical that this "embedded" can actually sample at 192k [02:51:53 CET] <kepstin> well, if you have a signal generator you can trust which can generate signals up to 96kHz, you could test it I guess? [02:57:46 CET] <Hello71> reminds me of the guy who wanted to calibrate his speakers with an uncalibrated microphone [03:08:13 CET] <kepstin> a lot of modern hifi systems actually ship with a microphone for doing speaker/room calibration, presumably the mic is built to spec and a "typical" response curve is stored in the hifi system firmware. [03:08:25 CET] <kepstin> mine has one, but I haven't bothered trying it. [03:14:11 CET] <Hello71> wouldn't such systems just come with calibrated speakers [03:14:31 CET] <Hello71> or do you mean the sound boxes are sold separately from the speakers [03:14:59 CET] <nicolas17> Hello71: maybe stuff that depends on speaker distance and such? [03:15:06 CET] <Hello71> right [03:15:13 CET] <Hello71> but my point is you wouldn't be calibrating the speakers [03:15:21 CET] <nicolas17> "room calibration" [03:25:27 CET] <AiNA_TE> does ffmpeg support burning picture based subs? [03:25:34 CET] <AiNA_TE> or do i need to ocr them first [03:30:40 CET] <nicolas17> AiNA_TE: do have subtitles already in a picture format and you want to burn them into the video stream? [03:31:52 CET] <AiNA_TE> yeah, they are vobsub as .idx/.sub [03:34:26 CET] <nicolas17> hmm I think you can use the overlay filter [03:53:21 CET] <igssl> https://pastebin.com/DL5LEnQR any idea what i'm doing wrong? trying to batch convert + upscale some webms to 1080p MP4s on windows [04:00:22 CET] <AiNA_TE> igssl, dont use %04d.webm as input [04:00:46 CET] <AiNA_TE> if your doing this in windows cli, either explicitly state input each time [04:00:50 CET] <AiNA_TE> or use a for loop [04:03:45 CET] <AiNA_TE> for %%a in ("*.webm") do (ffmpeg -i "%%a" -vf scale=1920x1080:flags=lanczos -c:v libx264 -crf 16 -preset slow -c:a copy %%a.mp4 ) [04:03:51 CET] <AiNA_TE> somthing like that [04:04:30 CET] <AiNA_TE> can you load the scaler like that also? [04:05:05 CET] <AiNA_TE> i always use "sws_flags spline" for example, outside of -vf [04:07:04 CET] <nicolas17> igssl: ffmpeg can't do "batch conversions", the %d stuff is to do stuff like load multiple images as frames and encode them into a single video [04:07:22 CET] <nicolas17> you have to make a script that runs ffmpeg separately for each file [04:08:10 CET] <igssl> well, FOR /F "tokens=*" %G IN ('dir /b *.webm') DO ffmpeg -i "%G" -vf scale=1920x1080:flags=lanczos -c:v libx264 -crf 16 -preset slow -c:a copy "%~nG.mp4" seems to be working [04:09:00 CET] <AiNA_TE> you should put your command after DO in () brackets [04:09:20 CET] <igssl> i am somewhat surprised at lanczos being better than every AI video rescaler i have tried, btw... waifu2x, topaz etc. all produce inferior results for these video files (overwatch gameplay) [04:09:36 CET] <nicolas17> AiNA_TE: the bracket-less way is how it worked in DOS so I guess it still works in modern systems :) [04:09:45 CET] <igssl> there isn't a better scaler i'm missing, is there? i know stuff like MadVR uses some kind of neural net jazz [04:10:01 CET] <AiNA_TE> i always use spline and like it's results [04:10:10 CET] <AiNA_TE> but diffrent scalers are better in diff situations [04:10:30 CET] <AiNA_TE> like waifu2x you mention is great for anime, but not live action for example [04:10:55 CET] <igssl> it has photo model too [04:12:22 CET] <nicolas17> waifu2x uses a neural network, right? [04:12:39 CET] <igssl> yep [04:12:55 CET] <nicolas17> I think for gaming you'd have to train it yourself with content from that particular game :P [04:58:59 CET] <AiNA_TE> nicolas17 i am playing around with the overlay filter but im getting the error "Filter overlay has an unconnected output" [04:59:02 CET] <AiNA_TE> you got any ideas [05:01:27 CET] <AiNA_TE> https://pastebin.com/raw/ubuG3W9b [05:01:38 CET] <analogical> how do I remove cover art from an audiofile?? [05:03:45 CET] <relaxed> analogical: ffmpeg -i input -map 0:a -c copy output [05:03:50 CET] <AiNA_TE> ffmpeg -i in.mp3 -vn -codec:a copy -map_metadata -1 out.mp3 [05:35:48 CET] <AiNA_TE> i worked it out, thx for pointing me in the right direction nicolas17 [10:02:19 CET] <Haemogoblin> Morning [10:03:02 CET] <Haemogoblin> Would here be a good place to pose a question about finding an old 32bit version of ffmpeg? [10:14:42 CET] <fling> Haemogoblin: how old? [10:16:02 CET] <Haemogoblin> old enough to function with audacity 2.0.6 [10:16:43 CET] <Haemogoblin> on a Apple mac running OS 10.4 [10:17:58 CET] <Haemogoblin> I've tried the 32bit 2.2.2-mac zip but audacity doesn't see the file and identify it is present [10:26:15 CET] <Haemogoblin> the audacity wiki led me to the ffmpeg project page, when I saw there was an IRC channel. I thought it might be worth a shot asking around. [10:28:32 CET] <Haemogoblin> I'm afraid I don't actually know which version of FFMPEG complimented version 2.0.6 of Audacity. However the build date is 2014, so I would imagine a version of FFMPEG from that time? [11:18:30 CET] <Haemogoblin> Hello? [11:58:18 CET] <kurosu> barhom: regarding your question in -devel that should have been in this channel, is it pixelation with some kind of pattern (GPU texture format or whatever at play) or too low a bitrate? [11:58:55 CET] <barhom> kurosu: Pixelation and, how do I explain, repeating frames - does that make sense? [11:59:09 CET] <barhom> Like its stuck between two frames jumping back and forth and then after 2 seconds it continues [11:59:30 CET] <kurosu> ok, maybe indeed some kind of data transfert issue [11:59:43 CET] <kurosu> or are you hitting some kind of profile+level bound in your decoder ? [12:00:15 CET] <kurosu> not like I think I can help you fix this, but what are is the level, profile and bitrate of the resulting video ? [12:01:09 CET] <barhom> Im transcoding with nvenc fast 900k, I havent specified level/profile [12:01:20 CET] <barhom> its a SD live transcode, 720x576 mpeg2 input source [12:25:13 CET] <kurosu> BtbN: ^ (also, output seems to be h264_nvenc, and I do hope pixelation isn't in the input) [12:27:43 CET] <BtbN> Are you swapping channels or am I missing messages? [12:28:51 CET] <BtbN> 0.9M encoding with nvenc won't look good though. Thats way too low of a bitrate. And why use fast? There's virtually no reason to ever use something other than the slowest preset. [12:29:19 CET] <JEEB> basically technically his stuff was end user discussion so technically kurosu was correct to pull him over here [12:37:24 CET] <barhom> JEEB: If we think its a bug should the discussion go here or in -devel? (I generally get way better answers in -devel) [12:38:00 CET] <barhom> kurosu: I found the bug, the problem was decoding on GPU is creating the pixelation [12:38:28 CET] <JEEB> well yea, unfortunately many developers don't want to be anywhere near users [12:38:35 CET] <JEEB> it's just masochists like me who are in here :P [13:02:50 CET] <kurosu> so it's true some hardware decoders have issue when they do not have enough to do/the resolution is too low [13:02:59 CET] <kurosu> *issues [13:03:31 CET] <kurosu> but mpeg2 & pal SD should be a pretty common place [13:03:31 CET] <BtbN> Hardware Decoders are generally perfect though [13:03:44 CET] <JEEB> the issue could be anywhere between the hwdec wrapper, driver, hardware and glue code [13:03:48 CET] <BtbN> There's a defined result for how to decode video. Not like encoders, where there is a quality difference [13:04:15 CET] <kurosu> thanks for explaining this to me :D [13:04:34 CET] <kurosu> yeah sorry, I meant, I did hear of that before [13:04:49 CET] <kurosu> not trying to shift blame somewhere/to someone [13:59:29 CET] <barhom> BtbN: Btw, when you said why use fast vs slowest. It is to cram in as much GPU transcodes as possible into 1 GPU at the cost of quality [14:00:08 CET] <BtbN> It's probably more cost effective to buy CPUs for that [14:07:39 CET] <barhom> BtbN: I decided on GPU after calculating costs. I ofc like x264 better, but it all depends on what the intended purpose is. [14:08:20 CET] <BtbN> You need a 2k¬+ GPU for more than two parallel sessions anyway [14:08:33 CET] <BtbN> and for 2k¬ you get a mad amount of CPU cores [14:08:53 CET] <barhom> The quadro p2000 isnt that expensive [14:09:55 CET] <barhom> p2000 combined with a cheap e3-v2 cpu [14:10:39 CET] <barhom> But I'm all ears, used hardware is no problem, what CPU are you thinking is the most cost effective? [14:10:44 CET] <DHE> v2? that's like 3 generations behind.. [14:12:19 CET] <barhom> DHE: Yeh but you get them cheaply. ¬200 for e3-1270v2 with 16gb ram. Slam in a quadro p2000 and you got a good encoding machine. [14:19:15 CET] <DHE> well, I've heard good things about the nvenc chips on Turing cards, and I did just happen to get an rtx2060 series card... but I would still want to test and validate it. I'd done that for older model cards though... [14:27:46 CET] <Abdullah> Is it possible to set an image with mp3 for first some minutes, and then another image for some minutes? [14:27:59 CET] <Abdullah> thus creating mp4 from them? [14:28:49 CET] <BtbN> Something like a server with Dual EPYC 7302. 32 cores, 64 thread. [14:29:09 CET] <BtbN> You can run a ton of lower quality x264 encodes on that. [14:29:18 CET] <DHE> can confirm (though I'm using intel right now) [14:29:36 CET] <BtbN> 32 Intel Cores cost slightly more than 2k¬ though [14:29:40 CET] <Abdullah> I'm using this command. ffmpeg -loop 1 -r 1 -i input.png -i input.mp3 -vcodec libx264 -acodec copy -shortest output.mp4 [14:29:53 CET] <DHE> 28 cores cost ~10,000 USD retail... [14:30:07 CET] <BtbN> Yeah, AMD wipes the floor with Intel then [14:30:13 CET] <BtbN> a 16 core Rome Epyc is 1k [14:30:25 CET] <Abdullah> Is it possible to set an image with mp3 for first some minutes, and then another image for some minutes? [14:30:46 CET] <DHE> oh absolutely, and it'll have more cache (probably?) and more memory channels and I believe higher speed memory support and more PCI-E lanes (not relevant to encoding unless you want 8 GPUs in a system) etc [14:31:10 CET] <BtbN> 128MB L3 [14:31:15 CET] <BtbN> 8 Channel memory [14:31:21 CET] <BtbN> 128 PCIe lanes [14:31:35 CET] <BtbN> PCIe 4.0 lanes even [14:32:47 CET] <Abdullah> offtopic ? [14:32:57 CET] <Abdullah> Is it possible to set an image with mp3 for first some minutes, and then another image for some minutes? [17:57:33 CET] <familiyaF> Hi, I want to mux SCC to mxf container as vanc. Is it possible to achieve the requirement via ffmpeg [18:00:06 CET] <JEEB> SCC is the text format with 608 captions? [18:00:17 CET] <familiyaF> yes [18:00:55 CET] <JEEB> also VANC as in actually in the video stream, or in f.ex. H.264 or MPEG-2 Video stream as separate packets? I do also know that MXF supports US captions as a separate stream? [18:02:03 CET] <johnjay> kepstin: interesting this video happened to come up for me: https://www.youtube.com/watch?v=3XDyQnY5GHI [18:02:21 CET] <familiyaF> No its not embedded in the video as side car. I have a SCC file which I want to mux in mxf container [18:02:54 CET] <JEEB> no I mean how do you want it exactly? or you do not care as long as it's in the MXF? [18:05:17 CET] <familiyaF> I want to insert it as a seperate stream within mxf [18:06:05 CET] <JEEB> oh wow [18:06:09 CET] <JEEB> we actually have an SCC reader now? :o [18:06:24 CET] <JEEB> I mean the format isn't too complex [18:06:34 CET] <JEEB> just caption packets with timecodes [18:07:03 CET] <JEEB> but yea, then you should be able to mux that into MXF unless the MXF muxer hasn't been updated accordingly [18:07:08 CET] <JEEB> the MXF reader at least supports caption tracks [18:09:07 CET] <familiyaF> I think we do have SCC reader in ffmpeg [18:09:56 CET] <JEEB> yes, that is what I was kind of surprised at :P [18:10:19 CET] <JEEB> since I found sccdec.c having mention of the codec id [18:13:59 CET] <familiyaF> I guess then I would need to update mxf muxer to add support mux 608 and 708 into MXF [18:14:33 CET] <JEEB> well you can just try [18:14:37 CET] <JEEB> if it works it works [18:16:29 CET] <familiyaF> Thanks [19:57:17 CET] <aughey> Using hwaccel, is it possible to access the texture on the video card to render directly without having to transfer to the main memory then back to the gpu? [19:57:58 CET] <JEEB> in general yes [19:58:18 CET] <JEEB> might be that some interfaces don't let you do that, but looking at how things were implemented in mpv etc with using the libavcodec hwaccels [19:58:34 CET] <JEEB> you can get d3d9 textures from dxva2 and d3d11 textures from d3d11va f.ex. [19:58:57 CET] <JEEB> vaapi has some sort of EGL interop thing with GL [20:04:41 CET] <aughey> Thanks. I'll take a look at those resources to see if I can find anything. I might be able to double buffer the processing/ copying so that I can keep the gpu busy [20:58:15 CET] <wondiws> hi furq [23:46:21 CET] <a8_> i want to convert a folder full of videos into H265 with half the bitrate, but after doing some googleing i am confused [23:46:54 CET] <a8_> How do i make ffmpeg target a certen bitrate and try not to go above it? [23:49:07 CET] <a8_> i found documentation but it confused me [23:50:01 CET] <JEEB> the average bit rate over the whole clip is -b:v for video. how well that works and how many passes are required depends on the encoder [23:51:43 CET] <JEEB> -maxrate and -bufsize are when you are also caring about fluctuations all the time (the buffer of X bits to which encoded pictures are added and removed). mostly used to keep bandwidth in line with some bandwidth limitations (so that if you have X amount of bandwidth, you should always be able to buffer fast enough etc) [23:54:43 CET] <a8_> can i set -maxrate to the same as i set -b:v? [23:54:50 CET] <kepstin> there's usually no reason to use -maxrate and -bufsize unless you're dealing with specific hardware or specification limitations (like upload bandwidth on a low-latency live stream, a fixed-bandwidth broadcast tv channel, or a blu-ray drive's specified read speed) [23:55:03 CET] <a8_> ok thanks guys so mutch [23:56:21 CET] <JEEB> yea, that was what I was trying to say :P it's for keeping within an alotted amount of bandwidth all the time [23:57:15 CET] <kepstin> in general with most software encoders, if all you want is to have the output file to be a certain size, then you should set -b:v and do a 2-pass encode. [23:57:21 CET] <JEEB> while without it encoders such as libx264 (which have proper rate control) will utilize a lot more bits in the spots where they're needed, and a lot less where they're not needed [23:59:40 CET] <a8_> ffmpeg -i h264Input.mp4 -c:v libx265 -b:v 6 -output h265output.mp4 [23:59:42 CET] <a8_> is this good? [00:00:00 CET] --- Wed Dec 25 2019

1 0

ffmpeg-devel.log.20191223
by burek 24 Dec '19

24 Dec '19

[00:00:00 CET] --- Mon Dec 23 2019

1 0

ffmpeg.log.20191223
by burek 24 Dec '19

24 Dec '19

[00:00:34 CET] <nicolas17> your division gives 48 kilobytes, now multiply by 8 to get kilobits [00:03:17 CET] <shibboleth> BtbN, reason i asked is that even ffmpeg master doesn't list nvenc/nvdec [00:03:50 CET] <shibboleth> even if compiled with --enable-nvdec/nvenc [00:04:29 CET] <BtbN> They're not hwaccel modes. [00:06:20 CET] <shibboleth> -hwaccel nvdec/nvenc does work as an lias without having to specify in/out codecs for cuvid though [00:06:26 CET] <shibboleth> alias [01:27:47 CET] <wondiws> what format are those vobsub subtitles? "file" gives me MPEG sequence, v2, program multiplex [01:28:25 CET] <wondiws> vobsub is the same format as the original DVD subtitles, albeit repackaged, although I'm not even sure if they are actually repackaged [02:12:09 CET] <cehoyos> BtbN: I just tested and I can confirm that "-f image2 -loop 1" reads the input file again and again. [09:36:58 CET] <veek> how do i join and sync two .ts files that I download manually [09:37:17 CET] <veek> just cat'ing them doesn't work [09:40:27 CET] <veek> nm got it working [09:41:53 CET] <JEEB> how well concatenating two MPEG-TS things works depends on how completely wonky the timestamps are compared to each other. that said, most of the cases where I've had to repeat content or so, it's worked alright. depends on the playback software of course [09:56:31 CET] <veek> thanks [11:35:54 CET] <wondiws> does anybody know by any change how to extract the subtitle stream from a *.vob file in linux? I use mpegdemux to extract the AC3 sound, and in Windows I use VobSub or VSRip (which is the same I believe?) to extract the subtitles, but how do you do this in linux? [11:36:20 CET] <wondiws> I tried mpegdemux to extract the subtitle stream the same way as the audio stream, but the resulting file didn't make sense to me [13:49:43 CET] <cehoyos> wondiws: Your question is difficult to understand: dvd subtitles only make sense inside a vob file (an mpeg program stream), you can export them to a raw stream but as you found out, the resulting file is not very useful [13:50:14 CET] <pink_mist> I think he's asking for OCR stuff [13:50:26 CET] <pink_mist> iirc vobsubrip is an ocr thing [13:50:27 CET] <wondiws> pink_mist, no [13:50:34 CET] <wondiws> I don't want to do OCR [13:50:46 CET] <cehoyos> What do you want to do? [13:51:43 CET] <wondiws> the subtitles are monochrome pictures in the MPEG stream. Vobsub/VSRip (in Windows) extracts them from a MPEG2/VOB stream and puts it in a *.sub file, which is again an MPEG2 stream, but with just the subtitles in it [13:51:57 CET] <wondiws> is there a similar program for linux like Vobsub or Vsrip? [13:52:34 CET] <JEEB> we can read the idx+sub thing at least [13:52:35 CET] <wondiws> you can then mux the *.sub file into a mkv file for instance, or use it as an external file [13:52:37 CET] <cehoyos> You are in the FFmpeg support channel, why do you expect to get support for any other software? (FFmpeg is able to read and write vob files, and it sounds as if your sub file actually is a vob file) [13:52:54 CET] <JEEB> cehoyos: he wants to explain his needs in the words of that other software [13:52:56 CET] <wondiws> yes, sub is MPEG2/VOB format too [13:52:58 CET] <JEEB> not getting help with that other software [13:53:18 CET] <furq> there is no vobsub muxer [13:53:23 CET] <wondiws> JEEB but ffmpeg also has code for vobsub, am I right? [13:53:27 CET] <JEEB> only for reading [13:53:29 CET] <JEEB> unfortunately [13:53:29 CET] <furq> ^ [13:53:40 CET] <JEEB> see libavformat/mpeg.c [13:53:44 CET] <JEEB> that has idx reading etc [13:53:55 CET] <furq> tcdemux from transcode will do it [13:54:05 CET] <furq> and i think mencoder as well but idk how [13:54:10 CET] <wondiws> furq, what is tcdemux? [13:54:32 CET] <furq> https://linux.die.net/man/1/tcdemux [13:54:48 CET] <furq> it's part of transcode which isn't developed any more so it fell out of a lot of package managers [13:54:50 CET] <wondiws> I will have a look [13:55:26 CET] <furq> mkvtoolnix will also do it but only if you go via mkv first iirc [13:55:27 CET] <wondiws> furq, then I will compile it myself ;) [13:55:53 CET] <furq> ffmpeg will mux an mkv with just a dvd subtitle stream and then mkvextract can remux that to vobsub [13:55:54 CET] <JEEB> but yea, I'm surprised "sub" was actually an MPEG-2 PS structure :P I would have guesstimated it to be just the packets one after another without the unnecessary parts [13:55:58 CET] <furq> that's probably the least annoying way of doing it [13:55:59 CET] <wondiws> furq, mkvtoolnix also allows me to extract vobsub subtitles from a mkv file, yes [13:56:08 CET] <wondiws> but it's not in an mkv file, it's in a vob file ;) [13:56:45 CET] <pink_mist> but you wanted to put it in a mkv file? [13:56:46 CET] <wondiws> furq, that's not a bad way actually [13:56:57 CET] <wondiws> pink_mist, I might [13:57:06 CET] <pink_mist> afaik mkvtoolnix should support getting things from vob and put in a mkv directly [13:57:13 CET] <wondiws> pink_mist, ah, even better [13:57:34 CET] <wondiws> the idx file is just plain text, so that should be easy to convert [13:57:39 CET] <wondiws> just the timestamps [13:58:19 CET] <wondiws> or I suppose I could just delete everything from a VOB/MPEG2 stream apart from the subtitles :P [13:58:24 CET] <wondiws> and rename the file to *.sub [14:00:41 CET] <wondiws> mkvtoolnix doesn't see subtitles in a vob\ [14:03:27 CET] <furq> mkvtoolnix has a vobsub muxer [14:03:45 CET] <furq> mkvextract foo.mkv tracks 2:bar.idx [14:04:09 CET] <furq> or whatever track mkvmerge --identify says [14:04:52 CET] <furq> unfortunately it doesn't read dvd structures so you'll still need some intermediate tool [14:06:32 CET] <wondiws> furq, exactly [14:06:53 CET] <furq> nvm i misread what you siad [14:08:20 CET] <furq> one of these days i should strip out all the actual transcoding stuff from transcode and make a standalone tccat and tcdemux [14:08:37 CET] <furq> i'm sure it all has hard dependencies on a bunch of awful shit though [14:31:44 CET] <wondiws> furq, indeed, I can't compile transcode [14:31:59 CET] <wondiws> CODEC_ID_AC3 and CODEC_ID_MP2 undeclared [14:32:09 CET] <DHE> AV_CODEC_ID_AC3 ? [14:32:50 CET] <wondiws> DHE, probably, I assumed the definitions have changed as this code is from 2003 or something. I didn't know those constants by heart myself though [14:33:06 CET] <DHE> 2003? I think that's a new record [14:35:11 CET] <wondiws> does <libavcodec/opt.h> still exist? [14:35:28 CET] <DHE> no [14:40:53 CET] <furq> yeah this is probably why it's dropped out of package managers [14:42:14 CET] <void09> can someone explain to me again the difference between having -ss/-to before/after the input file path ? [14:42:41 CET] <void09> I cannot really figure it out from the docs explanation [14:42:47 CET] <wondiws> furq, I compiled it though, without ffmpeg support, sorry ;) [14:46:28 CET] <DHE> void09: before -i means it applies to the input, seeking before video decoding. after -i means it applies to the output meaning the output discards input until it arrives at the indicated point in time [14:46:37 CET] <void09> ohhhhhhhhhhhhhh [14:46:39 CET] <void09> that easy, lol [14:47:00 CET] <void09> so the after -i is more accurate [14:47:08 CET] <DHE> before -i can instantly seek anywhere, but lacks precision due to keyframe requirements. after -i means you'll have to wait while the input is decoded, but it can do far more precise seeks (as long as you're not doing stream-copy) [14:47:11 CET] <void09> but in case of well indexed .mkv files, it should be the same ?. [14:47:16 CET] <furq> wondiws: if tccat/tcdemux work without ffmpeg support then that's great news for me [14:47:28 CET] <furq> i assume they do but i've never looked into it [14:47:38 CET] <void09> DHE: my scenario is serving files over http to do frame accurate encoding of segments [14:47:51 CET] <void09> think i can do it with just -ss -to before -i ? [14:47:52 CET] <wondiws> furq, it seems it works, but I have never used this before, so I have to get acquianted [14:48:04 CET] <void09> so the whole file does not get downloaded for a tiny segment [14:48:06 CET] <DHE> void09: only if every frame is a keyframe because the seek MUST land on a keyframe [14:48:34 CET] <void09> no, they're not always keyframes :\ [14:48:41 CET] <void09> but does not the mkv index help with this ? [14:48:52 CET] <void09> to figure out exactly where each timestamp is ? [14:48:54 CET] <DHE> the index is really just a keyframe list [14:49:01 CET] <wondiws> furq, yes, I can extract ac3 audio with tcextract [14:49:15 CET] <DHE> you can't seek to anything except a keyframe because you can't play video without starting on a keyframe [14:49:28 CET] <void09> yes, but can't ffmpeg go to the keyframe before the timestamp, if it's before, and just decode the missing frames until it gets to the precise point ? [14:49:38 CET] <DHE> unless you do crazy shenanigans like play audio with no video (because audio tends to be 100% keyframes by this definition) [14:50:04 CET] <DHE> void09: I don't think this is a feature ffmpeg has, but you could implement it yourself in a custom application. [14:50:48 CET] <void09> so i have to download 30GB for 3 seconds :\ [14:52:25 CET] <DHE> or maybe get creative? [14:53:36 CET] <wondiws> furq, or maybe I should port VSRip to linux :P [14:53:48 CET] <void09> DHE: i don't feel too creative today [14:54:04 CET] <void09> i will test, maybe it somehow works [15:27:18 CET] <Hello71> -ss is now fast as an input option. [15:28:16 CET] <Hello71> according to the wiki. [16:05:32 CET] <DHE> -ss was always fast as an option option, but it lacks precision [16:08:52 CET] <Hello71> uh [16:08:55 CET] <Hello71> right, the other one [16:09:16 CET] <DHE> oh? [16:13:27 CET] <void09> I am doing a ffmpeg -ss -to -i http:// encode to x264 in .mp4.with mp4 i get the video length as reported by vlc/mpv as being the total length of the input. if i use .mkv i get the right output length (1 second) [16:13:32 CET] <void09> what am i missing ? [16:13:55 CET] <void09> is it a bug or a feature ? [16:29:32 CET] <Hello71> it is accurate now [16:32:31 CET] <void09> Hello71: -ss is fast as an output option ? [16:32:44 CET] <void09> relevant paragraph ? [17:23:56 CET] <void09> need help with quoting http input source, ' and " do not work [17:24:29 CET] <void09> no wary around having spaces but replacing with %20 ? [17:32:43 CET] <void09> https://trac.ffmpeg.org/ticket/2031 [17:32:51 CET] <void09> Status changed from closed to reopened [17:32:51 CET] <void09> The commit was reverted. [17:32:52 CET] <void09> :)) [17:42:31 CET] <johnjay> hrm. what's the reason consoles were 240p? [17:43:26 CET] <johnjay> the NES for example is 256x240. this video is saying it was to be on 483 interlaced tv? https://www.youtube.com/watch?v=zwDPx6hP_4Y [17:46:43 CET] <c0fe> i am running a command to convert m2ts file to mkv and i am getting the following error messages: [17:46:45 CET] <c0fe> Duplicate element [17:46:46 CET] <c0fe> Truncating packet of size 257103 to 4206 [17:47:02 CET] <c0fe> Duplicate element [17:47:12 CET] <c0fe> "/media/power/seagate/g/disc1p1/00001.m2ts: End of file" [17:47:32 CET] <c0fe> i am running ffmpeg -i "/media/power/seagate/g/disc1p1/00001.m2ts" -scodec copy -acodec copy -vcodec copy -f matroska "/media/power/seagate/g/jap03.mkv" [17:47:33 CET] <c0fe> why? [17:49:11 CET] <c0fe> well looks like that file is corrupt because the next file throws up No wav codec tag found for codec pcm_bluray [17:49:12 CET] <c0fe> av_interleaved_write_frame(): Invalid argument [17:49:12 CET] <c0fe> Error writing trailer [17:55:29 CET] <nate> johnjay: Why? perhaps because the NES came out in 1985 and we didn't exactly have 1080p displays being rolled out back then lol [17:55:51 CET] <johnjay> well i didn't understand his explanation in the video to save money [17:56:00 CET] <johnjay> why not do 480i if that's what they had [17:56:19 CET] <johnjay> and if they didn't then... well it's really 30fps then isn't it? [17:56:36 CET] <johnjay> because the tv is 30 interlaced frames even and 30 interlaced frames odd [18:02:22 CET] <nate> johnjay: Format standards are going to be the main factor in frame/refresh rate, keeping in mind NTSC (as it would have been during NES' days), PAL and SECAM. As for the somewhat halved-ish resolution of at the time, it was indeed probably hardware limitations I would believe [18:03:14 CET] <johnjay> so it would literally be 1 interlaced frame, then black, then 2nd interlaced frame, then black, etc ? [18:03:20 CET] <johnjay> i guess that makes sense [18:03:35 CET] <johnjay> for some reason i don't remember NES game i play as a kid as interlaced with black lines lol [18:07:46 CET] <kepstin> the nes (and similar) 240p output is actually not quite that [18:08:08 CET] <kepstin> it actually still outputs 60 fields per second, it's just that they're all field 1 [18:08:29 CET] <kepstin> with most tvs this would cause it to scan the same set of 240 scan lines repeatedly, 60 times a second. [18:09:26 CET] <kepstin> (tvs usually are a bit blurry so the scan lines from the first vs second field overlap a bit, so the missing lines weren't usually super noticable, particularly on smaller screens) [18:11:33 CET] <johnjay> maybe i don't understand the crt then [18:11:56 CET] <johnjay> i thought it was part of the tech that you do 30 lines odd then 30 lines even alternate to get interlaced [18:12:39 CET] <johnjay> if you can do 60 times per second the odd lines then why not do that for movies as well? [18:22:08 CET] <Hello71> kepstin: so... 120p? [18:22:37 CET] <kepstin> no, it's 240p instead of 480i [18:23:18 CET] <kepstin> with 480i, it does 240 lines marked as field 1, then 240 lines marked as field 2, alternating. The tv offsets to the ones marked as field 2 to fill in the gaps between the lines from field 1 [18:23:24 CET] <Hello71> uh, right [18:24:15 CET] <kepstin> with the 240p output, it sends 240 lines marked as field 1, then another 240 lines marked as field 1 again. The result is that the tv draws the second "field" over top of where the previous was drawn, making it effectively progressive. [18:24:35 CET] <kepstin> (i forget if it actually outputs field 1 or field 2, but that doesn't really matter for this) [18:25:58 CET] <kepstin> the downsides are that you have less vertical resolution (enough to be noticeably less detail in live action video) and depending on the tv, the gaps between lines might be noticable [18:29:25 CET] <kepstin> this also means that doing video capture of a nes tv output is a pain, you need equipment that can handle the out-of-spec signal and do something reasonable with it. [18:29:43 CET] <kepstin> some modern hdtvs can't accept it as an input signal, for example :/ [18:29:55 CET] <johnjay> hrm. so this is basically a human issue not a technical one? i.e. having constant black lines is fine for an 8-bit console but not for live action? [18:30:21 CET] <kepstin> the consoles were so low resolution anyways due to lack of ram that it wasn't an issue [18:30:53 CET] <kepstin> later consoles with more memory switched to doing proper 480i output to get higher resolution [18:32:02 CET] <kepstin> (some, like the ps2 and wii, could actually do 480p on supported games with appropriate cables and equipment) [18:32:18 CET] <kepstin> that would be 480p 30Hz [18:32:43 CET] <johnjay> yes i still have my ps2 and it has the rgb things. [18:32:47 CET] <johnjay> forget what they are called [18:33:08 CET] <kepstin> the three-cable rca jacks are component video, yeah [18:33:48 CET] <kepstin> the ps2 normally still did 480i over component, you had to use a specific game with progressive support and activate it with certain keypresses or menu config to actually get progressive video. [18:35:49 CET] <JEEB> I think like gran turismo or something supported even 1080i [18:35:54 CET] <JEEB> on ps2 [18:36:26 CET] <TanaPanda> composite [18:36:36 CET] <kepstin> huh, i wouldn't think the ps2 supported the pixel clock high enough to do that [18:36:37 CET] <TanaPanda> red white yellow is composite [18:36:45 CET] <TanaPanda> red white blue red green is component [18:36:45 CET] <kepstin> TanaPanda: single yellow is composite [18:36:49 CET] <kepstin> the red/white is audio [18:37:03 CET] <TanaPanda> yea but they usually come in a set [18:37:44 CET] <TanaPanda> the circle with like 8 pin is s-video [18:37:57 CET] <TanaPanda> coax or f connector [18:38:01 CET] <TanaPanda> HDMI [18:38:07 CET] <TanaPanda> Digital Optical [18:38:16 CET] <TanaPanda> anything else? [18:38:18 CET] <TanaPanda> Display port [18:39:47 CET] <kepstin> huh, wow, so there's apparently around 5 or 6 ps2 games that did support 1080i, including gran turismo 4. that's kinda neat [18:40:01 CET] <kepstin> and, uh, nhl 2004 for some reason [18:41:46 CET] <johnjay> kepstin: i'm looking closely at the snes wiki article. it's saying there's 7 modes, the first four are "progressive" at ~240 [18:41:52 CET] <johnjay> so shouldn't capturing snes output be also as bad? [18:42:04 CET] <kepstin> probably, i bet it depends on the game [18:42:23 CET] <kepstin> a lot of these console games also switch modes dynamically, which is even more fun [18:43:13 CET] <johnjay> that's what it says here. 7 modes total, with switching [18:44:56 CET] <kepstin> the ps2 games with progressive or hd modes usually switched back to 480i for menus and cutscenes [18:47:00 CET] <johnjay> interesting, didn't know that [21:19:01 CET] <LFSVeteran> almost there.... [21:19:13 CET] <LFSVeteran> need to find the right codec and format for ffmpeg -ar 192k -f alsa -i plughw:1,0 -codec copy -b:a 192k -f mpegts icecast://live:str3am@127.0.0.1:8000/stream [21:19:42 CET] <LFSVeteran> the plan is to stream at 192k from plughw:1,0 [21:28:27 CET] <cehoyos> codec copy does not like b:a (and mpegts most likely doesn't like what -f alsa provides) [21:35:17 CET] <Hello71> I mean, even if it's allowed, it's probably dumb [21:55:41 CET] <LFSVeteran> I know, but what would be a good choice as codec / format [21:56:07 CET] <LFSVeteran> so I can replace copy and mpegts [22:38:22 CET] <LFSVeteran> ok got it working with mp3: ffmpeg -ar 192k -f alsa -i plughw:1,0 -c:a libmp3lame -b:a 192k -content_type audio/mpeg -f mp3 icecast://live:str3am@127.0.0.1:8000/stream [22:38:41 CET] <LFSVeteran> but that means compression [22:39:03 CET] <LFSVeteran> is there a way without compression ? [22:39:42 CET] <BtbN> Streaming uncompressed audio seems like a pointless waste [22:41:31 CET] <LFSVeteran> compression takes time... [22:41:52 CET] <LFSVeteran> delay isn't a problem as long it will be a continous stream [22:42:19 CET] <LFSVeteran> now I have some stops in the stream [22:51:25 CET] <BtbN> that must be a really crappy CPU if it can't sustain a single mp3 encoder [22:51:57 CET] <LFSVeteran> embedded [22:52:25 CET] <LFSVeteran> not a high speed PC cpu [23:05:10 CET] <BtbN> Even a first gen RPi could probably encode mp3 in real time. [23:08:58 CET] <LFSVeteran> hmm, maybe encoding isn't the problem then... [23:09:08 CET] <LFSVeteran> have to dive into it [00:00:00 CET] --- Tue Dec 24 2019

1 0

ffmpeg-devel.log.20191222
by burek 23 Dec '19

23 Dec '19

[00:00:00 CET] --- Sun Dec 22 2019

1 0

ffmpeg.log.20191222
by burek 23 Dec '19

23 Dec '19

[03:25:50 CET] <causasui> I've been making 'infinite loop' videos with `-filter-complex "[0]reverse[r];[0][r]concat"` and it works well. However I have a new use case, where I want to make such a loop with just a segment of a video. but passing `-ss` or `-st` switches seems to cause the filter to be ignored. do I need to do a streamcopy step before I run this filter? [03:49:49 CET] <causasui> that seems to do it yes [04:12:33 CET] <causasui> now my problem is that when I do `-ss '20'` it seems to work but when I do `-ss '21'` or even `-ss '20.05'` there doesn't seem to be any video in the output [17:38:07 CET] <snooky> hi all [20:44:23 CET] <Protected> Good evening. I'm experiencing a baffling issue when creating mpeg-dash files (format 'dash'). According to the documentation, "The segment filename might contain (..) an ffmpeg-specific "$ext$" identifier (...) ffmpeg will replace (...) with muxing formats extensions such as mp4, webm etc.". I'm using that in the segment filename templates, but ffmpeg copies $ext$ to the filename verbatim. [20:44:23 CET] <Protected> Is that a really recent feature? I'm using ffmpeg 4.1.4 [22:03:46 CET] <raisen> How can I update the streaming video once source.png changes on this command: ffmpeg -f image2 -loop 1 -i source.png -f mpegts "udp://127.0.0.1:3333?listen" ? I tried to cat > source.png, rename but didn't work [22:11:11 CET] <BtbN> I don't think it will ever re-read the file. [22:12:15 CET] <raisen> BtbN: that makes sense why it wouldn't work. Is there a way to reread the file? [22:13:02 CET] <TheAMM> One idea is to create a named pipe and then write the image into it every time you want to send a new one [22:13:23 CET] <TheAMM> or maybe even something like (while :; do cat source.png: sleep 1; done) | ffmpeg -i - ... [22:13:33 CET] <TheAMM> even just* [22:13:48 CET] <raisen> hmm, let me try.. very beginning ffmpeg user here [22:23:15 CET] <raisen> hmm, ffmpeg is complaining that the output file is empty and no video is played when i run ffplay. I tried with: (while :; do cat source.png; sleep 1; done) | ffmpeg -i - -f mpegts "udp://127.0.0.1:3333?listen" [22:23:26 CET] <Hello71> mpng? lol [22:24:08 CET] <TheAMM> Try -f image2pipe -i - [22:24:27 CET] <TheAMM> can't guide you to the end though, good luck if you're very beginning user, maybe someone else has a moment [22:25:35 CET] <raisen> TheAMM: Thanks for your help. I got the same result. I'll post on the mailing list [23:25:15 CET] <Abbott> I am trying to concatenate video files and I am using this command: drew@yoga% ffmpeg -i rare1.mp4 -i black1.mp4 -i hunt.mp4 -i black1.mp4 -i dare.mp4 -filter_complex "[0:v:0][0:a:0][1:v:0][1:a:0][2:v:0][2:a:0][3:v:0][3:a:0][4:v:0][4:a:0][5:v:0][5:a:0]concat=n=5:v=1:a=1[outv][outa]" -map "[outv]" -map "[outa]" final.mp4 [23:26:01 CET] <Abbott> but I get an error: https://bpaste.net/raw/OAXFW [23:26:09 CET] <Abbott> how can I propery concatenate all these files [23:26:58 CET] <nicolas17> you're giving it 6 inputs: 0 1 2 3 4 5 [23:28:07 CET] <nicolas17> I don't think [5] corresponds to any actual input file there [23:28:12 CET] <Abbott> oh, you're right [23:28:26 CET] <Abbott> removing that I get paste that error, but then I get an error about the files being different resolutions [23:32:39 CET] <Abbott> ffmpeg -i rare1.mp4 -i black1.mp4 -i hunt.mp4 -i black1.mp4 -i dare.mp4 -filter_complex "[0:v:0][0:a:0][1:v:0][1:a:0][2:v:0][2:a:0][3:v:0][3:a:0][4:v:0][4:a:0][5:v:0][5:a:0]concat=n=5:v=1:a=1[outv][outa],scale=w=1280:h=720" -map "[outv]" -map "[outa]" final.mp4 [23:33:00 CET] <Abbott> that doesn't work, how would I add the scale filter to that command to scale all of the inputs before the concat filter [23:33:47 CET] <nicolas17> [0]scale=whatever[scaled0]; and then feed scaled0 instead of 0 to the concat filter [23:34:09 CET] <nicolas17> each video will need its own scale filter [23:34:13 CET] <nicolas17> well, the ones that need scaling [23:37:35 CET] <Abbott> I'm not sure where to put that [23:37:37 CET] <Abbott> ffmpeg -i rare1.mp4 -i black1.mp4 -i hunt.mp4 -i black1.mp4 -i dare.mp4 -filter_complex "[0]scale=w=1280:h=720[scaled0][1]scale=w=1280:h720[scaled1][2]scale=w=1280:h=720[scaled2][3]scale=w=1280:h=720[scaled3][4]scale=w=1280:h=720[scaled4],[scaled0:v:0][0:a:0][scaled1:v:0][1:a:0][scaled2:v:0][2:a:0][scaled3:v:0][3:a:0][scaled4:v:0][4:a:0]concat=n=5:v=1:a=1[outv][outa]" -map "[outv]" -map "[outa]" [23:37:39 CET] <Abbott> final.mp4 [23:39:32 CET] <nicolas17> you need to separate with semicolons [23:39:41 CET] <nicolas17> [inputs]filter[outputs];[inputs]filter[outputs];[inputs]filter[outputs] [23:43:07 CET] <shibboleth> is it normal for -hwaccel to only show cuda/cuvid and not nvdec/nvenc? [23:44:20 CET] <BtbN> cuvid is legacy, and I'm pretty sure at this point only maps directly to cuda. [23:51:16 CET] <Abbott> ok got it. thanks nicolas17 [23:58:54 CET] <Abbott> If I want to use 2 pass encoding to make a 1.44M file out of a 30 second clip, I figure I use a bitrate of 1440K/30s = 48k, right? but the file I got out was 184k [23:59:28 CET] <nicolas17> no, you're mixing bytes and bits [00:00:00 CET] --- Mon Dec 23 2019

1 0

ffmpeg-devel.log.20191221
by burek 22 Dec '19

22 Dec '19

[00:00:00 CET] --- Sat Dec 21 2019

1 0

ffmpeg.log.20191221
by burek 22 Dec '19

22 Dec '19

[03:38:11 CET] <shibboleth> any idea why -hwaccel cuda shows up in nvidia-smi but cuvid does not? [09:38:15 CET] <raisen> is it possible to create a live video stream from a single image file using ffmpeg? this image would be updated frequently which would update the video frame, otherwise the video would constantly use the last image [10:27:21 CET] <snooky> hi all [10:27:57 CET] <snooky> is there a simple command to convert a m2ts into a mkv and remove the subtitles at the same time? [10:37:03 CET] <cehoyos> snooky: What did you try? (ffmpeg -i input -map -0:s -c copy out.mkv) [10:37:34 CET] <snooky> i have copy my bluray to my hdd... now i want it to copy on my nas [10:37:49 CET] <snooky> but i have the structure with STREAM folder and the m2ts files [10:38:21 CET] <snooky> how i now can make from this m2ts remove the subtitles and create an new m2ts or mkv [11:01:23 CET] <furq> snooky: use mkvtoolnix for that and use the mpls files as input [11:01:40 CET] <furq> there's not necessarily a 1:1 mapping between bluray titles and m2ts files [11:01:52 CET] <furq> and also the chapters are in the mpls [11:04:48 CET] <snooky> when i open the m2ts files [11:04:58 CET] <snooky> i cant disable the subtitle in vlc [16:26:36 CET] <phobosoph> hi [16:43:29 CET] <phobosoph> anyone here? :) [16:43:33 CET] <phobosoph> I got an interesting issue with ffmpeg [16:44:00 CET] <pink_mist> loads of people here. just ask your question [16:44:36 CET] <phobosoph> OK! So I use FFMPEG to stream-copy a RTSP webcam feed to Youtube Live RTMP server [16:44:57 CET] <phobosoph> it works quite well, full fps and full hd - but after some hours the bandwidth and framerate in ffmpeg stats drop and youtube stops recognizing the stream [16:45:13 CET] <phobosoph> I have to kill + restart ffmpeg, ffmpeg doesn't notice any issues while streaming and exits [16:45:24 CET] <phobosoph> how can I further diagnose what goes wrong? [16:45:33 CET] <phobosoph> This happens every time, I restart ffmpeg, the stream goes up in Youtube again [16:45:38 CET] <phobosoph> then after some hours the stream stops working [16:45:51 CET] <phobosoph> Have to terminate ffmpeg and restart it manually [16:45:59 CET] <phobosoph> maybe I am missing an option? [16:46:06 CET] <pink_mist> first thing to make sure of: are you on the latest git ffmpeg? [16:46:18 CET] <JEEB> you can check with -debug_ts if frames/packets are still being passed [16:46:37 CET] <phobosoph> pink_mist: good point, I use a docker image with latest ffmpeg [16:46:38 CET] <JEEB> it will log a lot of stuff, but will show you if the actual API usage ends up stopping somewhere [16:46:52 CET] <JEEB> -v verbose also is useful for having a bit of extra logging, but not too much [16:46:56 CET] <phobosoph> JEEB: ah, so I add -debug_ts to the ffmpeg invocation? :) [16:46:58 CET] <JEEB> -v debug tends to be a crapshoot [16:47:04 CET] <JEEB> phobosoph: yes [16:47:15 CET] <phobosoph> good I will do that, the output is logged as ffmepg runs in background [16:47:47 CET] <phobosoph> indeed, lots of stats are logged now :O [16:48:12 CET] <phobosoph> he, at some point I should use something like Prometheus to log the ffmpeg stats :D [16:48:24 CET] <JEEB> basically how a packet is read, how ffmpeg.c pokes the timestamps of that packet, pushing packet for decoding, pulling it to get a decoded frame etc [16:48:25 CET] <phobosoph> so I let this run until it fails again and then I can see some pattern in these stats? [16:48:29 CET] <phobosoph> ah [16:48:49 CET] <phobosoph> most interestingly ffmpeg happily sends the data and doesn't notice that youtube stopped using them [16:49:01 CET] <phobosoph> hmm, maybe youtube should also give the rtmp client a hint [16:49:23 CET] <phobosoph> ah,it would be great if this is finally fixed [16:49:41 CET] <JEEB> well this should tell you if the stuff within ffmpeg.c seems OK [16:50:03 CET] <JEEB> like, you don't get random timestamp jumps or pcakets just stop being pushed [16:50:35 CET] <phobosoph> JEEB: so when it starts to fail I put the log stuff into a pastebin so you can check it [16:51:00 CET] <JEEB> sure, also if you know the time when it started to bork you can check some time before that and onwards [16:51:04 CET] <JEEB> for any signs of weirdness [16:54:29 CET] <phobosoph> I also tried different streaming systems and networks and hosters, so it must be something either with youtube ingestion issues or ffmpeg config [18:16:13 CET] <phobosoph> stream still works, but this takes some hours, so I wait "paitently", lol [18:16:20 CET] <phobosoph> patiently [19:38:51 CET] <phk> hi, [19:38:51 CET] <phk> im not eble to run more than around 30 instances of ffplay in centos 7 server with dummy display [19:38:51 CET] <phk> (https://techoverflow.net/2019/02/23/how-to-run-x-server-using-xserver-xorg-…) [19:38:51 CET] <phk> no filter or anything, just simple command like DISPLAY=:1 ffplay -i udp://239.255.1.18:1234 -resize 384x216 -left 0 -top 0 -x 384 -y 216 >/dev/null 2>&1 & [19:38:53 CET] <phk> what can be issue? [19:39:33 CET] <nicolas17> what happens when you try? [19:39:43 CET] <phk> https://pastebin.com/fZCeFqFs [19:39:56 CET] <phk> give this kind of error [19:40:38 CET] <phk> it works fine till 30-32 instances thn gives above error [19:40:44 CET] <nicolas17> that's super strange o.o [19:40:55 CET] <phk> i tried with mpv too [19:41:12 CET] <phk> but mpv isnt though error, its freeze [19:41:38 CET] <phk> https://superuser.com/questions/538164/how-many-instances-of-ffmpeg-command… [19:41:48 CET] <phk> i tried this too, but it didnt help [19:42:15 CET] <nicolas17> the stack trace mentions pulseaudio, try -an to disable audio entirely [19:42:22 CET] <nicolas17> just to see if that's really the cause [19:43:18 CET] <phk> ok [19:46:27 CET] <phk> nicolas17: its working without audio, thank you [19:46:42 CET] <nicolas17> do you need audio? [19:49:29 CET] <phk> well, right now no [19:49:34 CET] <phk> but in future, yes [19:58:18 CET] <phobosoph> I am backi [19:58:20 CET] <phobosoph> so [19:58:21 CET] <barg> hi, I can't figure out how to remove the rotation from this file https://paste.ubuntu.com/p/752PV3h7zk/ [19:58:26 CET] <phobosoph> youtube stopped recongizing the stream again [19:58:30 CET] <phobosoph> I checked the detailed logs of ffmpeg [19:58:33 CET] <phobosoph> frame=169271 fps= 30 q=-1.0 size= 4241747kB time=01:34:02.60 bitrate=6158.2kbits/s speed=0.999x [19:58:39 CET] <phobosoph> as you can see, speed is not 1x anymore [19:58:42 CET] <phobosoph> it dropped to 0.999x [19:59:10 CET] <phobosoph> it drops and drops in the log [19:59:11 CET] <phobosoph> speed=0.533x [19:59:21 CET] <phobosoph> then I stopped it because youtube stopped recognizing the stream a long time ago [19:59:29 CET] <phobosoph> bitrate=6158.2kbits/s [19:59:42 CET] <phobosoph> so the bitrate is the same according to ffmpeg, but speed goes down to 0 slowly but steadily [19:59:59 CET] <JEEB> any changes from the debug_ts output? did something stop happening? [20:00:26 CET] <phobosoph> JEEB: yes! very much indeed [20:00:36 CET] <phobosoph> so all the time there are debug lines plus that speed line [20:00:45 CET] <phobosoph> in the end there is only the speed line, no debug lines [20:00:53 CET] <phobosoph> where the speed drops down [20:01:03 CET] <JEEB> right, so try and check when the last time packets went through then [20:01:28 CET] <phobosoph> ah [20:02:41 CET] <phobosoph> JEEB: https://pastebin.pl/view/0eea1bb8 [20:02:43 CET] <phobosoph> the last part [20:03:34 CET] <phobosoph> JEEB: then only the speed line that drops down to 0 (but I stopped ffmpeg before that, it would probably take a very long time): https://pastebin.pl/view/5d73fb7c [20:04:17 CET] <phobosoph> JEEB: so ffmpeg stops processing video packets? [20:05:25 CET] <JEEB> if demuxer -> no longer gets anything or anything similar, it seems like it's not getting anything from the input? [20:06:13 CET] <phobosoph> JEEB: hmm, when I open the ipcam stream it still works [20:06:25 CET] <phobosoph> JEEB: so the ipcam stream (RTSP) may have stopped for some time? [20:11:33 CET] <phobosoph> JEEB: any ideas for debugging this? :) [20:11:47 CET] <phobosoph> JEEB: I can also start ffmpeg again with different flags to capture other info [20:13:55 CET] <barg> also i don't understand is metadata rotation info something that is a cause of rotation or just a documentation of the video having been documented? [20:15:03 CET] <phobosoph> sorry, connection issue :( [20:15:09 CET] <phobosoph> so have you written something? [20:26:36 CET] <furq> barg: it causes the player to rotate the video on playback [20:57:46 CET] <barg> furq: any idea why i can't get rid of that metadata? https://paste.ubuntu.com/p/752PV3h7zk/ [21:31:19 CET] <barg> furq: ffmpeg -i VID_20191219_210054591.mp4 -vf "transpose=1" blah.mp4 <-- seems to have gotten rid of the rotational metadata [21:31:26 CET] <barg> but the other commands i'd done didn't [21:31:37 CET] <barg> i think it had a rotation of 270 degrees. so another 90 and it went to 0 and didn't say rotated. [21:31:50 CET] <barg> not sure if that's right but that's what it seems like [21:35:49 CET] <barg> also if all it does is cause the player to rotate, then why would that command take quite a while? [21:39:08 CET] <nicolas17> you're transcoding the entire video with that command [21:39:12 CET] <nicolas17> maybe the audio too [21:40:27 CET] <phk> how to map pid in ffplay? [21:40:39 CET] <phk> -map i:<pid> [21:40:43 CET] <phk> or -i:<pid> [21:40:51 CET] <phk> or #<pid> not working [22:27:44 CET] <barg> nicolas17: should I be specifying the video and audio codec with that command, to prevent a default being used? [22:28:05 CET] <nicolas17> the 'transpose' codec will *actually* rotate the video, and that will need transcoding [22:28:11 CET] <nicolas17> s/codec/filter/ [22:28:35 CET] <nicolas17> it decodes each frame, rotates the image, and encodes it back [23:24:00 CET] <barg> would this command just set the metadata rather than reencode the entire video? ffmpeg -i VID_20191219_210054591.mp4 -metadata:s:v rotate="0" rrrr.mp4 [23:24:22 CET] <barg> it seems to take a while too.. so seems like it may be reencoding too, so doesn't seem that different from -vf "transpose=.." [23:24:38 CET] <pink_mist> you didn't use -c copy, so it'll make a really crappy reencode probably [23:24:44 CET] <pink_mist> because the default settings are awful [23:27:23 CET] <barg> ok so should I use -c copy for both of those? the -metadata example, and the -vf "transpose=1" example? [23:27:44 CET] <pink_mist> for -vf transpose you can't [23:28:00 CET] <barg> why not? [23:28:12 CET] <pink_mist> because that needs to reencode [23:29:26 CET] <barg> ah that makes sense, thanks [23:34:50 CET] <barg> So for -vf "transpose=1" is it a good idea to specify a video and audio codec to make sure it doesn't use defaults? [00:00:00 CET] --- Sun Dec 22 2019

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