[Ffmpeg-devel-irc] ffmpeg.log.20170824

[burek]

[00:07:00 CEST] <furq> otherwise there's only one audio stream
[00:09:40 CEST] <c7j8d9> oh i see furq so can I -map 0:1 -map 0:a to copy one stream out of multiple?
[00:09:59 CEST] <furq> 0:a:0
[00:10:03 CEST] <furq> or whichever stream it is
[00:10:27 CEST] <furq> https://www.ffmpeg.org/ffmpeg.html#Advanced-options
[00:11:08 CEST] <c7j8d9> perfect! thanks furq
[00:36:28 CEST] <Cracki> kepstin, who's dealing with 4K content and still operating under 32 bit?
[00:37:05 CEST] <kepstin> people who accidentally download 32bit builds of ffmpeg rather than 64bit ones, mostly.
[00:37:24 CEST] <Cracki> I demand 16 bit builds
[00:38:10 CEST] <Cracki> I need my ancient cnc machine with parallel port to cut movie frames into papyrus
[00:44:28 CEST] <grublet> Cracki: i thought i was the only one
[00:44:47 CEST] <Cracki> <3 I love audible pulse frequencies
[00:48:36 CEST] <FishPencil> Is there a way to set the --install-name-dir for the output libs to @loader_path?
[01:57:33 CEST] <leif>  Can anyone tell me when I should use the AVFMT_* macros vs the AVFMT_FLAGS_* macros?
[01:58:34 CEST] <leif> I ask because it looks like AVFMT_* works for the AVFormatContext's 'flags' field, but the AVFMT_FLAGS_* macros appeaer right next to it.
[02:01:39 CEST] <commanderkeen> i wrote some code to take an input, demux, decode the video, run it through filter graph, encode it, then mux it to a file. it works other than i think my timestamps are messed up. ffmpeg - says the new video duration is 121:21:12.47 and the original is 0:0:14.08. any suggestions debugging the time issues?
[02:02:18 CEST] <commanderkeen> im getting the frame-rate from the original stream
[02:02:37 CEST] <commanderkeen> and time base
[02:05:04 CEST] <ankitbehera2670> hi
[02:05:22 CEST] <ankitbehera2670> I am a beginner in ffmpeg
[02:05:34 CEST] <ankitbehera2670> I want to use ffmpeg with c
[02:05:38 CEST] <leif> Ah, okay, it looks like AVFMT_* is for the AV(Input/Output) contexts.
[02:06:18 CEST] <ankitbehera2670> to write programs
[02:06:43 CEST] <commanderkeen> ankitbehera2670: have you looked at the examples?
[02:06:56 CEST] <ankitbehera2670> which examples
[02:07:40 CEST] <commanderkeen> ankitbehera2670: https://www.ffmpeg.org/doxygen/3.0/examples.html
[02:09:00 CEST] <ankitbehera2670> ok, I saw those examples
[02:09:28 CEST] <ankitbehera2670> but I wanted to find out a way to use those old source codes
[02:09:34 CEST] <ankitbehera2670> in dranger.com
[02:09:57 CEST] <commanderkeen> what do you mean find out a way to use them?
[02:10:46 CEST] <ankitbehera2670> I heard that pixel formats have been prefixed with AV_, and the PIX_FMT_* defines were moves to libavutil/old_pix_fmts.h
[02:11:04 CEST] <ankitbehera2670> I don't know it is true or not
[02:54:09 CEST] <Diag> can ffmpeg delay audio
[02:54:22 CEST] <Diag> For whatever reason my audio channel is ~5seconds fast
[02:58:20 CEST] <klaxa> see -itsoffset maybe?
[02:59:33 CEST] <klaxa> or the apad filter
[03:08:56 CEST] <Diag> erm
[03:09:08 CEST] <Diag> ill have a look
[03:32:33 CEST] <durandal_1707> Diag: adelay filter
[03:33:07 CEST] <Diag> durandal_1707: how did you see my message
[03:34:29 CEST] <durandal_1707> Diag: im allseeing
[03:34:47 CEST] <Diag> deleting ffmpeg now thanks
[03:56:39 CEST] <alexpigment> what does avresample do?
[03:56:48 CEST] <alexpigment> do I need to build with it?
[03:57:17 CEST] <durandal_1707> no
[03:57:41 CEST] <alexpigment> short and simple answer :) thanks durandal
[03:58:30 CEST] <durandal_1707> its swresample like library for libav folks only
[04:12:12 CEST] <alexpigment> durandal_1707: that was what I gathered from reading about it - I just needed to confirm to be on the safe side
[04:12:15 CEST] <alexpigment> thanks again
[04:54:50 CEST] <cryptodechange> DBZ bluray test (will probably need to DL as the preview video is compressed)... https://drive.google.com/open?id=0B6rDPzFC8IitYm1CSG02Z3gta2M (original) | https://drive.google.com/open?id=0B6rDPzFC8IitTlRHNTd6ZDVBLW8 (denoise/sharpened)
[04:55:00 CEST] <cryptodechange> Not sure what to do with the 'frame shake', is it more noticeable in the denoised version?
[05:09:22 CEST] <dan2wik> Is there a way I can compile ffmpeg faster?
[05:09:48 CEST] <dan2wik> I'm compile on a Pi 0 because cross compiling doesn't seem to be working
[05:10:13 CEST] <dan2wik> I don't know how much time it takes to compile them, but I don't need ffplay or ffserver
[06:38:03 CEST] <lioz_tandil> Hi, I need help with commands of ffmpeg. I need record a streaming of camera ip to mp4 using h264, but i need generate files *.mp4 with fixed duration in seconds. How can i do?
[06:41:21 CEST] <lioz_tandil> Any kind soul? :)
[07:07:41 CEST] <dan4wik> How do i use the -listen argument?
[07:12:11 CEST] <dan4wik> Alternatively, how would I go about listening for a connection for input?
[07:13:29 CEST] <dan4wik> If ffmpeg accepts input from stdin, I could get that going too.
[12:44:18 CEST] <jya> is there a way to set ffmpeg (ffplay / ffprobe) to print the http headers being sent and received?
[13:10:48 CEST] <Nacht> jya: I see HTTP headers when I use '-v 56' with ffmpeg
[13:11:00 CEST] <jya> Nacht: thanks will try
[13:13:39 CEST] <JEEB> why would you use the numeric value :<
[13:13:53 CEST] <JEEB> -v debug is much more readable, for example
[13:14:12 CEST] <jya> easier to remember maybe ? :)
[13:14:19 CEST] <jya> less character to type
[13:14:46 CEST] <jya> just wondering why our code get a moved temporarily error when doing a range request, but it works fine with ffmpeg
[13:26:24 CEST] <jcelerier1> hi :)
[13:27:16 CEST] <jcelerier1> what could be the cause of an error -11 (resource temporarily unavailable) upon av_send_packet ? I'm just trying to read a .aiff file and it reads approximately the first thousand samples then fails with this
[13:29:43 CEST] <jya> jcelerier1: increase the verbosity, you will see what the server responds with. (assuming you read over http)
[13:31:03 CEST] <jcelerier1> no, it's a local file
[13:31:23 CEST] <jcelerier1> this is my decoding code: https://github.com/OSSIA/i-score/blob/master/base/plugins/iscore-plugin-media/Media/AudioDecoder.cpp#L481
[13:35:08 CEST] <JEEB> umm
[13:35:32 CEST] <JEEB> jcelerier1: try printing out the `av_err2str(return_code)`
[13:36:11 CEST] <JEEB> I'm pretty sure you're actually getting EAGAIN
[13:37:20 CEST] <JEEB> or well, check it against AVERROR(EAGAIN)
[13:37:39 CEST] <jcelerier1> ah yes JEEB, that's it
[13:37:40 CEST] <JEEB> and see the documentation on the send/receive stuff :P
[13:38:10 CEST] <JEEB> it means that "I still have something stuck in my buffers so try the other side and then come back with that packet"
[13:39:09 CEST] <JEEB> https://www.ffmpeg.org/doxygen/trunk/group__lavc__encdec.html
[13:39:13 CEST] <jcelerier1> yep
[13:39:13 CEST] <JEEB> relevant documentation
[13:39:37 CEST] <jcelerier1> hmmm, so if I call avcodec_receive_frame(...) just afterwards I am greeted with "Invalid PCM packet, data has size 1 but at least a size of 4 was expected"
[13:40:29 CEST] <jcelerier1> maybe I should go the other way
[13:40:54 CEST] <jcelerier1> and try to read everything before calling av_send_packet ?
[15:59:08 CEST] <hamersaw> i'm looking to check the integrity of a few movies files that i suspect have been corrupted on a file transfer. i can manually execute the command in the terminal and it works fine, but when i try to script it the movie is sent to stdout (or written to the log file) what am i doing wrong?
[15:59:28 CEST] <hamersaw> ffmpeg -v error -i FILE -f null - 2> /tmp/log
[15:59:47 CEST] <hamersaw> obvioulsy FILE is the the video file and /tmp/log changes for each iteration
[16:03:41 CEST] <tezogmix> Any ideas on how to resolve an audio out of sync when converting a 50fps video to 25fps mp4? Normally, I've been able to do it fine without issue with: "ffmpeg -r 50 -i input.mp4 -preset veryfast -crf 15 -r 25 -c:a copy output.mp4" I checked the source to see if it was out of sync but it's not.
[16:06:37 CEST] <BtbN> Why are you overriding the input framerate?
[16:06:48 CEST] <BtbN> If it's not actually 50fps but something slightly different, that could easily cause your desync.
[16:08:34 CEST] <tezogmix> hey BtbN , it was for a specific device that couldn't handle framerates over 30fps... so i've been having success with changing them on sources that were 60fps (e.g. 60000/1001 to 30000/1001 or 60 to 30 or 50 to 25)
[16:08:53 CEST] <devinheitmueller> tezogmix: I agree with BtbN.  Unless the input.mp4 is somehow broken, get rid of the first -r 50.
[16:09:30 CEST] <BtbN> If it's an input device, that -r won't override the framerate but set it.
[16:09:43 CEST] <BtbN> For a video file input, it will plain override the actual framerate. Get rid of it and try again.
[16:10:26 CEST] <tezogmix> ah ok devinheitmueller & BtbN , so for future reference that initial "-r" prior to input file doesn't need to be there?
[16:10:41 CEST] <BtbN> It's a very bad idea to have it there.
[16:11:11 CEST] <devinheitmueller> Correct.  The order of arguments with ffmpeg is very important.  Having a -r 50 before the -i input.mp4 means to treat input.mp4 as 50 FPS regardless of what the actual files framerate is.
[16:11:29 CEST] <tezogmix> ok i see, i was just following some random command lines i had seen over the months, it didn't seem to be a problem with the other conversions i've done (probably over 50 times now)
[16:11:47 CEST] <devinheitmueller> The second -r 25 in your command line is what you want, since that will mean to take whatever the source framerate in the file is and make it 25 fps.
[16:12:17 CEST] <tezogmix> ok, i did learn about halving or keeping the framerate changes in direct integers
[16:12:29 CEST] <devinheitmueller> If the input.mp4 *really* is 50 FPS, then it probably wont matter.  But as soon as you provide an input file which is 29.97 or some other framerate you will have an avsync problem.
[16:13:04 CEST] <BtbN> and a way too slow or fast video
[16:13:04 CEST] <tezogmix> maybe it was a build issues, normally i update it before running but haven't updated ffmpeg in the last few weeks
[16:14:00 CEST] <tezogmix> any other info i can share of the input source that we could keep in mind (aside from removing the initial "-r 50") from mediainfo utility?
[16:14:45 CEST] <devinheitmueller> Unless the source is known to be broken, you should generally trust ffmpeg to properly figure out the properties of the input.
[16:20:18 CEST] <tezogmix> I just posted the mediainfo details here on pastebin (not sure if you all notice anything in particular to adjust into the command line conversion): https://pastebin.com/SwubLjKh
[16:20:49 CEST] <tezogmix> it was just a simple conversion meant only to lower the framerate and keep everything else as is.
[16:21:10 CEST] <devinheitmueller> Is that really 1080p/50 content?  Or did it just get improperly encoded and its really 1080i/50?
[16:22:04 CEST] <tezogmix> no idea devinheitmueller , that's how it is... i don't know what the source creator intended.
[16:22:44 CEST] <tezogmix> source plays fine on vlc/mpc-hc
[16:32:45 CEST] <tezogmix>  anyways, i downloaded the new ffmpeg build, got another source of same input file again and re-running this with your suggestions of removing that initial -r, thanks devinheitmueller & BtbN
[16:33:48 CEST] <tezogmix> have a good rest of your day/evening!
[17:10:49 CEST] <jcelerier1> how safe is it to have multiple resamples operating at the same time ?
[17:11:51 CEST] <klaxa> i don't see why it should be unsafe
[17:23:37 CEST] <alexpigment> hey guys
[17:24:12 CEST] <alexpigment> when using h264_qsv, if the system has an intel graphics card but it's not being used (no display attached), the qsv encoder will say that it's using partial acceleration and be really slow
[17:24:31 CEST] <alexpigment> is there a way to make this warning an actual error that prevents the encoding from occurring?
[17:24:51 CEST] <alexpigment> in my tests, "partial acceleration" is always significantly slower than x264
[17:30:10 CEST] <redrabbit> wondering if there is hardware encoders on my 6600k i could use
[17:30:20 CEST] <alexpigment> yes, there definitely is
[17:30:23 CEST] <alexpigment> you have quicksync
[17:30:26 CEST] <jkqxz> alexpigment:  Why would the presence of a display make a difference?  Does that actually mean "is running X and therefore some default device is found"?
[17:30:54 CEST] <alexpigment> jkqxz: i don't know. it's a dumb system to be honest
[17:30:57 CEST] <alexpigment> but it's how it works
[17:31:15 CEST] <redrabbit> atm i use ndivia hw encoders
[17:31:20 CEST] <alexpigment> like right now I can simulate this by having an nvidia card running the display but the intel GPU enabled
[17:31:31 CEST] <redrabbit> is there much of a difference between theses and intel
[17:31:44 CEST] <alexpigment> redrabbit: you can't generally use them both (with full acceleration) unless you have a display hooked up to both
[17:31:53 CEST] <alexpigment> redrabbit: nvenc is way better and faster
[17:32:07 CEST] <redrabbit> no need to bother then i have nvenc setup
[17:32:08 CEST] <jkqxz> alexpigment:  You should be able to specify the device to avoid that problem.
[17:32:16 CEST] <alexpigment> jkqxz: this is not for me
[17:32:25 CEST] <alexpigment> this is for an unknowable set of systems and hardware configs
[17:34:40 CEST] <jkqxz> Right.  Then using libmfx is quite a pain, yeah.
[17:36:28 CEST] <jkqxz> You could easily hack the source code to error out immediately in the case you described above.  Message is at <http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/qsvenc.c#l768>, can probably just return an error there.
[17:41:12 CEST] <alexpigment> Thanks for the heads up
[17:41:27 CEST] <alexpigment> I'm already doing a check to see if a test encode succeeds or fails, then enabling the option accordingly
[17:41:33 CEST] <alexpigment> Because it's a warning, it technically succeeds
[17:41:53 CEST] <alexpigment> I was just hoping there was an option you could put in the ffmpeg command line that effectly just treats warnings as fatal
[17:42:00 CEST] <alexpigment> It sounds like there isn't though...
[18:38:05 CEST] <Camusensei> Hello guys. I am struggling with .MTS files, I cannot get the audio out, what do you need? A dump of a ffprobe file.MTS?
[18:39:15 CEST] <c_14> it'd be a start
[18:41:02 CEST] <Camusensei> with output?
[18:41:32 CEST] <c_14> the console output yeah (it usually lists errors etc)
[18:41:38 CEST] <c_14> and it'll include the codecs in the file etc
[18:43:07 CEST] <Camusensei> http://sprunge.us/YAIf
[18:44:03 CEST] <c_14> Are you sure the input has audio?
[18:45:03 CEST] <c_14> And apparently that's not MTS, it's a tele-typewriter file
[18:45:53 CEST] <Camusensei> yeah, it contains a bunch of files and I' sure they have audio in them
[18:46:17 CEST] <c_14> "it contains a bunch of files"
[18:46:24 CEST] <c_14> Is it supposed to be a playlist?
[18:46:37 CEST] <Camusensei> yes
[18:46:56 CEST] <Camusensei> file 'file1.MTS'  and so on
[18:47:03 CEST] <c_14> The only playlists ffmpeg supports are hls and concat
[18:47:07 CEST] <c_14> aah, add -f concat before -i
[18:47:19 CEST] <Camusensei> http://sprunge.us/OQRA is an example
[18:47:28 CEST] <Camusensei> oooh
[18:47:30 CEST] <Camusensei> I see
[18:49:37 CEST] <Camusensei> Yes, it works now, thank you very much!!!!
[18:53:07 CEST] <c_14> np
[19:22:07 CEST] <alexpigment> jkqxz: I edited the source code, but it doesn't error out, unfortunately
[19:22:31 CEST] <alexpigment> It gives an error, sure, but it still goes through the process of creating the file
[19:23:25 CEST] <alexpigment> Does anyone else know how to make ffmpeg fail when an error is encountered?
[19:24:00 CEST] <alexpigment> The error in particular is: "[h264_qsv @ 0021fce0] Encoder will work with partial HW acceleration"
[19:26:07 CEST] <furq> maybe -xerror
[19:26:12 CEST] <furq> i forget if that bails out on any warning though
[19:26:26 CEST] <kepstin> that's not an error, yeah, it's just a warning.
[19:26:38 CEST] <alexpigment> I tried xerror
[19:26:51 CEST] <alexpigment> kepstin: well, I changed qsvenc.c to make it a warning
[19:26:52 CEST] <kepstin> would need special handling added to the qsv encoder to abort when a software fallback is used
[19:26:59 CEST] <alexpigment> er
[19:27:00 CEST] <alexpigment> rather,
[19:27:07 CEST] <alexpigment> i changed it from a warning to an error
[19:27:31 CEST] <alexpigment> this qsv encoder implementation is just a mess
[19:29:32 CEST] <alexpigment> At any rate, here's the area in the code that I changed: https://pastebin.com/1pEyEbAF
[19:29:54 CEST] <alexpigment> I just changed AV_LOG_WARNING to AV_LOG_ERROR
[19:30:02 CEST] <alexpigment> that obviously just changes the color from yellow to red
[19:30:11 CEST] <alexpigment> is there any way to make it fail if it gets that
[19:30:23 CEST] <kepstin> alexpigment: you need the encoder function in the qsvenc.c file to return an error code to libavcodec
[19:32:08 CEST] <alexpigment> do you think removing lines 3 and 4 would do the trick?
[19:32:16 CEST] <alexpigment> i don't really care if the error is accurate. i just need it to be fatal
[19:33:27 CEST] <kepstin> just add a "return AVERROR(EINVAL);" after your log message print
[19:33:55 CEST] <kepstin> or whatever other error code you feel like
[19:34:11 CEST] <alexpigment> so the next line after the semicolon?
[19:34:27 CEST] <alexpigment> (i'm not a programmer - sorry for the dumb question)
[19:34:30 CEST] <kepstin> between lines 3 and 4 in your paste, yeah
[19:34:34 CEST] <alexpigment> cool
[19:34:41 CEST] <alexpigment> i'll try it
[19:34:43 CEST] <alexpigment> thank you very much
[20:02:32 CEST] <alexpigment> kepstin: you just saved me hours of headache. thank you very much again
[21:06:42 CEST] <thebombzen> if I'm trying to do a bit of editing on 44.1 kHz audio (specifically afade), should I use aresample=48k before or after the other effects?
[21:07:58 CEST] <furq> huh
[21:08:04 CEST] <ChocolateArmpits> thebombzen, there's nothing about that filter that would require 48kHz audio. Do you expect 48kHz output or something ?
[21:08:23 CEST] <thebombzen> yea, if I want to do anything with it like encode it to Opus
[21:08:26 CEST] <furq> yeah you don't need to do that and if you want to do that for some other reason then i would assume it makes no difference
[21:09:10 CEST] <thebombzen> the reason I'd ask is if I want to encode it to Opus, it'll have to be 48 kHz anyway. I was wondering if there might be a reason to resample it beforehand
[21:10:21 CEST] <furq> i mean i guess it might make the fade slightly higher quality
[21:10:26 CEST] <furq> it won't be audible though
[21:10:57 CEST] <ChocolateArmpits> the effect processing result won't be distorted
[21:11:08 CEST] <ChocolateArmpits> but yeah, don't expect to hear anything
[21:11:20 CEST] <furq> i wouldn't bother
[21:11:51 CEST] <furq> especially not if you're not definitely going to encode this to opus
[21:11:56 CEST] <furq> and do nothing else with it
[21:13:15 CEST] <TD-Linux> thebombzen, there is no difference unless the intermediate filters are poor quality
[21:13:43 CEST] <thebombzen> alright thanks
[21:13:58 CEST] <furq> especially if you don't actually know if you're going to have to resample it to 48k
[21:14:09 CEST] <thebombzen> well I do actually know that
[21:14:20 CEST] <furq> you sounded very speculative when you mentioned opus
[21:14:29 CEST] <thebombzen> that was just a hypothetical example
[21:14:35 CEST] <thebombzen> I'm combing it with other mostly 48 kHz audio in an NLE
[21:14:46 CEST] <furq> if you're definitely resampling to 48k then it couldn't hurt to do it first
[21:14:56 CEST] <thebombzen> and I'm going to export it and UL it to YouTube later
[21:15:06 CEST] <thebombzen> which will resample it to 48 kHz for Opus
[21:15:37 CEST] <ChocolateArmpits> and libvorbis and aac
[21:15:58 CEST] <ChocolateArmpits> multiple formats there
[21:16:58 CEST] <furq> vorbis and aac will be 48k if the source is
[22:13:56 CEST] <leif> Does FFMPEG compile with pthreads by default in os x? (The build I'm using didn't have the --enable-pthreads flag: https://gist.github.com/LeifAndersen/074d6e4b3246daa2658374a8eac2b8b6 )
[22:14:32 CEST] <BtbN> I'd guess pthreads are autodetect
[22:17:08 CEST] <leif> BtbN: okay, thanks.
[23:20:37 CEST] <squarecircle> ohai
[23:21:08 CEST] <squarecircle> is there a way to see if my ffmpeg instance is using the graphics card for accelerated decoding?
[23:22:07 CEST] <squarecircle> if I start multiple instances, and I have multiple graphic cards, do they access one card or does they are distributed on the cards?
[23:23:59 CEST] <squarecircle> at the moment, I am calling ffmpeg on a server without real graphics card, only contains a MGA G200EV
[23:24:15 CEST] <squarecircle> the vm I'm testing the code only uses the virtualized grpahics adapter
[00:00:00 CEST] --- Fri Aug 25 2017