[Ffmpeg-devel-irc] ffmpeg.log.20170823

[burek]

[00:12:42 CEST] <Fenrirthviti> Is there an option for a custom vbv buffer size for NVENC?
[00:14:55 CEST] <bigpresh> I'm trying to get ffmpeg to receive an RTSP feed from a camera and send it to ffserver; I'm using `ffmpeg -i rtsp://... http://127.0.0.1:8090/feed1.ffm`, and getting "Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height"
[00:15:12 CEST] <bigpresh> Is there a way to say "just use the bitrates etc of the RTSP stream"?
[00:16:19 CEST] <alexpigment> Fenrirthviti: I've used -bufsize and -maxrate with nvenc in the past
[00:16:50 CEST] <alexpigment> It's possible it ignored the parameters (it's been a while since I tested), but it didn't throw any errors and I believe it actually honored them
[00:17:13 CEST] <Fenrirthviti> I have a few users who claim that streaming to an nginx-rtmp server with a vbv buffer of over 7k causes disconnects
[00:17:21 CEST] <Fenrirthviti> I'm 99.9% sure they're full of shit, but, whatever.
[00:17:32 CEST] <Fenrirthviti> I regularly stream to mine at 35-40k with no custom buffers set :\
[00:18:13 CEST] <alexpigment> Yeah, I'm not sure what would cause that. But either way, it's worth trying and confirming (or calling bullshit)
[00:21:10 CEST] <bigpresh> (I've tried '-codec copy' which I would hope would copy all details of the codecs for each stream, but that may well be a misunderstanding of how it would work)
[00:29:20 CEST] <kepstin> bigpresh: "-codec copy" tells ffmpeg to literally just copy the already encoded data from the input to the output, rather than decoding it, filtering, then re-encoding.
[00:30:21 CEST] <kepstin> bigpresh: that said, you're unlikely to get help with ffserver here, very few people use it and it's basically unmaintained :/
[00:32:29 CEST] <bigpresh> Oh, really? Arse.
[00:33:30 CEST] <bigpresh> Any recommendations on something similar that's actively maintained?  Basically, I want to grab video & audio from a couple of IP cameras to a central box that can then re-stream them, as that box has much better network link than the cameras do
[00:33:42 CEST] <bigpresh> (and the poor little things don't handle multiple viewers of their streams well :) )
[00:34:52 CEST] <klaxa> you could *try* my mkv server
[00:35:10 CEST] <klaxa> https://github.com/klaxa/mkvserver_mk2
[00:35:42 CEST] <klaxa> or nginx-rtmp or hls with a webserver
[00:36:39 CEST] <bigpresh> Interesting, thanks - I shall look in to those
[00:37:08 CEST] <bigpresh> I started setting up ZoneMinder, figured that would be good - then realised I need audio :)
[00:41:49 CEST] <bigpresh> I shall have a look at all of those optoins; nginx-rtmp stands out as maybe the easiest, and I'm already running nginx :)
[00:42:29 CEST] <klaxa> hls should be similarily easy
[06:18:19 CEST] <dafna_> Hi, if I set my own get_buffer2 when decoding a video
[06:18:32 CEST] <dafna_> and I also use multithreaded,
[06:19:10 CEST] <dafna_> that is thread_count in the avcodeccontext is > 1
[06:19:26 CEST] <dafna_> the n I need to handle my program with locks ?
[08:00:20 CEST] <crux0> Anyone know if ffmpeg has support for FastTracker files? I've downloaded a whole bunch of chiptune mods, and many of them are still in fasttracker format. I'm trying to play them using mpv, but ffmpeg doesn't seem to natively support .ft
[10:30:40 CEST] <thebombzen> uh, I just had FFmpeg default to aac for a webm encode
[10:30:42 CEST] <thebombzen> is this a bug?
[10:32:22 CEST] <JEEB> thebombzen: sounds like either your FFmpeg is old enough or you're not picking the webm mode correctly in matroskaenc
[10:32:31 CEST] <JEEB> see around line 1288 in matroskaenc.c
[10:32:33 CEST] <thebombzen> I built it recently
[10:32:50 CEST] <thebombzen> also this is ffmpeg.c, so I shouldn't have to use this
[10:32:54 CEST] <JEEB> if (mkv->mode == MODE_WEBM && !(par->codec_id == ....)
[10:33:11 CEST] <JEEB> there's a definite avcodec check there for the thing you're trying to push in there
[10:34:08 CEST] <thebombzen> hm, I can't reproduce
[10:34:17 CEST] <thebombzen> ah, gg, I had -c copy and no -c:a
[10:34:28 CEST] <thebombzen> closed: resolved. resolution: user is a dumb
[10:53:45 CEST] <fbeaufort> Hello everyone! I'd like to convert "properly" a mp4 file to flac file.
[10:53:45 CEST] <fbeaufort> The audio flac file I get can be read in Google Chrome but not in Firefox.
[10:53:45 CEST] <fbeaufort> Firefox folks tell me it's because the "SampleDescriptionBox" reads 0 for the samplerate and it must be a valid value.
[10:53:45 CEST] <fbeaufort> Would you happen to know how I can set the SampleDescriptionBox samplerate value from ffmpeg CLI?
[10:53:45 CEST] <fbeaufort> For info, here's the command I used:
[10:53:46 CEST] <fbeaufort> ffmpeg -i bear-1280x720.mp4 -map 0:0 -acodec flac -strict -2 -ar 192000 bear-flac-192kHz.mp4
[11:19:26 CEST] <fbeaufort> Note that when I don't change SampleDescriptionBox, audio file can be decoded in Firefox.
[11:20:52 CEST] <fbeaufort> FYI jya
[11:21:46 CEST] <fbeaufort> jya FYI
[11:22:18 CEST] <jya> fbeaufort: what do you mean by "when I don't change SampleDescriptionBox" ?
[11:22:42 CEST] <fbeaufort> when I don't set the "-ar" option in ffmpeg
[11:23:16 CEST] <fbeaufort> See more at https://github.com/GoogleChrome/samples/pull/538#issuecomment-324270570
[11:23:21 CEST] <jya> it's very weird that -ar would actually set the rate in the AudioSampleDescriptionBox to 0
[11:23:26 CEST] <jya> rather counter intuitive
[11:25:42 CEST] <fbeaufort> Would you happen to know where the code for this live in ffmpeg?
[11:26:27 CEST] <fbeaufort> QQ: How did you create your flac-in-mp4 files at https://github.com/mozilla/gecko-dev/tree/master/dom/media/mediasource/test/flac?
[11:27:34 CEST] <jya> fbeaufort: I asked BBC :)
[11:28:00 CEST] <jya> that was before ffmpeg had supports
[11:28:32 CEST] <jya> let me give you the link for the flac support in ffmpeg, afaik we submitted that
[11:34:20 CEST] <jya> fbeaufort: this was the commit adding flac in mp4 support https://github.com/FFmpeg/FFmpeg/commit/7dc4200c3828195ef33c8a6572891ecda5058cd6#diff-ee9f1598eabbf54de15b68d4df0535d3
[11:37:44 CEST] <jya> fbeaufort: at a glance, looks like it's because the track duration is 0, so it returns an average bitrate of 0
[11:38:01 CEST] <jya> https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/movenc.c#L559
[11:38:26 CEST] <jya> oops sorry, that's bitrate
[11:45:44 CEST] <jya> fbeaufort: did you create a ffmpeg bug already?
[11:45:52 CEST] <fbeaufort> Not yet.
[11:46:11 CEST] <fbeaufort> I'd like to understand what is going on ;)
[11:46:14 CEST] <jya> i think i see why it does so
[11:46:19 CEST] <jya> it always write 0
[11:46:56 CEST] <jya> this stupid pdf reader on my mac, is unable to perform a search on the ISO 14496-12 document
[11:49:25 CEST] <jya> https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/movenc.c#L1032
[11:49:33 CEST] <jya> if the rate is greater than 65536, it writes 0
[11:50:47 CEST] <jya> that can't be it, it's a 16 bits int, but in the spec it's 32 bits
[11:53:03 CEST] <dafna> after allocating and opening a avcodeccontext with avcodec_open2, then when decoding finishes, is it valid to call just       avcodec_flush_buffers( p_context ); and avcodec_free_context(&p_context); without calling avcodec_close ?
[11:53:28 CEST] <dafna> I ask this because this is what I see in the implementation in VLC
[11:54:32 CEST] <jya> fbeaufort: where do I get the bear-1280x720.mp4 file?
[11:54:56 CEST] <jya> found it
[11:55:02 CEST] <fbeaufort> :)
[11:55:20 CEST] <fbeaufort> also at https://storage.googleapis.com/media-session/bear-1280x720.mp4
[11:56:47 CEST] <jya> not the most uplifting video
[11:58:46 CEST] <jya> fbeaufort: I can play the file created via build.57/bin/ffmpeg -i bear-1280x720.mp4 -map 0:0 -acodec flac -strict -2 -ar 192000 bear-flac-192kHz.mp4
[11:58:52 CEST] <jya> in firefox that is
[11:59:08 CEST] <jya> using ffmpeg from the 3.3 branch
[11:59:27 CEST] <jya> ffmpeg version n3.3.3-1-ge51e07c34e
[12:00:07 CEST] <fbeaufort> I used static release 3.3.3 from https://www.johnvansickle.com/ffmpeg/
[12:00:17 CEST] <fbeaufort> I'll try with most recent build and let you know
[12:00:31 CEST] <fbeaufort> Does that matter that I've tried with Firefox on macOS?
[12:01:13 CEST] <jya> no
[12:03:18 CEST] <jya> that command line doesn't generate a fragmented mp4, maybe it's the fragmentation side of thing that mess things up
[12:05:34 CEST] <jya> the file you generated has Lavf57.75.100 has version
[12:05:50 CEST] <jya> 3.3 branch is Lavf57.71.100
[12:09:15 CEST] <fbeaufort> fragmented or not is the same result for me.
[12:09:54 CEST] <fbeaufort> I can reproduce with latest builds from https://www.johnvansickle.com/ffmpeg/
[12:11:04 CEST] <jya> fbeaufort: pfffft, my bad, I actually made a change to the code which caused it to work
[12:11:12 CEST] <fbeaufort> :D
[12:11:26 CEST] <jya> I can't match what they are writing in the spec.
[12:11:40 CEST] <jya> seems to me that it works because they write a 16 bits int followed by a 0
[12:12:54 CEST] <fbeaufort> Let me try to confirm your thoughts by creating files with different rates
[12:14:41 CEST] <jya> oh, I'm certain that if you set a rate of 48kHz it will wor
[12:14:42 CEST] <jya> k
[12:14:53 CEST] <jya> because my change was doing just that
[12:15:07 CEST] <jya> I replaced the 0 there https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/movenc.c#L1033 with 48000
[12:15:12 CEST] <jya> and it played just fine
[12:17:25 CEST] <fbeaufort> You're right. It works fine!
[12:17:32 CEST] <dafna> ?
[12:18:22 CEST] <jya> fbeaufort: the definition of the AudioSampleEntry is defined in 12.2.3.2
[12:18:37 CEST] <jya> https://irccloud.mozilla.com/pastebin/jZR7Uu75/
[12:20:30 CEST] <fbeaufort> So here we are. There is indeed a bug ;)
[12:23:05 CEST] <jya> i can't find the definition of the format ffmpeg write
[12:23:10 CEST] <jya> I must be overlooking something
[12:23:52 CEST] <fbeaufort> Is that as simple as using `avio_wb32` when writing sample rate?
[12:24:52 CEST] <jya> no... because the value written is a 16.16 fixed point
[12:25:16 CEST] <fbeaufort> oops noted
[12:38:11 CEST] <jya> i don't undestand how that code ever worked, and if they are just using a different set of spec than I have
[12:42:20 CEST] <fbeaufort> It looks like this code hasn't been touched for 6 years.
[12:42:25 CEST] <fbeaufort> https://github.com/FFmpeg/FFmpeg/blame/5f4e555dc7caea343838e9f6f218525bb80216bf/libavformat/movenc.c
[12:42:29 CEST] <fbeaufort> This may explain why
[12:45:23 CEST] <jya> maybe it was different in the quicktime spec
[12:46:45 CEST] <jya> very lucky that anything works today though... so long as rate > 0 it's fine for us. even if the value is buggy. and after that, we actually read the rates in the specialised box that follows. mp3, opus, aac, flac, all have their codec data that will include the actual sampling rate
[12:48:35 CEST] <jya> but having said that, even if we wrote a 16.16 value
[12:48:40 CEST] <jya> you can fit 192kHz in there
[12:50:07 CEST] <jya> you need to use a SamplingRateBox
[12:50:14 CEST] <jya> and ffmpeg has no support for that
[12:50:21 CEST] <jya> so no 192khZ in mp4 for you
[12:50:33 CEST] <fbeaufort> ;(
[12:53:27 CEST] <jya> creating a 192kHz output from a 48kHz file was a bit silly anyway :)
[12:53:54 CEST] <jya> if you wanted to capture a whale singing I could understand, but here's it's a poor bear locked in a cage
[12:54:46 CEST] <fbeaufort> I named this bear "poupi". Please.
[12:56:03 CEST] <fbeaufort> Should we still file a bug or at least make the ffmpeg team aware of this?
[12:58:08 CEST] <jya> you can create a bug that the SamplingRateBox (as per ISO 14496-2 12.2.3.2) is to be used if the value is greater than INT16_MAX
[12:59:14 CEST] <jya> they have provision for that inthe SoundDescription V2, but i can't find this anywhere in the spec
[12:59:20 CEST] <jya> must be a QT thing
[13:00:14 CEST] <jya> and if they properly create a SamplingRateBox, need to file a bug in Firefox that our mp4 demuxer doesn't support it :)
[13:01:46 CEST] <jya> ah indeed, you must enable the MOV mode
[13:31:03 CEST] <fbeaufort> I'm having an error on https://trac.ffmpeg.org account registration.
[13:31:03 CEST] <fbeaufort> http://i.imgur.com/EJyCD9e.png
[13:31:12 CEST] <fbeaufort> Do you mind filing it if that works for you?
[14:45:44 CEST] <stimple> greetings, earthlings
[14:47:07 CEST] <stimple> is it correct that when ffmpeg converts .srt to ass, it chooses which line to display the new text on?
[14:47:36 CEST] <klaxa> what do you mean with line?
[14:47:48 CEST] <stimple> this is line 1
[14:47:50 CEST] <stimple> this is line 2
[14:47:59 CEST] <stimple> i want subs to scroll up
[14:48:23 CEST] <stimple> like they do on youtube
[14:48:28 CEST] <stimple> or a terminal
[14:48:29 CEST] <stimple> or irc
[14:48:55 CEST] <klaxa> that's not really the default for ass subs
[14:49:23 CEST] <stimple> yes
[14:53:43 CEST] <stimple> it's possible the libass format doesn't support this behavior
[14:53:57 CEST] <stimple> trying to find the spec
[15:27:42 CEST] <stimple> i don't see a way to do it with ass format
[15:40:23 CEST] <stimple> i don't find any options to pass to ffmpeg for how to convert .srt to .ass
[15:41:03 CEST] <fbeaufort> jya Let me know if you've filed it already. It looks like it works now.
[15:48:36 CEST] <jya> fbeaufort: I haven't yet.... I need to lodge a bug on our side to have our new mp4 demuxer properly parse "SoundDescription"
[15:48:58 CEST] <jya> I had amended stagefright to support it, but we're about to disable that demuxer
[15:49:16 CEST] <jya> and we also need to add support for the SamplingRateBox
[15:53:28 CEST] <fbeaufort> Here is ffmpeg ticket: https://trac.ffmpeg.org/ticket/6609#ticket
[15:54:27 CEST] <fbeaufort> Do you have some doc/video about Firefox Media Stack?
[15:54:27 CEST] <fbeaufort> I've already watched your Paris Video Tech talk and enjoyed it.
[16:01:17 CEST] <jya> :)
[16:23:33 CEST] <jya> fbeaufort: I've added details to the bug.
[16:24:12 CEST] <fbeaufort> Yeah! Thanks.
[16:32:31 CEST] <jya> fbeaufort: incidentally, even if FF doesn't support yet SamplingRateBox , it should work because per spec samplingrate is to be written with a value of 1<<16 which is greater than 0
[17:02:31 CEST] <stimple> ok i was misled into coming here, sorry - cheers
[17:09:53 CEST] <charly> hello
[17:09:57 CEST] <charly> I'm using avio/avformat/acodec to decode an mpegts stream and it works very well
[17:10:09 CEST] <charly> but i don't know how to detects when the input mpegts stream change (codec, framerate ...)
[17:10:16 CEST] <charly> there is a way to do it ?
[17:12:28 CEST] <DHE> charly: there's not direct notification system. basically you have to watch the decoder's AVCodecContext values to see if they change while running
[17:12:43 CEST] <DHE> also mpegts is a strange one because it uses the AVProgram type to specify which streams are members of the (active) feed. you'll have to watch the AVProgram for changes and then reconsider all listed streams
[17:15:45 CEST] <charly> DHE, thank you, i will try this
[17:27:41 CEST] <Kadigan> Hello. I just wanted to know if there's a foolproof way of determining whether the video is interlaced, progressive or progressive-segmented-frame?
[17:27:55 CEST] <Kadigan> (programmatically using ffmpeg, I should add)
[17:28:11 CEST] <Kadigan> (or possibly ffprobe, at that)
[17:35:45 CEST] <JEEB> Kadigan:  you can get the coding mode values at most
[17:36:03 CEST] <Kadigan> Please elaborate, I'm not sure I understand.
[17:37:34 CEST] <JEEB> there usually is a coding mode value in a video stream. what the actual content is is completely separate. additionally in some formats there are per-picture flags you can receive after decoding
[17:38:24 CEST] <JEEB> what the actual content type is cannot be understood without looking at the decoded pictures
[17:40:34 CEST] Action: Kadigan sighs
[17:41:00 CEST] <Kadigan> I typically notice there's something wrong when I see a whole bunch of drop=X.
[17:41:31 CEST] <Kadigan> I was hoping there was a way I could detect this <before> I start the decode proper, in order to select the flags (ie. use deinterlacing or just recombine)
[17:44:05 CEST] <Kadigan> CANON support doesn't immediately know either, so I guess I'll have to wait for their action. Also, I'm running build N-86436-g4f9e958b04, and it <still> doesn't properly recognize the MXF files that come off of Canon XC10 -- I'm getting long lists of "[NULL @ 0x7ff99a0a7600] Truncating likely oversized PPS (9469 > 4096)".
[17:44:25 CEST] <Kadigan> Or am I doing something wrong there?
[17:44:31 CEST] <Diag> exucse that please
[17:44:52 CEST] <Kadigan> Diag: that's an, um, interesting handle.
[17:45:10 CEST] <Diag> Theres someone who goes by the name MrProcom in another channel
[17:45:16 CEST] <Diag> and once someone made the name MrProlapse
[17:45:18 CEST] <Diag> and grouped it
[17:45:30 CEST] <Diag> It has since been ungrouped
[17:46:12 CEST] <Kadigan> Well, the channel's PG13 and a prolapse is a legit medical condition that can happen due to a number of events... so...
[17:46:27 CEST] <Diag> I can just imagine though
[17:46:32 CEST] <Diag> ~this channel is publically logged~
[17:47:32 CEST] <Diag> wew
[17:47:35 CEST] <Diag> i grouped it for them
[17:55:01 CEST] <Diag> jesus christ i thought i left this channel
[18:37:16 CEST] <tableflopper> Hello - question on merging in subtitle tracks into a video stream.  Can I specify the subtitle file as a URL? e.g.   ffmpeg -i myvid.avi -i http://host/mysubtitle.srt
[18:38:57 CEST] <tableflopper> The ultimate goal is to stream in the subtitles incrementally, since they would be dynamically generated.  Possibly using a pipe for the subtitle file?
[18:40:28 CEST] <Diag> tableflopper: i dont know the answer to this specifically, but could you not just have a script that just wgets that link then executes ffmped
[18:59:53 CEST] <tableflopper> Thanks Diag that could technically work.  Would prefer a memory constant approach like streaming it into ffmpeg as needed as these files will be pretty big
[19:00:31 CEST] <Diag> ah
[19:45:54 CEST] <ney> Need help in streaming with ffmpeg, someone out there?
[19:52:10 CEST] <ney> Someone available now?
[19:54:12 CEST] <BtbN> All crystal ball operators and seers are gone. Forever.
[19:56:22 CEST] <ney> Whoa, bad news man! heheh. I'm having a more trivial problem.
[19:57:22 CEST] <ney> I'm streaming from ffmpeg to a NGinx server. The image is black and white and stripped. Any tip where is the misconfiguration?
[20:00:19 CEST] <samgoody> Hi. Quick question: I compiled ffmpeg and put it into /usr/bin/local/ffmpeg.
[20:01:02 CEST] <samgoody> Whereas the apt-installed version (which must have come with the default install of the OS) is at /usr/bin/ffmpeg
[20:02:24 CEST] <samgoody> And /usr/bin/ffmpeg trumps /usr/bin/ffmpeg when I do $ ffmpeg
[20:03:05 CEST] <ney> Hey Sam, you tried to uninstall the previous version first?
[20:03:25 CEST] <samgoody> No. I wanted to know if there is ome reason not to
[20:03:53 CEST] <BtbN> usr/bin/local is highly unusual
[20:04:02 CEST] <BtbN> do you mean local/bin?
[20:04:12 CEST] <samgoody> I had understood that /usr/local/ are for self compiled programs
[20:04:20 CEST] <BtbN> It is
[20:04:24 CEST] <BtbN> but you said /usr/bin/local
[20:05:12 CEST] <samgoody> hmm, you are right. That is illogical.
[20:05:59 CEST] <furq> you can probably set your $PATH such that /usr/local/bin comes before /usr/bin
[20:06:15 CEST] <furq> debian doesn't usually put anything in /usr/local/bin so presumably anything in there should take precedence anyway
[20:06:20 CEST] <samgoody> So, for starters I shouldput my compiled version into /usr/local/bin/
[20:06:24 CEST] <furq> yes
[20:06:40 CEST] <samgoody> And thats what I suspected, but didnt want to mess with the path without asking first
[20:08:08 CEST] <kepstin> I'm kind of surprised that debian doesn't already have /usr/local/bin earlier in the path?
[20:08:40 CEST] <kepstin> note that if you're using a bash shell where you ran 'ffmpeg' before installing the new one, you might need to run 'hash -r' to reset its cached path lookup
[20:08:59 CEST] <samgoody> $ echo $PATH
[20:08:59 CEST] <samgoody> /usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin:/usr/games:/usr/local/games
[20:09:25 CEST] <samgoody> So, now it should be first. But it isnt
[20:09:49 CEST] <samgoody> $ ffmpeg // shows ffmpeg version 2.8.11-0ubuntu0.16.04.1
[20:10:41 CEST] <kepstin> samgoody: ok, so do you still get the wrong one after running 'hash -r' in that shell or opening up a new terminal?
[20:10:58 CEST] <kepstin> if so, then what does the output of 'ls -l /usr/local/bin/ffmpeg' look like?
[20:13:47 CEST] <samgoody> After logging out and in, it now has the correct version, thank you
[20:14:26 CEST] <ney> Hi guys, I'm streaming from ffmpeg to a NGinx server. The image is black and white and stripped. Any tip where is the misconfiguration?
[20:16:00 CEST] <ianbytchek> Greetings! Can anyone point in the right direction with translating `ffmpeg -i "src.jpg" -vf "palettegen=max_colors=45" -y "palette.png"` into c code? I did go throuh ffmpeg_filter.c and ffmpeg_opt.c and have a working code, but it produces an empty image&
[20:16:15 CEST] <ianbytchek> Is there a knowhow or some guidance on this?
[20:19:44 CEST] <ney> I'll have to go now, I'm almost there. Thanks!
[20:24:35 CEST] <durandal_1707> ianbytchek: did you check for EOF?
[20:27:44 CEST] <ney> To furq and devinheitmueller: Thanks for the support man, I'm able to make a C++ program to stream using ffmpeg to NGinx server without problem (at least, until now). Next step is to embed on my RPi B+ and see what happens!
[20:31:28 CEST] <ianbytchek> durandal_1707: yes, if i try the full conversion, it works, the problem is with using a palettegen on it's own. i think i might be overcomplicating things, not sure where to start looking though.
[20:32:52 CEST] <ianbytchek> durandal_1707: ffmpeg uses lots of custom structs for simple filtergraph, will palettegen work if used with buffer and buffersink filters? is output or anything else would need to be setup differently?
[20:33:00 CEST] <durandal_1707> ianbytchek: palettegen consumes by default all stream and you need to tell him when its received all frames
[20:33:45 CEST] <durandal_1707> ianbytchek: it should work with buffer filters too
[20:34:27 CEST] <durandal_1707> just need to signal eof with special item i cant remmember exactly what
[20:36:29 CEST] <ianbytchek> durandal_1707: that's good to hear. i use avfilter_graph_parse_ptr with simple `palettegen=max_colors=64` desciption. i want to know how ffmpeg knows that the output png must be 16x16, what pix format, etc. does it get this info from filter at some point?
[20:37:58 CEST] <durandal_1707> ianbytchek: its 16x16 because paletteuse needs it that way
[20:38:45 CEST] <durandal_1707> and paletteuse gets info from previous filter
[20:40:28 CEST] <ianbytchek> durandal_1707: i understand that. but how does ffmpeg know that? i do avformat_alloc_output_context2, avcodec_alloc_context3, avformat_new_stream, avcodec_open2, avcodec_parameters_from_context, avio_open, then set up frames and filter graph. all manually. but perhaps i should call something else from palettegen to do the autosetup of the output?
[20:41:26 CEST] <durandal_1707> no, buffersink is setup already what it will receive
[20:43:33 CEST] <ianbytchek> durandal_1707: it just ticked a box in the head, you did tell something about eof, it was something very specific. i must be confusing this with something else. can you tell more about that? what should i invoke specifically?
[20:44:11 CEST] <ianbytchek> durandal_1707: * the last time we talked about it, it fixed something back then but i ended up using a different configuration and didn't need it.
[20:44:18 CEST] <durandal_1707> look at comments of headers for clues
[20:44:27 CEST] <ianbytchek> durandal_1707: cool.
[20:47:18 CEST] <durandal_1707> ianbytchek: you sent NULL as avframe to buffersrc to mark eof
[21:00:07 CEST] <ianbytchek> durandal_1707: hmm. is this with avcodec_send_frame(codecContext, NULL)?
[21:00:40 CEST] <durandal_1707> ianbytchek: no, you send to buffersrc
[21:19:54 CEST] <ianbytchek> durandal_1707: k. testing, but now working. just to be sure, this is av_buffersrc_add_frame. i see this is used in configure_filtergraph in ffmpeg itself.
[21:20:29 CEST] <durandal_1707> yes
[21:21:44 CEST] <durandal_1707> you also need to setup filtergraph with palettegen as only filter, with also sink and src filters
[21:38:36 CEST] <ianbytchek> durandal_1707: yes, that's what i have. i must be going wrong somewhere else. gonna figure it out. thanks for you help again! ffmpeg is a beast, can get my head around it.
[21:51:17 CEST] <SolarAquarion> 2 errors detected in the compilation of "/tmp/tmpxft_0000575e_00000000-7_vf_scale_cuda.cpp1.ii".
[21:51:19 CEST] <SolarAquarion> make: *** [ffbuild/common.mak:95: libavfilter/vf_scale_cuda.ptx] Error 2
[22:43:19 CEST] <alexpigment> drawing a blank here, is there a way to tell if a binary is 32-bit or 64-bit?
[22:43:35 CEST] <alexpigment> or do I have to look for something in the configuration flags
[23:02:17 CEST] <microchip_> alexpigment: on Linux? file /path/to/file
[23:02:26 CEST] <alexpigment> Windows
[23:02:29 CEST] <alexpigment> but no worries
[23:02:51 CEST] <alexpigment> I figured out the issue isn't 32-bit/64-bit related
[23:02:52 CEST] <alexpigment> thanks though
[23:38:02 CEST] <kepstin> there are people who run into 32bit issues on windows; it's kind of surprising how much memory x264 can use on HD+ content ;)
[23:55:52 CEST] <furq> alexpigment: for future reference, there's a windows port of file which will tell you
[23:56:00 CEST] <furq> and also you can check in task manager if it's running
[23:56:18 CEST] <furq> any 32-bit processes will have *32 next to the process name
[23:56:55 CEST] <c7j8d9> is there a way to convert the audio of a movie and add it as a new stream keeping the original too.
[23:56:58 CEST] <c7j8d9> ?
[23:58:25 CEST] <furq> c7j8d9: -map 0:v -map 0:a -map 0:a -c:a:0 aac -c:a:1 copy
[23:59:46 CEST] <c7j8d9> thanks furq. why the second -map 0:a?
[00:00:00 CEST] --- Thu Aug 24 2017