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April 2016
- 1 participants
- 60 discussions
[00:38:55 CEST] <pfelt> hey all. not sure where to even check for this, but is there a style guide for new filter submissions?
[00:40:56 CEST] <ubitux> style guide?
[00:43:00 CEST] <ubitux> there is the overview doc/writing_filters.txt i mentioned to you last time, then regarding coding style you just follow the style you see in other filters, then for submission click on "contribute" in the left menu of ffmpeg.org
[00:49:41 CEST] <pfelt> yeah. it was more of a coding style question
[00:49:58 CEST] <pfelt> i've still got a lot of cleanup to do before i can submit, i just thought i'd ask tho
[00:51:40 CEST] <rcombs> four-space indentation, spaces around braces, newline before the opening brace in a function, ~80-character lines unless it's inconvenient&
[00:51:52 CEST] <rcombs> really just look around a bit and match the norm
[00:52:53 CEST] <wm4> basically k&r
[00:53:23 CEST] <pfelt> alternative question
[00:53:46 CEST] <pfelt> the basis for my filter is the movie filter. i've modified it a bit. new filter or patchset against movie?
[00:54:05 CEST] <ubitux> patchset against movie
[00:54:36 CEST] <ubitux> you can define a new filter within the same file if deemed necessary
[00:54:44 CEST] <ubitux> see for insteance blend/tblend
[00:55:07 CEST] <ubitux> assuming it's more appropriate than just adding an option
[00:56:55 CEST] <pfelt> the major change is a few new options to allow for changing the stream on the fly
[00:57:08 CEST] <pfelt> so it really *can* be a patset
[00:58:18 CEST] <rcombs> http://www.wired.com/2016/04/ingenious-way-iranians-using-satellite-tv-beam… neat
[01:00:23 CEST] <thardin> sounds like a use case for amateur satellites
[01:00:42 CEST] <thardin> if someone can design a good X-band modem
[01:55:09 CEST] <pfelt> ok, another seemingly silly question. if i call av_strdup() do i need to av_freep() it?
[01:55:25 CEST] <ubitux> yes
[01:56:01 CEST] <pfelt> hmm. ok. when i had the code in i'd segfault so i wasn't sure. it's probably something stupid i'm doing
[02:00:00 CEST] <DHE> you using av_freep properly? it wants a pointer to a pointer
[02:01:48 CEST] <pfelt> then no. i didn't look that deep into the code to actually see that
[02:02:08 CEST] Action: pfelt was trying av_freep(char *) not av_freep(char **)
[02:02:26 CEST] <pfelt> not sure if that showed up right or showed up as an emoticon :D
[02:03:39 CEST] <pfelt> indeed the docs do show it saying pointer to a pointer. i stopped at the av_freep(void * ) definition
[02:08:32 CEST] <rcombs> I'm not sure why the prototype takes void* instead of void**
[02:11:16 CEST] <pfelt> isn't it *technically* right? just a tiny bit misleading?
[02:23:11 CEST] <DHE> void** means you can do one dereference safely before hitting the "black box" of (void*)
[02:23:31 CEST] <DHE> the idea is that it will NULL out the variable for you reducing the odds of a use-after-free
[02:23:53 CEST] <DHE> a lot of API functions do the same thing, like av_frame_free and av_packet_free
[02:54:31 CEST] <cone-016> ffmpeg 03Michael Niedermayer 07master:ea41ab098711: avformat/mpegts: factor duplicate seek back code into mpegts_resync
[02:54:31 CEST] <cone-016> ffmpeg 03Michael Niedermayer 07master:dbe1dd59e0d3: avformat/mpegts: Skip over broken 0x80 headers
[03:03:32 CEST] <rcombs> yeah, nothing invalid about using void* there, it's just not as clear
[03:23:43 CEST] <rcombs> wow, they're (widevine) not even using AES-NI
[03:35:38 CEST] <jamrial> rcombs: whatever happened to your aesni stuff?
[03:35:49 CEST] <jamrial> the cpu flag is in, but not the actual asm
[03:35:54 CEST] <rcombs> been sitting there for a while
[03:36:10 CEST] <rcombs> I forget if anything was actually blocking or if it just hadn't been explicitly OK'd
[03:36:45 CEST] <jamrial> re-send the bits that haven't been commited, then
[04:47:18 CEST] <pfelt> DHE: if you're still around, that makes sense. i haven't had to worry too much about it because src_movie.c already has done it. but yeah. i'm following you now
[05:03:59 CEST] <DHE> also you're supposed to finish your drink if you find a function whose prototype is (void***)
[05:04:00 CEST] <DHE> :)
[06:43:01 CEST] <Timothy_Gu> ubitux: seems like another one of those bit flips of yours http://fatebeta.ffmpeg.org/log/x86_64-archlinux-gcc-checkheaders/2016042404…
[06:43:16 CEST] <Timothy_Gu> ' ' == 0b00100000
[06:43:31 CEST] <Timothy_Gu> '$' == 0b00100100
[09:06:13 CEST] <cousin_luigi> Greetings.
[09:59:39 CEST] <flux> timothy_gu, pretty nice, I don't recall ever having seen such an issue :)
[10:44:59 CEST] <cousin_luigi> Is there an official position in regard to libdca and librtmp? Should internal encoders/decoders be preferred to them, in the stable branch?
[10:45:08 CEST] <cousin_luigi> Well, latest stable.
[11:36:37 CEST] <ubitux> Timothy_Gu: ... ok well, i'm going to try something
[11:36:44 CEST] <ubitux> i disabled one of the disk from the raid
[11:36:50 CEST] <ubitux> will see if it helps
[11:36:56 CEST] <ubitux> might be a bug in the firmware or something
[12:09:37 CEST] <BtbN> Hm, I expected more reactions to non-nonfreee nvenc.
[12:19:25 CEST] <cone-550> ffmpeg 03Michael Niedermayer 07master:c84ba07db4ab: avformat/mux: Check that deinit is set before calling it
[12:37:27 CEST] <cone-550> ffmpeg 03Diego Biurrun 07master:02cd8bb9cb43: h264: Clean up #includes
[12:37:28 CEST] <cone-550> ffmpeg 03Derek Buitenhuis 07master:64ae08ffec1a: Merge commit '02cd8bb9cb4381c50f42a9284098cc8a33d397db'
[12:40:54 CEST] <cone-550> ffmpeg 03Michael Niedermayer 07master:2e67a99fbc6b: avfilter/vf_drawtext: Check return code of load_glyph()
[12:43:03 CEST] <Daemon404> wm4, it's also going to be painful afaict.
[12:44:26 CEST] <nevcairiel> probably
[12:44:42 CEST] <nevcairiel> all this h264 refactoring has generally been painful, and the codec doesnt run faster for it, and it usually introduces bugs
[12:44:47 CEST] <nevcairiel> so, meh all around
[12:45:02 CEST] <Daemon404> i dont know what the end goal is
[12:45:14 CEST] <Daemon404> just nicer code?
[12:47:46 CEST] <wm4> wouldn't be a problem if there'd be only 1 code base
[12:48:45 CEST] <wm4> also btw., at this point, all blame for reunification not happening can be put on Libav
[12:49:11 CEST] <Daemon404> i wouldnt say all
[12:49:25 CEST] Action: Daemon404 was in the room last year, saw more than that
[12:49:56 CEST] <iive> wm4: all blame have always been on libav
[12:50:00 CEST] <cone-550> ffmpeg 03Diego Biurrun 07master:f4d581cda389: lavc: Deduplicate zigzag_scan table
[12:50:01 CEST] <cone-550> ffmpeg 03Derek Buitenhuis 07master:a5a6621616aa: Merge commit 'f4d581cda3897f66c1dda7586b93f86a591dbbef'
[12:50:03 CEST] <wm4> iive: no
[12:50:18 CEST] <wm4> well I guess there are remaining "factors" in the ffmpeg side
[12:50:24 CEST] <Daemon404> carl factor?
[12:50:49 CEST] <wm4> names withheld
[12:52:36 CEST] <iive> the universal excuse
[12:53:01 CEST] <iive> honestly didn't you get tired at blaming carl for everything?
[12:55:27 CEST] <Daemon404> only when he gets tired of living up to his end of the deal
[12:55:28 CEST] <wm4> maybe whenc arl stops vocally blaming Libav for everything
[12:55:49 CEST] <iive> like?
[12:56:13 CEST] <Daemon404> right, im not getting into this, on a Sunday
[12:56:23 CEST] Action: Daemon404 goes back to doing productive thigns
[12:57:28 CEST] <iive> this is en example of poisonous atmosphere
[12:57:58 CEST] <Daemon404> yes, cause by carl
[12:58:07 CEST] <iive> when somebody is smeared, without even having to point concrete and actual "wrongs"... just hyping and reiterating old rumors
[12:58:19 CEST] <Daemon404> because i tire of having to explain it every goddamn time
[12:58:32 CEST] <iive> Daemon404: ok, give me example for carl obstructing reunification in the last year.
[12:58:40 CEST] <Daemon404> were you in the room?
[12:58:56 CEST] <Daemon404> about every single thing that came up he tried to smack on Libav
[12:59:29 CEST] <Daemon404> now seriously, im not discussing bloody fork politics on Sunday
[12:59:29 CEST] <iive> wasn't "the room" a year ago?
[12:59:34 CEST] <Daemon404> no, 6 months or less
[12:59:41 CEST] <wm4> * Daemon404 goes back to doing productive thigns <- that's wrong on a sunday too
[13:00:35 CEST] <iive> wm4: were you "in the room"?
[13:00:41 CEST] <wm4> no
[13:01:10 CEST] <iive> the conversations in the room were kept "secret" on purpose
[13:01:12 CEST] <rcombs> who cares who "started it"
[13:01:39 CEST] <iive> so nobody would be able to sabotage the unification.
[13:01:45 CEST] <cone-550> ffmpeg 03Diego Biurrun 07master:a7da517f6a5c: h264data: Move all data tables from a header to a .c file
[13:01:46 CEST] <cone-550> ffmpeg 03Derek Buitenhuis 07master:9809371aa540: Merge commit 'a7da517f6a5c472f46f67dd33bb6b95ccc919923'
[13:02:11 CEST] <Shiz> that worked out less well than expected it seems
[13:02:25 CEST] <iive> my point exactly.
[13:02:48 CEST] <iive> it was just temporarily measure, for month or two...
[13:09:47 CEST] <ubitux> iive: pretty sure blaming ppl won't help solving the situation, unless you are in the state of mind of harassing until they're fed up an leave. unfortunately, this is not anywhere close ethically viable, and apparently ppl are so stubborn on both sides that they somehow keep doing their shit. so what we want instead is having cooperation, and in this case, blame has the opposite effect
[13:10:40 CEST] <iive> ubitux: tell that to Daemon404
[13:10:55 CEST] <wm4> in any case, we'd like for carl to adjust his behavior, but he NEVER changes
[13:11:07 CEST] <wm4> and iive started this whole blaming discussion too
[13:11:34 CEST] <iive> wm4: actually you did.
[13:11:43 CEST] <ubitux> wm4: i didn't notice any particular behavior issue with that regard from carl recently
[13:12:04 CEST] <iive> ubitux: and that was my point entirely
[13:13:17 CEST] <wm4> iive: no I didn't
[13:14:19 CEST] <iive> wm4: you used blame and libav in a sentence first :P
[13:14:37 CEST] <wm4> I didn't blame any persons, just the behavior of Libav as a project
[13:15:08 CEST] <iive> wm4: it was Daemon404 that blamed a person. for something he has said in a secret meeting
[13:15:37 CEST] <cone-550> ffmpeg 03Vittorio Giovara 07master:922b7e6d86e6: intrax8: Use local destination buffers
[13:15:38 CEST] <cone-550> ffmpeg 03Derek Buitenhuis 07master:1a9ffc59d27c: Merge commit '922b7e6d86e6197dfa5ebde602dd12cc66fb5f1f'
[13:15:42 CEST] <Shiz> i have to say this is one of the most counterproductive arguments i've read
[13:15:54 CEST] <wm4> and it always goes this way
[13:16:33 CEST] <iive> he said, she said..
[13:17:20 CEST] <cone-550> ffmpeg 03Vittorio Giovara 07master:65127450add5: intrax8: Make x8_init_block_index not use mpegvideo fields
[13:17:21 CEST] <cone-550> ffmpeg 03Derek Buitenhuis 07master:627115b3e43f: Merge commit '65127450add50c3bca307edc0517d2e8382717a0'
[13:17:22 CEST] <iive> wm4: i'm still interested what you had in mind, when you blamed libav (as project)
[13:19:31 CEST] <wm4> we've made plenty of concessions towards Libav... michaelni stepped down as leader, and we've made proposals how the unification could go
[13:19:36 CEST] <wm4> but Libav showed zero interest
[13:26:32 CEST] <iive> that's not surprising.
[13:30:56 CEST] <cone-550> ffmpeg 03Vittorio Giovara 07master:68127e1bf803: intrax8: Keep a reference to the context idctdsp
[13:30:57 CEST] <cone-550> ffmpeg 03Derek Buitenhuis 07master:83e0b1b1d7b4: Merge commit '68127e1bf8037a6e0acd6401cc8c5da950e3fa0a'
[13:47:44 CEST] <cone-550> ffmpeg 03Vittorio Giovara 07master:577393321c38: intrax8: Carry over the loopfilter value in ff_intrax8_decode_picture
[13:47:45 CEST] <cone-550> ffmpeg 03Vittorio Giovara 07master:eaeba6f241e0: intrax8: Pass the output frame to the decoding function
[13:47:46 CEST] <cone-550> ffmpeg 03Vittorio Giovara 07master:65f14128c4bc: intrax8: Use a constant buffer instead of a ScratchpadContext
[13:47:47 CEST] <cone-550> ffmpeg 03Vittorio Giovara 07master:8072345e9f86: intrax8: Keep a reference to the GetBitContext reader
[13:47:48 CEST] <cone-550> ffmpeg 03Vittorio Giovara 07master:1eaae7abb8f2: intrax8: Reference the current AVCodecContext
[13:47:49 CEST] <cone-550> ffmpeg 03Vittorio Giovara 07master:159323897f54: intrax8: Add a local BlockDSPContext and initialize it
[13:47:50 CEST] <cone-550> ffmpeg 03Derek Buitenhuis 07master:e5c655b09817: Merge commit '577393321c389ad2973bec6168a8045c94a9e099'
[13:47:51 CEST] <cone-550> ffmpeg 03Derek Buitenhuis 07master:36dad146935a: Merge commit 'eaeba6f241e0de0e797be10f8fda967ef8489e64'
[13:47:52 CEST] <cone-550> ffmpeg 03Derek Buitenhuis 07master:93c61c980d63: Merge commit '65f14128c4bcf8fcd9d3ba1e20b7a22057c9cfb0'
[13:47:53 CEST] <cone-550> ffmpeg 03Derek Buitenhuis 07master:50fa5715bcc3: Merge commit '8072345e9f86d88fbc4a15c17cb03f1e4701c9a5'
[13:47:54 CEST] <cone-550> ffmpeg 03Derek Buitenhuis 07master:52692eed615c: Merge commit '1eaae7abb8f208fefb4e8b9e983e61b2499206a3'
[13:47:55 CEST] <cone-550> ffmpeg 03Derek Buitenhuis 07master:896fce850966: Merge commit '159323897f545e7405fb9db234e0ba123e174376'
[13:49:15 CEST] <cone-550> ffmpeg 03Ico Doornekamp 07master:edf54887e293: rtpdec_jpeg: fix low contrast image on low quality setting
[13:49:16 CEST] <cone-550> ffmpeg 03Derek Buitenhuis 07master:4fe4c5c3761a: Merge commit 'edf54887e2935a30f9d9a46dd806802c3c867c0e'
[13:51:53 CEST] <cone-550> ffmpeg 03Martin Storsjö 07master:cdb1665f70de: aarch64: Make transpose_4x4H do a regular transpose
[13:51:54 CEST] <cone-550> ffmpeg 03Derek Buitenhuis 07master:87b8e9500874: Merge commit 'cdb1665f70def544ddab3e3ed3763ef99c8b3873'
[16:10:56 CEST] <wm4> [FFmpeg-devel] Cannot compile ffmpeg-0.10.16
[16:10:58 CEST] <wm4> hell
[16:11:19 CEST] <wm4> outdated Oracle crap
[16:11:28 CEST] <JEEB> oracle this time?
[16:13:57 CEST] <wm4> "please excuse me, that I bother you with this old version of ffmpeg, however,
[16:13:57 CEST] <wm4> javafx 8 requires libavcodec53 and libavformat53 on linux systems, which, as
[16:13:57 CEST] <wm4> far as I know, requires that I compile ffmpeg 0.10.16. "
[16:14:54 CEST] <JEEB> lol
[16:14:59 CEST] <JEEB> not even main part of java :P
[16:22:54 CEST] <anadon> Can someone come over to #ffmpeg to help me with a libav question. It's been a few days of me asking to no response.
[16:24:22 CEST] <JEEB> wm4: so something like this https://github.com/jeeb/ffmpeg/commit/41b8aa0dfe3dfdf1016225854b271a586454e… ?
[16:24:33 CEST] <JEEB> with regards to the final version of the PGS fix
[16:25:09 CEST] <wm4> looks good to me
[16:25:57 CEST] <JEEB> also with this the non-CCIR macros could just be removed since IIRC nothing uses them any more in lavc :)
[16:26:12 CEST] <wm4> awesome
[16:26:12 CEST] <JEEB> people found out how often full range YCbCr was being used in paletted formats
[16:26:52 CEST] <nevcairiel> its not about full-range ycbcr, its really more about limited range rgb, which should never exist
[16:27:35 CEST] <JEEB> I thought it was just that they thought that the input values were already in the full range and thus quantization wouldn't be required
[16:27:56 CEST] <JEEB> and the _CCIR ones would just do it so that the range goes from limited to full
[16:30:39 CEST] <jkqxz> Limited-range RGB is an abomnination. Really nothing in software should ever see it, but the requirement in CEA-861/HDMI to use it on the wire is inconsistently implemented and leads to horrible fail.
[16:32:18 CEST] <JEEB> there, posted the patch on ML
[16:32:43 CEST] <wm4> jkqxz: indeed, fuck that shit
[16:33:02 CEST] <JEEB> pretty much
[16:38:25 CEST] <jkqxz> (The only consistently-working answer to the RGB limited-range problem if you are an HDMI receiver is "pretend you are actually DVI receiver". If you need audio, sucks to be you.)
[17:47:10 CEST] <JEEB> michaelni: would that PGS patch I posted be something considered for backporting since as far as I can tell PGS subpictures have had the wrong RGB range and colorimetry for ages and I guess it would be nice to fix it
[22:17:13 CEST] <michaelni> JEEB has that been pushed to master ? (i dont want to backport things that have not yet been pushed to master)
[22:17:29 CEST] <JEEB> not yet since no comments on the latest iteration. just noted for the future if it gets merged
[22:17:59 CEST] <michaelni> ok, then please backport or ping me when its pushed
[22:18:07 CEST] Action: michaelni will forget
[22:18:17 CEST] <JEEB> gotcha
[23:05:51 CEST] <cone-749> ffmpeg 03Jan Ekström 07master:9779b6262471: pgssubdec: fix subpicture output colorspace and range
[23:53:35 CEST] <durandal_1707> size is 8x8 yet strachpad is of 42 size?
[23:57:07 CEST] <durandal_1707> its so side libav devs ignore my fixes
[23:57:15 CEST] <durandal_1707> *sad
[00:00:00 CEST] --- Mon Apr 25 2016
1
0
[00:06:03 CEST] <cousin_luigi> Greetings.
[00:07:11 CEST] <Prelude2004c> hello
[00:07:16 CEST] <Prelude2004c> anyone around to help ?
[00:08:39 CEST] <cousin_luigi> Can libdca and librtmp be safely disabled? Are the internal encoders/decoders at least on par?
[00:11:09 CEST] <furq> libdcadec is the internal implementation now
[00:11:31 CEST] <furq> you can no longer link with it as of a few weeks ago
[00:12:18 CEST] <furq> i've used the internal librtmp a bit and it seems fine but i can't really say with any authority
[00:12:22 CEST] <furq> s/lib//
[00:19:53 CEST] <Prelude2004c> hey furq.. you helped me quite a bit last time.. have an odd problem ( audio seems fine now )
[00:19:58 CEST] <Prelude2004c> http://pastebin.com/ustVEMaa
[00:37:29 CEST] <anadon> I'm getting garbage frames from the linked code and I'm not sure what could be going wrong: http://pastebin.com/3MGivRdv
[07:36:07 CEST] <bencc> is it possible to format ffmpeg errors so I'll know they comes from fffmpeg?
[09:34:29 CEST] <pa> Hi
[09:34:48 CEST] <pa> does anybody here have a dvb-s2 PC card?
[10:09:09 CEST] <Franciman> Hi all
[10:09:21 CEST] <Franciman> How can I generate scenechanges for a video?
[10:09:44 CEST] <Franciman> I mean, finding scene changes*
[10:11:15 CEST] <furq> -vf select=gt(scene\,.4)
[10:11:41 CEST] <furq> or -vf select=gt(scene\,.4),showinfo if you want text output
[10:24:06 CEST] <Betablocker>
[10:28:18 CEST] <Franciman> sorry, here I am. furq I'm a noob with ffmpeg, should I write ffmpeg "-vf ...." ?
[10:34:16 CEST] <Franciman> ok it was like that :P
[10:37:58 CEST] <Betablocker> trying commands with ffmpeg is fun, it is good to play around when you are new
[10:38:02 CEST] <Betablocker> hammer on !
[10:38:23 CEST] <Franciman> ffmpeg -vf "select=gt(scene\,.4),showinfo" -i vid.mp4
[10:38:29 CEST] <Franciman> I'm trying like that, but I get this error:
[10:38:37 CEST] <Franciman> Option vf (set video filters) cannot be applied to input file vid.mp4 -- you are trying to apply an input option to an output file or vice versa. Move this option before the file it belongs to.
[10:38:37 CEST] <Franciman> Error parsing options for input file vid.mp4.
[10:38:37 CEST] <Franciman> Error opening input files: Invalid argument
[10:41:16 CEST] <Franciman> ok found out this too. But what should be the output format?
[10:48:07 CEST] <Franciman> I'd like to have their timestamps
[11:49:16 CEST] <pandb> i wrote a program based on the muxing.c example on ffmpeg.org, and when I try to open the file it generates with mplayer, i get no video, but this series of messages appears on my terminal: pastebin.com/WXvz4aCf
[11:50:07 CEST] <pandb> write operations with av_write_frame and av_interleaved_write_frame appear successful
[11:50:32 CEST] <pandb> and the resulting file's size is consistent with my expectations
[11:53:00 CEST] <pandb> i'm hoping the problem indicated by mplayer's output is just one of a missing function or setting
[11:55:26 CEST] <Mavrik> Are you writing the trailer and closing the muxer?
[11:57:07 CEST] <pandb> no, just hitting ctrl-c
[11:57:19 CEST] <pandb> that's worked fine for me before
[11:57:20 CEST] <pandb> lol
[11:57:35 CEST] <pandb> the video is being generated continuously
[11:59:49 CEST] <pandb> im trying to broadcast to twitch.tv, but was hoping to get feedback on the process by saving to file
[12:01:04 CEST] <pandb> when I try to broadcast to a url i get messages that say 'buffer underflow st=0 bufi=#### size=74897
[12:01:34 CEST] <pandb> and bufi keeps increasing until it reaches 74897, then those messages stop
[12:01:52 CEST] <Mavrik> So.
[12:01:56 CEST] <Mavrik> You wrote a program that writes MP4.
[12:02:09 CEST] <Mavrik> And then you kill it before it has chance to wrap up the container and write the MOOV atom? :)
[12:03:03 CEST] <pandb> i guess so
[12:03:25 CEST] <qmr> I like to MOOV it MOOV it
[16:07:50 CEST] <Phrk_> Hello when i try to open a sftp file i got this : libssh: Error opening sftp file: SFTP server: No such file
[16:08:26 CEST] <Phrk_> i use this ffplay sftp://user:pass@ip:port/path.mkv
[16:09:02 CEST] <Phrk_> if i mount with fuse it's working, but i want to be able to play video without mounting
[16:19:26 CEST] <turbolego> sup
[16:20:10 CEST] <turbolego> does anyone know why this command results in a mp4 video that has not been speed up?
[16:20:13 CEST] <turbolego> ffmpeg -i E:\fraps\Movies\Linjeduell.mkv -vf "setpts=0.5*PTS" Linjeduell2.mp4
[16:30:47 CEST] <anadon> I'm getting garbage frames from the linked code and I'm not sure what could be going wrong: http://pastebin.com/3MGivRdv
[16:50:31 CEST] <Macey> Hi All, need some help.... Trying to stream to Flash Media Server. Older version of FFMPEG as embedded in app.
[16:50:34 CEST] <Macey> -c:a libvo_aacenc -ar 48000 -ab 48k -c:v libx264 -s 320x240 -b 200k -pix_fmt yuv420p -f flv
[16:50:53 CEST] <Macey> -s 320x240 & -pix_fmt yuv420p are apparently invalid options
[16:51:25 CEST] <furq> which version of ffmpeg
[16:59:42 CEST] <Macey> furq AVcodec-56.dll 56.41.100.0
[17:00:35 CEST] <Macey> ffmpeg.avcodec 56.41.100
[17:00:35 CEST] <Macey> ffmpeg.avformat 56.36.100
[17:00:35 CEST] <Macey> ffmpeg.avfilter 5.16.101
[17:00:35 CEST] <Macey> ffmpeg.avutil 54.27.100
[17:00:35 CEST] <Macey> ffmpeg.swscale 3.1.101
[17:02:59 CEST] <furq> those options should definitely exist in ffmpeg 2.7
[17:03:27 CEST] <furq> maybe try -vf scale=320:240,format=pix_fmts=yuv420p
[17:06:46 CEST] <Macey> thanks furq i'll give it a go
[17:08:04 CEST] <furq> -b 200k should be -b:v 200k
[17:08:13 CEST] <furq> maybe that's screwing up the option handling
[17:12:54 CEST] <topia_> hello all
[17:13:39 CEST] <topia_> how do i run ffmpeg in linux?
[17:14:00 CEST] <topia_> i went to the download page, and was redirected to a ubuntu page, which just game be source code
[17:14:10 CEST] <topia_> is there no pre compiled binaries available?
[17:14:38 CEST] <Macey> furq, perfect thanks for your help
[17:15:40 CEST] <kuroro> im planning to use ffmpeg in my android app and have looked at how other apps have integrated it. I see most using JNI and linking against libffmpeg.so, and a few that just calls out to a statically built ffmpeg executable. i'm currently leaning towards the latter approach as it simpler (no need to worry about JNI, familiar command line options). Would you guys suggest the same?
[17:19:36 CEST] <furq> topia_: http://johnvansickle.com/ffmpeg/
[17:19:56 CEST] <furq> your distro should have packages for it, but older versions of ubuntu won't have it or will have a very old version
[17:20:10 CEST] <topia_> tyvm furq
[17:20:22 CEST] <c_14> kuroro: what are you trying to do (what does your app need ffmpeg for)
[17:25:09 CEST] <topia_> furq i get the error "The program 'ffmpeg' is currently not installed. You can install it by typing:sudo apt install ffmpeg"
[17:25:18 CEST] <topia_> when trying to call it from a terminal
[17:25:37 CEST] <JEEB> if it's not in your PATH you will have to call it specifically in its location
[17:25:44 CEST] <topia_> in windows i just need to have the binary in a dir, and then i can do ffmpeg whatever from command line
[17:25:48 CEST] <furq> ./ffmpeg
[17:26:00 CEST] <JEEB> that's because windows has "." in PATH :P
[17:26:00 CEST] <topia_> i was in the same directory as ffmpeg in terminal jeeb
[17:26:04 CEST] <topia_> oh
[17:26:13 CEST] <topia_> will try ./
[17:26:20 CEST] <JEEB> most *nix systems don't have "." in PATH for various reasons
[17:26:22 CEST] <furq> just install it to /usr/local
[17:26:25 CEST] <furq> or er
[17:26:27 CEST] <furq> /usr/local/bin rather
[17:27:03 CEST] <furq> debian/ubuntu don't install anything in /usr/local, it's reserved for your manually installed programs
[17:27:06 CEST] <JEEB> yes, that is one location that is often in the default PATH and not used by the system's own packaging
[17:27:28 CEST] <JEEB> `echo $PATH` to see your current defined PATH :P
[17:28:32 CEST] <kuroro> c_14: to join video clips
[17:29:05 CEST] <kuroro> and maybe add subtitles
[17:29:16 CEST] <c_14> probably use the commandline. If what you're trying to do works via commandline, that will probably be easier. Also (helps) get around various possible licensing issues.
[17:31:43 CEST] <kuroro> i see. i guess there are certain tasks that necessitate linking against a shared lib, and thats probably why some apps decided to go the JNI route
[17:32:36 CEST] <kuroro> tnx for the suggestion
[18:10:54 CEST] <pa> i was looking at the ffmpeg webcam capture page: https://trac.ffmpeg.org/wiki/Capture/Webcam
[18:11:06 CEST] <pa> more specifically at: ffmpeg -f v4l2 -framerate 25 -video_size 640x480 -i /dev/video0 output.mkv
[18:11:18 CEST] <pa> but what if i want ffmpeg to transcode for me on the fly?
[18:11:28 CEST] <pa> like not just capture but actually encode the live stream?
[18:14:32 CEST] <c_14> it does that
[18:14:34 CEST] <c_14> by default
[18:14:38 CEST] <c_14> (unless you specify -c copy)
[18:19:25 CEST] <Phrk_> anyhelp : ffplay sftp://user:pass@ip:port/path.mkv [ffmpeg] libssh: Error opening sftp file: SFTP server: No such file
[18:20:28 CEST] <c_14> that file doesn't exist in that location on that server
[18:21:58 CEST] <Phrk_> yes it exist
[18:22:17 CEST] <Phrk_> if i use http or fuse via ssh it works
[18:22:54 CEST] <Phrk_> if i browse sftp with firefox he show the file can download it but not play
[18:24:05 CEST] <pa> c_14, but how do i specify the codec?
[18:24:15 CEST] <c_14> pa: -c:v foobar
[18:24:29 CEST] <pa> and if i dont, what happens?
[18:24:33 CEST] <pa> like in that line
[18:24:37 CEST] <c_14> it uses the default video codec for that container
[18:24:41 CEST] <pa> ah okay
[18:24:42 CEST] <pa> thanks!
[18:24:50 CEST] <c_14> for mkv that is libx264 if your build comes with that otherwise mpeg4
[18:25:04 CEST] <pa> last thing: how i choose mpeg-ts container?
[18:25:13 CEST] <c_14> change the extension to .ts
[18:25:23 CEST] <c_14> or add -f mpegts
[18:25:28 CEST] <pa> ah great
[18:25:29 CEST] <pa> thanks
[18:25:50 CEST] <c_14> Phrk_: maybe try :port//path.mkv ?
[18:26:08 CEST] <c_14> Is the file /path.mkv or is it relative to the user's home directory?
[18:29:44 CEST] <Phrk_> it's not on home, it's chroot sftp
[18:29:59 CEST] <Phrk_> c_14, no don't work with //
[18:30:37 CEST] <Asuran> hello, i got problems with following command: ffmpeg -i "... file ..." -c:v libvpx-vp9 -pass 2 -b:v 1000K -threads 8 -speed 1 -tile-columns 6 -frame-parallel 1 -auto-alt-ref 1 -lag-in-frames 25 -c:a libopus -b:a 64k -f webm out.webm
[18:30:47 CEST] <Asuran> it results in slow transcoding
[18:30:56 CEST] <Asuran> and blaming about invalid aspect ratio (h264)
[18:31:13 CEST] <Asuran> and much overflow + trys and worst: only 1 cpu used
[18:31:16 CEST] <c_14> Phrk_: hmm, no idea. is there any other warning/error in the ffmpeg output?
[18:31:22 CEST] <Asuran> any suggestion?
[18:32:36 CEST] <Phrk_> c_14, no, every player that use ffmpeg say the same error message
[18:32:53 CEST] <Asuran> http://pastebin.com/wBy7bwqi
[18:33:14 CEST] <Asuran> fflogger, http://pastebin.com/wBy7bwqi
[18:34:36 CEST] <c_14> Asuran: that h264 message is from the decoder
[18:34:51 CEST] <Asuran> yup
[18:35:07 CEST] <Asuran> so what im supposed to do about it
[18:36:19 CEST] <c_14> Besides getting a better source?
[18:36:41 CEST] <c_14> Phrk_: no idea. Is it only with that files or with all files you try?
[18:37:14 CEST] <Asuran> not possible i guess and whats the problem with it? why only 1 thread used?
[18:37:45 CEST] <Phrk_> c_14, all files, on all directory
[18:39:22 CEST] <c_14> Asuran: as to the speed libvpx is slow. 0.6x realtime isn't that unrealistic afaik. As to why it only uses one thread, no idea. libvpx can be finicky sometimes but it should work
[18:39:58 CEST] <Asuran> ah realtime, but i dont care if its realtime
[18:40:07 CEST] <Asuran> i dont want streaming
[18:40:18 CEST] <Asuran> i just got these from: http://wiki.webmproject.org/ffmpeg/vp9-encoding-guide
[18:40:32 CEST] <c_14> Asuran: that was just the speed thing ffmpeg has in its output
[18:41:12 CEST] <pa> i see there's a NVENC encoder for ffmpeg? is it there out of the box? or is it a plugin i have to download?
[18:41:40 CEST] <c_14> you need the header from nvidia and then you need to --enable-nvenc
[18:41:52 CEST] <JEEB> and --enable-nonfree for now
[18:41:54 CEST] <c_14> Phrk_: let me test on my system
[18:42:08 CEST] <JEEB> there's a patch for the nonfree to be removed due to how the header was relicensed recently tho
[18:42:09 CEST] <pa> ah so i have to rebuild
[18:42:09 CEST] <pa> ok
[18:42:10 CEST] <pa> thnxk
[18:42:17 CEST] <pa> whats a fast encoder otherwise?
[18:42:22 CEST] <c_14> x264
[18:42:22 CEST] <JEEB> libx264
[18:42:23 CEST] <pa> something not too crappy but really fast
[18:42:28 CEST] <pa> ok thanks
[18:42:33 CEST] <c_14> adjust crf and preset to your liking
[18:42:52 CEST] <JEEB> people at work often try to tell me how they'd like to try out HW encoding
[18:42:54 CEST] <Asuran> ok i changed the command to the first example (VOD)? what means VOD? video on demand?
[18:43:07 CEST] <c_14> Asuran: probably
[18:43:12 CEST] <JEEB> then I have to tell them that HW encoding is usually only useful in a very limited set of cases
[18:43:14 CEST] <Asuran> thanks guys
[18:43:19 CEST] <Asuran> i appriaciate youre help :)
[18:43:40 CEST] <JEEB> at least HW encoding is no more as bad as it was back with the whole CUDA nonsense
[18:43:49 CEST] <JEEB> expensive heaters for everyone!
[18:47:02 CEST] <Asuran> only one thing bothers me now: how to see current frame progress and how much is leftß
[18:47:03 CEST] <pa> can i put h264 into ts?
[18:47:07 CEST] <furq> sure
[18:47:08 CEST] <pa> i seem to get errors..
[18:48:33 CEST] <c_14> Phrk_: works just fine here. Might be something with your sftp server
[18:55:59 CEST] <pa> if i choose -c:v h264, i always get:
[18:56:11 CEST] <pa> no frame!
[18:56:15 CEST] <pa> last message repeated N times
[18:58:41 CEST] <Asuran> is -tile-columns codec related?
[18:59:49 CEST] <furq> pa: that's a decoder error
[19:00:15 CEST] <furq> Asuran: ffmpeg -h encoder=libvpx-vp9
[19:00:22 CEST] <furq> it's specific to libvpx afaik
[19:02:09 CEST] <pa> furq, but if i don't specify the encoder, it works.. (like without -c:v h264)
[19:05:15 CEST] <Asuran> intresting ffmpeg crashed
[19:05:18 CEST] <Asuran> second time
[19:05:24 CEST] <Asuran> seems to be the source fault
[19:05:41 CEST] <Asuran> with crash i mean i didn't got any message
[19:06:24 CEST] <pa> ah
[19:06:26 CEST] <pa> i see
[19:06:37 CEST] <pa> so -c:v is intended as input codec?
[19:06:59 CEST] <Asuran> nope output
[19:07:00 CEST] <c_14> pa: depends where you write it
[19:07:16 CEST] <furq> pa: if you put it before -i
[19:07:21 CEST] <furq> which you normally don't need to do
[19:07:31 CEST] <pa> so after is output?
[19:07:36 CEST] <furq> yes
[19:07:43 CEST] <pa> ok i see the problem now :)
[19:07:44 CEST] <pa> thanks
[19:28:55 CEST] <izco> what should be the instruction to save file to your computer when i have an m3u8 link and a cookie that is required to download the file?
[19:36:35 CEST] <klaxa> izco: https://ffmpeg.org/ffmpeg-protocols.html#HTTP-Cookies ?
[19:36:37 CEST] <klaxa> does that help?
[19:38:31 CEST] <izco> will try, cheers
[20:44:36 CEST] <Mooj> greetings, you marvelously populated channel! i have come with a question about ffmpeg
[20:45:35 CEST] <Mooj> so here's the deal: I'm using ffmpeg in a python program i'm writing to convert .ogg files to .wav
[20:45:46 CEST] <Mooj> and then im doing stuff with the wav data
[20:46:24 CEST] <Mooj> but for some reason, the wav conversion always has 1073741823 audio frames, regardless of the original file
[20:46:52 CEST] <Mooj> which amounts to the beginning of that data being what i actually want, and then about 99.99% zeros
[20:47:44 CEST] <Mooj> now I could conceivably write a function to chop off that excess useless data, but that seems very un-elegant
[20:47:51 CEST] <Mooj> is there a better solution for this?
[20:48:36 CEST] <JEEB> are you using the FFmpeg API or the ffmpeg cli application?
[20:49:19 CEST] <Mooj> uh....i believe the latter, but im honestly not 100% sure
[20:49:34 CEST] <JEEB> at least I have in general noticed that ffmpeg cli writes relatively correct WAV files (except with the encoder delay included, which usually isn't that many frames) :P
[20:50:05 CEST] <Mooj> i mean i know theres an option for -aframes
[20:50:21 CEST] <Mooj> but im not sure how to have it detect the correct number of audioframes to use there
[20:50:30 CEST] <JEEB> yeah, but by default you should be getting the correct amount (plus encoder delay)
[20:50:40 CEST] <JEEB> like with ffmpeg -i hurr.ogg durr.wav
[20:51:55 CEST] <JEEB> so you're either using a bugged version or your compiler breaks things or if it happens with current HEAD then we have a booger
[20:52:17 CEST] <Mooj> eh.....here's a little more detail about what im doing: http://pastebin.com/9KTQe8ZC
[20:52:39 CEST] <Mooj> so im just piping the ogg through the converter
[20:52:48 CEST] <Mooj> annnnd yeah..
[20:53:03 CEST] <JEEB> oh
[20:53:24 CEST] <JEEB> not sure if you can get exact length from OGG
[20:53:32 CEST] <JEEB> I mean, without decoding it first
[20:53:39 CEST] <JEEB> (completely)
[20:53:55 CEST] <Mooj> isnt that what ffmpeg does when it converts though?
[20:54:03 CEST] <Mooj> i mean, i assumed
[20:54:07 CEST] <JEEB> yes, but then it can't update the value
[20:54:10 CEST] <JEEB> because the output is a pipe
[20:54:21 CEST] <JEEB> if the value is stored in the WAVE header
[20:54:33 CEST] <JEEB> try doing that with a file
[20:54:52 CEST] <JEEB> and see if that gets you correct length. if not, then things might be broken in your wave parser as well
[20:55:09 CEST] <Mooj> hm. okay hold on
[20:55:36 CEST] <JEEB> if you were using the API you would get the PTS of the last PCM sample
[20:55:49 CEST] <JEEB> or well, you would also have the PCM sample count of course
[20:57:00 CEST] <Mooj> ahhhh i have no idea what im doing xD
[20:57:57 CEST] <Mooj> sorry, this is my first time trying to write a program to do things with audio files
[20:58:41 CEST] <Mooj> so..i guess i will try to just have ffmpeg spit out a wav and run with that
[20:59:18 CEST] <JEEB> that way you would get the length correct, yes
[20:59:39 CEST] <JEEB> because ffmpeg would first write the data and then at the end seek to the header and update the value accordingly
[20:59:46 CEST] <JEEB> at least that's how I imagine that thing working
[20:59:59 CEST] <Mooj> yeah fair enough
[21:00:11 CEST] <JEEB> most things that handle WAV from stdin usually don't care about the length field
[21:00:16 CEST] <JEEB> and just read until stdin dies out
[21:00:52 CEST] <Mooj> it's a bit temporary of a fix since in the final version i will want to be able to handle streaming audio data
[21:00:57 CEST] <Mooj> but it'll be a start
[21:01:22 CEST] <Mooj> i assume reading until stdin dies out means just stopping once you hit 500 0s?
[21:01:54 CEST] <JEEB> the reading side should notice that as soon as ffmpeg stops pushing stuff in there
[21:02:09 CEST] <JEEB> I only imagine the python wrapper tries to do some fancy things
[21:02:14 CEST] <JEEB> and makes up data
[21:03:03 CEST] <Mooj> yeah...from what i can tell its just preallocating the memory, and then not chopping off the excess
[21:10:18 CEST] <furq> http://vpaste.net/3A1sf
[21:10:20 CEST] <furq> i'm bored apparently
[21:10:46 CEST] <furq> but yeah the length is at 0x4C
[21:11:16 CEST] <furq> er, 0x4A even
[21:13:47 CEST] <Mooj> so...it can tell how long the audio should be?
[21:14:06 CEST] <JEEB> no
[21:14:19 CEST] <JEEB> that's the spot where ffmpeg would then seek back
[21:14:21 CEST] <JEEB> if it was a file
[21:14:29 CEST] <JEEB> it's not so there's a placeholder value there
[21:14:34 CEST] <furq> ^
[21:14:34 CEST] <JEEB> capisce?
[21:15:01 CEST] <Mooj> gotcha
[21:53:20 CEST] <Mooj> yep, works like a charm when not in a pipe
[21:53:36 CEST] <Mooj> thanks for the help, you guys!
[23:36:05 CEST] <Prelude2004c> hey guys.. any suggestion.. .thi sis what is happening.. i have a channel being encoded.. and i set PVR to record a show... for some reason after i do the playback on the show... the segments are 6 seconds long.. basically the playback is -6 so it only plays for 6 sends from the end... odd.. its like end of file is at 0 and beginning i have no idea.. :(.. i dont even know how to explain it
[23:36:18 CEST] <Prelude2004c> anyone have any vauge idea on what i am dealing with here ?
[23:36:26 CEST] <Prelude2004c> i only play last 6 seconds of any recording
[00:00:00 CEST] --- Mon Apr 25 2016
1
0
[00:03:51 CEST] <BBB> super-creepy
[00:03:53 CEST] <BBB> ohwell
[00:03:59 CEST] <BBB> I dont see how my thing fixes anything though
[00:04:10 CEST] <BBB> it allows conversion
[00:04:13 CEST] <BBB> this is more like input
[00:04:32 CEST] <JEEB> + cross-library dependencies
[00:04:57 CEST] <JEEB> well basically I would like it if getting r,g,b or y,cb,cr values for one another would be centralized somewhere
[00:05:43 CEST] <JEEB> together with general conversion functionality for planes
[01:18:56 CEST] <BBB> JEEB: but most of it is much more purpose-specific than that
[01:19:14 CEST] <BBB> JEEB: e.g. in vf_colorspace, we convert to intermediates that are 15bit with limited overflow (but not absent)
[01:19:22 CEST] <BBB> JEEB: swscale has different rules
[01:19:30 CEST] <BBB> JEEB: and typcailly youd want clipped 8bit to 8bit
[01:19:33 CEST] <BBB> so its all different
[03:20:02 CEST] <cone-768> ffmpeg 03Michael Niedermayer 07master:a39e3e5e7b12: avutil/aes: Remove duplicate include
[11:13:26 CEST] <cone-659> ffmpeg 03Carl Eugen Hoyos 07master:0499012a5592: lavf/isom: Support Perian tag NELL for nellymoser.
[13:46:34 CEST] <wm4> wat...
[13:48:38 CEST] <fritsch> hehe
[14:17:54 CEST] <rcombs> ubitux: http://puu.sh/osIs4/c7dd3182b9.aac <-- does this look like AAC to you
[14:18:39 CEST] <rcombs> erm actually http://puu.sh/osIv2/29b9046e18.aac
[14:23:23 CEST] <ubitux> why do you ask me? i don't know much about aac
[14:23:34 CEST] <ubitux> close to nothing tbh
[14:24:50 CEST] <rcombs> uh I was aiming for atomnuker and failed spectacularly
[14:25:02 CEST] <rcombs> this is a good indicator that it's time for me to get some sleep
[14:27:24 CEST] <atomnuker> rcombs: looks fairly high entropy, might be quantized coefficients, doesn't look like huffman codes from scalefactors
[14:28:12 CEST] <rcombs> lavc doesn't like it, but I saw what looked very much like ADTS frame headers
[14:29:19 CEST] <rcombs> (playing around with Amazon Video)
[14:30:50 CEST] <atomnuker> you'd think they'd use encryption, wouldn't they?
[14:31:37 CEST] <rcombs> this is post-encryption, I _think_
[14:31:43 CEST] <rcombs> erm, decryption
[14:32:18 CEST] <rcombs> (set breakpoint on decryption function; let function run; examine contents of buffers)
[14:33:48 CEST] <atomnuker> they have to decoder the data somewhere too, and chances are they'll send full packets there
[14:34:12 CEST] <rcombs> Chrome provides two paths; one where the decryption plug-in also decodes, and one where it does not
[14:34:17 CEST] <rcombs> this takes the latter path
[14:35:40 CEST] <rcombs> doesn't help that I'm looking at a stripped Chrome binary and poking around in memory mostly-blind
[14:37:15 CEST] <wm4> can't you build a non-stripped one?
[14:37:23 CEST] <rcombs> I might try
[14:37:34 CEST] <wm4> just make sure you have gigabytes of memory
[14:37:34 CEST] <rcombs> unfortunately chromium is a behemoth of misery
[14:37:39 CEST] <JEEB> can chromium even use the drm libraries for chrome?
[14:37:45 CEST] <rcombs> I'd imagine so
[14:37:51 CEST] <JEEB> ok
[14:38:02 CEST] <rcombs> like, I don't see a _technical_ reason it couldn't
[14:38:08 CEST] <JEEB> and yes, building chromium is a very... *special* effort
[14:38:13 CEST] <rcombs> they're just PPAPI plugins
[14:38:42 CEST] <rcombs> they run 'em out-of-process and pass "decrypt" messages around
[14:38:43 CEST] <JEEB> at my previous job it was rather simple to trigger a xwalk build, which contained chromium :)
[14:39:01 CEST] <JEEB> and then you'd go for a long coffee break
[14:39:13 CEST] <JEEB> (when you updated your own middleware component)
[14:40:11 CEST] <rcombs> I was really looking for video and got sidetracked when I saw something that looked vaguely like AAC
[14:40:29 CEST] <rcombs> and manually parsed the ADTS header and went "!!!!!" because it looked sane
[14:40:34 CEST] <JEEB> :D
[14:40:38 CEST] <rcombs> (stereo, 48kHz&)
[14:40:39 CEST] <JEEB> I know that feeling
[14:40:59 CEST] <JEEB> in the end the last time I poked a DRM scheme I ended up just hooking ReadFile that the player was using
[14:41:09 CEST] <JEEB> (yes, it was the shitty type of DRM where you have to install a rootkit player)
[14:41:25 CEST] <rcombs> I was hoping to trap the VideoToolbox decode call
[14:41:41 CEST] <rcombs> but apparently they're not using that
[14:41:57 CEST] <rcombs> so& I guess they're using software decoding at expense of battery life and performance
[14:42:02 CEST] <rcombs> for the sake of CONTENT SECURITY
[14:42:05 CEST] <JEEB> yup
[14:42:26 CEST] <JEEB> and video is shown through the "secure route" available on macs and windows, amirite?
[14:42:27 CEST] <rcombs> also I guess they must pay for the codec licenses then
[14:42:32 CEST] Action: rcombs shrugs
[14:42:47 CEST] <JEEB> yes, google already pretty much pays the maximum amount
[14:42:50 CEST] <JEEB> so they don't really care
[14:42:51 CEST] <JEEB> :)
[14:43:16 CEST] <rcombs> I just took a screenshot and it came out normal
[14:43:20 CEST] <JEEB> löl
[14:43:30 CEST] <rcombs> so& I dunno whether or not "secure route" means that shouldn't work
[14:43:38 CEST] <JEEB> it should be a black box with the secure route
[14:43:43 CEST] <JEEB> so yeah, they're not even doing that :)
[14:44:28 CEST] <JEEB> I think netflix's implementation tried to use that one? never tried though
[14:45:24 CEST] <rcombs> also the screen recording feature in QuickTime works
[14:45:28 CEST] <wm4> and at least win d3d11 api is full of stuff for the secure route
[14:45:53 CEST] <wm4> keep in mind that DRM is for restricting the _consumer_, not pirates
[14:46:08 CEST] <JEEB> yup
[14:46:22 CEST] <rcombs> <insert xkcd link>
[14:46:26 CEST] <JEEB> and for various legal reasons
[14:46:27 CEST] <wm4> but if a simple screenrecorder works, then lol
[14:46:45 CEST] <JEEB> wm4: well they're not using the secure route in the amazon case it seems
[14:47:30 CEST] <rcombs> this is really more of a challenge than anything practical
[14:48:09 CEST] <rcombs> I guess since there must be an H.264 decoder in this binary somewhere, I could just& look for it, and trap on that
[14:48:22 CEST] <rcombs> can't be _that_ hard to find, looking at a mountain of disassembly
[14:49:52 CEST] <wm4> at some point they got to output video
[14:50:11 CEST] <rcombs> make it only work on the built-in display
[14:50:14 CEST] <wm4> knowing nothing about anything, this sounds like the easiest point of attack
[14:50:17 CEST] <rcombs> (am I evil enough)
[14:50:46 CEST] <rcombs> wm4: yeah but then you have to re-encode :P
[15:32:06 CEST] <JEEB> can someone confirm what the default value is for codec context's height f.ex.?
[15:32:11 CEST] <JEEB> -1 or 0
[15:38:20 CEST] <wm4> just check <= 0
[15:39:52 CEST] <JEEB> k
[16:42:08 CEST] <nevcairiel> rcombs: chromium can most definitely use the widevine drm module from chrome, and even play amazon video with it
[16:42:36 CEST] <nevcairiel> i set that up a couple weeks back, only took b uilding my own chromium to enable it
[17:36:30 CEST] <shibly> Hi, anyone there?
[18:37:48 CEST] <shibly> Anyone there?
[18:38:13 CEST] <fritsch> ask your question and check the topic of this channel ...
[18:39:01 CEST] <shibly> https://github.com/FFmpeg/FFmpeg , is it the official source code mirror of ffmpeg project?
[18:39:37 CEST] <nevcairiel> its an official mirror, yes
[18:39:52 CEST] <wm4> the official git is git://source.ffmpeg.org/ffmpeg.git
[18:40:04 CEST] <wm4> the github mirror often is behind by up to 1 hour or so
[18:40:24 CEST] <BtbN> doesn't github auto-pull?
[18:40:38 CEST] <JEEB> also why on earth is this person double-asking
[18:42:05 CEST] <shibly> If i want to be one of the maintainer of ffmpeg project, what should i have to do?
[18:42:28 CEST] <shibly> I have downloaded the source code of ffmpeg from here: https://www.ffmpeg.org/download.html
[18:42:32 CEST] <iive> start working on something and send patches.
[18:44:23 CEST] <shibly> I don't see any ticket or bug list there: https://github.com/FFmpeg/FFmpeg
[18:44:32 CEST] <JEEB> that's because that's just a mirror
[18:44:39 CEST] <JEEB> for the code
[18:44:48 CEST] <nevcairiel> shibly: start here https://www.ffmpeg.org/developer.html
[18:45:12 CEST] <JEEB> also I'm not really sure if you are going to be successful unless you have something specific in FFmpeg that you want to improve
[18:45:14 CEST] <nevcairiel> also https://www.ffmpeg.org/bugreports.html
[18:45:22 CEST] <JEEB> and yes, thanks nev for linking those
[18:45:26 CEST] <iive> btw, ffserver needs a lot of care and nobody else wants it. (wink) (wink)
[18:45:37 CEST] <JEEB> lol
[18:45:55 CEST] <JEEB> it needs to be redesigned and redone pretty much :P
[18:49:40 CEST] <wm4> could someone with access cleanup this shit https://github.com/FFmpeg/FFmpeg/pulls
[18:50:10 CEST] <BtbN> https://github.com/FFmpeg/FFmpeg/pull/201 ...what?
[18:50:34 CEST] <wm4> this happens often
[18:50:49 CEST] <nevcairiel> people try to use it as a bug report function but fail, and then this shit happens
[18:52:00 CEST] <BtbN> Hm, I wonder if it's worth it to start another discussion about whether the MIT-Licensed NVENC header makes it possible to drop nonfree with nvenc enabled.
[18:52:32 CEST] <BtbN> I think the last one was drowned out because it was also bundling the header
[18:52:37 CEST] <nevcairiel> if andreas was the only one complaining the last time, he is pretty much gone now =p
[18:53:01 CEST] <BtbN> People were primarily complaining about the bundled header
[18:53:21 CEST] <BtbN> Which I still think is a good idea, but I'll keep that for later, as it would side-track the license discussion
[21:30:28 CEST] <cone-103> ffmpeg 03Michael Niedermayer 07master:38433345e503: avcodec/utils: Assert that the number of consumed bytes in avcodec_decode_audio4() is <= the input size
[22:06:13 CEST] <ubitux> http://sprunge.us/dLRO is there a name for this algorithm?
[22:09:02 CEST] <fritsch> ubitux: https://en.wikipedia.org/wiki/Summed_area_table
[22:09:54 CEST] <ubitux> thanks, this saves me a large stupid comment
[22:11:07 CEST] <ubitux> crazy how math can make a simple thing unreadable
[22:11:30 CEST] <fritsch> that's by intention
[22:11:55 CEST] <ubitux> :(
[22:11:57 CEST] <fritsch> the problem is: formulating something 100% correct makes the notation a bit non funny
[22:12:25 CEST] <ubitux> mathematician should move to code
[22:12:29 CEST] <fritsch> also your example, seems to be a special version
[22:17:18 CEST] <fritsch> ubitux: and btw. in your image some color would have helped to understand the +a right away :-) cause the "top left coordinate" is ambigious again
[22:17:24 CEST] <fritsch> as there are many rectangles
[22:18:04 CEST] <ubitux> top-left origin of M maybe
[22:18:12 CEST] <ubitux> it's hard to put colors in ascii
[22:18:34 CEST] <fritsch> yes of M
[22:23:19 CEST] <wm4> ubitux: vt100 terminal escapes
[22:23:46 CEST] <ubitux> yeah, it will be perfect in source code
[22:23:56 CEST] <ubitux> syntax color override
[22:24:23 CEST] <ubitux> if i'm lucky it might even creates so digraph/trigraph
[22:24:30 CEST] <ubitux> to fuck things perfectly
[22:24:41 CEST] <ubitux> some*
[22:30:33 CEST] <atomnuker> color codes in C are just a step away from having HolyC's 3D models inside files having code
[22:37:29 CEST] <thardin> ffmpeg on templeos when?
[00:00:00 CEST] --- Sun Apr 24 2016
1
0
[01:28:26 CEST] <CoffeeFluxx> oh, is there a way to include chapters when muxing to mp4?
[04:18:20 CEST] <fling> How do I capture two sources at once?
[04:18:39 CEST] <fling> I mean I have two video inputs
[04:25:34 CEST] <relaxed> fling: ffmpeg -i one -i two
[04:27:47 CEST] <fling> -map 0:0 -map 1:0 -map 2:0
[04:27:52 CEST] <fling> relaxed: thanks :>
[04:45:30 CEST] <fling> why does first webcam freeze at the frame 15? -> https://bpaste.net/show/363f92d64635
[04:47:26 CEST] <fling> but everything works just fine when I capture separately
[04:47:27 CEST] <fling> hmm hmmmm
[11:04:22 CEST] <Wader8> hello
[11:04:48 CEST] <Wader8> trying to batch convert whole folder of files with this, doesn't quite work http://stackoverflow.com/questions/5784661/how-do-you-convert-an-entire-dir…
[11:05:40 CEST] <Wader8> basically it's a folder that stuff gets put into with same exact settings, then, ocassionally i would run to transcode it and delete the sources
[11:06:44 CEST] <furq> what about it doesn't work
[11:08:19 CEST] <Wader8> the cmd window closes, maybe I have to add other parameters to ffmpeg
[11:09:42 CEST] <Wader8> but i used the solution the people talked in the comment below the one with double quotes, i may try the original, while my filenames don't contain spaces i think, but they use underscores, I usually don't use spaces in such files thank god, i only use spaces on desktop shortcuts and light stuff
[11:09:52 CEST] <Wader8> i could use the original solution then,
[11:14:25 CEST] <Wader8> ah
[11:14:30 CEST] <Wader8> seems like old stuff
[11:14:33 CEST] <Wader8> doesn't work at all
[11:19:57 CEST] <Wader8> okay this worked http://forum.videohelp.com/threads/356314-How-to-batch-convert-multiplex-an…
[11:33:01 CEST] <Wader8> hi again
[11:33:08 CEST] <Wader8> this warning is always shown: Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
[11:33:27 CEST] <Wader8> i use loglevel verbose and v 9
[11:34:03 CEST] <Wader8> it's all ok, it's just that im not sure what the warning is about, doesn't matter if I use x264 params crf= or -crf
[13:51:44 CEST] <spirou> are there a way to make ffmpeg not change background color on some messages? I really don't like black background on stuff...
[13:54:42 CEST] <furq> you can set AV_LOG_FORCE_NOCOLOR to get rid of colours entirely
[13:55:03 CEST] <furq> i don't think there's a way to disable just background colours
[13:55:08 CEST] <furq> maybe file a feature request
[14:20:38 CEST] <spirou> ok
[14:23:08 CEST] <spirou> I guess best would be to make it possible to have it only do "bold" instead of setting a color, becuase then it would be the color I set my terminal window to use for bold :-)
[14:54:21 CEST] <spirou> would "PCM" be uncompressed audio? (nondestructive I mean)
[14:54:33 CEST] <Draz> Padding on video appears grey: Applying pad filter to video stream gives non-black padding. http://pastebin.com/vuHEVSGZ
[14:55:02 CEST] <Draz> I've tried a few things, as documented in pastebin. Help appreciated.
[14:56:10 CEST] <printline> Hi. What is the simplest way to read a video stream in h.264 from a webcam using ffmpeg as encoder? I've considered reading it in via UDP, but that would require depacketizing and possibly handling for dropped frames, which I want to avoid. Is there a way to read the stream output of ffmpeg directly from RAM, perhaps as a subprocess?
[14:58:24 CEST] <spirou> I found the description (if it is correct or not I down know) about what files my tv can use from the usb-stick, and for .mkv it is "video: H.264, MPEG1,2,4 audio:EAC3/AC3" but for .mp4 it instead lists "video: MPEG4, Xvid * H.264 audio: PCM/MP3" so I'm curious about what the PCM would be
[14:58:37 CEST] <spirou> and how to set ffmpeg to do that if I would like to
[14:59:02 CEST] <spirou> *don't know
[15:02:40 CEST] <furq> spirou: https://en.wikipedia.org/wiki/Pulse-code_modulation#Implementations
[15:04:06 CEST] <furq> i have no idea why it wouldn't support mp3 in mkv though
[15:04:40 CEST] <furq> personally i would probably just remux and then see if it works, those docs are always flaky
[15:07:14 CEST] <spirou> yeah I was thinking about testing some different variants and see what it really can do
[15:09:23 CEST] <spirou> and when it say it can do pcm, would that mean everything listed on https://trac.ffmpeg.org/wiki/audio%20types and making a mp4 with it would be using for example -acodec pcm_s16le
[15:15:53 CEST] <spirou> aha I would need to write -f s16le -acodec pcm_s16le for that. I test with one of those too. and a mkv/mp3 version.
[15:20:59 CEST] <furq> just `-c:a pcm_s16le out.mp4` should do
[15:21:17 CEST] <furq> i'd try it with aac and mp3 though since that should cover most of what you've got
[15:21:34 CEST] <furq> pcm audio will be ~10x the size
[15:21:45 CEST] <spirou> ok
[15:22:25 CEST] <spirou> what is "global headers" btw? I assume the warning is about it is adding those becuase mkv needs that
[15:23:46 CEST] <furq> something like that
[17:22:30 CEST] <DHE> API question, is there a reason multi-threaded decoding and/or encoding is not the default?
[17:33:24 CEST] <kepstin> it is in codecs which support automatically determining a suitable number of threads
[17:34:07 CEST] <kepstin> oh, in the api, huh.
[17:34:14 CEST] Action: kepstin doesn't know if it's different there
[17:37:00 CEST] <DHE> I'm just following the examples and not getting multi-threaded decoders. seems my CPU needs it to reliably decode H264
[17:37:15 CEST] <shibly> Hi, what does ffmpeg do?
[17:37:27 CEST] <JEEB> ffmpeg or FFmpeg
[17:37:30 CEST] <JEEB> FFmpeg is project
[17:37:30 CEST] <DHE> I've been explicitly enabling it for encoders because I have specific needs
[17:37:38 CEST] <JEEB> ffmpeg is the command line tool included in FFmpeg
[17:38:10 CEST] <shibly> What's the difference between ffmpeg and FFmpeg?
[17:38:12 CEST] <JEEB> also please don't double-post on -devel :P
[17:38:16 CEST] <JEEB> I just noted the difference :P
[17:38:23 CEST] <kepstin> DHE: do these files play fine with e.g. ffplay?
[17:38:31 CEST] <JEEB> one is a tool in a project and the other is the project itself
[17:38:32 CEST] <shibly> What type of project?
[17:38:38 CEST] <JEEB> software project
[17:38:42 CEST] <kepstin> DHE: you could look at what that tool does to enable multithreaded decoding and copy it if so :/
[17:38:56 CEST] <shibly> What does that FFmpeg project do?
[17:39:12 CEST] <JEEB> FFmpeg contains libraries to handle reading, decoding, filtering, encoding and writing/passing multimedia data
[17:39:20 CEST] <DHE> kepstin: I know HOW to. The program works. I'm asking if there's a reason it isn't the default. H264 in realtime is a bit flaky in terms of being able to keep up
[17:39:35 CEST] <JEEB> FFmpeg also contains tools such as ffprobe, ffmpeg and ffplay that are example usages (kind of) of those libraries
[17:39:36 CEST] <DHE> I just turned on multithreaded decoding to try and keep up
[17:40:24 CEST] <JEEB> when you talk of ffmpeg you generally talk of the ffmpeg tool within FFmpeg, but how you phrased your question made me want to affirm what you wanted to ask about
[17:40:38 CEST] <JEEB> s/affirm/confirm/
[17:40:54 CEST] <shibly> How can i read multimedia data? Suppose i have a file named p.avi
[17:41:08 CEST] <shibly> What would i do reading multimedia data?
[17:41:18 CEST] <JEEB> do you just want to use the ffmpeg tool or actually the APIs?
[17:41:21 CEST] <DHE> traditionally get raw RGB or YUV frame pixels
[18:24:31 CEST] <shibly> There was some problem with internet here.
[18:24:46 CEST] <shibly> How can i read p.avi data with ffmpeg?
[18:24:58 CEST] <shibly> What can i do with reading p.avi data?
[18:25:55 CEST] <JEEB> < JEEB> do you just want to use the ffmpeg tool or actually the APIs?
[18:26:15 CEST] <shibly> I want to use api
[18:26:20 CEST] <JEEB> then take a look at the examples
[18:26:28 CEST] <JEEB> you will be opening the file with avformat
[18:26:41 CEST] <JEEB> you will be then presented packets from the streams and you usually feed them to avcodec for decoding
[18:26:56 CEST] <JEEB> from which you received decoded pictures in the colorspace they are in (usually YCbCr)
[18:27:00 CEST] <JEEB> and so forth
[18:27:20 CEST] <JEEB> shibly: take a look at these examples https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples
[18:27:46 CEST] <JEEB> demuxing and decoding is the first thing I guess
[18:28:02 CEST] <JEEB> after that you can look into what you want to do with that data you get out of it
[18:37:14 CEST] <shibly> https://github.com/FFmpeg/FFmpeg , is it the official source code mirror of ffmpeg project?
[18:39:23 CEST] <JEEB> it's an official mirror and in general has more fancy C syntax highlighting etc
[18:39:32 CEST] <JEEB> so if I want to quickly link something I usually do that :P
[18:39:44 CEST] <JEEB> for actual cloning you will want to use git.videolan.org
[18:39:49 CEST] <JEEB> since that's where the main repo resides
[18:39:58 CEST] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=summary
[18:41:02 CEST] <andrey_utkin> how could I get AVCC-formatted H.264 packets out of libx264 encoder (AFAIU it produces Annex B)? Would it be viable to implement BSF for this?
[18:43:01 CEST] <DHE> andrey_utkin: there's a BSF for the opposite direction actually...
[18:43:08 CEST] <JEEB> http://git.videolan.org/?p=x264.git;a=blob;f=x264.h;h=5581ab9b9971ea8eb9a1b…
[18:43:22 CEST] <JEEB> andrey_utkin: you can already get the packets out of x264 in that format by setting the b_annexb to correct value. you will not get extradata pre-created of course...
[18:43:36 CEST] <JEEB> but that way you already get the length in the beginning
[18:43:49 CEST] <JEEB> so you don't have to poke the NAL units
[18:45:50 CEST] <DHE> multi-threaded decoding seems to have solved my issues...
[18:54:13 CEST] <andrey_utkin> JEEB, oh nice, thanks a lot, that allows me to reencode selected GOPs (with others remuxed) in mp4 files without convertion to mpegts and back.
[18:54:42 CEST] <andrey_utkin> with stitchable x264 option of course
[19:07:33 CEST] <Amitari> Anyone who knows how to batch-remux a whole directory of .ts-files into MKV?
[19:07:47 CEST] <Amitari> Doing it manually with MKVToolnix is so bothersome.
[19:08:01 CEST] <BtbN> ffmpeg -i something.ts -c copy something.mkv
[19:09:11 CEST] <Amitari> Thanks, but how do I make them get saved with the same file name, but a different extension? Just switching "something" to an asterisk usually causes problems.
[19:10:01 CEST] <BtbN> Use your shell to iterate over all the files and modify the filename string
[19:11:07 CEST] <Amitari> Uh, I don't know how to do that. :(
[19:11:12 CEST] <Amitari> Isn't there any other way?
[19:11:45 CEST] <BtbN> Well, you could it manualy for every single file...
[19:12:19 CEST] <Amitari> Ouch!
[19:12:23 CEST] <Amitari> You sure there isn't any way?
[19:12:34 CEST] <BtbN> The one i just mentioned
[19:12:41 CEST] <BtbN> No idea what's so bad about that
[19:12:55 CEST] <anadon> I'm getting garbage frames from the linked code and I'm not sure what could be going wrong: http://pastebin.com/3MGivRdv
[19:12:57 CEST] <Amitari> I don't understand what you mean.
[19:12:59 CEST] <BtbN> it's a plain for loop over *.ts, and some ${f/.ts/.mkv}
[19:13:14 CEST] <Amitari> Oh, I'm just not very well-versed in that.
[19:14:30 CEST] <Amitari> But I don't understand what you mean with "for loop over" and all that jazz.
[19:17:23 CEST] <furq> Amitari: what os
[19:17:39 CEST] <Amitari> GNU/Linux, Arch, BASH.
[19:18:06 CEST] <furq> for f in *.ts; do ffmpeg -i "$f" -c copy "${f%.*}.mkv"; done
[19:19:20 CEST] <Amitari> Thanks!
[19:19:26 CEST] <Amitari> I'll save that to my command collection. :P
[19:20:23 CEST] <furq> you may also want to add -map 0 if you have multiple audio/subtitle tracks
[19:20:58 CEST] <Amitari> Where in the command should that be added?
[19:21:06 CEST] <furq> between " and -
[19:21:51 CEST] <Amitari> And this won't cause any negative effects if I do it on a clip that doesn't have multiple tracks?
[19:21:56 CEST] <furq> no
[19:22:17 CEST] <Amitari> Good.
[19:22:19 CEST] <Amitari> Thanks! :D
[21:03:15 CEST] <q3cpma> Hello, I'm trying to extract some subrip from a mkv with "ffmpeg -i my.mkv -map 0:3 -c:s copy out.sub" but it gives me Could not write header for output file #0 (incorrect codec parameters ?): Operation not permitted. Any idea why?
[21:06:28 CEST] <relaxed> q3cpma: try, ffmpeg -i my.mkv -map 0:3 -c:s copy out.srt
[21:07:41 CEST] <q3cpma> relaxed: Now it says "Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument" (ffprobe says that it's dvd_subtitle)
[21:08:15 CEST] <q3cpma> relaxed: With "[srt @ 0x2388640] Unsupported subtitles codec: dvd_subtitle" before. Expected.
[21:08:52 CEST] <relaxed> pastebin the command and output
[21:09:19 CEST] <furq> you can't automatically convert dvd_subtitle to srt
[21:09:20 CEST] <relaxed> dvd subtitles are images
[21:09:34 CEST] <furq> i'm not sure what container to use there though
[21:09:40 CEST] <q3cpma> relaxed: Yeah I know, that's why my original command was -> out.sub
[21:10:08 CEST] <furq> try without -c:s copy
[21:10:25 CEST] <q3cpma> At least my subs work now, I'm using a pretty old ffmpeg (2.8.6)
[21:10:25 CEST] <furq> vobsub and dvd_subtitle aren't the same iirc
[21:10:39 CEST] <q3cpma> furq: What's the extension for it, then?
[21:10:52 CEST] <furq> i guess .sub for vobsub
[21:11:12 CEST] <furq> iirc ffmpeg doesn't have very good support for this but i could be getting it conflated with its poor support for ifo
[21:11:13 CEST] <q3cpma> furq: Without -c:s copy, I get "Encoder (codec microdvd) not found for output stream #0:0"
[21:12:47 CEST] <relaxed> what are you trying to do with these subs?
[21:13:10 CEST] <q3cpma> relaxed: They don't show while using mpv or ffplay (but they do with a mpv/ffmpeg git)
[21:13:14 CEST] <furq> you should be able to mux them directly into an output video
[21:13:20 CEST] <q3cpma> So i was trying to examine them.
[21:13:22 CEST] <furq> i don't know if there's a standalone container for dvd_subtitle
[21:13:34 CEST] <furq> and i don't know if ffmpeg can convert them to anything which does have one
[21:13:53 CEST] <q3cpma> Could I put i into a .vob?
[21:13:54 CEST] <furq> you could just mux them into an mkv i guess
[21:15:28 CEST] <q3cpma> Well, anyway, I can watch my shit with git ffmpeg/mpv, so I suppose there's no problem.
[21:15:36 CEST] <q3cpma> Thanks for the help.
[21:15:45 CEST] <relaxed> why use anything else? :)
[21:15:57 CEST] <q3cpma> relaxed: Cause gentoo
[21:16:17 CEST] <furq> is cpma still going
[21:16:24 CEST] <furq> i see promode.org is still dead
[21:16:41 CEST] <q3cpma> furq: Still play it with a friend. But yeah, the mod is dead (or finished)
[21:16:53 CEST] <furq> doubtless there are still some diehards playing ca on q3dm6
[21:17:08 CEST] <q3cpma> furq: I'd say cpm22 more
[21:17:17 CEST] <furq> not for ca
[21:17:29 CEST] <furq> i don't think i ever saw a public server on any other map
[21:17:43 CEST] <q3cpma> duel is all that matter :)
[21:17:54 CEST] <furq> aerowalk is pretty great
[21:18:10 CEST] <furq> it's better in qw though
[21:18:17 CEST] <q3cpma> qw?
[21:18:20 CEST] <furq> quakeworld
[21:18:31 CEST] <furq> the original and best
[21:18:31 CEST] <q3cpma> Must admit I never tried.
[21:18:35 CEST] <relaxed> that's the floating arena map?
[21:18:54 CEST] <furq> https://www.youtube.com/watch?v=kcgh9UNIN60
[21:19:04 CEST] <q3cpma> relaxed: Nah, vertical and small map with teleporters
[21:19:23 CEST] <q3cpma> https://i.ytimg.com/vi/ny4y30-N7wo/maxresdefault.jpg
[21:20:07 CEST] <q3cpma> furq: Looks nice. Is the netcode as good as cpma?
[21:20:32 CEST] <furq> it's pretty good
[21:20:37 CEST] <furq> idk if it has the fancy lag compensation stuff cpma has
[21:21:48 CEST] <q3cpma> Actually, I don't think a cpma player would feel like a stranger. Looks pretty similar.
[21:22:07 CEST] <emilsp> hello, what's the state of h/w accelerated video encode on android, if there is such a thing ?
[21:22:11 CEST] <furq> cpm's movement is based on qw's
[21:22:21 CEST] <furq> aside from the stuff which is from q2 and q3
[21:25:54 CEST] <Mavrik> emilsp, yp, MediaCodec
[21:26:21 CEST] <Mavrik> 18+ for a decendly compatible implementation
[21:27:34 CEST] <emilsp> Mavrik, I'm looking for a C only imlementation
[21:28:15 CEST] <Mavrik> yeah, no.
[21:28:17 CEST] <emilsp> is there nothing useable on the native side ? The application I am developing would not be running as a regular android app but rather as a development tool
[21:28:20 CEST] <emilsp> oh well
[21:28:46 CEST] <Mavrik> It's a silly requirement though.
[21:28:55 CEST] <Mavrik> If you want to interact with Android, you have to do it via Java.
[21:28:56 CEST] <emilsp> since I want to access /dev/graphics/fb0, I will need adb access rights
[21:29:02 CEST] <Mavrik> huh.
[21:29:08 CEST] <emilsp> the same with input events
[21:29:13 CEST] <Mavrik> What are you trying to do?
[21:29:27 CEST] <Mavrik> Since that sounds very wrong.
[21:29:29 CEST] <emilsp> I am trying to write a non-intrusive android application automation thing
[21:29:43 CEST] <emilsp> this is for my uni project, so it is very wrong
[21:29:57 CEST] <emilsp> because 'in the real world' I would be using HDMI output to obtain the video stream from a phone
[21:30:18 CEST] <Mavrik> Is there a reason why aren't you using Android's API to take screen?
[21:30:27 CEST] <Mavrik> And then use the same API to move surfaces to HW encoder_
[21:30:27 CEST] <emilsp> is there one ?
[21:30:30 CEST] <Mavrik> Yes.
[21:30:39 CEST] <emilsp> could you please refer me to it ?
[21:31:19 CEST] <Mavrik> There's an opensource app that does that: https://github.com/JakeWharton/Telecine/blob/master/telecine/src/main/java/…
[21:31:28 CEST] <Mavrik> There's also an adb command that starts recording to an mp4 file.
[21:32:19 CEST] <emilsp> I need to annotate the stream frame by frame with the unix timestamp, so just piping to a unix socket (instead of a file) wouldn't work to push the stream to a host
[21:32:27 CEST] <emilsp> thank you very much Mavrik
[21:32:48 CEST] <Mavrik> Well, then take the frame, add timestamp and pass it to mediacodec encoder
[21:33:47 CEST] <emilsp> I was expecting to just have a struct, holding the delta frame/reference frame and the actual timestamp :(
[21:34:07 CEST] <Mavrik> huh
[21:34:16 CEST] <Mavrik> Well depends what you're doing.
[21:34:34 CEST] <emilsp> Mavrik, wouldn't your previous suggetion imply that I just overlay the timestamp on the actual frame ?
[21:35:09 CEST] <Mavrik> I guess, isn't that what you want? :P
[21:35:30 CEST] <emilsp> no, I'd rather not mess with the frame itself
[21:35:54 CEST] <emilsp> since I am going to send the frames off to a host anyway, I could just 'package up' each frame with the timestamp
[21:39:25 CEST] <emilsp> so, in an ideal world, I could memcopy from /dev/graphics/fb0, get_timespec(); pass the memcopied framebuffer to the encoder, receive a new frame from the encoder and push both the timestamp and the frame off to a server, so the server would basically receive the something along the lines {'timestamp':1333564552423, alpha_frame:"", delta_frame:""}
[21:39:35 CEST] <emilsp> is what I am envisioning to do completely mental
[21:39:39 CEST] <emilsp> ?
[21:46:40 CEST] <Mavrik> Yes.
[21:46:52 CEST] <Mavrik> Because there's no guarantee fb0 will actually have what you want or work.
[21:46:59 CEST] <Mavrik> Because Android isn't your desktop Linux distro.
[21:47:07 CEST] <Mavrik> Use proper APIs for that in Java.
[21:48:58 CEST] <emilsp> Mavrik, the ADB command that you suggested does exactly the same thing
[21:49:14 CEST] <Mavrik> ok.
[21:49:28 CEST] <emilsp> it's just that I have no experience with video encoding and decoding at all
[21:50:25 CEST] <emilsp> and wrt to the adb command, the only difference is that it outputs the framebuffer to stdout or to a file without grabbing timestamps
[21:53:35 CEST] <Mavrik> It's not outputing the framebuffer from /dev
[21:53:45 CEST] <Mavrik> But it creates a surface composer client
[21:53:49 CEST] <Mavrik> Just like proper API does.
[21:54:12 CEST] <emilsp> https://android.googlesource.com/platform/frameworks/base/+/jb-release/cmds…
[21:55:36 CEST] <Mavrik> screenrecord is this: https://android.googlesource.com/platform/frameworks/av/+/kitkat-release/cm…
[21:55:51 CEST] <anadon> I'm getting garbage frames from the linked code and I'm not sure what could be going wrong: http://pastebin.com/3MGivRdv
[21:56:16 CEST] <Mavrik> It uses private APIs which you don't have guaranteed as a user.
[21:56:33 CEST] <emilsp> Mavrik, I am aware of that
[21:56:41 CEST] <emilsp> and yes, you are correct
[21:56:51 CEST] <emilsp> I guess I'll have to do this at the Java level
[21:59:12 CEST] <Mavrik> You can try doing it via framebuffer but it might not do what you want
[21:59:24 CEST] <Mavrik> Those APIs aren't stable or standard across devices
[22:00:36 CEST] <emilsp> yes, I am awareo of that, I feel like you have made me reonsider the way I approach the problem
[22:01:12 CEST] <emilsp> but still, all the examples using MediaCodec deal with streams, which essentially means that i don't get to granularly 'pick out' my frames
[22:03:31 CEST] <Mavrik> But you want to encode video to h.264 right?
[22:04:14 CEST] <emilsp> anything that is optimal, really
[22:04:33 CEST] <emilsp> I don't expect to be able to just push raw pixel values over wifi :(
[22:04:43 CEST] <Mavrik> That's kind of important.
[22:04:47 CEST] <Mavrik> You're talking about picking out frames.
[22:04:50 CEST] <Mavrik> But videos ARE streams.
[22:04:55 CEST] <emilsp> I am aware of that
[22:04:59 CEST] <Mavrik> So not sure which part did I misunderstand? :)
[22:05:13 CEST] <Mavrik> How many frames do you want to send per second?
[22:05:29 CEST] <emilsp> so, video's are streams where a frame may consist of a reference frame or a combination of a reference frame and delta frames, right ?
[22:06:11 CEST] <Mavrik> Um, video is a sequence of frames.
[22:06:13 CEST] <Mavrik> They have timestamps.
[22:06:22 CEST] <Mavrik> When encoded they may reference other frames to be fully constructed.
[22:06:29 CEST] <emilsp> yes
[22:06:36 CEST] <Mavrik> But usually you don't care about that since encoders / decoders handle that for you.
[22:06:39 CEST] <emilsp> but the timestamps are local to the video, are they not ?
[22:06:48 CEST] <Mavrik> You essentially get a sequence of timestamped frames.
[22:07:25 CEST] <Mavrik> Yes, timestamps are in video format specific units and don't reflect actual wall clock
[22:08:00 CEST] <emilsp> so, if I were to take a systemwide timestamp at the start of the recording, would or would I not be able to then 'correlate' the video timestamps with the system timestamp at the start of the recording ?
[22:08:48 CEST] <Mavrik> You could :)
[22:08:50 CEST] <emilsp> I am expecting to correlate different frames with different input events, (touch inputs), so I can deteremine what parts of the UI have changed and thus signify what input is epxected to drive an application to the next step
[22:09:00 CEST] <Mavrik> Provided that timestamps are of course correct.
[22:09:01 CEST] <emilsp> but would it be precise ?
[22:09:10 CEST] <emilsp> hmm, and what about dropped frames ?
[22:09:19 CEST] <Mavrik> Dropped frames just aren't there.
[22:09:38 CEST] <Mavrik> Remember, video has to be precise to the ranges of 1/60th of a second or even more ;)
[22:09:42 CEST] <emilsp> but the timestamps for the next frames are still 'relative' to the first frame, right ? not just +1
[22:09:48 CEST] <Mavrik> Even though, I have a feeling that you'd probably get away with screenshots.
[22:09:55 CEST] <Mavrik> Since you usually don't get 60 input events per second.
[22:10:02 CEST] <emilsp> you'd be surprised
[22:10:31 CEST] <Mavrik> The issue you'll mostly have is that a lot of players don't really handle variable framerate well :)
[22:10:32 CEST] <emilsp> anyway, thank you very much for not being dismissive and being helpful and constructive in this discussion
[22:10:56 CEST] <Mavrik> the timestamps for frames are always properly relative between themselves
[22:11:11 CEST] <Mavrik> e.g. if you have a 25fps video with 1/90000 timebase
[22:11:21 CEST] <Mavrik> first frame will have timestamp 0, second 3600, third 7200, etc.
[22:11:42 CEST] <emilsp> I am going to implement 'the player' anyway, since I will basically map input events with frames on the host side and analyze the thing frame by frame and I think also in a streamish manner to determine movement
[22:11:45 CEST] <Mavrik> and 25th (at exactly 00:00:01.000 play time) will have timestamp of 90000
[22:12:08 CEST] <emilsp> yes
[22:17:38 CEST] <Mavrik> emilsp, consider if you can get away by just sending a stream of JPEG screenshots
[22:17:45 CEST] <Mavrik> Those aren't all that bandwidth intensive
[22:18:01 CEST] <emilsp> those are resource intensive, are they not ?
[22:18:41 CEST] <Mavrik> less than encoding video
[22:18:47 CEST] <emilsp> well, at least when I attempted to solve this problem in an 'easier' way in the past, piping adb screencap -p | dos2unix >> screenshot gave me about 5fps
[22:18:50 CEST] <Mavrik> they're bigger bandwidth wise
[22:19:06 CEST] <Mavrik> that sounds inefficient :)
[22:19:19 CEST] <Mavrik> What's your required FPS?
[22:19:41 CEST] <emilsp> honestly, I've no idea, but I guess 5fps wouldn't really cut it
[22:20:54 CEST] <Mavrik> Try to check that first, dragging in video encoding if you don't need it is a lot of unnecessary work :)
[22:21:25 CEST] <Mavrik> After all, even most security cameras get away with MJPEG ;)
[22:24:06 CEST] <emilsp> well, this is all in all a resarchy PoC thing to get a piece of paper that says I can do what I've been doing for about two years already, so I'm guessing resorting to the least amount of effort thing (which would be adb screencap -p | dos2unix ) isn't excatly what I want to do here
[22:24:52 CEST] <emilsp> and in the real world, I would be getting HDMI output, so I guess I have to make something that would emulate a unix fd that outputs a video stream
[22:25:12 CEST] <emilsp> otherwise the PoC doesn't really replicate the way this would work, if it even would be able to work
[23:21:56 CEST] <mixfix41> ffplay isn't playing to hdmi through rpi2 on slackware arm
[23:22:07 CEST] <mixfix41> with no display how to get it hearing sound?
[23:22:16 CEST] <sfan5> try using a proper video player
[23:22:44 CEST] <c_14> set up your audio device correctly (alsa/pulse)
[23:26:26 CEST] <mixfix41> maybe I gotta enable mixing lets see
[23:34:31 CEST] <Prelude2004c> hey guys.. need some urgent help
[23:34:33 CEST] <Prelude2004c> http://pastebin.com/ustVEMaa
[23:34:41 CEST] <Prelude2004c> see problem in pastebin below
[00:00:00 CEST] --- Sun Apr 24 2016
1
0
[03:39:21 CEST] <jamrial> Daemon404, wm4: valgrind is complaining after either the latest merge batch or the new decode api
[03:40:21 CEST] <jamrial> vc1 tests only
[04:50:13 CEST] <cone-224> ffmpeg 03Michael Niedermayer 07master:268b5ae10aa6: doc/APIchanges: Fix bistream typo
[08:11:03 CEST] <cone-511> ffmpeg 03Lou Logan 07master:3eafbbe70a10: doc/APIchanges: Fix bitsream typo
[09:12:45 CEST] <wm4> hm will take a look at the vc1 valgrind thing later
[10:51:05 CEST] <fritsch> wm4: do you have an email adresse that I can use when writing to the intel mesa / drm folks for a EGL image interop possibility for hevc-10bit?
[10:52:50 CEST] <wm4> fritsch: you could use the one I have on github and the ffmpeg mailing list
[10:53:01 CEST] <fritsch> i looked into your last commit
[10:53:04 CEST] <fritsch> wm4@nowhere
[10:53:07 CEST] <fritsch> but i will find it :-)
[10:53:12 CEST] <wm4> https://github.com/wm4/
[10:53:25 CEST] <wm4> also, remind those guys that D3D11 provides this just fine on Windows
[10:53:36 CEST] <fritsch> Will do
[10:54:43 CEST] <fritsch> wm4: G16, GR1616 would be fine I think for now
[10:54:51 CEST] <fritsch> but let's see what they reply to us
[10:55:04 CEST] <fritsch> i added everyone that contributed to the last mesa changes
[10:56:06 CEST] <wm4> yep, I'd expect a 16 bit fixed-point texture
[10:57:54 CEST] <fritsch> sent
[10:58:06 CEST] <fritsch> if something is technically wrong, does not matter - they know we are coming now :-)
[10:58:09 CEST] <fritsch> and need something
[10:59:14 CEST] <wm4> thanks for doing this
[10:59:44 CEST] <fritsch> i have a selfish own interest in that :-)
[10:59:58 CEST] <fritsch> as I want to watch future live tv
[11:00:17 CEST] <fritsch> nope - the real reason is: i know the oss world stuff for too long ... have seen too many ugly workarounds
[11:00:26 CEST] <fritsch> that all had some disadvantages at a point
[11:00:33 CEST] <fritsch> so we need to do it right from the beginning
[11:00:39 CEST] <fritsch> and that's the way to go
[11:02:27 CEST] <wm4> yeah, with the EGL interop, video rendering with vaapi finally becomes less of a joke
[11:03:06 CEST] <wm4> those who are into doing TVs with Linux seem to be more into using Intel's fixed-function video rendering, though
[11:03:18 CEST] <wm4> to the point of declaring video rendering with OpenGL a ridiculous idea
[11:03:39 CEST] <fritsch> yeah - I also had such a discussion
[11:03:44 CEST] <fritsch> with a wayland dev ...
[11:03:50 CEST] <wm4> me too
[11:03:51 CEST] <fritsch> an gstreamer highpriest
[11:04:58 CEST] <fritsch> kodi's some million users with an upcoming perfect BXT setup also have some influence
[11:05:05 CEST] <fritsch> so - it will work out - I am sure
[11:06:14 CEST] <wm4> surely it will
[14:22:00 CEST] <Daemon404> wm4, did you look into that valgrind thing?
[14:26:08 CEST] <wm4> Daemon404: not yet, but will do today
[14:27:01 CEST] <Daemon404> ok
[15:37:36 CEST] <atomnuker> Gramner: can you check if I'm doing something obviously wrong in this piece of assembly: https://0x0.st/Zez.diff ?
[15:39:35 CEST] Action: Daemon404 wonders why firefox wont display diff as text
[15:45:07 CEST] <durandal_1707> don't use Firefox
[15:51:13 CEST] <wm4> get the "Open in Browser" extension and never worry again
[15:55:56 CEST] <wm4> meh every time I pull and rebuild, the build first goes well, and then errors because configure wasn't rerun
[15:56:36 CEST] <Daemon404> this happens when new codecs or demuxers are added
[15:56:52 CEST] <wm4> I know (but usually it's due to new filters)
[15:57:25 CEST] <Daemon404> ideally the build system would be smart enough
[15:57:29 CEST] <Daemon404> (based on date of all*.c)
[15:57:31 CEST] <Daemon404> ideally.
[15:58:10 CEST] <wm4> in theory it prints a warning message at some point
[15:58:54 CEST] <wm4> so, how do I run fate with valgrind?
[16:00:23 CEST] <nevcairiel> configure --toolchain=valgrind
[16:01:03 CEST] <nevcairiel> or rather --toolchain=valgrind-memcheck
[16:01:06 CEST] <wm4> uh, so rebuilding again
[16:06:56 CEST] <Shiz> can't make it easy now can you
[16:07:28 CEST] <wm4> well I still don't know how to make fate use vaslgrind
[16:09:39 CEST] <wm4> seems like I needed TARGET_EXEC=valgrind
[16:10:09 CEST] <wm4> I did "make fate-vc1 -j4 SAMPLES=../fate/ TARGET_EXEC=valgrind" but nothing shows up, seems all fine
[16:13:36 CEST] <Daemon404> wm4, try with THREADS=N? i dunno
[16:13:44 CEST] <Daemon404> or wait for jamrial to appear
[16:13:57 CEST] <wm4> added THREADS=4, nothing
[16:14:12 CEST] <nevcairiel> wm4: if you configure with that it should use the target exec automatically
[16:14:45 CEST] <wm4> nevcairiel: it appears it didn't
[16:14:48 CEST] <BBB> Daemon404: is the h264 bug still there?
[16:16:17 CEST] <Daemon404> BBB, yes
[16:18:19 CEST] <BBB> Daemon404: where is the file?
[16:18:24 CEST] <BBB> and trac thing?
[16:22:30 CEST] <Daemon404> https://trac.ffmpeg.org/ticket/5458
[16:22:33 CEST] <Daemon404> all info and file there
[16:37:10 CEST] <BBB> did you try with er turned off?
[16:37:16 CEST] <BBB> assuming the bug is b/c the file is damaged
[16:41:27 CEST] <BBB> Ill look at it maybe after lunch or so, have had a long long week so a little downtime would be nice
[16:49:20 CEST] <Gramner> atomnuker: does that asm work correctly?
[16:55:20 CEST] <atomnuker> Gramner: no, every 2 samples appear darker and seem to be clipping
[16:55:57 CEST] <atomnuker> so I was wondering if maybe the values were too large for pmuhswr
[16:55:59 CEST] <Gramner> you seem to be reversing the bytes in each pixel
[16:56:22 CEST] <Gramner> or is that supposed to happen?
[16:57:04 CEST] <atomnuker> yes, they have different endianess
[17:00:03 CEST] <Gramner> 10-bit values are fine with pmulhrsw
[17:01:15 CEST] <Gramner> 0x7fff for multiply by 1 starts to fail with inputs larger than 0x4000
[17:02:13 CEST] <Gramner> but you don't reverse the bytes in the c version?
[17:02:27 CEST] <Gramner> av_le2ne32 is a no-op on x86
[17:12:52 CEST] <JEEB> wm4: btw do you think that we should be exporting pgs and other ycbcr based subpictures as 4:4:4 with alpha?
[17:13:23 CEST] <wm4> JEEB: maybe we should - but currently we can't
[17:13:51 CEST] <wm4> they should be exported as their native format, paletted yuv
[17:13:58 CEST] <wm4> which is satan
[17:20:22 CEST] <BBB> paletted yuv
[17:20:23 CEST] <BBB> omg
[17:20:38 CEST] <BBB> next comes paletted bayer
[17:21:14 CEST] <Daemon404> i sitll, and always will, maintain exporting paletted "colorspaces" is dumb.
[17:21:58 CEST] <nevcairiel> if anything subtitles should be converted to AVFrame somehow, then you can export subtitles using de-palettized yuv
[17:22:34 CEST] <wm4> de-palettizing is an attractive idea, but could make things slower in some scenarios
[17:22:43 CEST] <wm4> we can just hope it doesn't matter
[17:23:27 CEST] <Daemon404> slower, or "now its only 800x realime insteam of 1000x"?
[17:24:12 CEST] <nevcairiel> the only source for PGS is blu-ray, which is 1080p video or some such, pgs subtitle decoding pales in comparison to even parsing the bitstream of the video =p
[17:35:30 CEST] <BBB> I think for transcoding the idea of having access to the original paletted buffer makes sense
[17:35:41 CEST] <BBB> but thats about the only thing I can think of where its sensible
[17:39:52 CEST] <Daemon404> BBB, we have other various "reasons" like passing it to libavfilter
[17:40:04 CEST] <Daemon404> to benefit in some way for ... things?
[17:40:08 CEST] <wm4> subtitles can't be represented as single AVFrame
[17:40:16 CEST] <wm4> so it's not that simple wrt. lavfi
[17:40:38 CEST] <wm4> (at least pgs subs can have multiple bitmaps)
[17:40:39 CEST] <Daemon404> they could if it had a duration
[17:40:54 CEST] <BBB> multiple bitmaps?
[17:41:01 CEST] <wm4> Daemon404: that's also not enough
[17:41:02 CEST] <nevcairiel> lavfi doesnt do subtitles right now either way =p
[17:41:02 CEST] <BBB> like highlight on/off or so?
[17:41:17 CEST] <wm4> BBB: just multiple bitmaps
[17:41:29 CEST] <nevcairiel> BBB: it can have multiple distinct images on the screen at the same time, with their own palette even if they wanted to etc
[17:41:54 CEST] <Daemon404> i would just render them as one... it's a bluray
[17:41:56 CEST] <Daemon404> res is known.
[17:42:23 CEST] <nevcairiel> pgs has a fiel for presentation size either way
[17:42:26 CEST] <nevcairiel> field*
[17:42:27 CEST] <Daemon404> i would admit its kind of annoying i there are weird timign overlaps
[17:42:40 CEST] <JEEB> yeah, presentation size is there anyways
[17:42:43 CEST] <nevcairiel> and you would potentially need to re-render it when one of them times out
[17:42:44 CEST] <wm4> Daemon404: yeah, I suggested that too once
[17:42:48 CEST] <nevcairiel> which is weird
[17:43:10 CEST] <Daemon404> nevcairiel, this is how dvd subs were coded iirc
[17:43:16 CEST] <Daemon404> as in on the dvd itself.
[17:43:25 CEST] <JEEB> on DVDs only one could be active anyways
[17:43:27 CEST] <JEEB> at a given time
[17:43:28 CEST] <JEEB> IIRC
[17:43:28 CEST] <wm4> don't dvd subs just use rle
[17:43:36 CEST] <Daemon404> wm4, not what i meant
[17:43:38 CEST] <wm4> so they code the whole frame
[17:43:39 CEST] <nevcairiel> dvds only have one active object, so if they want two, they need to code a big image containing both
[17:43:45 CEST] <Daemon404> i meant dvd subs essentially re-render
[17:43:48 CEST] <Daemon404> like nevcairiel was describing
[17:43:50 CEST] <wm4> yeah
[17:43:51 CEST] <Daemon404> due to format limitations
[17:43:58 CEST] <wm4> and highlights are satan
[17:44:08 CEST] <wm4> they can't really be done with libavcodecs decoder
[17:44:10 CEST] <nevcairiel> but pgs is more dynamic, screwing this dynamic and just making everything dumb again seems less then ideal
[17:44:15 CEST] <Daemon404> wm4, i heard you love per-character sub styles
[17:44:38 CEST] <nevcairiel> wm4: you know whats fun, animated dvd subtitles
[17:44:45 CEST] <Daemon404> nevcairiel, it's almost as if subtitles are out of place in a video/audio library.
[17:44:48 CEST] <Daemon404> almost.
[17:44:59 CEST] <wm4> nevcairiel: they just have high bitrate right
[17:45:34 CEST] <Daemon404> i used to have a piece of code that rendered after effects generated overlays (like fansubbers use for signs) as bluray subs
[17:45:37 CEST] <nevcairiel> wm4: no, the subtitle itself is static, it just codes multiple palettes to fade them in and out =p
[17:45:38 CEST] <Daemon404> one per frame
[17:45:38 CEST] <wm4> bitmap sub decoders in lavc are almost reasonable
[17:45:44 CEST] <Daemon404> i wonder if wm4 would murder me if i dug it up
[17:45:45 CEST] <wm4> what gets me are text subtitle converters in lavc
[17:45:54 CEST] <wm4> Daemon404: possible
[17:46:18 CEST] <wm4> the best thing I've ever seen was a mkv file with 300MB or so of ASS extradata
[17:46:18 CEST] <Daemon404> youll be happy to know i stopped fansubbing before we could render bluray subs
[17:46:21 CEST] <Daemon404> so it never did anything.
[17:46:24 CEST] <BBB> overlay rendering into bitmap is silly
[17:46:25 CEST] <wm4> with useless aegisub crap in it
[17:46:30 CEST] <wm4> (easily half of the file size)
[17:46:31 CEST] <BBB> a proper player overlays them as separate textures
[17:46:40 CEST] <Daemon404> BBB, of course.
[17:46:52 CEST] <BBB> and then timing overlap doesnt matter
[17:46:58 CEST] Action: BBB goes pick up kids from school
[17:47:02 CEST] <BBB> will look at h264 issue later
[17:47:03 CEST] <BBB> bye
[17:47:06 CEST] <Daemon404> aloha.
[17:47:52 CEST] <JEEB> anyways, I guess for now we'll just have to do the PAL8 dance
[17:48:16 CEST] <JEEB> also I always thought the PAL there was for NTSC,PAL,SECAM PAL
[17:48:18 CEST] <Daemon404> not me becaus ei dont care about subtitles!
[17:48:44 CEST] <JEEB> :)
[17:51:12 CEST] <JEEB> also I do wonder why there's YUV_TO_RGB1 and 2. first one takes your chroma and the second takes your r,g,b and luma
[17:52:34 CEST] <Daemon404> JEEB, because mutlimedia is neverending pain, to quote wm4.
[17:52:53 CEST] <JEEB> yup
[18:22:22 CEST] <cone-582> ffmpeg 03Anton Khirnov 07master:e3dfef8e3c85: qsvdec_h2645: switch to the new BSF API
[18:22:23 CEST] <cone-582> ffmpeg 03Anton Khirnov 07master:704a39769719: rtmpdh: add an stdio.h include
[18:22:24 CEST] <cone-582> ffmpeg 03Derek Buitenhuis 07master:a980cc47cd01: Merge commit 'e3dfef8e3c85a64dbe6388117303f5819fa3c6a2'
[18:22:25 CEST] <cone-582> ffmpeg 03Derek Buitenhuis 07master:53a9b164c560: Merge commit '704a39769719d2e1ae3f13bfc562b51c9cd002d7'
[18:25:28 CEST] <wm4> ah, skipped
[18:25:35 CEST] <Daemon404> i mailed Ivan
[18:25:59 CEST] <Daemon404> also apparently qsv decoder has crashed for 11 days
[18:26:01 CEST] <Daemon404> and nobody noticed
[18:26:07 CEST] <Daemon404> according to him
[18:26:21 CEST] <Daemon404> it took less time for someone to notice libnut...
[18:51:19 CEST] <jamrial> wm4: did you look at vc1 valgrind memleaks?
[18:51:54 CEST] <jamrial> judging by the commits since the last good run, the new decode api is most likely at fault
[18:53:02 CEST] <wm4> jamrial: ah, so just memory leaks? what fate command line would show this?
[18:53:22 CEST] <wm4> just running with valgrind I see no additional output
[18:54:02 CEST] <jamrial> http://fate.ffmpeg.org/report.cgi?time=20160422161453&slot=x86_64-archlinux…
[18:54:24 CEST] <jamrial> configure with --toolchain=valgrind and run any of those vc1 tests
[18:55:10 CEST] <jamrial> valgrind-memcheck, rather
[18:55:29 CEST] <nevcairiel> thats what I said
[18:55:30 CEST] <nevcairiel> works for me
[18:58:11 CEST] <wm4> it doesn't use valgrind by default for me
[18:58:19 CEST] <wm4> and enabling it explicitly doesn't show any warnings
[18:59:18 CEST] <jamrial> let me see if i can reproduce it
[19:04:00 CEST] <jamrial> yep
[19:05:12 CEST] <jamrial> configure --enable-gpl --toolchain=valgrind-memcheck --samples=/path/to/samples && make fate-vc1_ilaced_twomv
[19:06:40 CEST] <jamrial> using gcc 5.3 on linux x86_64
[19:06:46 CEST] <jamrial> wm4: ^
[19:07:33 CEST] <jamrial> or try "valgrind --error-exitcode=1 --malloc-fill=0x2a --track-origins=yes --leak-check=full --gen-suppressions=all --suppressions=/path/to/ffmpeg-src/tests/fate-valgrind.supp /path/to/ffmpeg-build/ffmpeg -nostdin -nostats -cpuflags all -flags +bitexact -hwaccel none -threads 1 -thread_type frame+slice -i /path/to/samples/vc1/ilaced_twomv.vc1 -flags +bitexact -fflags +bitexact -f framecrc -"
[19:10:24 CEST] <wm4> ah, --toolchain=valgrind-memcheck was PEBKAC, and now fate fails as well
[19:10:31 CEST] <wm4> *not working
[19:19:16 CEST] <wm4> jamrial: does fate with valgrind generally fully succeed?
[19:20:41 CEST] <wm4> api-threadmessage fails
[19:21:09 CEST] <wm4> but I'm fairly sure it prints system-dependent crap anyway, or worse
[19:23:29 CEST] <BBB> Daemon404: I can not reproduce the different output
[19:23:34 CEST] <BBB> Daemon404: or corruption
[19:23:37 CEST] <BBB> Daemon404: it works fine for me
[19:25:19 CEST] <Daemon404> are you sure
[19:25:22 CEST] <Daemon404> even carl reprod it
[19:25:34 CEST] <Daemon404> which file are you testing, mine or carl's?
[19:27:22 CEST] <jamrial> wm4: supposedly, yeah. ubitux's boxes show that every other test passes
[19:29:50 CEST] <jamrial> configure with --disable-stripping (or use ffmpeg_g if you run the command above), otherwise the backtrace will not be useful
[19:30:02 CEST] <wm4> it's a minor memory leak, do you want to review it or can I just push?
[19:30:28 CEST] <jamrial> if it's trivial and it fixes it then just push
[19:31:02 CEST] <cone-423> ffmpeg 03wm4 07master:27adf9f9cdbc: avcodec/utils: fix minor memory leaks on avcodec_open2() failure
[19:32:47 CEST] <BBB> Daemon404: carl's
[19:33:05 CEST] <Daemon404> i cannot vouch for his cut
[19:33:22 CEST] <BBB> $ ./ffmpeg -threads 8 -i ~/Downloads/nondeterministic_cut.h264 -f framemd5 -nostats -v 0 -|md5
[19:33:23 CEST] <BBB> f69630d6c78fa2b4a1a838ea13fc5a76
[19:33:26 CEST] <BBB> it never changes
[19:34:44 CEST] <Daemon404> ill check, but like i said, i cant vouch for carl's cut.
[19:36:09 CEST] <Daemon404> especially since he uploaded an annexb stream, which means he ran a BSF on it
[19:40:18 CEST] <Daemon404> BBB, i cant reproduce it with carl's file
[19:40:21 CEST] <Daemon404> but i CAN with mine
[19:40:26 CEST] <BBB> hm...
[19:40:26 CEST] <Daemon404> i dont know how he made his cut.
[19:40:50 CEST] <BBB> do the frame numbers in his seq correspond to where you see issues?
[19:40:57 CEST] <BBB> (in his crc diff)
[19:40:59 CEST] <BBB> or not even that?
[19:41:06 CEST] <BBB> if not, we should remove his comment as unrelated
[19:41:16 CEST] <BBB> I dont want multiple issues in one trac entry
[19:41:18 CEST] <Daemon404> i have no idea what his cut is from or how it is related
[19:41:40 CEST] <BBB> ok, can repro with yours
[19:41:47 CEST] <BBB> $ for n in `seq 0 10`; do ./ffmpeg -threads 8 -i ~/Downloads/nondeterministic.mp4 -f framemd5 -nostats -v error -|md5; done
[19:41:47 CEST] <BBB> 503761ed866c82ab06a5717757b4cc1d
[19:41:48 CEST] <BBB> 0aa0670d052dd0e4fd0de28222a7e717
[19:42:34 CEST] <Daemon404> i managed to repro with his... once
[19:42:40 CEST] <Daemon404> it might just be really hard with a small sample?
[19:42:46 CEST] <Daemon404> this is a race cond after all
[19:43:06 CEST] <BBB> perhaps
[19:44:31 CEST] <Daemon404> btw insterestingly, using your cmdline above, i get 53a96fb1350e6dbd5a35053c8f5282a6
[19:44:39 CEST] <Daemon404> except once in a while
[19:45:43 CEST] <BBB> $ for n in `seq 0 1000`; do md5=`./ffmpeg -threads 8 -i ~/Downloads/nondeterministic_cut.h264 -f framemd5 -nostats -v error -|md5`; if [ $md5 != f69630d6c78fa2b4a1a838ea13fc5a76 ]; then echo $md5; fi; done
[19:45:43 CEST] <BBB> a5308eca9ff7e6c3adc6a882290ff1a0
[19:45:44 CEST] <BBB> 9d7716a42b36ebf6367229de95ec5238
[19:45:45 CEST] <BBB> that works
[19:46:30 CEST] <BBB> ok lets work with what we have
[19:48:17 CEST] <BBB> did framemd5 output change?
[19:48:21 CEST] <BBB> my checkout may be a little old
[19:48:27 CEST] <BBB> that may cause your md5 being different
[19:48:28 CEST] <Daemon404> yeah i think so
[19:48:31 CEST] <Daemon404> thats probably why
[20:08:32 CEST] <cone-423> ffmpeg 03Jan Sebechlebsky 07master:f9d7e9feec2a: avformat/tee: Fix leaks in tee muxer when open_slave fails
[20:08:33 CEST] <cone-423> ffmpeg 03Jan Sebechlebsky 07master:2063f3ed9c2f: avformat/tee: Handling slave failure in tee muxer
[21:07:48 CEST] <JEEB> https://github.com/jeeb/ffmpeg/commit/8e1b13e440182391ec826c4723fa910881e09…
[21:07:51 CEST] <JEEB> comments pl0x
[21:09:28 CEST] <wm4> why do you determine hdtv vs. sd it lazily?
[21:09:38 CEST] <wm4> and then stick to the result?
[21:09:59 CEST] <JEEB> if you're OK with it changing with every presentation segment I can do that, too
[21:10:33 CEST] <JEEB> it's just that cehoyos's patch had the default -1 value and seemed to try to do that - although it had some weird logic there
[21:10:35 CEST] <wm4> I just see no reason to it lazily... although the current code structure could be used to force it via a private option
[21:10:59 CEST] <wm4> also not too fond of extending this cryptic macroshit
[21:11:22 CEST] <JEEB> well it's basically that or we start dragging zimg or BBB's proper colorimetry thing in
[21:11:36 CEST] <wm4> what proper colorimetry thing?
[21:11:57 CEST] <JEEB> he made the new colorspace filter I think
[21:12:47 CEST] <JEEB> (cehoyos's second patch was http://up-cat.net/p/35963179 )
[21:12:58 CEST] <JEEB> (see how it overrides things to SD :P)
[21:13:42 CEST] <JEEB> but yeah, I do hate the macroshit
[21:13:56 CEST] <JEEB> but in the short term there's just the macroshit unless we start dragging in dependencies
[21:14:05 CEST] <wm4> "avctx->pix_fmt = AV_PIX_FMT_PAL8;" lol didn't spot that yet
[21:14:24 CEST] <wm4> (it's just context, but it's "fun" on its own)
[21:17:37 CEST] <TMA-1> Hi all, I know there is a built-in DCT in ffmpeg, but are there any wavelet transforms that I could make use of?
[21:19:26 CEST] <JEEB> wm4: so basically I just unlazify it
[21:21:42 CEST] <JEEB> wm4: https://github.com/jeeb/ffmpeg/commit/b1f16b679cf331ba9c264c2a564d2e852e690…
[21:21:45 CEST] <JEEB> unlazified
[21:22:18 CEST] <JEEB> argh, the comment
[21:22:24 CEST] <wm4> yeah, better, although there's still no need for the ctx->hdtv field
[21:22:45 CEST] <JEEB> ?
[21:23:00 CEST] <wm4> you could just use avctx->height and compare it directly when converting the colors, tight?
[21:23:02 CEST] <wm4> *right
[21:23:16 CEST] <JEEB> the video track might not be the original
[21:24:05 CEST] <JEEB> as far as I can see the video width and height have to follow the original presentation video's size
[21:24:07 CEST] <wm4> it does this: ret = ff_set_dimensions(avctx, w, h);
[21:24:12 CEST] <wm4> which means avctx->height is set to h
[21:24:18 CEST] Action: JEEB facepalms
[21:25:45 CEST] <JEEB> so I should check for (h < 0 || h > 576) in the conversion spot?
[21:25:53 CEST] <JEEB> the default was -1 , right?
[21:26:16 CEST] <JEEB> (I have no idea if our parser will go into this spot without proper height/width)
[21:26:52 CEST] <JEEB> if the default's zero then not much can be done about that and I'll just check for h > 576
[21:27:18 CEST] <wm4> no idea, I thought the size was always known
[21:27:46 CEST] <JEEB> not sure if a presentation segment has to always come before a palette segment
[21:28:19 CEST] <wm4> ah good point
[21:28:52 CEST] <JEEB> and the private option kind of lets you have a default that you know how it works, as opposed to avctx's which I don't remember the workings of at all
[21:29:20 CEST] <JEEB> although since the valid values of height are known
[21:29:30 CEST] <JEEB> I could check if h == 480 || h == 576
[21:29:37 CEST] <JEEB> and otherwise HDTV
[21:29:55 CEST] <JEEB> which should hopefully include the avctx default
[21:31:42 CEST] <JEEB> so which solution do you think is better?
[21:32:56 CEST] <wm4> the palette is always copied on output
[21:33:20 CEST] <wm4> so the color conversion could be moved to output (in display_end_segment)
[21:33:44 CEST] <wm4> of course that's a bit more complicated
[21:34:16 CEST] <JEEB> yeaaah...
[21:36:15 CEST] <JEEB> but isn't that already an RGB'ized palette btw?
[21:36:23 CEST] <JEEB> in display_end_segment
[21:40:50 CEST] <jamrial_> TMA-1: check jpeg2000dwt.c, maybe
[21:43:26 CEST] <atomnuker> TMA-1: you can use vc2's DWT transforms
[21:43:44 CEST] <atomnuker> they're simpler to use and set up
[21:43:48 CEST] <atomnuker> or snow
[21:45:46 CEST] <JEEB> wm4: I guess I'll go with the extra field for now... while I probably know the default values for avctx->height I'm lazy
[21:52:03 CEST] <wm4> JEEB: this would still output a first bogus subtitle if it goes wrong?
[21:52:55 CEST] <JEEB> if there is no presentation segment yet and you are watching subtitles made for SD, yes
[21:53:14 CEST] <JEEB> because the default is HD
[21:54:57 CEST] <TMA-1> atomnuker: Ah I didn't see them before. Do you know if they have a db2 transform?
[21:55:10 CEST] <JEEB> of course this is all due to the design where we convert to RGB at this point in the process (and the fact that we're converting at all)
[21:55:34 CEST] <JEEB> but I don't care about PGS enough to start reworking the whole "decoder"
[22:02:57 CEST] <atomnuker> TMA-1: db2 transform?
[22:03:37 CEST] <atomnuker> Daubechies?
[22:07:41 CEST] <TMA-1> atomnuker: yes, sorry
[22:07:53 CEST] <TMA-1> Daubechies D4 transform (which I believe is also known as 'db2').
[22:08:36 CEST] <atomnuker> not sure what kind of Daubechies vc2/dirac uses, but either way that transform's missing from the encoder
[22:09:03 CEST] <atomnuker> only LeGall, Deslauries-Dubuc 9/7 and Haar are implemented on the encoder side
[22:09:39 CEST] <atomnuker> I think libschroedinger has Daubechies though
[22:11:18 CEST] <TMA-1> Huh, didn't know about libschroedinger, will check that out. Thanks atomnuker.
[22:37:34 CEST] <fner> Hello. Is this a good place to ask about building libav* libs for windows?
[22:44:12 CEST] <Compn> fner : no, try #ffmpeg
[22:44:18 CEST] <Compn> this channel is for devs
[22:56:38 CEST] <fner> thanks!
[23:18:03 CEST] <wm4> wow
[23:20:50 CEST] <ln-> could be spam
[23:21:13 CEST] <iive> it is definitely a spam
[23:21:47 CEST] <iive> the problem is how tecsolution account have been created and elevated
[23:23:11 CEST] <iive> michaelni:
[23:25:34 CEST] <BtbN> i wonder what the point of this kind of spam is. Noone in their right mind would respond to that.
[23:28:00 CEST] <iive> it's in the wiki on the website
[23:28:09 CEST] <iive> they abuse ffmpeg ranking
[23:29:15 CEST] <iive> these are fishing sites.
[23:30:09 CEST] <BtbN> it's just a phone number?
[23:31:54 CEST] <iive> just disable the account
[23:32:02 CEST] <iive> and delete these pages, if you have admin rights
[23:32:15 CEST] <iive> never heard of microsoft support call scam?
[23:34:01 CEST] <BtbN> I don't have any special permissions on trac
[23:40:04 CEST] <michaelni> accont deleted
[23:40:31 CEST] <BBB> tnx
[23:40:43 CEST] <BBB> isnt there a badwords wiki page?
[23:41:18 CEST] <michaelni> yes
[23:41:24 CEST] <BBB> I forgot the exact name
[23:42:17 CEST] <michaelni> me too :(
[23:43:16 CEST] <michaelni> https://trac.ffmpeg.org/wiki/BadContent
[23:43:35 CEST] <michaelni> also he passed captcha checks
[23:44:22 CEST] <BBB> its probably a person, not a machine
[23:44:37 CEST] <BBB> the time between page creations was very long
[23:45:10 CEST] <BBB> if you give me delete permissions Ill simply delete all the pages he created :-p
[23:46:47 CEST] <BBB> and maybe add (?i)helpline to BadContent (Im not a maintainer either so I cant do that either)
[23:47:17 CEST] <JEEB> yeah, spammers have noticed that it can actually be cheaper to pay people to post spam than to develop a system that keeps up to speed with CAPTCHA systems
[23:48:10 CEST] <BBB> BadIP even better ;)
[23:48:40 CEST] <michaelni> BBB, whats your user name on trac ?
[23:48:46 CEST] <BBB> rbultje
[23:50:18 CEST] <michaelni> you have wiki delete powers now, also ive added you to the developers group you seem not to have been in it or i missed you
[23:52:56 CEST] <BBB> all pages deleted
[23:53:06 CEST] <BBB> I suppose fflogger will catch that?
[23:53:10 CEST] <BBB> maybe not
[23:57:46 CEST] <BBB> JEEB: and colorspace conversion in a decoder yikes
[23:57:47 CEST] <BBB> :D
[23:57:57 CEST] <JEEB> yes
[23:58:37 CEST] <JEEB> it's all eww and I wish your thing or zimg could be used instea :P
[23:58:51 CEST] <JEEB> but that's how those subtitle decoders seem to work
[23:59:09 CEST] <JEEB> you call those macros for all parts in the palette
[23:59:26 CEST] <JEEB> and then that's it and you've got PAL8 :P
[00:00:00 CEST] --- Sat Apr 23 2016
1
0
[00:16:01 CEST] <Guster_> hi there
[00:16:19 CEST] <Guster_> can anyone "new" help me plese with ffmpeg
[00:23:27 CEST] <Draz> Padding on video appears grey: Applying pad filter to video stream gives non-black padding. http://pastebin.com/vuHEVSGZ
[00:24:16 CEST] <Draz> I've tried a few things, including applying format filters and outputting with a different codec, but no success.
[00:24:29 CEST] <Draz> Help appreciate.
[00:24:41 CEST] <Draz> *appreciated.
[00:25:10 CEST] <Draz> Guster_, "When asking a question be precise, detailed...". What's your issue?
[00:25:46 CEST] <Guster_> cant unicast with ffmpeg to ffserver
[00:26:20 CEST] <Guster_> there is log from ffmpeg
[00:26:21 CEST] <Guster_> http://pastebin.com/fXw9qw6V
[00:26:44 CEST] <Guster_> at ffserver just Fri Apr 22 00:12:30 2016 127.0.0.1 - - [POST] "/feed1.avi HTTP/1.1" 200 1282
[04:03:52 CEST] <anadon> People kicking around?
[04:28:26 CEST] <DHE> anadon: standard procedure is to ask your question and wait
[04:36:43 CEST] <anadon> DHE: I know, but some channels stay dead for a very long time. I just ping once in awhile.
[04:38:30 CEST] <DHE> on the other hand it's late in north america, night in europe, and you've wasted valuable time playing IRC tag already.
[04:39:15 CEST] <anadon> DHE: Don't heap on a mistake.
[04:40:13 CEST] <DHE> sorry, just trying to make a point
[05:37:40 CEST] <wyatt8740> Is there a way to make ffmpeg print ONLY info? (meaning I don't want it to print warnings or errors, but only print the information about encoding that you'd get if it were going smoothly)
[05:38:54 CEST] <wyatt8740> I get spammed by distracting 'Past duration [number here] too large' when combining two streams (video and audio devices) into a single capture
[05:39:53 CEST] <anadon> Could you grep through the output
[05:40:07 CEST] <wyatt8740> not while preserving colors, since I'd have to redirect stderr
[05:40:12 CEST] <wyatt8740> info seems to print to stderr too
[05:40:33 CEST] <wyatt8740> and it behaves differently by not using colors if you put it through a pipe
[05:47:34 CEST] <wyatt8740> also, grepping seems to prevent the progress from being displayed
[05:47:40 CEST] <wyatt8740> probably due to buffering or something
[05:48:31 CEST] <anadon> I'd wait for someone more expert than me (I'm not at all)
[05:48:51 CEST] <wyatt8740> fiddling with socat but I never was able to make this work for anything else so my hopes are low
[05:49:20 CEST] <wyatt8740> (once tried to log all communication between my real RS-232 terminal and my PC with socat, it failed miserably)
[05:49:32 CEST] <anadon> That's dedication
[05:50:33 CEST] <wyatt8740> the terminal is really cool and I'm trying to make it work as well as possible :\
[05:50:44 CEST] <wyatt8740> the termcap definitions seem to be missing a few things
[05:51:08 CEST] <wyatt8740> also I was able to discover and report a bug in GNU Nano where syntax highlighting caused the entire screen to redraw with every keystroke
[05:51:31 CEST] <wyatt8740> it happened regardless of terminal but in modern terminal emulators it was so fast no one noticed :|
[05:52:17 CEST] <anadon> wyatt8740: Sounds l=more like you should be bugging #gnu or something.
[05:52:19 CEST] <wyatt8740> http://i.imgur.com/DKBqQ2b.jpg
[05:52:27 CEST] <wyatt8740> no, this is a ffmpeg problem
[05:52:30 CEST] <wyatt8740> I got the nano bug fixed
[05:52:31 CEST] <wyatt8740> :)
[05:52:49 CEST] <wyatt8740> and this ffmpeg problem is not 1985 terminal related
[05:53:05 CEST] <anadon> nethack?
[05:53:16 CEST] <wyatt8740> that's rogue
[05:53:22 CEST] <wyatt8740> I do have nethack here too though
[05:53:33 CEST] <anadon> On a 1985 machine?
[05:53:49 CEST] <wyatt8740> no, lol, that's a terminal
[05:54:09 CEST] <wyatt8740> my i5-3350p desktop PC has rogue and nethack on it :\
[05:54:27 CEST] <wyatt8740> also a serial port
[05:56:05 CEST] <wyatt8740> anyway, if anyone can help me to make ffmpeg not print errors or warnings but still print info messages, please speak up :)
[06:35:35 CEST] <furq> wyatt8740: -v quiet -stats
[06:39:54 CEST] <furq> also i'm disappointed that someone with a 3161 on his desk isn't using irssi
[06:54:20 CEST] <wyatt8740> furq: nice, you recognized the model
[06:54:23 CEST] <wyatt8740> did you use one?
[06:59:08 CEST] <wyatt8740> also when I find a pic I'll show you - I do use irssi on it
[06:59:14 CEST] <wyatt8740> just not in that pic :\
[07:06:11 CEST] <wyatt8740> durandal_1707: I forget, isn't your name taken from Roland's sword? (Roland the knight of charlemagne)
[07:06:36 CEST] <furq> yeah i definitely didn't type "ibm terminal" into google images and then find another picture of your desk
[07:06:48 CEST] <furq> that's not how i know the model number
[07:09:29 CEST] <wyatt8740> furq: oh, okay. But there are about a million and a half IBM terminals
[07:09:41 CEST] <wyatt8740> and I would've been too lazy :\
[07:09:56 CEST] <furq> well your desk is on the second row so it worked out great
[07:10:06 CEST] <wyatt8740> lol
[07:10:09 CEST] <wyatt8740> is it now?
[07:10:31 CEST] <wyatt8740> so it is
[07:10:36 CEST] <wyatt8740> with the amber screen I need to fix
[07:13:48 CEST] <wyatt8740> most of their offerings
[07:13:54 CEST] <wyatt8740> *don't even work on normal PCs
[07:13:58 CEST] <wyatt8740> you need to have a IBM mainframe
[08:17:55 CEST] <wyatt8740> I really think ffmpeg is one of the most awesome free/OSS projects out there. I'd say it along with the linux kernel and maybe QEMU (...and ffmpeg and qemu are both things that were started by Fabrice Bellard. Wow).
[09:17:16 CEST] <thebombzen> does anyone know if the drawtext filter supports WOFF2?
[09:28:35 CEST] <xbloodhound> I have been trying to add two separate animations as png watermarks using something like this -filter_complex "[0:v][1:v][2:v] overlay=0:0" but I am getting error that says too many inputs specified for overlay filter etc. so my question is it possible to add two overlays at all or am I doing something wrong?
[09:29:53 CEST] <xbloodhound> I only know -filter_complex "[0:v][1:v] overlay=0:0" works at least
[09:32:00 CEST] <furq> xbloodhound: [0:v][1:v]overlay=0:0[tmp];[tmp][2:v]overlay=0:0[out]
[09:43:39 CEST] <xbloodhound> ugh that worked thx
[09:43:58 CEST] <xbloodhound> been messing around for a while
[09:59:25 CEST] <termos> is there a way to reduce the blocking time of avio_open? it's taking 500ms each call to open rtmp:// sockets, it's too long when opening 20+ of them
[11:04:54 CEST] <Eiken_> when i apply a cube lut generated in nuke the output looks exactly the same
[11:05:21 CEST] <Eiken_> -vf lut3d=/mnt/home/erjo/testlut.cube
[11:05:24 CEST] <Eiken_> is this incorrect?
[11:50:54 CEST] <durandal_1707> wyatt8740: nope
[11:58:43 CEST] <aspiers> hi all, any ideas why ffmpeg 3.0.1 would record audio perfectly via pulseaudio, but when I add -f x11grab for screencasting, the audio is truncated at the last ~3s of every capture ?
[11:59:02 CEST] <aspiers> and BTW a/v sync is perfect in this scenario
[11:59:52 CEST] <aspiers> I used: ffmpeg -thread_queue_size 512 -f pulse -i default -ac 1 -f x11grab -video_size 1280x720 -framerate 30 -i :0.0+0+0 -y test.mp4
[13:41:03 CEST] <GusterChe> hi there
[13:41:23 CEST] <GusterChe> me again like last night :D
[13:41:41 CEST] <GusterChe> same problem with my ffmpeg and ffserver :D
[13:42:37 CEST] <GusterChe> anyone with experience to pvt? :D
[13:54:32 CEST] <GusterChe> there are all outputs from ffserver and ffmpeg with ffserver.conf http://pastebin.com/1KxwWQDZ if soneone know what i did wrong please help
[14:01:05 CEST] <GusterChe> is anyone online? xD
[14:13:17 CEST] <GusterChe> hi oldcode
[14:14:36 CEST] <GusterChe> hi fracting :D
[14:15:07 CEST] <BtbN> Don't use ffserver if you can't figure stuff out on your own.
[14:27:08 CEST] <GusterChe> there are all outputs from ffserver and ffmpeg with ffserver.conf http://pastebin.com/1KxwWQDZ if soneone know what i did wrong please help
[15:26:00 CEST] Last message repeated 1 time(s).
[15:29:35 CEST] <Joeboy> Hi. I'm trying to convert some stereo files to mono, but the resulting files seem to get converted from s24le to s16le. Is there a way I can make ffmpeg not do that?
[15:29:38 CEST] <Joeboy> I'm doing:
[15:30:05 CEST] <Joeboy> ffmpeg -y -i input.wav -map_channel 0.0.0 junk.wav -map_channel 0.0.1 output.wav
[15:36:17 CEST] <furq> -c:a pcm_s24le
[15:36:36 CEST] <Joeboy> Actually -acodec copy seems to work
[15:36:37 CEST] <furq> or maybe -c:a copy will work but i don't think it'll make any difference
[15:36:50 CEST] <furq> yeah that'll do it
[15:36:56 CEST] <Joeboy> Sorry, should have experimented more before asking
[15:38:03 CEST] <furq> if you don't specify then ffmpeg will pick whatever output codec it thinks is best for the output container
[15:38:14 CEST] <furq> which for wav is pcm_s16le
[15:38:39 CEST] <GusterChe> furq: can u help me with my problem?
[15:39:08 CEST] <furq> i can tell you to not use ffserver because it's basically abandoned and it sucks
[15:39:10 CEST] <furq> does that help
[15:39:38 CEST] <GusterChe> yes xD
[15:39:52 CEST] <GusterChe> and, i have one question more if u have some time :D
[15:40:38 CEST] <furq> well that depends on how long it would take to answer the question
[15:40:41 CEST] <GusterChe> if i want to use nginx and rtmp module. do u can give me guildlines? :)
[15:40:55 CEST] <furq> https://gist.github.com/fur-q/d7028f51c38f7d0bb56e
[15:40:56 CEST] <furq> how's that
[15:41:28 CEST] <furq> you can ignore the lua stuff if you don't care about auth
[15:43:37 CEST] <GusterChe> and how to set unicast to server with ffmpeg?
[15:44:09 CEST] <furq> it's rtmp, it only does unicast
[15:56:37 CEST] <GusterChe> furq: thanks, this works fine for now
[15:56:53 CEST] <GusterChe> i go to test it for another things
[15:57:05 CEST] <GusterChe> thanks a lot and cya!
[17:56:39 CEST] <Joeboy> Ok, I asked a question earlier and thought I'd got it sorted but it turns out I hadn't. So I'm going to ask it again.
[17:57:07 CEST] <Joeboy> I'm trying to convert an s24le stereo file into two s24le mono files.
[17:57:58 CEST] <Joeboy> (actually lots of files)
[17:58:17 CEST] <Joeboy> I got as far as: ffmpeg -y -i $f -map_channel 0.0.0 junk.wav -acodec copy -map_channel 0.0.1 "gve-london-audio/oktava/$f"
[17:58:48 CEST] <Joeboy> But that gives me stereo files, for some reason.
[17:59:06 CEST] <Joeboy> without the -acodec copy, I get s16le mono files.
[17:59:51 CEST] <Joeboy> I just want to do a raw copy of one track from the stereo file into a mono file.
[18:01:02 CEST] <Joeboy> ffmpeg (at least my version) doesn't seem to have any 24 bit output options
[18:09:24 CEST] <durandal_1707> Joeboy: -codec:a pcm_s24le
[18:13:08 CEST] <Joeboy> durandal_1707: Thanks!
[20:27:12 CEST] <johnnny22-afk> i have a mpegts that starts at some pts, is there a way to re-encode that pts/dts, so that it starts at 0 ?
[20:28:23 CEST] <c_14> ffmpeg should do that by default
[20:49:25 CEST] <johnnny22-afk> m'ok, but I have a 39h mpegts, which has a rollover around 13h or so from the beginning. ffprobe seems to tell me that the duration is about 13h long. After re-encoding it, ffprobe still tells me that the duration is about 13h. I was hoping to make the duration be ~26h or so, being that max duration of a mpegts without rollovers.. Should I be adjusting something else ?
[20:49:55 CEST] <johnnny22-afk> maybe it's the DTS value that should be changed ?
[20:52:49 CEST] <c_14> ffprobe doesn't actually go through the entire video to find the duration, it usually either uses a value stored in the container
[20:56:17 CEST] <aspiers> hi all, any ideas why ffmpeg 3.0.1 would record audio perfectly via pulseaudio, but when I add -f x11grab for screencasting, the audio is truncated at the last ~3s of every capture ?
[20:56:22 CEST] <aspiers> I used: ffmpeg -thread_queue_size 512 -f pulse -i default -ac 1 -f x11grab -video_size 1280x720 -framerate 30 -i :0.0+0+0 -y test.mp4
[20:57:09 CEST] <c_14> aspiers: is this a regression?
[20:57:39 CEST] <aspiers> c_14: since the last time I tried it, yes - but that was at least 6-12 months ago
[20:58:05 CEST] <aspiers> c_14: A/V sync is still flawless, however
[20:59:37 CEST] <c_14> Can you test with a recent git build? If it still doesn't work with that, open a ticket on trac and mention that it's a regression (preferably mention which release it last worked with)
[21:00:09 CEST] <aspiers> c_14: I already downloaded a static build from git, but it was missing support for pulseaudio :-/
[21:00:16 CEST] <aspiers> and I don't have time to compile one unfortunately
[21:01:16 CEST] <c_14> You can make a bugreport and mention that you only tested with 3.0.1, but the first thing you're going to be asked is if you can still reproduce with a build from git
[21:01:44 CEST] <aspiers> hah
[21:02:26 CEST] <aspiers> yeah, that's fair - although it would only take someone who already has a git build ~20s to copy/paste my command and see if it's the same for them ...
[21:02:40 CEST] Action: c_14 doesn't have pulse
[21:02:42 CEST] <c_14> therefore can't test
[21:02:45 CEST] <aspiers> ok
[21:04:30 CEST] <johnnny22-afk> c_14: aah ok ,right. In my case it's a .ts file, so I'm not sure where it takes that duration value.
[21:04:53 CEST] <johnnny22-afk> c_14: is there a way to make it check the whole file to compute the duration of a particular .ts file ?
[21:06:27 CEST] <johnnny22-afk> maybe -formatprobesize to the whole size of the file ?
[21:07:19 CEST] <c_14> There is -count_packets and -count_frames, but I don't think either is used to get a duration, you can use ffmpeg -i input -f null /dev/null and the last line of output will contain the total duration
[21:08:42 CEST] <johnnny22-afk> yeah, i guess that would be one way :)
[21:09:08 CEST] <johnnny22-afk> adding -c copy for speed : P
[21:11:17 CEST] <CoffeeFluxx> Heyo! I ran a command to mux some h264 video and aac audio into mp4, but somehow it's creating an empty srt track http://pastebin.com/raw/q8T2Ja4K
[21:11:56 CEST] <CoffeeFluxx> Any idea what I'm doing wrong? Neither of the files I'm muxing in has subs, so I'm a bit confused where that came from
[21:12:14 CEST] <c_14> CoffeeFluxx: the m4a presumably had a subtitle trac
[21:12:25 CEST] <c_14> can you pastebin the complete console output of the first command?
[21:12:32 CEST] <c_14> also, it's not srt. It's mov_text
[21:12:32 CEST] <CoffeeFluxx> sure
[21:13:40 CEST] <CoffeeFluxx> http://pastebin.com/raw/ccnWubmm
[21:15:02 CEST] <c_14> hmm, weird
[21:15:14 CEST] <c_14> you should be able to get rid of it with -sn though
[21:16:38 CEST] <CoffeeFluxx> yeah, i checked both the input files and it wasn't there
[21:19:53 CEST] <CoffeeFluxx> it's still there, even after adding -sn
[21:20:08 CEST] <CoffeeFluxx> lemme pastebin
[21:20:24 CEST] <c_14> add -map_metadata -0
[21:22:01 CEST] <CoffeeFluxx> http://pastebin.com/raw/ytfy4Nth
[21:25:16 CEST] <CoffeeFluxx> added the -map_metadata -0, but it's still there
[21:26:41 CEST] <c_14> eeh, honestly no clue. Try updating to a newer build? Does the mov_text stream always show up or only with those two files?
[21:28:19 CEST] <CoffeeFluxx> no clue, normally i'm just muxing to mkv so i use a different program
[21:31:59 CEST] <c_14> Does a subtitle track show up when muxing those 2 files to mkv instead of mp4?
[21:35:45 CEST] <CoffeeFluxx> conversion fails
[21:35:45 CEST] <CoffeeFluxx> http://pastebin.com/raw/fSYti06j
[21:36:16 CEST] <c_14> well, that's fun
[21:36:20 CEST] <c_14> try mpegts?
[21:37:54 CEST] <CoffeeFluxx> same error
[21:40:13 CEST] <c_14> try muxing to mkv with the older build (if you still have that one)
[21:42:21 CEST] <CoffeeFluxx> nope :\
[21:42:23 CEST] <CoffeeFluxx> lemme grab one
[21:44:17 CEST] <CoffeeFluxx> hmm i can't find older windows builds
[21:45:17 CEST] <kepstin> the issue with your most recent command is that the m4a file has a piece of cover art that's being encoded as video
[21:45:31 CEST] <kepstin> you'll want to use some -map parameters to select specific streams
[21:45:42 CEST] <kepstin> e.g. -map 0:a -map 1:v for your most recent example
[21:45:51 CEST] <kepstin> that'll also solve the subtitle issue
[21:46:55 CEST] <CoffeeFluxx> i thik i tried that earlier and it didn't get rid of the subs
[21:47:03 CEST] <CoffeeFluxx> the video is definitely correctly in the output
[21:47:07 CEST] <CoffeeFluxx> i'll try it again, though
[21:50:14 CEST] <CoffeeFluxx> yeah the subs are still there
[21:54:09 CEST] <c_14> It's probably a bug. I'd open an issue on trac, though if there's no good way to reproduce the issue... If you could upload the files (or parts thereof) to reproduce that would probably help
[21:54:24 CEST] <CoffeeFluxx> huh, re-encoded the audio with -map 0:a and it appeared there too
[21:57:18 CEST] <CoffeeFluxx> i'll file a report, though unfortunately i can't upload the files :\
[21:57:48 CEST] <CoffeeFluxx> i'll see if i can repro by encoding some other stuff the same way, though
[22:19:57 CEST] <spirou> if I want to let ffmpeg convert 5.1 sound in a video to normal stereo, would it be very complicated parameters?
[22:20:40 CEST] <JEEB> if input audio channels are mapped in a normal way and in a single track - it should be as simple as -ac 2 after input
[22:21:11 CEST] <JEEB> it only gets complicated if you have to start doing channel mapping or multiple tracks into channels of a single track
[22:21:13 CEST] <spirou> since I have to let it reencode the sound anyway to make the TV accept it, I guess I could tell it to so stereo
[22:22:57 CEST] <spirou> (I have a mp4, that the tv didn't like, and when I had ffmpeg put it in a mkv instead (with -c copy) there was no sound, so I assume the tv don't like that kind of sound, or that kind of sound in a mkv)
[22:23:02 CEST] <spirou> "Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 447 kb/s (default)"
[22:24:38 CEST] <spirou> JEEB: ok so like this then? ffmpeg -i in.mp4 -ac 2 -vcodec copy out.mkv
[22:24:54 CEST] <JEEB> add -c:a aac -b:a 192k or so
[22:25:20 CEST] <JEEB> if it complains about experimentality then update FFmpeg or add -strict experimental which enables "experimental" features
[22:26:08 CEST] <JEEB> and you might want to use -c:v copy for video in the short way as well in later commands :)
[22:27:15 CEST] <spirou> aha, the "-c:v copy" would be the new way of say "-vcodec copy"
[22:27:22 CEST] <JEEB> yup
[22:30:09 CEST] <spirou> yeah I had to write -strict experimental after the input file. I assume it is same thing as -strict -2 as it suggested
[22:30:31 CEST] <spirou> btw, what do "[matroska @ 0xa79c120] Codec for stream 0 does not use global headers but container format requires global headers" means?
[22:31:41 CEST] <JEEB> spirou: yeah - except the text-based value is generally more understandable than "-2" which is the internal value :)
[22:31:57 CEST] <JEEB> also you should generally update your FFmpeg
[22:32:08 CEST] <JEEB> the aac encoder was vastly improved recently
[22:32:14 CEST] <JEEB> and the experimental flag was removed
[22:32:36 CEST] <JEEB> among other improvements
[22:35:12 CEST] <spirou> ah yeah I'm still on "7:2.8.3+git1~trusty1", I haven't updated for some time... the newest would be "7:3.0.0+git1~trusty1" it says here
[22:37:01 CEST] <spirou> I'll should remember to let it do its updates before doing anything I want to store permanently
[22:39:33 CEST] <epifanio> Hi
[22:39:50 CEST] <spirou> Ideally I should try to make some ffprobe-using script that could tell if the TV would be OK with a file...
[22:39:55 CEST] <spirou> hello epifanio
[22:41:33 CEST] <epifanio> Im dealing with 2 windows applications (for which I do not have access to the src code), the first one takes as input a directory with png and generates a .avi as output. The second application takes the .avi (previously generated) as input, and try to make a mosaic from each frame
[22:41:56 CEST] <epifanio> i want replace the first application with some open source code, using ffmpeg
[22:42:32 CEST] <CoffeeFluxx> submitted a bug report - any suggestions in the meantime on how to work around the issue?
[22:42:53 CEST] <epifanio> i do know how to generate a video using a sequence of pngs with ffmpg, but my output is not compatible wiht the input video needed by the second application
[22:43:42 CEST] <epifanio> so, I tried to run ffmpeg -i goodoutput.avi on the file generated by the first application, this is the output https://gist.github.com/e7ef38889217aee0fa1e1aefceeaf2a8
[22:44:39 CEST] <epifanio> can you suggest me the correct syntax/codec i need to use with ffmpeg to generate a video of the same type as the one described by the ffmpeg -i output?
[22:44:45 CEST] <kepstin> epifanio: should be as simple as setting the codec to match, "-c:v rawvideo -pix_fmt bgr24"
[23:13:34 CEST] <epifanio> kepstin: Thank you! Im trying with: !cat *.png | ffmpeg -framerate 5 -f image2pipe -c:v png -i - -c:v rawvideo -pix_fmt bgr24 test3.avi
[23:14:26 CEST] <epifanio> ive a video output, but the framerate is not accepted this the log: https://gist.github.com/31592fd73aa4569a73dd73a30422e31a
[23:15:34 CEST] <epifanio> debug
[23:16:19 CEST] <epifanio> the app which triys to open the video goes in segfault
[23:17:58 CEST] <epifanio> this the log of ffmpg -i on the output video i used: https://gist.github.com/74270e975ae79348c44c63db6467cd3c
[23:17:58 CEST] <epifanio>
[23:18:15 CEST] <epifanio> the fps are set correctly
[23:20:51 CEST] <epifanio> ill try again ffmpeg -i on both files and post here to double check their format match
[23:24:03 CEST] <epifanio> the output is https://gist.github.com/epifanio/f9ec086b2f946a56583cf46f1de47fcf left.avi is the video which works as input for the app Im using, test3.avi is the one i generated with ffmpeg
[23:26:24 CEST] <kepstin> epifanio: I don't see any notable differences.
[23:26:46 CEST] <kepstin> if the application that tries to open it segfaults, well, that's a pretty bad application bug...
[23:27:01 CEST] <kepstin> that said, you might want to try making the height/width a multiple of 8 or 16.
[23:27:12 CEST] <epifanio> it is a vey bad app which they are forcing me to use .. :(
[23:27:26 CEST] <kepstin> at this point, you're just debugging that application :/
[23:29:44 CEST] <epifanio> they just dont care, they give me an app which work (please let me use this term & I know) that takes a directory with some png and make a video out of it & my problem is that i have 70.000 images and i need to group them in subfolders of 20 images each .. then make videos for each subfolder
[23:30:31 CEST] <epifanio> so i wanted to script my way to generate those videos, on my unix machine .. rhater than klick who knows how many times on a windows application gui
[23:31:22 CEST] <epifanio> i saw a little difference in the metadata: encoder : Lavf57.25.100
[23:31:41 CEST] <epifanio> and the size: 330x330 vs 1360x1024
[23:33:47 CEST] <kepstin> yeah, I suspect the size is the main issue
[23:36:24 CEST] <epifanio> how can i force the output to match this size? which is actually the original size of the png, a sample is: http://epinux.com/hbc_img/selection/left/201503.20150619.181140656.204627.p…
[00:00:00 CEST] --- Sat Apr 23 2016
1
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[00:54:52 CEST] <cone-241> ffmpeg 03Michael Niedermayer 07master:9f36ea57ae6e: avcodec/avpacket: Fix off by 5 error
[01:16:17 CEST] <Daemon404> where exactly does this 5 come from...
[01:20:44 CEST] <rcombs> 4 bytes size, 1 byte type
[02:31:46 CEST] <BBB> durandal_170: is tm2rt in any way related to tm2?
[02:33:41 CEST] <durandal_170> Dunno, not much
[03:32:11 CEST] <KGB> [13FFV1] 15ablwr opened pull request #17: CSS stylesheet now locally available (06master...06master) 02https://git.io/vwCkP
[08:43:40 CEST] <pfelt> evening all. in ffmpeg_opt.c what is the intended difference between init_complex_filters() and configure_complex_filters() ? my filter init() function is getting called twice with two different contexts from those two functions and its messing my code up
[09:16:46 CEST] <pfelt> grumble. i just found this in code: /* this graph is only used for determining the kinds of inputs * and outputs we have, and is discarded on exit from this function */
[09:17:02 CEST] <pfelt> that isn't good. it screws up the fifo i'm trying to use
[13:22:07 CEST] <durandal_1707> michaelni: could you backport few commits?
[13:26:25 CEST] <michaelni> durandal_1707, sure, which ?
[13:29:12 CEST] <durandal_1707> michaelni: latest takdec and apedec
[13:31:33 CEST] <KGB> [13FFV1] 15michaelni closed pull request #17: CSS stylesheet now locally available (06master...06master) 02https://git.io/vwCkP
[14:02:11 CEST] <cone-648> ffmpeg 03Michael Niedermayer 07master:0de99ab06f70: avcodec: Add avpriv_codec_get_cap_skip_frame_fill_param()
[14:02:11 CEST] <cone-648> ffmpeg 03Michael Niedermayer 07master:a0b92788a885: avformat/utils: Remove use of caps_internal as it is not public API
[14:10:22 CEST] <michaelni> durandal_1707, 9149e9c0baaec122bc3da925d6068dffa60b5427 and 38797a8033d061ade58b30b8ac86da222fe42a84 ?
[14:10:45 CEST] <Daemon404> durandal_1707, why dont you learn to use cherry pick
[14:10:48 CEST] <Daemon404> instead of just complaining
[14:10:50 CEST] <Daemon404> and being lazy
[14:11:00 CEST] <Daemon404> it's getting kind of annoying.
[14:12:34 CEST] <durandal_1707> michaelni: yes
[14:14:50 CEST] <durandal_1707> Daemon404: I don't like to mess with biblical branches in fear I will fsck smthing up
[14:15:04 CEST] <durandal_1707> *nonlocal
[14:15:20 CEST] <Daemon404> that's a bs excuse. it's a cherry pick of a localized file, which you wrote.
[14:15:51 CEST] <wm4> michaelni, Daemon404: ok if I push the decode API patches?
[14:16:05 CEST] <Daemon404> wm4, fine by me.
[14:17:00 CEST] <michaelni> wm4, should be ok
[14:18:13 CEST] <wm4> should I squash patch 2/3 into 1/3?
[14:18:34 CEST] <michaelni> no oppinion ...
[14:18:57 CEST] <nevcairiel> 2 is just a fix on top of 1, isnt it
[14:19:11 CEST] <wm4> in a way
[14:20:16 CEST] <wm4> I'll squash it
[14:21:29 CEST] <wm4> hm I guess I'll add a version bump too
[14:21:33 CEST] <wm4> it was skipped, apparently
[14:29:52 CEST] <cone-648> ffmpeg 03Aaron Boxer 07release/3.0:5d79566ab3dd: avcodec/j2kenc: Add attribution to OpenJPEG project:
[14:29:53 CEST] <cone-648> ffmpeg 03Marios Titas 07release/3.0:47f0d80ee01b: avfilter/src_movie: fix how we check for overflows with seek_point
[14:29:54 CEST] <cone-648> ffmpeg 03Michael Niedermayer 07release/3.0:6103a8453fad: Changelog: Make formating consistent
[14:29:55 CEST] <cone-648> ffmpeg 03Michael Niedermayer 07release/3.0:325d0b64d5a4: avcodec/pngdec: Fix alpha detection with skip_frame
[14:29:56 CEST] <cone-648> ffmpeg 03Michael Niedermayer 07release/3.0:14fdebc4ffcb: avcodec/bmp_parser: Ensure remaining_size is not too small in startcode packet crossing corner case
[14:29:57 CEST] <cone-648> ffmpeg 03Ivan 07release/3.0:76573c5239fb: avcodec/h264: Fix for H.264 configuration parsing
[14:29:58 CEST] <cone-648> ffmpeg 03Michael Niedermayer 07release/3.0:264c9fe6aada: avcodec/avpacket: Fix off by 5 error
[14:29:59 CEST] <cone-648> ffmpeg 03Paul B Mahol 07release/3.0:6fdd122b11e5: avcodec/apedec: fix decoding of stereo files with one channel full of silence
[14:30:00 CEST] <cone-648> ffmpeg 03Paul B Mahol 07release/3.0:7626fb6cbf5f: avcodec/takdec: add code that got somehow lost in process of REing
[14:30:50 CEST] <Daemon404> oh shit i broke libnut
[14:31:27 CEST] <Daemon404> er
[14:31:33 CEST] <Daemon404> hey michaelni where do i *get* libnut?
[14:31:42 CEST] <Daemon404> google provides a bunch of weird github forks
[14:33:54 CEST] <cone-648> ffmpeg 03wm4 07master:7fc329e2dd62: lavc: introduce a new decoding/encoding API with decoupled input/output
[14:33:55 CEST] <cone-648> ffmpeg 03wm4 07master:656b07b5a969: lavf: use new decode API
[14:34:16 CEST] <wm4> unfortunately I have no idea how to fix ffmpeg to use the new API
[14:35:02 CEST] <wm4> and I suppose removing the DTS "trick" on draining would not be accepted
[14:35:16 CEST] <Daemon404> hack, you mean hack
[14:36:02 CEST] <Daemon404> the fact that i cannot even find the repo, really does say something about libnut
[14:36:41 CEST] <Daemon404> git://git.ffmpeg.org/nut seems to be it
[14:36:48 CEST] <Daemon404> but it is literally only reference from multimediawiki
[14:36:52 CEST] <Daemon404> which is hardly canonical
[14:37:21 CEST] <Daemon404> it isnt listed anywhere.
[14:37:23 CEST] <Daemon404> ^ michaelni
[14:37:33 CEST] <wm4> can't we just drop libnut support
[14:37:44 CEST] <Daemon404> oh... its a git mirror
[14:37:45 CEST] <wm4> is there even a single half-way non-insane argument for keeping it
[14:37:47 CEST] <Daemon404> of svn://svn.mplayerhq.hu/nut
[14:37:50 CEST] <Daemon404> even better
[14:38:05 CEST] <Daemon404> wm4, the only argument i can think of is that it helps to have two implementations of nut to keep it "honest"
[14:38:14 CEST] <wm4> lol.
[14:39:11 CEST] <Daemon404> holy crap that svn co is slow
[14:40:10 CEST] <nevcairiel> why didnt you use the git mirror
[14:40:49 CEST] <Daemon404> i wanted to see if upstream even still existed
[14:42:26 CEST] <Daemon404> real 0m25.744s
[14:42:32 CEST] <Daemon404> for ~10 files
[14:42:57 CEST] <Daemon404> ... huh... there is no configure... just config.mak
[14:43:00 CEST] <Daemon404> which you edit by hand
[14:43:01 CEST] <Daemon404> ghetto.
[14:44:06 CEST] <wm4> the server must be located in someone's basement, running on dialup
[14:44:38 CEST] <Daemon404> yes that sounds like mplayer's svn server
[14:45:11 CEST] <Daemon404> wm4, btw it took almost 2 weeks for someone to notice libnut.c didnt even compile.
[14:45:23 CEST] <wm4> as expected
[14:47:14 CEST] <Daemon404> ff_parse_specific_params
[14:47:16 CEST] <Daemon404> nice fucn name
[14:55:24 CEST] <Daemon404> shit wrong email in my git config
[15:59:43 CEST] <Daemon404> oh hey
[16:00:16 CEST] <Daemon404> BBB / michaelni - i got persmission to share a sample which triggers a race in the h264 decoder (causing broken output)
[16:04:30 CEST] <BBB> awesome!
[16:04:42 CEST] <BBB> can you file a trac ticket and poke me with it friday?
[16:04:51 CEST] <BBB> if michaelni fixes it earlier, thats ok also
[16:05:05 CEST] <Daemon404> BBB, i will file a ticket
[16:05:17 CEST] <Daemon404> however i cannot guarantee carl wont close it as FUCKYOU
[16:07:15 CEST] <BBB> hahahahhahahaha
[16:07:17 CEST] <BBB> well, ok
[16:07:27 CEST] <BBB> if the file is somewhere, Ill look anyway
[16:07:32 CEST] <BBB> I dont mind carl so much
[16:07:40 CEST] <BBB> as long as he doesnt delete the file
[16:08:49 CEST] <Daemon404> the file will be hosted elsewhere
[16:08:57 CEST] <Daemon404> its probsbly too large for trac upload
[16:09:05 CEST] <Daemon404> (266 mb)
[16:15:08 CEST] <Daemon404> BBB, ^
[16:16:30 CEST] <fritsch> Daemon404: i wonder how often you end up in /dev while navigating to your ~/dev directory
[16:16:51 CEST] <Daemon404> not often
[16:16:57 CEST] <fritsch> hehe, but it happens
[16:16:57 CEST] <Daemon404> since i type cd ~/dev/<...>
[16:17:02 CEST] <Daemon404> itll fail with /dev
[16:17:02 CEST] <Daemon404> ;)
[16:18:34 CEST] <JEEB> I usually have ~/projects/
[16:19:04 CEST] <JEEB> (and I just learned that irssi can tab-autocomplete random paths)
[16:19:26 CEST] <Daemon404> ... so it can
[16:19:27 CEST] <Daemon404> wtf
[16:19:46 CEST] <JEEB> I knew it did it with stuff like /script or /config
[16:19:47 CEST] <Daemon404> 10 years of use and i didnt know this.
[16:19:52 CEST] <JEEB> inorite
[16:20:58 CEST] <durandal_1707> Why people use libnut when there is superior implementation already in lavf
[16:22:08 CEST] <cone-648> ffmpeg 03Derek Buitenhuis 07master:be5fde92ff6c: libnut: Pass stream to ff_parse_specific_params
[16:22:09 CEST] <BtbN> because they are nuts
[16:23:27 CEST] Action: JEEB slowly claps
[16:24:11 CEST] <wm4> no, because they think the lib is needed
[16:24:19 CEST] <wm4> because if not, why would we support it?
[16:24:40 CEST] <wm4> (the truth is that we support it because we never delete anything, not even totally useless things)
[16:24:55 CEST] <Daemon404> also our wiki for building includes roughly 100000 libraries
[16:25:01 CEST] <Daemon404> so everyone follows those instructions
[16:27:10 CEST] <atomnuker> wm4: we delete stuff, 2 aac encoders, libstagefright, something else I can't remember...
[16:28:10 CEST] <Daemon404> but not libutvideo
[16:28:40 CEST] <wm4> is that other aac encoder really deleted
[16:32:16 CEST] <atomnuker> libaacplus was
[16:37:16 CEST] <rcombs> and libvo_aac
[16:43:46 CEST] <wm4> but not libfaac or libfdk
[16:44:03 CEST] <Daemon404> libfdk should not be removed
[16:44:27 CEST] <atomnuker> the latter has it's place, the former will be gone as soon as I find time to rewrite the fast coder
[16:55:28 CEST] <cone-648> ffmpeg 03Martin Storsjö 07master:fab8156b2f30: avio: Copy URLContext generic options into child URLContexts
[16:55:29 CEST] <cone-648> ffmpeg 03Derek Buitenhuis 07master:f8e89d8a297e: Merge commit 'fab8156b2f30666adabe227b3d7712fd193873b1'
[17:02:08 CEST] <cone-648> ffmpeg 03Martin Storsjö 07master:136c3438bbdb: tcp: Use rw_timeout for setting the connect/listen timeouts
[17:02:09 CEST] <cone-648> ffmpeg 03Martin Storsjö 07master:3ee2ec5ec1e3: unix: Use rw_timeout for setting the connect timeout
[17:02:10 CEST] <cone-648> ffmpeg 03Derek Buitenhuis 07master:ba2d37e9ada5: Merge commit '136c3438bbdb56a5d2f1f0f486f180641dc6dda0'
[17:02:11 CEST] <cone-648> ffmpeg 03Derek Buitenhuis 07master:7e3d3a6a895f: Merge commit '3ee2ec5ec1e39a438f89302d949c93a1b5d365a2'
[17:04:04 CEST] <cone-648> ffmpeg 03Diego Biurrun 07master:b298b36fc008: fate: Only run SRTP test if SRTP code is enabled
[17:04:05 CEST] <cone-648> ffmpeg 03Diego Biurrun 07master:3dfbf32b95ae: build: Drop redundant removal of compiled object files
[17:04:06 CEST] <cone-648> ffmpeg 03Derek Buitenhuis 07master:420806f0d14a: Merge commit 'b298b36fc008ad94a24929fe770c8189d96bcac4'
[17:04:07 CEST] <cone-648> ffmpeg 03Derek Buitenhuis 07master:53107052b2a1: Merge commit '3dfbf32b95ae69a9b15cd4912bc1c68fa16b4093'
[17:04:41 CEST] <cone-648> ffmpeg 03Diego Biurrun 07master:439929859ae0: testprogs: Clean up #includes
[17:04:42 CEST] <cone-648> ffmpeg 03Derek Buitenhuis 07master:32c044cbc640: Merge commit '439929859ae0eb9542d3bb8a0c856bd5a1d1ec48'
[17:50:20 CEST] <cone-648> ffmpeg 03Diego Biurrun 07master:3b08d9d932ee: testprogs: K&R formatting cosmetics
[17:50:21 CEST] <cone-648> ffmpeg 03Derek Buitenhuis 07master:5f587b1daf90: Merge commit '3b08d9d932eef09403074d5af31e10d8011e840b'
[17:52:24 CEST] <cone-648> ffmpeg 03Anton Khirnov 07master:a638e9184d63: vf_fade: make sure the slice end is always in the frame
[17:52:25 CEST] <cone-648> ffmpeg 03Derek Buitenhuis 07master:aa40df483b21: Merge commit 'a638e9184d63e57e67901f34afe919fd56fd3ac4'
[17:53:28 CEST] <Daemon404> who is our qsv person
[17:53:37 CEST] <Daemon404> qsvdec_h2645: switch to the new BSF API
[17:53:40 CEST] <Daemon404> looks nontrivial to merge
[17:57:13 CEST] <jamrial_> Daemon404: Ivan Uskov maybe?
[18:02:06 CEST] <nevcairiel> like those people are going to be of much help =p
[20:15:47 CEST] <cone-648> ffmpeg 03Paul B Mahol 07master:c85d04251df6: avcodec: add TrueMotion 2.0 Real Time decoder
[20:24:43 CEST] <cone-648> ffmpeg 03Michael Niedermayer 07master:7b7c338e9a73: avfilter: Add AV_OPT_FLAG_FILTERING_PARAM to where it was missing
[21:09:32 CEST] <wm4> how does h264 code interlaced video? separate fields, or something worse?
[21:10:20 CEST] <JEEB> the cable tv mafia made there be two modes
[21:10:23 CEST] <JEEB> MBAFF and PAFF
[21:10:35 CEST] <JEEB> and all the implementors rejoiced
[21:10:39 CEST] <JEEB> (not)
[21:10:47 CEST] <wm4> yeah I heard it's complex
[21:12:00 CEST] <nevcairiel> mbaff sounds good in concept, but implementation wise its of course complex
[21:13:27 CEST] <atomnuker> afaik MBAFF is more efficient
[21:22:19 CEST] <nevcairiel> definitely can be
[21:34:58 CEST] <iive> it would have been quite a bummer if it was complex, slow and inefficient
[21:35:06 CEST] <iive> also, interlace must die
[21:42:51 CEST] <fritsch> next gen live tv standard has it whiped
[21:42:56 CEST] <fritsch> hevc 10 bit uhd progressive
[21:50:12 CEST] <Shiz> 12 bit!1
[22:02:21 CEST] <iive> well, iso14496-2 (Mpeg4 ASP) also had it wiped out... but it came back.
[23:21:14 CEST] <pfelt1> afternoon all. i've gotten pretty far on my filter to allow me to swap streams without killing ffmpeg, however i've hit a small problem. every time i swap streams my audio and video get out of sync by about 1 second. only thing i can think of is that the audio data is in a buffer waiting to be output (something like multiple frames per packet). could that be the case, and if so, is there a way to just snarf all the remaining frames from the buffer and
[23:39:03 CEST] <pfelt1> hmm. actually that can't be it. the av_frame_get_best_effort_timestamp() is resetting to 0. it's just that the decoder is able to pull audio before video
[00:00:00 CEST] --- Fri Apr 22 2016
1
0
[00:02:38 CEST] <johnnny22-afk> that's a bummer :P
[00:04:58 CEST] <johnnny22-afk> And if I recombine the head after ?
[00:05:21 CEST] <johnnny22-afk> The advantage of mpegts is that I can just cut the head of the file even while it's playing.
[00:05:39 CEST] <johnnny22-afk> as long as I don't remove the part where the playhead is at.
[00:05:49 CEST] <kepstin> yeah, as far as I know that's the only format which supports that
[00:06:29 CEST] <johnnny22-afk> only format that supports audio & video with sync'ing right ? obviously I could probably go with some rawvideo and rawaudio !? hehe
[00:06:33 CEST] <kepstin> since it is after all designed as a streaming format for broadcasts, where someone can turn on a TV at any time and start at the first bit their receiver picks up
[00:06:37 CEST] <kepstin> heh
[00:06:50 CEST] <johnnny22-afk> what about that 'nut' format ?
[00:07:07 CEST] <kepstin> i'm pretty sure that also has some required global headers at the start
[00:07:08 CEST] <johnnny22-afk> nvm, it seems to say nonseekable
[00:07:37 CEST] <johnnny22-afk> what's that other format that ffserve uses humm
[00:08:05 CEST] <johnnny22-afk> FFM2
[00:08:47 CEST] <kepstin> i dunno exactly what you're trying to do, but using a custom player that connects to a server which handles all the seeking, segmentation/etc and provides a continuous mpeg-ts (or whatever) stream might be something to look at.
[00:09:53 CEST] <johnnny22-afk> very true, I did ponder about that too. I might revisit that idea.
[00:10:38 CEST] <johnnny22-afk> But I'd probably have to go with rtsp to handle seeking etc.. right ?
[00:11:04 CEST] <kepstin> looks like ffm/ffm2 is just an internal circular buffer format for ffserver, it's not supported by any other tools and isn't even compatible acress ffmpeg versions :/
[00:11:19 CEST] <kepstin> or have out-of-band signalling in some other way, yeah
[00:12:01 CEST] <johnnny22-afk> And the best bet is probably some UDP stream straight to the player.
[00:12:47 CEST] <kepstin> if you're able to, sure :) Use rtp for that unless you have a good reason to do otherwise.
[00:18:28 CEST] <johnnny22-afk> I'll think of that
[01:08:32 CEST] <jacobwg> Does anyone know why transcoding this mkv into an mp4 with an additional audio track is causing that additional audio track to be in a different alternate group? And do you know how I might go about making both tracks be in the same alt group? https://gist.github.com/jacobwgillespie/baf9223ac2c58c5bd73e373dda137395
[01:13:18 CEST] <pzich> jacobwg: I haven't dealt with multiple audio tracks, so I can't really, but I recommend checking out these: https://trac.ffmpeg.org/wiki/AudioChannelManipulation and https://trac.ffmpeg.org/wiki/Map
[01:14:58 CEST] <jacobwg> pzich: I'll take a look - this "alternate group" thing is specific to the container I believe - I'm digging into the source now to see how it's computed: https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/movenc.c#L2419
[01:17:03 CEST] <pzich> ah yeah, that's far beyond my area of knowledge. good luck :-/
[01:27:05 CEST] <johnnny22-afk> kepstin: if i was to go with a streaming server solution, wouldn't that server have to deal with those same issues ?
[01:28:26 CEST] <kepstin> Sure, but you can use formats that the clients wouldn't understand, you can get cross segment seeking correct, etc.
[01:32:05 CEST] <johnnny22-afk> Seems like still a hassle to make sure the server can support that. But you do have a point, that it makes the player side simpler. While I'm still not sure of such server that would support nicely those seeking within multi-segment mpegts's or other formats. I guess I could use mpeg-dash too in such a case on the server-side.
[01:32:52 CEST] <johnnny22-afk> But, out of the box, what server supports this nicely, while supporting the possibility to remove segments or pieces at the head of the file to recreate a sliding window system.
[01:33:54 CEST] <johnnny22-afk> Seems like I'd have to implement quite a bit.
[01:51:39 CEST] <pfelt1> say i've got three ffmpeg commands. two of them create streams of data and then the third pulls that data to do something with it. is there any better way than to use a fifo between them?
[04:55:01 CEST] <fling> Downgrading to 4.1.21 solved the issue. Is it something wrong with v4l2 or with what?
[06:42:14 CEST] <Prelude2004c> hey guys.. good evening to everyone
[06:42:41 CEST] <Prelude2004c> question.. silencedetect.. anyone familiar with it ? silencedetect=n=-50dB:d=5 for example.. i want to say if it detects silence for longer than say 5 seconds.. write to a log file or something.. how does one do that
[09:45:16 CEST] <termos> is there a way to avoid avformat_open_input reading from the input before I am ready? My init phase takes some time and by the time I'm finished I get a huuuuge burst of frames because somehow they are bufferet up when I try to av_read_frame. Ideas?
[10:09:23 CEST] <t4nk407> Hey :) I gto the same problem this guy has : http://ffmpeg-users.933282.n4.nabble.com/wall-clock-time-s-in-setpts-filter…
[10:09:29 CEST] <t4nk407> any suggestions ?
[10:10:02 CEST] <ggugi> hi everyone
[10:45:49 CEST] <Xen0_> is there a reason i always get this error?
[10:45:51 CEST] <Xen0_> Stream #0:3 -> #0:0 (dvd_subtitle (dvdsub) -> ssa (native))
[10:45:51 CEST] <Xen0_> Press [q] to stop, [?] for help
[10:45:51 CEST] <Xen0_> [ssa @ 0xfe4180] Only SUBTITLE_ASS type supported.
[10:45:51 CEST] <Xen0_> Subtitle encoding failed
[10:46:13 CEST] <Xen0_> everything i read says it should work fine but never does
[11:03:52 CEST] <furq> Xen0_: i don't think you can automatically convert dvd subtitles to a text subtitle format
[11:05:38 CEST] <Xen0_> oh
[11:07:13 CEST] <Xen0_> hmm
[11:09:12 CEST] <furq> there are a million different windows tools that will do it
[11:09:28 CEST] <furq> i think subextractor is the cool one at the moment but i've not had to do it in years
[11:09:46 CEST] <furq> as far as *nix goes i don't know of anything other than avidemux
[11:10:09 CEST] <Xen0_> yea in on linux
[11:10:25 CEST] <Xen0_> ffmpeg -i input.mkv -filter_complex "[0:v][0:s]overlay[v]" -map "[v]" -map 0:a <output options> output.mkv
[11:10:33 CEST] <Xen0_> if i use this
[11:10:49 CEST] <Xen0_> and say video is 0:1
[11:11:06 CEST] <Xen0_> do i replace the other [v] with 1
[11:11:20 CEST] <t4nk407> Hey :) I gto the same problem this guy has : http://ffmpeg-users.933282.n4.nabble.com/wall-clock-time-s-in-setpts-filter…
[11:11:21 CEST] <furq> no
[11:12:26 CEST] <furq> Xen0_: i'm pretty sure mkv supports dvd subtitles so you should be able to use -c:s copy and avoid burning them in
[11:13:21 CEST] <Xen0_> i need them burned in for my end usage is the issue
[11:13:44 CEST] <Xen0_> my final output is webm encodes
[11:14:24 CEST] <furq> well using video filters will force a transcode, so you probably want to encode it straight to webm
[11:14:43 CEST] <furq> unless one of your output options is -q:v 0
[11:16:01 CEST] <Xen0_> im ok with reencoding after i get them burned in
[11:16:30 CEST] <Xen0_> but when i try i it with -vf subtitle=foo.mkv i still get an error
[11:20:45 CEST] <furq> that only works with text subtitles
[11:22:18 CEST] <Xen0_> ahh
[11:22:36 CEST] <Xen0_> explains it
[11:24:39 CEST] <Xen0_> i think just got the overlay filter to work
[11:25:18 CEST] <Xen0_> ill have to reencode to webm still but im ok with that
[11:38:35 CEST] <t4nk407> is there a way, to start 4 streams with one command, of wich one is in the upper left, one in the upper right , one in the lower left and one in the lower right of the original video ?
[12:38:24 CEST] <Godspeed990> Hi, I am trying to push a video to a video decoder. I have a mp4 file. I need a PES file. Which option for ffmpeg should I use. I am currenlty extracting a h264 stream. Which I can't play?
[12:44:28 CEST] <Eiken_> anyone know if its possible to use CDL values with ffmpeg somehow?
[14:14:48 CEST] <t4nk421> hey :) any Idea, why this won't work : ffmpeg -f dshow -video_size 1280x720 -rtbufsize 999999k -i video="screen-capture-recorder" -r 20 -c copy -threads 4 -f mpegts udp://239.255.1.1:1234
[14:15:28 CEST] <t4nk421> I don't want to transcode the stream, because I think that would cause to high latency
[14:23:10 CEST] <furq> t4nk421: mpegts can't contain rawvideo or pcm audio
[14:23:14 CEST] <furq> which i assume is what you get out of dshow
[14:27:54 CEST] <t4nk421> ah that is it , thank you ;)
[14:44:58 CEST] <t4nk421> are there other muxers similar like mpegts wich can stream raw streams ?
[15:32:19 CEST] <ek_> Hello, a colleague is trying to install ffmpeg using homebrew on his Mac, but the ffmpeg.org site seems to be refusing connections on port 443; even the mailing list links on that site are broken since they use HTTPS. Does anyone know if there is a way around this problem?
[15:33:13 CEST] <furq> https is working fine for me
[15:34:44 CEST] <ek_> Hmm. Maybe the problem is on my end, then. Thank you.
[15:36:05 CEST] <maziar> i want to create manifest like this http://pastebin.com/U5Q92qUq , i can create *.m3u8 but i don't know how to create master.m3u8 to refer to some quality, please kindly help me
[15:39:41 CEST] <ek_> quit
[15:58:49 CEST] <neuro_sys> I have a set of PNGs with black background on top a textured polygon moving only. I'd like to overlay this on top of a video, how would I go about doing that?
[15:59:32 CEST] <furq> neuro_sys: https://ffmpeg.org/ffmpeg-filters.html#colorkey
[16:01:01 CEST] <neuro_sys> furq: thanks!
[16:09:04 CEST] <neuro_sys> Could someone explain the terms used in here? I couldn't make sense of the manual description: ffmpeg -i video.mkv -i image.png -filter_complex '[0:v][1:v]overlay[out]' -map '[out]' out.mkv
[16:09:18 CEST] <neuro_sys> [0:v] means the first input's video stream
[16:09:27 CEST] <neuro_sys> [1:v] means the second input's video stream
[16:09:58 CEST] <c_14> technically, streams. But yes
[16:10:01 CEST] <neuro_sys> overlay is the overlay filter. So how do filters get their parameters?
[16:10:06 CEST] <neuro_sys> by prefixed with streams?
[16:10:21 CEST] <c_14> filters take inputs and have outputs
[16:10:29 CEST] <c_14> they also have options which have defaults if not explicitly set
[16:10:55 CEST] <neuro_sys> the inputs are the ones inside brackets to the left of filter names, and the output is to the right?
[16:11:04 CEST] <c_14> yep
[16:11:50 CEST] <neuro_sys> so parameters are indexed? because we didn't give those streams any names for the overlay filter to make sense.
[16:12:09 CEST] <neuro_sys> https://ffmpeg.org/ffmpeg-filters.html#overlay-1
[16:12:14 CEST] <neuro_sys> It doesn't speak about ordering here
[16:12:28 CEST] <c_14> it does
[16:12:33 CEST] <c_14> >It takes two inputs and has one output. The first input is the "main" video on which the second input is overlaid.
[16:12:39 CEST] <neuro_sys> oh yes silly me
[16:15:01 CEST] <neuro_sys> In this example: -filter_complex "[0:v][1:v] overlay=25:25:enable='between(t,0,20)'" I see that it's possible to assign values to filters using equals sign?
[16:15:14 CEST] Action: neuro_sys scans through the manual to find the meaning of that
[16:15:27 CEST] <c_14> those are filter options
[16:15:59 CEST] <c_14> 4.1 Filtergraph Syntax
[16:16:04 CEST] <c_14> https://ffmpeg.org/ffmpeg-filters.html#Filtergraph-syntax-1
[16:18:37 CEST] <neuro_sys> beautiful
[16:24:00 CEST] <neuro_sys> http://i.imgur.com/Sn2PyKS.gif
[16:24:02 CEST] <neuro_sys> xD
[16:24:21 CEST] <neuro_sys> it's the best tool
[16:27:07 CEST] <maziar> i want to create manifest like this http://pastebin.com/U5Q92qUq , i can create *.m3u8 but i don't know how to create master.m3u8 to refer to some quality, please kindly help me
[16:35:16 CEST] <m3gab0y> hey all :) anyone up for a command line challenge? I want to setup complex filter overlays but can't get it right
[17:00:45 CEST] <anadon_> I'm trying to use a FILE* to access decoding an H264 encoded video (so stdin and files can both be used) but all the examples I'm pulling up rely on a function call which take a filepath. Where should I be looking to do this?
[17:07:02 CEST] <neuro_sys> what is a filter that does the inverse of colorkey?
[17:08:14 CEST] <c_14> presumably overlay over a color?
[17:08:50 CEST] <neuro_sys> c_14: indeed
[17:09:21 CEST] <c_14> then there's your answer
[17:09:50 CEST] <neuro_sys> yeah thanks, I'm checking
[17:15:53 CEST] <Mavrik> anadon_, you'll probably gonna have to create your on avio_context and implement read() calls
[17:16:27 CEST] <anadon_> Mavrik: That sounds hard. Is it?
[17:17:05 CEST] <Mavrik> Well, you call https://ffmpeg.org/doxygen/3.0/avio_8h.html#a853f5149136a27ffba3207d8520172… to allocate the avio_context
[17:17:20 CEST] <Mavrik> And you'll need a read_packet function that will read data.
[17:17:31 CEST] <Mavrik> And then set that aviocontext to "pb" variable in avformat_context.
[17:19:11 CEST] <anadon_> Mavrik: Do you think named pipes might me a better way for me to go about this?
[17:19:53 CEST] <Mavrik> How would named pipes help?
[17:20:03 CEST] <Mavrik> Are you calling the ffmpeg binary or are you using libav API?
[17:20:38 CEST] <kepstin> on linux, you might be able to hack it by opening /dev/fd/# as a file
[17:21:01 CEST] <anadon_> Mavrik: API -- it looks like I'd need to know many details of the h264 format, which really isn't worth it.
[17:21:12 CEST] <Mavrik> Huh, what?
[17:21:34 CEST] <Mavrik> Named pipes seems convoluted way of doing a solution when you have the AVIO API right there for that exact purpose.
[17:21:42 CEST] <Mavrik> No idea why'd you have to know the details.
[17:21:50 CEST] <Mavrik> You'll just get a read (or seek) call and you react on it.
[17:22:27 CEST] <anadon_> kepstin: Doesn't that have files that behave a little diferently when it comes to stuff like seeking? I'd have to know if any file operations of the fd/# would be aversely affected.
[17:23:39 CEST] <anadon_> Mavrik: Could the read function work as a pull of 1 byte at a time from a FILE*?
[17:24:24 CEST] <kepstin> anadon_: re-opening an fd via the /dev/fd/# interface is equivalent to duplicating the file handle, so stuff like seeking, etc. will be independent.
[17:25:43 CEST] <Mavrik> anadon_, not sure I understand the question
[17:25:46 CEST] <Mavrik> look at the signature of the method
[17:25:49 CEST] <Mavrik> you get a buffer and size
[17:25:53 CEST] <Mavrik> and you fill the buffer with file data
[17:26:03 CEST] <anadon_> kepstin: yikes. So that may or may not break. I wonder why there's not support for using FILE* over a file address, unless the URL case is really that common.
[17:26:24 CEST] <kepstin> anadon_: I really can't think of any reason why you'd want to have ffmpeg reading from a file *and* read from it yourself
[17:26:28 CEST] <kepstin> that doesn't even make sense
[17:27:39 CEST] <anadon_> Mavrik: intuitively, the function suggests some weird parsing magics.
[17:27:48 CEST] <Mavrik> ?
[17:28:23 CEST] <anadon_> kepstin: What do you mean? I'm used to FILE* as a generic handle.
[17:29:04 CEST] <anadon_> Mavrik: I'm working through the documentation -- this is my first project in libav and I'm running into roadbumps.
[17:29:24 CEST] <kepstin> FILE* isn't really a generic handle, that's just a C library interface around fds, which are the actual generic handle :)
[17:31:55 CEST] <anadon_> kepstin: 'kay
[17:31:58 CEST] <t4nk281> hi tryng to use ffmpeg3 on windows 7 ¿ where is the .exe file ????
[17:33:01 CEST] <t4nk281> eoeoeo not dificult question ...
[17:33:14 CEST] <kepstin> but yeah, if you're gonna be using libav* to read media files, you should be letting the library handle all of the io in most cases. Reading from stdin is a bit of a special case - having libav* open /dev/fd/0 aka /dev/stdin and then *not touching stdin from your code* might be best? I'm not totally sure.
[17:33:22 CEST] <anadon_> kepstin: http://images.wikia.com/hunterxhunter/images/8/80/Alluka_nanika.png
[17:33:56 CEST] Action: kepstin hasn't seen HxH, so the reference goes over his head ;)
[17:34:00 CEST] <t4nk281> please ... I am in a hurry
[17:34:19 CEST] <anadon_> kepstin: The character only says " 'kay "
[17:34:42 CEST] <kepstin> t4nk281: I assume you probably downloaded the source code. If you want a build, try https://ffmpeg.zeranoe.com/builds/
[17:34:58 CEST] <kepstin> t4nk281: and random people in this chat channel are not obligated to provide you support.
[17:35:03 CEST] <furq> kepstin: you're way too nice
[17:36:24 CEST] <t4nk281> Thank you : "ffmpeg-3.0.tar.bz2" seems ot to be a build ?
[17:36:44 CEST] <c_14> no, that's the source
[17:37:05 CEST] <t4nk281> ok thank again I go to your link
[17:38:46 CEST] <t4nk281> bye !
[18:08:56 CEST] <anadon_> kepstin: in sws_getContext, arguments 3 and 6 take something like "AV_PIX_FMT_YUV420P" -- hardcoding this is probably a bad idea. What's the pointer path in AVCodecContext or AVFormatContext?
[18:10:29 CEST] <JEEB> http://ffmpeg.org/doxygen/trunk/structAVCodecContext.html
[18:10:41 CEST] <JEEB> ctrl+f AV_PIX_FMT
[18:28:34 CEST] <anadon_> What is the correct include path for avformat.h? It doesn't seem to be under avformat/ or avutil/ .
[18:29:20 CEST] <thebombzen> anadon_: try libavformat/avformat.h
[18:29:21 CEST] <furq> it should be in libavformat
[18:29:26 CEST] <thebombzen> oh sniped
[18:29:32 CEST] <furq> ;_;
[18:29:56 CEST] <thebombzen> thx for reminding me to rebuild ffmpeg
[18:30:08 CEST] <furq> i wonder if freebsd have updated yet
[18:30:17 CEST] <anadon_> Thanks!
[18:33:17 CEST] <anadon_> "avcodec_alloc_frame()" is in the libavutil/format.h include, isn't it?
[18:33:52 CEST] <anadon_> nv, got it
[18:42:01 CEST] <anadon_> I know I'm doing something dumb here....
[18:42:36 CEST] <anadon_> does sws_getContext() or sws_scale() throw "[swscaler @ 0xafb940] bad dst image pointers" ?
[19:06:19 CEST] <anadon_> Can someone tell me what I'm doing dumb here? It's causing a segfault later in the program, likely because I'm using something incorrectly. http://pastebin.com/Czgw58QG
[19:07:23 CEST] <DHE> outdata[1] on line 19 isn't valid
[19:08:26 CEST] <anadon_> Howso?
[19:09:26 CEST] <anadon_> yes, and doesn't the RGB32 codec only write out to channel 0?
[19:10:07 CEST] <anadon_> and then...
[19:10:08 CEST] <DHE> what's on line 19?
[19:10:09 CEST] <anadon_> yup
[19:10:53 CEST] <anadon_> Let's play 'how much sleep did I really get last night'
[19:11:26 CEST] <DHE> oh geez, I'll be playing that game tomorrow...
[19:18:49 CEST] <anadon_> Any idea is sws_getContext() has any issue converting to strange resolutions? (32X18)
[19:19:43 CEST] <DHE> if this is linux, compile with -g and run under valgrind
[19:45:15 CEST] <anadon_> DHE: I'm still getting corrupted frames back.
[19:51:00 CEST] <anadon_> I need a second set of eyes on this. It should be easy: http://pastebin.com/3GwKxwqZ
[19:59:15 CEST] <anadon_> Relocated -- did I miss anything?
[20:04:09 CEST] <jack54> Can anyone help me with capturing all unique frames from an OS X display to a QuickTime compatible H.264 file?
[20:04:28 CEST] <jack54> I already have: ffmpeg -framerate 120 -f avfoundation -i "1" -c:v libx264 -qp 18 -preset ultrafast out.mkv
[20:05:14 CEST] <jack54> I think I need to add a decimate filter that cuts out duplicate frames and tweak the output so it's a 60 FPS movie in 420/NV12 format that QuickTime understands
[20:32:18 CEST] <anadon_> jack54: mind taking a look a t my code? http://pastebin.com/qPFq6eW1
[20:32:28 CEST] <anadon_> I'm getting corrupted frame
[20:33:18 CEST] <jack54> anadon_: I can't help you -- I am just stuck with a question myself :)
[21:04:24 CEST] <utack> i guess this is pretty stale and will not be adressed? https://trac.ffmpeg.org/ticket/1885
[21:04:33 CEST] <utack> not that it is high priority or anything, just wondering
[21:24:16 CEST] <andrey_utkin> is concatdec with "-auto_convert 0" supposed to fail on mp4 files? it does: test script: https://gist.github.com/andrey-utkin/53c33e11b68eaa54d67aa25aaa40dd58 output: https://gist.github.com/andrey-utkin/86ca4c63e8340b015edf4f538ba08456
[21:24:32 CEST] <prelude2004cZzzz> hey guys.. looking for help.. can someone tell me if this looks right ? http://pastebin.com/y34iadDv >> i keep getting random audio sync issues and things come out of sync for some reason
[21:25:37 CEST] <prelude2004cZzzz> ? that is what i did
[21:27:20 CEST] <sfan5> um
[21:27:24 CEST] <relaxed> is there a reason you're not using nvenc through ffmpeg?
[21:27:30 CEST] <sfan5> ffmpeg can support nvenc as an encoder
[21:27:35 CEST] <sfan5> without any piping workaround
[21:29:34 CEST] <andrey_utkin> prelude2004cZzzz, no exact command line and no command output at all
[21:31:11 CEST] <prelude2004cZzzz> yes found quality working better with the library direct
[21:31:19 CEST] <prelude2004cZzzz> shows great improvement over the ffmpeg version
[21:31:29 CEST] <prelude2004cZzzz> but i have tried with both and still the same issue
[21:31:39 CEST] <prelude2004cZzzz> something is throwing off the sound sync
[21:32:02 CEST] <prelude2004cZzzz> am i not preserving timestamps correctly?
[21:32:04 CEST] <relaxed> did you look at the options listed from "ffmpeg -h encoder=nvenc" ?
[21:32:07 CEST] <sfan5> that's most likely caused by that piping workaround
[23:11:29 CEST] <Guster_> hi there
[23:11:47 CEST] <Guster_> i have little problem with ffmpeg on ubuntu
[23:12:19 CEST] <Guster_> does someone have some time for me please? :D
[23:13:03 CEST] <sfan5> just ask your question
[23:13:06 CEST] <sfan5> someone will help if they can
[23:13:14 CEST] <Guster_> okey
[23:13:44 CEST] <Guster_> when i try to feed my test stream from web cam on server in my local network
[23:14:10 CEST] <Guster_> i get a error message and break the feed
[23:15:00 CEST] <Guster_> Thu Apr 21 23:12:57 2016 127.0.0.1 - - [POST] "/feed1.avi HTTP/1.1" 200 1282
[23:15:00 CEST] <Guster_> this is message on ffserver
[23:15:00 CEST] <Guster_> av_interleaved_write_frame(): Broken pipe
[23:15:00 CEST] <sfan5> damn bot
[23:15:00 CEST] <sfan5> some more console output would be nice
[23:15:08 CEST] <sfan5> yeah there it is
[23:15:14 CEST] <Guster_> ok i will paste
[23:15:56 CEST] <Guster_> there
[23:15:57 CEST] <Guster_> http://pastebin.com/fXw9qw6V
[23:16:00 CEST] <Guster_> ^^
[23:17:03 CEST] <sfan5> does ffserver output an error?
[23:17:26 CEST] <Guster_> noup
[23:17:31 CEST] <Guster_> Thu Apr 21 23:12:57 2016 127.0.0.1 - - [POST] "/feed1.avi HTTP/1.1" 200 1282
[23:17:35 CEST] <Guster_> header 200
[23:17:48 CEST] <Guster_> and some number after
[23:19:23 CEST] <Guster_> i have to go afk for 15minutes
[23:19:27 CEST] <Guster_> brb
[23:20:13 CEST] <sfan5> hm
[23:21:14 CEST] <sfan5> ah
[23:21:14 CEST] <sfan5> Guster_: [tcp @ 0x29d4fc0] Connection to tcp://localhost:8090 failed (Connection refused)
[23:29:21 CEST] <juls> hi all
[23:36:58 CEST] <Guster_> sfan5: here i am
[23:41:40 CEST] <Guster_> can anyone else help me please?
[00:00:00 CEST] --- Fri Apr 22 2016
1
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[00:01:20 CEST] <wm4> why wouldn't it
[00:02:59 CEST] <Angus> ahh, cool.
[00:03:48 CEST] <Angus> Calling vdpau_init after avcodec_open2 is sane?
[01:16:49 CEST] <cone-039> ffmpeg 03Petru Rares Sincraian 07master:0b32aff83d87: fate: Add test for mts2 (mss4) codec
[02:29:35 CEST] <cone-039> ffmpeg 03Michael Niedermayer 07master:9a6ae7fbb2fd: fate: Add test for ticket 1833 (Ogg/Vorbis Chapters)
[04:35:15 CEST] <rcombs> nevcairiel: can you think of any reasons why auto-bsf would need to support multiple bitstream filters on a single stream
[09:35:21 CEST] <nevcairiel> rcombs: its probably unlikely to happen
[13:20:07 CEST] <durandal_1707> michaelni: one can only mentor single project per gsoc policy
[13:20:34 CEST] <durandal_1707> dunno about backup mentors
[13:21:03 CEST] <durandal_1707> can one be backup and major at same time
[13:23:45 CEST] <michaelni> durandal_1707, what policy ?
[13:28:04 CEST] <durandal_1707> michaelni: one can mentor only one project
[13:28:07 CEST] <michaelni> durandal_1707, just tried, i can assign myself (and them unassgned) as mentor to 2 projects
[13:28:34 CEST] <durandal_1707> I read it somewhere...
[13:28:40 CEST] <michaelni> where ?
[13:38:49 CEST] <durandal_1707> michaelni: can't find it anymore
[14:42:19 CEST] <Daemon404> i may be juvenile, but i find this very amusing: http://ffmpeg.org/pipermail/ffmpeg-devel/2016-April/193306.html
[15:59:10 CEST] <jamrial> nevcairiel: re adtstoasc, since rcombs sent a patch to test it we can drop mine
[16:01:50 CEST] <jamrial> mine adds an extra dependency to what should depend on latm muxer only after all
[16:02:17 CEST] <wm4> michaelni: can you ok or not-ok my decoder API patches? (especially the lavf change)
[16:04:13 CEST] <michaelni> wm4, moment, let me re-take a quick look
[17:01:50 CEST] <michaelni> wm4, done
[17:03:21 CEST] <wm4> thanks
[18:19:42 CEST] <jsebechlebsky> Hello, is Nicolas G. here?
[18:21:17 CEST] <ubitux> rarely
[18:21:47 CEST] <ubitux> when he is, he uses Cigaes as nickname
[18:21:57 CEST] <ubitux> but he only comes on special occasions
[18:29:08 CEST] <jsebechlebsky> Thanks for info :), I guess I'll just send e-mail then
[18:30:26 CEST] <pfelt> morning all. i've got something of an oddity that i'm strugging to debug. i have two ffmpeg commands, one pulls a udp stream, transcodes it a bit, and writes it to a named pipe. the second reads from that named pipe using the movie filter. if i run the pipe reader in gdb using the non stripped version, then i start the writer it almost always dies as the reader breaks the pipe. however when running in gdb the process itself is still running
[18:30:43 CEST] <pfelt> if i then re-run the writer, data flows and the stream comes up on output
[18:31:12 CEST] <pfelt> *then* if i rerun that sequence, just not in gdb, the reader terminates when it breaks the pipe
[18:32:01 CEST] <pfelt> anyone have any ideas on what i can do to debug this? i'm at a loss on where to go from here as i've never seen something run in gdb and not run when not in gdb (sorry for the add double negative phrasing there)
[18:35:56 CEST] <cone-241> ffmpeg 03Tobias Rapp 07master:ef59c6f7e89a: fate: add readvitc filter test
[19:03:08 CEST] <Shiz|test> pfelt: seems like it may be better fit for the user chan
[19:08:56 CEST] <pfelt> didn't seem like there were many dev types in there last time i asked
[19:24:03 CEST] <cone-241> ffmpeg 03Henrik Gramner 07master:8dd3ee9dddca: x86inc: Fix AVX emulation of scalar float instructions
[19:24:04 CEST] <cone-241> ffmpeg 03Anton Mitrofanov 07master:42be240ad687: x86inc: Fix AVX emulation of some instructions
[19:24:05 CEST] <cone-241> ffmpeg 03Anton Mitrofanov 07master:4bd5583acea5: x86inc: Improve handling of %ifid with multi-token parameters
[19:24:06 CEST] <cone-241> ffmpeg 03Anton Mitrofanov 07master:e428f3b30ceb: x86inc: Enable AVX emulation in additional cases
[22:48:45 CEST] <cone-241> ffmpeg 03Paul B Mahol 07master:38797a8033d0: avcodec/takdec: add code that got somehow lost in process of REing
[00:00:00 CEST] --- Thu Apr 21 2016
1
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[00:26:38 CEST] <Prelude2004c> hey guys.. question.. i am running itsoffset -6 seconds to basically fix up some stuff... but the cc data is really behind now. is there something i can use to speed up closed caption and set it faster or slower to sync with the audio ?
[01:01:27 CEST] <gusto> hey
[01:01:44 CEST] <gusto> any idea why ffmpeg has problems resolving when there are many A records?
[01:02:03 CEST] <gusto> but I see that the problem also affects "host" but not "dig" interestingly
[06:59:38 CEST] <dongs> if I wantted to use vf=prespective to make video fit into http://www.wwvalue.com/cont/2012/05/2012052101display-17.jpg how do I calculate the points?
[07:00:42 CEST] <dongs> is that just x,y coordinates for each corner?
[07:19:48 CEST] <dongs> hm, sorta.
[07:19:54 CEST] <dongs> but it ends up taking up whole screen
[07:20:29 CEST] <dongs> not sure how to constrain it to original
[07:22:58 CEST] <dongs> perspective looks also fliipped
[07:23:01 CEST] <dongs> halp :(
[07:34:33 CEST] <dongs> https://www.dyne.org/software/frei0r/ absolutely awful fucking font
[07:34:48 CEST] <dongs> entire page is unreadable, I *STILL* cannot find a list of all the shit included in those "effects"
[09:39:44 CEST] <Gringham> Hi , I am receiving an incoming unicast and send it out via multicast (ffmpeg -i unicastadress -f mpegts multicast). Is there a way, to add a seperate audiofile or stream while doing this ?
[09:40:08 CEST] <Gringham> to the multicast*
[09:40:13 CEST] <dongs> sure
[09:40:24 CEST] <dongs> duno how it'll deal wiht modifying PAT/PMT and shit tho
[09:41:26 CEST] <Gringham> this is acceptable in the first test I think
[09:42:23 CEST] <dongs> just add another -i crap.aac and hope it will know what to do
[09:42:50 CEST] <dongs> or you mihgt have to have a copy of ffmpeg streaming it "looped" and then add that as input source
[09:44:38 CEST] <Gringham> thx, I will try that
[09:45:50 CEST] <dongs> for pmt modification youre gonna need to use -metadata: thing i think
[10:38:59 CEST] <Gringham> if I use -vcodec rawvideo to stream a video , how can I open it on the other side ?
[10:58:39 CEST] <dongs> err
[10:58:46 CEST] <dongs> isnt that uncompressed
[11:52:43 CEST] <hc2p> hey, how do i secure the rtmp input? with ffmpeg -listen 1 -i rtmp://user:pass@streaming.myip.com/live/streamtoken i still can connect to it regardless of correct username/password and streamtoken.
[13:20:53 CEST] <_Timon> How would I convert from h265 to h264 without any quality loss?
[13:21:05 CEST] <JEEB> by using lossless encoding
[13:21:10 CEST] <JEEB> which of course will be huge
[13:21:12 CEST] <_Timon> I found some lines but they all included a -crf suffix which changes the bitrate as far as I know
[13:21:19 CEST] <_Timon> file size is irrelevant
[13:21:26 CEST] <furq> not changing the bitrate isn't going to prevent quality loss
[13:21:40 CEST] <JEEB> then use -q:v 0 as that is always set to lossless mode in x264
[13:21:42 CEST] <furq> quite the opposite
[13:21:50 CEST] <JEEB> crf 0 is only lossless when the x264 is built for 8bit encoding
[13:22:08 CEST] <JEEB> the resulting file will be huge and not decode'able by most HW decoders
[13:22:30 CEST] <JEEB> but at least it will be exactly the same as the decoded HEVC stream, since you use lossless coding
[13:22:34 CEST] <_Timon> My h265 file is 3:45 long 1,5GB 3840x2160 60fps 56.2Mbps
[13:23:02 CEST] <_Timon> What would you guys recommend?
[13:23:36 CEST] <furq> i would recommend trying various crf settings and seeing which one gives the least bad set of compromises
[13:23:54 CEST] <JEEB> personally I would say go for the source and re-encode that, but I guess for you the HEVC file is the "source" :P
[13:24:01 CEST] <furq> or reevaluating your need to reencode it, if possible
[13:24:06 CEST] <_Timon> ya hevc is source sadly :/
[13:24:52 CEST] <_Timon> I'll try a lossless first
[13:24:54 CEST] <_Timon> see where that leads me
[13:25:10 CEST] <furq> i can't imagine any situation where lossless is what you want, but who knows
[13:25:36 CEST] <JEEB> well it is faster to decode with lavc than HEVC
[13:25:45 CEST] <JEEB> so if you just want to play it locally using some lossless format is OK
[13:25:53 CEST] <JEEB> if you need HW decoding support you'll have to do lossy
[13:26:09 CEST] <furq> is it really faster to decode lossless x264 than h265
[13:26:15 CEST] <JEEB> yes
[13:26:22 CEST] <JEEB> HEVC decoder misses a lot of optimizations
[13:26:31 CEST] <JEEB> in which case it is a case for nonzero CRF (start 23 and move up if it looks good and down if it looks bad)
[13:26:37 CEST] <furq> that doesn't match up with my experience but it's been a while since i dealt with lossless x264
[13:27:02 CEST] <JEEB> well the more bit rate you have the more data you have to process, but HEVC in general is slower to decode than AVC
[13:27:11 CEST] <JEEB> (and "helped" even more by the lack of asm in mainline FFmpeg)
[13:27:22 CEST] <_Timon> It's not about "looking good" in my case
[13:27:25 CEST] <JEEB> nev's FFmpeg fork has some assembly optimizations from openhevc ported
[13:27:26 CEST] <_Timon> it's about maxing out my hardware
[13:27:27 CEST] <furq> well for 60fps 4k i can see that being the bottleneck over decoder optimisations
[13:27:40 CEST] <JEEB> depends, really :P
[13:27:55 CEST] <JEEB> I wouldn't be surprised if HEVC being just that much slower is the main thing here
[13:27:57 CEST] <furq> maxing out your hardware?
[13:28:00 CEST] <JEEB> but not like I haven't tested :P
[13:28:13 CEST] <JEEB> *not like I have done proper benchmarks
[13:28:45 CEST] <JEEB> with the openhevc optimizations from nev's repo you can get quite nice speed-up, but that will most probably never get merged into mainline
[13:28:52 CEST] <JEEB> because they're written with intrinsics
[13:28:57 CEST] <JEEB> not pure asm
[13:29:00 CEST] <JEEB> >_>
[13:31:54 CEST] <_Timon> wait.. so "-h265-params" changes h265 to h264?
[13:32:05 CEST] <_Timon> That doesn't make alot of sense to me
[13:32:07 CEST] <JEEB> there's no such option
[13:32:09 CEST] <furq> what
[13:32:12 CEST] <furq> also yeah, that
[13:32:28 CEST] <JEEB> there's x264-params and x265-params, which controls parameters given to the library of one or the other encoder
[13:33:05 CEST] <_Timon> okay...
[13:34:16 CEST] <JEEB> so when you're encoding with libx264 the other option does absolutely nothing since there's no such parameter defined in it, vice versa for the other one :P
[13:34:43 CEST] <_Timon> with what encoder am I encoding when im converting to 265 to 264?
[13:35:01 CEST] <_Timon> I guess 264?
[13:35:05 CEST] <JEEB> it should say in the log but even more specifically you should be specifying it yourself with -c:v or vcodec :P
[13:35:17 CEST] <JEEB> otherwise ffmpeg will pick a random encoder for you and you will have a tons 'o fun
[13:35:31 CEST] <_Timon> I'll google some more
[13:35:33 CEST] <_Timon> nothing is making sense
[13:35:50 CEST] <JEEB> well you are doing ffmpeg -i INPUT PARAMS OUTPUT, right?
[13:35:56 CEST] <_Timon> yep
[13:36:02 CEST] <JEEB> if you don't set a video or audio encoder in PARAMS
[13:36:07 CEST] <JEEB> guess what happens
[13:36:22 CEST] <JEEB> (hint: ffmpeg looks at your output container/format and decides from a list)
[13:36:26 CEST] <_Timon> It'll just look at what it is you're converting and basing it's encoder on that information
[13:36:31 CEST] <JEEB> no
[13:36:33 CEST] <JEEB> never
[13:36:38 CEST] <_Timon> So it's not using logic?
[13:36:40 CEST] <_Timon> :P
[13:36:43 CEST] <JEEB> your input never matters at that point since it's been decoded
[13:38:45 CEST] <JEEB> it's using "logic" depending on your output container/format which might or might not be what you want
[13:38:55 CEST] <_Timon> okay..
[13:39:01 CEST] <JEEB> which is why you always specify video and audio encoders with -c:v/a or -vcodec/acodec
[13:39:04 CEST] <JEEB> latter being the old way
[13:39:09 CEST] <JEEB> and -c:v|a being the new way
[13:39:20 CEST] <JEEB> AVC encoder would be libx264 :P
[13:40:53 CEST] <_Timon> Seems like it doesn't accept any input when im not defining a crf value
[13:43:50 CEST] <_Timon> ok crf 25 puts the video down to 9887 Kbps overall bitrate
[13:44:45 CEST] <_Timon> so that's not going to work
[14:25:42 CEST] <Infiltrator> How do I use VBR with opus? I tried -aq but it says "libopus @ 0x8d2f60] Quality-based encoding not supported, please specify a bitrate and VBR setting."
[14:26:21 CEST] <hc2p> has someone answered my question? sorry, i was disconnected. Question was regarding how to secure the ffmpeg rtmp connection in listening mode. It seems that whatever i set for appname and playpath everyone can just stream to it
[14:28:18 CEST] <kepstin> Infiltrator: vbr is the default with opus, just specify a bitrate.
[14:29:02 CEST] <kepstin> (in the default vbr mode, bitrate is basically used as a quality level rather than target bitrate)
[14:29:21 CEST] <Infiltrator> Ah okay. So just -abitrate?
[14:29:47 CEST] <kepstin> or -b:a if you want to use the short modern syntax :)
[14:30:24 CEST] <Infiltrator> kepstin: I'm chucking it into a script, so when I came back later I don't want to be scratching my head. :P
[14:30:29 CEST] <Infiltrator> Thanks.
[14:31:45 CEST] <Infiltrator> But abitrate doesn't work anyway. :\
[14:32:17 CEST] <kepstin> The vbr mode in opus is set so that everything encoded will be basically the same quality, and over a large corpus of files the average bitrate over everything will be close to the requested bitrate.
[14:33:31 CEST] <kepstin> ah, there was never an '-abitrate' option, it was '-ab'. Probably etter to use '-b:a' for consistency tho.
[14:33:38 CEST] <Infiltrator> Do you mean that there is no point setting the bitrate?
[14:33:51 CEST] <Infiltrator> Yeah, I just ended up doing -b:a.
[14:33:53 CEST] <kepstin> hmm? no, it sets the quality level
[14:34:15 CEST] <Infiltrator> What did you mean then by "basically the same quality"?
[14:34:21 CEST] <kepstin> add a "with the same bitrate setting" after "everything encoded"
[14:34:29 CEST] <Infiltrator> Ah okay.
[14:35:06 CEST] <Infiltrator> By the way, if I'm converting away from mpeg4, I don't need to worry about the B-frame warning, right?
[14:35:41 CEST] <kepstin> yeah, that's a warning from the decoder. If you're re-encoding, it's not an issue.
[14:35:50 CEST] <Infiltrator> Cool.
[14:35:54 CEST] <Infiltrator> Thanks for the help.
[14:38:21 CEST] <Infiltrator> Is there a good reference for the options to give to various codecs? I had a look at https://ffmpeg.org/ffmpeg-codecs.html#libopus but it didn't say much about opus.
[14:38:42 CEST] <Infiltrator> And even less about hevc.
[14:41:45 CEST] <furq> ffmpeg -h encoder=libx265
[14:42:13 CEST] <furq> most of the options are shared between codecs so they're not documented separately on the website
[14:53:24 CEST] <fling> DTS -295798475818, next:22628395 st:0 invalid dropping
[14:53:29 CEST] <fling> PTS -295798475818, next:22628395 invalid dropping st:0
[14:53:46 CEST] <fling> ^ video and audio out of sync when capturing from a webcam with 4.5.1 kernel
[14:54:05 CEST] <fling> tested with ffmpeg 2.8.6 and master.
[14:54:15 CEST] <fling> Is this a known behavior? How do I fix? ^
[14:54:37 CEST] <fling> Or should I just use the older kernel?
[15:01:01 CEST] <fling> The command I use -> https://bpaste.net/show/31b6d6944b89
[15:31:49 CEST] <__raven> hi (again) :)
[15:32:23 CEST] <__raven> i need to extract subtitle track out of a hls stream but i only get several different errors now
[15:32:47 CEST] <__raven> do i need the map option, the c:s:0.2 or just 0 option?
[15:32:53 CEST] <__raven> do i need a copy or srt?
[15:33:26 CEST] <furq> pastebin the errors
[15:33:33 CEST] <furq> or ideally the command line and full output
[15:33:45 CEST] <__raven> just a moment
[15:38:17 CEST] <__raven> http://pastebin.com/1GdJjznR
[15:38:31 CEST] <__raven> it should be the 0.2 unknown
[15:40:44 CEST] <furq> you probably just want -map 0:s
[15:41:05 CEST] <__raven> furq: map 0:2 i tried already :(
[15:41:07 CEST] <furq> and using -c:s copy probably won't work as hls is most likely using webvtt subtitles
[15:41:20 CEST] <furq> well yeah that's apart from the error
[15:41:46 CEST] <furq> "Consider increasing the value for the 'analyzeduration' and 'probesize' options"
[15:41:56 CEST] <__raven> furq: what would be the best way to get the original text-stream out though?
[15:42:23 CEST] <furq> -map 0:s -c:s copy out.vtt
[15:43:45 CEST] <__raven> 0:s or 0:2?
[15:43:52 CEST] <furq> either
[15:44:06 CEST] <furq> 0:s is all subtitle streams
[15:44:15 CEST] <furq> which in this case is identical to 0:2
[15:54:24 CEST] <__raven> furq: now i get Cannot map stream #0:2 - unsupported type. :(
[16:08:10 CEST] <a1fa> hello, i am trying to capture audio using ffmpeg and alsa, and i am having some issues
[16:08:22 CEST] <a1fa> arecod -l gives me card 3 as acceptable input
[16:09:21 CEST] <a1fa> arecord -l
[16:09:23 CEST] <a1fa> card 3: Headset [Logitech G35 Headset], device 0: USB Audio [USB Audio]
[16:09:33 CEST] <a1fa> [alsa @ 0xe7d4c0] cannot set channel count to 2 (Invalid argument)
[16:09:33 CEST] <a1fa> hw:3: Input/output error
[16:09:38 CEST] <a1fa> this is the ffmpeg error i get
[16:10:26 CEST] <a1fa> ffmep -f alsa -i hw:3,0 test.wav is the command i used
[16:12:06 CEST] <a1fa> https://obsproject.com/mantis/view.php?id=387
[16:12:08 CEST] <a1fa> it could be a bug
[16:14:53 CEST] <a1fa> adding -ac 1 fixes my problem :)
[16:14:54 CEST] <a1fa> lol
[16:16:20 CEST] <a1fa> is it possible to use a playback device as recording hw?
[16:16:44 CEST] <a1fa> without using loop back?
[16:20:05 CEST] <maziar> chrome can not play m3u8
[16:20:22 CEST] <maziar> why chrome can not play m3u8? how should i do ?
[16:23:54 CEST] <Bombo> maziar: try vlc media player
[16:24:52 CEST] <maziar> Bombo i want to play it on web, like youtube
[16:26:56 CEST] <Bombo> maziar: how about firefox?
[16:27:11 CEST] <maziar> Bombo, same
[16:28:26 CEST] <Bombo> maziar: internet exploder?
[16:28:31 CEST] <Bombo> maziar: opera?
[16:31:47 CEST] <maziar> Bombo same error, but in safari its ok
[16:32:49 CEST] <Bombo> maziar: maybe complain to the developers of chrome/firefox then
[16:35:02 CEST] <Snoo20664> it's not an error
[16:35:12 CEST] <Snoo20664> you just need to associate it with something
[16:35:29 CEST] <Snoo20664> or just download it with ffmpeg and play it locally
[16:38:58 CEST] <Bombo> HTTP Live Streaming (also known as HLS) is an HTTP-based media streaming communications protocol implemented by Apple Inc. as part of its QuickTime, Safari, OS X, and iOS software.
[16:39:11 CEST] <Bombo> https://en.wikipedia.org/wiki/HTTP_Live_Streaming
[16:39:42 CEST] <Bombo> maybe thats why it works with safari ;)
[16:39:55 CEST] <maziar> Snoo20664 you mean we can not play m3u8 on chrome ?
[16:41:19 CEST] <Bombo> ffplay seems to be able to play HLS
[16:42:32 CEST] <Bombo> maziar: wikipedia says 'Unsupported on desktop OSes' so chrome on android and iOS works i guess
[16:43:05 CEST] <maziar> Bombo let me check
[16:43:19 CEST] <Bombo> maziar: tried with ffplay?
[16:43:33 CEST] <Bombo> oh you want it in a browser...
[16:43:39 CEST] <maziar> Bombo no, what is this ?
[16:43:51 CEST] <kepstin> in some cases, you might be able to play HLS in browsers by using a javascript player that uses media source extensions, but iirc DASH is generally preferred for that
[16:44:04 CEST] <Bombo> maziar: do you know what ffmpeg is? (this channel name)
[16:44:14 CEST] <kepstin> but if you're targetting desktop browsers, and are encoding stuff in advance, just make an mp4 file
[16:44:23 CEST] <kepstin> (with -movflags faststart)
[16:44:35 CEST] <Bombo> maziar: (it's a player)
[16:45:22 CEST] <maziar> Bombo i know what is ffmpeg
[16:45:46 CEST] <maziar> i want to play m3u8 on web browser with JWPlayer, that i bought
[16:46:46 CEST] <Bombo> jwplayer is in the list
[16:47:04 CEST] <Bombo> 'Adobe Flash, iOS, Android'
[16:47:15 CEST] <kepstin> looks like jwplayer uses a flash fallback for hls, because browsers don't support it natively
[16:47:41 CEST] <kepstin> maziar: I'd suggest reading https://support.jwplayer.com/customer/portal/articles/1430218-using-hls-str… - this isn't an ffmpeg issue at this point.
[16:48:10 CEST] <maziar> kepstin thank you i will read it and notify you
[16:48:22 CEST] <kepstin> they recommend having an mp4 file for desktop browsers
[16:49:47 CEST] <Bombo> maziar: https://en.wikipedia.org/wiki/HTTP_Live_Streaming#Clients
[16:50:17 CEST] <maziar> Bombo i will read it too
[16:50:41 CEST] <maziar> kepstin but website like youtube, don't use m3u8 ?
[16:51:07 CEST] <kepstin> youtube uses a combination of webm files, mp4 files, and DASH with a fancy javascript player.
[16:51:22 CEST] <kepstin> and possibly also HLS for apple devices, i'm not sure
[16:56:32 CEST] <Bombo> maziar: what do you want to do anyway? really stream live? or just play files?
[16:58:31 CEST] <Bombo> https://www.youtube.com/html5
[17:05:53 CEST] <maziar> Bombo play movie in different quality and it can be adaptive bitrate, its mean that quality can be changed by user network speed
[17:09:59 CEST] <xviper33> Hey there, just a quick question I couldn't find an answer to using google. I can use "-disposition:a:0 default" to give an audio track the "default" tag. But how do I remove a "default" tag from a track that already had it set in the input file?
[17:10:31 CEST] <maziar> Bombo i want something like this http://pastebin.com/U5Q92qUq
[17:17:30 CEST] <kepstin> maziar: you're almost certainly going to have to look at coding a javascript player with media source extensions. This is well out of the scope of ffmpeg :/
[17:18:02 CEST] <xviper33> Figured it out by guessing. By using -disposition:a:1 "-default". Kinda weird approach but seems to work
[17:18:23 CEST] <maziar> kepstin i know that, because of that i bought jwplayer
[17:19:44 CEST] <kepstin> jwplayer doesn't appear to support what you want to do.
[17:20:07 CEST] <kepstin> although I could be wrong. That said, you should be asking jwplayer support, not random people in this irc channel :)
[17:22:45 CEST] <maziar> kepstin would you please check these sites : http://www.filimo.com http://www.aparat.com
[17:23:17 CEST] <maziar> kepstin would you please check them and tell me which technology they use ?
[17:27:14 CEST] <explodes> Does the probe buffer for av_probe_input_format get saved for later use in the returned AVInputFormat?
[17:28:44 CEST] <kepstin> maziar: no, please read the sites yourself to find out what technology they use.
[17:38:30 CEST] <maziar> kepstin unfortunately i can not find the technology
[19:02:14 CEST] <fling> fixed with a reboot ^ wtf
[19:05:08 CEST] <fling> It is now only slightly out of sync
[19:05:12 CEST] <fling> I'm now updating ffmpeg
[19:13:20 CEST] <fling> the video is freezing sometimes somehow hmm hmmm
[19:26:34 CEST] <fling> hmm hmm fps is somehow way too low
[19:26:53 CEST] <fling> I'm now downgrading the kernel
[19:27:40 CEST] <pfelt> hey all, i'm playing with some modifications to the movie filter and it looks like ffmpeg is calling the .init function twice. is that normal?
[19:31:12 CEST] <fling> [matroska @ 0x55f75933c580] Non-monotonous DTS in output stream 0:0; previous: 37023, current: -1567689; changing to 37023. This may result in incorrect timestamps in the output file.
[19:46:33 CEST] <maxcnc> hi can i reorder the timeindex of a vob as the recorder did all 4 vidio sequences in one vob sequence and the menue frames in between i need them to be seperate
[19:46:53 CEST] <maxcnc> i stitched all vob together with cat
[19:46:57 CEST] <maxcnc> on linux
[19:47:29 CEST] <maxcnc> but the timindexes jump on etch sequence
[20:36:35 CEST] <blue_misfit101> does -c:a aac have a pure VBR mode or must you specify a bitrate?
[20:36:56 CEST] <blue_misfit101> looking to encode highest possible quality stereo AAC for an intermediate file
[20:39:01 CEST] <benlieb> what's the state of the art for html5 in browsers? do I still need mp4 and ogg?
[20:39:03 CEST] <benlieb> formats
[20:39:26 CEST] <blue_misfit101> only if you care about really old / fringe browsers or operating systems
[20:39:42 CEST] <blue_misfit101> basically everything can play HLS now if you use a decent player like video.js or flowplayer
[20:40:04 CEST] <Mavrik> benlieb, VP8 in WebM container + H.264 in MP4 is usually good.
[20:40:31 CEST] <benlieb> Mavrik: are there browsers that can't play that?
[20:41:02 CEST] <TD-Linux> nope
[20:41:07 CEST] <Mavrik> Hrmf, Lynx? :P
[20:41:16 CEST] <Mavrik> But all major ones should handle those two.
[20:41:36 CEST] <benlieb> last I looked (couple of years) there was one browser that needed ogg
[20:41:40 CEST] <benlieb> can't remember what
[20:41:44 CEST] <blue_misfit101> benlieb, I'd look into ABR encoding (HLS or DASH) if you're not doing it already
[20:41:46 CEST] <Mavrik> benlieb, Firefox
[20:41:51 CEST] <blue_misfit101> you'll get much better user experience with ABR
[20:42:00 CEST] <TD-Linux> http://caniuse.com/#search=H.264%7CVP8
[20:42:02 CEST] <Mavrik> benlieb, and chromium
[20:42:10 CEST] <Mavrik> benlieb, but FF now supports VP8
[20:42:13 CEST] <Mavrik> together with chrome etc.
[20:42:32 CEST] <TD-Linux> in fact if you're patient you can get basically the same support with VP9 instead of VP8
[20:42:44 CEST] <TD-Linux> http://caniuse.com/#search=H.264%7CVP9
[20:42:46 CEST] <Mavrik> Hrmf, did FF enable VP9 yet?
[20:42:52 CEST] <Mavrik> Ah.
[20:42:55 CEST] <benlieb> wow, I feel like your speaking a foreign language. Is there a one-stop-spot to understand all of this?
[20:42:56 CEST] <Mavrik> Certanly go with VP9 then.
[20:43:12 CEST] <Mavrik> benlieb, you're getting links right now?
[20:43:18 CEST] <Mavrik> And any video tutorial on the web? :/
[20:43:21 CEST] <blue_misfit101> benlieb, nope :D you've got a lot of research ahead of you
[20:43:50 CEST] <benlieb> lol, the VP9 is Heckler & Koch's latest handgun and the first striker fired HK since the renowned...
[20:44:18 CEST] <blue_misfit101> lololol
[20:44:52 CEST] <TD-Linux> benlieb, I also forgot to ask, I assume you're talking about video, not just audio
[20:44:58 CEST] <benlieb> yes video
[20:46:22 CEST] <TD-Linux> yeah webm (vp8+vorbis or vp9+opus) has more or less replaced ogg (theora+vorbis)
[20:46:33 CEST] <benlieb> Can you break down what this means a bit more: VP9 in WebM container + H.264 in MP4? Wikipedia says VP9 is a coding format. Isn't that what H.264 is? And mp4?
[20:46:52 CEST] <benlieb> or is that vid / audio
[20:46:59 CEST] <TD-Linux> H.264 is a video format, but mp4 is the container
[20:47:08 CEST] <TD-Linux> mp4 is usually H.264+AAC
[20:47:17 CEST] <blue_misfit> ^
[20:47:20 CEST] <benlieb> and WebM container?
[20:47:30 CEST] <TD-Linux> VP8+vorbis or VP9+opus
[20:47:42 CEST] <benlieb> so what does using both together mean?
[20:47:49 CEST] <TD-Linux> one is video and one is audio
[20:48:14 CEST] <benlieb> Which is which? I need a vid container and an audio container?
[20:48:23 CEST] <benlieb> I thought the vid container contained the audio
[20:48:29 CEST] <TD-Linux> no, just one container, and an audio and video format that fit inside the container
[20:49:01 CEST] <TD-Linux> note that ffmpeg will pick appropriate audio and video codecs for you based on extension
[20:49:09 CEST] <benlieb> right so there seems to be two containers suggested here: VP9 in WebM container + H.264 in MP4
[20:49:16 CEST] <TD-Linux> yup
[20:49:56 CEST] <benlieb> so is this recommendation to have two videos, one in each of these formats: VP9 in WebM container + H.264 in MP4?
[20:50:09 CEST] <TD-Linux> yes
[20:50:51 CEST] <Mavrik> yp
[20:50:58 CEST] <TD-Linux> (your initial thought of ogg + mp4 would have been fine too, just that ogg theora is a bit long in the tooth)
[20:53:19 CEST] <benlieb> I guess my question is can I get by with offering just one video? I run an instructional video website that I started 9 years ago. It mostly just works and I don't touch it much, but I want to make some upgrades, and one important one is offering streaming
[20:53:54 CEST] <benlieb> This page seems to suggest that the only browser that can't handle MPEG-4/H.264 is opera mini http://caniuse.com/#search=H.264%7CVP8
[20:54:37 CEST] <TD-Linux> benlieb, opera mini prerenders web pages on the server and so doesn't support video at all
[20:54:55 CEST] <benlieb> I'm not sure I care about open mini to be honest
[20:55:00 CEST] <benlieb> opera
[20:55:03 CEST] <TD-Linux> you shouldn't care about it
[20:55:23 CEST] <TD-Linux> the one thing that the chart is missing though is that Firefox and I think IE depend on Windows's H.264
[20:55:49 CEST] <TD-Linux> so people on Windows XP or the N versions of Windows won't have it
[20:55:50 CEST] <benlieb> TD-Linux: so FF on mac won't play it?
[20:56:03 CEST] <TD-Linux> bencoh, FF on mac will because it can use mac's H.264 decoder
[20:57:17 CEST] <benlieb> TD-Linux: so the only people that won't be able to play mp4/H264/AAC are older windows users?
[20:57:31 CEST] <TD-Linux> benlieb, or windows N users
[20:58:06 CEST] <benlieb> I'm not clear on what that means exactly
[20:58:12 CEST] <benlieb> thanks for your help by the way
[20:58:15 CEST] <TD-Linux> it's the european version of windows
[20:58:18 CEST] <TD-Linux> np
[20:59:06 CEST] <benlieb> Let's see if I have this right. so firefox users on windows in Europe won't be able to play it.
[20:59:30 CEST] <benlieb> TD-Linux: ^
[21:00:30 CEST] <TD-Linux> yes.
[21:00:43 CEST] <benlieb> oh, I forgot IE users in europ
[21:01:27 CEST] <Mavrik> Also Linux users.
[21:01:32 CEST] <Mavrik> (Some of them.)
[21:01:37 CEST] <benlieb> damn
[21:01:51 CEST] <benlieb> linux can't play mp4?
[21:02:04 CEST] <benlieb> Mavrik: ^
[21:02:11 CEST] <Mavrik> MP4 is a container.
[21:02:17 CEST] <benlieb> H.264
[21:02:21 CEST] <TD-Linux> benlieb, if they haven't installed a copy of lavc
[21:02:32 CEST] <TD-Linux> sorry for all the conditionals and fine rules here...
[21:02:43 CEST] <TD-Linux> I don't know if this is documented anywhere hmm
[21:02:48 CEST] <Mavrik> TD-Linux, all distrost have lavc pretty much, most ship without H.264 decoder ;)
[21:03:24 CEST] <benlieb> I'm going to just assume that if someone is using linux, they know how to get to a video they want to watch
[21:03:37 CEST] <TD-Linux> overall I'd say you get 95% of the way there with mp4 and basically 100% with mp4 and webm
[21:04:26 CEST] <benlieb> I think I can be ok just offering one format then. It's a lot of work and a lot of storage to keep copies in multiple formats.
[21:04:44 CEST] <benlieb> Is AAC the recommended audio encoding?
[21:06:45 CEST] <TD-Linux> for mp4 it's basically your only choice for compatibility
[21:08:13 CEST] <benlieb> TD-Linux: ok, I'll go with mp4 / h.264 / AAC. Thanks for your help!
[21:08:25 CEST] <benlieb> also is H.264 decoder the same as H.264 codec?
[21:08:42 CEST] <benlieb> I think codec = encoder / decoder?
[21:09:17 CEST] <TD-Linux> technically yes but "format" and "codec" tend to be interchanged a lot
[21:10:06 CEST] <Mavrik> Encoder - software that encodes
[21:10:11 CEST] <Mavrik> Decoder - software that decodes
[21:10:27 CEST] <Mavrik> Codec - usually interchangable with format of encoded data
[21:11:14 CEST] <benlieb> anyone have a book recommendation or website that might cover all of this in some detail?
[21:11:32 CEST] <TD-Linux> encoding, or browser support?
[21:12:08 CEST] <benlieb> Audio, Video formats, containers, codecs...
[21:12:50 CEST] <TD-Linux> https://developer.mozilla.org/en-US/docs/Web/Guide/HTML/Using_HTML5_audio_a…
[21:13:13 CEST] <benlieb> I assume ffmpeg can encode into VP9? Where does it get that codec? I installed ffmpeg several years ago, would I have to upgrade ffmpeg, or just install the codec?
[21:14:14 CEST] <kepstin> depends how you got your ffmpeg in the first place. I'd expect most current builds have vp8/9 encoding enabled
[21:14:34 CEST] <benlieb> TD-Linux: that link is just for the HTML side of things. I'm pretty good on that part.
[21:14:38 CEST] <kepstin> at a minimum, you'd need to install a recent libvpx and rebuild ffmpeg.
[21:15:00 CEST] <benlieb> kepstin: is there a flag I can check what codecs I have available/
[21:15:01 CEST] <benlieb> ?
[21:15:10 CEST] <kepstin> ffmpeg -codecs
[21:15:38 CEST] <TD-Linux> you'll want to upgrade to a recent ffmpeg to get the much improved aac encoder as well.
[21:16:11 CEST] <benlieb> looks like I got it: D.V.L. vp9 Google VP9
[21:16:17 CEST] <benlieb> What is D.V.L ?
[21:16:22 CEST] <kepstin> read the labels at the top
[21:16:32 CEST] <benlieb> ah
[21:16:40 CEST] <benlieb> had to scroll way up
[21:16:57 CEST] <benlieb> this one must be evil: DEVIL. mjpeg Motion JPEG
[21:17:03 CEST] <benlieb> lol
[21:17:41 CEST] <benlieb> looks like I'm running ffmpeg version 2.2.1 Copyright (c) 2000-2014
[21:18:59 CEST] <benlieb> ooh now I'm remember the pain of the install back then.
[21:19:40 CEST] <benlieb> I needed freetype, theora, libass, etc
[21:20:28 CEST] <kepstin> what os are you on? you can probably get a prebuilt or packaged ffmpeg...
[21:21:43 CEST] <benlieb> mac
[21:32:05 CEST] <benlieb> ok upgraded to 2.8.5
[21:32:25 CEST] <benlieb> TD-Linux: what is the improvement in the aac encoder?
[21:35:03 CEST] <kepstin> for the built-in aac encoder? the fact that the quality is now actually useful.
[22:47:32 CEST] <jacobwg> Is this the right place to get help with a MP4 media encoding issue (the file was transcoded with ffmpeg, though the question is only tangentially related to ffmpeg)? I'm experiencing issues with QuickTime (and by extension, my Apple TV) playing multiple audio channels at the same time, though only under certain conditions (I have one file that works and
[22:47:32 CEST] <jacobwg> one that doesn't, with seemingly unrelated differences) and am looking for a second pair of eyes to look at what's going on.
[23:20:18 CEST] <pfelt1> hey all, i've got something of a silly question. on a -filter_complex i'm playing with a filter that works as a passthrough on video. if my input has audio and video, and i do a [0:v]filter; in my filter, how do i join it back up with the audio for ultimate output from the filtering?
[23:21:31 CEST] <durandal_1707> same with audio
[23:21:40 CEST] <durandal_1707> 0:a
[23:22:46 CEST] <pfelt1> yeah, i know that, but, for example: with the last video output be automatically added to the last audio output, or do i need to add a filter of some kind to say "[inputVideo][inputAudio]jointogether" as the last thing in my filter
[23:23:23 CEST] <kepstin> pfelt1: the audio and video are always separate until the muxer
[23:23:27 CEST] <pfelt1> that should read: &. for example: will the .....
[23:23:49 CEST] <pfelt1> ok. so i don't need anythign special at the end to join them back up. the muxer will do it for me
[23:24:06 CEST] <kepstin> pfelt1: at worst, you might need to use a separate -map option for each of the video and audio
[23:24:24 CEST] <kepstin> if your filter chain has a named audio output and named video output, for example.
[23:25:04 CEST] <pfelt1> 'movie=filename=/tmp/tmp1.fifo:streams=dv+da:pts=2 [str]; [str:v]zmq'
[23:25:21 CEST] <pfelt1> is what i've got right now, and it's failing, but i'm sure it's because i've got somethign specified wrong
[23:25:49 CEST] <kepstin> why are you using 'movie' in the filter_complex string rather than using an input with -i ?
[23:26:09 CEST] <pfelt1> i'm actually adding some functionality to it to allow me to restart the stream
[23:26:16 CEST] <pfelt1> (via a process_command)
[23:26:39 CEST] <pfelt1> and that's what zmq is for
[23:26:52 CEST] <kepstin> right now, the 'movie' filter in the filter chain only has a single output pad '[str]' which will have the video
[23:27:14 CEST] <pfelt1> it should have two. one for the dv and one for the da
[23:27:24 CEST] <kepstin> but you only gave it one
[23:27:46 CEST] <kepstin> i think...
[23:27:59 CEST] <pfelt1> oh snap
[23:28:05 CEST] <pfelt1> i just saw the last example
[23:28:10 CEST] <pfelt1> i need to name both on output
[23:28:59 CEST] <kepstin> yeah, and to include a named output in the resulting file, add a -map '[outputname]'
[23:29:20 CEST] <kepstin> (if you do that, you should name all outputs, and add a -map for each of them)
[23:29:30 CEST] <pfelt1> ugh. ok
[23:34:32 CEST] <kepstin> but yeah, a single filter pad will only ever have one stream on it; audio and video always have to go through different pads. In this case, the :v on [str:v] is redundant at best, I'm not sure how that's handled.
[23:35:02 CEST] <kepstin> probably best to use just [str] to avoid confusion (or rename it, to make it obvious that it's the video)
[23:35:58 CEST] <johnnny22-afk> what's the biggest PTS that a mpeg-ts can have ?
[23:36:03 CEST] <pfelt1> it'll be interesting to see what happens when i send the process_command to switch streams
[23:36:33 CEST] <pfelt1> johnnny22-afk: i believe it's a 64bit int
[23:36:40 CEST] Action: pfelt1 guesses 2^64
[23:37:29 CEST] <johnnny22-afk> I'm guessing that makes a big pts, more than 48h
[23:37:31 CEST] <pfelt1> that's not right
[23:37:39 CEST] <pfelt1> A time stamp is a 33 bit number that is a sample of a counter driven by a 90 kHz clock
[23:37:48 CEST] <pfelt1> so 2^33
[23:38:10 CEST] <johnnny22-afk> which makes 8589934592
[23:38:27 CEST] <johnnny22-afk> what are those, seconds ? frames ?
[23:39:21 CEST] <kepstin> the clock is 90kHz
[23:39:40 CEST] <pfelt1> yeah. khz
[23:39:41 CEST] <kepstin> so apparently it's designed to be a little more than 24H
[23:40:19 CEST] <johnnny22-afk> humm makes more sense from what I've heard, but I wasn't sure of the intrisic details :)
[23:40:33 CEST] <johnnny22-afk> and then the mpeg-ts pts roll-over back to 0:00 I assume.
[23:40:43 CEST] <kepstin> yeah, looks like they just picked the smallest number of bits that can fit 24H
[23:40:56 CEST] <johnnny22-afk> broadcasting people.. eh !:P
[23:41:00 CEST] <kepstin> i'd expect in most cases on a continous stream it would count to 24H then roll over
[23:41:29 CEST] <johnnny22-afk> and how do players or ffmpeg handle playing such a stream that rolls over, doesn't that cause issues when seeking around ?
[23:41:36 CEST] <johnnny22-afk> or ffplay
[23:41:37 CEST] <kepstin> but a lot of broadcast streams aren't actually continuous, of course, you have commercial splicing etc.
[23:41:50 CEST] <kepstin> mpeg-ts isn't designed to be seekable.
[23:42:06 CEST] <johnnny22-afk> even when on disk ?
[23:42:33 CEST] Action: kepstin wonders how the bdmv format does seeking, actually
[23:43:09 CEST] <kepstin> probably has some extra constraints on the mpeg ts on blurays so you have to have monotonic pts or something
[23:45:00 CEST] <kepstin> if you have an mpeg-ts file with timestamp discontinuities, without an external index of some sort, I don't think you can seek much better than picking a file position based on average bitrate and resyncing :/
[23:45:30 CEST] <johnnny22-afk> m'ok, humm makes me think
[23:45:56 CEST] <kepstin> it is designed as a streaming format, after all.
[23:46:04 CEST] <johnnny22-afk> maybe I should change the fileformat, while keeping the v and a codecs intact :)
[23:47:55 CEST] <johnnny22-afk> I'd like a fileformat that doesn't have a header or a tail, that can be truncated anywhere (or almost) without making the fileformat unplayable. Ideally a fileformat that can take more than 24 hours.
[23:48:00 CEST] <johnnny22-afk> and is seekable :P
[23:48:23 CEST] <johnnny22-afk> Obvioulsly, I could use ffmpeg to regenerate the pts, so as to not create discontinuities.
[23:48:34 CEST] <kepstin> an interesting note on DVDs is that seek points are actually handled by having external data files storing *sector* offsets within specific VOB files to mark chapters, etc.
[23:49:25 CEST] <johnnny22-afk> that's probably not obvious to do on the fly as you're saving the stream to disk, or maybe it is.. but players probably wouldn't like this.
[23:50:22 CEST] <johnnny22-afk> I could also just rotate to another file after 24h, or make sure to remove the head of the file so that not more than 24h exists on the file.
[23:50:37 CEST] <kepstin> yeah, segmenting the file is probably your best option
[23:51:15 CEST] <johnnny22-afk> for 24h rollover sure. but not within that 24h stretch (i think).. cuz players really hate dealing with multiple files and seeking in them.
[23:51:36 CEST] <kepstin> you could probably even use the HLS muxer to do it, and then you'd have an m3u8 playlist file that HLS-supporting players can use - it even handles seeking across files.
[23:51:53 CEST] <johnnny22-afk> currently seeking in the mpegts file I'm generating seems smooth... but I do have my worry about pts's existing twice, or what happens when it roll-over back to 0:00
[23:52:28 CEST] <johnnny22-afk> yeah, but I've had bad experience seeking on a m3u8 that doesn't contain a end-list tag.
[23:53:02 CEST] <johnnny22-afk> merging those ts files and seeking within that mpegts file seems just so much smoother. By far.
[23:54:08 CEST] <johnnny22-afk> I could seek by segment in a HLS, but finer than that just exhauses the player.
[23:59:41 CEST] <kepstin> if you just need an arbitrarily long file that is readable and seekable as it's being encoded, mkv might work. But you can't just chop the head off an mkv file like you can with mpeg-ts.
[00:00:00 CEST] --- Thu Apr 21 2016
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