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September 2014

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ffmpeg-devel.log.20140929
by burek 30 Sep '14

30 Sep '14

[00:29] <cone-228> ffmpeg.git 03Michael Niedermayer 07master:755f7929c6aa: avcodec/mpegvideo_enc: Only enforce rc_max_available_vbv_use on first encoding attempt [00:29] <cone-228> ffmpeg.git 03Michael Niedermayer 07master:3f5095f213a0: avformat/hlsenc: fix outter typo [11:45] <ubitux> BBB: do you know if there are security concerns about chromium linking to a system version of ffmpeg? [11:45] <ubitux> like, can the probing be trigger to exploit a crash in a random codec or format [11:46] <ubitux> or something along these lines? [11:46] <ubitux> (because of the system ffmpeg being full featured instead of a very small subset) [11:48] <wm4> I'm not BBB or google, but obviously the answer is yes [11:48] <wm4> in addition to ABI problems [11:48] <wm4> the latter makes system ffmpeg a nightmare for application devs anyway [11:48] <ubitux> well, maybe they do the probing differently [11:49] <ubitux> like, forcing the format according to mime types or whatever [11:50] <Daemon404> ubitux, why do you ask? [11:50] <Daemon404> because chrome will always bundle their own .... always [11:51] <ubitux> there is a -Duse_system_ffmpeg=0/1 [11:51] <ubitux> my distro sets it to 0, like probably many/all of them [11:51] <Daemon404> do you mean *Chrome* [11:51] <Daemon404> or chromium [11:51] <Daemon404> (sorry, i wasnt clear) [11:51] <ubitux> chromium [11:51] <Daemon404> oh ok [11:51] <Daemon404> that makes more sense. [11:52] <ubitux> isn't chrome just the windows version of chromium? [11:52] <nevcairiel> chrome is their binary distribution, it also exists for linux [11:52] <ubitux> ok [11:53] <Daemon404> ubitux, main difference is bundle flash [11:53] <Daemon404> bundled* [11:53] <wm4> at least it makes no sense for chrome to enable obscure game format decoders and such, whose only value is probably adding security issues [11:53] <ubitux> wm4: right, but the question is more about whether the code is triggerable or not [11:54] <ubitux> if they do force codec and format, then it might be fine [11:55] <Daemon404> ubitux, i dont see why they wouldnt though [11:55] <Daemon404> it's such a small subset that it's actually preferable [11:56] <ubitux> wouldn't what? [11:56] <ubitux> force or probe? [11:56] <Daemon404> hard set / enable vp8, vp9, h264, aac [11:56] <Daemon404> during init [11:56] <Daemon404> i mean they HAVE to have their own probe stuff, for their hw playback [11:57] <Daemon404> which is what most users get [11:57] <ubitux> ah right, you can indeed enable only some codecs/format, forgot that [11:57] <Daemon404> yeah [11:57] <Daemon404> just dotn do av_register_all() etc [11:57] <ubitux> yeah right ok [11:57] <Daemon404> and manually register the few you need [11:57] <ubitux> then it should be fine :) [11:57] <wm4> huh? since when can you register individual codecs [11:57] <ubitux> avcodec_register(...)? [11:58] <wm4> ubitux: and how do you get the codec ptr? [11:58] <ubitux> avcodec_find_decoder()? [11:58] <ubitux> avcodec_register(avcodec_find_decoder("h264")) ? [11:58] <wm4> but that finds only registered codecs [11:59] <ubitux> ah [12:00] <Daemon404> wow [12:00] <Daemon404> yeah [12:00] <Daemon404> what the hell? [12:01] <Daemon404> wow that's pretty bad... now it bothers me [12:01] <wm4> avcodec_register is basically just useless global state [12:02] <wm4> someone in Libav actually made plans how to remove that, but never posted a patch [12:02] <Daemon404> it seems that everything is always init'd anyway [12:02] <Daemon404> so idneed usless [12:02] <Daemon404> mind you for chrome, they only enable what they need in configure [12:02] <wm4> yeah, that's the right way for this stuff [12:04] <Daemon404> wm4, anyway, i dont think chrome uses probing [12:04] <Daemon404> probably just find_decoder(AV_CODEC_ID_H264) [12:04] <wm4> they probably have their own demuxers and still use libvpx too... [12:04] <Daemon404> libvpx is still faster for vp9 decoding on their biggest platform [12:05] <wm4> arm? [12:05] <wm4> i386? [12:05] <Daemon404> 32bit windows [12:05] <Daemon404> like most of the world [12:05] <Daemon404> that isnt in a bubble [12:06] <wm4> nah, 64 bit windows is getting pretty popular [12:06] <wm4> but 32 bit might still be easier to distribute [12:06] <wm4> also the future is mobile [12:06] <Daemon404> 64 bit chrome is sitll not in the release channe [12:06] <Daemon404> its in beta though [12:06] <wm4> (unfortunately) [12:14] <rcombs> wm4: Chrome uses libavformat for demuxing, can confirm [12:14] <wm4> rcombs: even webm? [12:14] <rcombs> yup [12:15] <wm4> libavformat doesn't do DASH though [12:15] <rcombs> you can throw an MKV with H.264 at it and it'll chug along happily, but it'll fail if the required probe time for the file is longer than 5 seconds [12:15] <rcombs> neither does Chrome, natively [12:15] <rcombs> DASH in Chrome involves MSE and a lot of JS [12:15] <ubitux> so it does actually call the probing code from ffmpeg? [12:15] <wm4> then how are they doing youtube? [12:16] <wm4> though from what I hear, youtube uses a subset of dash... just separate audio and video tracks [12:16] <rcombs> ubitux: yup [12:17] <rcombs> wm4: YouTube on CC, for example, uses MSE and segments [12:17] <ubitux> so, if it links to a system ffmpeg, will it be able to play like... nut/ffv1 videos? [12:17] <rcombs> YouTube on desktop just uses separate audio and video tracks, and probably loads them separately and plays at the same time [12:18] <rcombs> ubitux: I'm not sure if it ever links to system ffmpeg; I think it always bundles its own (libffmpeg_sumo) [12:18] <ubitux> rcombs: there is a -Duse_system_ffmpeg=0/1 [12:18] <rcombs> oh, well then [12:18] <kierank> ubitux: no because they ship their own ffmpeg [12:18] <kierank> they aren't stupid [12:18] <ubitux> i know right [12:18] <kierank> their own ffmpeg only has stuff they need [12:19] <kierank> 11:17 AM <"ubitux> so, if it links to a system ffmpeg, will it be able to play like... nut/ffv1 videos? [12:19] <kierank> so no [12:19] <wm4> even open source applications ship their own ffmpegs [12:19] Action: kierank does [12:19] <rcombs> possibly, but I haven't tested it; they might special-case MKV (and possibly others) and otherwise not call into lavf at all [12:19] <ubitux> kierank: -Duse_system_ffmpeg=1 will link against the system ffmpeg i suppose, which is not a subset [12:19] <kierank> ah [12:20] <ubitux> and i'm wondering about the impact on security it has (assuming it works in the first place) [13:04] <cone-810> ffmpeg.git 03Michael Niedermayer 07master:8ba694548782: avcodec/utils: Fix off by 1 error causing unneeded allocation in ff_fast_malloc() [15:19] <wm4> why does the libavformat mpegts demuxer seek around in the stream [15:19] <wm4> isn't mpegts designed to not need that [15:19] <nevcairiel> it should only do that on opening to determine the duration, or when it loses sync [15:20] <wm4> seems to be mostly for resync [15:23] <cone-810> ffmpeg.git 03Michael Niedermayer 07master:cf32181b7011: avcodec/put_bits: Add rebase_put_bits() [16:10] <rcombs> so, free SSL [16:21] <BtbN> "free" "SSL" [17:21] <ubitux> [] find ass headers without "Format:" line in styles, but still defining some styles [17:22] <ubitux> of course, ffmpeg doesn't like it [17:22] <wm4> ubitux: yes [17:22] <wm4> vsfilter doesn't read the format lines at all [17:22] <ubitux> i wonder why we do [17:22] <wm4> libass also fails in this case [17:23] <wm4> there's no reaosn to [17:23] <ubitux> not really [17:23] <wm4> just makes the code more complicated [17:23] <ubitux> it seems to like it well [17:23] <wm4> unfortunately, ffmpeg _still_ outputs ass that assumes you can change the format lines [17:23] <ubitux> :D [17:23] <ubitux> i know i know :) [17:23] <ubitux> i'm working on it [17:23] <wm4> libass falls back to default ass/ssa format line if it fails [17:23] <ubitux> right, but ffmpeg doesn't fallback [17:24] <ubitux> it justs... return -1 ;) [17:24] <wm4> s/fails/is missing/ [17:24] <nevcairiel> the damn thing has a header indicating which fields it has, why would you not assume! [17:24] <wm4> ubitux: if you want to do it "correctly", you have to read vsfilter sources [17:24] <ubitux> nevcairiel: Format: is an alias for Comment:, but be careful because Comment: actually is a timed event [17:24] <ubitux> so Format: is the real comment field [17:25] <ubitux> wm4: yeah probably [17:26] <ubitux> anyway, since i'm going to change the decoded text form, it will fix that at the same time [17:26] <wm4> e.g. vsfilter ignores even sections [17:26] <wm4> it just takes whatever headers it can find [17:26] <wm4> and I forgot how it distinguishes ssa and ass [17:27] <ubitux> V4+ Styles vs V4 Styles [17:27] <ubitux> probably [17:28] <wm4> possibly [19:28] <ubitux> today i found a -lowres user [19:29] <wm4> ubitux: me too [19:29] <ubitux> really? [19:30] <wm4> a mplayer user who tried to play something on his underpowered device with mpv [19:30] <wm4> not sure if he actually used it [19:30] Action: wm4 sees deprecated API warnings when compiling ffmpeg.c... wat [19:30] <ubitux> his nickname started with 'f'? [19:32] <ubitux> wm4: yes we wait for our users to test the api before we do [19:33] <wm4> no, not 'f' [19:33] <ubitux> ok, then it's indeed 2 different users [19:33] <wm4> fascinating [19:33] <nevcairiel> unless its the same user under different names [19:35] <ubitux> the guy was complaining that he couldn't play "hd" videos on his eeepc anymore because libav dropped it (not more mplayer -lavdopts lowres=1) [19:42] <wm4> h264: The maximum value for lowres supported by the decoder is 0 [19:42] <wm4> when using lowres=1 [19:44] <ubitux> try a mpeg4 video [20:33] <cone-810> ffmpeg.git 03Timothy B. Terriberry 07master:a05f5052fef3: sdp: Make opus declaration conform to the spec [20:33] <cone-810> ffmpeg.git 03Michael Niedermayer 07master:77ab7407c272: Merge commit 'a05f5052fef3b3743fab7846da12861d8a8098ec' [20:38] <Compn> yeah i think you can only lowres some strange h264 (maybe no slices? ) but mpeg4 should still work [20:38] <Compn> and mpeg2 of course [20:39] <Compn> i'm surprised some eeepc was using it :) [20:39] <Compn> or underpowered device* [21:02] <cone-810> ffmpeg.git 03Luca Barbato 07master:e3a00acde05c: hevc: Initialize mergecand_list to 0 [21:02] <cone-810> ffmpeg.git 03Michael Niedermayer 07master:46807921f768: Merge commit 'e3a00acde05c925617dc19b5373969d864bf8414' [21:10] <cone-810> ffmpeg.git 03Michael Niedermayer 07master:2cd7c99498b1: h264: reset ret to avoid propagating minor failures [21:10] <cone-810> ffmpeg.git 03Michael Niedermayer 07master:56c47364c3b5: Merge commit '2cd7c99498b1bcb450c328459cf85a686ce83456' [21:35] <someone-noone> hello. I capture my mic & webcam with avfoundation framework, but resulting file is out of sync. Here's what I run: http://pastebin.com/EZRRCXST [21:35] <someone-noone> I think it's due to the fact that my webcam starts not immediately as mic and that's why video little bit late [21:36] <someone-noone> I looked at avfoundation.m and saw that first pts values is set in read_packet() , before devices are opened [21:36] <someone-noone> so pts values should be correct, I can't figure out why it's happening, can someone help? [21:44] <cone-810> ffmpeg.git 03Justin Ruggles 07master:19133e96d30e: lavf: fix memleaks in avformat_find_stream_info() [21:44] <cone-810> ffmpeg.git 03Michael Niedermayer 07master:a2d5f6b9dbcc: Merge commit '19133e96d30e3f80dbae236ef081aedef419a6bf' [21:45] <michaelni> someone-noone, probably best you mail thilo or open a ticket on trac [21:46] <someone-noone> michaelni, thanks [22:08] <cone-810> ffmpeg.git 03Michael Niedermayer 07master:1441641d786a: avcodec/mpegvideo_enc: Allocate only a small buffer and reallocate as needed [22:27] <akira4> Hi! I'm new here. I wanted to apply for OPW under ffmpeg. Could someone guide me about how to start? [22:28] <hawken> Hi, was wondering if there were plans for mvc support in ffmpeg. I'd like to participate but I don't know where to start [22:29] <llogan> akira4: have you seen this link? https://trac.ffmpeg.org/wiki/SponsoringPrograms/OPW/2014-12 [22:32] <akira4> llogan, I did check the page but couldn't figure out where exactly to start [22:33] <llogan> do any of the listed projects interest you? do you have a project idea you would like to work on? [22:35] <akira4> llogan, I am interested in the subtitles project. [22:35] <llogan> i think ubitux can help you with that. [22:36] <ubitux> oh? [22:36] <llogan> you're listed as a mentor [22:36] <ubitux> yeah right [22:36] <ubitux> but i wasn't briefed on the opw process [22:36] <ubitux> so yeah sure i can answer subtitles questions, but not so much about opw [22:37] <akira4> I see. Thanks [22:37] <akira4> Also how do I start with the qualification task? [22:38] <ubitux> mmh let me what i put there [22:38] <ubitux> "write one subtitles demuxer and decoder (for example support for Spruce subtitles format). This is in order to make sure the subtitles chain is understood." mmh alright [22:39] <ubitux> let me see if i can find something that looks like a specifications [22:39] <ubitux> http://documentation.apple.com/en/dvdstudiopro/usermanual/index.html#chapte… [22:40] <ubitux> alright, this looks like a good starting point [22:40] <ubitux> not sure if we have samples around [22:41] <akira4> cool. Thanks [22:41] <ubitux> http://www.eso.org/~lchriste/trans/eyes/subtitles/soundtrack.stl [22:41] <ubitux> alright, here is one [22:42] <ubitux> akira4: are you familiar with libavformat/libavcodec and the demux/decode process or not at all? [22:42] Action: rcombs would love to see subtitles in lavfi, and would probably help with it [22:42] <akira4> ubitux, No I'm not actually. [22:43] <ubitux> rcombs: yes that's the last step, but there are a few things before, to give me enough time to redesign the api so it's possible :P [22:43] <ubitux> akira4: alright, so... [22:43] <ubitux> akira4: basically, see this first: http://ffmpeg.org/ffmpeg.html#Detailed-description [22:43] <ubitux> (just the ascii graph and the explanation below) [22:44] <ubitux> poke me when you're done, i'll explain how subtitles fit into this [22:44] <akira4> I see. [22:44] <akira4> cool [22:44] <akira4> I'll do that [22:44] Action: hawken looks around [22:45] <hawken> So, I don't have anything to do with OPW being a guy but I still want to see mvc happen :P Can I take part in that? [22:45] <ubitux> hawken: sorry, i don't know anything about mvc, you should probably ask michaelni [22:45] <hawken> Okay.. [22:45] <hawken> michaelni: ^ ? [22:45] <ubitux> iirc several people tried to approach the mvc problem over the time [22:45] <hawken> I'd like as much information as I can get [22:45] <ubitux> we had mixed results [22:46] <hawken> I tried making something from scratch to analyze the files but I feel I only scratched the surface [22:46] <CoRoNe> Good afternoon, [22:46] <ubitux> i think at some point one guy sent the worst patch dump ever [22:46] <ubitux> hawken: maybe try "site:ffmpeg.org ffmpeg-devel mvc" on google [22:46] <hawken> https://github.com/hawken93/mvc-decoder <-- so I didn't get to actual MVC but I think I made a nice m2ts debugger [22:46] Action: hfvjkab is reminded about about people trying to add cuda to x264 [22:47] <hfvjkab> WTFis up with my nick? [22:47] <hawken> Ah I saw that.. [22:47] <hawken> so that draw me towards making my own stuff since I totally don't know where to start in a project as big as ffmpeg :P [22:47] <hawken> I'm a total newb lol :3 [22:48] <ubitux> https://encrypted.google.com/search?hl=en&q=site%3Affmpeg.org%20ffmpeg-deve… [22:48] <hawken> mhm [22:48] <hawken> Okay I'll be going through all the results now :P [22:49] <ubitux> akira4: i realize that description doesn't actually explain much; can you start building ffmpeg and get maybe a .srt file somewhere? [22:49] <CoRoNe> Good afternoon, [22:49] <CoRoNe> Possible bug: whenever I'm trying to stream opus over rtp, ffmpeg by default prints: [22:49] <CoRoNe> SPD: [22:49] <CoRoNe> ... [22:49] <CoRoNe> a=rtpmap:97 opus/48000 [22:49] <CoRoNe> ...causing the stream to be mono. If I manually append '/2' to 'opus/48000', the stream is stereo like the source. [22:49] <CoRoNe> I there anything you developers can do about it? [22:50] <akira4> ubitux, Alright I'll try doing that. [22:50] <CoRoNe> libavformat / sdp.c: line 606 [22:53] <llogan> CoRoNe: submitting a patch to ffmpeg-devel will allow us to see it in full context and allow others who are not in IRC to possibly comment. [22:54] <CoRoNe> I'm sorry, I'm not familiar with that, nor am I a developer. Would creating a thread on http://ffmpeg.zeranoe.com/builds/ suffice too? [22:55] <ubitux> CoRoNe: http://ffmpeg.org/bugreports.html or http://ffmpeg.org/developer.html#Contributing [22:55] <ubitux> depending on what you want to do [22:57] <CoRoNe> oh yes, there's a bugtracker of course. I'll create a new ticket. Thank you [23:17] <akira4> ubitux, I'm done with the building of the source code part. Should I read the documentation that you provided? [23:18] <ubitux> akira4: it will just take you 2 min [23:18] <ubitux> akira4: do you have a .srt file at hand? [23:18] <akira4> ubitux, yep I have many .srt files with me [23:19] <ubitux> ok; so do you have a ffprobe tool built in the source directory? [23:20] <akira4> ubitux, yes I do. [23:21] <ubitux> try running ./ffprobe -show_packets -show_data foo.srt|less [23:21] <ubitux> this will show you the demuxing process [23:21] <akira4> I see. [23:21] <ubitux> basically, the srt demuxer (in libavformat/srtdec.c) will fill "packets" [23:22] <ubitux> a packet is a simple structure, which has a few fields, notably pts, data and size [23:22] <ubitux> and duration [23:22] <ubitux> (and a few other things you can see here) [23:23] <ubitux> the data is basically supposed to be kind of opaque [23:23] <ubitux> in the case of the srt, you can see that it contains basically the text for each event [23:23] <akira4> yes [23:23] <ubitux> but it can have markup, right [23:23] <ubitux> typically, it's the event copied verbatim [23:23] <ubitux> so with <font ...> and stuff like that [23:24] <ubitux> other subtitles demuxer will output similar packets, with their markups as well [23:24] <akira4> I see. [23:24] <ubitux> for microdvd typically, you'll get stuff like ${c:...} [23:24] <ubitux> and same for every other formats [23:24] <ubitux> anyway [23:25] <ubitux> these packets, you can send them directly to a muxer [23:25] <ubitux> for instance with ffmpeg, you can do ffmpeg -i in.srt -c copy out.srt, and only the demuxer and muxer will be in the chain [23:26] <ubitux> the demuxer will output timed packets, and the muxer will re-create a file by printing timestamps and the payload (data) [23:26] <ubitux> similarly, you can do ffmpeg -i in.srt -c copy out.mkv [23:26] <akira4> oh [23:26] <akira4> wait so [23:26] <akira4> if i'm getting this right [23:26] <akira4> we're basically taking a .srt file with markup [23:27] <akira4> and creating a file with timed events corresponding to the text? [23:27] <ubitux> you should get the exact same file at the end [23:28] <ubitux> can you open libavformat/srtenc.c ? [23:28] <ubitux> the srt_write_packet() is the main callback of the muxer [23:29] <ubitux> it takes a packet with a pts and duration, and print the string "00:01:02:03 --> 04:05:..." [23:29] <ubitux> and the payload [23:29] <akira4> okay. [23:30] <ubitux> the idea is that some containers (or formats) accepts packets of different known tags [23:30] <ubitux> so your srt demuxer is outputing packets, with the codec "subrip" (that's the name of the markup) [23:30] <ubitux> several muxers can take these packets [23:30] <ubitux> the srt muxer is obviously one, but the matroska (mkv) muxer also accepts them [23:30] <ubitux> it means the muxer know how to store these packets [23:31] <akira4> I see. [23:31] <ubitux> in the case of the srt muxer, it will create a new .srt file with just the timestamps printed as is, and the text [23:31] <ubitux> and matroska has its own way of storing the timestamps [23:31] <ubitux> so you'll have something like 32-bits for a timestamps, it will be stored as binary [23:32] <ubitux> (not sure if that's exactly that, but you get the point) [23:32] <akira4> yeah [23:32] <ubitux> so, is that fine with you so far? [23:32] <akira4> one thing [23:32] <ubitux> i'm going to go on the decoding process now [23:32] <ubitux> ok [23:32] <akira4> so the srt file that was created out.srt [23:33] <akira4> is in some ways different from in.srt ? [23:33] <ubitux> ideally, it should be the same [23:33] <ubitux> in practice you might have slight differences [23:33] <ubitux> let me thing of an example.. [23:34] <ubitux> right, imagine you have a lot of empty lines at the beginning of in.srt [23:34] <ubitux> the srt demuxer will ignore them [23:34] <ubitux> and just outputs packets loosing that "information" [23:34] <akira4> okay [23:34] <ubitux> similarly, you could imaging a in.srt with timestamps written in a weird manner [23:35] <ubitux> like, i don't know, if the timestamps are written like 00001:02:03.04 --> ... [23:35] <ubitux> when reading the timestamps and storing it in AVPacket.pts, the demuxer will loose that weird "0" padding [23:36] <ubitux> only the timestamps value itself is kept [23:36] <akira4> and that would lead to losing of data if the .srt file is opened by a container that isn't compatible with it? [23:36] <ubitux> the muxer will probably print 01:02:03.04 --> ... [23:36] <ubitux> you can't actually send the packet to a muxer that doesn't support it [23:37] <ubitux> if the muxer doesn't accept subrip packets, you'll have to convert them, that's the nextttstep [23:37] <ubitux> (convert them from one markup to another) [23:37] <ubitux> this is basically the same as audio and video [23:37] <ubitux> if you have a mkv with h264 in it, and you want to put that h264 into ogg, you can't [23:37] <ubitux> because ogg doesn't accept h264 packets, and you'll have to convert these packets [23:38] <ubitux> OTOH, you can demux h264 packets from a mkv file, and just remux them into a mp4 file [23:38] <ubitux> because both mkv and mp4 accept h264 packets [23:40] <akira4> hold on. Let me read the whole thing. it happened too fast. [23:40] <ubitux> right, sorry [23:41] <akira4> so the whole idea is that if there is a packet that a muxer doesnt support we convert it? [23:41] <ubitux> yes [23:41] <akira4> and the packets can be different if they have different tags? [23:41] <ubitux> that's the decoding/encoding process i was going to explain [23:41] <akira4> okay [23:42] <ubitux> what do you mean by different packets and different tags? [23:42] <ubitux> a subrip packet will be different from a microdvd packet yes [23:42] <akira4> I'm not sure what tags mean [23:42] <akira4> I was gonna ask that [23:42] <ubitux> i don't remember talking about tags, but you did :p [23:43] <akira4> sorry. I think I got confused [23:43] <ubitux> i'm going to give you more example before i continue [23:43] <akira4> cool. thanks :) [23:44] <ubitux> a .sub file (microdvd) contains lines like this: "{1400}{1500}hello world" [23:44] <ubitux> 1400 is the starting frame, and 1500 is the ending one; you can consider them as timestamps for now [23:44] <akira4> ok [23:44] <ubitux> now microdvd also has a markup system [23:45] <ubitux> it looks like this typically: "{1400}{1500}{c:$ff0000}hello world" [23:45] <ubitux> this will make the text in red [23:45] <ubitux> anyway, the demuxer has no knowlead of the markup [23:45] <ubitux> and in this case, it will output a packet that looks like this: [23:46] <ubitux> AVPacket { pts=1400, duration=100, data="{c:$ff0000}hello world" } [23:46] <ubitux> (it's a C-struct, right?) [23:46] <akira4> yep [23:46] <ubitux> if you have a .srt file, it's different [23:46] <ubitux> you will probably get something like this: [23:47] <akira4> I see. [23:47] <ubitux> AVPacket { pts=56, duration=100, data="<font color=red>hello world</font>" } [23:47] <ubitux> and so, as you guess, the srt muxer can not accept the microdvd packets [23:47] <ubitux> and in the same way the microdvd can not accept the srt packets [23:48] <ubitux> otherwise, you would end up with files like with this: [23:48] <ubitux> {1400}{1500}<font color=red>hello world</font> [23:48] <ubitux> and this is an invalid file [23:48] <akira4> hmm. [23:48] <ubitux> that's why the microdvd muxer only accept microdvd packets [23:48] <ubitux> and the srt muxer only accepts srt packets [23:49] <ubitux> for example, matroska accepts the srt packets, but not the microdvd ones [23:49] <ubitux> because that's how it was designed, it only accepts the srt markup [23:49] <ubitux> did i lost you again? [23:49] <akira4> nope [23:49] <akira4> I got everything :) [23:50] <ubitux> cool [23:50] <ubitux> should i move on then? [23:50] <akira4> yes [23:50] <ubitux> alright, now it's a bit tricky [23:50] <ubitux> i'm going to make a comparison with how audio and video are handled [23:51] <akira4> okay [23:51] <wm4> oh, someone is going to help out with subtitles stuff... that's nice [23:51] <ubitux> if you have h264 packets, the data is the compressed data that only the decoder can translate to images [23:51] <ubitux> every video decoders (they are in libavcodec/) only understands one kind of packets [23:51] <ubitux> they do output "raw" frames [23:52] <akira4> I see. [23:52] <ubitux> there are various different form of "raw" but all of them are generics [23:52] <akira4> what exactly are frames? [23:52] <ubitux> typically, some decoders will output RGB [23:52] <ubitux> some other will output YUV, but that's all pure raw that can be piped to image processing code [23:52] <wm4> please note that this separation is sometimes a bit non-sensical with subtitles [23:52] <ubitux> wm4: wait wait ;) [23:53] <wm4> because subtitles are complicated [23:53] <wm4> ok ok [23:53] <ubitux> wm4: i'm trying to make things simple first :P [23:53] <ubitux> akira4: when talking about frames, i'm talking about the AVFrame structure [23:53] <ubitux> it contains a lot of information about the decoded "frame" or image [23:54] <ubitux> (or slice of sound in the case of audio) [23:54] <akira4> Oh. I see. [23:54] <ubitux> it's basically the exploitable data [23:54] <ubitux> a decoder task is to transform a AVPacket into a AVFrame (in the case of audio/video) [23:54] <ubitux> the AVPacket is the opaque form that only demuxer and muxers understand [23:54] <ubitux> and the AVFrame is the decoded form, which is usable by everyone [23:55] <akira4> ok [23:55] <ubitux> so if you want to display a frame in your video player, you have to decode it, to get the picture itself, and blend it to the screen or whatever [23:55] <wm4> it's worth noting that a packet (AVPacket) is basically always a byte blob [23:56] <ubitux> yes, the data is supposely opaque, it just carries timing information and says what the data blob type is [23:56] <wm4> so it's basically a byte array plus timestamps (and, very rarely, some "side" data) [23:56] <akira4> ok. [23:56] <ubitux> so now we reach the point where you can get a clue about what the ascii graphic @ http://ffmpeg.org/ffmpeg.html#Detailed-description means [23:57] <akira4> yeah. [23:57] <ubitux> the decoded frames box is where you plug your filters typically (to alter the image itself) [23:57] <ubitux> ok, so this is how the audio and video works [23:57] <ubitux> subtitles are a PITA so it's a bit different [23:57] <akira4> the filters can be used on both audio and video frames right? [23:57] <ubitux> yes [23:58] <ubitux> nowadays both audio and video are stored in the AVFrame structure [23:58] <wm4> ubitux: we should explain how audio/video is displayed [23:58] <wm4> since video players are, you know, a very common use case of ffmpeg [23:58] <ubitux> i don't think that's necessary now [23:58] <ubitux> i'm trying to make sure the demuxer/muxer and decoder/encoder process is well known [23:59] <ubitux> so akira4 can write a demuxer/decoder for a simple subtitle format [23:59] <ubitux> and actually understand what's going on [23:59] <ubitux> anyway [23:59] <ubitux> akira4: so, are you ok so far? [23:59] <akira4> Yep. [00:00] --- Tue Sep 30 2014

1 0

ffmpeg.log.20140929
by burek 30 Sep '14

30 Sep '14

[00:16] <dashcloud> Hi, I'm looking for a commandline to encode video from my capture card to SD h264 in real time. If it matters, I'll be editing it afterwards to trim out things. [00:19] <c_14> https://trac.ffmpeg.org/wiki/Encode/H.264 [00:24] <dashcloud> thanks! do you think an AMD Athlon II X2 B22 is fast enough to do real-time encoding of SD resolution h264? [00:25] <c_14> Hmm, you'd have to try. [00:25] <c_14> Using -preset ultrafast will help. [00:26] <dashcloud> thanks [00:28] <iive> i'm with i3 2.9GHz (2 physical cores, 2 virtual). I can encode SD at 3 times the realtime speed. [00:28] <iive> that is, with the normal default setting. [00:29] <iive> btw, are you capturing though v4l2 (aka raw image) or dvb (aka digital tv) [00:33] <dashcloud> v4l2 [00:35] <iive> libx264 it is then :) [01:01] <Peter_Occ> here is a command line which seems to work fine [01:01] <Peter_Occ> ffmpeg -f mjpeg -r 8 -i http://user:pass@192.168.0.6/video/mjpg.cgi -r 10 /var/www/html/videos/videoMain2.mp4 [01:01] <Peter_Occ> Why would I have set it up this way, with the rate for input at 8 while the rate for output is 10? I remember doing that on purpose but I don't remember why. [02:52] <waressearcher2> I have one video 10 minutes long that is with 24 fps, and other 10 minutes long that is with 29.970 fps, when I concatenate them together using command: "ffmpeg -f concat -i list.txt -c copy out.avi" in place where they are joined there is a lag or pause for few seconds and if I add audio to that out.avi when it reaches 10 minutes there goes that [02:52] <waressearcher2> lag then video continue playing but the audio seek forward for few minutes, what I did to remedy it is added "-r 24" to ffmpeg command but will it worsen the quality ? or should I've added "-r 29.970" instead ? [02:53] <c_14> Wait, that helped? [02:53] <c_14> -r as an output option should not do anything if -codec copy was used. [02:53] <c_14> iirc [02:54] <waressearcher2> I don't remember if I removed "copy" option [02:55] <waressearcher2> but anyway, if I concatenate two videos, 24fps and 29.970fps, should I increase to upper value or decrease to lowest value fps ? [02:56] <c_14> I'd increase just to avoid frame dropping. [02:56] <c_14> But it doesn't really matter. [02:56] <c_14> Might depend on if you want to stay more PAL or more NTSC conform. [04:45] <sebastianlacuest> Hi everyone! [04:45] <sebastianlacuest> does anybody knows about qmg file format support in ffmpeg? [05:00] <c_14> Don't think so. [05:05] <active8> in linux, when ffmpeg runs and all those output messages fly, do the errors (in red) go to stderr ( fd 2) or stdout (fd 1) - i take it all normal output text goes to stdin; and what about those warnings that I see in yellowish orange? I'd like to catch them in some info dialogs I can pop up with zenity (pretty cool, that) [05:06] <JodaZ> active8, all messages go to stderr [05:07] <active8> thanks JodaZ. I'll try catching it. be nice to catch the errors separately. guess I'll have to filter on key words in the output. [05:10] <JodaZ> active8, use the return code [05:13] <active8> JodaZ, won't that just return success or failure. I was going to use two info popups. one for warnings and/or errors (the dialog icon is specific to the zenity dialog type) and one for normal output. be interesting to see if zenity preserves the message colors. [05:13] <JodaZ> active8, thing is, since encoding is supposed to be deterministic, you can always rerun it to get the error :) [05:15] <JodaZ> oh, you making some kinda gui [05:15] <JodaZ> hmm [05:15] <active8> the script I'm writing should figure out whether the dialog needs to have and exclamation icon, info, or error icon [05:16] <JodaZ> doesn't new ffmpeg output colored error info? (or was that libav) [05:16] <c_14> It does. [05:16] <JodaZ> colour would correspond nicely to your error classification [05:17] <JodaZ> but you propably shouldn't show anything poping up for anythign below red, and even for red it shouldn't abort ffmpeg or interrupt it or anything [05:17] <active8> bash script calls zenity for output info. zenity does list, checkboxes, text views, etc. radios - I could set codec, formats, pick files, all that good stuff and build the command froom the selections and not have to process commandline options to the script. [05:18] <c_14> JodaZ: I'm pretty sure red is fatal, isn't it? [05:18] <JodaZ> dunno man [05:18] <active8> i might find a way to ... easy q for quit echo q >&1 - i think [05:19] <active8> or >&0 stddin [05:19] <JodaZ> zenity sounds annoying >_> [05:19] <louis__> what should encode lossless faster, huffyuv or ffv1? [05:21] <active8> what's annoying is trying to remember that option t (or whatever) means a different thing for every stinkin linux cmd that has that option. -t for list in tar, -l for list in unzip. when I come home with a 3gp or amr on my phone, I want to just open the script, set whatever and do the comversion to mp3 or mp4 [05:21] <active8> or whatever i want without having to look up a command in my note. the script is my notebook [05:22] <JodaZ> active8, theres handbrake [05:22] <active8> you get handbrake to run on CentOS [05:22] <JodaZ> maybe i can interest you in a desktop os that is not trying to be rhel? [05:24] <JodaZ> but there are rpm's for it aparently [05:24] <active8> well, in a VM guest, debian sux and that travesty BKlinux died, IIRC. but it ran like a pig in mud when it wasn't locking up - cinelerra just locked. [05:25] <JodaZ> ?_? [05:25] <JodaZ> vm? [05:25] <JodaZ> handbrake is available for windows [05:26] <JodaZ> and centos [05:26] <JodaZ> dunno what more you could want [05:26] <active8> what was before that? AV Linux. buncha locking up. they wouldn't install right from a livecd, either. [05:27] <c_14> louis__: According to several short tests, they both take about the same length of time. But you should probably test with your source and your settings as I didn't take all variables into account (and didn't test that thoroughly). [05:28] <active8> handbrake is NOT available for CentOS in any of the repos I trust - the ones that won't break things. Now in windows, on the laptop, I can make exceptions. Doze is made to break anyway and I can keep XP in one piece with no effort [05:28] <JodaZ> ... [05:28] <c_14> The ffv1 output was about half the size of the huffyuv output though (again, might depend on the source and encoding settings). [05:28] <JodaZ> great os you got there if packages break the system [05:28] <JodaZ> but ya, suit yourself :) [05:29] <active8> thanks JodaZ . I'll keep handbrake in mind along with some of the other 'doze video and sound apps I've found usable. [05:33] <louis__> c_14: kdenlive says 45 minutes to encode ffv1, but 13 for huffyuv [05:33] <louis__> i am skeptical since the huffyuv video came out to be 29 gb for 27 minutes [05:33] <active8> JodaZ, it's for cert training - not meant for gui fooey. server and devel box. prob is that without carefully maintaining exclude lists in the repo.d files, it can accidentally replace a dependency from another app. at least that's what would happen with VLC til i found the nux-dextop repo. [05:33] <active8> but what distro / desktop OS did you have in mind, JodaZ ? [05:34] <c_14> louis__: I only tried with a short sample file, but both took about 35 seconds. [05:34] <c_14> I can try with something longer. [05:35] <active8> sumthing that will run in a VM guest without trouble? note that USB passthrough can be flakey with KVM/qemu [05:35] <JodaZ> replace a dependency? never had dep trouble with debian, while all the rpm plebs always go around installing random rpm's to save their lifes [05:35] <c_14> louis__: let me use the git version of ffmpeg while I'm at it. [05:36] <active8> JodaZ, debian ran like crap in a VM. Not sure why [05:36] <active8> but YAY! [05:36] <JodaZ> maybe because you brought up a gui on the shitty emu'd gpu [05:36] <JodaZ> or you didn't have kvm but original real deal qemu xD [05:36] <active8> I got Ardour to compile into a safe dire where I can run it directly without hosing anything [05:37] <active8> lots of ifs and and buts with VMs , lol 8) [05:37] <louis__> how are you c_14 ? [05:37] <active8> Maybe one day I can package my Ardour into an RPM and script that for easy updates [05:37] <louis__> thanks for all the help with the v4l stuff a few weeks back [05:39] <c_14> I'm fine, no problem. [05:39] <louis__> is it possible for huffyuv to do lossless video at only 169106 kilobits per second? [05:39] <louis__> that seems a little low [05:40] <nmz787> when I use libx264 trying to write video from opencv my computer freezes, but IYUV and DIV3 codecs work [05:40] <louis__> 1080p30 source [05:41] <c_14> I'm getting 130000k about for huffyuv 720p24 [05:41] <c_14> And the bitrate aside, the compression ratio is usually the more interesting part. Is the source raw? [05:42] <louis__> c_14: rawvideo capture [05:42] <louis__> i work with rawvideo because my wimpy laptop processor likes it better [05:42] <louis__> capturing lossless x264 works, but trying to edit it is a nightmare [05:43] <louis__> there's 5 seconds of lag just when you hit play in kdenlive [05:55] <c_14> Right, I've got some huffyuv encodes running now. Will do the ffv1 tests once they're done. [06:13] <c_14> Hmm, looking at some io stats, huffyuv at least might be IO limited on my machine. [06:46] <c_14> louis__: At least on my machine, both ffv1 and huffyuv take about the same amount of time and ffv1 creates files that are ~40% as large as the huffyuv encode, but as I said above I might be IO-limited. [07:35] <socksandsandals> is there an API in ffmpeg to take one AVPicture and draw it on top of another one at a given (x,y) offset? [08:26] <aditya3098> Hi, I have a live audio stream, and would like to use ffmpeg to relay this stream, with a minuite of latency [08:27] <aditya3098> In other words, if a a sound is sent over the main stream, then in a minuite, the same sound is sent over the new stream [08:27] <aditya3098> I tried other channels and was asked to use liquidsoap/gstreamer, but can I do it with ffmpeg [15:55] <svkoskin> I'm having an issue with -shortest option. I'm trying to convert a .ts file which has around 7œ minutes of H.264 video and 9œ minutes of audio to an .mp4 file, which should have 7œ min length (so that the part which only has audio track gets discarded, as I suppose -shortest should work that way). However, I get a .mp4 file with a longer audio stream than [15:55] <svkoskin> the video stream, so it seems -shortest is not working correctly. Here is the command line command and its output http://pastebin.com/k8r7dva6 [17:57] <quebre> hi [17:58] <quebre> i joined two movies into one, which is now: NVEExport.mpg, File size: 7.44gb, Audio: AC-3, Audio bitrate: 448kbit/s, Video codec: MPEG-2, Video bit rate: 3,0Mbit/s, Resolution 720x576 [17:58] <quebre> so how do i put it on my double layer DVD to play on dvd plaer ? [19:04] <sky99> I am trying to get a question answered about ffmpeg and I have tried the mailing list for for some reason anything I send there seems to fall into a black hole and never gets posted. [19:05] <sky99> My question is this: I have video files in .ts format that were obtained via a satellite receiver. [19:06] <sky99> They contain one video stream but mulyiple audio streams. [19:06] <sky99> multiple [19:07] <sky99> Each audio stream appears to have two channels (left and right) but one contains background sounds, another dialogue and so on. [19:08] <sky99> Is there any way using ffmpeg to merge all those audio streams down to a single stream, without changing the video at all? [19:09] <sky99> What i need to do is actually slightly more complicated than that nut that is the key part that I cannot figure out how to do. [19:09] <sky99> nut=but [19:29] <jberger1> hello [19:34] <jberger1> I've just upgraded to ffmpeg 2.4.1 and adjusted to the modified syntax of the overlay feature. Nearly all seems to be fine, except a input/output error messages about the (different) overlay .png files. The resulting .mov file works as expected, but I am unable to get more information about the reason, -loglevel debug does not help. [19:39] <jberger1> Debug output: http://pastebin.com/BYXdtUdh [19:54] <JodaZ> sky99, https://trac.ffmpeg.org/wiki/AudioChannelManipulation [20:12] <vlatkozelka> hi ... how to change ffprobe output format to xml instead of the plain text default ? i read its possible but can't figure out how [20:12] <JEEB> http://www.ffmpeg.org/ffprobe.html [20:13] <vlatkozelka> i read that [20:13] <JEEB> then you should know [20:13] <vlatkozelka> but doesnt show what option to use [20:13] <JEEB> with little scrolling I see the option [20:14] <JEEB> and for the value I can see it tells me to scroll further [20:16] <vlatkozelka> i get it the value is xml [20:16] <vlatkozelka> but what comes before that ? [20:16] <JEEB> hint: the example in the documentation in that point has the value "json" [20:16] <JEEB> I mean, seriously. it's written there and I found it with little scrolling. [20:17] <JEEB> and I don't even use ffprobe like that so it's not like I already knew what I was looking for :| [20:17] <vlatkozelka> maybe im just tired ... [20:17] <someone-noone> hello. I capture my mic & webcam with avfoundation framework, but resulting file is out of sync. Here's what I run: http://pastebin.com/EZRRCXST [20:18] <someone-noone> I think it's due to the fact that my webcam starts not immediately as mic and that's why video little bit late [20:18] <someone-noone> Can ffmpeg fix it without offsets? [20:18] <JEEB> vlatkozelka, ctrl+f for "json" upwards from the listing of the formats :P you should get the example there [20:19] <vlatkozelka> yeah found that [20:19] <vlatkozelka> but [20:19] <vlatkozelka> its not really usefull i though im doin something wrong [20:19] <vlatkozelka> all it does is <ffprobe> </ffprobe> [20:19] <vlatkozelka> everything else is still printed as it would always do [20:20] <JEEB> the xml format has a comment in there [20:20] <JEEB> Note that the output issued will be compliant to the ffprobe.xsd schema only when no special global output options (unit, prefix, byte_binary_prefix, sexagesimal etc.) are specified. [20:20] <JEEB> anyways, no idea otherwise, that's general ffprobe usage then and I have nfi about that \o/ [20:31] <Polarwinds> JEEB, I finished my transcoding with the script you gave me. it reduced the whole series from 27 gigs to 13 :) [20:46] <gopi> I am trying to add watermark to a video [20:46] <gopi> this is my code [20:46] <gopi> ffmpeg -i bigbunny.mp4 -i /media/movies/watermark/wm_720p.png -filter_complex 'overlay=10:main_h-overlay_h-10' -t 1 test.mp4 [20:47] <gopi> I am getting following error "/media/movies/watermark/wm_720p.png: Input/output error' [20:48] <gopi> what is this error? How do I debug it.. [20:51] <vlatkozelka> try [0:v][1:v] overlay [20:52] <gopi> what does those parameters do? [20:52] <vlatkozelka> specifies what ur overlaying over what [20:53] <vlatkozelka> 0:v is the video , 1:v is the png [20:53] <gopi> oh..mapping.. [20:53] <gopi> i am getting same error [20:53] <gopi> this is the code ffmpeg -i bigbunny.mp4 -i /media/movies/watermark/wm_720p.png -filter_complex '[0:v][1:v] overlay=10:main_h-overlay_h-10' -t 1 test.mp4 [20:55] <vlatkozelka> -map 0:v:0 -map 0:a:0 -c:v copy -c:a copy [20:55] <vlatkozelka> 1 sec ill show u a test i just did [20:57] <vlatkozelka> gopi , try this [20:57] <vlatkozelka> http://pastebin.com/QfnVtMMZ [20:58] <gopi> ok.. [20:58] <vlatkozelka> thats the result i got [20:58] <vlatkozelka> http://oi59.tinypic.com/1428tau.jpg [20:58] <gopi> give me a minute [21:01] <gopi> vlatkozelka, I am getting same [21:01] <gopi> ffmpeg -i bigbunny.mp4 -i /media/movies/watermark/wm_720p.png -filter_complex "[1:v] scale=20:20 [s] ; [0:v][s] overlay=10:10" -map 0:v:0 -map 0:a:0 -c:v libx264 -c:a copy test.mp4 [21:02] <gopi> what does this error means? "/media/movies/watermark/wm_720p.png: Input/output error' [21:02] <gopi> can't it find the file? [21:02] <vlatkozelka> maybe [21:03] <vlatkozelka> try putting -f mpeg right before the test.mp4 [21:04] <gopi> i ll trye [21:04] <gopi> http://pastebin.com/7WGvdvhP [21:05] <gopi> that's the error report [21:05] <vlatkozelka> hmm [21:05] <vlatkozelka> can u try [21:05] <vlatkozelka> ffplay /media/...../ .. jpg [21:05] <vlatkozelka> just ffplay the picture see if it works [21:08] <gopi> i can't do ffplay..since i am running it as server.. [21:08] <gopi> i did ffprobe [21:09] <svkoskin> do you have right to write to /media/movies/watermark/ [21:09] <gopi> http://pastebin.com/J5WcdfSh [21:09] <gopi> yeah [21:09] <gopi> i set 777 [21:10] <gopi> folder watermark have 777 permission [21:18] <someone-noone> hello. I capture my mic & webcam with avfoundation framework, but resulting file is out of sync. Here's what I run: http://pastebin.com/EZRRCXST [21:19] <someone-noone> I think it's due to the fact that my webcam starts not immediately as mic and that's why video little bit late [21:19] <someone-noone> Can ffmpeg fix it without offsets? [21:22] <someone-noone> or: can I tell ffmpeg to recalculate pts values not how they actually written with avfoundation or some muxer, but when the first packet comes [21:22] <gopi> i got the watermark working.. [21:23] <gopi> I am still getting Input/Output error.. [21:23] <gopi> but watermark and encoding working.. [21:23] <gopi> thanks a lot.. [21:24] <vlatkozelka> ? [21:24] <vlatkozelka> its working but giving an error ? how ? and what did you do ... im interested [21:24] <vlatkozelka> gopi ^ [21:28] <gopi> vlatkozelka, watermark and encoding is working.. [21:28] <gopi> but I am still getting this message /media/movies/watermark/wm_720p.png: Input/output error [21:29] <gopi> I don't know why [21:29] <vlatkozelka> wierd but [21:29] <vlatkozelka> what did u do to make it work ? [21:29] <gopi> I didn't make any changes.. [21:30] <vlatkozelka> lol [21:30] <vlatkozelka> well if its working dont break it i guess [21:30] <gopi> It must have been working before as well...i was trying to figure out the error message...i didn't see the output [21:30] <gopi> lol [21:30] <gopi> i should have looked at the output.. [21:31] <gopi> stupid me :P [21:31] <gopi> lol [21:31] <vlatkozelka> programmers always care about errors , its warning we dont give a fuck about ;) [21:31] <gopi> But, I am still curious..why I am getting input/Output error.. [21:31] <gopi> lol...debugging is the fun side of programming.. [21:32] <gopi> :) [21:32] <gopi> Do you have any idea, how we can trace this error? [21:32] <gopi> i am curious to figure it out.. [21:32] <vlatkozelka> no not really ... [21:32] <vlatkozelka> maybe post on the forums [21:32] <vlatkozelka> or in the mailing list [21:33] <gopi> yeah..i will do.. [21:37] <gopi> vlatkozelka, I have posted to mailing list.. [21:38] <gopi> lets wait.. [21:49] <spectralsun> I created some video files from images, but I am having issues when concatting them with existing video files that were created by phones [21:50] <c_14> https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20med… [21:50] <spectralsun> c_14: https://dpaste.de/si7M [21:51] <spectralsun> it's complaining about pixel format when I use this method [21:51] <c_14> That won't work. [21:51] <c_14> Have you tried reading the link I gave you? [21:51] <c_14> Try using the concat demuxer. [21:51] <c_14> That or the filter. [21:52] <c_14> Since you're reencoding anyway. [21:52] <c_14> The only video format (I know of) that you can concat with `cat' is mpegts. [21:52] <spectralsun> I'll try with the filter [21:52] <spectralsun> I also tried with the mmcat scrip [21:53] <spectralsun> https://dpaste.de/BB9T [21:53] <spectralsun> Output file #0 does not contain any stream [21:53] <c_14> >/tmp/mcs_v_all: Operation not permitted [21:53] <c_14> Something's going wrong. [21:54] <spectralsun> indeed. [21:54] <c_14> I also would not recommend that script. I've had mixed results with it. [21:54] <c_14> Sometimes it works fine for people, sometimes you need to patch certain parts, and at other times it just doesn't work. [21:55] <c_14> Also, it might not break anything, but are you sure you want this: [..]b/nsextreme/../nsextreme/[..] ? [21:57] <spectralsun> c_14: those were paths generated by the django app [21:57] <spectralsun> getting better results with the concat demuxer [21:58] <c_14> Better or good? [21:58] <spectralsun> it made a file, but the video clip that is an actual clip from a phone, is all messed up [21:58] <spectralsun> I'm going to see if converting it all to the same scale helps [21:59] <spectralsun> the file plays in ffplay [21:59] <c_14> Scale shouldn't matter that much, (except for maybe with codec copy). What has to match is things like pixel format etc. [22:01] <spectralsun> the video clip created from the image plays correctly, but the video from the actual video file looks like this: http://i.imgur.com/7Y1O76A.png [22:04] <c_14> Ye, looks like something doesn't match up between the 2 videos. [22:04] <c_14> Try with the filter or pastebin the ffprobe of both files. [22:06] <spectralsun> Okay, would it help if all the files I'm concatting are the same resolution? [22:06] <c_14> Are you reencoding during concat or not? [22:07] <spectralsun> Yeah, I want them all to come out as .mp4 [22:08] <spectralsun> basically I'm taking a 6 second clip from videos, and having splash image + 6 second clip + another splash image [22:08] <spectralsun> sometimes the 6 second clip is a .mov, but it will usually be .mp4 [22:10] <c_14> If you're reencoding, differing sizes shouldn't be an issue, if you're only remuxing (ie with -codec copy) it might be. [22:13] Action: hawken looks around [22:14] <hawken> Wondering if there are any plans with making ffmpeg read bluray mvc (.ssif-files) and if I can contribute :P [22:16] <c_14> Related to OPW or on your own? [22:23] <hawken> c_14: OPW? I just want to be able to read those files :P but I won't mind putting some work in ffmpeg if I know where to start :P [22:23] <hawken> I'm just your regular computer science student [22:27] <ubitux> hawken: ask in #ffmpeg-devel instead [22:27] <ubitux> this is where the development on FFmpeg happens [22:27] <hawken> Mkay :P [22:44] <voip1__> Hello guys, please help me [22:44] <voip1__> I have problem and i am not sure my problem related with my source or with ffmpeg parameters. [22:44] <voip1__> When i am trying to play my source with VLC, video stops and starts. Stream source provider says on his side everything is ok, and he is able normal play with VLC [22:44] <voip1__> Please chek: http://pastebin.com/dEmLZYEz [22:46] <spectralsun> c_14: https://dpaste.de/kMfc#L23 [22:47] <c_14> splash.mp4 needs an audio track, even if it's just silence. [22:48] <spectralsun> ok, how can i add one? [22:48] <c_14> you can use aevalsrc=0 [22:49] <spectralsun> c_14: thanks, trying that now [22:49] <spectralsun> c_14: Unable to find a suitable output format for 'aevalsrc=0' [22:50] <c_14> For the current command you can just use -filter_complex 'aevalsrc=0:d=2[as];[0:0][as][1:0][1:1] concat=n=2:v=1:a=1[v][a]' instead of what you have as a filter_complex. [22:51] <spectralsun> ah [22:52] <spectralsun> c_14: Input link in1:v0 parameters (size 568x320, SAR 0:1) do not match the corresponding output link in0:v0 parameters (568x320, SAR 640:639) [22:54] <spectralsun> that was when i tried to use a splash.mp4 that was converted to the same resolution as the video file [22:54] <spectralsun> here is when I use the original splash: [22:54] <spectralsun> Input link in1:v0 parameters (size 568x320, SAR 0:1) do not match the corresponding output link in0:v0 parameters (1280x720, SAR 1:1) [22:55] <c_14> Try piping in1 through setsar=1 [22:55] <c_14> '[1:0]setsar=1[vs]' [22:55] <c_14> and then use [vs] instead of [1:0] in the concat [22:56] <spectralsun> c_14: 'aevalsrc=0:d=2[as];[1:0]setsar=1[vs]; [0:0] [as] [vs] [1:1] concat=n=2:v=1:a=1 [v] [a]' ? [22:56] <spectralsun> still same error [22:58] <c_14> scale=iw*sar:ih,setsar=1 [22:58] <c_14> ie [1:0]scale=iw*sar:ih,setsar=1[v1];[0:0]scale=iw*sar:ih,setsar=1[v2] [22:58] <c_14> then use v1 and v2 in the concat [22:59] <spectralsun> c_14: could you show me a full example in my context please? [23:01] <c_14> 'aevalsrc=0:d=2[as];[1:0]scale=iw*sar:ih,setsar=1[vs];[0:0]scale=iw*sar:ih,setsar=1[v2s]; [v2s] [as] [vs] [1:1] concat=n=2:v=1:a=1 [v] [a]' [23:01] <spectralsun> ok, we are getting somewhere [23:01] <spectralsun> that worked [23:01] <spectralsun> :) [23:02] <spectralsun> I did this: [23:02] <spectralsun> 'aevalsrc=0:d=2[as];[1:0]scale=iw*sar:ih,setsar=1[v1];[0:0]scale=iw*sar:ih,setsar=1[v2]; [v1] [as] [v2] [1:1] concat=n=2:v=1:a=1 [v] [a]' [23:02] <spectralsun> it rendered the clips in the wrong order though [23:02] <c_14> switch v1 and v2 [23:03] <c_14> either after the setsar (for correctness) or in the concat input. [23:04] <spectralsun> 'aevalsrc=0:d=2[as];[0:0]scale=iw*sar:ih,setsar=1[v1];[1:0]scale=iw*sar:ih,setsar=1[v2]; [v1] [as] [v2] [1:1] concat=n=2:v=1:a=1 [v] [a]' -map '[v]' [23:04] <spectralsun> I changed the input streams before the scale parameter [23:05] <c_14> That works too. [23:05] <spectralsun> now its in the correct order, but the splash plays for too long and you only see a few seconds of the video clip [23:05] <c_14> Can you pastebin the command output? [23:07] <spectralsun> c_14: https://dpaste.de/ucnw [23:09] <c_14> That one's still in the wrong order, isn't it? [23:09] <c_14> Also, you should change d=2 to d=4 since splash.mp4 is 4 seconds long. [23:09] <spectralsun> yes, you're right [23:09] <spectralsun> ok, I'll try that [23:10] <c_14> You might just want to remux the splash mp4 directly with the silent audio [23:11] <c_14> ffmpeg -i splash.mp4 -af aevaslrc=0 -c:v copy -c:a [whatever] -shortest out.mp4 [23:12] <spectralsun> I accidently made the splash 4 seconds, it really is supposed to be 2. Can I add silent audio with this command: [23:12] <spectralsun> ffmpeg -f image2 -r 1/2 -i Downloads/nsextreme_logo.jpg -vcodec libx264 -pix_fmt yuv420p -b:v 128k ~/code/nsextreme/nsextremeweb/nsextreme/video/clips/splash.mp4 [23:13] <c_14> yeah, just add -af aevalsrc=0 -shortest [23:14] <spectralsun> I did that, but still only seeing stream 0:0 for video [23:15] <c_14> In splash.mp4? [23:15] <spectralsun> Yep [23:15] <spectralsun> ffmpeg -f image2 -r 1/2 -i Downloads/nsextreme_logo.jpg -vcodec libx264 -pix_fmt yuv420p -b:v 128k aevalsrc=0 ~/code/nsextreme/nsextremeweb/nsextreme/video/clips/splash.mp4 [23:16] <spectralsun> sorry that was an old try [23:16] <spectralsun> it has the -af flag [23:16] <c_14> Try using -f lavfi -i aevalsrc=0 -shortest [23:16] <c_14> You should place that before -vcodec [23:18] <spectralsun> okay, that worked. [23:22] <spectralsun> Still the same result on the ouput [23:22] <spectralsun> for the concatted file [23:29] <voip1__> guys, please help me [23:29] <voip1__> I have problem and i am not sure my problem related with my source or with ffmpeg parameters. [23:29] <voip1__> When i am trying to play my source with VLC, video stops and starts. Stream source provider says on his side everything is ok, and he is able normal play with VLC [23:29] <voip1__> Please chek: http://pastebin.com/dEmLZYEz [23:29] <spectralsun> c_14: output.mp4 should be around 4 seconds, but ffprobe says it's 8 [23:30] <spectralsun> https://dpaste.de/kKOH [23:30] <spectralsun> line 81 is when I do ffrobe [23:31] <c_14> ah, the fps don't match up [23:32] <c_14> append an ,fps=29.97 before the [v1] [23:33] <c_14> right after the setsar=1 [23:34] <spectralsun> Okay, thats better, but the splash screen is only up for a second [23:34] <spectralsun> the video clip plays fine after a flash of the splash screen [23:34] <spectralsun> ffrobe reporting correct video time [23:36] <c_14> So the splash screen shows for 1 second instead of 2? [23:37] <spectralsun> not even a second, its just a flash of it [23:37] <spectralsun> plays in vlc! thats a good sign [23:38] <spectralsun> but yeah, it shows the splash for a second, then it immediately shows the first frame of the 2nd clip [23:38] <spectralsun> during when it should be showing the splash [23:38] <c_14> Might now why, hmmm. Recreate the splash with -r 29.97 as an output option and then run the command again without the ,fps= [23:39] <c_14> *know [23:41] <spectralsun> c_14: ffmpeg -f image2 -r 1/2 -i Downloads/nsextreme_logo.jpg -f lavfi -i aevalsrc=0 -shortest -vcodec libx264 -r 29.97 -pix_fmt yuv420p -b:v 128k -vf scale=568:320 ~/code/nsextreme/nsextremeweb/nsextreme/video/clips/splash568x320.mp4 [23:41] <spectralsun> I am using -r twice ? [23:42] <c_14> Once as an input option, once as an output option. [23:42] <spectralsun> is that correct ^? [23:42] <c_14> should be [23:42] <spectralsun> I used that splash output and it doesn't hang on the frame of the next video, however it still only shows the splash screen for a split second [23:43] <c_14> the duration of the splash video is as it should be though? [23:43] <spectralsun> no [23:43] <spectralsun> it's just a flash of it [23:43] <c_14> I mean the file. [23:43] <spectralsun> then into the next clip [23:43] <spectralsun> no [23:43] <spectralsun> its 00:00:00.12 [23:45] <spectralsun> c_14: how do I set the length of that file? Are you sure I should be using -r twice? [23:45] <spectralsun> I want the splah to be 2 seconds long [23:46] <c_14> I'm testing right now. [23:48] <spectralsun> ok [23:51] <c_14> According to the wiki what I said _should_ work, but I found an alternative. [23:51] <c_14> ffmpeg -loop 1 -i nsextreme_logo.jpg [foobar] -t 2 splash.mp4 [23:51] <c_14> keep the -r 29.97 as an output option [23:52] <spectralsun> c_14: sweet, working now :) [23:54] <spectralsun> so, this requires me to render the splash to the same size as the video I'm concatting to [23:55] <spectralsun> if I attempt to use a larger resolution splash, I get this error again: [23:55] <spectralsun> Input link in1:v0 parameters (size 568x320, SAR 1:1) do not match the corresponding output link in0:v0 parameters (1280x720, SAR 1:1) [23:55] <spectralsun> is it possible to have ffmpeg resize the splash during the combining of the clips? [23:57] <c_14> you can try activating the unsafe flag in concat [23:57] <spectralsun> I would prefer to do it this way, since we may get new higher resolutions in the future. (as opposed to having premade splash clips at identicle expected resolutions) [23:59] <spectralsun> c_14: I don't think that will solve my question [23:59] <spectralsun> I want to have it convert all video streams to a common resolution, I'll be using the video clip as that resolution [00:00] --- Tue Sep 30 2014

1 0

ffmpeg-devel.log.20140928
by burek 29 Sep '14

29 Sep '14

[02:38] <cone-658> ffmpeg.git 03Thilo Borgmann 07master:dd16a0d8cced: lavd/avfoundation: Add basic PCM audio support. [02:57] <michaelni> anyone has any oppinons about "[FFmpeg-devel] [PATCH 4/4] hevc: reuse edge emu buffer for coefficients" ? [02:57] <michaelni> if not ill probably apply it [02:59] <iive> what does it do? use emuedge buffer for idct ? [03:03] <iive> nope, tu.coeffs .. [03:10] <cone-658> ffmpeg.git 03Joakim Roubert 07master:3edb9aaea6aa: avformat/hlsenc: Added parameter -hls_allow_cache [04:46] <cone-658> ffmpeg.git 03Michael Niedermayer 07master:4d9a46fe31e0: avcodec/libx264: mark unchanged pointer arguments as const [04:46] <cone-658> ffmpeg.git 03Michael Niedermayer 07master:3956b12fdec5: avcodec/libx265: enable psnr reporting when requested by the user [05:33] <Timothy_Gu> Nice, FFmpeg is in Debian sid now [07:09] <pross> Do we prefer planar or interleaved audio (for new decoders?) [09:05] <nevcairiel> usually one of those makes more sense for the implementation [09:06] <nevcairiel> most codecs encode the channels separately and the decoders just used to interleave them after, in that case, use planar [10:07] <pross> nevcairiel: ok [11:02] <cone-238> ffmpeg.git 03Reimar Döffinger 07master:b7082d953fda: configure: add noexecstack to linker options if supported. [11:27] <pross> i need help with gcc asm syntax. anyone game? i want to 'pslldq' a c array. [11:30] <ubitux> are you really adding some inline code? [11:31] <pross> uh huh [11:31] <pross> __asm__( [11:31] <pross> "movdqu (%1), %%xmm0\n\t" [11:31] <pross> "pslldq $1, %%xmm0\n\t" [11:31] <pross> "movdqu %%xmm0, (%1)\n\t" : [11:31] <pross> "+r" (x)); [11:56] <Polarwinds> Hello, is there a way to convert 10bit mkv's with subtitles to mp4? [13:19] <pross> got it! [13:23] <kierank> nevcairiel: how did you make ffmpeg attempt to decode the substream? [13:23] <nevcairiel> if you just remove the limitation it'll attemtp to do it, but fail immediately [13:23] <nevcairiel> even the truehd headers seem to be missing [13:24] <kierank> I commented out m->max_decoded_substream = FFMIN(m->num_substreams - 1, 2); [13:24] <nevcairiel> commenting out is wrong, it still needs to be set to m->num_substreams -1 [13:24] <nevcairiel> just remove the ffmax [13:24] <kierank> ah [13:24] <nevcairiel> eh, ffmin [13:24] <kierank> oh i see max_decoded_substream is an index [13:27] <kierank> ah fails now [13:59] <kierank> nevcairiel: hmm yeah it's weird [13:59] <kierank> no restart header [14:12] <kierank> I'm doing this because my timestamps are doubles, and can't be [14:12] <kierank> expressed in integer timebases. [14:12] <kierank> !!!!! [14:12] <kierank> wm4: [14:12] <wm4> :) [14:12] <wm4> thought about changing this [14:12] <wm4> but I can't really see any reason to [14:12] <wm4> I mean, take Matroska: timestamps are rounded to ms boundaries... [14:13] <Daemon404> ns [14:13] <Daemon404> ms is just what libs do. [14:13] <wm4> but "nobody needs this except you" is probably still a valid reason to reject it [14:13] <wm4> Daemon404: well, timestamps within clusters are 16 bits [14:13] <wm4> so you can't realistically use ns, I think [14:14] <nevcairiel> use moar clusters [14:14] <wm4> though there's also the "timecode scale" [14:14] <Daemon404> though tbh, i dont know why people wouldnt store timestamps as dts/pts in the first place [14:14] <Daemon404> when designing a format [14:14] <wm4> simplicity!!!111111111111111 [14:15] <wm4> and then let's randomly use monotonic pts and reordered pts [14:15] <nevcairiel> SimpleBlock has a 16-bit timecode, Block uses a full variable-length mkv number [14:15] <Daemon404> lol [14:15] <Daemon404> matroska? simple? [14:15] <wm4> nevcairiel: are you sure? [14:15] <cousin_luigi> Greetings. [14:16] <nevcairiel> at least i thought thats how it worked, maybe i'm mixing it up with another element [14:16] <kierank> 1:13 PM <"Daemon404> ns --> wrong [14:16] <kierank> 27mhz [14:16] <wm4> nevcairiel: from what I can tell, they always use 16 bits [14:16] <Daemon404> nobody uses mhz as suc ha measurement outside of broadcast [14:16] <wm4> but block duration is a full timestamp [14:16] <kierank> Daemon404: the 27 is more important [14:16] <Daemon404> for broadcasters [14:17] <kierank> for content that's ntsc based [14:17] <kierank> which is probably the vast majority of content still [14:17] <Daemon404> for content that is in any way based in any sane format, ms is fine [14:17] <nevcairiel> in any case, if you really need ns precision, you can always use more clusters at a slightly increased overhead [14:17] <Daemon404> nevcairiel, so, nobody [14:17] <Daemon404> ;P [14:17] <nevcairiel> you can also use 0.1ms or whatever floats your boat [14:18] <wm4> I even thought about "fixing" these timestamps by correlating them with the FPS value stored in the matroska header [14:18] <Daemon404> wm4, that seems like it would cause a lot of issues [14:18] <wm4> yeah [14:18] <Daemon404> unelss youre quite careful [14:36] Action: cousin_luigi is not quite sure this is the right place to ask this, but do you have any idea why ffplay is not being built using these options http://sprunge.us/FEYG ? [14:37] <cone-228> ffmpeg.git 03Michael Niedermayer 07master:a52f443714b5: avcodec/vc1dec: fix is_intra block vs. mb bug [14:44] <nevcairiel> one of these days i should compare the vc1 decoder against microsofts reference [14:45] <nevcairiel> the output, i mean [14:46] <wm4> is vc1 decoding bitexact? [14:47] <nevcairiel> afaik the spec is, yes, our decoder, probably not [14:50] <nevcairiel> could probably compare it to the hwaccel output through ffmpeg, thats easier to test than a binary decoder [14:51] <nevcairiel> the hwaccel is bitexact to the software h264 decoder, so at least i know that the tests work :D [14:52] <JEEB> :) [14:52] <JEEB> IIRC the reference thingy was not only painfully slow but also hard to plug into anything [14:52] <nevcairiel> well i was thinking of the default microsoft decoder, would assume its bitexact [15:02] <Hoschi__> hi folks, could someone explain me the sentence between the row# 895 ->899 in the following file? http://ffmpeg.org/doxygen/2.4/mux_8c_source.html#l00867 [15:12] <cone-228> ffmpeg.git 03Michael Niedermayer 07master:04152a31cb41: avcodec/vc1dec: Fixing is_intra, 2nd attempt [15:25] <LigH> Greetings. [15:25] <LigH> Do you already know about issues compiling libutvideocodec in libavcodec? [15:26] <JEEB> it's unmaintained [15:26] <JEEB> the library that is [15:27] <LigH> Hmm. So it may not make much sense to try to create a ticket? [15:27] <JEEB> you can, but if it's mostly related to the library then I don't think much will be done about it since it's a third party component [15:28] <LigH> Well, GCC complains it cannot finf intmath.h, which seems to be a part of libavcodec, though. [15:35] <Daemon404> as teh author of that wrapper [15:35] <Daemon404> i want to delete it [15:36] <Daemon404> but people are unhappy if i do. [15:37] <LigH> Well, I am no real developer, I have not much experience with C compilers... I can only try to read and interpret what I believe to understand. [15:38] <LigH> For building ffmpeg, I use the media-autobuild_suite by jb_alvarado. [15:38] <LigH> It creates an MSYS2 environment for a Win64 cross compile with GCC. [15:38] <Daemon404> sure [15:38] <LigH> Then it downloads all required sources from CVS/SVN/GIT as required. [15:39] <Daemon404> but my point was: what do you need the libutvideo wrapper for/ [15:39] <Daemon404> or is it one of those "the recipe/package maintainer decided to enable it for no reason" things [15:39] <LigH> Probably to support Ut video AVIs as source in ffmpeg? [15:39] <Daemon404> there are native encoders and decoders for ut video [15:39] <Daemon404> is a* [15:40] <LigH> Then the configuration of ffmpeg might contain the wrong flags? [15:40] <Daemon404> libutvideo has to be explicitly enabled [15:40] <Daemon404> so someone has explicitly set it. [15:41] <LigH> If you have a recommendation, I will tell jb_alvarado about it. If enabling libutvideo is not required for ffmpeg to support Ut video format, I will tell him not to enable it. [15:41] <JEEB> for decoding it hasn't been for a very long time [15:41] <JEEB> kostya implemented it I think in 2011 or so? [15:42] <Daemon404> ... like 2 weeks after i wrote the wrapper [15:45] <iive> is the library encoder too? [15:46] <JEEB> yes, and I implemented the encoding in 2012 [15:46] <iive> so, are there any features that are present in the library and not in ffmpeg? [15:47] <JEEB> I have no idea if the library was upkept with any of the updates that the VFW encoder got, we plugged in the BT.709 stuff [15:49] <iive> so it's the opposite. Virtually there is no reason to use the library if you are not hunting bug in the ffmpeg ut implementation. [15:50] <LigH> Okay ... I just found that the media-autobuild_suite uses it only on one of two build calls (I believe only in the static building, not in the shared). [15:51] <LigH> May be a "forgotten to remove" mistake? ... Will try to patch the script and see what happens. [15:51] <iive> LigH: anyway, ffmpeg should build with the library present and enabled. fill bugreport anyway. [15:52] <iive> if nobody wants to maintain it, then it could be removed. [15:52] <iive> I wonder if libav have removed it already? Diego loves to nix such "cruft". [15:53] <LigH> I am just afraid that the response may be: "You are using m-a_suite, so that's their fault, not ours, *obviously*". Some people like to respond this way... ;) [15:55] <Daemon404> iive, it was never in libav. [15:55] <Daemon404> i wrote it. it's not useful anymore. [15:56] <wm4> KILL KILL KILL [15:59] <LigH> https://trac.ffmpeg.org/ticket/3984 [16:16] <iive> LigH: try to change the include to "#include "libavutil/intmath.h" and see if it compiles. [16:22] <cone-228> ffmpeg.git 03Michael Niedermayer 07master:b28f470df09c: avutil/pca: Make argument of ff_pca_add() const [16:26] <LigH> iive: Without "--enable-libutvideo", ffmpeg -decoders|-encoders both contain support for "Ut Video". [16:37] <LigH> Ciao... \o [16:39] <JEEB> iive, expecting him to actually do any source code modification? even if it's simple he won't just do it and go behind the thing that "he's just a user" [16:40] <iive> well, it's optional :) [16:40] <iive> he got his problem solved. I wanted a confirmation of a solution. [17:18] <arwa> hey could anyone brief me with --> how to get started with writing filters for libavfilters? [17:20] <wm4> doc/writing_filters.txt [17:21] <arwa> okay thanks :D [17:44] <ubitux> i need to fix the lena reference in that tutorial [17:46] <Daemon404> -_- [17:48] <ubitux> :( [18:26] <ubitux> -0, 11, 11, 1, 38016, 0xe1e821ca [18:26] <ubitux> +0, 11, 11, 1, 38016, 0xe1e621ca [18:26] <ubitux> oh wow. [18:26] <ubitux> dat crc change. [18:27] <ubitux> -0, 11, 11, 1, 518400, 0x722bc6e8 [18:27] <ubitux> +0, 11, 11, 1, 518400, 0x77cbc6e8 [18:27] <ubitux> same here [18:27] <ubitux> fear. [18:27] <durandal_1707> of what? [18:27] <ubitux> a52f443714b5c2a40ed272d8445f4c39220a4b69 [18:27] <ubitux> vc1 [18:28] <Daemon404> wut [18:31] <nevcairiel> ubitux: i saw those, i wondered how that is possible [18:32] <nevcairiel> in theory a single bit change should scramble the entire crc, shouldnt it [18:32] <ubitux> maybe there is a change at regular interval [18:32] <ubitux> or something like this [18:33] <Daemon404> well, for some codecs, it is trivial to force the decoded freame to any crc you want :P [18:33] <Daemon404> you could have much fun with fate [18:35] <durandal_1707> is it ok to use bytestream2 in probe function? [19:04] <cone-228> ffmpeg.git 03Michael Niedermayer 07master:d22e88d120eb: avcodec/x86/fmtconvert: Fix operand size in ff_int32_to_float_fmul_array8_sse* [19:06] <michaelni> ubitux, these are not the crcs you are searching for ... these are adler32 checksums [19:07] <ubitux> ooh ok [19:08] <Daemon404> obi wan niedermayer? [19:08] <michaelni> :) [19:20] <nevcairiel> hm, the hevc decoder has a bug, everytime it parses an in-band sps/pps/vps group, it seems to discard 2 video frames for output on this sample [19:24] <michaelni> probably best to send a mail to mraulet or open a ticket [21:38] <cone-228> ffmpeg.git 03Michael Niedermayer 07master:579a0fdc211f: avutil/lls: Make unchanged function arguments const [22:38] <cone-228> ffmpeg.git 03Michael Niedermayer 07master:b1a0fccd023d: avformat/hlsenc: export inner muxer timebase [23:20] <llogan> Why do trac bug reporters often leave "Patches should be submitted to the ffmpeg-devel mailing list and not this bug tracker."? [00:00] --- Mon Sep 29 2014

1 0

ffmpeg.log.20140928
by burek 29 Sep '14

29 Sep '14

[01:39] <imperfect-> wc [01:47] <MikeJoel> what command line option do I need to set to prevent degrigation in the video? If I create an mp4 everything looks fine the whole way through. If I run the same command yet change the output file to avi or swf it runs fine up to almost the end then the video starts blocking up and pixelating like a very low quality video. This is making video from pictures (pictures files are not changes) [01:56] <MikeJoel> Appears it starts degrading when the fade ins (which is the only movement in the video) stops and the same image is just repeated until the end of the video [02:55] <MikeJoel> http://www.parksandsonspiano.com/Testsite/output.avi is the video [02:59] <MikeJoel> like I was saying. You can see that near the end of the video it goes blocky [02:59] <MikeJoel> it does not happen in mp4 [02:59] <MikeJoel> the problem also shows up in swf [03:07] <iive> MikeJoel: interesting issue. [03:07] <iive> i'd recommend you to use constant quality encoding, e.g. -vqscale 2 or however you set fixed quantizer [03:08] <iive> bitrate encoding sometimes could do strange things. [03:09] <iive> also, look up mv0 and cbp, these might be option of flags, they help for static images. may need mbd=1 or/and trellis [03:09] <iive> sorry gtg. [03:09] <MikeJoel> ok thanks for the pointers [03:11] <iive> also, why not encode in h264? -c:v libx264 -crf 22 (it have different scale for qualitzation , 18 is like mpeg1/2/4 quant=2) [03:16] <MikeJoel> q:v seems to have fixed it [03:17] <iive> sure. just have in mind the fourcc fmp4 might not play on all devices. xvid/divx are more popular fourcc for it. [03:18] <iive> i'm off [05:04] <JodaZ> has anyone looked at http://files.plexapp.com/elan/ffmpeg/PlexTranscoder.tar.bz2 and checked if some changes might be worth backporting? [06:36] <spectralsun> I'm trying to make a video from an image, but I get the error " [06:36] <spectralsun> deprecated pixel format used, make sure you did set range correctly" [06:36] <spectralsun> ffmpeg -r 1/2 -i Downloads/logo_narrow.jpg -vcodec libx264 out.mp4 [06:37] <spectralsun> is the command i tried [06:37] <spectralsun> I get that error message in VLC when I try to play out.mp4 [06:37] <c_14> Does ffplay play the video? [06:38] <spectralsun> c_14: yes, it does! [06:38] <spectralsun> ok, so we're getting somewhere [06:38] Action: c_14 blames vlc [06:38] <spectralsun> I think the error might actually be [06:38] <c_14> Try a nightly? [06:38] <spectralsun> main vout display error: Failed to resize display [06:38] <spectralsun> because it shows that deprecated pixel yellow text in ffplay too [06:38] <c_14> Can you pastebin an ffprobe of the file and the relevant ffplay output? [06:40] <spectralsun> c_14: https://dpaste.de/0svC [06:42] <c_14> You can try forcing yuv420p instead of yuvj444p [07:23] <spectralsun> c_14: how do i do that? [07:24] <c_14> -pix_fmt yuv420p [07:26] <spectralsun> nope, didn't fix the vlc issue [07:26] <spectralsun> it's not a big deal, it just needs to be able to play in the browser [08:20] <mos_basik> I am trying to install ffmpeg on CentOS 6 following this guide: http://wiki.razuna.com/display/ecp/FFMpeg+Installation+on+CentOS+and+RedHat. [08:20] <mos_basik> I am stuck at the "Install Libvpx" step. The git checkout command as shown failed with "error: pathspec 'tags/v.1.3.0' did not match any file(s) known to git." Looking at the changelog in the master branch seemed to show it was already version 1.3.0, so I decided to ignore this step for now. [08:20] <mos_basik> The configure command failed with "nasm -Ox -f elf64 does not support section alignment (nasm <=2.08?)". The installed nasm is version 2.07, and Google says that's default for CentOS 6. Getting an updated version nasm looks pretty heavy for my power level. [08:20] <mos_basik> I thought it odd that yasm needed to be configured and installed and was conveniently the step before libvpx. Did I miss something that is supposed to make the process use yasm instead of nasm? Is it easier than it looks to get an updated version of nasm? [08:20] <mos_basik> This isn't the first time I've tried to compile ffmpeg myself, but I have never succeeded. On Ubuntu I eventually just used someone else's ppa, but I think here I'm stuck if I'm not able to get it working. [08:29] <JodaZ> mos_basik, use prebuilt binary? :) [08:32] <JodaZ> mos_basik, or try this guide: https://trac.ffmpeg.org/wiki/CompilationGuide/Centos [08:40] <mos_basik> JodaZ, that tutorial looks almost the same, actually. The order that the codecs are installed appears different, but that's about it [08:41] <JodaZ> libvpx checkout is different [08:41] <mos_basik> hmm, i see. the source is different [08:47] <mos_basik> same error (nasm <=2.08?). still researching [09:00] <mos_basik> is it feasible to install .rpm's built for Fedora or openSUSE on CentOS? [09:21] <JodaZ> mos_basik, http://ffmpeg.zeranoe.com/builds/ you can just extract that and run it [09:21] <JodaZ> (or put it in your path or whatever) [09:22] <JodaZ> o wait, nvm [09:23] <JodaZ> mos_basik, http://johnvansickle.com/ffmpeg/ these ones i meant [09:25] <mos_basik> Might do that. Found an rpm for a newer version of nasm, tried to install. needed a newer version of glibc. found an rpm for that, it needs newer versions of like five different things. [09:37] <seasc> good morning, i want to record from my webcam, but i have no sound, what am i missing? :: ffmpeg -v quiet -f v4l2 -s 640x480 -i /dev/video0 "/home/sea/webcam-out.mpg" [09:37] <seasc> its a linux system [09:41] <c_14> https://trac.ffmpeg.org/wiki/Capture/ALSA [09:42] <seasc> copied wrong line: ffmpeg -v quiet -f v4l2 -s 640x480 -i /dev/video0 -f alsa -i default -c:a mp3 "/home/sea/webcam-out.6.mpg" [09:44] <seasc> ty c_14 [09:47] <seasc> simple as that is working already: ffmpeg -v quiet -f v4l2 -s 640x480 -i /dev/video0 -f alsa -i default "/home/sea/webcam-out.mpg -- <3 ffmpeg :) [09:49] <mos_basik> JodaZ, going with the johnvansickle seems to be working. Hopefully it works with the use I have in mind. [09:50] <mos_basik> An installation question - what is best practice for setting up a static build like that on a multi-user system such that anyone can access it? Put it somewhere in /bin and put it in the path? [09:51] <c_14> mos_basik: /usr/local/bin maybe? [09:52] <mos_basik> c_14, looks good according to this http://devopsbootcamp.readthedocs.org/en/latest/_images/hierarchy.jpg. I'll do that. [10:00] <Akagi201> I am compiling ffmpeg with rtmp support (cross compile to TI dm365). But I always get `librtmp not found` [10:00] <Akagi201> here is my compile configs https://gist.github.com/anonymous/918600bf6f9460003ad6 [10:00] <Akagi201> I have put all needed libs and headers to /home/akagi201/libs-dm365 folder [10:02] <c_14> Can you pastebin your config.log? [11:58] <Polarwinds> Hello, is there a way to convert 10bit mkv's with subtitles to mp4? [12:01] <Polarwinds> http://pastebin.com/gbwaAg1u [12:01] <Nitori> -c:s copy [12:01] <Nitori> ? [12:02] <Polarwinds> it doesnt convert the video track to 8bit [12:02] <Nitori> because you copy it [12:02] <Polarwinds> how would i encode it then? [12:03] <Nitori> "-c:v libx264" or something like that [12:03] <Polarwinds> http://pastebin.com/gbwaAg1u [12:03] <Polarwinds> what would i do? [12:03] <Polarwinds> if i have the code put in i can just use it as a template [12:04] <Nitori> you want it hardsubs? [12:04] <Polarwinds> yes [12:04] <Polarwinds> please [12:06] <Polarwinds> http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=17&t=1417 for reference [12:06] <Nitori> ok [12:06] <Polarwinds> thank you [12:06] <Polarwinds> could you just make it so it doesnt hard sub it? [12:07] <Nitori> you do NOT want hardsubs? [12:07] <Nitori> what is it you want? D: [12:07] <Polarwinds> and ill the sub as an external files, which is easier? [12:07] <Polarwinds> i want it to be 8bit from 10bit [12:07] <Polarwinds> what ever is easier [12:07] <Polarwinds> i do want hard sub, and non hard sub [12:07] <Polarwinds> if possible [12:08] <Nitori> oh dear [12:08] <Polarwinds> what? [12:08] <Nitori> hard sub might be a little bit harder, if the mkv contains fonts [12:08] <Polarwinds> it will :/ [12:08] <Polarwinds> so can we do without? [12:08] <Nitori> we'd have to extract them, put them to your system fonts, and so on [12:09] <Polarwinds> i can just load the .ass seperatly [12:09] <Nitori> yes, but without the fonts it's going to look ugly [12:09] <Polarwinds> i am on windows [12:09] <Nitori> I know.. [12:09] <Nitori> the easiest way would probably to copy everything over, and just to reencode the video [12:09] <Polarwinds> hmm, what do you sugguest [12:10] <Polarwinds> okay [12:10] <Polarwinds> that would work [12:13] <Polarwinds> so? is it possible [12:13] <Nitori> wait a sec [12:13] <Polarwinds> okay [12:14] <Polarwinds> %m [12:14] <Polarwinds> hmm [12:17] <Nitori> hm [12:17] <Polarwinds> difficult? [12:17] <Nitori> having trouble copying the fonts [12:17] <Polarwinds> oh [12:17] <Polarwinds> i have the mkv file on my pc [12:18] <Polarwinds> it has to subtitle tracks, should i take out the one i dont want? [12:18] <Nitori> no, I cannot, for whatever reason, copy the .ass file into the mp4 container Oo [12:18] <Nitori> sure you can [12:18] <Polarwinds> hmm [12:18] <Polarwinds> interesting [12:18] <Polarwinds> but you can convert it without the .ass? [12:19] <Polarwinds> that's what i mostly need [12:19] <Polarwinds> is that code [12:19] <loadbang> Hi all, just updated to 2.4.1, it appears -pattern_type glob is broken. Has anything changed or is it a bug? I just get an error *.jpg: No such file or directory [12:21] <Nitori> [mp4 @ 0x13cb760] Could not find tag for codec ttf in stream #2, codec not currently supported in container [12:21] <Nitori> Oo [12:22] <Polarwinds> what does that mean [12:22] <loadbang> ok, https://trac.ffmpeg.org/ticket/3948 [12:22] <Nitori> it means, my ffmpeg doesn't want to put ttf into the container. I probably did something wrong [12:22] <Polarwinds> oh, can you fix it, so i just have to switch the name of the mkv to something else each time? [12:25] <Polarwinds> :/ [12:25] <Nitori> dunno [12:26] <Polarwinds> oh [12:26] <Polarwinds> so ? [12:26] <Nitori> well, probably. [12:26] <Polarwinds> for now i just need it so it converts the mkv to mp4 [12:27] <Nitori> without anything would be: ffmpeg -i "Coppelion - 01.mkv" -map 0:v -map 0:a -c:a copy -c:v libx264 -crf 18 -preset slow "Coppelion - 01.mp4" [12:27] <Nitori> maybe [12:28] <Polarwinds> i am going to try [12:28] <Nitori> that would *only* result in an mp4 with audio and reencoded video [12:28] <Polarwinds> can you make the audio to aac 640? [12:28] <Nitori> 640? [12:28] <Polarwinds> then send me the script :) [12:28] <Polarwinds> kbs [12:28] <Nitori> you mean 64? [12:28] <Polarwinds> no 640kbs [12:29] <JEEB> btw, if you want hardsubs straight off a matroska file with the styles kept, I recommend looking at mpv's encoding module [12:29] <Nitori> that's bullshit [12:29] <Polarwinds> no? [12:29] <Polarwinds> flac downsamples to 640kbs aac lc [12:29] <Nitori> oh, so it's flac [12:29] <Polarwinds> flac wont work in mp4 [12:29] <Polarwinds> :p [12:30] <Nitori> maybe you have to add something like: -c:a aac -strict -2 -b:a 640k [12:30] <Nitori> dunno if that's correct [12:30] <Polarwinds> let me try the script [12:30] <Nitori> and remove the -c:a copy [12:30] <Polarwinds> okay [12:30] <JEEB> https://github.com/mpv-player/mpv/blob/master/DOCS/encoding.rst [12:30] <JEEB> this should be able to hardsub the matroska straight off with the styles, basically [12:31] <JEEB> (which unfortunately ffmpeg is not really good at atm) [12:31] <Nitori> I never had problems to hardsub with ffmpeg. at least if the fonts are available in the system [12:32] <Polarwinds> what is the - 2 in the audio thing mean? [12:32] <JEEB> I remember vf ass needing you to extract the ASS and then manually installing the fonts [12:32] <Polarwinds> and the strict? [12:32] <JEEB> Polarwinds, -strict experimental [12:32] <Nitori> I use vf subtitles [12:32] <Polarwinds> oh [12:32] <Polarwinds> sorry i am new, i only used the basic function [12:33] <Nitori> JEEB, yes, extracting both ass and fonts. it's a bit of a hassle, indeed [12:33] <JEEB> yup, which is why mpv's encoding thingy is much simpler [12:33] <Nitori> never tried [12:33] <JEEB> it can't do stream copy, but in this case both V and A are re-encoded [12:33] <Polarwinds> http://pastebin.com/gbwaAg1u [12:33] <Polarwinds> does that look right guys? [12:34] <JEEB> yeah, but you probably wanna try out mpv [12:34] <JEEB> just simpler [12:34] <JEEB> no extraction etc [12:34] <Nitori> I have no fu*king idea what those first lines are for [12:34] <JEEB> http://mpv.srsfckn.biz/ [12:34] <JEEB> windows builds [12:34] <Polarwinds> sorry :C [12:34] <JEEB> see #mpv-player @ freenode for help (and the encoding.rst I linked) [12:35] <Polarwinds> why wont ffmpeg work ? [12:35] <JEEB> well it will, but you will have to do X steps instead of just one [12:35] <JEEB> which should be much simpler for you :P [12:35] <Nitori> in your paste, you added "ffmpeg" as the first argument to ffmpeg.exe [12:35] <Nitori> which is wrong [12:35] <Polarwinds> what? [12:36] <Polarwinds> idk :p i am confused [12:36] <Nitori> yea, I suggest you do what JEEB suggested, and try it with mpv instead [12:36] <Nitori> mpv is also a nice video player [12:36] <Nitori> especially for anime ;-) [12:36] <Polarwinds> http://puu.sh/bR3IX/bc259b833c.png [12:37] <Polarwinds> is the only error [12:37] <Polarwinds> so what do i do? [12:37] <Nitori> well, the file doesn't exist. that's what it says [12:37] <Polarwinds> hmm [12:37] <Nitori> you probably have to edit it, to include the complete path [12:38] <JEEB> anyways, for this particular use case, mpv is probably going to be simpler to use (and it uses FFmpeg's libraries in the background) [12:38] <Polarwinds> okay [12:39] <Polarwinds> i downloaded mpv [12:39] <Nitori> and then join #mpv-player [12:39] <Polarwinds> nothing so far :/ [12:39] <JEEB> start with playing the file to make sure you have the input file name right and you can see it rendering the subtitles right [12:40] <JEEB> and yeah, then we can move the discussion there [12:40] <Polarwinds> hmm [12:40] <Polarwinds> i still got the file doesnt exsist D: [12:44] <seasc> Polarwinds, if you're using linux, i made a script to ease such tasks: http://forums.fedoraforum.org/showthread.php?t=299182 [12:45] <seasc> Polarwinds, subtitles still beta though [12:45] <Polarwinds> http://pastebin.com/cyaWa55u [12:45] <Polarwinds> then i just need to encode it without them [12:45] <Polarwinds> what am i doing wrong? [12:46] <Polarwinds> http://puu.sh/bR4bh/2efed04422.png [12:46] <Polarwinds> the right file is there [12:46] <JEEB> you are writing the file name wrong, as I said [12:47] <JEEB> "c:\ffmpeg\Coppelion -01.mkv" compared to what is in that screenshot [12:47] <Polarwinds> so what do i change? [12:47] <Polarwinds> sorry [12:48] <seasc> we dont know what the proper file name is... however, i'd assume there is another space after the dash... "abc - 1.mkv" [12:48] <JEEB> we do, he just posted a screenshot :P [12:48] <Polarwinds> its Coppelion -01 [12:48] <Polarwinds> that's it [12:48] <JEEB> that's not what your screenshot says [12:48] <Polarwinds> D: [12:48] <JEEB> do you per chance have bad eyesight? [12:49] <Polarwinds> i am wearing glasses... [12:49] <seasc> just as my assumption was :D [12:49] <JEEB> well so do I [12:49] <JEEB> but I can still clearly see that space after the dash [12:49] <JEEB> in your screenshot [12:49] <JEEB> that is why you use tab-autocompletion in a shell or a command line, or heck - just right-click the file name and select "rename" to select the file name so you can copy it :P [12:50] <JEEB> anyway, I recommend you go back to mpv since it will most probably give what you want in a more simple manner, without extracting the subtitles and fonts [12:50] <Polarwinds> "C:\ffmpeg\bin\ffmpeg.exe" -i *C:\ffmpeg\Coppelion - 01.mkv" -map 0:v -map 0:a -c:a aac -strict -2 -b:a 640k -c:v libx264 -crf 18 -preset ultrafast"*C:\ffmpeg\Coppel [12:50] <Polarwinds> is what i have [12:51] <Polarwinds> hmm i could renameit [12:51] <JEEB> ... [12:51] <seasc> now this looks better than previously :p [12:51] <JEEB> yes, now the file name and path looks correct :P [12:52] <Polarwinds> it worked [12:52] <Polarwinds> now i got this : http://puu.sh/bR4rI/0b40304949.png [12:52] <seasc> i'm more confused by his: "-map 0:v" and "-map 0:a" i'm used to use the real id's from ffmpeg -psnr -i video... would they just include all audio and video streams? [12:52] <Polarwinds> i need to convert both [12:52] <Polarwinds> :/ [12:52] <JEEB> seasc, yeah - it would select all of that type [12:53] <Polarwinds> the files is flac and i want to convert it to aac lc 640kbs [12:53] <Polarwinds> so? [12:53] <Polarwinds> thanks for helping me learn [12:54] <seasc> Polarwinds, there are some weird symbols between 'ultrafast' and the output file [12:55] <seasc> JEEB, ty [12:55] <Polarwinds> http://pastebin.com/cyaWa55u [12:55] <Polarwinds> there is my error [12:55] <Polarwinds> error is the puush [12:55] <seasc> again... put a space betwen arguments [12:56] <seasc> there is none between ultrafast and the outputfile [12:56] <Nitori> and there is a quotation mark at the beginning in front of the mkv filename missing [12:56] <Nitori> and at the end [12:56] <Polarwinds> oh [12:57] <seasc> it is ther at the beginng of outputfile, but tthe end is missing.. yes [12:57] <seasc> ahh the input misses the leading quote, and the output the trailing quote ;) [12:57] <Polarwinds> http://pastebin.com/cyaWa55u [12:58] <Polarwinds> i did the quotes around the file names now? [12:58] <Nitori> oh dear.. [12:58] <JEEB> no you did not [12:58] <Polarwinds> sorry :C [12:58] <JEEB> also stop hurling yourself from one place to another [12:58] <Polarwinds> sorry. [12:58] <JEEB> if you want to use mpv, just stay on that track [12:58] <Polarwinds> ill try? [12:58] <JEEB> you're just making this harder to yourself [12:58] <JEEB> uhh [12:58] <JEEB> I goddamn spoonfed you a command line [12:58] <Polarwinds> :c [12:59] <JEEB> since we now know the full path to your source file [12:59] <Polarwinds> :/ [12:59] <Polarwinds> sorry :c [13:00] <Polarwinds> i really want to use ffmpeg [13:00] <JEEB> no, for what you are doing (hardsubbing ASS straight off of a matroska file) you do NOT want to use it [13:00] <JEEB> it just adds extra manual steps, why would you do it [13:00] <Polarwinds> cause my tablet cant play 10bit mkv :/ [13:00] <JEEB> ... [13:01] <JEEB> MPV FUCKING LOADS EVERYTHING FROM THE FUCKING FILE [13:01] <JEEB> YOU DON'T NEED TO EXTRACT [13:01] <JEEB> CAN'T YOU FUCKING READ [13:01] <Polarwinds> D:< [13:01] <Polarwinds> slowly [13:01] <JEEB> I'm trying to give you a simpler fucking alternative, I spoonfed you a fucking command line after the file name became known [13:02] <Polarwinds> mpv wont open? [13:02] <Polarwinds> i left clicked it [13:02] <JEEB> just like ffmpeg, as I said [13:02] <JEEB> it's a COMMAND LINE APPLICATION [13:02] <Polarwinds> okay [13:03] <seasc> try: C:\ffmpeg\bin\ffmpeg.exe -i *"C:\ffmpeg\Coppelion - 01.mkv" -map 0:v -map 0:a -c:a aac -strict -2 -b:a 640k -c:v libx264 -crf 18 -preset ultrafast "*C:\ffmpeg\"Coppelion-01.mp4" [13:03] <seasc> err [13:03] <seasc> try this: C:\ffmpeg\bin\ffmpeg.exe -i *"C:\ffmpeg\Coppelion - 01.mkv" -map 0:v -map 0:a -c:a aac -strict -2 -b:a 640k -c:v libx264 -crf 18 -preset ultrafast "*C:\ffmpeg\Coppelion -01.mp4" [13:04] <seasc> i dont care about the output name ^^ he can rename later [13:04] <Polarwinds> it says invalid arguement [13:05] <seasc> yea, try the 2nd one [13:05] <Polarwinds> still? [13:05] <Polarwinds> :/ [13:05] <seasc> whats the error? [13:06] <Polarwinds> http://puu.sh/bR575/e629f5ff72.png [13:06] <Polarwinds> its the exact same D: [13:06] <Polarwinds> the file is the same name as the code >.< [13:07] <seasc> make the error area larger, that little thing doesnt say anything [13:09] <Polarwinds> it just says invalid aruguement [13:09] <Polarwinds> that is it o [13:10] <Fjorgynn> :D [13:10] <Polarwinds> ? [13:11] <seasc> One last try, and if there is an error... upload ALL of the console output !! [13:11] <seasc> http://paste.fedoraproject.org/137124/41190266/raw/ [13:13] <seasc> ctrl+a -> ctrl+c -> go to console and then right-click and paste it [13:13] <Polarwinds> http://puu.sh/bR5tY/17710dafbb.png [13:14] <seasc> ok thats my bad... windows doesnt handle them i guess :( [13:14] <Polarwinds> it can [13:14] <Polarwinds> D: [13:14] <Polarwinds> i have done it before [13:14] <seasc> but me out.. you have not provided enough area to figure errors... bye [13:15] <Polarwinds> bye :/ [13:20] <someone-noone> Hello. I have a nut file with uyvy422 frames and I want to convert it to mkv with libx264, but I got error "Unable to parse option value "-1" as pixel format". [13:21] <Fjorgynn> i see [13:21] <someone-noone> Here's a full stdout\stderr http://pastebin.com/uBdWwvH0 [13:21] <someone-noone> Why is it happening? And how can I fix it? [13:22] <someone-noone> I tried to provide different pix_fmt options (yuv420p and uyvy422), but it doesn't help [13:22] <seasc> try adding: -map 0 [13:23] <someone-noone> Same error with: ./ffmpeg -i ../Screen/out.nut -map 0 -c:v libx264 out.mkv [13:24] <someone-noone> perhaps it's due to disable-everything option. Perhaps I missed smth? There is a filter error in the end as you may see: [graph 0 input from stream 0:0 @ 0x7fee29d0cf20] Error applying options to the filter. [13:24] <someone-noone> Error opening filters! [13:26] <someone-noone> in ./ffmpeg -pix_fmts I can see that both yuv420p and uyvy422 are supported [13:28] <seasc> sry me cant help with this :( [13:39] <seasc> might the *nut require some special codecs? because ffmpeg seems very capable in autodetection usualy [14:10] <cousin_luigi> Greetings. [14:11] Action: cousin_luigi is compiling ffmpeg 2.4.1 with these options: http://sprunge.us/FEYG but ffplay is not being built. What would you recommend to look at? [14:29] <Twoll> hi [14:29] <Twoll> someone to help me to grab SSE3Tools.dll plz? [14:59] <Hoschi_> hi folks, could someone explain me the sentence between the row# 895 ->899 in the following file? http://ffmpeg.org/doxygen/2.4/mux_8c_source.html#l00867 [15:03] <JEEB> if we had an input packet, after interleaving its content to another, set its contents to zero and re-initialize it is how it looks like [15:03] <JEEB> and then the local pointer is set to NULL so it can't be re-used in that function [15:05] <Hoschi_> hie JEEB -> does this one not leave a dangling pointer? [15:06] <JEEB> the packet is re-initialized so I guess it's not freed so you can re-use it again [15:06] <JEEB> of course I haven't looked into the contents of an avpacket in a very long time by now :P [15:06] <JEEB> so if there's a pointer there and interleave_packet doesn't do anything to it [15:07] <JEEB> that said, looking at interleave_packet [15:07] <JEEB> there's if (in) av_free_packet(in); in there [15:07] <JEEB> so it does free the content of the packet in there, and then the actual avpacket is just re-initialized by the code you linked [15:15] <Hoschi_> thank you, i need to debug a more deeply to understand the interleaving function, because i think my timestamp calculation is not accurate enough. After i upgraded from V. 2.1 -> 2.4, my program allocates a lot of memory :-( [15:19] <seasc> @ someone-noone : just found (at the bottom) http://trac.ffmpeg.org/wiki/Encode/H.264 might help you [15:24] <seasc> Could someone please give a an advice how to remove the 'pixelish' start (as in the ~first minute) of encoded videos(?), afterwards the quality is just fine... I thought it would be better when adding buffer (maxrate, minrate, bufsize) but it acutaly seems to be about the same :( [15:24] <seasc> cmd used: ffmpeg -v quiet -i "01 To You, Two Thousand Years Later.mkv" -strict -2 -map 0:0 -b:v 1024K -vf scale=1920x1080 -c:v libx264 -minrate 2048K -maxrate 2048K -bufsize 1024K -b:a 256K -c:a ac3 -c:s srt -map 0:3 -map 0:1 "01 To You, Two Thousand Years Later.0.mkv" [15:25] <seasc> also, 2nd question: subtitle prints some position values.. just at 'random' occasions, but not always, just rarley [15:25] <seasc> i'm using the same subtitle codec as input file [15:31] <seasc> http://pastebin.com/spperGp6 [15:38] <kepstin-laptop> seasc: if you can do 2-pass encoding, that should help the relative bit allocations somewhat. If you don't need the minrate/maxrate/bufsize then don't use them, they reduce quality somewhat. [15:39] <kepstin-laptop> (if you want 1-pass and don't need an exact filesize, use -crf rather than setting a bitrate) [16:00] <seasc> kepstin, problem beeing, 2pass videos are not played by my TV :( -- i do not need an exact filesize, however, i want it small with good quality, (to me) that is easer accomplished with bitrates... so i may translate the crf suggestion to increase bitrate? [16:00] <seasc> oh (s)he left [18:11] <seasc> which codec to use for 'wav' files? I'm on linux and having wavpack and wavesynth avilable, both fail... am i missing the right one? [18:17] <seasc> http://pastebin.com/uaW0kGS2 [18:17] <sacarasc> seasc: Encoding to? -c:a pcm_s16le [18:18] <seasc> sacarasc, thank you [19:37] <JodaZ> mos_basik, glibc is one of the things you do not just update [19:38] <JodaZ> mos_basik, because everything depends on it, thus updating breaks everything [19:52] <mos_basik> JodaZ,i see that now [00:00] --- Mon Sep 29 2014

1 0

ffmpeg-devel.log.20140927
by burek 28 Sep '14

28 Sep '14

[01:34] <cone-646> ffmpeg.git 03Luca Barbato 07master:d995f0a1550c: log: Unbreak no-tty support on 256color terminals [01:34] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:5a5d454dcce5: Merge commit 'd995f0a1550c9148140eab34c7d770e90925c004' [01:40] <cone-646> ffmpeg.git 03Andreas Cadhalpun 07master:883e98bc2575: probe: Bump the score for mime type matching [01:40] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:b7a55388706b: Merge commit '883e98bc25755327565d8ae0f565a6c886f24816' [01:48] <cone-646> ffmpeg.git 03James Almer 07master:26cd7b1e1ad5: x86/fmtconvert: add ff_int32_to_float_fmul_array8_{sse,sse2} [02:12] <cone-646> ffmpeg.git 03Luca Barbato 07master:6cfbe1de5ac6: avconv: Use only audio and video to guess discontinuities [02:12] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:ac6b5bb834c0: Merge commit '6cfbe1de5ac6c57c41459626f7ac32841d63ace8' [02:26] <cone-646> ffmpeg.git 03Luca Barbato 07master:9d2cee52d37c: configure: Assume a standard-compliant default libc [02:26] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:eb1ac04e59cb: Merge commit '9d2cee52d37c7340f85a5d41110282aac03e6855' [02:58] <cone-646> ffmpeg.git 03Rémi Denis-Courmont 07master:1c80c9d7ef80: hwaccel: Call ->get_format again if hwaccel init fails [02:59] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:01831fd00490: Merge commit '1c80c9d7ef809180042257200c7b5f6b81d0b0e2' [03:35] <cone-646> ffmpeg.git 03Rémi Denis-Courmont 07master:ce4231632114: h264: Always invoke the get_format() callback [03:35] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:28b1698a7ccc: Merge commit 'ce42316321145c435029dc288c2b5f1d48519dde' [03:54] <cone-646> ffmpeg.git 03Rémi Denis-Courmont 07master:4e03b2097ca5: mpeg12: Always invoke the get_format() callback [03:54] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:be5dd8aa5558: Merge commit '4e03b2097ca59843ed65424a572b7acd5345f088' [04:03] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:2ec4a84dca60: ffplay: Separate AVFrame from AVSubtitle [09:31] <cone-814> ffmpeg.git 03Paul B Mahol 07master:bfdf0f078a74: avfilter/signalstats: add more pixel formats [12:35] <cone-385> ffmpeg.git 03Christophe Gisquet 07master:f183febcdafb: hevc: remove CodingTree [13:07] <cone-385> ffmpeg.git 03Michael Niedermayer 07master:233d1acb7095: tests/fate/microsoft: add -flags +bitexact to ilaced_twomv [13:57] <cone-385> ffmpeg.git 03lvqcl 07master:e58fc44649d0: avutil/x86/cpu: fix cpuid sub-leaf selection [15:17] <ubitux> so... 2.4.1 in sid, nice [15:19] <iive> in Sid? [15:19] <JEEB> debian unstable [15:20] <nevcairiel> indeed [15:20] <nevcairiel> much better than experimental [15:26] <iive> that's great. For some reason I thought that sid is already released version of debian. [15:27] <nevcairiel> no, sid is never a released version, its the permanent name for the unstable edition, it gets a new name when it gets promoted to testing, and the unstable branch retains the name sid [15:29] <iive> aha, this explains where i've heard it. [15:41] <wm4> now it just needs to make it to testing [15:41] <wm4> then as I understand it'll be part of a release [16:09] <nevcairiel> wm4: imo unlikely before the current testing is actually released [16:10] <nevcairiel> but ubuntu pulls from sid kinda automatically, so it might get into 15.04 [16:10] <nevcairiel> too late for 14.10 [16:10] <iive> the plans for jessie release are for november, aren't they? [16:17] <nevcairiel> dont think so, november has like the first freeze date, with further freezes for fixes in december and doc updates in january [16:19] <iive> yes [16:25] <Compn> libav doesnt like ffmpeg in debian as far as i can tell [16:25] <Compn> or maybe some of the comments in the debian threads/bugs [16:28] <wm4> of course they don't like it [16:28] <wm4> just as ffmpeg didn't like it being kicked out of ffmpeg [16:28] <wm4> this is 100% politics [16:31] <ubitux> we're not actually kicking them out [16:31] <ubitux> i'm extremly happy we can finally get back in debian though [16:31] Action: wm4 too [16:32] <wm4> finally I can drop some more ancient shitcode from my code [16:32] <wm4> e.g. the old subtitle demuxer exists only because libav can't handle most subtitle formats [16:32] <nevcairiel> so you're essentially dropping libav compat then? [16:32] <wm4> I could port these ffmpeg changes back to libav myself, but jesus christ... [16:33] <wm4> nevcairiel: no [16:33] <wm4> it just means libav-mpv can handle not as many formats as ffmpeg-mpv [16:33] <ubitux> not having subtitles support is gonna brings you some mad users :p [16:33] <nevcairiel> you could've done that before [16:34] <ubitux> you should probably wait until it's available in ubuntu [16:34] <ubitux> not exactly sure when [16:34] <nevcairiel> can only help to increase pressure on the distros to move [16:36] <wm4> my userbase is tiny and most of these subtitle formats are rarely used, so that would be little pressure [16:37] <ubitux> you still keep the srt and microdvd internal decoders? [16:38] <wm4> currently they're still around [16:39] <ubitux> i guess you plan to move away from them when i'll change the decoded state of subs? [16:39] <wm4> probably [16:39] <wm4> I should check what exactly "my" srt decoder seems to do better than the ffmpeg one [16:39] <wm4> I recall there were some small issues [16:40] <ubitux> i was worried about it when Aurelien wrote the one in ffmpeg after i did it in ffmpeg [16:40] <ubitux> in mplayer* [16:41] <ubitux> but i didn't look in details [16:41] <ubitux> the srt decoder needs to move to the bprint first [16:41] <ubitux> that's the only one that doesn't and i'm a bit uncomfortable at these snprintf with the fixed buffer [16:42] <ubitux> not exactly a priority right now though [16:43] <wm4> I think I prefer such fixed buffers over the bprint automagic, thougj [16:43] <wm4> *though [16:45] <ubitux> really? [16:45] <ubitux> is it just a feeling or there is a technical ground? [16:45] <wm4> with bprint it would be allocating and deallocating all the time [16:46] <wm4> considering libav* does this all the time anyway, this is probably a weak concern [16:46] <ubitux> not really [16:46] <ubitux> bprint uses the stack for the first 1k iirc [16:46] <ubitux> then it fallbacks on heap [16:46] <wm4> I know [16:46] <ubitux> also, it can be put in the context [16:46] <ubitux> like i did with SAMI typically [16:47] <ubitux> so re-used after each event [17:34] <cone-844> ffmpeg.git 03Clément BSsch 07master:ff5f5cbb4653: avformat/assdec: output ASS packets [17:34] <cone-844> ffmpeg.git 03Clément BSsch 07master:9ec52e5533cf: avformat/assdec: make sure "Comments:" events end up in the header [17:34] <cone-844> ffmpeg.git 03Clément BSsch 07master:0aa5c5a89e2b: avformat/assdec: reindent after previous commit [17:34] <cone-844> ffmpeg.git 03Clément BSsch 07master:3a6fa38fb1bd: avformat/assdec: add myself to Copyright [17:34] <cone-844> ffmpeg.git 03Clément BSsch 07master:0e6bad1229d4: avformat/assenc: mux an event header if none are found in extradata [17:53] <cone-844> ffmpeg.git 03Clément BSsch 07master:71f6275169ce: avformat/assdec: make sure to free allocated data in case of error [17:56] <cone-844> ffmpeg.git 03Clément BSsch 07master:180d7f330bd9: avcodec/microdvddec: remove old compat code [17:56] <cone-844> ffmpeg.git 03Clément BSsch 07master:5ec90c90c604: avcodec/subviewerdec: remove old compat code [17:56] <ubitux> alright [17:56] <ubitux> better working ground now [18:56] <cone-844> ffmpeg.git 03Luca Barbato 07master:551746680462: configure: Provide a safe default for unknonw libcs [18:56] <cone-844> ffmpeg.git 03Michael Niedermayer 07master:5672107ed696: Merge commit '551746680462538bb5ad7adf55fb1248190428ec' [20:58] <cone-844> ffmpeg.git 03Michael Niedermayer 07master:d85ebea3f3b6: avcodec/ac3enc_template: fix out of array read [00:00] --- Sun Sep 28 2014

1 0

ffmpeg.log.20140927
by burek 28 Sep '14

28 Sep '14

[01:24] <thearchitectnlog> i m on mountain lion i m trying the following ffmpeg -f qtkit -list_devices true -i "" [01:25] <thearchitectnlog> m getting Unrecognized option 'list_devices'. [01:25] <thearchitectnlog> Error splitting the argument list: Option not found [01:25] <thearchitectnlog> can anyone help me ? [01:28] <thearchitectnlog> ? [01:30] <thearchitectnlog> http://pastebin.com/CbfbrHmt [01:31] <thearchitectnlog> done can u please help [01:32] <sacarasc> Try updating. I am not sure, but the option may have been added some time in the last 6 months which means 2.2 wouldn't have it. [01:33] <thearchitectnlog> ffmpeg version is fmpeg version 2.2.4 [01:35] <thearchitectnlog> i install ffmpeg with brew now [01:38] <thearchitectnlog> anyway what i trying to do is stream my mac web cam to my nodejs server [01:40] <thearchitectnlog> sacarasc: http://pastebin.com/d30WYgW0 can u please check this 2 [01:45] <thearchitectnlog> ? [02:00] <thearchitectnlog> anyone [02:00] <thearchitectnlog> ? [02:10] <sacarasc> thearchitectnlog: Now is the time to be patient. [02:14] <sacarasc> I don't know what is wrong, but people come and go in this channel all the time. Eventually someone should help if they can. [02:41] <thearchitectnlog> what is the latest macos of ffmpeg ? [02:42] <sacarasc> As it is possible to compile yourself, the latest release is 2.4.1. But you could also get the git version. [03:32] <ScottSteiner> is it possible to convert a subtitle so that its timing is delayed by 3 seconds? [07:09] <nykac> can someone help me figure out how to overlay video? [07:10] <nykac> I'm trying to stream x11grab with a webcam overlay to ffserver [07:10] <nykac> my command is http://paste.ee/p/KDshU [07:11] <nykac> I was using http://paste.ee/p/JCKJ2 (based off the wiki) but it wasn't working so I tried to get it working with a simplified filter [08:21] <dahat> Can anyone see why my attempt to build rtmpdump under mingw with the MSVC toolchain is failing? http://pastebin.com/MZ5287jZ [08:32] <drv> dahat: it looks like GCC-style options are being passed to CL, which isn't going to work... [08:40] <dahat> I must have missed that part... GCC arg(s) are being passed to CL? [08:42] <dahat> drv: I am puzzling over this because there are plenty of warnings through most of the log but nothing fatal until the end... and even then not very actionable [08:44] <drv> the MSVC equivalent of -o is /Fo, which is the first problem [08:51] <dahat> That seems to get me closer... also seems to be some issues with .o files being expected when .obj files are the intermediate output... I'll have to dig into that part tomorrow, thanks! [08:53] <drv> that is exactly what /Fo specifies (the object file name) [12:00] <_Arif_> Hello! [14:46] <ScottSteiner> is it possible to convert a subtitle so that its timing is delayed by 3 seconds using ffmpeg? [15:39] <eristisk> I upgraded my Debian system recently and now BOTH mplayer and VLC are playing the MOV files from my 5D Mark III back with very noticeable graphical glitches [17:18] <hjudt> hi. can anyone help me with converting a 1920x1080p movie to a 16:9 pal-dvd? i'm having problems with overscan, the stand-alone dvd player cuts off borders, while software players play it back properly. by research i stumbled over the pad filter, but all examples i found use an older ffmpeg syntax with -padding-top which doesn't work anymore. can anyone give me a generic command line example or tell me how the pad filter works please? [17:33] <hjudt> ok, i finally found the information about the pad filter on a website. [18:48] <Fjorgynn> hey hey [19:15] <arwa> I was trying to set up the environment for ffmpeg. Its giving me this error:- libfdk_aac not found. But I have already installed libfdk_aac. Could anyone please help me out with this?? [19:16] <JEEB> see config.log [19:17] <__jack__> install libfdk-aac-dev:amd64 [19:20] <arwa> My system is 32-bit. So, I don't think amd64 will work. :/ [19:20] <__jack__> ha [19:20] <__jack__> libfdk-aac-dev:i386 then [19:23] <arwa> And I looked in config.log, libfdk-aac is not present in it. What do I do next? [19:23] <JEEB> when did you look at it? did configure now give you no errors and it output the listing at the end which also lists fdk in the audio encoders part? [19:23] <JEEB> oh [19:23] <JEEB> not present [19:23] <JEEB> well yes [19:24] <JEEB> now look at where the check was that failed for it [19:24] <JEEB> if it gave you an error about it [19:24] <JEEB> __jack__, I'm thoroughly surprised any distro distributes fdk. Is this a source-based distro? [19:24] <JEEB> because binary distribution is limited according to fdk's license [19:25] <mark4o> binary is fine as long as it is not combined with gpl code [19:25] <JEEB> go look at the license [19:32] <mark4o> JEEB: [19:32] <mark4o> You must retain the complete text of this software license in the documentation and/or other materials [19:32] <mark4o> provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. [19:32] <mark4o> this? [19:32] <mark4o> You must make available free of charge copies of the complete source code of the FDK AAC Codec and your [19:32] <mark4o> modifications thereto to recipients of copies in binary form. [19:32] <mark4o> ^ or this? [19:33] <JEEB> mark4o, You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized [19:33] <JEEB> by appropriate patent licenses. [19:34] <mark4o> how is that different for binary vs. souruce? [19:36] <JEEB> IANAL and I don't remember this exactly, but at least looking at previous things, source code distribution is often not too much looked into, while binaries are something licensors actually care about. That said, at this point I'd probably point towards one of folks on -devel who've actually done the research [19:38] <mark4o> JEEB: ok, I know people have looked into it and will not distribute binary ffmpeg linked with it, but I thought it was because the GPL did not allow it [19:38] <mark4o> (and linking with GPL code such as x264 is more important) [19:39] <JEEB> yeah, that was a separate thing, where fdk isn't compatible with GPL. [19:39] <JEEB> although it might be related to the similar part of the license, of course [19:57] <darkfrog> can you use ffmpeg to work with audio-only files? [19:57] <JEEB> yes [19:58] <darkfrog> awesome. :) [20:13] <arwa> thanks :). It worked [22:35] <imperfect-> Anyone know if I can normalize audio with those pre-encoded videos that explode with loudness all the time? [22:35] <c_14> Look at the compand filter. [22:36] <imperfect-> compand, k. ty. [22:37] <imperfect-> compand? or compound? [22:37] <c_14> https://ffmpeg.org/ffmpeg-filters.html#compand [22:38] <imperfect-> Google failed me [22:39] <imperfect-> Hrm. wonder how I would use it to deal w/ with major spikes [22:40] <c_14> Look at the examples. [22:40] <imperfect-> Yeah I'm reading [22:41] <imperfect-> Here is another noise gate, this time for when the noise is at a higher level than the signal (making it, in some ways, similar to squelch):? [22:41] <imperfect-> I don't understand 'signal' in that passage [22:42] <imperfect-> And the other question I have is can I get it to just process the audio track and not the video? [22:49] <c_14> Signal is audio signal. [22:49] <c_14> use -af compand=arguments -c:v copy [23:51] <isweet1> Hey guys. I've got a command-line app that generates PNGs on the fly and writes them to stdout. I want to pipe this PNG stream into ffmpeg and then convert it to a video which will be streamed off a v4l2 device. [00:00] --- Sun Sep 28 2014

1 0

ffmpeg-devel.log.20140926
by burek 27 Sep '14

27 Sep '14

[00:16] <cone-763> ffmpeg.git 03Carl Eugen Hoyos 07master:8de7bdd756b6: Show correct pix_fmt for vp9 videos != AV_PIX_FMT_YUV420P. [00:16] <cone-763> ffmpeg.git 03Carl Eugen Hoyos 07master:475e3799cdcc: Support libopenjpeg 2.x via libopenmj2. [00:16] <cone-763> ffmpeg.git 03Carl Eugen Hoyos 07master:88c937fdc8a8: Autodetect jpg images. [00:29] <cbsrobot_> ubitux: do you mind if I add to the opw Misc Libavfilter extension something like: add >8bit support to libavfilter/drawutil.c ? [00:54] <Timothy_Gu> Hey all, why does https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel/ return 404? [00:54] <reynaldo> Timothy_Gu: llogan said michaelni disabled mailman due to some problem.bug [00:55] <reynaldo> problem and/org bug I mean [00:57] <Timothy_Gu> reynaldo, oh [00:58] <Timothy_Gu> reynaldo, llogan, michaelni, do you know when it will be back up? [01:03] <reynaldo> Timothy_Gu: no idea [01:03] <llogan> Timothy_Gu: i'm not sure [01:17] <wm4> ubitux, BBB: so, as you know, vp9 packets in webm sometimes contain two frames [01:18] <wm4> but libavcodec wants them neatly separated, and the parser usage in libavformat takes care of that [01:18] <wm4> but this also means that the decoder does not return a frame sometimes, and that the PTS for 1 frame will be lost [01:19] <wm4> that the decoder doesn't return a frame is very odd, I don't think any other decoder behaves this, as long as the packets are not corrupted, and frame skipping is disabled [01:22] <llogan> ...it's not deleted though. just disabled temporarily. [01:39] <rcombs> don't quite a lot of decoders sometimes not return a frame? E.g. if running multithreaded? [01:41] <wm4> only at start of decoding [01:41] <wm4> once you get the first video frame, decoding should always return a frame [01:41] <wm4> unless you direct it to drop frames, or maybe if an error of some kind happened [01:51] <iive> hum... doesn't the decoder return pts in the avframe? [01:51] <iive> timestamp that he gets as dts. [01:51] <wm4> the pts/dts is unset [01:52] <wm4> because the returned frame originates from the parsed packet, which doesn't have and can't have a pts [01:52] <iive> ok [01:52] <wm4> I'm not quite sure what vp9 issue this was, something about golden frames [01:52] <iive> still, afair, the decoder signals when frame is dropped and when there is no frame available. [01:52] <wm4> and how this is not an issue for libvpx, because it can handle multiple frames in a packet [01:53] <wm4> iive: dunno, does it? but makes no difference here anyway, because the pts association is gone [01:53] <iive> the code that handles the initial lag should handle this case... [01:54] <iive> didn't you just said that there is pts/dts ? [01:54] <iive> there is no pts/dts [01:56] <wm4> yes, possible the decoder could reorder it correctly, or something [01:58] <iive> oh, you mean that you are loosing the last frame [01:58] <iive> this is an issue with all b-frame lag formats, afair, it was workarounded by calling decode without new packet [01:59] <wm4> no [01:59] <wm4> that's not my issue [02:00] <iive> oh.. [02:07] <iive> so golden frames are frames that are not displayed, but later may be used for prediction with mv00 and cbp=0, aka emulation of b-frames. [02:11] <iive> so these hidden frames don't have pts. [02:12] <iive> i'm off. have fun. [02:27] <michaelni> about the ML, just the web interface is disabled, mailman should be working fine through the email interface [02:39] <Compn> so moderation is down then ? [02:40] <Compn> or can be moderated via email? never tried that [02:54] <michaelni> Compn, the webinterface should be up again in a day or 2 [03:00] <Compn> no worries [04:49] <cone-964> ffmpeg.git 03Thilo Borgmann 07master:a69c70e148d9: lavd/avfoundation: Split adding a device and getting the device configuration into separate functions. [05:13] <cone-964> ffmpeg.git 03Thilo Borgmann 07master:92827e186999: lavd/avfoundation: Using the actual stream index instead of hardcoded value. [05:13] <cone-964> ffmpeg.git 03Thilo Borgmann 07master:1ea7a3e04e42: lavd/avfoundation: Simplify debug message generation. [07:29] <rcombs> ^ we have confirmation that the numbers-in-usernames issue is fixed [08:39] <Ritika> reynaldo: Hi! [09:53] <ubitux> wm4: i don't know for the vp9 parser [09:55] <nevcairiel> the timestamp thing is just a design limitation of the generic parsing code [09:55] <nevcairiel> redesign welcome? :p [09:55] <wm4> is that so [09:55] <nevcairiel> that the vp9 decoder swallows soem frames is just a property of the codec otherwise [09:56] <nevcairiel> well, if the parser detects two frames in one packet, it splits the packet and assigns the timestamp to the first frame [09:56] <nevcairiel> in the vp9 case, this first frame is the one frame that gets swallowed later, and the timestamp lost [09:57] <nevcairiel> but of course the parser has no way to know this stuff, since its design is limited [09:57] <durandal_1707> what happened with mailing list archive? [10:15] <ubitux> wm4: can you send me a sup file? [10:15] <ubitux> would be nice to have the muxer though [10:15] <ubitux> so i could extract some weird pgs and try stuff with them [10:17] <durandal_1707> there is already patch for muxer [10:17] <ubitux> yes but pending [10:18] <ubitux> still waiting for an iteration, right? [10:18] <ubitux> i'll push the sup demuxer in a moment but i'd like a file first [10:19] <wm4> ubitux: can we try DCC? [10:19] <ubitux> go ahead [10:19] <wm4> wow it works [10:19] <ubitux> :) [10:19] <wm4> I just notice the file is WTFish huge though [10:20] <ubitux> awesome speed [10:20] <wm4> yeaaah [10:22] <wm4> ubitux: there's also this one http://samples.mplayerhq.hu/sub/BluRay/title03_track2.sup [10:22] <ubitux> ah, thanks [10:23] <ubitux> 36M meh [10:38] <cone-409> ffmpeg.git 03wm4 07master:7dd2005e1ac3: avformat: add SUP/PGS subtitle demuxer [10:43] <rcombs> blimey, was that not there before? [10:48] <ubitux> blimey? [10:48] <av500> https://www.google.com/search?q=blimey [10:48] <nevcairiel> he must be english [10:49] <rcombs> not even [10:49] <rcombs> while that expression isn't particularly common in the US, I thought it was pretty well-known internationally just because of its use in British literature and TV [10:50] <ubitux> oh, ok [10:51] <ubitux> that sup demuxer really feels simpler that the vobsub one [10:57] <wm4> nice [11:14] <cone-409> ffmpeg.git 03Clément BSsch 07master:9ae83ece8a8e: avformat/srtdec: simpler and more lenient probing [12:23] <cone-409> ffmpeg.git 03Thilo Borgmann 07master:f2254e36c048: lavd/avfoundation: Use microseconds as common timebase. [12:29] <cone-409> ffmpeg.git 03Diego Biurrun 07master:4d5b99dacdbb: caf: Give context structure a consistent name [12:29] <cone-409> ffmpeg.git 03Michael Niedermayer 07master:b0458e62c8fc: Merge commit '4d5b99dacdbb56373e5abe2ad35fc87b5c0c5bd2' [12:40] <cone-409> ffmpeg.git 03Diego Biurrun 07master:b97f6ef95629: pcm-dvd: Move a variable to a smaller scope [12:40] <cone-409> ffmpeg.git 03Michael Niedermayer 07master:4c57be124f42: Merge commit 'b97f6ef956293fe06ae1e99bd295f42613081f37' [12:59] <cone-409> ffmpeg.git 03Michael Niedermayer 07master:ee2e5acdb13f: avcodec/dxtory: remove unused variables [12:59] <cone-409> ffmpeg.git 03Peter Ross 07master:1964251be7df: libavformat/iff: print error message when DSDIFF compression type is not supported [20:54] <cone-646> ffmpeg.git 03Tim Walker 07master:b5fe13448b13: vc1: Use logical instead of bitwise or for coded_inter [20:54] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:bde8eb7fa33d: Merge commit 'b5fe13448b13bfdd48a75b7e12387e11e3526d7c' [21:00] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:a97328afef0c: vc1: Use logical instead of bitwise or for twomv [21:00] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:ecf552657889: Merge commit 'a97328afef0ccebfc8c3d9f9fdb8e93cbf1058ab' [22:06] <cone-646> ffmpeg.git 03Tim Walker 07master:b39ebcddd47d: fate: Add VC-1 interlaced twomv test [22:06] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:b7caf8940123: Merge commit 'b39ebcddd47daf37659796aaa7d068668086507a' [22:18] <cone-646> ffmpeg.git 03Tim Walker 07master:4c2fd4b26234: ac3enc: allow Dolby Pro Logic II as a preferred downmix mode. [22:18] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:15fda8bcf1ea: Merge commit '4c2fd4b262347273afe97865ba451a1abde43ae6' [22:27] <cone-646> ffmpeg.git 03Tim Walker 07master:cc4992aaf3db: ac3enc: allow Dolby Pro Logic IIz as the Dolby Surround EX mode. [22:27] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:c895fa7f66c0: Merge commit 'cc4992aaf3dbb0af88d9727983d75636baf1f8cc' [22:39] <cone-646> ffmpeg.git 03Tristan Matthews 07master:7703995a2e87: flac: Remove unused headers [22:39] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:7618d360ca0b: Merge commit '7703995a2e8760e66ff35148bf029e0f9e30dbe6' [23:54] <cone-646> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:e3e051c42c48: libavfilter/af_join: avoid derreferencing possible null [23:54] <cone-646> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:32288234a200: libavfilter/af_channelmap: avoid derreferencing possible null [23:54] <cone-646> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:a9ea79bb4976: libavfilter/af_amix: avoid derreferencing possible null [00:00] --- Sat Sep 27 2014

1 0

ffmpeg.log.20140926
by burek 27 Sep '14

27 Sep '14

[01:55] <kryo_> hey, do i have to re-encode video to use the -g option? [01:55] <kryo_> i'm currently using it with -vcodec copy [01:57] <kryo_> apparently it's doing nothing [01:59] <sacarasc> Yes, you do. [01:59] <sacarasc> It changes frames. [02:00] <kryo_> damn [02:00] <kryo_> don't really want to re-encode [03:03] <MikeJoel> I am still having trouble with converting an swf to mp4. It converts but instead of the whole swf being seen it appears to be enlarging the top left corner of the swf to fill the screen for the mp4... here is what I get when I run ffmpeg http://www.parksandsonspiano.com/Testsite/ffmpeg.html [03:03] <MikeJoel> (the above address is the command prompt output not the video) [03:04] <llogan> Input stream #0:0 frame changed from size:400x400 fmt:yuvj420p to size:195x181 fmt:yuvj420p [03:05] <MikeJoel> huh.. I also tried it by telling it 400x400 still made no difference [03:07] <MikeJoel> any idea how to prevent that? [03:11] <MikeJoel> I just remain the flash to be sure nothing in the video was doing it... still same problem. [03:11] <relaxed> remain? [03:11] <MikeJoel> remade - sorry [03:11] <relaxed> put the swf up somewhere [03:13] <MikeJoel> http://www.parksandsonspiano.com/Testsite/intro.swf [03:15] <relaxed> the first few frames are 400x400 and then it switches the frame size to 195x181 as llogan pointed out [03:18] <MikeJoel> I ffprobe the swf and it says 400x400. When I watch it, it stays 400x400 all the way through [03:20] <relaxed> but it does not. if you have the images and audio and it shouldn't be too hard to create it using ffmpeg [03:21] <MikeJoel> yeah - I was going to switch to that if I couldn't do it... [03:21] <MikeJoel> I am slowing the video down to see each frame [03:21] <relaxed> https://trac.ffmpeg.org/wiki/Create%20a%20video%20slideshow%20from%20images [03:23] <MikeJoel> I just saw it... it looks like it shrinks down to a block in the middle of the screen. [03:23] <MikeJoel> grr [07:08] <kaotiko> hi [07:22] <relaxed> hi [09:50] <wintershade> hi everyone [10:05] <EvolE> hello [10:17] <wintershade> a quick question about libx264 here. I'm doing a ffmpeg -i <input> -c:v libx264 -preset slow -crf 21 -c:a libfdk_aac -vbr 5 <output.mkv> and getting less than 4 FPS on average. is this normal? [10:18] <wintershade> my CPU is an old Intel core2duo, but I remember having higher framerates in the past with the same input video format (DVD VOB, 720x576) [10:18] <wintershade> tia [10:22] <Mavrik> wintershade, yes, that is normal for "slow" preset :) [10:22] <Mavrik> with medium preset you get like 20-40 fps on a modern quad processor [10:24] <wintershade> Mavrik: oh... hmm ok. so how good would it be performance-wise to get an 8-core i7 CPU? [10:24] <wintershade> I'm getting 11 FPS on medium :-/ [10:24] <Mavrik> A proper 600$ Xeon 8-core? [10:24] <Mavrik> or the fake 4-core + hyperthreading i7? [10:24] <flarunt> i think the settings within the presets change over time, so preset slow is probably more cpu intensive now than it was in the past [10:24] <Mavrik> mhm [10:24] <Mavrik> we run medium or faster for real-time encodes mostly [10:25] <wintershade> Mavrik: I don't believe I could get a laptop with a proper 8-core Xeon, at least not with my budget (or, well, at least not until I write a bestseller). [10:26] <wintershade> so, I should just be patient and wait a few days if I want to encode on slow? [10:27] <wintershade> actually, I'm interested about both. anyone got any info on 8-core and fake 4-core+HTs? what would the framerates be? [10:27] <Mavrik> well, yes [10:27] <Mavrik> or just don't use slow :) [10:27] <Mavrik> as I said, I'm getting like 15-30 fps on my i5 2500k [10:28] <wintershade> Mavrik: not using slow is kinda pointless... my VOB source is at around 6-7 mbps bitrate, and the medium with -crf 20 gives it a... 6,5 mbps [10:28] <wintershade> lol [10:29] <flarunt> increase the crf then [10:29] <wintershade> flarunt: wouldn't that degrade video quality? [10:31] <flarunt> of course but if your h264 encode has the same bitrate as mpeg2 source then clearly something is wrong. the bitrate should be a third or less. [10:32] <Mavrik> mhm :) [10:38] <wintershade> flarunt: okay, got any ideas on what might be wrong? [10:42] <wintershade> my current line is ffmpeg -i <input> -c:v libx264 -preset medium -crf 21 -c:a libfdk_aac -vbr 5 -cutoff 20000 -threads 8 test.mkv [10:51] <Mavrik> what do you mean wrong? [10:54] <wintershade> well, flarunt said that something is wrong since I am getting bitrates very similar to the DVD source. [10:55] <flarunt> use a higher crf to force the bitrate down and then check the output [10:56] <reactormonk> How do I capture audio output from another program with ffmpeg via pulse? [10:57] <wintershade> flarunt: well okay, I'll try that. although I'm not the biggest fan of such things. I'll check on other video sources, it might be that the bitrate is going crazy due to a noisy/grainy source. [10:58] <flarunt> it could also explain why you are getting a lower fps than you expected [10:59] <wintershade> it does? [10:59] <wintershade> wow. didn't know that one :-/ [10:59] <wintershade> but it... kinda makes sense. [11:00] <wintershade> but as Mavrik says, my FPS is normal for my CPU and the preset... [11:00] <wintershade> okay, yeah. the video is a bit grainy. using hqdn3d lowers the bitrate down by half. [11:01] <flarunt> when the complexity of the video increases i often see the fps drop, then rise again when not much is happening on screen [11:01] <wintershade> another question. how much does the CPU clock vs. number of cores affect the encoding speed? [11:01] <wintershade> i.e. would it be more worth it to get a 4-core 4GHz CPU or a (true) 8-core 2.4 GHz CPU? [11:01] <wintershade> if anyone has any idea on that... [11:02] <flarunt> it scales pretty linearly i think [11:02] <wintershade> so they'd probably do the same? [11:02] <flarunt> multiply clockspeed by number of cores. bear in mind that each new generation of intel cpus is roughly 10% faster at the same clock speed. [11:03] <wintershade> ok. also... if I may ask. a 4-core Intel or a 4-core AMD? experiences? [11:03] <wintershade> I haven't worked with AMD for a decade now, so I have no idea whatsoever how good they are. [11:03] <flarunt> intel [11:03] <wintershade> ...I thought as much. [11:04] <wintershade> I know Intel rocks with DMBS, but video encoding/processing is something I actually know very very little about. [11:04] <Mavrik> Intel is lightyears ahead, especially for x264 [11:05] <wintershade> cool. I'm asking because I heard some people swear their lives and sould on AMD, but those people are mostly gamers so I reckon it's hardly comparable. [11:05] <flarunt> i have a 2600k and a fx8350. i regret buying the fx8350. [11:05] <wintershade> flarunt: why? [11:06] <flarunt> slow than an i5/i7 and much more power hungry [11:06] <wintershade> okay, two more questions hardware-wise. [11:06] <wintershade> first, I have an NVIDIA graphics card. can I use it to speed up my encoding? or is x264 doing it already? [11:06] <Mavrik> no [11:06] <wintershade> Mavrik: no for latter, or for both? [11:07] <Mavrik> x264 has some limited support but it'll trash your quality [11:07] <Mavrik> GPUs aren't really good for quality encodes [11:07] <wintershade> Mavrik: gotcha. so I should disable the opencl extensions? [11:07] <Mavrik> I didn't notice any difference either way [11:08] <flarunt> http://mewiki.project357.com/wiki/X264_Settings#opencl [11:09] <wintershade> flarunt: ok thanks for that bit of info. much appreciated! [11:09] <wintershade> second question. I'm currently constructing a Gentoo-based cluster from three old PCs laying around back home. they would run a 32-bit OS, since two out of three PCs are 32-bit. the average speed of each CPU is 2GHz. would that be any good for x264? Does x264 even support cluster-based processing? [11:10] <wintershade> (I'm asking because I've googled around for this a bit, but didn't find anyhting official/useful) [11:11] <flarunt> http://forum.doom9.org/showthread.php?t=127611 [11:11] <flarunt> not sure if that is 2pass or crf [11:12] <wintershade> hmm. this sounds good! [11:12] <wintershade> I hope I could get something like that for Linux, though... [11:13] <wintershade> okay, I have to run now. thanks everyone for your help! this was very useful. cheerio~ [11:14] <Mavrik> wintershade, I would advise against it [11:14] <Mavrik> you'll get bad quality if you try to slice up your videos for pretty much no gain [11:14] <Mavrik> either use the machines to each encode their own video or just throw them into a trash bin and get a single powerful encoder... way cheaper (due to power use difference) in the long run [12:47] <timothy> hi, I have a little problem with new ffmpeg (2.4.1) [12:49] <timothy> ok [12:50] <timothy> http://pastie.org/pastes/9596524/text?key=pkmzeb8mt82ahzpfqwmg7a [12:51] <timothy> the error is: "Error while filtering." with 2.3.x it used to work [12:56] <timothy> sacarasc: and enabling debugging / logging does not help [12:58] <sacarasc> That filter is beyond me, someone might come along who understands, though. [12:58] <timothy> ok, i can try to find which filter part does not work [13:00] <relaxed> timothy: it's recommended to use -filter_complex with multiple inputs now [13:02] <relaxed> https://www.ffmpeg.org/ffmpeg-filters.html#Examples-50 [13:04] <timothy> overlay=10 so 10 is the second input? [13:05] <relaxed> yes [13:08] <timothy> '[0:v]scale=iw*min(1920/iw\,1080/ih):ih*min(1920/iw\,1080/ih), pad=1920:1080:(1920-iw*min(1920/iw\,1080/ih))/2:(1080-ih*min(1920/iw\,1080/ih))/2 [scale]; [scale][1:v] overlay=main_w-overlay_w-10:main_h-overlay_h-10 [out]' [13:08] <timothy> so something like that? [13:08] <timothy> yes, there is something wrong [13:11] <timothy> '[0:v]scale=iw*min(1920/iw\,1080/ih):ih*min(1920/iw\,1080/ih), pad=1920:1080:(1920-iw*min(1920/iw\,1080/ih))/2:(1080-ih*min(1920/iw\,1080/ih))/2 [scale]; [scale][1:v] overlay=main_w-overlay_w-10:main_h-overlay_h-10' [13:11] <timothy> it seems correct now [13:11] <timothy> thx relaxed [15:14] <chutchut> hey all [15:14] <chutchut> is it possible to stream RTSP over SSL? [16:51] <wintershade> hi everyone [16:54] <Fjorgynn> hi me [17:06] <chutchut> hey guys X) [17:10] <Fjorgynn> :o [17:16] <chutchut> is it possible to stream RTSP over SSL? [17:18] <Fjorgynn> maybe [17:45] <chutchut> know any links/resources Fjorgynn ? X) [17:50] <mjuszczak> Using a build of ffmpeg 2.4 off of a PPA but it doesn't seem to have qt-faststart. Is that because it's 2.4 or is that because the PPA didn't enable it? [17:51] <chutchut> think it comes in a separate (utils) package mjuszczak [17:52] <chutchut> alternatively there is https://github.com/danielgtaylor/qtfaststart which is a python version mjuszczak [17:53] <mjuszczak> oh, thank you! [17:54] <chutchut> apt-get install libav-tools (un ubuntu) i believe for the C version mjuszczak [17:54] <mjuszczak> thanks! [17:55] <timothy> qt-faststart does not work with mp4, right? [17:55] <sacarasc> mjuszczak: It's not needed any more. [17:55] <mjuszczak> sacarasc: what do you mean? [17:55] <chutchut> think it does timothy? why not sacarasc ? [17:56] <sacarasc> ffmpeg -i blah -movflags faststart blah.mp4 [17:56] <sacarasc> That will do it. [17:56] <chutchut> ahh yea its built-in [17:56] <timothy> also with -vc copy? [17:56] <timothy> I need it too :P [17:56] <chutchut> but for screwy files from other sources, qt-faststart is still useful [17:56] <sacarasc> timothy: It should work, yes. [17:56] <timothy> ok, thanks [17:57] <sacarasc> Also, chutchut, getting people to install a fork of something in the original channel is fun! [17:57] Action: sacarasc disappears into the ether. [17:58] <chutchut> lol sorry sacarasc, wasnt 'getting' anyone to do anything though.. Purely useful advice X) [17:58] <chutchut> not good to be blind or narrow-minded eh X) [17:59] <chutchut> again, sorry for being informative, my bad X [18:27] <quebre> does any of you know how can i encode my .avi movie to MPEG-1 ? so then i can record movie on CD and play it on my DVD player... [18:33] <Mavrik> MPEG-1? O.o [18:34] <Mavrik> quebre, shouldn't ffmpeg -i <file>.avi -target ntsc-dvd -q:a 0 -q:v 0 output.mpg work? [18:34] <Mavrik> or film-dvd or pal-dvd? [18:34] <quebre> but why dvd ? [18:35] <quebre> i only have CD discs, so trying to create VCD format and then burn it [18:35] <quebre> but it's magic for me [18:35] <Mavrik> DVD describes a format [18:35] <flarunt> it doesnt really matter if you burn the output onto cd or dvd.. your authoring software should be able to make an SVCD which is mpeg2 [18:35] <c_14> use -target vcd [18:35] <c_14> or pal-vcd or ntsc-vcd [18:35] <quebre> ok [18:35] <c_14> You'll still have to author it though. [18:35] <quebre> pal-vcd i think.. [18:35] <quebre> author ? [18:36] <flarunt> you cant just put the mpeg on the disc, it has to be burnt in a certain way [18:36] <quebre> oh, hmm can you recommend me some software that can do it ?.. [18:37] <c_14> Have never used any, you can probably just search for vcd authoring software on the internet or vcd in your favorite package manager. [18:37] <flarunt> i think you can use something like VCDGear to create an image to burn [18:37] <flarunt> but software like nero etc should be able to do it anyway [18:38] <quebre> alright ! [18:38] <quebre> thank you [18:38] <quebre> if i do the -target pal-vcd then ffmpeg will choose MPEG-1 or MPEG-2 ? i readed on site that MPEG-1 is best choice as most dvd players will play it [18:41] <c_14> mpeg1video [18:56] <quebre> that did not ended well... [18:56] <quebre> ffmpeg -i ocean.avi -target pal-svcd ocean.mpeg [18:56] <quebre> input size is 698MB output is 1.47gb [18:56] <quebre> that wont fit on CD... [18:57] <c_14> not svcd, vcd [18:57] <quebre> hmmm [18:57] <quebre> but is it not possible to convert to svcd without making the output bigger than input ? [18:58] <c_14> svcd uses a bitrate of 2040k [18:59] <c_14> If that is higher than the input bitrate, it will get bigger. [19:00] <flarunt> a 700mb mpeg4 video has a greater level of compression, so your mpeg1 vcd will be larger and i guess youll have to split it over multiple cds. but really your dvd player should be able to play dvds! [19:02] <quebre> it is able.. but im out of DVD cds ... :( [19:03] <quebre> but ! [19:03] <quebre> i just read on front of my DVD player 'MPEG4' supported [19:03] <quebre> so i can use mpeg-4 for svcd for better compression ? [19:04] <flarunt> you should be able to just burn a normal data disc with the file on it [19:04] <flarunt> assuming your .avi is mpeg4 [19:06] <quebre> ffmprobe says it's mpeg4 [19:06] <quebre> ;) [19:42] <zooptwo> Hello, I am having a problem converting a file to pcm_s16le It is forcing a metadata of encoder=Lavf54.63.104 This I am sure is normally not a problem, but as I am doing this for a phone system it does nto support this and fails to play the file. [19:42] <zooptwo> Any Idea what I could do? [19:43] <c_14> Wait, what. [19:43] <c_14> The phone reads the Encoder metadata tag and then decides not to play the file? [19:43] <zooptwo> I is unable to read it and fails. [19:43] <zooptwo> it thinks that it is not a valid audio file. [19:43] <zooptwo> This is Asterisk 1.8 by the way. [19:44] <zooptwo> Sure one minute. [19:45] <quebre> can i tell ffmpeg to scale any-size video to max 700MB ? [19:45] <flarunt> quebre: yes but I think you'll have to do a 2-pass encode rather than crf [19:46] <c_14> use 2-pass cbr with (video bitrate * length) + (audio bitrate * length) < 700mb [19:46] <quebre> ugh [19:46] <quebre> how would it look in command line ?... [19:47] <flarunt> or pick a suitably high crf and hope for the best.. [19:48] <c_14> https://trac.ffmpeg.org/wiki/Encode/MPEG-4 [19:50] <zooptwo> http://pastebin.com/U7FA6kBi [19:53] <llogan> zooptwo: could you try using ffmpeg code from current git master? [19:53] <llogan> you can just configure and make, no need to install [19:54] <zooptwo> I can try [19:54] <llogan> a bare configure should be fine in this case [19:54] <zooptwo> FYI this E-mail list thread seemed to try to be doing the same thing. [19:54] <flarunt> perhaps you need to output to raw pcm instead of wav? [19:54] <zooptwo> http://ffmpeg.org/pipermail/ffmpeg-user/2011-June/001365.html [19:55] <flarunt> http://stackoverflow.com/questions/4854513/can-ffmpeg-convert-audio-to-raw-… [20:07] <zooptwo> Because of other limitations not related to asterisk itself. a .wav file is required. [20:07] <zooptwo> I have tried using a pcm file and even just renameing it to .wav but no love. [20:11] <flarunt> its a long shot but have a look at RIFFStrip.. http://mark0.net/soft-riffstrip-e.html [20:11] <llogan> first step is to try with git master [20:31] <zooptwo> llogan, it would appear that it does the same thing. [20:32] <zooptwo> This is the right get source right? git://source.ffmpeg.org/ffmpeg.git [20:34] <zooptwo> flarunt, Thanks I will look into that. I think that the RIFF is actually important though. Maybe I am wrong about this meta tag then. [20:34] <zooptwo> I was just looking for difference in the audio files and that is the only thing I could find different. [20:34] <zooptwo> I created the audio file first with ffmpeg and asterisk does not like it. I use sox on the from created from ffmpeg and then it works. [20:35] <zooptwo> the difference I saw was just the metadata. So I could be wrong. It could be something else that is making it unhappy. [20:35] <llogan> zooptwo: yes, that's the correct repo. thanks for testing. [20:37] <zooptwo> I need ot get some sleep. Thanks for the help. [21:08] <darkfrog> am I supposed to use avconv instead of ffmpeg now? [21:09] <llogan> darkfrog: you can use whatever you want. we only support ffmpeg from FFmpeg here [21:10] <darkfrog> llogan: what's the relationship between avconv and ffmpeg? [21:10] <llogan> avconv is made by a fork of FFmpeg called Libav [21:11] <darkfrog> llogan: okay...looking for an ffmpeg ppa for Ubuntu 14.04 then... [21:12] <llogan> https://launchpad.net/~mc3man/+archive/ubuntu/trusty-media [21:13] <darkfrog> llogan: thanks [21:15] <darkfrog> I'm writing a web application that needs to be able to do several things: 1.) read video details, 2.) create a screenshot from the video, 3.) transcode the video to h264 [21:15] <darkfrog> I'm assuming ffmpeg is capable of all this via command-line? [21:17] <llogan> yes [21:43] <Arlis> hi [21:47] <kaotiko> hi [22:03] <Arlis> alio [22:05] <Arlis> something is here? [22:05] <Arlis> hi [22:06] <darkfrog> is there a way to transcode a video and resize to a specific width and height (maintaining aspect ratio) and crop the excess? [22:06] <Arlis> ok [22:06] <Arlis> then you can tell me [22:06] <llogan> darkfrog: see the scale and crop filters [22:07] <Arlis> why need \bin\ffprobe.exe [22:07] <llogan> "-vf scale=640:-2" will scale to 640 pixels wide, and -2 will 1) make the height whatever it needs to be to preserve aspect, and 2) will make the height divisible by 2 [22:09] <darkfrog> llogan: what if I have specific dimensions like 240x180 and I want it scale to either width or height and crop the other (centered)? 640:-2 is fine if I know the video is wider than the size I'm scaling too, but that doesn't handle if it's taller. I suppose I can determine this myself by reading the video details via ffprobe, but was hoping ffmpeg had the ability to do this automatically. [22:11] <llogan> darkfrog: sorry, but i'm not quite following you. and i have to go now, but I'll be back in an hour or so. someone else may be able to help until then. [22:12] <darkfrog> llogan: no worries, thanks anyway [22:14] <darkfrog> The question is, if I have a video that is 500x360 I want the resulting video to be 240x180 and horizontally centered with the extra 10 pixels of width cut off (it should scale it to 250x180 and crop five pixels on the left and right) [22:15] <darkfrog> similarly, if I have a video that is 480x400 it should scale to 240x200 and crop 10 pixels above and 10 pixels below [22:22] <Diogo> hi this is the best way to generate multiple ts with different quality http://pastebin.com/iybS6mXn [22:56] <llogan> Diogo: they are all using the same value for the rate control [22:58] <Diogo> I lnow [22:59] <Diogo> But i dont know if this command is efficient or i need to change something [23:00] <llogan> it is not efficient because you could probably just scale once [23:01] <llogan> https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs#Multipleencoding… [23:02] <llogan> unless each output needs to be scaled to a different size. [23:29] <darkfrog> llogan: did you see my explanation? [23:30] <llogan> yes, but i didn't understand. you'll just have to read the docs of the scale and crop filters [23:32] <darkfrog> llogan: man page or is there a better documentation? [23:33] <llogan> man ffmpeg-filters [23:34] <darkfrog> I think I can do what I want by a scale then crop filter [23:35] <darkfrog> I guess I'll have to figure out the values manually [00:00] --- Sat Sep 27 2014

1 0

ffmpeg-devel.log.20140925
by burek 26 Sep '14

26 Sep '14

[00:10] <cone-600> ffmpeg.git 03Martin Storsjö 07master:a2efbecc4ed1: libavformat: Move avc mp4 startcode parsing to a shared file [00:10] <cone-600> ffmpeg.git 03Michael Niedermayer 07master:b8d3f7fc21b9: Merge commit 'a2efbecc4ed12d287cf29856418c4da4a7648d95' [00:16] <jamrial> michaelni: if i change the prototypes for these functions, do i have to do it for every arch? [00:16] <michaelni> yes but it should be trivial for 32bit based archs [00:52] <cone-600> ffmpeg.git 03Thomas Volkert 07master:ddf5fb71ee9c: rtpenc: HEVC/H.265 support [00:52] <cone-600> ffmpeg.git 03Michael Niedermayer 07master:d8ddac363e77: Merge commit 'ddf5fb71ee9c8b2d9a23c0f661a84896cd7050fc' [01:09] <cone-600> ffmpeg.git 03Martin Storsjö 07master:2f172f1ae984: rtsp: Clear the session id on redirects [01:09] <cone-600> ffmpeg.git 03Michael Niedermayer 07master:6441d522e9fe: Merge commit '2f172f1ae984b763c06069adb51d0053b3834f4b' [02:43] <cone-600> ffmpeg.git 03James Almer 07master:c99a88281489: avcodec/idctdsp: change {put,add}_pixels_clamped to ptrdiff_t line_size [02:51] <cone-600> ffmpeg.git 03James Almer 07master:4f4f08e6f00f: x86/idctdsp: port {put,add}_pixels_clamped to yasm [05:43] <cone-600> ffmpeg.git 03Michael Niedermayer 07master:303fc9b36833: avcodec/mpegvideo_enc: write a log message at verbose level if frame re-encoding occurs due to VBV [09:19] <Ko_deZ> ubitux: Thanks for the answer. I saw the time on my post, and realized I had 2 minutes to catch my train. I failed :-/ Anyway, I will look into that, thank you. [09:41] <durandal_1707> how many opw slots we have? [09:43] <ubitux> durandal_1707: afaik we only have one [10:24] <nevcairiel> afaik you have to come up with your own funding if you want more [12:06] <michaelni> ubitux, some companies donate to OPW wthout specifing a project, and their donation is used for whichever applications+applicants+project looks best so if there are alot of applicants wanting to work on ffmpeg and their applications and abilities are great we likely would get more slots, thats at least how i understand it [12:27] <Jul13t> michaelni, I'm working on my proposal for the MPEG, I've not yet been assigned my qualification task yet [12:34] <Daemon404> michaelni, did anything change recently in how we probe images? [12:34] <Daemon404> i can bisect, but now suddenly it doesnt work at all for me. [12:37] <michaelni> Daemon404, yes there where a few changes related to image2/3(pipe) [12:37] <Daemon404> ok [12:37] <Daemon404> pretty much all image probing fails for me now [12:37] <Daemon404> as an API user i mean [12:38] <michaelni> Jul13t, did you ask paul / thilo ? [12:38] <Jul13t> michaelni, he's just given me what I need to to. Thanks :) [12:38] <michaelni> Jul13t, ok great then [12:39] Action: Jul13t doing Thilo's assignment [12:55] <Daemon404> hmm actually its not probing [12:55] <Daemon404> it fails at decode... hmm [12:55] Action: Daemon404 checks [12:59] <J_Darnley> WTF? Does that ALS things really put floating-point and lossless together? [13:00] <Daemon404> dts-hd-ma does too [13:00] <Daemon404> as does lagarith [13:00] <Daemon404> as such, we have an x86 fp emu in the codebase [13:00] <J_Darnley> :O [13:01] <J_Darnley> I knew lagarith did but that's cause it is pants-on-head but the other thing is for bluray isn't it? [13:02] <Daemon404> yes ok the problem is probing... [13:02] <Daemon404> avformat_find_stream_info succeeds, no streams are found or set [13:03] <Daemon404> video_codec_id also not set [13:03] <Daemon404> :| [13:05] <Daemon404> seriously broken [13:05] Action: Daemon404 guess bisect time [13:09] <cone-763> ffmpeg.git 03Martin Storsjö 07master:d07ddc93e29a: avcodec: Fix a doxy comment to refer to the right function [13:09] <cone-763> ffmpeg.git 03Michael Niedermayer 07master:a6fd685fedb1: Merge commit 'd07ddc93e29a4fdf12cd426953a7ecd5abe5e36d' [13:22] <Jul13t> michaelni, while working on my qualification task, is there any format applications should follow apart from the one on the Gnome wiki? [13:23] <Jul13t> I mean perculiar to ffmpeg may be [13:25] <Daemon404> 'format applications' ? [13:30] <J_Darnley> I think that mean guides or examples to be followed, like the gnome one. [13:30] <kierank> possibly Jul13t means code style? [13:30] <Jul13t> Daemon404, i'm talking of the the application format [13:30] <Jul13t> submitting the application to the gnome student app portal [13:31] <Daemon404> oh ok [13:31] <Jul13t> as for the coding styles i think there's a wiki for that on code conventions [13:31] <Daemon404> ffmpeg has no application format [13:31] <Jul13t> and stuff, and it's very clear [13:31] <Daemon404> afaict [13:31] <Jul13t> Daemon404, guess i'll just follow the format given by Gnome, [13:31] Action: Jul13t hopes its safe ;) [13:31] <ubitux> Jul13t: see http://ffmpeg.org/developer.html#Contributing [13:32] <Jul13t> ubitux, thanks [13:32] <Daemon404> michaelni, found the offending commit [13:33] <Daemon404> michaelni, b3fd2b175c90b4766034095e74e1f5112b1547ef breaks all image stream probing [13:37] <michaelni> Daemon404, how can i reproduce it ? [13:37] <Daemon404> avformat_open_input(&ctx->afctx, file_path, in_format, NULL); [13:37] <Daemon404> avformat_find_stream_info(ctx->afctx, NULL); [13:38] <Daemon404> then inspect the avformatcontext [13:38] <Daemon404> in_format here is NULL [13:39] <michaelni> do you have a testfile & command line ? [13:39] <Daemon404> command line is no applicate [13:39] <Daemon404> it's API. [13:39] <Daemon404> not* [13:39] <Daemon404> ffmpeg.c doesnt reproduce it. you only broke it for API users. [13:40] <Daemon404> http://chromashift.org/img/ef1.d2v_001_0.png is teh url i opened [13:40] <Daemon404> to test [13:40] <Daemon404> but any will fail. [13:40] <michaelni> why does it differ between ffmpeg and API users ? [13:41] <Daemon404> because ffmpeg has a bunch of hacks [13:41] <Daemon404> very liekly [13:41] <Daemon404> its not my problem to investigate ffmpeg.c [13:42] <Daemon404> i can certainly write a tiny test c file and submit it, but it will only be a few api calls. [13:44] <michaelni> "./demuxing_decoding ef1.d2v_001_0.png video audio" works too [13:45] <cone-763> ffmpeg.git 03Katerina Barone-Adesi 07master:28f5cd312c9d: fate: Switch ra4-288 test from framecrc() to pcm() [13:45] <Daemon404> unsurprising as it is mostly copypasted [13:45] <Daemon404> i will write a test file. [13:45] <cone-763> ffmpeg.git 03Michael Niedermayer 07master:9b53691f7318: Merge commit '28f5cd312c9da9072108edf8b7685d009374ea96' [13:46] <michaelni> Daemon404, thx [13:51] <cone-763> ffmpeg.git 03Gabriel Dume 07master:95d312d6c82e: concat: K&R formatting cosmetics [13:51] <cone-763> ffmpeg.git 03Michael Niedermayer 07master:8f3d3e5f8373: Merge commit '95d312d6c82e7485f43c3ddda0f369af0e41c83b' [14:00] <cone-763> ffmpeg.git 03Gabriel Dume 07master:86a361081d3e: lmlm4: K&R formatting cosmetics [14:00] <cone-763> ffmpeg.git 03Michael Niedermayer 07master:59f49667569b: Merge commit '86a361081d3eb4e999d75bc2a462f01d88cbd2e5' [14:08] <cone-763> ffmpeg.git 03Gabriel Dume 07master:4620affa24ee: m4vdec: K&R formatting cosmetics [14:08] <cone-763> ffmpeg.git 03Michael Niedermayer 07master:454f98b881fc: Merge commit '4620affa24eedb167482198aa04126bcedd05620' [14:24] <Daemon404> michaelni, msg'd you a code test [14:24] <Daemon404> reverting b3fd2b175c90b4766034095e74e1f5112b1547ef causes the 'good' output. [15:31] <michaelni> Daemon404, fixed [15:31] <cone-763> ffmpeg.git 03Michael Niedermayer 07master:1dbdcb4a8c76: avformat/img2dec: fix error code at EOF for pipes [15:33] <Daemon404> michaelni, thanks! [15:54] <Daemon404> michaelni, you missed one EOF [15:55] <Daemon404> - return AVERROR(EIO); /* signal EOF */ [15:55] <Daemon404> + return AVERROR_EOF; /* signal EOF */ [15:55] <Daemon404> ^^ [15:58] <Daemon404> michaelni, ping after that one is fixed because there is another bug which is not related... also in img2... [16:09] <Daemon404> michaelni, ok so attempting to seek first (to see if it is a seekable file) leaves it in a fucked up state [16:10] <Daemon404> seeking to 0 causes av_read_image to never ready any packets ever again [16:10] <Daemon404> this i only after b3fd2b175c90b4766034095e74e1f5112b1547ef [16:11] <Daemon404> i dont see how this can possibly be correct behavior [16:22] <Daemon404> afctx->pb->seekable is true too [16:22] <Daemon404> sorry but this fix just seems buggy as fuck.. [16:28] <cone-763> ffmpeg.git 03Michael Niedermayer 07master:2497914a1846: avformat/img2dec: pass error code and signal EOF [17:48] <cone-763> ffmpeg.git 03Michael Niedermayer 07master:6d35aba167a2: avformat/img2dec: initialize pkt->pos for image pipes [17:48] <cone-763> ffmpeg.git 03Michael Niedermayer 07master:ce6e46be7229: avformat/img2dec: enable generic seeking for image pipes [17:57] <michaelni> Daemon404, maybe fixed, can you find more issues in it ? [17:57] <ubitux> "please hurt me" [17:57] <Daemon404> lol [17:57] <Daemon404> let me try. [17:58] <Daemon404> michaelni, nope it seems to work now [18:01] <Daemon404> and many thanks of course [18:02] <reynaldo> guess no one would object removing the "copied straight from gsoc" banner at https://trac.ffmpeg.org/wiki/SponsoringPrograms/OPW/2014-12 ? [18:02] <reynaldo> michaelni ^ [18:03] <michaelni> reynaldo, sure, remove it [18:04] <michaelni> but then IMHO it doesnt matter, we could also leave it ;) [18:04] <reynaldo> ppl will actually look at that for reference, I think it matters [18:05] <reynaldo> the amount of info is overwelming enough to make it start with "this shite is c&p-ed crap, info needs to be updated" [18:05] <reynaldo> 8) [18:09] <reynaldo> michaelni: is anyone chasing mentors/backups to update that page already? Other than the discussion on MPEG ALS I see nothing more on the ml [18:14] <michaelni> reynaldo, no idea, i just ask on a "as needed" basis ATM [18:15] <cone-763> ffmpeg.git 03Thilo Borgmann 07master:e113692c899f: lavu/ffmpeg_opt: Check return value of avcodec_find_encoder(). [18:48] <kierank> michaelni: where did all the opw money come from [19:20] <wm4> kierank: *chirping crickets* [19:46] <reynaldo> llogan: there are two msgs from me to ffmpeg-devel@ been held due to a too-many-recipients situation, please drop the first one and let the second pass [19:46] <reynaldo> I CCed the mentors whos emails I have on file, that's why they got held [19:58] <michaelni> kierank, see reynaldos mail [20:00] <kierank> ah [20:04] <wm4> if someone picks up the vdpau opw task, I could probably provide some guidance for that [20:07] <llogan> reynaldo: i can't access mailman right now. it's 404. [20:07] <reynaldo> llogan: noticed, tried to drop the first one by myself but went 404 too [20:08] Action: llogan attempts to summon beastd [21:02] <cone-763> ffmpeg.git 03James Almer 07master:4b892e469bc9: x86/cavsdsp: fix buffer alignment in cavs_idct8_add_mmx() [21:02] <cone-763> ffmpeg.git 03James Almer 07master:a829870b2f4d: avcodec/svq1enc: align buffer used by simd functions [21:36] <wm4> haha [21:37] <wm4> another image opening regression [21:37] <wm4> what's even going on [21:39] <Compn> blame carl :P [21:39] <Compn> ehe [21:48] <gnafu> "But Caaaaaarl, that /kills/ people." [22:20] <cone-763> ffmpeg.git 03Carl Eugen Hoyos 07master:c0f9df30dd02: lavc/x86/idctdsp.h: Fix make checkheaders. [22:27] <ubitux> Andreas hasn't packaged 2.4.1? [22:27] <ubitux> or maybe that's what the warning is about [22:51] <ubitux> https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel/ ? [22:51] <ubitux> something went wrong? [23:04] <bryno> but... where's the mail? :( [23:06] <wm4> the ML got accidentally deleted! [23:22] <J_Darnley> wm4 what? [23:58] <cone-763> ffmpeg.git 03Michael Niedermayer 07master:a43bcaafe20d: avcodec/dxtory: Support dxtory-2.0.127.avi [00:00] --- Fri Sep 26 2014

1 0

ffmpeg.log.20140925
by burek 26 Sep '14

26 Sep '14

[00:15] <relaxed> MikeJoel: http://johnvansickle.com/ffmpeg/ [00:35] <w00ds> howdy [00:36] <w00ds> think this be a good format to convert mkv/h.264 to mp4/h.265? [00:36] <w00ds> ffmpeg -i INPUTFILE.MKV -y -c:v libx265 -c:a copy -preset ultrafast -qp 0 OUTPUTFILE.mp4 [00:38] <c_14> It'll work, if that's what you're asking. [00:38] <c_14> Not sure if it'll be any smaller than the h.264 file though, also the quality won't be better. [00:39] <microchip_> wodim: x265 is still not mature yet so don't expect wonders [00:40] <wodim> what? [00:40] <w00ds> c_14: will it be ALOT worse quality you think? [00:40] <w00ds> it'll definitely do a MUCH smaller file size though [00:40] <microchip_> wodim: sorry, meant w00ds [00:41] <w00ds> i factored it out, will be closer to like 2.3GB output file compared to source file 11.5GB [00:41] Action: c_14 hasn't done a lot of H.265 encoding, but you probably won't see the difference. [00:41] <w00ds> c_14: thats awesome, thanks [00:41] <w00ds> microchip_: it seems to be pretty awesome though [00:41] <w00ds> especially consider Plex Server is able to read it ... [00:42] <w00ds> here's a question though ... [00:42] <c_14> It'll burn your cpu. [00:42] <microchip_> w00ds: you think now it's awesome? Wait until it's ready. [00:42] <w00ds> if i convert this file to be for example 2.3GB file size using h.265 ... when it transcodes to work on a non-compliant device, will it transcode it back to being 11.5GB? [00:43] <w00ds> or just convert it to be h.264 at the size 2.6GB still? [00:43] <c_14> That depends on what plex does. [00:43] <w00ds> good point ... [00:43] <w00ds> wondering if converting my stuff to x265 would be worth it [00:43] <Nosomy|off> hevc still is experimental [00:44] <w00ds> well im thinking in a better remote streaming outcome ... [00:44] <w00ds> a client would be able to download a 2.3GB file faster than a 11.5GB file [00:44] <c_14> You also have to take into account if that client can then play the H.265 file. [00:44] <w00ds> err not faster ... but keep up with a movie that is a smaller size. [00:44] <w00ds> there are no clients that play h.265 (aside from PC) [00:45] <w00ds> just don't know if it converts it back to 11.5GB before it streams to a client [00:45] <c_14> Then they will at some point have to encode to H.264 (or something else) which will burn cpu and increase the file size (or ruin the quality). [00:45] <w00ds> so the transcoding back to h.264 will jump the size back up to 11.5GB? [00:46] <w00ds> in theory. [00:46] <Nosomy> lol [00:46] <c_14> Depends on how the source was encoded and how it's encoded back, but it could. [00:46] <Nosomy> lossy is always lossy... [00:46] <w00ds> hmm [00:46] <c_14> And the quality will be worse, it might not be _visually_ worse, but it'll be worse. [00:46] <w00ds> true [00:47] <w00ds> well im trying it out ... see how it works out [00:47] <w00ds> because then a lot of homes could easily remote stream from home with their 10mbt upload speed [00:47] <w00ds> not needing a full 100mbit/1gbit to get fast enough speed ... [00:47] <c_14> If they could encode H.265. [00:47] <c_14> In a reasonable time frame. [00:47] <w00ds> decode you mean? [00:47] <w00ds> the client decode... [00:48] <c_14> you were talking about upload speed, the person who uploads needs to encode [00:48] <w00ds> well if it was already encoded in h.265 [00:48] <w00ds> be quicker to send 2.3GB than 11.5GB ... [00:48] <c_14> Ye, you'll need to check if the clients can decode H.265 in realtime though. [00:49] <c_14> Especially at higher resolutions. [00:49] <w00ds> true, when a h.265 client is out [00:49] <w00ds> heh [00:49] <c_14> Try it out though and see if it works. [00:49] <w00ds> will do [00:49] <w00ds> im also going to try out plex's "cloud sync" feature [00:49] <c_14> The main reason I haven't bothered that much with it is because it takes forever to encode. [00:50] <w00ds> which transcodes video to google drive for example [00:50] <w00ds> c_14: ya it does take a long time if you don't do ultrafast [00:50] <c_14> I can't remember what settings I was using, but afaik I calculated 20 or so days for a 2 hour 1080p bluray rip. [00:51] <w00ds> weee ... 76 degrees celsius temperature for my CPU [00:51] <w00ds> lol [00:51] <c_14> That was 4 or 5 months ago. [00:51] <w00ds> http://i.imgur.com/MVDTdyV.png [00:51] <c_14> yep, and now it'll stay like that. For a while. [00:53] <w00ds> aren't there encoding/transcoding companies? [00:53] <c_14> Companies that'll encode video for you? [00:53] <w00ds> ya [00:54] <c_14> No clue. There might be. But tbh all you need to do is buy/rent a bunch of cpu time. [00:54] <c_14> But I like having control of my encodes. [00:55] <w00ds> tue true [00:55] <w00ds> $26/month http://myskyhost.com/NL-Encoding-RDP.html [00:57] <w00ds> maybe cheaper just get a kimsufi box [00:57] <w00ds> heh [00:58] <c_14> I usually just use one of my computers. [00:58] <c_14> I have a couple standing around so if one of them is busy it doesn't bother me that much. [00:58] <c_14> Plus it makes for a great heater. [00:58] <w00ds> ya true. [00:59] <w00ds> would more cpus be better or more threads? [00:59] <w00ds> example, 4c/8threads @ 2.6gighz or 4c/4threads @ 3.1gighz [01:00] <c_14> Probably more cores. [01:00] <c_14> Hyperthreading is mainly for desktop loads. [01:00] <c_14> ie doing lots of things at once [01:01] <c_14> Ie lots of short-running threads while encoding is several long-running threads. [01:01] <w00ds> so faster cores will matter more than threads, k got it. [01:12] <vmBenLubar> Does -c:v libvpx-vp9 -lossless 1 work? It seems to wash out the colors, at least when I play it back in vlc. [02:04] <kryo_> hey guys [02:05] <kryo_> i'm trying to encode an MP4-TS as flv and send it to a rtmp:// url [02:05] <kryo_> w21 [02:07] <kryo_> any ideas on how to get ffmpeg t odo this? [02:10] <c_14> ffmpeg -i file -c copy -f flv rtmp://foobar [02:10] <c_14> And the word you are looking for is remux, not encode. Unless you actually want to change the audio/video data and not just the format. [02:13] <kryo_> ah sweet [02:13] <kryo_> remux is even better [02:13] <kryo_> thought i might need to encode cause of the TS format [02:23] <c_14> Not sure you have to, but never streamed to rtmp either. [02:28] <kryo_> c_14: one more question [02:28] <kryo_> can i make it loop the video? [02:29] <c_14> Not ffmpeg-internally. You said the input file was mpeg-ts, right? [02:29] <c_14> If it is, you can just use: `while true; do cat file.ts; done | ffmpeg -i - -c copy -f flv rtmp://foobar' [02:31] <kryo_> looks good to me [02:31] <kryo_> thanks a lot [02:38] <Riviera> I hardcoded subtitles by using "-vf subtitles=file.srt", yet could not find an easy way of specifying the font size (I want a larger font). Is there any? [02:44] <c_14> Short of converting the srt to ass and changing the fontsize there, you can mess around with the original_size setting. [02:44] <c_14> Though, tbh. Just convert to ass and change the fontsize. [02:52] <Riviera> c_14: hm, funny hack, i'll try that; thanks :) [02:54] <circ-user-iNQ84> Hi, I'm building ffmpeg to include libvorbis and libvpx for Android. On configuration I get 'ERROR: libvorbis not found'. How do I tell ffmeg where to find libvorbis? My configuration command is: [02:54] <circ-user-iNQ84> ./configure \ --prefix=${DIR_SYSROOT} --arch=${CPU} --target-os=linux \ --extra-ldflags="-L${DIR_SYSROOT}lib" --extra-cflags="-I${DIR_SYSROOT}include" \ --enable-cross-compile --cross-prefix=${PREFIX}- --sysroot=${DIR_SYSROOT} \ --disable-shared --enable-static --enable-small \ --disable-all --enable-ffmpeg \ --enable-avcodec --enable-avformat --enable-avutil --enable-swresample --enable-avfilter \ [02:54] <c_14> Where is libvorbis installed? [02:55] <circ-user-iNQ84> ${DIR_SYSROOT}/lib/libvorbis.a [02:55] <circ-user-iNQ84> ${DIR_SYSROOT} is the android system root [02:56] <circ-user-iNQ84> I don't know if this means that libvorbis was not found or if it was not built correctly [02:56] <c_14> What does `pkg-config --exists libvorbis' return? [02:56] <circ-user-iNQ84> Nothing [02:57] <c_14> echo $? [02:57] <c_14> The actual return code, not output. [02:57] <circ-user-iNQ84> How do I get that? $$ echo $ [02:57] <circ-user-iNQ84> ? [02:57] <dahat> Are there any known 'newer' RTMP servers that spit out either RTMPE or RTMPS that libRTMP is known not to work with at this time? [02:57] <c_14> `echo $?' [02:58] <circ-user-iNQ84> I mean where should I put that [02:58] <circ-user-iNQ84> it just prints a $ [02:58] <circ-user-iNQ84> Oh! It's 1 [02:59] <c_14> Right, how did you install libvorbis? [02:59] <circ-user-iNQ84> make install [02:59] <circ-user-iNQ84> I looked through thr logs [02:59] <circ-user-iNQ84> It was put in the android sysroot [02:59] <c_14> do you have ${DIR_SYSROOT}/include/vorbis/vorbisenc.h ? [03:00] <circ-user-iNQ84> Yes [03:00] <circ-user-iNQ84> I mean pkg-config shouldnt find libvorbis [03:00] <circ-user-iNQ84> pkg-config is the host machine's pkg-config [03:00] <circ-user-iNQ84> I don't have a pkg-config for android [03:01] <c_14> It doesn't matter anyway, the configure script doesn't use pkg-config for libvorbis. [03:01] <c_14> does ${DIR_SYSROOT} have a trailing slash? [03:01] <circ-user-iNQ84> Yes [03:02] <c_14> aaah [03:02] <circ-user-iNQ84> Should I remove it? [03:02] <c_14> you might need --extra-cxxflags="-I${DIR_SYSROOT}include" [03:02] <c_14> I think libvorbis is c++, not c. [03:02] <circ-user-iNQ84> So I need that for compiling fmpeg or libvorbis? [03:03] <c_14> for compiling ffmpeg [03:03] <c_14> Just add it to above configure line. [03:03] <c_14> *the above [03:04] <circ-user-iNQ84> Nothing [03:04] <circ-user-iNQ84> The same error [03:04] <c_14> Can you pastebin config.log ? [03:05] <circ-user-iNQ84> One sec [03:07] <circ-user-iNQ84> http://pastebin.com/ftuEEK0Q [03:10] <circ-user-iNQ84> There are these lines at the end. [03:11] <circ-user-iNQ84> something about ranlib [03:11] <c_14> yep [03:11] <c_14> Saw them as well, trying to find out why that might be happening. [03:12] <circ-user-iNQ84> Do I need any environment variables setup in order to build? [03:13] <c_14> I'm guessing the libvorbis configure cannot correctly detect the ranlib binary and therefore doesn't generate the indexes. [03:13] <circ-user-iNQ84> http://www.mega-nerd.com/erikd/Blog/CodeHacking/MinGWCross/pkg-config.html [03:13] <circ-user-iNQ84> Is this related? [03:14] <c_14> probably not [03:18] <c_14> You could probably either directly run ranlib on the vorbis libraries, or you could check the Makefile that ./configure (for libvorbis) outputs and checks what it sets RANLIB as, and then make sure it's set correctly so it references the android ranlib [03:20] <circ-user-iNQ84> It's using the wrong ranlib in libvorbis Makefile [03:20] <circ-user-iNQ84> RANLIB = ranlib [03:20] <c_14> You can probably just correct it so it uses android ranlib. [03:20] <circ-user-iNQ84> So, is there an environment variable I can set to fix this when building libvorbis? [03:20] <circ-user-iNQ84> Or should I fix it manually and run make? [03:21] <c_14> I'd fix it manually. [03:21] <c_14> Not sure if there's a variabl. [03:21] <c_14> *variable [03:22] <c_14> Might want to file a bug with libvorbis though, since that would probably be a bug in their configure script. [03:22] <circ-user-iNQ84> Oh! Cool! [03:22] <circ-user-iNQ84> Let me try if this works first. [03:25] <circ-user-iNQ84> Not working. [03:25] <circ-user-iNQ84> I'll check config.log again. [03:26] <circ-user-iNQ84> Let me send a pastebin, it's something about ogg. [03:27] <circ-user-iNQ84> http://pastebin.com/FMdaD9jL [03:29] <c_14> >ogg: no archive symbol table (run ranlib) [03:29] <c_14> looks like the same thing, just for ogg this time [03:29] <circ-user-iNQ84> Oh! Nice! [03:36] <circ-user-iNQ84> Making sure it works. Will be back in a bit. [04:11] <circ-user-iNQ84> Now, I'm having a problem with libvorbis, turns out it's not built correctly too. [04:11] <circ-user-iNQ84> Should I ask about this here? [05:02] <MrJoestar> I'm trying to use " ffmpeg -i input.mkv -filter_complex "[0:v][0:s]overlay[v]" -map [v] -map 0:a <output options> output.mkv ", but I got the message " no matches found: [v]". [06:16] <gcl5cp> Overlay area (0,0)<->(720,480) not within the main area (0,0)<->(480,270) or zero-sized [06:18] <gcl5cp> I'm trying to overlay a greater image than video, overlay=-120:-100, to concatenate a small video with other with more resolution. [06:19] <gcl5cp> is this posible? [06:35] <pentanol> gcl5cp hi, video overlay on the video? [06:36] <gcl5cp> ffmpeg -i w.png -i v.mp4 [06:40] <gcl5cp> i need a 720x480 video. Original is 480x270, y don't want use -s, will be "pixeled". so i want to use a png watermark(border frame) and insert(overlay) video in center. [06:41] <pentanol> you must use scale [06:45] <pentanol> format=rgba,scale=720x480 [06:53] <gcl5cp> so, it is not possible [06:55] <gcl5cp> preserve original(low) resolution [07:12] <pentanol> as i remember in ffmpeg swscaler make expand the picture [07:15] <pentanol> ffmpeg -i w.png -vf format=rgba,scale=720x480 -i v.mp4 [07:15] <pentanol> pardon, ffmpeg -i w.png -vf format=rgba,scale=720:480 -i v.mp4 [08:38] <kryo_> hey i want to remux an mpeg-ts to flv and play it back to a rtmp:// address [08:38] <kryo_> can i do that with ffmpeg? [08:39] <pentanol> sure [08:39] <kryo_> i'm using -c copy [08:39] <kryo_> but it goes much too fast... :P [08:40] <kryo_> how to slow down? XD [08:44] <pentanol> -c ? for subtitles? [08:44] <pentanol> need to go... [08:44] <kryo_> lol [09:00] <EvolE> xD [09:01] <EvolE> kryo_: what goes too fast? [09:01] <kryo_> ffmpeg is sending the stream over rtmp [09:01] <kryo_> and it's going at 2500+ fps [09:02] <kryo_> i believe i need to to go 30 fps [09:04] <EvolE> that's strange. doesn't it recognize source fps ? [09:05] <kryo_> i think it's essentially just remuxing the video and sending it over rtmp asap [09:05] <kryo_> setting -r 30 does nothing o_O [09:05] <EvolE> hmm... maybe you are right. did you try -re option? [09:06] <EvolE> before the input [09:09] <kryo_> well that work excellent, thanks! [09:09] <kryo_> however, the audio isn't being sent o_O [09:09] <kryo_> ffmpeg -re -i fil.mp4 -bsf:a aac_adtstoasc -acodec copy -c copy -f flv rtmp:// [09:15] <EvolE> if you are using "-c copy" then you probably don't need -acodec copy [09:16] <EvolE> or you can use both "-acodec copy" and "-vcodec copy" [09:16] <EvolE> but i'm not sure if it's the case why audio isn't sent [09:16] <kryo_> the audio is still detected [09:17] <kryo_> maybe it's being sent in the wrong format [09:17] <EvolE> maybe. [09:19] <kryo_> ha just had to use -acodec libmp3lame [09:21] <kryo_> thanks EvolE :D [09:22] <EvolE> kryo_: np, good it helped) [14:44] <csepulvedab> hi guys! [14:47] <csepulvedab> guys, you know how i can transcode a video using the same x264 option uses in another video? [14:47] <csepulvedab> there is an option for this or i must parse the output of ffprobe? [16:23] <McSalty> Hi, I would like to use a stream from an ipcam (http://x.x.x.x/mjpg/video.mjpg) and map it to a video loopback device (via v4l2loopback) so that I can access the stream via an application requiring a valid webcam. I am only allowed to use ffmpeg directly (no wrappers or other tools). How can I do this? [16:23] <McSalty> I already have a virtual device and I can pipe a video to it, but the stream doesn't work yet. [16:24] <disconnected> hey, I am compressing mp4 files using: ffmpeg -y -i input_file -qscale 31 -vcodec libxvid -acodec copy output_file. Why does this create bigger files sometimes? For 7.4 MB file it created 36.5MB output file and for file 42.5MB -> 15.1MB. Is there any property that determines if it produces bigger file or not? [16:42] <klaxa|work> disconnected: the bitrate/quality of the input files? [16:42] <klaxa|work> those feel like properties that will determine whether or not the resulting file (with constant quality) will be smaller or bigger [16:43] <disconnected> from medianfo, the overall bit rate is 184 Kbps (with variable bit rate mode), the video bit rate is 55.6 Kbps [16:44] <disconnected> the created bigger file has 771 Kbps video bit rate with Constant bit rate mode [16:44] <disconnected> is there any universal command that would just create less quality file somehow? [16:44] <disconnected> (or rather universal parameters...) [16:45] <klaxa|work> i don't think so, you can parse mediainfo or ffprobe and generate command line parameters from that [16:45] <disconnected> if not - how can I calculate/is this defined somehow what is the threshold above which the generated file will be bigger? [16:49] <disconnected> (btw. the input file is marked by the ffmpeg as: Stream: Video h264 (Baseline) (avc1), yuv420p(tv), <here resolution which varies - 1280x720 and 1920x1080), here probably bitrate? which varies: 1874 kb/s and 55kb/s [16:53] <wintershade> hi everyone [16:57] <McSalty> Hi guys, I'm getting closer tot he solution. Right now the only problem is that the output format (video4linux2) is not suitable for video1 [16:57] <McSalty> ffmpeg -i http://username:password@x.x.x.x/mjpg/video.mjpg -f video4linux2 /dev/video1 [17:01] <ubitux> what version of ffmpeg? [17:02] <ubitux> McSalty: ah i forgot to add the video4linux2 alias [17:02] <ubitux> McSalty: use -f v4l2 [17:05] <McSalty> ubitux, version 0.8.16-6:0.8.16-1 it doesn't work with -f v4l2 [17:05] <ubitux> you're probably not using ffmpeg [17:05] <ubitux> we are in 2.4 btw, just upgrade [17:06] <McSalty> that's the problem, I'm not allowed to upgrade. I have to stick to this version. If it keeps failing, I'll have to suggest to upgrade [17:06] <wintershade> McSalty: you can just download the binary, extract it somewhere in your /home directory and run it from there [17:07] <wintershade> McSalty: which OS are you using anyway? [17:07] <McSalty> wintershade, the system is running crunchbang [17:08] <McSalty> wintershade, so I don't have to load new modules. then I guess I will just download the binary and try that first. thanks a lot, I'll try it right away [17:08] <wintershade> McSalty: odd. can you paste(bin) the console output when you just type "ffmpeg" in it? tia [17:11] <McSalty> wintershade, sure: http://pastebin.com/tg79eBdi [17:11] <wintershade> McSalty: it *could* work. Linux 2ndGen package management system is there to make things easier for you, but it's not always the only way - just the most convenient one for most users. OTOH, I know people who always install just the bare bones of their Linux distro, the last thing being X and perhaps WindowMaker or OpenBox. All the other applications, they just download the binary in .tgz/.rpm/.deb/.whatever, extract it somewhere in [17:11] <wintershade> their /home and run it from there. [17:11] <wintershade> McSalty: whoa, you really do have 0.18.16 version of ffmpeg. how old is that??? [17:12] <wintershade> and I thought my Gentoo stable has some old packages... [17:13] <McSalty> wintershade, ha, I know now, I didn't realize that. is it really that old? I'll just try the updated version, things will probably be much better:) [17:14] <wintershade> McSalty: as ubitux said, the current branch is 2.4. that's way above 0 :D I've just updated to 2.3.3 on my Gentoo the other day. will probably return to 2.2, until guys at oDesk decide to compile their oDeskTeam against newer ffmpeg libs. [17:41] <bluepr0> hello! Im trying to find out if its possible with ffmpeg to start transcoding on a certain time position for a HTTP stream. For example an stream given by peerflix [17:54] <McSalty> wintershade, I've sent you a pm [17:55] <McSalty> wintershate, I've sent you a pm [18:19] <gcl5cp> thank pentanol, "scale=720:480" but i am looking for enlarge resolution without scale video, is like add a frame(border) to a photo making canvas bigger but same image resolution. [19:44] <alexblzn> hello, is it possible to convert XESC to AVI format with ffmpeg? [19:50] <sacarasc> From the ffmpeg -codecs output, I would say maybe. [19:52] <alexblzn> I'm taking a look in the list. [19:53] <alexblzn> Thanks. [21:19] <sheshkovsky> Hello guys, I'm trying to build an application to burn subtitles on videos. The specific task to do is burning Persian (Right-to-left language like arabic) subtitles. But I have problem with it. When I got several words simuntaneously without any Enter between them, the ffmpeg push the lines to up. for example if I have three lines, the third lines comes on first line, then the second line, and then the first line prints on the [21:19] <sheshkovsky> Can anyone help me with this problem? [21:33] <sheshkovsky> I've posted a question on stackoverflow about my problem, if anyone could help me please send the answer there. Thanks. Here's the link: http://stackoverflow.com/questions/26046586/how-to-fix-ffmpeg-mencoder-push… [21:35] <benlieb> of [21:35] <benlieb> How do I cut a section of movie between two time stamps? [21:35] <benlieb> it seems like -t and -to both represent durations... [21:36] <c_14> You want the middle part or you don't want the middle part? [21:39] <benlieb> c_14: I want the part between the two times [21:39] <c_14> ffmpeg -ss start time -t duration -i input output [21:39] <c_14> Or, ffmpeg -i input -ss start_time -to end_time output [21:40] <c_14> s/start time/start_time [21:40] <benlieb> c_14: the -to docs say "Stop writing the output at position" [21:40] <benlieb> this is position of the output file, no? [21:41] <c_14> depends [21:42] <c_14> If you use -to without -ss or with -ss as an output option, it's the timestamp of the input file. [21:42] <benlieb> if I wanted to get from 03:00 to 04:00 would I use -to 04:00 or -to 01:00 [21:42] <benlieb> so when -ss and -to are use as output options they refer to the input file? [21:43] <benlieb> That's weird [21:43] <c_14> ffmpeg -i file -ss 03:00 -to 04:00 output or ffmpeg -ss 03:00 -t 01:00 -i input output or ffmpeg -ss 03:00 -i input -t 01:00 output or ffmpeg -ss 03:00 -i input -to 01:00 output [21:43] <benlieb> about -ss docs say When used as an output option (before an output filename), decodes but discards input until the timestamps reach position [21:43] <c_14> yep [21:43] <benlieb> this clip is actually towards the end of a 2 hour vi [21:43] <benlieb> vid [21:44] <c_14> https://trac.ffmpeg.org/wiki/Seeking%20with%20FFmpeg [21:44] <c_14> That might be a bit more helpful. [21:44] <c_14> If you don't understand something, tell me and I'll fix it/help you. [21:47] <benlieb> c_14: ffmpeg -i video.mp4 -ss 00:01:00 -to 00:02:00 -c copy cut.mp4 [21:47] <benlieb> will do what I want [21:47] <benlieb> but will it also decode 0:00 to 1:00 [21:47] <benlieb> ? [21:47] <c_14> yes [21:48] <benlieb> so there's no way to do a fast seek and use two timestamps [21:48] <benlieb> which is odd [21:48] <c_14> Fast seeking throws away the timestamps. [21:48] <benlieb> because it's my most common use case [21:49] <benlieb> as of now I use a duration library to interpret two time stamps and calculate the -t [21:49] <c_14> Ie if I fast-seek 20 minutes into a video, the first timestamp I encounter after seeking is 00:00:00 not 00:20:00 [21:49] <benlieb> but it doesn't handle milliseconds [21:49] <c_14> It does _unless_ you use -c copy. [21:49] <c_14> Or do you mean the library? [21:49] <benlieb> the lib [21:50] <benlieb> c_14: I actually can't find a single ruby lib that handles duration w milliseconds. [21:50] <benlieb> except ones that represent 'time', which is linked to the unix epoque [21:51] <c_14> Where do you get the timestamps from? Because you can just give ffmpeg timestamps as floats in seconds.milliseconds [21:51] <c_14> "floats", that is [21:51] <benlieb> c_14: that wouldn't solve the problem of needing a duration [21:52] <benlieb> That's what the lib was doing. [21:52] <benlieb> I'm getting them from a aegisub [21:52] <benlieb> subtitling program, which has a nice interface for mapping sections of video, even though i'm not using subtitles ;) [21:53] <c_14> you have a from and a to timestamp, right? [21:53] <c_14> In the form xx:xx:xx.xx ? [21:53] <benlieb> it also uses the easily parseable .ass format [21:53] <benlieb> yep [21:54] <c_14> You might need to make your own function to calculate duration with milliseconds... [21:55] <c_14> That, or find a way to get ffmpeg not to throw away timestamps when keyframe-seeking. [22:00] <benlieb> c_14: yeah [22:01] <benlieb> prob will end up writing my own duration class [22:01] <benlieb> grr [22:02] <benlieb> all good though I guess [22:09] <benlieb> c_14: I wonder if ffmpeg is the right tool for what I'm doing [22:10] <benlieb> I have fallen into this odd service of converting instructional dance DVDs into a digitally distributable format. [22:11] <benlieb> It requires adding intros, and splitting the dvd at various points, adding music, etc. But I'm not super happy with the product I've achieved. I've managed to get a relatively automated system together, but it's buggy and unwieldy. [22:11] <benlieb> But here's an example of the 'finished' product: [22:11] <benlieb> c_14: http://www.idance.net/en/packs/465-the-22-foundation-patterns-of-west-coast… [22:11] <benlieb> do you have any tools in mind that might be better suited for this? [22:12] <benlieb> using a gui would take way too long and be more tedious. [22:14] <ChocolateArmpits> Did you look into Avisynth? [22:15] <benlieb> ChocolateArmpits: hm. Hadn't heard of it. [22:15] <benlieb> ChocolateArmpits: how does it compare to ffmpeg? [22:16] <ChocolateArmpits> Avisynth allows to use video filters to process video, you can write filters, share filters [22:16] <ChocolateArmpits> It's a filter processor [22:16] <ChocolateArmpits> of sorts [22:17] <benlieb> a filter processor? [22:17] <ChocolateArmpits> well, ok, Wiki gives a better description of a "GUI-less video editor" [22:17] <benlieb> I've found it extremely unintuitive to work with ffmpeg to do basics of cropping, cutting, contacting etc. [22:18] <benlieb> is it more intuitive? [22:18] <ChocolateArmpits> Well if "vid1 ++ vid2" to get a concatted video is intuitive, then yes [22:19] <ChocolateArmpits> http://en.wikipedia.org/wiki/AviSynth [22:19] <ChocolateArmpits> Just read the page [22:19] <benlieb> that sounds good. [22:20] <benlieb> considering this is what I do do to add an intro (already made) in ffmpeg. https://gist.github.com/pixelterra/c46cfc699caba052d543 [22:21] <benlieb> ug [22:21] <ChocolateArmpits> You can load Avisynth scripts into ffmpeg as a regular input, but only 32bit version is supported [22:22] <ChocolateArmpits> 32bit compile of ffmpeg that is [22:23] <ChocolateArmpits> There is also Vapoursynth, similar thing, but newer and uses Python http://www.vapoursynth.com/ [22:24] <benlieb> Those look really cool. [22:24] <ChocolateArmpits> You can generate text in ImageMagick, save as an image file with alpha and then apply it in Avisynth over a video via Layer filter [22:25] <ChocolateArmpits> I use Avisynth myself to generate videos with different video/subtitle combinations [22:27] <benlieb> ChocolateArmpits: Awesome. Is Vapoursynth more or less powerful / popular? [22:27] <ChocolateArmpits> Vapoursynth doesn't have as many filters, so obviously less functionality, but it should be faster than Avisynth, at least from my few tests [22:28] <benlieb> ChocolateArmpits: speed isn't my main concern [22:29] <ChocolateArmpits> In general Vapoursynth tries to position itself as a logical continuation to Avisynth [22:29] <ChocolateArmpits> even though there are projects, such as Avisynth+ which try to bring Avisynth more up to date by combining features such as 64bit support, proper multithreading [22:29] <benlieb> ChocolateArmpits: since you're in ffmpeg chanel, you clearly still need / use ffmpeg? [22:30] <ChocolateArmpits> I'd probably go insane having to use an NLE every time I want to transcode a video [22:30] <benlieb> nle? [22:30] <ChocolateArmpits> Non-Linear Editor [22:30] <ChocolateArmpits> You're not familiar with broadcast terms? [22:31] <ChocolateArmpits> or rather, video in general [22:31] <benlieb> ChocolateArmpits: lol, nope [22:31] <ChocolateArmpits> It stands for video editor software [22:31] <ChocolateArmpits> editing* [22:31] <ChocolateArmpits> Such as Premiere, Final Cut [22:31] <benlieb> I'm a web developer, and somehow stumbled into video accidentally with a project [22:32] <benlieb> now I have thousands of videos on my hand and a huge stack of dads to process. [22:32] <ChocolateArmpits> I see [22:35] <ChocolateArmpits> Any more questions ? [22:35] <benlieb> so for my general use case I need to chop a dvd into pieces, add an intro, make previews, and stills [22:35] <benlieb> ChocolateArmpits: ^ [22:35] <benlieb> well I need to make the intro too [22:36] <benlieb> I'm using .ass, a background video, and an mp3 to make a pretty simple intro [22:36] <ChocolateArmpits> The intro consists of just an overlayed text that fits the topic the video is for ? [22:37] <benlieb> I grab the text from the db and use an .ass template file to output another .ass file that I use for the ffmpeg filter. [22:37] <benlieb> could avisynth do all of this? [22:38] <ChocolateArmpits> Well, in this case Avisynth will only overlay the subtitle [22:38] <ChocolateArmpits> via a subtitle plugin, Assrender or VsFilter(mod) [22:38] <benlieb> can it do cropping / cutting? [22:38] <ChocolateArmpits> Yes [22:38] <ChocolateArmpits> Trim(in_point,out_point) [22:39] <benlieb> does it do audio noise reduction? [22:39] <ChocolateArmpits> Eh, it's not really audio-oriented [22:39] <benlieb> ChocolateArmpits: can it layer multiple videos with alpha channels ? [22:40] <ChocolateArmpits> Yes, Layer(bg_video,top_video) [22:40] <ChocolateArmpits> You can use video as a mask for another video [22:40] <benlieb> that sounds like more what I need. [22:40] <benlieb> but no audio processing? [22:41] <ChocolateArmpits> Well there are filters, both internal and external http://avisynth.nl/index.php/External_filters#Audio_Filters http://avisynth.nl/index.php/Internal_filters#Audio_processing_filters [22:41] <ChocolateArmpits> Externals seem to offer SoX, so maybe that has some noise correction feature [22:41] <ChocolateArmpits> You can also use SoX CLI outside avisynth [22:42] <ChocolateArmpits> At least I do all my sound processing outside Avisynth and then mux with ffmpeg [22:43] <benlieb> yeah, this is all immensely complex. i probably bit off more than is reasonable [22:43] <benlieb> I have also been storing data in a db that corresponds to all of the parameters needed to (re)generate all resources from their source file [22:43] <benlieb> this includes intro music, font size and color etc, video background, dimensions... [22:44] <ChocolateArmpits> Is the noise quantifiable? [22:44] <benlieb> so if something goes wrong or missing and is noticed 1 year from now, I'd be able to regenerate all videos, thumbs etc from the db. [22:44] <benlieb> ChocolateArmpits: hrn? [22:44] <benlieb> hm? [22:44] <benlieb> ChocolateArmpits: the noise is different for every recording [22:45] <benlieb> I've been manually using audacity [22:45] <ChocolateArmpits> Oh, so different frequencies and levels ? [22:45] <benlieb> I'm basically converting dvds to purchasable sections [22:48] <ChocolateArmpits> ok, Sox has a noise reduction filters, http://sox.sourceforge.net/sox.html search for "noisered" [22:48] <ChocolateArmpits> But if the noise pattern isn't systematic it will be hard to apply it [22:49] <Diogo> Hi this is possíble using generate ts for hls change the time of each segment [22:49] <benlieb> ChocolateArmpits: you usually have to sample it. [22:49] <benlieb> ChocolateArmpits: but iMovie has a great noise filter that is very smaret [22:49] <benlieb> smart [22:56] <ChocolateArmpits> Hmm, do the videos have the same audio mix level ? [22:58] <ChocolateArmpits> benlieb ^ [22:59] <benlieb> nope, but it's not really that important [22:59] <benlieb> I mostly am concerned with making attractive intros, and cutting videos easily. maybe some filters for contrast etc [23:00] <ChocolateArmpits> Well, you can detect audio silence, when it goes below a threshold level, then use the value to sample silent section and then use the sampled noise for the noisered [23:00] <ChocolateArmpits> So it's possible to some degree to automate [23:00] <ChocolateArmpits> You would have to average the levels beforehand so the silence is at relatively the same level through the recordings [23:01] <ChocolateArmpits> Silence detection can be done with ffmpeg https://www.ffmpeg.org/ffmpeg-filters.html#silencedetect [23:01] <ChocolateArmpits> from there you would move with the extracted value to SoX [23:03] <benlieb> ChocolateArmpits: would it be wisest to start with avisynth or go directly to vaporsynth? [23:03] <ChocolateArmpits> If you don't know Python, then Avisynth [23:04] <benlieb> ChocolateArmpits: I know basic python [23:04] <benlieb> do you think the features of VS are catching up to AS? [23:06] <ChocolateArmpits> I have only started following it only recently, but some of the essential plugins as getting native rewrites, rather than using a plugin to load avisynth native plugins, so there is some implied migration [23:08] <ChocolateArmpits> I would probably choose Avisynth, as I'm not familiar with Python, I only installed Vapoursynth to see how it performs and general interest [23:08] <ChocolateArmpits> My future outlook on Vapoursynth is better than on Avisynth though [00:00] --- Fri Sep 26 2014

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