Ffmpeg-devel-irc

Download

ffmpeg-devel-irc@ffmpeg.org

September 2014

1 participants
60 discussions

ffmpeg-devel.log.20140914
by burek 15 Sep '14

15 Sep '14

[00:02] <jamrial> it should be functional. we have fate tests for it. no idea if it's feature complete, though [00:02] <jamrial> ask mraulet [00:08] <ubitux> ass demuxer will soon output "mkv ass" packets [00:08] <nevcairiel> i thought you wanted to get away from mkv ass [00:08] <ubitux> then i'll work on making the decoded sub using that form [00:08] <ubitux> nevcairiel: the opposite [00:08] <ubitux> i don't want the decoded form to include the timestamps [00:08] <nevcairiel> well no complaints here, directshow uses mkv-ass [00:10] <nevcairiel> for the longest time ffmkv just converted everythign away from the mkv format tho [00:10] <ubitux> yes, i fixed that [00:10] <ubitux> since last major bump it now outputs the correct packet format [00:20] <cone-934> ffmpeg.git 03Michael Niedermayer 07master:5312fb13b47c: RELEASE_NOTES: add version numbers [01:22] <cone-934> ffmpeg.git 03Andreas Cadhalpun 07master:8f537420ae71: doc: mention in APIChanges that AVProbeData must be initialized due to the new mime_type field [02:02] <cone-934> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:4c8be331bcd5: ffserver: use correct error for stream not found [02:02] <cone-934> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:2638af2fc5ec: ffserver: drop custom skip_spaces() impl [03:18] <cone-934> ffmpeg.git 03Andreas Cadhalpun 07master:365b9cf6245e: doc: mention important API changes in the RELEASE_NOTES [07:31] <Timothy_Gu> ubitux, would you please apply parts of my release notes if you have time as I won't be able to finish it and michaelni is releasing it tomorrow? Thank you so much. [07:35] <ubitux> Timothy_Gu: huh? [07:35] <ubitux> you mean i have to split and reword what you started? [07:40] <wm4> btw. I just realized I didn't post my PGS patch again [07:40] <wm4> too late for 2.4? [07:42] <ubitux> i don't see the branch created yet [07:42] <ubitux> what pgs sub btw? [07:42] <wm4> extracted bluray bitmap subs [07:47] <ubitux> Timothy_Gu: anyway, working on it. [07:47] <ubitux> wm4: ah, the .sup thing? [07:50] <ubitux> Timothy_Gu: i don't know what's the state of the API/ABI in 2.4, but we bumped major so... i would guess a lot of things changed in both [07:50] <wm4> ubitux: yes [07:53] <ubitux> dunno, it will require a minor bump [07:53] <ubitux> it's not branched out yet, but the library versions are already written to the RELEASE_NOTES [07:54] <ubitux> that can be adjusted of course but well.. [07:54] <jamrial> ubitux: add a line about utf16 support to changelog/release_notes while at it since it wasn't done when the code was committed [07:55] <ubitux> right [07:55] <ubitux> i need help with the API/ABI stuff [07:55] <ubitux> someone needs to summarize what happened [07:55] <ubitux> we had a major bump, so API & ABI "broke", right? [07:56] <jamrial> ABI yes, API sorta [07:56] <jamrial> we dropped some deprecated functions with the bump [07:56] <jamrial> check the link i posted some hours ago. they are listed there [07:56] <ubitux> i'm pretty sure the FF_API_* stuff changed behaviours [07:57] <ubitux> the bump at least made matroska output the correct ass packets :p [08:00] <jamrial> wonder if people will stop reinventing the wheel (aka, rewritting their own demuxers) for their projects from now on, then :p [08:01] <ubitux> there are a few missing features in our mkv demuxer / lavf api [08:01] <jamrial> anyway yeah, the deprecated functions were all inside a couple FF_API_* that we didn't bother to postpone [08:04] <rcombs> namely, editions and ordered chapters, and maybe segment linking as well [08:04] <wm4> and these things are not trivial at all [08:04] <rcombs> it's true! [08:05] <wm4> although mpv would use the lavf demuxer even if the demuxer just exports ordered chapter metadata [08:05] <wm4> (instead of implementing them fully) [08:05] <rcombs> I have a patch laying around that adds support, but it's a bit of a mess [08:05] <rcombs> and it's based on an old branch that was previously rejected [08:14] <ubitux> i like how libav release not mention an api compatibility and in the same file says "A number of additional APIs have been introduced ..." [08:15] <ubitux> notes* [08:15] <wm4> ubitux: additional APIs don't mean breaking existing APIs [08:15] <wm4> but I suppose adding mimetypes to AVProbeData is somewhat of a break [08:16] <wm4> (it actually broke at least 1 application) [08:16] <wm4> still pretty innocent [08:16] <ubitux> "innocent" erm [08:17] <wm4> technically there's no API break [08:17] <wm4> but it went less "bumpy" than intended [08:17] <wm4> oh wait, you said deprecated APIs are gone? [08:20] <ubitux> it seems it is in FFmpeg but i haven't confirmed in Libav [08:23] <ubitux> maybe they are [08:25] <ubitux> so, will 2.4 be a LTS? [08:30] <jamrial> don't we pretty much support every version to a certain extent? [08:30] <jamrial> i mean, a month ago there was a point release of the 0.10 branch [08:31] <ubitux> that's because it's branched out of libav 0.8 or something [08:31] <ubitux> so the same commits are backported or something [08:34] <ubitux> meh, using my pro address. [08:41] <ubitux> ah i forgot some changes from Timothy_Gu [08:42] <ubitux> actually i forgot to actually check the diff [08:57] <ubitux> ok, back to useful stop now [08:57] <ubitux> rewarding stuff* [09:24] <rcombs> so, iirc the thing that primarily prevented ordered chapter parsing from making it into ffmpeg master was discovering the segment files, because Matroska just gives you a UUID, rather than a filename or a path [09:25] <rcombs> so the patch that added ordered chapters a while back was refused partially because it stuck directory listing in AVIO, and it was considered not to belong there [09:34] <wm4> I don't appreciate it either if ffmpeg just randomly opens files [09:35] <nevcairiel> well, anything else is plagued with huge complexity [09:45] <ubitux> wm4: can be optional ofc [09:45] <ubitux> but if ffmpeg/ffplay are able to do that transparently, that would be could [09:45] <ubitux> it will mean we could have correct playback, and we could also use ffmpeg to stack up all the files into one [09:46] <wm4> I'm still a bit of the opinion that piecing the segments is the job of code between player and demuxer, not in the demuxer [09:49] <rcombs> wm4: so, when transcoding, you would have that job be handled by ffmpeg.c? [09:49] <ubitux> just made the ass demuxer output ass packets, yay. [09:49] <rcombs> ubitux: as in, with the timestamps stripped? [09:49] <wm4> rcombs: ffmpeg.c simply could handle it as playlist (and could use the some code to concat files) [09:49] <ubitux> rcombs: yes [09:49] <rcombs> :D [09:49] <ubitux> not a huge patch actually [09:50] <wm4> ubitux: oh, nice [09:50] <wm4> so there's only 1 packet format [09:50] <ubitux> http://b.pkh.me/0001-avformat-assdec-output-ASS-packets.patch [09:50] <rcombs> is matroskadec.c outputting that format as well, now? [09:50] <ubitux> yes [09:51] <rcombs> goody [09:51] <ubitux> the only remaining ssa code stuff remains in... nut! [09:51] <wm4> that's nuts [09:51] <ubitux> but i don't want to care about [09:51] <wm4> (hey, better than an ASS pun) [09:51] <ubitux> haha [09:54] <ubitux> ok it works nicely now [09:54] <ubitux> the order from the original file is stored through readorder when doing ffmpeg -i input.ass -c copy output.mkv [09:54] <ubitux> and it can be reconstructed properly when doing the other way around [09:55] <ubitux> now i need to fix the decoded form in AVSubtitle->ass [09:55] <ubitux> so it's also reconstructed properly when transcoding [09:56] <wm4> you've probably broke concat of ass tracks [09:56] <rcombs> how did that work? [09:56] <rcombs> (concat demuxer?) [09:57] <ubitux> wm4: mmh good point :) [09:58] <ubitux> wm4: but that's true for many formats i would guess [09:58] <ubitux> not sure what could be done about that [09:58] <rcombs> how does that work with the header? [09:58] <ubitux> it doesn't [09:58] <ubitux> :D [09:58] <wm4> ordered chapters have the same problem [09:59] <rcombs> ubitux: move readorder to a field in AVSubtitle? [09:59] <wm4> that's why ordered chapters sometimes work better in mpv than other players [09:59] <rcombs> (except not, because that'd require a whole new packet format and that's a mess) [09:59] <ubitux> rcombs: won't help for remuxing [09:59] <rcombs> so, shift ReadOrder in the text when concatenating? [09:59] <ubitux> which is the purpose of concat [10:00] <ubitux> yes but where does this logic belong? [10:00] <ubitux> an exception in concat demuxer? [10:00] <ubitux> we probably need capabilities in codecs [10:01] <rcombs> good question [10:01] <ubitux> to say which packets are concat compliant [10:01] <ubitux> and which aren't [10:01] <rcombs> could have an API to provide offsetting? [10:02] <ubitux> yeah maybe [10:04] <rcombs> you'd need to know the maximum ReadOrder in the stream, though [10:04] <rcombs> so you'd have to have read the whole thing [10:05] <wm4> keep in mind that libass will drop packets with duplicate readorder [10:05] <ubitux> there is something funny btw [10:05] <ubitux> the matroska specs starts at readorder=1 [10:06] <ubitux> but every file i have here starts at 0 [10:06] <ubitux> so the code i wrote expect the readorder to start at 0 as well [10:06] <ubitux> but if a matroska is muxed "properly", starting at readorder=1 [10:06] <wm4> matroska spec actually being correct? can't have that [10:07] <ubitux> the muxer will wait until the end of the stream for the readorder entry 0 [10:08] <rcombs> ubitux: and then give up on 0 and start from 1? [10:10] <wm4> the "spec" isn't very precise about this [10:11] <wm4> I find the "1" only in an example [10:11] <wm4> and mkvtoolnix uses 0 as base - I'd just assume it's an error [10:12] <rcombs> can we at least assume nothing's dumb enough to go negative? [10:22] <ubitux> rcombs: the muxer doesn't write the readorder, so there is just a stack orderered by those [10:22] <ubitux> and the muxer waits for the next expected readorder in that stack [10:22] <ubitux> and for every packet it tries to flush that stack as long as the expected readorder matches the readorder [10:23] <ubitux> if there is a gap, it's waiting for more [10:23] <ubitux> and at the end, it flushes the stack, wheither there is gaps or not [10:25] <ubitux> can anyone have a look to the 0.6 branch cut? [10:25] <ubitux> in doc/APIchanges [10:26] <rcombs> what about it? [10:26] <ubitux> i had to move it [10:26] <ubitux> because it doesn't match what i found in 0.6 branch [10:26] <ubitux> i suspect some nastiness [10:27] <rcombs> I notice a few "NOTE: this was backported to 0.6"s [10:27] <ubitux> grumbl [10:27] <ubitux> indeed [10:29] <wm4> 0.6, isn't that prehistoric? [10:30] <ubitux> it is, but i was trying to have proper cuts in the APIchanges file [10:30] <wm4> ah [10:30] <wm4> oh [10:30] <ubitux> but with backported changes, that's indeed confusing [12:50] <cone-371> ffmpeg.git 03Clément BSsch 07master:b97e27082bb3: doc/APIchanges: attempt to split releases [12:50] <cone-371> ffmpeg.git 03Michael Niedermayer 07master:fa1b5631964b: doc/APIchanges: Update hashes and dates [12:59] <ubitux> so, i'm going to create a new sub struct [12:59] <ubitux> in libavutil, and just for text subtitles [13:00] <wm4> omg [13:00] <ubitux> AVTextSubtitle { char *dialog; int64_t start; int64_t end; } [13:00] <ubitux> do we need anything else? [13:01] <wm4> format [13:01] <ubitux> why? [13:01] <wm4> because there are various subtitle formats [13:01] <ubitux> it's for generic (ass) decoded markup [13:01] <ubitux> every decoder will output this [13:01] <ubitux> every text subtitles decoder* [13:01] <wm4> you probably want to have more flexibility here [13:01] <ubitux> indeed [13:02] <wm4> so you're not stuck with ass if you find something better [13:02] <ubitux> so a format for "ass" vs "text" [13:02] <ubitux> alright, and maybe assv5 later [13:02] <ubitux> fine with me [13:02] <ubitux> what else? [13:02] <wm4> but I'm still unsure what the general architecture should be, so I can't suggest much [13:03] <wm4> e.g. using "decoders" to convert subtitles is not ideal, to say the least [13:03] <ubitux> it works relatively well so far [13:03] <wm4> not really [13:04] <J_Darnley> Make the API private you you can change it as much as you like? [13:04] <ubitux> it's for the user [13:04] <wm4> ubitux: for example, you could argue an ASS decoder should take ASS as input, and output a list of bitmaps to blend as output [13:04] <ubitux> it will be heap allocated through a helper so we can add field, but the 4 first field at least will be for public use [13:05] <wm4> but then, to render srt, you'd have to chain 2 decoders (????) [13:05] <ubitux> erm [13:05] <ubitux> i don't think that logic belongs in the decoders [13:05] <J_Darnley> A subtitle renderer shold definitely be separate [13:06] <J_Darnley> As should an OCR module [13:06] <wm4> J_Darnley: but vobsub and pgs also use decoders for packet->bitmaps [13:06] <ubitux> the rendering is too close related to other stream ( the video one) [13:06] <ubitux> we'll make an AVBitmapSubtitles later [13:06] <nevcairiel> if you're redesigning stuff, i want a renderer that can give me the subtitle bitmaps for me to do with as I please, not only one that is designed to render it directly onto the video [13:07] <ubitux> for the bitmap sub codecs, or if you want a rasterizer codec [13:07] <wm4> maybe nicolas george is right, and it should all just be a giant graph (actually he certainly is right, but it shouldn't be lavfi) [13:07] <nevcairiel> ie. no avfilter [13:07] <ubitux> now, i need comment for the closed caption stuff [13:08] <ubitux> do we want to handle that here? [13:08] <ubitux> (aka within the suggested AVTextSubtitles thing) [13:08] <nevcairiel> if ffmpeg would provide a cc -> text decoder, i would certainly use it [13:09] <wm4> don't CCs have 4 streams in one packet? [13:12] <ubitux> so, how am i going to call the new av sub decode routine so it doesn't conflict with a potential dev from libav? [13:12] <ubitux> who bets they're going to design a new subtitles api? [13:12] <nevcairiel> of course they will [13:12] <wm4> I haven't heard anything about such ideas [13:12] <ubitux> :( [13:12] <wm4> elenril is not interested in subs, so it won't happen either [13:13] <wm4> koda was once interested in backporting subtitle stuff, but apparently gave up on that [13:13] <wm4> I wouldn't worry [13:13] <ubitux> ok [13:21] <ubitux> is it better to have the start/end ts associated with a time base field, or people would prefer a AV_TIMEBASE ts, or ms or something? [13:22] <ubitux> (having in tb field makes simplifications in internals and allow reducing the number of inaccuracy with the different rescaling) [13:23] <wm4> like with packets and avframes, the timebase should probably be external? [13:27] <ubitux> yeah it would be the stream tb i guess [13:27] <wm4> if it ends up as ASS anyway, you could hardcode the ASS timebase [13:27] <wm4> which is 10ms [13:28] <ubitux> i don't want to [13:28] <ubitux> just like you said, it can change in the future, i added a format field [13:28] <ubitux> and assv5 will probably have a better resolution [13:28] <ubitux> also, the ts isn't in the data anymore, so it can be expressed in any tb [13:31] <nevcairiel> 10ms resolution? that seems kinda coarse [13:31] <nevcairiel> cant even match video frames properly [13:33] <wm4> yeah [13:37] <saste> wm4, are you going to VDD? [13:38] <wm4> no [13:39] <saste> most irc people don't go to such meetings indeed [13:40] <saste> so there won't be much to discuss, probably... [13:40] <ubitux> does anyone mind if i add this to the RELEASE_NOTES? : [13:40] <ubitux> As usual, if you have any question on this release or any FFmpeg related [13:40] <ubitux> topic, feel free to join us on the #ffmpeg IRC channel (on [13:40] <ubitux> irc.freenode.net) [13:41] <Compn> i dont [13:42] <cone-371> ffmpeg.git 03Clément BSsch 07master:5eef7042d8a2: Changelog/RELEASE_NOTES: prepare for 2.4 [13:42] <ubitux> thx [13:49] <rcombs> you don't mind, or you don't have any question on this release or any FFmpeg related topic? [13:49] <rcombs> :P [13:51] Action: ubitux bugs [13:53] <Compn> simple binary answer to binary question [15:43] <cone-371> ffmpeg.git 03Clément BSsch 07fatal: ambiguous argument 'refs/heads/release/2.4': unknown revision or path not in the working tree. [15:43] <cone-371> Use '--' to separate paths from revisions [15:43] <cone-371> refs/heads/release/2.4:HEAD: Changelog/RELEASE_NOTES: prepare for 2.4 [15:58] <cone-371> ffmpeg.git 03Andrew Stone 07release/2.4:04361427e65a: Revert "lavf: eliminate ff_get_audio_frame_size()" [15:58] <cone-371> ffmpeg.git 03Anton Khirnov 07release/2.4:7dfccac20c0c: electronicarts: do not fail on zero-sized chunks [15:58] <cone-371> ffmpeg.git 03Diego Biurrun 07release/2.4:e8f2823f0651: vsrc_movie: Adjust a silly typo from b977b287f61fea48ecd6251d54a26334213b7ec6 [15:58] <cone-371> ffmpeg.git 03Diego Biurrun 07release/2.4:d04fb118684f: error_resilience: Drop asserts from guess_mv() [15:58] <cone-371> ffmpeg.git 03Diego Biurrun 07release/2.4:d2bad216f775: mpeg12enc: Add missing #include for PICT_FRAME [15:58] <cone-371> ffmpeg.git 03Diego Biurrun 07release/2.4:63795fe5b967: setpts: Add missing inttypes.h #include for PRId64 [15:58] <cone-371> ffmpeg.git 03Diego Biurrun 07release/2.4:0263750a0db7: vfwcap: Add fallback define for HWND_MESSAGE [15:58] <cone-371> ffmpeg.git 03Luca Barbato 07release/2.4:8c91414803e4: vc1: Fix the skip condition [15:58] <cone-371> ffmpeg.git 03Anton Khirnov 07release/2.4:c2d6cc2971b3: mpegenc: limit the maximum muxrate [15:58] <cone-371> ffmpeg.git 03Anton Khirnov 07release/2.4:7c4685507498: avconv: fix the muxrate values for -target [15:59] <cone-371> ffmpeg.git 03Anton Khirnov 07release/2.4:e2a89f7f0f8f: avconv: fix parsing the AVOptions for -target [15:59] <cone-371> ffmpeg.git 03Luca Barbato 07release/2.4:ee099059e71e: vc1: Initialize start_code_found to 0 [15:59] <cone-371> ffmpeg.git 03Luca Barbato 07release/2.4:e62f08ca8d6e: pulse: Add a wallclock option to be compatible with other other captures [15:59] <cone-371> ffmpeg.git 03Anton Khirnov 07release/2.4:1f52f82a55a5: doc/APIchanges: fill in missing hashes and dates [15:59] <cone-371> ffmpeg.git 03Vittorio Giovara 07release/2.4:5694831e0693: matroska: list supported extensions [15:59] <cone-371> ffmpeg.git 03Vittorio Giovara 07release/2.4:110841c3ab1d: avcodec: add stream-level stereo3d side data [15:59] <cone-371> ffmpeg.git 03Vittorio Giovara 07release/2.4:152e09fde7f6: matroskadec: parse stereo mode on decoding [15:59] <cone-371> ffmpeg.git 03Vittorio Giovara 07release/2.4:5b740d1eaa63: matroskaenc: convert avstream stereo3d side data during encoding [15:59] <cone-371> ffmpeg.git 03Diego Biurrun 07release/2.4:4cde8bae4927: license: Mention that vf_interlace is GPL, not LGPL [15:59] <cone-371> ffmpeg.git 03Reinhard Tartler 07release/2.4:b5d4f49e3cb1: Prepare for 11_beta2 Release [15:59] <cone-371> ffmpeg.git 03Michael Niedermayer 07release/2.4:480633c6c2e1: avcodec: fix missing doxygen comment marker [15:59] <cone-371> ffmpeg.git 03Anton Khirnov 07release/2.4:9d3e69ae3013: Add release notes for 11. [15:59] <cone-371> ffmpeg.git 03Diego Biurrun 07release/2.4:07b0ccf5116c: Mark 11 release in the changelog [15:59] <cone-371> ffmpeg.git 03Sean McGovern 07release/2.4:7d8ebb877408: Fix RELEASE identification [15:59] <cone-371> ffmpeg.git 03Diego Biurrun 07release/2.4:4f2d4b98fc98: doc: Fix syntax and logical errors in avconv stream combination example [15:59] <cone-371> ffmpeg.git 03Diego Biurrun 07release/2.4:f851477889ae: Prepare for 11 release [15:59] <cone-371> ffmpeg.git 03Michael Niedermayer 07release/2.4:547cad8c81c1: Merge commit 'f851477889ae48e2f17073cf7486e1d5561b7ae4' into release/2.4 [16:01] <cone-371> ffmpeg.git 03Michael Niedermayer 07master:da2186be81b5: MAINTAINERS: Add 2.4 to maintained releases, drop 2.3 [16:01] <cone-371> ffmpeg.git 03Michael Niedermayer 07release/2.4:66ac5b96e80a: MAINTAINERS: Add 2.4 to maintained releases, drop 2.3 [16:53] <cone-371> ffmpeg.git 03Michael Niedermayer 07master:68bca03951b3: doc/examples: remove unneeded NULL checks [16:53] <cone-371> ffmpeg.git 03Michael Niedermayer 07master:808db3e68778: Changelog: add 2.4 [17:16] <cone-371> ffmpeg.git 03Michael Niedermayer 07master:b227be34db76: avcodec/mjpegenc: the AMV encoder doesnt support yuv422 [17:51] <cone-371> ffmpeg.git 03Michael Niedermayer 07release/2.4:c16e80ee3d1c: doc/examples: remove unneeded NULL checks [17:52] <cone-371> ffmpeg.git 03Michael Niedermayer 07release/2.4:703bd3164736: Changelog: add 2.4 [17:52] <cone-371> ffmpeg.git 03Michael Niedermayer 07release/2.4:ace90ee26550: avcodec/mjpegenc: the AMV encoder doesnt support yuv422 [18:28] <ubitux> wm4: btw, now that i think about the concat & ass muxer with readorder [18:28] <ubitux> i think it won't be hard to handle that situation [18:29] <ubitux> the ass muxer need to be adjusted to handle discontinuities [18:29] <ubitux> and doing this in a simple way is possible [18:29] <ubitux> just keeping track of every occured readorder, and if you get it twice, you force-flush the current stack, and reset expected readorder, then go on this [18:30] <ubitux> i'll eventually take the time to do that [18:31] <ubitux> unrelated, what is public in AVCodec? [18:31] <wm4> you still can't write the same readorders to the file [18:31] <wm4> if you e.g. mux to mkv [18:31] <ubitux> ah, finally found the public/private cut in the comments. [18:32] <wm4> someone should also take the time to move non-public fields to an internal struct [18:32] <ubitux> wm4: that's write, the mkv muxer needs adjustment [18:32] <ubitux> right* [18:36] <cone-371> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n2.4': unknown revision or path not in the working tree. [18:36] <cone-371> Use '--' to separate paths from revisions [18:36] <cone-371> refs/tags/n2.4:HEAD: avcodec/mjpegenc: the AMV encoder doesnt support yuv422 [19:09] <cone-371> ffmpeg.git 03Clément BSsch 07master:fcfa3ebed134: avcodec/utils: remove avcodec_ prefix for internal symbol [19:21] <ubitux> 2.0 tag was added in the main line but not the following? [19:37] <michaelni> n2.0 is exactly on the last commit that is in both release/2.0 and master [19:39] <ubitux> and every other release was tagged within a commit in the branch itself? [19:41] <michaelni> v0.8 is on libavs master branch and as we merge that also on ours [19:42] <michaelni> the tag isnt though in our repo [19:46] <ubitux> i'm thinking of post n2.0 [19:46] <ubitux> like n2.1, n2.2, .. [19:48] <michaelni> they are all after the branchpoints, you can see this with git log master..n2.1 vs. git log master..n2.0 [19:49] <ubitux> mmh i see [19:50] <ubitux> ok [20:00] <cone-371> ffmpeg.git 03Clément BSsch 07master:b24d74e44ae5: avcodec/microdvddec: indent fix [20:29] <cone-371> ffmpeg.git 03Deb Mukherjee 07master:04b0dda85319: avcodec/libvpxdec: Adds decode support for formats other than 420 [20:34] <ubitux> we need to send srt and ssa decoders to hell [21:21] <wm4> amazing, ffmpeg has public fields that are not public [21:21] <wm4> e.g. AVFrame.channels [21:21] <wm4> it's designated as public, but "Code outside libavcodec should access this field using: * av_frame_get_channels(frame)" [21:22] <BtbN> Just a deprecated way of accessing it, what's so special about it? [21:23] <ubitux> michaelni: re: Ticket #3929 / b96d864 don't we need http://pastie.org/9553303 as well? [21:24] <ubitux> (no idea how to test) [21:25] <michaelni> probably [21:28] <wm4> ubitux: someone requested audio visualization of this kind: http://www.youtube.com/watch?v=tYfxYBmHTp8 [21:29] <ubitux> cute [21:29] <nevcairiel> i assume those foreign chars are lyrics [21:29] <wm4> yes [21:29] <nevcairiel> looks like something milkdrop could potentially do [21:30] <ubitux> sounds kind of complex for ending up in ffmpeg [21:30] <ubitux> but maybe J_Darnley will propose something so user could write such thing :p [21:30] <nevcairiel> should opengl milkdrop and make it a open-source library or stuff [21:33] <J_Darnley> You rang? [21:34] <J_Darnley> Well, avs might be able to some of that. [21:34] <J_Darnley> I have seen moving block presets. [21:35] <J_Darnley> Waves are easy. [21:36] <J_Darnley> Text is not. I'm ignoring the avs' only text renderer as it is a mess of windows API calls. [21:39] <ubitux> J_Darnley: did you have time to look at eq btw? [21:40] <J_Darnley> Oh [21:40] <J_Darnley> no I completely forgot. [21:40] <ubitux> :( [21:40] <J_Darnley> I'm never good at remembering to do things. [21:44] <J_Darnley> Hey, eq is only 240 lines [21:45] <ubitux> eq2 about the same as well iirc [21:45] <ubitux> and if you strip the asm there isn't much remaining [21:46] <ubitux> the most boring parts are probably the filter boilerplate (doc/writing_filters.txt is your friend) and the documentation [21:47] <J_Darnley> I think I've already got the boiler plate done into vf_example [21:47] <J_Darnley> (there's something else I never finished) [22:17] <J_Darnley> Does anyone here know the range of values the vf_eq allowed for brightness and contrast? From the code I would guess -100 to +100. [22:19] <J_Darnley> oh, I bet the mplayer docs can tell me. [22:22] <J_Darnley> yes, -100 to +100 [22:24] <ubitux> you probably want to rescale that to -1 .. 1 [22:39] <J_Darnley> And use a float? I'll put that on the todo list. [23:17] <J_Darnley> Would ptrdiff_t be the right type to pass linesize/stride into a function so that I wouldn't have to sign/zero extend in an assembly function? [23:18] <nevcairiel> yes [23:18] <nevcairiel> linesize is the offset between two lines, ie. the difference, hence ptrdiff [23:18] <nevcairiel> :) [23:18] <J_Darnley> just checking. [23:18] <J_Darnley> In that same situation, would size_t be the right type for width and height? [23:19] <nevcairiel> technically yes, but if its inside ffmpeg, we still use int everywhere for that [23:19] <J_Darnley> I know. I'm just planning ahead, a little. [23:23] <ubitux> J_Darnley: just keep it consistent :p [23:27] <J_Darnley> yeah, on second thought, I won't. [23:27] <J_Darnley> I'm using ints in an array index anyway. [23:32] <J_Darnley> okay, that might be everything [23:33] <J_Darnley> now for the makefile and allfilters [23:35] <J_Darnley> Oh, I think I might be missing a few includes... [23:46] <ubitux> ah, i just understood the meaning of "bi-directional" in the fribidi context [23:46] <ubitux> it's simply that arabic text can be in 2 directions: right to left by default, but left to right for numbers [23:47] <nevcairiel> yeah, its really annoying to handle [23:47] <nevcairiel> $work has its own tiny bidi library [23:48] <nevcairiel> had to fix some punctuation issues at one time, really learned a lot about bidi rendering that day [23:48] <ubitux> bidi is not enough anyway, you need proper context shaping [23:48] <ubitux> i wonder what's the purpose of this half-assed library [23:48] <rcombs> ubitux: half-assed non-thread-safe library! [23:49] <rcombs> (unless you're on a particularly recent version or passed a particular configure arg) [23:50] <ubitux> no he didn't release yet [23:50] <rcombs> yeah, so you'd have to be on git [23:50] <ubitux> i doubt anyone is running a git of fribidi [23:50] <rcombs> I am! [23:50] <ubitux> haha [23:50] <rcombs> (but it's me and about 6 other people, one being the fribidi developer and the other 5 being libass guys) [23:51] <ubitux> why do you guys need the git of fribidi? [23:51] <rcombs> because it's thread-safe! [23:51] <ubitux> could just rebuild the stable version, but ok :) [23:51] <ubitux> also, not present in any release [23:51] <ubitux> i see "Support 4-byte UTF-8 sequences" [23:52] <ubitux> it's kind of frightening for such library [23:52] <rcombs> sure, but we were poking at it before figuring out that there was a configure option that'd solve the problem [23:52] <ubitux> i wonder how it's handled in pango & friends [23:53] <ubitux> i didn't check if pango builds the internal fribidi copy with it or not [23:53] <rcombs> and now we have a hack that puts the relevant fribidi call behind a mutex, but only if your fribidi was configured to be non-threadsafe, which we can actually check at runtime [23:54] <ubitux> btw, you do need fribidi even when you have hb, right? [23:54] <rcombs> yup :| [23:54] <ubitux> erm. [23:55] Action: ubitux needs to figure out what exact task these 2 libraries do, and where they overlap [23:55] <rcombs> when we first ran into this problem, I asked if we could just remove fribidi in favor of harfbuzz, and was saddened to find that we couldn't [23:56] <J_Darnley> Oh damn! There still isn't a per-plane size factor in pixfmt descriptions. [23:58] <ubitux> J_Darnley? [23:58] <ubitux> rcombs: remember what was missing? [23:59] <rcombs> ubitux: no, I think zgreg was the one who knew the details on that [23:59] <rcombs> my understanding was that they're 2 different libs that do entirely separate things, and you happen to need both to shape Arabic and such [23:59] <rcombs> but I don't have an excellent understanding of the topic, so take that with a grain of salt [00:00] --- Mon Sep 15 2014

1 0

ffmpeg.log.20140914
by burek 15 Sep '14

15 Sep '14

[02:04] <nsfx> Hello, any tips on how to avoid blurry/pixelated text on the x11grab source in this script? http://pastebin.com/HTnRGpSs [02:18] <EvolE> nsfx: did you try to increase bitrate of the video "-b:v 1000k -minrate 1000k -maxrate 1000k" ? you can try 2000, 2500 and 3000 and see if it helps [02:19] <nsfx> EvolE: I have, doesn't seem to make a difference. I also tried using a beefier pixel format [02:20] <nsfx> I think it's something in the filter_complex causing this because if I work with just the x11grab source it looks fine [02:21] <EvolE> nsfx: is "inkject" your twitch channel? [02:22] <nsfx> EvolE: Yes [02:22] <sacarasc> nsfx: Also, avconv is not supported in this channel. Try #libav. [02:23] <nsfx> sacarasc: I figured I'd x-post here :) more populated [02:23] <nsfx> Sorry about that [02:28] <_dunno_> nsfx have u tried different presets? [02:30] <nsfx> _dunno_: Yes I've tweaked just about everything outside the filter_complex param [02:30] <nsfx> Looks the same even on veryslow [02:30] <nsfx> Here's what the output looks like to get an idea http://i.imgur.com/RSL3NcC.png [02:31] <sacarasc> nsfx: You're doing 30 images at 1280x720 within 125kB. They're not gonna look good! (And Twitch supports 3k.) [02:33] <_dunno_> nsfx take a screenshot and zoom in - guess it'll look the same - it's your X11 not ffmpeg/avconv [02:34] <_dunno_> nsfx -vf unsharp might help [02:35] <EvolE> nsfx: and seems like something is wrong with colors on the video? [02:35] <nsfx> sacarasc: When I stream my entire desktop it looks way sharper, this aliasing/color loss only happens when I mix video sources [02:36] <nsfx> _dunno_: Sorry I didn't get what you meant. What do you mean by zooming in / X11's fault? [02:36] <EvolE> so probably something wrong is in "-filter_complex" parameter [02:36] <sacarasc> Waiting for OBS Multiplatform, then. :D [02:37] <nsfx> I suppose I can running the equivalent cmmand through ffmpeg proper [02:37] <nsfx> can try* [02:38] <EvolE> if you can upload 3 different normal videos I can try to play with them xD [02:38] <sacarasc> nsfx: ^ Won#t have to install with them. [02:38] <nsfx> Thanks sacarasc, I'll try that [02:42] <nsfx> Ah I just fixed it [02:42] <nsfx> I had a feeling this would fix it [02:43] <nsfx> Instead of overlaying the x11grab onto the v4l source, I overlayed the v4l source on the x11grab [02:43] <nsfx> Now it looks great [02:44] <_dunno_> nsfx so plz post another pic for the before-after experience [02:44] <nsfx> 1 sec [02:47] <nsfx> _dunno_: http://i.imgur.com/0pXC7c0.png [02:50] <EvolE> nice [02:51] <_dunno_> ? am i blind? differences are marginal - at best [02:52] <nsfx> _dunno_: -g 60 -keyint_min 30 -b:v 1000k -minrate 1000k -maxrate 1000k \ [02:53] <nsfx> -pix_fmt yuv420p -preset ultrafast -tune film \ [02:53] <nsfx> Oops wrong paste [02:53] <nsfx> _dunno_: The window on the right is the video output, the window in the middle is the actual window [02:55] <EvolE> _dunno_: you need to look at the top right corner, on previous screenshot there were almost no colors on that scroreboard [02:57] <_dunno_> okay i can see the diff there [02:58] <_dunno_> nsfx OT: NES like Nintendo? gng - ghouls and goblins? [03:01] <nsfx> _dunno_: Yeap [03:03] <_dunno_> nsfx don't you think 1280x768 is way to small for NES streaming? Considered using 4k, yet? ;) [03:04] <nsfx> Hahaha [06:08] <Soul_keeper> ffmpeg -i input -c:a libvorbis -b:a 192k -vn -sn output.oga Is the default behavior here CBR or ABR for the audio ? [06:14] <Soul_keeper> "Set bitrate expressed in bits/s for ABR." ok found it in the docs, nm [06:16] <bencc1> to record audio on a server I need a dummy alsa soundcard? [09:15] <mozzarella> guys help [09:27] <mozzarella> I'm trying to stream a video to twitch.tv [12:18] <hachi> hello folks, I'm getting Non-monotonous DTS in output stream with -vsync cfr... and without as well. Does anyone know why -vsync cfr doesn't seem to help that situation? [13:17] <xtruder> hi/ Does ffmpeg support multi bitrate streaming like FME? [15:51] <pa> can i somehow tell ffmpeg (via command line/api) to start decoding(or transcoding) from the nth keyframe, instead of time offset? [16:49] <brontosaurusrex> what would be equal to : lame -V 2 ? [19:57] <Belliash> hello [19:57] <Belliash> I got problem while pushing stream to rtmp server... my stream looses audio [19:59] <mozzarella> Belliash: how do you do that [20:00] <mozzarella> I tried to push a video to a rtmp server and failed miserably [20:01] <EvolE> how you both do that?) [20:02] <mozzarella> EvolE: no YOU tell us how to do it [20:02] <EvolE> -f flv rtmp://... ? [20:03] <mozzarella> ffmpeg -i TEDxGE2014_Stallman05_HQ.ogg -f flv rtmp://live.twitch.tv/app/live_privatezzzzzzzzzzzzzzzzzzzz [20:03] <mozzarella> but it doesn't work [20:03] <mozzarella> "Could not write header for output file #0 (incorrect codec parameters ?): Invalid data found when processing input" [20:06] <EvolE> well it can't read the file then, not write to rtmp [20:06] <EvolE> can you give me TEDxGE2014_Stallman05_HQ.ogg or it's too big? [20:07] <EvolE> i would like to try that [20:09] <Belliash> cvlc http://A.B.C.D/stream :sout="#transcode{acodec=mpga,ab=128,channels=2,samplerate=44100}:duplicate{dst=std{access=file,mux=flv,dst=-}}" | ffmpeg -i - -vcodec copy -acodec copy -f flv rtmp://A.B.C.D/stream [20:09] <Belliash> when i pipe it to VLC its working [20:09] <Belliash> looks like ffmpeg breaks audio [20:10] <mozzarella> EvolE: http://audio-video.gnu.org/video/TEDxGE2014_Stallman05_HQ.ogg [20:10] <EvolE> mozzarella: thx, will try that when after download [20:11] <mozzarella> I can watch it just fine with totem, so I don't think it's corrupt or anything like that [20:12] <EvolE> Belliash: try to set codec to aac or mp3 at least [20:12] <Belliash> ffmpeg -i - -vcodec copy -acodec mp3 -ac 2 -ab 128000 -ar 44100 -strict experimental -f flv [20:12] <Belliash> also not working [20:13] <EvolE> mozzarella: is there a torrent of that, speed is 300 kB/sec ( [20:13] <Belliash> when connecting to RTMP, vlc says audio uses undf codec [20:14] <mozzarella> EvolE: maybe I can upload it on a faster server? we'd need to find one first [20:15] <EvolE> Belliash: why vlc should say anything if it only sends it to ffmpeg? [20:15] <Belliash> EvolE: I can open another copy of VLC and connect to RTMP [20:16] <Belliash> then it says audio codec is undf [20:16] <Belliash> but when i do cvlc .... | vlc - [20:16] <Belliash> audio works pretty fine [20:18] <EvolE> did you try to mux in ts on input vlc ? [20:18] <EvolE> Belliash: [20:19] <Belliash> EvolE: yes [20:19] <Belliash> no difference [20:20] <EvolE> Belliash: is that input stream is on internet or it's local? can i try it? [20:21] <Belliash> EvolE: its local [20:21] <Belliash> do you need any information about it? [20:26] <EvolE> Belliash: just tried to stream with ffmpeg from file to twitch and it works [20:29] <EvolE> Belliash: aac and mp3 works for me [20:30] <EvolE> Belliash: have some lags, but im not sure why.. [20:31] <EvolE> Belliash: why don't you get input stream with ffmpeg? [20:33] <mozzarella> EvolE: does it work on your end? [20:34] <EvolE> Belliash: 3 min left for that file... [20:34] <EvolE> mozzarella: [20:34] <EvolE> oops) [20:41] <LiohAu> hey guys, I would like to understand one thing, is ffmpeg using x264? or is it using its own h264 codec ? [20:41] <LiohAu> or both are valid answers? [20:41] <JEEB> x264 is the only H.264 encoder available via libavcodec [20:42] <JEEB> but the decoder is an internal libavcodec one, and x264 has no decoders [20:43] <EvolE> mozzarella: you need to encode that video to h264 and audio to aac or mp3 to send it to twitch [20:43] <LiohAu> JEEB: I have a library that has a USE_X264 define, that is based on this: $(shell [ -f $(SYSROOT)/include/x264.h ] || [ -f $(SYSROOT)/local/include/x264.h ] || [ -f $(SYSROOT_ALT)/include/x264.h ]) [20:43] <LiohAu> the thing I don't understand, is that eitheir the library use x264, or it uses ffmpeg [20:44] <JEEB> x264.h is provided by x264 [20:45] <mozzarella> EvolE: how do I change it to h264? [20:45] <LiohAu> ok, so I understand the define, but why is this define used to make a choice between x264 vs ffmpeg since ffmpeg rely on x264 ? [20:45] <JEEB> please ask an esper [20:46] <JEEB> because I have no fucking idea about what the fuck you are talking about, you might just fucking go poke the people who made whatever you're trying to use [20:46] <EvolE> mozzarella: use "-vcodec libx264" in ffmpeg command line [20:47] <LiohAu> JEEB: nevermind [20:54] <mozzarella> EvolE: how can I pause, rewind, etc.? [20:54] <mozzarella> I'd like to be able to do that [20:55] <Belliash> why passing -vcodec h264 makes ffmpeg reencoding video which is already encoded using h264? [20:56] <JEEB> because you are specifically telling it to re-encode [20:56] <JEEB> everything else but -c:v (or -vcodec) copy will re-encode the video [20:57] <Belliash> can i somehow tell it to reencode only if source codec is different? [20:57] <JEEB> no [20:58] <JEEB> you will have to do a pass with either ffmpeg or ffprobe to see what the video track is [20:58] <JEEB> and then decide to either re-encode it or not [21:00] <foonix> btw, could someone look/test the 2.4 version for getting pixel format from source when using -probesize max? [21:04] <EvolE> mozzarella: if you want to rewind it while streaming live you better stream the screen and the video player on it) [21:06] <mozzarella> EvolE: ( ;____;) [21:07] <mozzarella> can I at least pause it? [21:09] <EvolE> mozzarella: and what will go to the stream if you pause it? [21:10] <mozzarella> EvolE: a still image [21:10] <EvolE> im not sure you can do it with ffmpeg) why not to stream with obs or stream full screen? [21:11] <mozzarella> EvolE: what's obs? [21:12] <EvolE> mozzarella: free streaming program on windows [21:12] <sacarasc> And OS X. [21:13] <mozzarella> EvolE: I use linux [21:17] <EvolE> mozzarella: then stream your screen with ffmpeg [21:17] <mozzarella> EvolE: how? [21:18] <EvolE> mozzarella: here is some sample script the guy showed yesterday: http://pastebin.com/HTnRGpSs you can take parameters from it probably [21:20] <xtruder> iam trying to send multiple bitrate/resolution streams to rtmp server, just like FME does (rtmp:.../streamkey1, rtmp:.../streamkey2), however server recognizes only one. Any idea why? (working fine with fme) [21:26] <EvolE> xtruder: how you doing that? [21:30] <xtruder> EvolE: ffmpeg -analyzeduration 10M -probesize 10M -re -i rtmp://url -c:a copy -c:v h264 settings -f flv "rtmp://key1" -c:a copy -c:v h264 settings -f flv "rtmp://key2" [21:31] <xtruder> tried forcing keyframes too, no effect [21:37] <mozzarella> EvolE: does it track the window? what if I move it around? [21:39] <EvolE> mozzarella: don't know, i didn't try that script [21:40] <dada__> hi there. would anyone know if it's possible to make a video containing h264 and flac? I tried the following https://dpaste.de/G0Es [21:40] <c_14> You can't put flac in mp4. [21:41] <dada__> would it work if I switched to mkv? [21:41] <c_14> yep [21:41] <dada__> thanks I'm gonna have a look to see which arguments I need to use for that [21:42] <dada__> this is for upload to youtube in a script so I can always just re-encode it but I'd prefer to copy it so that it's faster :) [21:43] <dada__> looks like I can just set out to tmp.mkv instead of tmp.mp4 and it works [22:04] <brontosaurusrex> dada__: you could use alac with mp4 i think [22:13] <dada__> whoops, changing output to .mkv actually broke all videos apparently [22:13] <dada__> now youtube can't process them [22:13] <dada__> brontosaurusrex: that's a nice idea, I'll try it [22:41] <rickbol> been successfully using MP4Box to concatenate 2 ffmpeg generated mp4 files together into a single mp4 file. Is there a way I can concatenate using ffmpeg? [22:42] <sacarasc> The concat filter might be able to do it. [22:49] <nicolas_> hello everyone , i would have a question, i want to convert a video track of an mkv file and leave everything else as it is (audiotrack, chaptermarks etc), this has to be agnotic of the input file (the number of auditracks changesfrom file to file ), has anyone an idea how i can achieve this ? [22:51] <rickbol> nicolas_: are you still storing into an mkv container? [22:54] <rickbol> sacarasc: I'm running an older version of ffmpeg (0.7.15). ffmpeg -filters doesn't list a "concat" filter. [22:54] <sacarasc> rickbol: Upgrade. [22:55] <sacarasc> Use one of them if you have to. :D [22:55] <nicolas_> rickbol: yes, old.mkv has an mpeg2 video track, and new.mkv should have everything the same but with an h264 video track [22:55] <dada__> would anyone know why the -t argument seems to be ignored here? https://dpaste.de/W2ea#L26,41 [22:56] <sacarasc> dada__: Try using it as an output option. [22:57] <rickbol> nicolas_: I'm using an old ffmpe, so I expect that the syntax I know wouldn't e correct for a current ffmpeg, but the audio is copied with -acodec copy [22:58] <rickbol> nicolas_: I don't know tje syntax to copy chaptermarks, etc. [23:03] <dada__> sacarasc: thanks [23:09] <rickbol> sacarasc: just downloaded the current static build, and ffmpeg -filters doesn't list a "concat" switch either. [23:10] <sacarasc> http://ffmpeg.org/ffmpeg-all.html#concat-3 [23:12] <rickbol> sacarasc: my bad! I printed the system version, not the new static build. :-/ [23:14] <Belliash> hey again [23:14] <Belliash> I got one more question... can I force ffmpeg to keep all audio tracks? [00:00] --- Mon Sep 15 2014

1 0

ffmpeg-devel.log.20140913
by burek 14 Sep '14

14 Sep '14

[00:53] <cone-879> ffmpeg.git 03Andreas Cadhalpun 07master:d5e802609a00: doc: document the addition of the AVProbeData.mime_type field and it's implications [00:53] <cone-879> ffmpeg.git 03Andreas Cadhalpun 07master:bcac0f401001: lavf/format.c: use AVPROBE_SCORE_MIME instead of AVPROBE_SCORE_EXTENSION for matching mime types [01:32] <llogan> once again FFmpeg distracts me from my boring work and now I'm behind. [01:33] <llogan> oh well. it's friday and I have a keg and liver to kill. [01:34] <llogan> FFmpeg Dev Days at my dumpartment. [08:21] <ux> erm do we really want the "Behaviour changes" section to document API changes? [11:28] <pross> vp10: http://www.cnet.com/news/googles-web-video-ambitions-run-into-industry-real… [11:30] <wm4> maybe they should have written a spec after all [11:30] <wm4> ""Our goal is to get codec development to Web speed," he said." wat [11:31] <nevcairiel> google doesnt really care about wide-spread adoption, they are happy if they can use it to save bandwidth with youtube -> chrome/android :p [11:32] <wm4> (that last code in combination with the intention to release a new codec all 2 years) [11:32] <wm4> s/code/quote/ [11:32] <J_Darnley> How are they going to make video useful on a mobile without anytime for someone to make a HW decoder? [11:33] <nevcairiel> phones are getting faster and faster every year, which also means energy requirements for software decoding are going down [11:33] <J_Darnley> Perhaps this time they'll get their alpha channel correct. [11:33] <pross> a new codec ever 18 months. quantity over quality :) its good to see the on2 monkeys are still running things [11:34] <wm4> J_Darnley: did they decide yet which of the 3 methods to use? [11:34] <J_Darnley> Although I have no idea why a video would want one. [11:35] <pross> J_Darnley: gif successor [11:36] <nevcairiel> i've actually seen that happen quite o ften recently, people starting to post small webm's instead of terrible-quality gifs [11:38] <J_Darnley> I don't see why that requires an alpha channel. All these gifa that should be videos are square. Usually because that are from videos. [11:39] <pross> VP10 will have spherical coordinate system :) [11:39] Action: J_Darnley wonders WTF is going wrong with his typing lately. [11:47] <rcombs> J_Darnley: it's useful within a production process [11:49] <rcombs> e.g. for overlay animation [11:50] <wm4> because video production will use lossy formats...? [11:50] <J_Darnley> Don't you have a "pro" codec that for that? [11:50] <rcombs> wm4: yes, actually [11:50] <rcombs> but it's usually ProRes or similar [11:51] <rcombs> the only use for alpha in video I can think of is& pre-rendered subtitle typesetting [11:51] <rcombs> *consumer video [12:17] <JEEB> lol at that VP10 article [13:27] <ubitux> the article doesn't mention that unless you have the infrastructure of google you can't actually bench vp9 properly [13:28] <ubitux> encoding 4k with libvpx? see you next year [13:28] <ubitux> maybe they improved the performances since last time though [13:28] <wm4> yeah, and practical HEVC encoders will be around much sooner than Vp9 [13:31] <JEEB> practical HEVC encoders are pretty much here now, even if x265 is crappy it has kind of climbed over the fence [13:33] <cone-641> ffmpeg.git 03Michael Niedermayer 07master:8572dbc5dbeb: avcodec/ac3en: use FF_ALLOC(Z)_ARRAY() [14:37] <ubitux> any idea when does the readorder really matters in ass muxed in mkv? [14:37] <ubitux> for the rendering itself it shouldn't make any difference, right? (that's the purpose of layer) [14:38] <ubitux> i suppose that's just in order to reconstruct the .ass file identically if necessary? [14:38] <wm4> ubitux: it also matters for rendering... somehow [14:38] <ubitux> how so? [14:39] <ubitux> isn't Layer for that purpose? [14:39] <wm4> if you have two events at the same times (same start/end TS), the one with the higher order is visible [14:39] <wm4> sure you can use layer to disambiguate this [14:39] <ubitux> mkv/ass has both [14:40] <wm4> but you can still write a subtitle file that doesn't use layer and has such overlapping events [14:41] <ubitux> mmh. [14:41] <ubitux> ok. [14:42] <wm4> not sure why it's stored in the file though [14:42] <ubitux> ReadOrder is mkv specific [14:42] <wm4> yes, but why does it have it [14:43] <wm4> it's somewhat useful to eliminate duplicate events though [14:43] <wm4> but in theory you wouldn't need it; you could recreate it on demuxing [14:43] <ubitux> i would guess it's in order to reconstruct the original .ass [14:43] <ubitux> typically if you stack OP,ED and then other events [14:44] <ubitux> and if you want to reconstruct the original .ass not based on the ts but that original file order [14:44] <wm4> oh right, muxing in mkv would force monotonic timestamps [14:44] <ubitux> for esthetic purpose [14:44] <wm4> while an ass file can arbitrarily shuffle them [14:45] <ubitux> i'm trying to consider making the .ass demuxer outputs ASS packets (the mkv muxed form) [14:50] <ubitux> saste: no opinion on the lavfi/ass patchset? [14:51] <ubitux> michaelni: do we use that AV_LOG_LEVEL granularity somewhere else? [14:51] <ubitux> i'm not exactly comfortable using that [14:54] <wm4> what lavfi/ass patchset? [14:55] <wm4> ubitux: for the libass log level thing, note that many log levels are rarely used or unused [14:55] <wm4> I don't think trying to map them to libav* ones in an overly sophisticated manner will help with anything [14:55] <ubitux> < wm4> what lavfi/ass patchset? // the 3 small patches you seem to have found just now [14:55] <ubitux> wm4: that's my thought as well [15:16] <cone-641> ffmpeg.git 03Clément BSsch 07master:d86cf4a91de2: avformat/vobsub: fix NULL dereference [15:23] <michaelni> ubitux, dunno, but intermediate levels where definitly intended to be usable [15:25] <ubitux> ok [15:25] <ubitux> for ass that's hardly testable anyway [15:25] <ubitux> these intermediate levels are not used so far [15:32] <cone-641> ffmpeg.git 03Clément BSsch 07release/2.3:148d9cd12227: avformat/vobsub: fix NULL dereference [15:34] <ubitux> olol just found a mkv in my samples that has the Dialogues with timestamps [15:34] <wm4> ubitux: huh? [15:35] <ubitux> mmh my bad [15:36] <wm4> relief! [15:36] <ubitux> yeah [15:37] <ubitux> i was in my backport state [15:37] <ubitux> where the demuxed was fucking the packets :p [15:37] <ubitux> demuxer* [15:39] <ubitux> ffprobe -show_packets -show_data e [16:34] <ubitux> nice the latest notice from freenode [16:42] <J_Darnley> If there is a breach perhaps that is a fake message which is now capturing password changes. [16:44] <wm4> J_Darnley: nice one [16:44] <J_Darnley> paranoia++; [16:45] <J_Darnley> :) [16:45] <J_Darnley> I changed my password anyway. What can they get out of the old or new random strings? [16:46] <JEEB> hm? [16:46] <J_Darnley> If you didn't get the message... [16:46] <JEEB> oh [16:47] <J_Darnley> [Sat 16:33] -tomaw- [Global Notice] Earlier today the freenode infra team noticed an anomaly on a single IRC server. We have identified that this was indicative of the server being compromised by an unknown third party. [16:47] <JEEB> finally found it in the backlog of #ffmpeg [16:47] <JEEB> xchat being useful and pushing notices from servers to a random channel's log [16:47] <J_Darnley> I won't bother with the rest for now [16:47] <wm4> JEEB: not hexchat? [16:48] <J_Darnley> Yeah, mIRC isn't any better. My message was in #lua [16:48] <JEEB> no, not hexchat. I'm using the official builds due to a moment of rage back in '07 or so [16:48] <JEEB> when I actually went and bought a license for it [16:49] <Case> J_Darnley: I'm sure mirc has config for that. My global notices always go to server window [16:49] <J_Darnley> hmm [16:50] <Case> Options -> IRC and there "Show in active" [16:50] <Case> untick everything and they appear in server window [16:51] <Case> at least that's my assumption [16:51] <Case> I have only checkmark on Whois [16:51] <J_Darnley> ditto [16:53] <J_Darnley> are they "ctcp" ? [16:54] <J_Darnley> Ifo then it might be in under the "Events" button on that page. [16:54] <J_Darnley> *if so [16:56] <Case> now I'm actually not sure if I get global notices in server window all the time or if I just happen to have it active every time global notice comes [16:56] <Case> when I'm not actively following anything I go to server window so it's easy to spot any channel that has activity [17:15] <cone-934> ffmpeg.git 03Michael Niedermayer 07master:b11d1889ef60: avcodec/bmp_parser: fix parsing a single bmp which has a fsize < its header [17:58] <cone-934> ffmpeg.git 03Michael Niedermayer 07master:3c6d824b8002: avcodec/bmp_parser: simplify [17:58] <cone-934> ffmpeg.git 03Michael Niedermayer 07master:ceb2fe027fec: avcodec/bmp_parser: delay frame end detection to the next header or EOF [18:17] <ubitux> is "\n" (0x5C 0x6e) really supported in ASS? [18:17] <ubitux> (and not "\N") [18:21] <wm4> ask vsfilter [18:22] <JEEB> #irrational-typesetting-wizardry might be useful, on rizon [18:22] <JEEB> nielsm/jfs and friends :3 [18:22] <ubitux> i have a mkv with those at least [18:41] <JEEB> 2: No word wrapping, wide lines will extend beyound the edges of the screen. Both \n and \N force line breaks. [18:41] <JEEB> looking at @ http://docs.aegisub.org/3.2/ASS_Tags/ [18:42] <JEEB> <Plorkyeran> it's treated as \N in one wrapping style and as a space in all others [18:42] <JEEB> <Plorkyeran> i.e. it's dumb and useless [18:43] <JEEB> <jfs> yes the \n versus \N really dates back to the SSAv3 to SSAv4 transition, as far as I understand [18:43] <ubitux> haha [18:43] <ubitux> ok [18:44] <JEEB> basically old SSA didn't have automatic line wrapping [18:44] <JEEB> so old scripts were full of linebreaks [18:45] <JEEB> <jfs> but to let subbers take advantage of automatic breaking without changing scripts, the \n break was made into an ignored code except in one particular mode [19:06] <kepstin> hmm, looks like my K6 w/mmx box is in working order with a linux distro installed. I guess I should set up FATE on it and see how long it takes to run :) [19:09] <J_Darnley> At least it has mmx! [19:12] <kepstin> no 3dnow tho; it's one of the older k6 models. [19:14] <kepstin> that's hilarious; the linux distro installed on it is gentoo. [20:33] <cone-934> ffmpeg.git 03Clément BSsch 07master:b96d864fd685: avcodec/mjpegdec: Fix chroma width rounding [21:08] <ubitux> michaelni: can i ask for the assenc patch i just sent to be part of 2.4? [21:09] <ubitux> just to be clear, it allows to reconstruct .ass files perfectly from mkv's [21:09] <ubitux> (it reorders the line as they were originally) [21:09] <ubitux> lines [21:12] <michaelni> ubitux, it will be part of 2.4 if its pushed before we branch release/2.4 [21:12] <michaelni> or if its backported [21:12] <ubitux> that's why i'm asking for a special review before you branch out ;) [21:12] <ubitux> (you're the author of the assenc afaik) [21:13] <ubitux> (libav just released 11) [21:19] <jamrial> just wondering, since this version is the one that may make it to debian, how about making it ffmpeg 3.0 instead of 2.4? [21:19] <jamrial> it's also the first version after we bumped major for all libraries [21:19] <ubitux> you sound quite optimistic jamrial [21:20] <ubitux> it's been dying on the "NEW" queue for almost 6 months and you think something is going to change? [21:20] <jamrial> i did say may :P [21:20] <ubitux> :) [21:20] <ubitux> that's too much optimism already [21:20] <ubitux> joke aside, i'm fine with 3.0 but the changelog will be quite poor for a major release [21:21] <jamrial> true, it's too close to 2.3 after all [21:26] <jamrial> wm4: utf16 support isn't mentioned in changelog. it probably deserves a line [21:35] <michaelni> ubitux, review sent [21:38] <michaelni> ubitux, also if the entries are almost sorted often then keeping track of the last element of the linked list and adding there should be faster than searching through the whole list [21:38] <michaelni> i think the code searches from the start for inserts [21:38] <ubitux> yeah right [21:39] <michaelni> might be good enough instead of a change of the data structure to something else [21:40] <michaelni> i dont really know how badly these elements are sorted normally [21:41] <ubitux> maybe i should start from the end instead [21:41] <ubitux> i can give you a real case example of the evolution of the list, just a moment [21:45] <ubitux> michaelni: http://pastie.org/pastes/9551335/text [21:48] <ubitux> but actually from the start looks like the best thing to do [21:48] <ubitux> because that's likely where you add it [21:50] <ubitux> there is one case though where it might matters [21:50] <ubitux> is when the ReadOrder starts at 1 [21:50] <ubitux> and the muxer has to wait until every single Dialogue is cached [21:51] <ubitux> in this case, every events will have to be added at the end [21:51] <ubitux> i guess we can add an initial check for that [21:52] <ubitux> like "is it to be added at the end?" with fallback start lookup from the beginning [21:52] <michaelni> there are many options, you also could start the search from where the last element was inserted [21:53] <michaelni> that would help with cases like 1 4 7 9 2 3 5 8 10 6 [21:54] <ubitux> in the common use case i pasted above it's the least optimal [21:54] <ubitux> like, adding 43 when you have "59 60 61 62 63 64 65 66 67 68 69 70 71" [21:54] <ubitux> ah mmh wait [21:54] <ubitux> my bad [21:54] <ubitux> 59 wasn't the latest added [21:54] <ubitux> yeah right, i guess i can do that [21:55] <michaelni> theres also libavutil/tree.h [21:55] <michaelni> or you could write a splay tree implementation [21:55] <michaelni> nearly infinite options :) [21:55] <ubitux> won't that be totally overkill? [21:55] <michaelni> yes [22:05] <ubitux> michaelni: so something along those lines would be fine with you? http://pastie.org/9551362 [22:05] <ubitux> mmh missing something. [22:07] <ubitux> fixed @ http://pastie.org/9551365 [22:07] <ubitux> so looking from last added to the end, then beginning to last added [22:13] <michaelni> ubitux, looks fine from a quick look [22:13] <ubitux> cool [22:13] <ubitux> then please poke me before you plan to branch out 2.4 [22:14] <cone-934> ffmpeg.git 03Michael Niedermayer 07master:296cd9c43277: avformat/mpegts: Improve probe heuristic by considering the overall frequency of 0x47 headers [22:14] <ubitux> (thank you) [22:16] <michaelni> ubitux, i intend to make the branch today or tomorrow, that is intend that does not mean i will [22:17] <michaelni> that is assuming i dont run into any major issues [22:18] <ubitux> ok, i'll push in the next hour then [22:23] <michaelni> ok [22:38] <cone-934> ffmpeg.git 03Michael Niedermayer 07master:8407cbbfc942: Fix "passing argument 1 of av_free discards const qualifier from pointer target type" [22:38] <cone-934> ffmpeg.git 03Michael Niedermayer 07master:52c85b194b2d: avdevice/lavfi: dont assign variables to themselfs [22:44] <ubitux> Command:This is a "command" event, which means SSA will execute the specified program as a background task. [22:44] <ubitux> omg [22:45] <iive> ubitux: is that a joke? [22:46] <ubitux> iive: http://moodub.free.fr/video/ass-specs.doc [22:46] <ubitux> that file name is already a joke, but well. [22:47] <ubitux> Sound: This is a "sound" event, which means SSA will play the specified .wav file. [22:47] <ubitux> Movie: This is a "movie" event, which means SSA will play the specified .avi file. [22:47] <Compn> so many people have problems with the word 'ass' :P [22:47] <ubitux> i like those as well btw [22:50] <J_Darnley> it just means "donkey"... [22:51] <ubitux> Compn: "ass", "specs" and "doc" are the jokes [22:56] <ubitux> does anyone know if there are mkv with other event than Dialogues muxed in? [22:56] <ubitux> such as Comment [23:08] <Timothy_Gu> ubitux: WIP release notes: https://gist.github.com/TimothyGu/cadf487de75189e49b4e [23:09] <Timothy_Gu> I find it kinda hard to pick out the "most exciting features" in 2.4 [23:10] <ubitux> just a moment, i'll have a look [23:10] <Timothy_Gu> UTF-16? dctdnoiz? Icecast? [23:11] <ubitux> no need to make a paragraph on that [23:11] <ubitux> just tell that we released 2.3 very recently so the Changelog is short [23:11] <Timothy_Gu> What's the difference between dctdnoiz and owdenoise and hqdn3d? [23:11] <ubitux> but OTOH this release is meant to be LTS (need confirmation from michael) if debian follow it [23:11] <Timothy_Gu> Ok [23:12] <ubitux> dctdnoiz is DCT based, with an original color correlation, owdenoise is wavelet based, and hqdn3d is... old/dunno [23:13] <cone-934> ffmpeg.git 03Clément BSsch 07master:89d42da5c62c: avformat/assenc: honor ReadOrder [23:13] <ubitux> Timothy_Gu: you can add that ffmpeg now reconstruct .ass files from .mkv in original order now [23:13] <ubitux> :p [23:15] <ubitux> Timothy_Gu: please drop the AVProbe thing from that file [23:15] <ubitux> it really doesn't belong here at all [23:16] <Timothy_Gu> OK... [23:18] <jamrial> Timothy_Gu: you're missing a couple things present in changelog, like h261/h265 in RTP or LZMA compression in TIFF [23:19] <jamrial> also, "FFmpeg 2.4 is API-, but not ABI-compatible with the previous major release" is not exactly true http://upstream-tracker.org/compat_reports/ffmpeg/2.3.3_to_current/src_comp… [23:31] <ubitux> Timothy_Gu: you might want to look for asm contributions [23:31] <ubitux> and summarize them in a sentence [23:31] <ubitux> iirc there is at least idet that got some improvements asm wise [23:32] <nevcairiel> hevc got transform_add asm, it makes no big dent in overall decoding time, but if you want to mention stuf... [23:32] <cone-934> ffmpeg.git 03Michael Niedermayer 07master:36ea35bbc0e3: avformat/utils: free s->pb for image2 as it can be used with and without a file [23:34] <jamrial> IMO, better wait until we get sao and idct in the tree to do that [23:34] <ubitux> still no idct? really? :( [23:36] <jamrial> it's written as intrinsics in openhevc [23:36] <ubitux> is plepere working on this or someone else? [23:36] <jamrial> sao as well, but plepere already ported it to yasm (or is in the middle of it). he hasn't posted it to the ml yet, though [23:36] <jamrial> parts of it at least [23:37] <nevcairiel> i was surprised how much sao asm really helped the performance, didnt expect that really [23:37] <jamrial> https://github.com/OpenHEVC/FFmpeg/commits/rext [23:45] <nevcairiel> hm, the x265 lead keeps advertising their own hevc decoder and always downplays ffmpeg, implying it lacks features and whatnot (although he doesn't say it, but he always makes a point to say stuff like"and ours even supports 10-bit"), its really annoying me sometimes :P [23:48] <JEEB> yeah, MCW seems to have a history of marketing bullshit uber alles [23:52] <cone-934> ffmpeg.git 03Michael Niedermayer 07master:c2a7607a1ca6: RELEASE_NOTES: Add next versions name (Fresnel) [23:52] <jamrial> did he imply we don't support 10bit? because we do. 9 and 12bit as well, even... [23:53] <nevcairiel> well i read it that way anyway, if you compare your decoder to ffmpeg one line above, and in the next line write "we support the full spec, including 10-bit", i read it that way [23:53] <nevcairiel> did i lie to him in claiming that we support RExt btw? i saw a bunch of changes and new pixel formats, but i'm not 100% sure its actually entirely functional [00:00] --- Sun Sep 14 2014

1 0

ffmpeg.log.20140913
by burek 14 Sep '14

14 Sep '14

[00:43] <BtbN> Is it possible to tell ffmpeg to transcode in realtime? [00:43] <c_14> -re [00:43] <BtbN> Like, even if it can go 500 fps, it should stick with the input material framerate? [00:43] <vlatkozelka> yep -re [00:43] <vlatkozelka> that helped me alot btw c_14 :) [00:43] <llogan> as an input option [00:47] <BtbN> ah, it's an input option! [00:47] <BtbN> yep, that's why i didn't find it propably [00:49] <BtbN> wow, twitch realy didn't like me sending them an entire 20 minute video in like 30 seconds [00:49] <berzerka> hi. i built the latest ffmpeg version because i want to link vlc to it. it seems that all symbols are stripped off the object file on installation. [00:49] <BtbN> it's completely broken now, looping over the same 6 seconds since 5 minutes [00:50] <berzerka> the version in the source tree shows all the symbols, but in the installed .so there are no symbols defined. [00:53] <berzerka> just found --disable-stripping, lets give it a try.. [01:57] <berzerka> which ffmpeg version supplies libavcodec 55? [02:00] <berzerka> oh, got it from http://www.ffmpeg.org/download.html [02:06] <Baked_Cake> finally got my homegroup working [02:06] <Baked_Cake> think my problem was boh pc's had the sameo name [02:07] <Baked_Cake> oops [05:01] <valder21> hey all wondering if somebody can do a quick code review on my audio decoder stuff. I can't figure out how to add sound to my videos. All of the videos come out silent even though I believe I am writing to the stream audio samples. [05:01] <valder21> I've posted the issue on SO : http://stackoverflow.com/questions/25813484/what-am-i-doing-wrong-with-my-a… There is a link there to the full source of my project. [05:03] <valder21> I also get this error when I try to open my audio codec, I can't find any reference as to what it means: "Could not open audio codec: Experimental feature" This happens in the open_audio function. [07:52] <haspor> hello, anyone could help me with decoding adpcm_swf to pcm data ? [08:37] <bencc> when trying to build on ubuntu 14.04 with --enable-x11grab I'm getting intrin.h: No such file or directory [08:48] <relaxed> bencc: did you install build-essential? [08:57] <bencc> relaxed: yes [08:57] <bencc> relaxed: I'm trying to use this https://github.com/pyke369/sffmpeg [08:59] <bencc> relaxed: your static build doesn't include enablex11grab? http://johnvansickle.com/ffmpeg/ [09:16] <Reelee> Can someone please help me use -map on MPTS stream to select specific audio channel? Here is part of the MPTS stream: http://pastebin.com/kRyxTLZC. Thank you! [09:21] <relaxed> bencc: correct [09:22] <Nosomy> -map 0:28 for get a mdk. [09:22] <relaxed> bencc: apt-get build-dep libav-tools [09:27] <michaelni> haspor, can you describe the problem and show us the code that fails [09:28] <Reelee> Nosomy: first I use -map 0:p:11 to get Program 11, but when I use -map 0:28 as 2nd parameter, I get "Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument" and "sample rate not set" [09:29] <haspor> all right, i have flv file that has audio stream (adpcm_swf), i want to decode that to raw pcm data, its 22050hz mono s16 stuff [09:30] <haspor> but all i get is crackling and static noise [09:30] <haspor> with mp3/aac it works just fine [09:40] <Nosomy> last ffplay build, still with problem of playback mkv files with hevc stream, 'm having to use an old version, which works well for this. [09:50] <valder21> Hi all, I am looking for help. I don't understand how to get my audio to be heard on my output video. I'm pretty sure I'm following examples I've seen around the web closely. Am I not accounting for PTS/DTS properly? All of my videos come out silent. When I try to write out an audio frame it results with an error (code -22). "Invalid Argument" I can't seem find what that really means if I'm calling [09:50] <valder21> avcodec_decode_audio4() [10:11] <valder21> ? [10:13] <Fjorgynn> :/ [10:13] <Fjorgynn> I am sleeping [10:13] <Fjorgynn> please be atient [10:13] <Fjorgynn> patient [10:23] <Nosomy> ffplay version N-66249-gbcac0f4 http://pastebin.com/2aFqMywt <- mkv with hevc, playback fail [10:40] <bencc> relaxed: I'll try it. thanks [10:54] <bencc> I'm trying to capture desktop and sound with [10:54] <bencc> ffmpeg -video_size 1024x768 -framerate 25 -f x11grab -i :0.0+100,200 -f alsa -ac 2 -i pulse output.flv [10:54] <bencc> getting Unknown input format: 'alsa' [10:54] <bencc> full log http://dpaste.com/0PTADJQ [10:55] <bencc> this is how I built ffmpeg http://dpaste.com/3JQN84J [12:55] <relaxed> bencc1: install libasound2-dev and recompile ffmpeg [12:56] <relaxed> it will tell you the supported indevs in configure's output. Make sure alsa is listed. [13:18] <bencc1> relaxed: works thanks [13:19] <bencc1> I'm using the following command but the audio and video are out of sync [13:19] <bencc1> /usr/bin/ffmpeg -f pulse -ac 1 -i alsa_output.pci-0000_00_05.0.analog-stereo.monitor -f x11grab -framerate 15 -s 1280x960 -i :0.0 -acodec pcm_s16le -vcodec libx264 -preset:v ultrafast -crf 0 -threads 0 capture.mkv [13:19] <bencc1> what can I do to make the video and audio in sync? [13:45] <relaxed> bencc1: try -r 60 instead [14:06] <haspor> here is the code im currently using to get audio stream from flvs: https://gist.github.com/anonymous/41155aa9d9a1ff933c99 [14:07] <haspor> i have no idea what might be wrong [15:03] <mrskman> Hi, is there any way how to set default audio stream in mkv? [15:05] <Peppery> Hi there, I am using this command to rotate and overlay a PNG over a video http://pastebin.com/ebWWj4gP - is there any way to have the PNG fill the entire video rather than manually specifiyng width and height? [15:18] <c_14> Peppery: none I know of [15:19] <Peppery> thanks, c_14. that's the conclusion I was coming to [15:22] <c_14> mrskman: technically with `-metadata:s:a:0 default=1', but iirc ffmpeg does not currently write it correctly. You can use mkvtoolnix to set it. [15:31] <mrskman> same with default/forced subtitles? [15:32] <c_14> yep, ffmpeg will copy them if the input file has them set as default/forced, but it won't set them if they're not. [15:33] <mrskman> ok, thanks [16:41] <gcl5cp> is there a python biding updated? [16:42] <klaxa> i don't think so [17:34] <bencc1> relaxed: "-r 60" instead of "-framerate 16" ? I'll try it. can you explain why? [19:14] <Mic_> Hi all. I'm new to ffmpeg. I am a student doing a case study on the project. I have the following question: I'm trying to gain some insight into the security vulnerabilities that were fixed in previous versions on the security page of the ffmpeg website, but I don't understand what's written there. It just shows something like the following: CVE-2014-5271 60bfa9154d0084bc8b007b984051a6bb82d9652c [19:15] <Mic_> What exactly does this mean and how can I get some more information about what was fixed? [19:15] <Mic_> Thanks to anyone who can help! [19:16] <JEEB> git clone git://git.videolan.org/ffmpeg.git && cd ffmpeg && git show 60bfa9154d0084bc8b007b984051a6bb82d9652c [19:16] <JEEB> or http://git.videolan.org/?p=ffmpeg.git;a=commit;h=60bfa9154d0084bc8b007b9840… [19:17] <JEEB> and then looking at the diff you can see what changed in both cases [19:40] <haspor> anyone could finally help with this: http://pastebin.com/veHU8VBx, why it fails to decode ADPCM_SWF [19:43] <Mic_> oh i see. thanks JEEB [21:09] <bencc1> when capturing the screen, is it possible to get the whole screen instead of specifying the resolution? [21:09] <bencc1> I'm using [21:09] <bencc1> ffmpeg -f pulse -ac 1 -i alsa_output.pci-0000_00_05.0.analog-stereo.monitor -f x11grab -framerate 15 -s 1280x960 -i :0.0 -acodec pcm_s16le -vcodec libx264 -preset:v ultrafast -crf 0 -threads 0 capture.mkv [22:36] <valder21> can somebody point me to an example of taking audio from an MP4 file and splicing it into another? I've tried a few different examples I've found off SO and ffmpeg examples. I just get silent videos. Here's my current code: [22:36] <valder21> http://pastebin.com/4ax7XMur [22:38] <valder21> the work is done in the renderMovieRequest function. primarily between lines 995 and 1038 [22:38] <valder21> do I need to convert the audio context to another format first? Am I missing a step? [22:44] <_dunno_> i have a huge file, like 10h+, and want to cut chunks of that file to different files: ffmpeg -i long_vid.mpeg2 -ss start1 -t end1 output1.avi ... -ss startn -t endn outputn.avi is this possible? or in other words - does ffmpeg know about such thing as a 'cutlist'? [23:40] <gcl5cp> pip search ffmpeg --> ffms, wffmpeg, ffmpegwrapper, myffmpeg. are they updated? [00:00] --- Sun Sep 14 2014

1 0

ffmpeg-devel.log.20140912
by burek 13 Sep '14

13 Sep '14

[00:08] <J_Darnley> I wonder if that email I just sent is verbose enough. [00:17] <jamrial> michaelni: don't forget to cherry pick c619e14c314b44d86a8d552259afb957c0b6775d as part of the fix for ticket #3943 [00:21] <rcombs> J_Darnley++, re: transforming -- into [00:26] <michaelni> jamrial, backport where, and what fix ? [00:27] Action: michaelni looks at "Commit: Carl Eugen Hoyos <cehoyos(a)ag.or.at>" [00:28] Action: michaelni looks at fflogger [00:28] <michaelni> jamrial, "<cone-520> ffmpeg.git Michael Niedermayer release/2.1" <--- carl backported these not me [00:33] <jamrial> ah, my bad. forgot to check who committed them [00:35] <jamrial> ok, i'll do it [00:38] <cone-722> ffmpeg.git 03James Almer 07release/2.1:fc82ba06ee87: avformat/oggparseopus: Check opus_duration() return value [00:52] <J_Darnley> rcombs: I just decided to stick what I saw in a @code section like the many other commands [00:55] <Timothy_Gu> ubitux, jamrial: What's the difference between the pixelutils SAD and the me_cmp sad? [00:58] <jamrial> the pixutils one was made to remove lavc dependency to several filters [01:06] <Timothy_Gu> jamrial: why did ubitux rewrite some asm for pixelutils instead of just using the avcodec one? [01:06] <Timothy_Gu> Also 94865 warnings on icc WTF: http://fate.ffmpeg.org/report.cgi?time=20140911211522&slot=x86_64-linux-gnu… [01:09] <nevcairiel> its full of crazy wrong warnings [01:15] <jamrial> because he also made some changes to it, like removing the MpegEncContext argument, or using separate linesizes for each source [01:20] <BBB> see, I tought something was broken [01:20] <BBB> I hadnt received any chats in days [01:50] <J_Darnley> Timothy_Gu: I'll address your comments, and anyone else's, in the morning. [01:50] <Timothy_Gu> J_Darnley: What's your time zone? I.e. what is "morning"? It's 4:50 PM here. [01:51] <J_Darnley> +2 [01:51] <J_Darnley> utc+2 that is [01:51] <Timothy_Gu> WTH you are awake 2:00 in the morning?? [01:52] <Timothy_Gu> But OK. Have a nice sleep! [01:52] <J_Darnley> Yes. I'm a lazy git out of work at the moment. [03:26] <cone-722> ffmpeg.git 03James Darnley 07master:e3fb2b0eb7c5: docs: add example around the suggested commit message format [03:30] <Timothy_Gu> holy shit I found an ffmpeg-0.3.2.tar.gz with timestamps from 2001 [03:33] Action: Timothy_Gu stashes that into his stash of pre-0.5 tarballs https://launchpad.net/ffmpeg/pre-0.5 [03:33] <wm4> open source archeology? [03:58] <Timothy_Gu> yep :) haven't found 0.3, 0.3.1, and 0.3.3 though [03:58] <Timothy_Gu> http://www.filewatcher.com/ really helps [04:00] <wm4> and 0.1, 0.2? [04:00] <wm4> do they even exist? [04:00] <Timothy_Gu> nope, only up to 0.3. See Changelog [04:01] <wm4> ah [04:53] <jamrial> ^that ticket could be considered a missing documentation case [04:54] <jamrial> there's apparently no x265 entry in encoders.texi [05:01] <Timothy_Gu> jamrial: Agreed. You'll have to ping Daemon404 though [05:02] <jamrial> i reopened the ticket and edited it to reflect this [10:04] <wm4> ubitux: apparently "milkdrop" is the best audio visualization ever [10:46] <ubitux> wm4: that's because you haven't seen the one from J_Darnley yet :s [10:50] <wm4> hm [10:52] <rcombs> ubitux: any chance you have example output? [10:52] <ubitux> i have no idea what it looks like :) [10:55] <rcombs> I wonder how practical it would be to port milkdrop to& not D3D [10:56] <wm4> rcombs: there's a GL port [10:56] <wm4> used by audacious [10:57] <rcombs> ProjectM? [10:57] <rcombs> (as opposed to the Melee-for-Brawl thing) [10:59] <wm4> yes [10:59] <rcombs> projectM-emscripten <-- wat [10:59] <wm4> you want visualizations in the browser too [11:00] <wm4> also you'll have to get rid of your "wat" whenever emscripten is mentioned [11:00] <wm4> this kind of stuff is "normal" now [13:44] <wm4> michaelni: I'm starting to think image2 patterns should be treated as a protocol [13:44] <wm4> think: image://somethng%02.jpg [13:44] <wm4> mplayer also does this (but with mf:// ) [13:44] <wm4> no more probe trouble [13:58] <thardin> no a bad idea [14:07] <thardin> not [14:53] <michaelni> wm4, i think we should fix the bug and not connect it to a redesign that will break other things like http and ftp with image%d.jpg [14:54] <michaelni> that is IMO apply one of the patchsets and leave the redesign to a seperate future patchset [15:14] <Compn> michaelni / wm4 : yes, i agree. adding a new way ex. mf://*jpg is better than changing the old design [15:14] <Compn> and breaking scripts :\ [16:21] <benoit-> michaelni, wm4: I'm fine with the strcmp("image2") as it seems better to rethink the design afterwards [17:16] <ubitux> michaelni: what is the finer level values you are talking about for the ass patch? [17:16] <ubitux> the value "3" is basically not used in libass [17:17] <ubitux> i dupped the 2nd value, in case something come up [17:17] <ubitux> same for the info one [17:17] <ubitux> the previous code had a finer granularity, but it was wrong [17:29] <ubitux> https://projects.archlinux.org/svntogit/packages.git/commit/trunk/PKGBUILD?… wtf [17:29] <ubitux> any idea what's going on here? [17:39] <nevcairiel> some dude with no clue how it works? [17:42] <nevcairiel> they also have --enable-dxva2 in a linux build script, so ... [17:44] <Compn> ubitux : maybe theres some disabling being done by user ? [17:44] <Compn> and they need to force these codecs enabled ? [17:44] <Compn> maybe its causing some build error somewhere [17:44] <ubitux> no, it's just braindead [17:44] <Compn> yes it looks braindead. [17:44] <ubitux> i tried that configure line with and without [17:44] <ubitux> the config.h is the same [17:45] <ubitux> (except the configure line) [17:45] <ubitux> nevcairiel: missed that... [17:46] <ubitux> i'm wondering about --enable-pic as well [17:46] <ubitux> and --enable-postproc which should be enabled as well [17:47] <nevcairiel> well i have a few redundant enables in my configure as well probably, but explicit decoders and enable-dxva2 on the wrong platform is really over the top :D [17:50] <ubitux> the guy is telling me that's for some coop with PPSSPP or something [17:50] <ubitux> that atrac3 used to be disabled without explicit flag (wtf?) [17:51] <ubitux> and the guy don't see anything wrong with keeping them in here [17:51] <ubitux> .. [18:02] <ubitux> nevcairiel: ok that's the same guy [18:02] <ubitux> https://projects.archlinux.org/svntogit/packages.git/commit/trunk/PKGBUILD?… [18:02] <ubitux> he doesn't seem to have much clue [18:02] <ubitux> this was reverted (fortunately), but see: https://projects.archlinux.org/svntogit/packages.git/commit/trunk/PKGBUILD?… [18:02] <ubitux> (the _asm part) [18:03] <kepstin-laptop> hmm, I was under the impression that some of the ffmpeg assembly didn't work on 32bit with pic enabled, because pic required an extra register. [18:04] Action: kepstin-laptop wasn't responsible for the arch package, of course. [18:04] <ubitux> then maybe --enable-pic should be dropped? [18:06] <kepstin-laptop> gentoo allows conditionally enabling pic with a use flag; I think it's required for some of the optional security-related stuff gentoo supports. But I don't think arch has anything like that. [18:13] <michaelni> ubitux, i meant that you could do something like (AV_LOG_WARNING + AV_LOG_ERROR)/2 if you want a intermediate value [18:13] <ubitux> ah, we can do that? [18:13] <michaelni> yes [18:20] <Compn> kepstin-laptop : its possible, theres an outstanding bug in gcc with ffmpeg. ehe [18:20] <Compn> i think they closed it though :\ [18:30] <cone-305> ffmpeg.git 03Michael Niedermayer 07master:321c3cd1a97b: avformat/img2dec: reduce bmppipe probe score [18:30] <cone-305> ffmpeg.git 03Michael Niedermayer 07master:917d14e6a20d: avformat/format: Run image2 probe again when file content data is available [19:02] <cone-305> ffmpeg.git 03Michael Niedermayer 07master:b3fd2b175c90: avformat/img2dec: Fail probing when no data is yet available and the filename contains no number/glob patterns either. [22:31] <J_Darnley> Oh my god! A patch for 586 cpus? [22:32] <iive> well, this means somebody is using it. [22:36] <J_Darnley> Perhaps they have just finished their last fate run and seen the failure. [22:36] <nevcairiel> you mean the first fate run since the cpu was new 20 years ago? [22:36] <nevcairiel> :D [22:37] <nevcairiel> it just took this long [22:37] <J_Darnley> That is the joke I tried and failed to make. :( [22:37] <nevcairiel> well the one change looks sensible, should disable i686 when using --cpu=i586 [22:37] <nevcairiel> the others.. i dont like [22:55] <kepstin-laptop> hmm, I should try to get my k6-2 box up and running, I might be able to get fate running on there. [22:57] <cone-305> ffmpeg.git 03Mikulas Patocka 07master:cdb3eee7c496: configure: Fix miscompilation for i586 [00:00] --- Sat Sep 13 2014

1 0

ffmpeg.log.20140912
by burek 13 Sep '14

13 Sep '14

[00:00] <tab1293> how can I specify the output to be a mpeg audio elementary stream? [00:00] <_dunno_> okay - it's something about the aspect ratio - when i halve x and y coordinates i can also downscale the video - why? and how to overcome? [00:03] <_dunno_> tab1293 where's the prob? if u'r using the right acodec and u'r still running into probs, maybe atag may solve it [00:04] <tab1293> _dunno_: here is the command I am running to transcode a mp3 file to a AAC HLS stream [00:04] <tab1293> ffmpeg -i sample.mp3 -codec a libfdk_aac -b:a 128k -map 0:a:0 -f segment -segment_time 10 -segment_list outputlist.m3u8 -segment_format mpegts 'output%03d.m4a' [00:05] <tab1293> with this command each segment is a mpeg transport stream [00:05] <tab1293> I would like it to be an elementary stream instead [00:15] <valder> is there a way to have av_dump_format output to Logcat? [00:27] <_dunno_> it'S quite trivial - when scale=x:y are not multiples of crop=x:y you have to put '-aspect cropX:cropY' to the cmdline [00:58] <pyBlob> is there an ffmpeg native filter that does the same as mp=eq2? [01:11] <pyBlob> ... the curves filter seems to be it [01:17] <ac_slater_> I know I should only say nice things ... but man libavformat/codec are not at all friendly. Documentation is scarce and I literally have to run everything through a dugger/callgraph to find out the simplest things - ie - how to set video size on rawvideo decoder/demuxer [01:17] <ac_slater_> +1 for ffmpeg, -1 for bad/lack of documentation :( [01:33] <c_14> Did you try reading the doxygen? [01:39] <ac_slater_> c_14: yea, it does it's just to tell you type info, etc, and there are some things regarding simple usage [01:39] <ac_slater_> c_14: sadly, it doesnt really do much to explain the plugin process or creating new ones. [01:39] <ac_slater_> :( [01:47] <llogan> ac_slater_: was anything in doc/examples useful? [01:48] <dahat> Q: I'm developing an app to play basic videos with FFMPEG as a backend... and currently it accepts a local path into avformat_open_input(), now I am trying to expand it to target RTMP files, currently I can do the following at the command line, how can I apply the same args to avformat_open_input()? ffmpeg -rtmp_app <AppName> -rtmp_conn <AuthToken> -rtmp_playpath <PlayPath> -i rtmp://streaming.example.com/ [01:49] <ac_slater_> llogan: sort of. I guess I always seem to do the things that boilerplate examples wont cover. They were helpful getting started though. [01:50] <ac_slater_> dahat: one way to start, is to compile the ffmpeg binary in debug mode, run `cachegrind` on it and get a call graph. [01:50] <ac_slater_> dahat: that's time consuming though [01:55] <dahat> ac_slater: given how much time I've spent trying to get FFMPEG to work in my environment... what's another few hours? ;) Q then... is --enable-debug=LEVEL the way to do that? And does it support non GCC compilers? [01:58] <dahat> Hrm... configure looks to add the -gX argument (where X is the debug level)... MSVC would use /Zi I believe [02:13] <ac_slater_> dahat: I know nothing about windows, but `-g` is used for clang and gcc ... I do `-ggdb3` and it works well [02:14] <ac_slater_> dahat: BUT, it looks like all you need to do is set options. [02:14] <ac_slater_> (for the demux/decoder) [02:15] <ac_slater_> dahat: specifically http://www.ffmpeg.org/doxygen/2.1/rtmpproto_8c.html#a01616ba57274bb27f2e3f8… [02:16] <ac_slater_> dahat: you can pass an AVDictionary to avformat_open_input with those values filled in [02:18] <ac_slater_> dahat: or maybe use AVOption ... I get those two mixed up and am facing the same issue [02:19] <ac_slater_> dahat: and by AVOption, I mean av_opt_set* [02:28] <relaxed> dahat ac_slater_ are you guys on the libav-user mailing list? [02:28] <ac_slater_> relaxed: I am, yes. [02:28] <ac_slater_> relaxed: I rarely get replies though ;), its ok [02:48] <dahat> relaxed: I am not [02:49] <dahat> ac_slater: AVOption or AVDictionary are a couple of ways I've suspected it could be done, but haven't found a bit of example code that demonstrates their use yet [02:52] <relaxed> ok, I was just thinking you might get some answers there. [02:52] <ac_slater_> dahat: https://libav.org/doxygen/master/group__lavf__decoding.html ... under "Opening A Media File" [02:53] <ac_slater_> (I've never gotten AVOptions to work via av_opt_set* ... one day [02:53] <ac_slater_> also, I know that's libav not ffmpeg... it still works. It just had a clear example [02:53] <ac_slater_> You can find some examples in the `avcodec.c` example in 2.2 [02:54] <ac_slater_> (and the other ones, but they are mostly for audio... same idea) [02:56] <llogan> ac_slater_: https://ffmpeg.org/doxygen/trunk/group__lavf__decoding.html [02:57] <ac_slater_> llogan: thanks! [02:57] <ac_slater_> I should have known [02:57] <ac_slater_> dahat: use that ^ [03:19] <Baked_Cake> windox malicous toools found my kms service onoes [05:04] <dahat> ac_slater: That looks like it might just do it! will try now, thanks :) [07:31] <dahat> ac_slater: thank you thank you thank you! your link gave me the insight needed to get my prototype functioning [07:32] <dahat> I've been silent for a while as I've been singing and dancing around the house in happiness that my prototyping of something is finally doing something I want [07:33] <dahat> bah! seems he has already left at a reasonable hour of the day... I should do the same... but there is too much to do! [09:00] <omnivibe> Here is my command, which is giving me errors: http://pastebin.com/NpmvCLpD [09:00] <omnivibe> when I leave out the "-profile:v main -level 4.0", it works... [09:01] <relaxed> pastebin the error [09:03] <omnivibe> http://pastebin.com/UnjHdRp4 [09:04] <relaxed> try -level 40 [09:05] <omnivibe> same error. Turns out anyway it's not the -level part that's a problem, but the -profile [09:07] <relaxed> it would help if you pastebined all the output [09:10] <relaxed> -profile:v main works here, maybe your build is old [09:12] <omnivibe> http://pastebin.com/jvur10kZ [09:13] <omnivibe> nah, it's a recent build I think... [09:13] <omnivibe> ffmpeg version 2.3.git Copyright (c) 2000-2014 the FFmpeg developers [09:14] <relaxed> that's the longest metadata output I've ever seen [09:15] <relaxed> are those edit points? [09:15] <relaxed> add -pix_fmt yuv420p [09:16] <relaxed> I think you were trying to use an unsupprted color space with the main profile [09:22] <toeshred> is there a way to change which audio track is the default (shown as "(default)" when looking at the streams)? [09:22] <omnivibe> oh, that did it! [09:22] <omnivibe> the pix_vmt [09:23] <toeshred> i tried swapping the order of the first and second audio track, but that doesn't change which track played by default. [09:23] <relaxed> toeshred: try again with -map_metadata -1 [09:23] <omnivibe> can you explain a bit about what that does, and do I need to pay attention to something for all the different source files to make sure there's no color shifts? [09:24] <toeshred> relaxed: ok, i'll give that a try. i did use -map originally, but not -map_metadata -1 [09:25] <relaxed> toeshred: they're different [09:26] <relaxed> omnivibe: certain profiles only support specific color spaces. http://en.wikipedia.org/wiki/H.264/MPEG-4_AVC#Profiles [09:28] <relaxed> and main only supports 4:2:0, which is why you needed -pix_fmt yuv420p. [09:30] <relaxed> while the input color space was yuv444p10le [09:32] <omnivibe> damn, it definitely lost some detail like that :( shadows are more crushed, picture is overall darker [09:32] <omnivibe> but I guess going from 4:4:4 to 4:2:0 is going to leave something to be desired... [09:33] <relaxed> do you need main 4:2:0? [09:34] <K4T> frame=2434691 fps= 34 q=17.0 size=26204796kB time=13:31:33.79 bitrate=4408.6kbit -> this is my log from ffmpeg console which is up from almost 24h. I noticed that now fps parameter = 34, but when I started ffmpeg it was 50 and it should be always 50 fps. Can someone tell me what happened? Right now my CPU usage is 15% and RAM 34%. [09:35] <omnivibe> yeah, targeting ipads [09:35] <omnivibe> (so main level 4.0) [09:40] <relaxed> omnivibe: see if -sws_flags accurate_rnd makes any difference [09:49] <omnivibe> Undefined constant or missing '(' in 'accurate_rn' [09:49] <omnivibe> oh hehe [09:49] <omnivibe> nm [10:02] <omnivibe> same problem with darkness :( [10:05] <relaxed> omnivibe: the algorithms are listed in "man ffmpeg-scaler", maybe chroma* ? [10:22] <omnivibe> tried -sws_flags full_chroma_int+full_chroma_inp, still dark :( [10:37] <relaxed> omnivibe: does it look dark when you play the input with ffplay? [10:45] <omnivibe> can't really do that since my virtual machine is console only [10:46] <relaxed> can you put a sample of the video up somewhere? [11:17] <Keshl> How does ffmpeg get keypresses while it's rendering? Does it use any special libraries or anything? [11:27] <saste> Keshl, no special library [11:28] <Keshl> Awesome. o_o. How's it do it, then? o.O I don't know of any special built-in functions for C++ that do it.. (ffmpeg's C++, right?) [11:37] <michaelni> omnivibe, if you think the darkness is a bug, then please open a bug report on trac with a reproduceable testcase [12:08] <K4T> about ffmpeg console output format - is it possible to get days number in time column too? Now format is: HH:mm:ss [12:08] <K4T> I would like to see days too [12:31] <paolo_> Hi i wanna stream my desktop with a webcam overlay. I am struggeling with the filters. when i am recording to a file the webcam is missing. and when i am using ffplay the video window won't open. i would appreciate some help. here is what I do : [12:31] <paolo_> ffmpeg -video_size 1680x1050 -framerate 25 -f x11grab -i :0.0+0,0 -vf "[in] scale=1152:720 [tmp]; movie=/dev/video0:f=video4linux2, scale=320:-1, fps [webcam]; [tmp][webcam] overlay=main_w-overlay_w-10:main_h-overlay_h-10 [out]" test.mp4 [12:37] <relaxed> paolo_: -filter_complex is recommended now for multiple inputs [12:39] <paolo_> relaxed: thank you for your answer. but it can't be used with ffplay right ? [12:42] <relaxed> ffplay can't open test.mp4 until ffmpeg is finished encoding. Use test.mkv as your output instead. [12:43] <paolo_> relaxed: i am using ffplay to create the filter. so i can se the results without encoding first. [12:45] <paolo_> relaxed: does the representation of the filtergraph remain the same between -vf and -filter_complex [12:45] <paolo_> ? [12:46] <relaxed> https://www.ffmpeg.org/ffmpeg-filters.html#overlay-1 [12:46] <omnivibe> Thanks for the help everyone [12:46] <omnivibe> peace [12:48] <relaxed> paolo_: ffplay doesn't not support filter_complex, which is annoying [12:48] <relaxed> does not* [12:53] <paolo_> relaxed: "Stream specifier 'in' in filtergraph description [...] matches no streams" [12:54] <paolo_> relaxed: i used ffmpeg and replaced vf with filter_complex. [13:00] <paolo_> i think i will switch to a stable version first an see if it still happens [13:00] <paolo_> relaxed: thank you very much for your help. i appreciate it. [13:01] <halt2> Hi All, I propably will ask something which was asked million times here before, but can't really find the answer so forgive me to asking it again, I have huge MOV files and I what to convert them to .MP$ to be able to play then with flow-player ( web ) if it's possible without much quality lost, I know nothing about codecs and encoding and etc so plz keep it simple [13:06] <__jack__> halt2: try something like this: ffmpeg -i input.mov -c copy output.mp4 [13:25] <someone-noone> Can ffmpeg encode vp6 somehow? [13:26] <ubitux> relaxed: you can use -f sdl - with ffmpeg if you want to use -filter_complex and visualize [13:26] <ubitux> (or -f gl, -f xv) [13:36] <paolo_> I am using -filter_complex now but the overlay is still missing. any hints ? [13:36] <paolo_> here is what I do now [13:37] <paolo_> ffmpeg -video_size 1680x1050 -framerate 25 -f x11grab -i :0.0+0,0 -filter_complex "[0:v] scale=1152:720 [tmp]; movie=/dev/video0:f=video4linux2, scale=320:-1, fps [webcam]; [tmp][webcam] overlay=main_w-overlay_w-10:main_h-overlay_h-10 [out]" -map '[out]' test.mp4 [13:37] <paolo_> I am sure the webcam is recording because the LED is on [13:51] <ubitux> paolo_: are you sure of the overlay position? [13:52] <ubitux> also, if you use -filter_complex, you can drop the movie= .. [13:52] <paolo_> ubitux: i am not sure about the overlay position. I have 2 screens. [13:53] <ubitux> try just "overlay" first [13:53] <paolo_> ubitux: i did [13:53] <ubitux> and? [13:53] <ubitux> nothing is overlayed? [13:53] <paolo_> it didn't word [13:53] <paolo_> *morked [13:53] <paolo_> lol [13:53] <paolo_> *worked [13:53] <paolo_> :-) [13:55] <paolo_> ubitux: i'll try another corner. 0:0 is the upper left ? [13:56] <paolo_> it doesn't work :-/ [13:56] <ubitux> going to have a look [13:56] <ubitux> just a moment [13:57] <ubitux> ffmpeg -video_size 1680x1050 -framerate 25 -f x11grab -i :0.0+0,0 -f v4l2 -i /dev/video0 -filter_complex "[0:v] scale=1152:720 [tmp]; [1:v] scale=320:-1, fps [webcam]; [tmp][webcam] overlay [out]" -map '[out]' [13:57] <ubitux> try this [13:58] <paolo_> ubitux: your settings are working [13:59] <ubitux> now why it didn't work with the amovie, i don't really know [13:59] <ubitux> maybe the pts were kind of broken or something [13:59] <ubitux> saste might know better [13:59] <brontosaurusrex> what would be a 2014 cli for converting to jpg image sequence? [14:00] <ubitux> ffmpeg -i ... out%03.jpg [14:00] <ubitux> ffmpeg -i ... out%03d.jpg [14:00] <brontosaurusrex> i have "-f image2 -an -qmax 1" as well [14:02] <paolo_> ubitux: thank you for the help ! [14:03] <brontosaurusrex> ubitux: thanks [14:04] <ubitux> paolo_: you can probably drop the fps filter btw [14:04] <paolo_> ubitux: OK :-) [15:00] <brontosaurusrex> ubitux: uhmm, how to control compression? [15:02] <ubitux> qscale, qmax, or something [15:02] <__jack__> Use -qscale:v (or the alias -q:v) as an output option. The full range for JPEG is 1-31 with 31 being the worst quality. I recommend trying values of 2-5. [15:03] <__jack__> (dixit some links on the web) [15:08] <brontosaurusrex> -q:v 1 would be max quality? [15:10] <__jack__> yep, I guess, but remember, you won't get a better quality than the soruce [15:10] <__jack__> source* [15:12] <brontosaurusrex> __jack__: sure, i have like 100 droplets and iam trying to make them all work with new ffmpeg, cant even remember why i ever need jpg seq at all now [16:10] <pmart> Hi. Should I use VBR mode of libfdk_aac? It is considered experimental but I much prefer VBR in general (e.g. CBR encodes silence with same bitrate). [16:13] <pmart> I'm talking about document with actor voices but no music [16:13] <halt2> __jack__: Thanks! [17:13] <sine0> does ffmpeg do batch resizing of images just as jpg [17:14] <__jack__> bah, use something else [17:16] <sine0> i am [17:16] <__jack__> like imagemagick, which may be far better than ffmpeg for such task (example: convert input.jpg -resize 64x64 output.jpg) [17:18] <ubitux> sine0: sure you can [17:19] <ubitux> __jack__: why? [17:20] <__jack__> dunno, in my head, ffmpeg is designed for "record, convert & stream audio & video", which is a bit out of "image resizing", and thus may not be the best tool for that [17:21] <ubitux> ffmpeg input.jpg -vf scale=64x64:force_original_aspect_ratio=decrease output.jpg should do exactly the same [18:35] <valder> hey all. I need some assistance. I'm having a problem adding audio to my video that I'm creating. I'm splicing videos from multiple sources into a single output. I can't seem to get the associated audio to be a part of the final output. All my videos come out silent. here's my code: http://pastebin.com/rAZkU3XZ the code I have to write audio to the output file starts at line 911. [18:35] <valder> any help would be appreciated. [18:38] <tab1293> can anyone tell me how I can output an elementary audio stream with ffmpeg? [19:09] <relaxed> tab1293: what type of stream? [19:23] <vlatkozelka> hi , im tryin to save a udp stream on a .ts file segments wiht -f segment -segment_time 60 , and using -copyts to keep the timestamps untouched ... but the -copyts is making me get 1 second long files (with right timestamps) and if i dont use -copyts i lose the timestamps [19:36] <stevehehehe> how's best to mp3 -> mp4 (mp3 + jpg) ? [19:37] <stevehehehe> specifically for youtube. this has been way harder than i thought it would be [20:32] <tab1293> relaxed: aac [21:01] <haspor> im trying to decode adpcm_swf audio stream of FLV file but all i get is really messed up sound, for mp3 and aac it works perfectly, any special tricks i need for my code to make it work? [21:02] <haspor> with ffplay.exe it works ok so the FLV file itself is ok [21:02] <relaxed> tab1293: ffmpeg -i input -map 0:a -c copy -f aac output.aac [21:03] <tab1293> relaxed: thank you [21:06] <relaxed> tab1293: that should be -f adts instead of -f aac [21:14] <stevehehehe> to create basically a mp4 of an mp3+image really shouldn't take this long, surely? [21:15] <relaxed> stevehehehe: use -shortest [21:16] <stevehehehe> could you give me an example commandline? i'm not too familiar with ffmpeg [21:16] <relaxed> what are you doing? [21:16] <stevehehehe> basically trying to upload an mp3 to youtube [21:16] <stevehehehe> with a nice image as a background/video [21:20] <relaxed> ffmpeg -i input.mp3 -loop 1 -i input.jpg -shortest -c:a copy -c:v libx264 -tune stillimage -movflags faststart output.mp4 [21:22] <relaxed> add -pix_fmt yuv420p [21:22] <stevehehehe> thanks thats going pretty fast [21:33] <brontosaurusrex> relaxed: is movflags faststart not the default behaviour? [21:42] <llogan> brontosaurusrex: it is not the default [21:43] <relaxed> movflags slowstart [21:45] <brontosaurusrex> ok, and thats the same as using that qt-faststart tool? [21:46] <relaxed> yes [21:48] <brontosaurusrex> just re-bash-ing some of my old scripts, but I found so many stupid things it will probably take me a week to fix it all... [21:52] <lipizzan> I'm using ffmpeg v0.7.15, and encoding dv to (mp4) h.264 using libx264 -vpre lossless_medium -crf 20. The bitrate can vary WIDELY depending on the source image. Is this to be expected? [21:53] <brontosaurusrex> lipizzan: yes, especially if you use -tune film < just my experience [21:53] <lipizzan> ~ 700kbs for a largely balck & white image with < 20% region change, vs 4000+ kbs for a frame of tree leaves. [21:57] <llogan> lipizzan: i have a feeling that 0.7 branch lacks the old preset "emulation" files that you're using (lossless_medium). but i may be wrong. [21:58] <lipizzan> I'm piping dvgrab to ffmpeg, but I paste the cmd shortly. [21:59] <llogan> doesn't have to be the whole file. just a short output will suffice [22:01] <llogan> but yes, input complexity will affect bitrate, and DV can be "complex" due to noise (such as from a low light situation using a crappy camcorder), etc. [22:07] <lipizzan> http://pastie.oeg/9548738 [22:07] <lipizzan> http://pastie.org/9548738 [22:18] <lipizzan> I guess my concern is making sure I'm encoding 640x480 source DV into h.264 efficiently and with decent quality. I don't wanna waste bits if i don't have to. [22:21] <llogan> lipizzan: i'd start with a more recent ffmpeg build. [22:22] <llogan> and use a real preset instead of "lossless_medium" [22:22] <llogan> which no longer exists [22:26] <llogan> ffmpeg -i - -i logo.png -filter_complex "[0:v]yadif=1:1,hqdn3d=3,format=yuv420p[bg];[bg][1:v]overlay=W-w-3:H-h-62,fade=in:0:30[out]" -map "[out]" -map 0:a:0 -c:v libx264 -preset slow -crf 24 -movflags +faststart -c:a libfdk_aac -vbr 4 -metadata title="title" output.mp4 [22:26] <llogan> or something like that [22:26] <llogan> you can dump movie filter [22:27] <lipizzan> llogan: I know, I know, but I'm using the linux distro, AVLinux v6.0.3, and this is what it has. Also, I'm doing alot with the script, and don't have time to rework it at the moment. [22:27] <llogan> format should be in the second filterchain instead of the first in my example [22:27] <llogan> you can just download a recent build (but you would have to use the native AAC encoder) [22:28] <llogan> not that libfaac is that great anyway [22:28] <lipizzan> llogan: can I use neroAccEncode? [22:28] <llogan> sure. you can pipe to it, then remux the audio from nero [22:30] <lipizzan> llogan: remux? Is that a separate step? I try to do this all in one pass. [22:30] <llogan> yes, a separate step [22:31] <lipizzan> llogan: I'll have to work on that since there is another step where I add a "sponsors" leader to the video. I want good audio quality, is neroAccEncode worth it? [22:32] <llogan> i don't know. you'll just have to compare it yourself and listen [22:33] <lipizzan> llogan: others, in the past, have indicated that it was probably preferrable to libfaac. [22:33] <llogan> since quality is subjective, and your level of "do I care" is too, then it will have to be up to you [22:34] <lipizzan> llogan: thx. probably worth my time to chek it out. [22:34] <llogan> i'm a lazy bastard, so i just use libfdk_aac (but i compile my own ffmpeg) [22:35] <lipizzan> llogan: I'm not familiar with libfdk_aac. I need to check it out. thx! [22:35] <llogan> https://trac.ffmpeg.org/wiki/Encode/AAC [22:36] <llogan> https://trac.ffmpeg.org/wiki/Encode/H.264 [22:36] <llogan> 264 - core 118 is old. [00:00] --- Sat Sep 13 2014

1 0

ffmpeg-devel.log.20140911
by burek 12 Sep '14

12 Sep '14

[01:05] <Timothy_Gu> See if they respond to my last question: https://lists.libav.org/pipermail/libav-devel/2014-September/063150.html [01:24] <Timothy_Gu> kierank: the link on http://obe.tv/ to your Opus blog post is dead [01:25] <Timothy_Gu> kierank: also the second "read more" on http://obe.tv/products [01:33] <cone-182> ffmpeg.git 03Diego Biurrun 07master:775a0b04f0cf: doc: Fix syntax and logical errors in avconv stream combination example [01:33] <cone-182> ffmpeg.git 03Michael Niedermayer 07master:5e3da256184e: Merge commit '775a0b04f0cf8102fe322b2ee03fe1a0633dea04' [01:40] <wm4> why is this linked on ffmpeg.org http://www.mplayerhq.hu/DOCS/tech/patches.txt [01:41] <wm4> that's negatively helpful [01:41] <wm4> starting with the fact that it assumes svn [01:41] <J_Darnley> because the docs are old [01:42] <wm4> anyway, this guy has a point: https://ffmpeg.org/pipermail/libav-user/2014-September/007400.html [02:13] <Timothy_Gu> wm4: link to the MPlayer doc? [02:13] <wm4> Timothy_Gu: see above, if you mean where it's linked: http://ffmpeg.org/contact.html#MailingLists [02:14] <Timothy_Gu> wm4: that's what i meant [02:14] <Timothy_Gu> interresting [05:12] <cone-1> ffmpeg.git 03Michael Niedermayer 07master:16de4d927e8b: ffmpeg: Do av_buffersink_set_frame_size() when reconfiguring the filtergraph not just when changing audio resample parameters [11:43] <rcombs> wait a second, why am I allowed to close or assign other people's tickets [11:47] <ubitux> because you opened it maybe [11:48] <rcombs> I get the same options on all tickets, though [11:48] <ubitux> are you going to use that power for evil? [11:48] <rcombs> I'd expect not [11:49] <rcombs> "oh no, the evil rcombs is pillaging the town and marking tickets as solved!" [11:49] <rcombs> "and assigning ASS-related bugs to wm4!" [11:49] <ubitux> ahah [11:49] <ubitux> LGTM [11:58] <ubitux> we should add harfbuzz support to drawtext :-° [11:59] <rcombs> mpv actually implements its OSD using libass [12:08] <rcombs> ^ does that happen often? [12:22] <ubitux> rcombs: there is a word blacklist somewhere [12:23] <ubitux> it used to happen often [12:24] <ubitux> eeeh no escaping possible with file with concat demuxer :( [14:13] <cone-520> ffmpeg.git 03Michael Niedermayer 07master:ea7ebadeb78c: avformat/rmdec: very basic MLTI support [15:25] <cone-520> ffmpeg.git 03Michael Niedermayer 07master:5303a644f588: avformat/img2: Make AVOptions available to img* demuxers defined in other files [15:25] <cone-520> ffmpeg.git 03Michael Niedermayer 07master:646d663f7137: avformat/img2_brender_pix: Add AVClass and image2 AVOptions [15:25] <cone-520> ffmpeg.git 03Michael Niedermayer 07master:8b4714614f8b: avformat/img2_alias & brender_pix: fix function names [16:54] <cone-520> ffmpeg.git 03Michael Niedermayer 07master:de4a4ca1a08d: avformat/img2_alias_pix: Add AVClass and image2 AVOptions [16:54] <cone-520> ffmpeg.git 03Michael Niedermayer 07master:a6153bde3dee: avcodec/pngdec: print the actual number of bytes left in the error case [16:54] <cone-520> ffmpeg.git 03Michael Niedermayer 07master:4a203b773702: avcodec/png_parser: Return the buffered data at the end [17:03] <ubitux> > Is it possible for me to use part of this patch for the FFmpeg website? [17:03] <ubitux> *sigh* - once more this feels like FFmpeg takes, takes, takes, and gives [17:03] <ubitux> nothing in return (let's not even delve into the whole negativity thing). [17:03] <ubitux> lol [17:06] <rcombs> wat [17:07] <ubitux> rcombs: yeah well they are upset because we steal everything from libav [17:08] <rcombs> :| [17:09] <ubitux> and here one guy was reviewing stuff from libav, and asked for the permission to pick a diff (which happens to be unlicensed) [17:09] <ubitux> but oh well :) [17:26] <ubitux> Timothy__Gu: stop stealing from libav!! [17:30] <Timothy__Gu> ubitux: I can only sigh [17:53] <J_Darnley> Woo! A complex visualization rendering at more than realtime! [17:53] <ubitux> ? [17:55] <J_Darnley> The vis filter I've been working on. [17:55] <ubitux> ok :) [17:56] <ubitux> can you spoil a bit? [17:58] <J_Darnley> Spoil a bit? [17:58] <ubitux> tell us more [17:58] <Daemon404> printf("3+2i"); [17:58] <Daemon404> compelx visualization! [17:58] <J_Darnley> :) [17:59] <J_Darnley> It is the port of Winamp's/Nullsoft's Advanced Visualization Studio I've been working on for a couple of months now. [18:01] <Daemon404> ah... what was their stuff called... it was somethign silly [18:01] <Daemon404> like MILK [18:01] <kepstin-laptop> wow, been a while since i've used that :) [18:01] <kepstin-laptop> no, MILK was the successor to AVS [18:01] <J_Darnley> There was also Milkdrop [18:01] <Daemon404> ah ok. [18:01] <Daemon404> its been a while [18:01] <Daemon404> ;) [18:01] <kepstin-laptop> J_Darnley: any chance you hung out in #avschat back in the day? [18:01] <J_Darnley> That's all on the GPU I think. [18:01] <J_Darnley> No [18:01] <kepstin-laptop> ah, pity. [18:01] <ubitux> or projectM maybe [18:04] <J_Darnley> I've been using Lua to replace the original expression library [18:04] <J_Darnley> and now I've stopped compiling the same expression over and over, and moved to LuaJIT [18:05] <J_Darnley> It now renders one preset at 640x480@60Hz at 39 fps. [18:06] <J_Darnley> Uh that should be 640x480@30Hz [18:08] <J_Darnley> I should put a video up somewhere. [18:37] <cone-520> ffmpeg.git 03Rong Yan 07master:8e8db16a2cf5: avutil/ppc/float_dsp_altivec: ppc: Fix bug in ff_vector_fmul_add_altivec() affecting little endian environments on POWER8 [18:50] <cone-520> ffmpeg.git 03Jörg Krause 07master:453944cc77b9: configure: fix check_cmd in check_host_cpp [20:03] <llogan> rcombs: you don't have to worry about closing those spam tickets. one of us (michael, carl, alexander, or me) will eventually banish it [20:04] <llogan> unfortunatly they seem to appear when I'm afk the majority of the time [20:04] <rcombs> llogan: yeah, figured; saw it and went "ooh, something to kill" [20:04] <llogan> it is satisfying to nuke it [20:06] <llogan> now if i weren't so bad at regex... [20:06] <rcombs> http://regexcrossword.com <-- play this for a while and you won't be [20:10] Action: llogan button mashes [20:12] <ubitux> why do i have the feeling that someone is probably doing exactly the same thing at the same moment when i'm writing some code for ffmpeg? [20:13] <ubitux> and like, i absolutely need to hurry up before he posts that patch(set) [20:18] <llogan> ubitux: http://www.youtube.com/watch?v=7YvAYIJSSZY [20:21] <ubitux> :( [20:46] <llogan> how was VDD? [20:46] <llogan> oh, wait wrong year [20:46] <llogan> duh, 20-21 sept [20:47] <llogan> i have a problem of not thinking before typing [20:48] <wm4> libav 11 still not released? [20:48] <wm4> it's always the same shit... [20:50] <nevcairiel> told you so [21:13] <llogan> damn it. i didn't see that the attachment was 14 MB. [21:15] <llogan> i wish mailman would bark about file size instead of unsubscribed when both are true. [21:19] <ubitux> ass expert (wm4, rcombs?); any opinion on the ass_set_font_scale() patch on libavfilter/vf_subtitles.c? [21:19] <wm4> ubitux: the first patch looks very wrong [21:19] <ubitux> (https://ffmpeg.org/pipermail/ffmpeg-devel/2014-September/162755.html) [21:19] <wm4> I'll reply [21:19] <ubitux> thank you [21:21] <ubitux> kind related, i'm probably going to add the libass hinting option btw, just like shaper [21:27] <ubitux> mmh we don't set the pixel_aspect [22:33] <J_Darnley> llogan: I wonder whether you should announce that we are dropping those old patch guidelines. [23:01] <llogan> J_Darnley: i just thought it was confusing to users and nobody reads it anyway [23:02] <J_Darnley> That's certainly true [23:02] <J_Darnley> I'm just reading the git howto for the first time. [23:03] <J_Darnley> (If it wasn't clear I was mostly joking about making an announcement) [23:03] <wm4> another suggestion: make it clear on the website that ffmpeg-devel is for patches [23:03] <wm4> a user was confused where he should send his patch to (he ended up on libav-user) [23:04] <J_Darnley> I would phrase that as "ffmpeg-devel is the place to send patches" [23:04] <J_Darnley> Otherwise it sounds like it should be nothing but patches. [23:05] <llogan> wm4: yeah, that's what I was going to do after dropping this MPlayer patch (if I can get support to do so). [23:05] <iive> patches and development of ffmpeg itself. [23:05] <llogan> s/patch/link [23:05] <wm4> nice [23:06] <llogan> in general I want to reduce verbosity and try to make it more clear to users [23:07] <J_Darnley> I support your change and I'll send an email shortly [23:07] <llogan> thanks! [23:08] <J_Darnley> Oh I see Carl has sent a reply [23:48] <J_Darnley> We need to stop what ever transforms our docs into html turning two hyphens into an em dash [23:52] <cone-520> ffmpeg.git 03Michael Niedermayer 07release/2.1:6f6b1d0ddb26: avformat/oggparseopus: factor opus_duration() out [23:53] <cone-520> ffmpeg.git 03Michael Niedermayer 07release/2.1:64908f70e436: avformat/oggparseopus: calculate pts/dts for initial packets after seeking [23:57] <kepstin-laptop> re closed ticket - due to the change in beheviour with the ffmpeg tool exit code in "errors", 2.2 isn't actually a compatible series with 2.1 :/ [00:00] --- Fri Sep 12 2014

1 0

ffmpeg.log.20140911
by burek 12 Sep '14

12 Sep '14

[00:18] <c_14> FFmpeg does not currently support motion interpolation. [00:32] <DoctorTrombone> Greetings. How would I go about merging several audio input streams (some stereo, some mono) from my input file into one stereo output file? [00:33] <Mavrik> DoctorTrombone, you'll probably have to use amix filter to mix them together [00:33] <DoctorTrombone> Ah, that sounds tricky. Could you go into a little detail on that? [00:35] <Mavrik> this might be a good starting point: https://trac.ffmpeg.org/wiki/AudioChannelManipulation [00:35] <DoctorTrombone> Thanks! [00:35] <Mavrik> DoctorTrombone, but basically, see how audiofilters work [00:36] <Mavrik> DoctorTrombone, and then just tell amix filter (documentation on ffmpeg-filters page) to merge them together into one [00:36] <DoctorTrombone> I'm confused, I thought you said this page was the one I wanted. [00:36] <Mavrik> Well you want it if you want to do anything useful. [00:36] <DoctorTrombone> I don't understand. [00:38] <c_14> There's more than one place with documentation. [00:39] <DoctorTrombone> c_14: I don't follow. [00:39] <c_14> There's the wiki and there are the man pages. [00:39] <DoctorTrombone> c_14: and I read both [00:39] <DoctorTrombone> There's a reason I came to the IRC channel. [00:40] <c_14> You have one input file with multiple audio streams that you want to merge into one Stereo stream? [00:41] <DoctorTrombone> Yeah, my input file has around six audio streams. [00:42] <c_14> And most streams are stereo but some are mono? [00:42] <DoctorTrombone> c_14: That is correct, yes. [00:44] <llogan> you'll nned to provide the complete ffmpeg console output of "ffmpeg -i input", and a description of where each channel for each input stream should end up in the appropriate channel in the output. [00:45] <DoctorTrombone> What? [00:45] <llogan> we can't give an example without knowing what you want to do [00:45] <DoctorTrombone> llogan: I just want to mix them all down to a stereo output file, with stereo tracks in stereo, and mono tracks on both channels. [00:45] <DoctorTrombone> llogan: That might be true, but I explained what I wanted to do, didn't I? [00:45] <llogan> now you did [00:46] <c_14> And they all have the same sample rate and the same format? [00:46] <c_14> And, pastebin the output. [00:46] <DoctorTrombone> llogan: I did before, you may have missed it. [00:46] <DoctorTrombone> c_14: Yes, same rate and format. [00:47] <DoctorTrombone> http://pastebin.com/7uDPb736 [00:48] <DoctorTrombone> The video track is basically garbage, and I don't want one of the audio tracks but I can't tell which one at the moment. [00:50] <DoctorTrombone> Honestly I'd probably use Sox if it understood .bik, but its format support is pretty limited. [00:51] <c_14> -i file -filter_complex '[0:1][0:2][0:3][0:4][0:5][0:6][0:7]amerge=inputs=7,pan=stereo:c0<c0+c2+c4+c5+c6+c8+c9:c1<c1+c3+c4c+5+c7+c8+c10[aout]' -map '[aout]' [00:51] <c_14> That should be it, or at least close enough. [00:51] <DoctorTrombone> c_14: That looks brilliant. [00:51] <DoctorTrombone> let me try it out [00:53] <DoctorTrombone> It didn't work http://pastebin.com/idKPegAj [00:55] <c_14> you messed up when copying what I typed [00:55] <c_14> +c4+c5+c7 [00:55] <c_14> not +c4c+5+c7 [00:55] <DoctorTrombone> c_14: I used copy-paste, that's rather odd. [00:56] <c_14> Hmm, if my syntax is so deprecated, why is it used on both the wiki and the manpages... [00:57] <llogan> ignore that [00:57] <DoctorTrombone> c_14: Thanks so much :) FFmpeg was pretty easy to understand, but edge cases are killers for any powerful tool. [00:57] <c_14> np [00:57] <c_14> llogan: I know it shouldn't break anything, I'm just wondering. [00:58] <llogan> you may be able to use -ac 2 instead of pan. [00:58] <DoctorTrombone> llogan: you mean after the whole merge inputs bit? [00:58] <llogan> not sure if that will magically/lazily work [00:58] <llogan> yes [00:59] <DoctorTrombone> It worked fine, but having something like that for future reference is good. [01:00] <valder> Here's a really naive question, but if I read video in from any movie file (ie mp4, mov, etc) can I assume the data is in RGB color space? Are all finished movie files in RGB? Can they be in YUV or some other format? [01:00] <DoctorTrombone> Mm, that works as well. Thanks a bunch folks, and good day to you all. [01:00] <Mavrik> valder, no, almost no video formats are in RGB [01:01] <Mavrik> RGB is a terrible color space for video storage, vast majority will use YUV as their preferred color spae [01:01] <Mavrik> *space [01:01] <valder> ok.. I thought YUV was camera only. [01:01] <valder> thanks [01:03] <valder> So I have a routine that seems to be writing video files out.. however, I'm not sure if it is properly handling all types of files. Would anybody mind doing a review? [01:04] <valder> Here's the pastebin of the relevant code: http://pastebin.com/Q2eZLxKB [01:06] <Mavrik> valder, you should probably keep timebase as AVRational and make sure you convert it properly when combining frames [01:06] <Mavrik> or your video won't play well [01:07] <Mavrik> also remember, alot of video formats have YUV420 enforced, so make sure your input videos use that color space or encoding will fail [01:08] <valder> Mavrik: how do I go about verifying that? (YUV420) the assumption is that the user can supply any video file. If it isn't supported (?) I should be able to tell and reject the file right? (how do I do that?) [01:08] <Mavrik> input codec context has pix_fmt that tells you that [01:08] <Mavrik> you can also use swscale to convert it [01:09] <valder> right. so I commented out swscale b/c I didn't know if that was actually doing naything. [01:10] <valder> I guess I should leave it in. :) [01:10] <Mavrik> also you seem to be missing write_frame funcion :P [01:10] <valder> oh let me paste the entire code [01:10] <Mavrik> also stuff like that is horribly uncool: numBytes=avpicture_get_size(PIX_FMT_RGB24, pCodecCtx->width, pCodecCtx->height); [01:11] <Mavrik> you're persuming that your frames are always in RGB24 (which I doubt they will be :) ) [01:12] <valder> http://pastebin.com/LZurNs7C [01:13] <valder> Mavrik: please tell me the "cool" way to do it. :) I found an example that did it that way and followed it. [01:13] <Mavrik> ah, well if you're grabbing frames from an existing video [01:13] <Mavrik> you'll have their color space passed as pix_fmt in the codec context [01:13] <Mavrik> then use that when allocating data right? :) [01:14] <valder> so for each input video, I want to inspect the pix_fmt value to determine if I need to convert it or not. Right now I don't think I'm doing that correctly(or at all). [01:15] <valder> so I think my code is geared for a specific video file type. If I try to mix in another one, it will crash (is that what you expect?) [01:15] <valder> that might be why I"m having issues, I'm making the assumption the input videos are of the same format o [01:15] <Mavrik> yes [01:16] <Mavrik> you're also making the assumption that all videos use same PTS/DTS and timebase :) [01:17] <valder> So what do I need to do to stream line it all? [01:17] <valder> 1) on openign an input file I need to check pix_fmt [01:17] <valder> 1a) if it's not the right pix_fmt I need to convert it [01:18] <valder> 2) check the PTS/DTS and timebase (frames per second?) adjust reading in of the frames accordingly. say I want a target of 24fps. [01:18] <valder> I should then be able to mix in the files right? [02:08] <valder> Mavrik? [03:04] <active8> I lost the note on the command I used to slow down a video and keep the music pitch the same. It was not the filter PTS method - that produced crap sound. I think it was all ffmpeg - changing fps and rate, but may have been done otherwise, though I really don't recall stripping out the sound, slowing vid and aud separately and remuxing. Pretty sure it was all done wiith one ffmpeg command. Could someone please tell me how it's done? I m [03:04] <active8> anaged to get 1/4 speed of flying fretboard fingers with good sound and no pitch shift. [03:15] <vasilydernis> hey guys, i have a large file of say 700 seconds and i need to reduce it down to 600 seconds in length using 10 equal segments.. is this possible to do using ffmpeg in 1 command? [03:15] <vasilydernis> or do i have to split the larger file using -ss and -t and then concat the segments? [03:16] <vas123> i tried using select filter to no avail :( [03:19] <c_14> Create a filter complex with 10 trim filters and concat it. [03:20] <vas123> c_14: could you point me to where i can read up more on this? [03:22] <c_14> https://ffmpeg.org/ffmpeg-filters.html#trim [03:22] <c_14> https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20med… [03:22] <active8> Example of the vid is youtube rush la villa strangiato acoustic - it's the intro [03:24] <c_14> active8: https://trac.ffmpeg.org/wiki/How%20to%20speed%20up%20/%20slow%20down%20a%20… [03:26] <vas123> c_14: so something like this? http://video.stackexchange.com/questions/10396/how-to-concatenate-clips-fro… [03:26] <active8> as i said, i did not use PTS and PTS produces crappy scratchy audio. I'm sure it was done with a resampling at a diff rate and a change in FPS. If not, If I DID use PTS, then there was more to it than at the link you posted cause the audio is good. Pretty sure whatever I did, I learned it here. At least it was refined from help I got here. [03:27] <c_14> Remember what month/day it was? [03:27] <c_14> The channel is publicly logged. [03:27] <c_14> Heck, if you know the month and your nick, you can just grep. [03:29] <c_14> Oh, and if you find it ping me and I'll add it to that wiki page. [03:30] <active8> ok. I guess the file timestamp might be a clue [03:30] <active8> !welcome [03:30] <active8> !topic [03:31] <c_14> You can usually get the topic with /topic [03:33] <active8> Ok some channels have "!" commands. Where's the log? [03:34] <c_14> http://ffmpeg.gusari.org/irclogs/ [03:38] <Kirito> Whenever I use -threads 0 running ffmpeg, I've noticed that the progress indicator seems to glitch. What I mean is the bitrate and output size stops updating, but the time progress continues going up [03:39] <Kirito> But the video itself seems to encode just fine [03:39] <Kirito> Any idea why this may be..? [03:39] <c_14> What version you running? [03:39] <Kirito> ffmpeg version N-53307-g5a65fea [03:39] <Kirito> Debian Wheezy build [03:40] <Kirito> Actually it's not specific to -threads 0, it just seems to be a recent issue I've noticed [03:41] <Kirito> Or rather it's only doing it when encoding with mkv as the container [03:43] <Kirito> for i in *; do ffmpeg -threads 8 -i "$i" -map 0 -vcodec libx264 -tune animation -crf 18 -preset slow -c:a copy -c:s copy -c:d copy -c:t copy "./8bit/$i"; done --- is the command being run. Perhaps it has something to do with all the data I'm copying here, since ffmpeg complains that it can't recognize the codes of the font attachments the mkv container I'm encoding has? [03:45] <c_14> Shouldn't be. [03:45] <c_14> Try pressing d a few times to see if that refreshes the dispaly. [03:47] <Kirito> Nope, it just prints "error parsing debug value0 // debug=0" [03:53] <active8> c_14 the log is from the end of day 2-18-2014 to 2-19-2014 with my nick, active8. At 00:15 logan pasted the filter bit from the wiki. That's what I used last night and the sound is horrible. The one diff is that back then, I clipped out just the intro part - I didn't slowmo the whole vid. Could it be I used a copy codec or q:v 0 ? It's an mp4 and correct me if I'm wrong, that would involve both a re-encoding of h.264 and AAC audio to pu [03:53] <active8> t the remaining frames back together, resulting in info loss. [03:59] <c_14> What encoding settings? [04:04] <c_14> Kirito: no clue [04:05] <Kirito> Not a big issue anyways, I'll try and track down what specifically causes it later and submit a bug report to Debians tracker I guess, thanks [04:08] <active8> c_14: you asking me about encoding settings? [04:08] <active8> or Kirito? [04:09] <active8> I used no specific settings [04:09] <c_14> active8: you [04:09] <active8> mp4 to mp4 letting ffmpeg use the default - not sure if that is -c:v mpeg4 or libx264 [04:10] <c_14> And the audio sounds bad? [04:11] <active8> i am getting closer. I clipped the intro and ran (wait one while I clean it up) [04:11] <active8> ffmpeg -i quitar-intro.mp4 -filter_complex "[0:v]setpts=2*PTS[v];[0:a]atempo=0.5[a]" -map "[v]" -map "[a]" -q:v 1 -q:a 1 guitar-intro-half-speed.mp4 [04:12] <c_14> I think a higher q is better quality... [04:12] <c_14> Not sure though. [04:17] <active8> I think lower is higher. remember testing that but ffmpeg reports 0 is not right - that it should be 1-5 yet I recall a page where it was run up to -q:v 10 (maybe a different encoder supports higher values and the example was for a different container. AVI, IIRC) So after running the above, it sounds like what I did in Feb. Running it again to get 1/4 speed results in the same quality also - which was not so good. either I should spec t [04:17] <active8> he encoder to libx264, or i need to run filter_complex like it is in the trac wiki on the line above the last line (IOW -filter_complex "[0:v]setpts=2*PTS[v];[0:a]atempo=0.5[a],[0:a]atempo=0.5[a]") but I'm not sure if that syntax will work [04:18] <active8> q isn't quality, BTW it's quantization and maybe that's why lower is better - not gunna puzzle that through right now [04:18] <active8> 8) [04:18] <c_14> >Effective range for -q:a is around 0.1-10. This VBR is experimental and likely to get even worse results than the CBR. [04:19] <active8> ffmpeg said 1-5 when I tried 0 (which has worked in the past, IIRC. -qscale 0 worked, too, according to my notes/snips [04:20] <active8> trac wiki on doubling the audio slowdown/speedup: ffmpeg -i input.mkv -filter:a "atempo=2.0,atempo=2.0" -vn output.mkv [04:20] <active8> What I'm thinking for complex_filter: -filter_complex "[0:v]setpts=2*PTS[v];[0:a]atempo=0.5[a],[0:a]atempo=0.5[a]" [04:21] <active8> think if it can be done, either it's as-is or s/,/;/ [04:21] <active8> or [04:21] <active8> -filter_complex "[0:v]setpts=2*PTS[v];[0:a]atempo=0.5,atempo=0.5[a]" [04:28] <active8> c_14: awesome! I get better sound with: ffmpeg -i quitar-intro.mp4 -filter_complex "[0:v]setpts=4*PTS[v];[0:a]atempo=0.5,atempo=0.5[a]" -map "[v]" -map "[a]" -q:v 1 -q:a 1 guitar-intro-quarter-speed-complex2.mp4 [04:29] <active8> now maybe adding -c:v libx264 -c:a AAC might get even better. Is that the right codec and should it be right before the outfile name or before the filter? [04:31] <c_14> shouldn't matter. I usually put it before the output file though. [04:33] <active8> vlc says the audio is MPEG AAC Audio (mp4a) [04:33] <active8> ffmpeg says there are two: [04:35] <active8> DEA.L. aac AAC (Advanced Audio Coding) (encoders: aac libfdk_aac ) [04:36] <active8> and D.A.L. aac_latm AAC LATM (Advanced Audio Coding LATM syntax) [04:37] <active8> So the question is what's the diff between aac and libfdk_aac which I can find somewhere at trac or ffmpeg (the green pages) but not sure if there isn't a terse, simple explaination 8) [04:38] <c_14> Use libfdk_aac [04:38] <c_14> https://trac.ffmpeg.org/wiki/Encode/AAC [04:39] <active8> and thanks, c_14 for pointing me to the irc logs. Just knowing I used the atempo filter was enough to clue me in to the possibility that I set the quantization. 8) [04:39] <active8> triple thanks! [04:40] <ac_slater> hey all. I know this might not be a good place to ask (ill post to the mailing list too). I have a .yuv file created via ffmpeg. I want to read it via libavforat and avio. Any clues? [04:40] <ac_slater> I didn't see any examples of such a thing. I guess I could calculate the offsets and read the data myself via avio_file_map [05:20] <dahat> I'm developing an app to play basic videos with FFMPEG as a backend... and currently it accepts a local path into avformat_open_input(), now I am trying to expand it to target RTMP files, currently I can do the following at the command line, how can I apply the same args to avformat_open_input()? ffmpeg -rtmp_app <AppName> -rtmp_conn <AuthToken> -rtmp_playpath <PlayPath> -i rtmp://examplecom/ [06:12] <toeshred> Hello everyone, I am trying to change the default Audio stream. One is english with commentary, and the other is just english no commentary. I used the -map option to reorder so the 'no commentary' stream comes first, but the second audio stream always plays by default still, and has (default) next to it when I check the streams. I thought changing the stream order would also change the which audio stream [06:12] <toeshred> plays first, but I guess it doesn't. Is there a way to do this? [08:00] <valder> anybody around that can help me debug a segfault I'm getting in my code? I'm coding a native decoder/encoder for Android. It seems to work, as it generates a video however after it creates the video, the program segfaults as it returns to the java method that called it. [08:01] <valder> Here's the pastebin if it helps. http://pastebin.com/NeYE41hP [08:31] <dahat> I'm developing an app to play basic videos with FFMPEG as a backend... and currently it accepts a local path into avformat_open_input(), now I am trying to expand it to target RTMP files, currently I can do the following at the command line, how can I apply the same args to avformat_open_input()? ffmpeg -rtmp_app <AppName> -rtmp_conn <AuthToken> -rtmp_playpath <PlayPath> -i rtmp://streaming.example.com/ [08:34] <valder> dahat: would you mind reviewing what I wrote to see if there are any issues with it? my program keeps crashing when the render method returns back to the calling JNI method. [08:35] <dahat> valder: I can always look... though I cannot guarentee any degree of comptance with my review... coding against ffmpeg is something that is very new to me [08:36] <valder> dahat: ditto. :) [08:37] <dahat> I feel your pain... while there are a number of samples and bits of documentation out there... the general expectation seems to be 'figure it out yourself... then update the wiki or code with what you found'... which assumes quite a bit [08:43] <toeshred> Hello everyone, I am trying to change the default Audio stream. One is english with commentary, and the other is just english no commentary. I used the -map option to reorder so the 'no commentary' stream comes first, but the second audio stream always plays by default still, and has (default) next to it when I check the streams. How do I change what is "(default)"? [08:50] <valder> dahat: anything you can share would be great. I'm pretty sure I've made a few mistakes. [10:23] <termos> would it make sense to do run image blurring on YUV420P frames, or is it better to convert them to RGB? [10:37] <rivarun> c_14: on interpolation -- thanks, fwiw i found slowmoVideo which seems to do the trick [13:03] <anshul_mahe> is there any bad effect of joining B frame enabled video stream and non B frame enabled stream [13:17] <anshul_mahe> before concat I am using -vcodec libx264 -bf 0 in one video and in another video -vcodec libx264 [13:21] <termos> I am trying to create an AVFrame as decribed here http://stackoverflow.com/questions/12831761/how-to-resize-a-picture-using-f… but I end up with an AVFrame that has width,height = 0,0 and some strange linesize[] values. Any ideas? [13:57] <anshul_mahe> how to use --stitchable option in libx264 [14:01] <anshul_mahe> i am using x264 0.142.2 verion of x264 [14:03] <anshul_mahe> i got it -x264opts option [14:13] <albertid_> Hi, where can I find information about the variables/constants in ffmpeg filters? Like iw and ih. Is there a variable that holds the length of the video? [14:37] <ubitux> albertid_: these constants are filters specific [14:37] <ubitux> look at the filters documentation, they are documented [14:37] <ubitux> (or at least they should) [14:43] <albertid_> ubitux, I did. I was looking for a way to tell fade-out to start at "(length of video) - 1s", but I guess there is no way [14:44] <ubitux> the length of the video is never known [14:44] <ubitux> filters are stream based [14:44] <ubitux> you can extract an approximation to the best (or a wrong to the worse) from the container and use that [14:46] <ubitux> ffprobe -v error -of flat -show_entries format=duration input.mov [14:46] <ubitux> ffprobe -v error -of flat -show_entries stream=duration input.mov [14:58] <albertid_> ubitux, thx [15:25] <techfreak> sup guys [15:25] <techfreak> https://gist.github.com/Garont/2c57b5bdba72d286d148 [15:25] <techfreak> anyone can help me? [15:52] <techfreak> fucking hate ffmpeg [15:53] <sacarasc> Why? [15:54] <sacarasc> Looks like your file is broken... Claims to be mono, but isn't. [15:54] <ubitux> also 1.0 [15:54] <sacarasc> That too. [15:55] <techfreak> and how to fix that? [15:55] <relaxed> techfreak: http://johnvansickle.com/ffmpeg/ [15:55] <sacarasc> You could try with a more up to date ffmpeg... [16:27] <nehaljwani> How do use the pad filter? [16:27] <techfreak> relaxed: thanks, works fine [16:27] <nehaljwani> Suppose I have a video of res 1920x1440 [16:27] <nehaljwani> and I want to add padding at bottom to make it 1920x1080 [16:27] <nehaljwani> what should be the params to pad fileter? [16:28] <c_14> nehaljwani: You do know that 1440 > 1080, right? [16:28] <nehaljwani> oops [16:28] <sacarasc> http://ffmpeg.org/ffmpeg-all.html#Examples-80 shows examples of how to use the pad filter. [16:28] <nehaljwani> I should get some rest :-/ [16:29] <nehaljwani> what is the maximum 4:3 res that can fit in 1920x1080 [16:29] <relaxed> techfreak: you're welcome [16:30] <sacarasc> 1440x1080? [16:32] <nehaljwani> now I want to add padding so that the video is centralized.. what param should I give to pad? [16:33] <c_14> you can use x=iw/2 and y=ih/2 [16:33] <c_14> I think that was it anyways [16:33] <nehaljwani> what is iw/2 ? [16:33] <c_14> input width divided by 2 [16:33] <c_14> https://ffmpeg.org/ffmpeg-filters.html#pad [16:34] <c_14> aaaah, wait [16:34] <c_14> (ow-iw)/2 [16:34] <c_14> that was it [16:35] <nehaljwani> exactly. [16:35] <nehaljwani> now I understand the parameters [16:35] <nehaljwani> stupid googleing didn't help [16:35] <nehaljwani> thanks c_14 [16:40] <nehaljwani> why did debian guys fork ffmpeg and made avconv? [16:43] <nehaljwani> looks like it's a personal issue +1 to this :D [17:11] <pa> if i have a movie, say in mp4, mkv or some other nice container, and i get a byte offset, that is the exact offset of a keyframe, how do i find what keyframe that is / tell avconf to start transcoding exactly from that keyframe? [17:11] <sacarasc> avconv is part of libav, a fork of ffmpeg. Their channel is #libav. Or did you mean the libav* libraries? [17:12] <sacarasc> Wait, I brought up libav. [17:12] Action: sacarasc fails at think. [17:13] <pa> ah i thought that ffmpeg changed to libav completely.. so it's only a fork [17:13] <pa> weird [18:28] <t4nk438> i try to apply ``afade=t=in:st=0:d=3`` to a input audio stream but nothing changes. What can be the matter? [18:36] <t4nk438> sorry :*| http://pastebin.com/DV8qqrXa why afade filter doesn't work for this. [18:37] <t4nk438> http://pastebin.com/8HYcgixD [18:39] <c_14> Because if you don't name the arguments, the order is assumed to be type:start_sample:nb_samples:start_time:duration:curve: [18:39] <c_14> ignore that trailing ':' [18:40] <c_14> And 0 is not a valid number of samples to apply the effect for. [18:40] <c_14> Which is what it says on line 22 [18:40] <c_14> and 20 [20:13] <t4nk438> hi, is it a bug or normal behaviour? ``afade=t=out`` applied to the result of ``amovie`` makes it completely silent, ``afade=t=in`` applied to the result of ``amovie`` makes nothing. See http://pastebin.com/FupMtC4E The filters only apply as expected if input file is supplied via ``-i`` option. When i use ``amovie`` instead, things get corrupted. [21:09] <mychele> hi everybody. I'm trying to understand if I can do this thing with ffmpeg and ffserver. I would like to install ffmpeg on a raspberry pi (actually I'm doing that, it's compiling) and use it as an input for a ffserver feed. So far so good. Then I would like to stream this feed to devices with iOs and Android, but I don't understand clearly what can I do. Thank you very much. [21:37] <loadbang> When converting a directory of JPEG files in to a video, is the only way to do this is to rename all the JPEGs in a number sequnce? Wondering if there is a way to do everything in date order. [21:39] <llogan> loadbang: see the glob pattern [21:40] <llogan> http://ffmpeg.org/ffmpeg-formats.html#image2-1 [21:41] <llogan> ffmpeg -i -pattern_type glob -i "*.jpg" output [21:49] <loadbang> llogan: *.jpg: No such file or directory [21:50] <llogan> i don't know if it works in Windows [21:51] <loadbang> oh no, OS X some reason put instead of " [21:52] <llogan> you could use single quotes instead if you prefer [22:01] <dahat> Q: I'm developing an app to play basic videos with FFMPEG as a backend... and currently it accepts a local path into avformat_open_input(), now I am trying to expand it to target RTMP files, currently I can do the following at the command line, how can I apply the same args to avformat_open_input()? ffmpeg -rtmp_app <AppName> -rtmp_conn <AuthToken> -rtmp_playpath <PlayPath> -i rtmp://streaming.example.com/ [22:08] <shout-user93> hi [23:04] <_dunno_> 'error while opening encoder' <-- 'ffmpeg -ss 00:01:23.45 -i INFILE.MPEG2 -t 01:23:45.00 -vf yadif,crop=692:432:12:72,scale=352:288 -c:a libmp3lame -b:a 128k -c:v mpeg4 -b:v 1000k -vtag DX50 OUTFILE.AVI [23:09] <_dunno_> error while opening encoder for output stream #0.0 -- maybe incorrect parametres such as bit_rate, rate, width or heigth [23:10] <_dunno_> fflogger thanks for the reminder - but there is not much to paste - see above [23:13] <_dunno_> seems to be that scale part - but it's a multiple of 16 - and also within mpeg4p2 specifications, isn't it? [23:35] <_dunno_> as long as the scale values are bigger than the inputs original size it works - but i cannot downscale [00:00] --- Fri Sep 12 2014

1 0

ffmpeg-devel.log.20140910
by burek 11 Sep '14

11 Sep '14

[00:05] <cone-854> ffmpeg.git 03Pascal Massimino 07master:649b7a99463d: av_filter/x86/idet: use HADDD where appropriate [00:06] <Daemon404> i luckily dont care about systemd at all :D [00:15] Action: rcombs glances at launchd [00:16] <J_Darnley> How can you say that? Isn't it the one true init system? [00:16] <wm4> it should be ported to windows! [00:17] <J_Darnley> Yeah! I can'r believe Windows has so mnay places to start things. [00:18] <Daemon404> i dont particularily care what init system my linux box uses [00:18] Action: J_Darnley is appalled at his typing today. [00:18] <Daemon404> at least, it has not affected me in any practical way yet [00:18] <Daemon404> as a user. [00:22] <rcombs> seems like Windows's sucks and all the *nux ones work plenty well enough [00:22] <iive> wm4: i thought it is ported _from_ windows! [00:22] <wm4> I think it's most similar with launchd, so osx [02:38] <cone-854> ffmpeg.git 03Michael Niedermayer 07master:1b5ccae0f204: avformat/mp3dec: Improve seeking frame sync code [09:56] <rcombs> are we supposed to support targeting OS X 10.6 when building on a later OS X version? [09:56] <rcombs> (because right now, we don't if indevs are enabled, because the AVFoundation indev is enabled by default and there's no option to disable it) [09:58] <ubitux> no option to disable the AVFoundation indev? huh ? [09:58] <ubitux> --disable-indev=avfoundation ? [09:59] <rcombs> oh, derp, forgot that option existed [09:59] <rcombs> danke [11:58] <cone-557> ffmpeg.git 03Michael Niedermayer 07master:3294fc8c474b: avformat/concatdec: fix "warning: explicitly assigning a variable of type int to itself" [14:45] <thardin> does anyone in here know the state of very low bitrate video codecs? say 20 kbps and under [14:48] <kierank> hevc at like 5fps i guess [14:48] <kierank> but at that bitrate you're looking at high overhead for headers etc [14:49] <thardin> yeah, I noticed that [14:50] <thardin> ran some tests with x264, and just the NAL header is 4% [14:50] <thardin> is x265 useful right now? [14:53] <Daemon404> thardin, at such low bitrates, it could possibly be [14:53] <Daemon404> testing encouraged [14:54] <thardin> I'll try to remember when getting home [14:55] <thardin> should probably hack both to not output the version string, since that skews results too much [14:56] <kierank> yeah the version string causes vbv problems on the first frame in zerolatency as well [14:56] <thardin> no foreman_qcif.y4m that string ends up eating another 3% [14:56] <thardin> on [14:56] <thardin> found some interesting papers on this problem too [15:18] <cone-182> ffmpeg.git 03Michael Niedermayer 07master:8b7a39d7ea4b: avcodec/adpcmenc: Use FF_ALLOC_ARRAY_OR_GOTO() [15:19] <cone-182> ffmpeg.git 03Michael Niedermayer 07master:7fe59eeaaf22: avcodec/ac3enc_template: Use FF_ALLOC_ARRAY_OR_GOTO() [15:19] <cone-182> ffmpeg.git 03Michael Niedermayer 07master:c4a0c64f1450: avcodec/aacenc: Use FF_ALLOCZ_ARRAY_OR_GOTO() [15:20] <michaelni> thardin, might be fun to try snow, i dont remember how it looked at xy kbps but it looked better than mpeg4-asp [15:20] <michaelni> at low bitrate that is [15:25] <Compn> x264 had some ultra low bitrate stuff, if you have low motion [15:25] <Compn> snow was nice too :) [15:25] <Compn> http://radar.weather.gov/Conus/Loop/centgrtlakes_loop.gif [15:26] <Compn> its gonna rain [15:33] <J_Darnley> Oh god, does this increment fbp before after setting the j'th value? "fbp++[j] = whatever" [15:33] <J_Darnley> *before or after [15:36] <ubitux> after [15:39] <J_Darnley> thanks [15:56] <thardin> michaelni: not a bad idea. one paper I found used an H.264/DWT hybrid [18:05] <mateo`> ou ça ? [18:05] <mateo`> wooops worry :( [20:53] <wm4> no Libav 11 yet... [20:56] <nevcairiel> didnt they talk about releasing it all weekend [20:56] <nevcairiel> guess talk is easy, doing not so much? :) [20:57] <cone-182> ffmpeg.git 03Michael Niedermayer 07master:6f48c60956aa: avcodec/twinvq: Use FF_ALLOC_ARRAY_OR_GOTO() [20:57] <cone-182> ffmpeg.git 03Michael Niedermayer 07master:743d489c9a26: avcodec/mpegvideo: Use FF_ALLOCZ_ARRAY_OR_GOTO() [21:57] <iive> nevcairiel: "we are not paid to do it" [22:54] <cone-182> ffmpeg.git 03Michael Niedermayer 07master:7258b5eb65d8: avcodec/dnxhdenc: Use FF_ALLOCZ_ARRAY_OR_GOTO() [22:54] <cone-182> ffmpeg.git 03Michael Niedermayer 07master:3e56ae67d300: avcodec/h264: Use FF_ALLOCZ_ARRAY_OR_GOTO() [00:00] --- Thu Sep 11 2014

1 0

ffmpeg.log.20140910
by burek 11 Sep '14

11 Sep '14

[00:00] <c_14> You can't use audio filters when stream copying. Since you have a pcm stream, just reencode it when demuxing, or when muxing it doesn't matter. [00:00] <c_14> Add the asetpts audio filter when you reencode the audio. [00:01] <joules> c_14: can't explain it. [00:01] <c_14> works now? [00:03] <joules> no i'm annoyed because I'm sure I gave the same command and with the deint it took three hours and looked like crap in theend. [00:04] <joules> and yes it looked better with mcdeint (sport video). [00:04] Action: Suchiman working on 58GB MKVs take some time [00:05] <joules> Suchiman: x264 supports threads. [00:05] <Suchiman> yeah but if you just want to get the smaller Audio stream it still Needs to seek through the whole 58GB [00:06] <joules> Suchiman: probably mostly IO, working on a ssd scratch disk or ramdisk (I have 72GB). [00:07] <joules> ? [00:07] <Suchiman> yeah, USB 3 HDD, CPU is boring itself at 3% [00:08] <joules> oh no. USb3 is soo noob, [00:08] <joules> but I'm sure the actually HDD is the bottle necik. [00:08] <Suchiman> ffmpeg doing 75MB/s IO on HDD [00:08] <Suchiman> yeah [00:09] <joules> I use a SSD scratch disk and ramdisk for most editing. [00:09] Action: Suchiman should get that 128GB Hynix DDR4 module [00:09] <Suchiman> just need a CPU to handle that :/ xD [00:11] <joules> thunderbolt does 20G http://en.wikipedia.org/wiki/Thunderbolt_%28interface%29 [00:11] <joules> I have 12cores. [00:12] <Suchiman> c_14: it didn't seem to have any effect: http://pastebin.com/sqAVJRN1 [00:12] <Suchiman> joules: um... i fear my 4670K @ 4,4GHz can't stand your 12 cores [00:13] <joules> lol westmere-EP [00:14] <Suchiman> joules: uhh... talking about hyper threaded cores :P ? [00:15] <joules> that would make it 24hthreads. [00:15] <c_14> Suchiman: hmm, the audio should be about 02:07:45.01 ? [00:15] <Suchiman> c_14: yeah but according to ffprobe it's still a second longer [00:15] <Suchiman> and still at 48kHz [00:19] <joules> c_14: well it's doing the lossless version, and it will dump to rawvideo pretty fast, it's the whole deint that slows it down. i have no idea what happened :/ [00:21] <c_14> Suchiman: hmm, I have a couple things I'd like to try, but it depends on how patient you are. There's the quick and dirty solution, and there's the searching for why it's messing up solution. [00:22] <Suchiman> c_14: i've already wasted days on this shit, a few more hours can't hurt. i'm planning on digitalizing old VHS tapes [00:23] <Suchiman> unfortunately ffmpeg can't use the grabber device directly because the DShow filter for the device uses crossbar [00:23] <c_14> Right, try extracting the audio with -af aresample=async=1000 [00:24] <c_14> That'll stretch/squeeze samples to the audio timestamps. [00:24] <c_14> So in the hope that the audio timestamps are correct it'll squish the audio down to fit the timestamps. [00:24] <Suchiman> ahh, that sounds like an idea [00:26] <Suchiman> c_14: by the way, maybe it's worth noticing, when i captured the streams in directShow (i used a graph editor to construct a DShow graph manually) into a avi Container it always desynced so it seems to be a piece of magic by the MKV muxer filter [00:29] <c_14> Magic just makes everything better. [00:29] <Suchiman> ;) [00:30] <joules> sweet [00:30] <joules> Suchiman: just I"m fixing some old broadcast [00:31] <joules> mkv as lossless is good archive format, but for editing I go raw formats. [00:32] <Suchiman> codec* mkv is a Container ;) [00:33] <Suchiman> c_14: that Looks better [00:33] <Suchiman> Input: Duration: 02:19:02.92 -> Output: Duration: 02:19:02.85 [00:35] <joules> mkv takes rawvideo? [00:35] <c_14> I'd probably call that close enough. To whomever can discern a 70 millisecond desync I will give a prize. [00:35] <c_14> joules: sure, I said so earlier [00:35] <joules> alrighty then. [00:36] <Suchiman> c_14: gimme some minutes, i will denoise the Audio and then try to remux it [00:36] <joules> not sure if alot of gui editors support mkv though. Just been using avi for ages. [00:37] <Suchiman> joules: there is for example a mkv plugin for virtualdub (:s) but you're right, there are way more editors for avi [00:37] <joules> yeh I just downloaded lightworks the other day. [00:49] <joules> well 30G(lossless) is way better than 80G(raw). Almost done, see if lightworks can take it. [00:51] <Suchiman> joules: well 58GB lossless for me, i don't want to even try raw ;) [01:08] <Suchiman> c_14: :D [01:08] <c_14> Is that an it's working smiley? [01:09] <Suchiman> Original MKV: 02:19:02.92; Resampled Audio: 02:19:02.85; Remuxed MKV: 02:19:02.91; Sync: perfekt ;D [01:09] <c_14> yay [01:09] <Suchiman> many thanks, you've ended my misery [01:09] <c_14> np [01:09] <c_14> Now if only I could find that other guy with the same/a similar issue from earlier today... [01:11] <c_14> benbro: you still around? [01:11] <Suchiman> 00:21:57 <c_14> Suchiman: hmm, I have a couple things I'd like to try, but it depends on how patient you are. There's the quick and dirty solution, and there's the searching for why it's messing up solution. <-- anything else you've wanted to try for academic purposes? [01:12] <c_14> Nah, it was mainly if that didn't work to try adding some more resampling/pts setting options on top of that. [01:16] <Suchiman> then another question: while audacity does a very good job at denoising the audio, are there any filters in ffmpeg (Zeranoes builds) which can improve PAL movie quality? and if i cut the black borders of the video, VLC starts showing two sizes for the video, resolution and display resolution [01:17] <joules> can't polish a turd. [01:17] <joules> what do you mean improve :) [01:17] <Suchiman> joules: making it look less like turd ;) [01:18] <joules> http://www.ffmpeg.org/ffmpeg-filters.html#toc-setdar_002c-setsar [01:19] <joules> if you want to crop and scale, best to stick the standard scales. [01:20] <Suchiman> for example, it has some... "ghosting"? meaning objects are surrounded by conspicuous dark / bright contours and the video has "interlacing like" issues, you can slightly see that the first line is slightly offset to the second line, but every second line is aligned [01:21] <joules> color bleeding? [01:21] <Suchiman> probably, i will get a pic [01:21] <joules> can't do anything about that, you can deinterlace though. [01:23] <joules> so the pic with no aspect btw. [01:25] <Suchiman> http://suchi.teneon.de:8080/video.jpg [01:26] <Suchiman> joules: so if you look at the right door frame above the head thats what i mean by "interlacing like" and at the left side of the head you see this "ghosting" or color bleeding [01:27] <joules> umm 1920×1080? [01:27] <Suchiman> Screen pic in VLC [01:27] <joules> that is interlaced frames. [01:28] <joules> if you want to tidy it up, you should start with a noaspect screenshot. [01:29] <joules> theres also cropdetect, but I find it does too much. [01:30] <Suchiman> joules: it does nothing in my case, the lighting of the black is too bright [01:30] <debianuser> Hello, I have an unusual problem. I have an mkv file with subtitles (SRT), and I need to convert it to mp4, so I'm trying to embed subtitles into a video. And the hardest part is that the video is half-side-by-side 3D. I know how to keep it 3d (-x264opts frame-packing=3), but how can I embed subtitles there? Any hints to what I should look at? [01:30] <joules> Suchiman: you one windows? [01:31] <Suchiman> joules: yes [01:32] <debianuser> (I mean I can go with "-vf subtitles=subs.srt", but then subtitle spans both frames which breaks completly when switched to 3d) [01:36] <c_14> debianuser: maybe use the stereo3d filter to convert to a different stereoscopic image format like alternating frames maybe, throw the subtitles filter on that, and then convert back using the stereo3d filter? [01:37] <c_14> Not sure how that will look once it's 3d again, but... [01:38] <c_14> You could also try the above below 3d, but then the subtitles would only be on one eye... [01:38] <debianuser> interesting idea, thanks! let's see... [01:45] <debianuser> it _almost_ worked. :( I get two equal copies for left and right eye, but they're squeezed horizontally (because initially those were half-side-by-side). I guess now I'll need to play with scale filters... [01:48] <c_14> I'd probably scale the input to iw*2 do the stereoscopic filter, then subtitles, then stereoscopic then scale to iw/2 [01:48] <c_14> But I'm not sure the order makes a difference since it's up/downscaling anyway... [01:54] <debianuser> c_14: With scale+stereo3d+subtitle+stereo3d+scale it works! Thanks for the idea! [02:00] <c_14> np [02:00] <c_14> I honestly wasn't sure that would work though. [02:03] <joules> lightworks is a no go for matroska [02:04] <debianuser> It works. :) Now I'm looking for the way to increase subtitle font a bit. I can downscale, add subtitle then upscale, but that noticeably decreases image quality. [02:05] <Suchiman> joules: what did you mean by noaspect screenshot? [02:10] <joules> frame size [02:11] <joules> if cropdetect isn't cutting it you have to figure it out manually. [02:12] <joules> I use kruler (kde desktop) or you could just load it in an image editor. [02:12] <Suchiman> joules: ffplay remuxed.mkv -vf "crop=704:432:8:73[crop]; [crop]setdar=dar=16/9" gives http://suchi.teneon.de:8080/cropped.png [02:12] <joules> looks like lightworks can't even handle this avi. [02:13] <joules> is that vlc? [02:13] <Suchiman> um... no [02:13] <Suchiman> thats ffplay ;) doh [02:14] <joules> ok - framesize snapshot is more useful to people (well to me anyhow). [02:17] <joules> hmm and lightworks wants my money. [02:18] <Suchiman> pitty that VLMC does not get dev's [02:19] <benbro> c_14: I'm here [02:20] <c_14> About your audio length/sync problem from earlier: just for academic purposes/future reference could you try extracting the audio from the video again using -af aresample=async=1000 ? [02:21] <joules> kdenlive supports all (?!) ffmpeg formats it seems. [02:23] <benbro> c_14: what do you mean by extracting? [02:24] <c_14> eh, the one where you had your video as input and the wav as output [02:24] <joules> is there such a thing as lossless mpeg2video? [02:24] <benbro> c_14: this is what I have http://dpaste.com/24SB0MH [02:24] <c_14> let me check the logs [02:25] <benbro> c_14: I have mkv where the audio lags behind the video. [02:25] <benbro> when stretching the video by a factor of ~1.08992 with mkvtools the audio and video are almost in perfect sync [02:25] <benbro> so the factor is constant throughout the video [02:26] <joules> benbro: yeh i mentioned PTS to you yesterday. [02:26] <c_14> It was this paste: http://dpaste.com/3J4TYB9 [02:26] <c_14> They just went ahead and deleted it though. [02:27] <benbro> it's the sam epast5e [02:27] <benbro> past [02:27] <c_14> Does/did your source video have the audio lag? [02:27] <benbro> yes. I captured the screen with avconv and now I'm trying to fix it with ffmpeg :) [02:28] <c_14> ah, ok. So it was a similar issue but not quite the same. [02:28] <c_14> Just wanted to check because there could have been a more general solution. [02:28] <benbro> what is your solution? [02:28] <benbro> I saw something about -af asetpts=0.91875*PTS [02:29] <benbro> the factor I found is 0.9175 [02:29] <c_14> the solution would have been using the async feature of aresample, but that won't work if the source video is already misaligned. [02:29] <benbro> I don't understand why. 44.1/48 makes sense [02:29] <benbro> what do you mean by source video? [02:29] <benbro> I just captured audio and video from my desktop [02:29] <c_14> Yes, that captured video would be the source in this case. [02:30] <benbro> I don't know if it's misalined [02:30] <joules> benbro: didn't -vf setpts=0.999*PTS work? [02:30] <benbro> joules: how to use it? [02:30] <joules> ffplay to check. [02:31] <benbro> ffplay -vf setpts=0.999*PTS capture.mkv ? [02:32] <joules> tias [02:33] <benbro> what does 0.999*PTS do? [02:34] <joules> !pts [02:34] <joules> http://www.ffmpeg.org/ffmpeg-filters.html#setpts_002c-asetpts [02:35] <benbro> but why 0.999 ? [02:35] <benbro> is it arbitrary/ [02:36] <benbro> ? [02:36] <joules> 29.97fps [02:37] <joules> are you sure it's the video? [02:37] <benbro> what do you mean? [02:38] <joules> ok i'm done guessing..back to work. [02:38] <benbro> thanks :) [02:38] <benbro> my factor is 0.9175 not 0.999 [02:40] <joules> asetpts correct? [02:41] <joules> so it works then great?! [02:41] <benbro> 0.9175 almost work. I can find better number but I'll just use it [02:41] <benbro> I still don't understand why the capture is bad [02:41] <benbro> I'll try to compile ffmpeg on ubuntu and use it [02:42] <benbro> but I couldn't find a simple capture command (not even on the wiki) [02:42] <benbro> I also couldn't make it work on windows without installing other drivers (which I don't trust) [02:42] <joules> ¯ [02:43] <joules> avconv sux ¯\o/¯ [02:43] <benbro> ? [02:43] <benbro> yes [02:43] <benbro> so ffmpeg will give me better results when capturing the screen? [02:44] <c_14> https://trac.ffmpeg.org/wiki/Capture/Desktop ? [02:44] <joules> unless you are DIW :) [02:44] <joules> I'm really disappointed in lightworks. [02:44] <benbro> not sure what it means [02:44] <benbro> yet, it's not trivial to install ffmpeg on ubuntu [02:45] <benbro> avconv -f pulse -ac 1 -i alsa_output.pci-0000_00_05.0.analog-stereo.monitor -f x11grab -framerate 15 -s 1280x960 -i :0.0 -acodec pcm_s16le -vcodec libx264 -preset:v ultrafast -crf 0 -threads 0 capture.mkv [02:46] <benbro> ^^ this is what I'm using with avconv. it's not trivial but seems to be very common [02:46] <benbro> I'm not sure what all the options does [02:46] <c_14> What's up with that alsa input man... [02:46] <benbro> c_14: I don't know but that what worked for me [02:46] <benbro> without it, I couldn't capture audio [02:47] <c_14> If you're using pulse, couldn't you just ffmpeg -f alsa -i pulse [foobar] [02:47] <c_14> It'll use whatever's the default pulse audio capture device. [02:47] <benbro> let's start with something basic [02:48] <benbro> is there an "official" way to install ffmpeg on ubuntu 14.04? [02:48] <c_14> There's the compilation guide. [02:48] <benbro> that's hard [02:49] <c_14> It's mainly copy+paste tbh. [02:49] <benbro> https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [02:49] <benbro> that's not trivial [02:50] <benbro> and is complex for no reason [02:50] <c_14> It _is_ mainly copy+paste though. [02:50] <c_14> If you have suggestions, I will gladly edit the page. [02:50] <benbro> a deb package [02:50] <benbro> dumb users like me a scraed by such steps [02:50] <Nanashi> What else does "installing" do besides letting you type ffmpeg in a term? [02:50] <benbro> that's the first contact I have with ffmpeg [02:50] <benbro> but I just want to start using it [02:51] <benbro> not read some compliation guide [02:51] <Suchiman> mhh... cropping to exact video (702x430) size and then letter boxing to 16:9 ? or scaling? what would be the best solution? or even keeping it at 702x430? [02:51] <benbro> I'm not trying to be rude [02:51] <benbro> just honest [02:51] <benbro> I also tried this https://github.com/stvs/ffmpeg-static [02:52] <benbro> but it's not perfect either [02:52] <joules> Suchiman: for achiving no just set the DAR. (IMO), [02:53] <c_14> Nanashi: not much [02:53] <Suchiman> joules: what about the SAR? when setting DAR the result is somewhat [SAR 3440:3159 DAR 16:9] [02:53] <joules> Suchiman: also when you say letter boxing, do you intend on adding black bars? [02:53] <benbro> c_14: that's why I can't use ffmpeg on ubuntu https://github.com/pyke369/sffmpeg/issues/9 [02:53] <c_14> benbro: There are efforts to getting FFmpeg into the Debian (and then Ubuntu) repos. But it might take a while. [02:53] <Suchiman> joules: yeah, for making the encoder happy with a default ratio / something that is devideable by 16 [02:54] <Suchiman> c_14: are you an ffmpeg dev? [02:54] <benbro> c_14: first step could be an official debian/ubuntu repository :) [02:54] <benbro> by official I mean from the ffmpeg team [02:54] <c_14> Suchiman: I have patched FFmpeg, but I would not consider myself an FFmpeg dev. [02:56] <c_14> benbro: there's this ppa: https://launchpad.net/~jon-severinsson/+archive/ubuntu/ffmpeg [02:56] <benbro> yes, a personal ppa [02:56] <benbro> I can't trust it on my server [02:56] <benbro> I don't know this guy [02:56] <joules> Suchiman: 704×480 is a standard res (AFAIK). [02:56] <benbro> I'm sure he is a great dev but I don't know him [02:57] <joules> Suchiman: So not PAL? [02:58] <joules> i'm using ffmpeg on debian?! [02:58] <joules> or did I get it from dmm.org? [02:58] <joules> yah I did. [03:01] <Suchiman> joules: well it's shitty i think ;) the grabber grabs at 720x576 which is digital PAL?, the source material is analog PAL which is 704x576? or 702x576? and this contains an anamorph 16:9 video?! something like that, i hate analog shit ;) xD [03:01] <joules> 704 [03:03] <c_14> benbro: all I know is that the stones are rolling on getting FFmpeg into Ubuntu + Debian, and barring any major roadblocks it should get there soon". [03:03] <c_14> Where soon is by debian's definition. [03:03] <Suchiman> 20years* [03:04] <joules> lol [03:04] <joules> thus we have dmm [03:04] <Suchiman> whoever forked avconv, i hate him for confusing me that much [03:05] <joules> no hate some guy for being an maintainer@debian [03:05] Action: joules checks. [03:05] Action: joules mode +b debian [03:07] <Suchiman> [swscaler @ 00000000042f3d80] Warning: data is not aligned! This can lead to a speedloss <--- by the way, why is data misaligned ;) [03:07] <joules> what's the final res? [03:08] <Suchiman> 704x430 [SAR 215:198 DAR 16:9] [03:08] <joules> dunno [03:09] <Suchiman> interestingly VLC states that resolution is 704x450 while display resolution is 704x430 [03:12] <joules> needs deinterlacing too. [03:13] <joules> well I'm probably on +kb on #debian [03:20] <joules> http://www.lwks.com/index.php?option=com_content&view=article&id=100&Itemid… <-which one of the uncompressed are a good match? tried png/yuv4/rawvideo so far. [03:24] <c_14> do you need uncompressed or would lossless H.264 be ok? [03:33] <Suchiman> any recommendations on deinterlace filter? ffmpeg has many :o yadif seems to be good [03:41] <relaxed> use yadif [03:41] <Suchiman> ok [03:52] <active8> !paste [03:53] <active8> !welcome [03:53] <active8> !topic [03:53] <active8> nice [03:53] <joules> Suchiman: and if it's pal use bff but check it (ffplay). you will see it not working when theres jitter. [03:54] <joules> that's it, if lwks, can't take a simple avi yuv4 encoded. then I can use it. ;d [03:54] <joules> s/can/can't/ [03:54] Action: joules hnds are cold it's true. [03:55] <Suchiman> joules: whats the difference? could i really see it which one of These lines is the first, how would it influence the result [03:57] <Suchiman> joules: i can't tell if there is an difference between tff and bff [04:01] <joules> try a scene with some movement. [04:04] <joules> oh yadif=1:1 sorry. render the fields. [04:05] <joules> (playback) [04:05] <Suchiman> joules: so yadif=parity=0 doesnt seems to be different from yadif=parity=1 even in quick movement [04:08] <joules> hang on it's a movie. [04:09] <Suchiman> and send_field also doesn't seems to change the effect ;) [04:10] <joules> anyone? [04:11] <joules> Suchiman: it's not captured as fields (movie). [04:11] <Suchiman> so it doesn't really matter in the end? [04:11] <joules> look at pullup [04:12] <joules> never used it. [04:12] <joules> http://www.ffmpeg.org/ffmpeg-filters.html#pullup-1 [04:12] <Suchiman> is PAL teleclined? i think it's just played faster [04:13] <Suchiman> therefore also known as PAL acceleration [04:13] <joules> movies are over broadcast. [04:13] <joules> http://www.ffmpeg.org/ffmpeg-filters.html#dejudder [04:15] <Suchiman> i'm neither converting from film to NTSC or PAL to NTSC [04:16] <Suchiman> i'm in germany, PAL country [04:18] <Suchiman> so currently i'm at: ffplay -i remuxed.mkv -vf "yadif[deint]; [deint]crop=w=698:h=430:x=10:y=74[crop]; [crop]pad=width=704:height=432:x=3:y=1:color=black[dar]; [dar]setdar=dar=16/9" which gives already a good improvement over the source material [04:18] <joules> Suchiman: interlaced content is captured in fields and broadcast in frames so PAL@50, so a movie (progressive) played over broadcast are telecined [04:20] <joules> depends on the content, if it was broadcast captured video (interlaced) or broadcast telecined movie. [04:21] <Suchiman> DVB ftw :o [04:21] <Suchiman> die interlacing die!!!1 [04:23] <joules> pullup! [04:23] <joules> interlacing is ok [04:24] <Suchiman> joules: with which options? assuming it's 25fps broadcast and 24fps movie [04:24] <joules> consider it 25fps [04:25] <Suchiman> "PAL uses a similar system, "2:2 pulldown"." [04:25] <joules> actully I don't know. Never played with it. [04:27] <Suchiman> anyway, it's extremely late here, i must go to sleep now if i want to stand up in 4h, you guys are stealing my sleep ;D [04:32] <joules> lwks stole my morning. [04:41] <Suchiman> *short re-awakening* and thanks all for the help :) [04:52] <active8> I'm trying to grab audio only from the computer - using a youtube video. It's protected from downloadhelper of course, or I wouldn't have to do this. I'd rather be able to just grab audio while I do other things and grab it in raw PCM to use fewer CPU cycles. I pasted my efforts here: http://pastebin.com/Q9pf1pt2 [10:00] <termos> I keep getting immediate exit requested messages from av_interleaved_write_frame, they seem to happen at somewhat random intervals [10:00] <termos> 9 minutes to 25 minutes [10:00] <termos> after encoding has started [10:07] <Keshl> termos: If it's a bug with ffmpeg I can't help at all, but I've run into similar issues with other software. Just a few sanity checks real fast.. Is your system running hot, and is your disk filling up, are you running out of ram/swap? [10:11] <Keshl> Oh, and, if you're overclocking, back off .2 ghz even if you don't think you need to. Same with your RAM. The fact that this is a sustained load can mess with an otherwise appearntly stable overclock. x.x [10:12] <termos> Ah, I will try to monitor those things when it happens again. Could be worth mentioning that it happens when writing hls segments to disk [10:13] <termos> they shouldn't be running hot, the load is very small about 7-10% cpu load [10:14] <Keshl> Odd.. [10:50] <K4T> hello [10:50] <K4T> frame=116272 fps= 50 q=23.0 size= 1275863kB time=00:38:45.42 bitrate=4494.6kbits [10:50] <K4T> what is that 'q' parameter? [11:00] <relaxed> quantizer [12:08] <joules> anyone aware of crop and scale supporting timeline editing? Appears not with my version. I can see why, but I promise to retain the aspect ratio. [12:37] <Keshl> joules: I don't have an answer for you, but if you ever get one, _please_ tell me. Same deal here, same promise. <É< [13:03] <joules> Keshl: I don't think it would be supported. [13:05] <Keshl> It might be. Ffmpeg supports a bunch of weird stuff you wouldn't think it should.. XD [13:05] <joules> unless checks where coded in to make sure the output (crop/scale) is consistent. [13:05] <joules> For now I just punch out the frames. [13:10] <joules> I wonder what would happen if a encoder received a frame a difference size to the previous? Something bad would happen? [14:22] <emulator> I am trying to build the example transcoding.c, but for some reason once built it either gives me a message that the output file could not be opened ( not permitted) IF ran on a VOB file... or it segfaults when ran on an avi [14:23] <emulator> Can anyone help me figure out why the example seems to not work? [14:33] <valder> I am in desperate nee dof assistance. I'm writing some android native code using ffmpeg to mix video and audio together. I'm not sure how to figure out this issue I am having. The code keeps on crashing. I don't know how to get the error code or condition why I get sigsegv errors during execution of the av_interleaved_write_frame method. Is there anybody around that can look at what I wrote and help me figure [14:33] <valder> out what is wrong? [14:35] <valder> Here's the pastebin I have: http://pastebin.com/d4pbWVTk [14:50] <emulator> @valder What version are you using? [14:50] <valder> @emulator: of ffmpeg? the latest tag release [14:51] <emulator> Hmm... looking at your code... (but I may not be of much help) [14:52] <valder> v2.3.1 [14:54] <valder> @emulator: thanks. I can't figure why it fails on the av_interleaved_write_frame. My code is very similar to other SO examples I've come across. The input is a MP4 video file taken from the camera on my GalaxyTab [14:55] <valder> @emulator: what I'm trying to do is take that original video and return a shorter version of it basically. I know the inframe and outframe locations of each of the snippets I want to keep. [14:59] <valder> @emulator: please tell me if what I am doing is completely ass-backwards or somehitng.. I'm new to using ffmpeg. I'm used to working at higher levels. On iOS all of this stuff is handled in AVFoundation classes. [14:59] <emulator> @valder ... looking at the code... I don't see anything wrong... Have you checked your build options for ffmpeg. (No worries I am a newbie too.) [15:00] <valder> @emulator, what options am I looking for? [15:00] <valder> I basically copied somebody's instructions on building ffmpeg for Android. so I don't know what needs to be turned on/off. [15:01] <emulator> Honestly... I don't know... but I know that some errors on running can be caused by mis-configured ffmpeg builds. [15:01] <emulator> I am trying to figure out why an example from ffmpeg website is not working on THIS fedora box... [15:02] <valder> @emulator: here's my build script: http://pastebin.com/t8KDVehz [15:04] <emulator> @valder :'( [15:06] <valder> @emulator: perhaps I'm missing a few flags.. I found a few other scripts that enable different types of muxers [15:06] <emulator> @valder I don't see anything wrong... but you could try manually going to your build directory and checking what options the configure script has and see if you notice anything. (./configure --help ) See if I overlooked anything [15:06] <emulator> @valder that is what I am thinking might be the problem... [15:06] <valder> @emulator: you mean the ffmpeg build right? [15:06] <emulator> yes. [15:07] <valder> @emulator, any idea on how I can get the error message other than a segfault to appear in logcat? [15:08] <emulator> Nope :( [15:08] <valder> @emulator: thanks [15:08] <emulator> @valder you could pepper your source with debug statements.... that could help. [15:10] <valder> @emulator: I did, that's how I figured out where it crashes [15:10] <valder> notice the DLOG statements? DLOG is my macro to android logging [15:10] <emulator> @valder: Yeah.. I assumed that. [15:11] <emulator> @valder: but I wasn;t going to assume and take it as fact. [15:11] <valder> @emulator: only issue is I don't know how to print out an arbitrary object. [15:12] <valder> for instance the args to av_interleaved_write_frame, how can I do a dump of those values? [15:12] <klaxa|work> looks like twitter in here [15:15] <emulator> Have you tried just passing them into a debug message? That *sometimes* works [15:18] <emulator> @valder: you may also want to build with debug. [15:19] <valder> @emulator: mind if we go into private chat? seems our conversation is annoying a few people. [15:25] <klaxa|work> if you are referring to me, i was just trying to make a witty comment [15:27] <emulator> Apparently we didn't get it... [15:27] <klaxa|work> s-sorry [15:27] <emulator> No worries... (I don't mind) [15:29] <klaxa|work> i mean you both were obviously talking to each other, i prefer not to prefix my messages with a nick then, but that's a matter of taste [15:30] <klaxa|work> and etiquette i guess [15:31] <emulator> Ehh... I like making it easy for users... and it gets confusing following a conversation when multiple are going on(different channels, different etiquette) [15:31] <valder> klaxalwork: I didn't want to exclude anybody if they had anything to add to the conversation. private chats exclude additional help [15:32] <valder> different channels different rules. klaxalwork: if you have anything that can help me I would greatly appreciate it. [15:32] <klaxa|work> hmm yes i guess i would have been better off just reading :) [15:33] <klaxa|work> i am not exactly familiar with the C API of ffmpeg [15:34] <valder> neither am I that's why I'm here. :) SO only goes so far it seems. :) [15:35] <klaxa|work> does it crash at the very first call of av_interleaved_write_frame() ? [15:35] <klaxa|work> or does it write a few frames and crashes then? [15:36] <klaxa|work> because the pFormatCtx variable is initialized by opening a file, i'm not sure you can just use it and write there too [15:37] <valder> crashes on first call [15:38] <klaxa|work> does av_interleaved_write_frame() also take an OutputFormatContext? [15:38] <valder> do I need to create a new AVFormatContext var and pass that to av_interleaved_write_frame? Or are you saying I'm missing some configuration for pFormatCtx [15:39] <klaxa|work> i think you have to allocate a new formatcontext, yes [15:40] <valder> so is each formatContext tied to a specific file or write call? [15:41] <klaxa|work> i'm not sure myself, but the documentation for av_interleaved_write_frame() says it's a "media file handle" so i would think you have to use a different file for writing than you use for reading [15:41] <klaxa|work> did you look at the remuxing.c example code? [15:42] <klaxa|work> maybe take a look at: https://www.ffmpeg.org/doxygen/trunk/remuxing_8c-example.html#a25 [15:42] <valder> hmm ok.. I think I got confused.. been looking at too many examples. some showed writing using only a single context and others with two [15:42] <klaxa|work> avformat_alloc_output_context_2() [15:42] <klaxa|work> oh? [15:42] <klaxa|work> maybe i'm completely wrong [15:43] <klaxa|work> i only took snippets from remuxing.c too [15:43] <valder> yeah.. it could also be that the code I used as reference was older too [15:43] <klaxa|work> those parts worked though [15:44] <valder> i've had to go through hell the last few weeks trying to rebuild this code i'm working on. easily could have munged things up.. thanks1 [16:45] <emulator> I still can't get the transcoding.c example to work... does anyone know why? Or can give pointers or anything? [16:50] <emulator> It creates the file... just fails to open it. I think. [16:53] <emulator> Passing into it an avi causes a segfault... [16:55] <emulator> Figured it out. :-) THANKS [17:43] <valder> klaxalwork: in the sample code you linked to, the output file is built using the input file's codec as a reference. What if I already know what output codec I want to use? Where is the documentation for that? I assume its a constant somewhere where I can say I want the output format to be MP4/H.264 or something right? [17:46] <valder> n/m just noticed klaxalwork logged out. [17:47] <valder> can somebody assist me again? :) in https://www.ffmpeg.org/doxygen/trunk/remuxing_8c-example.html#a25 The output file is conifugred using the input file's information. How do I create the output file without using the input file? [20:00] <atr2006> libx264 requires that frame width and height are even numbers. Is it restriction of H.264 standard or just its implementation in libx264? [20:01] <JEEB> it requires what your picture type requires [20:02] <JEEB> 4:2:0 is mod2 width and height [20:02] <JEEB> 4:2:2 is mod2 width and mod1 height [20:02] <JEEB> 4:4:4 is mod1 width and height [20:02] <JEEB> chroma subsampling :P [20:04] <atr2006> oh yes, i suspected that it is due to chroma subsampling :) thank you very much [20:04] <JEEB> np [20:23] <BtbN> h264 even requires the width/height to be multiples of 16 [20:23] <BtbN> not just even numbers [20:24] <BtbN> That's why 1080p videos are encoded as 1920x1088 [20:24] <BtbN> + a flag saying crop the bottom 8 pixels [20:25] <JEEB> yes, but that isn't visible to the user [20:25] <JEEB> but yes, if we're really pedantic AVC encodes in macroblock sizes [20:25] <JEEB> I think the limitations change a bit with interlacism, too [20:26] <JEEB> in HEVC the internal size depends on the coding tree maximum size [20:26] <JEEB> which can be up to 64x64 [20:29] <emulator> In C how do you prevent the AVCodecContext from making the output significantly smaller? For simplicity I am working with the transcoding example, but I can't seem to figure out how to not let it get shrunk to about 1/4 the original size. [20:33] <onyx> c_14: is there a way to make the video rendering not stop when 1 frame is missing? [20:34] <emulator> Would that be qmin and qmax? To prevent the shrinkage? [20:35] <BtbN> Why do you even reencode it if you don't want it to change at all? [20:36] <BtbN> The only real point in re-encoding is switching to a diffrent codec or reducing its size [20:36] <emulator> I want to be able to control the size of the file... not not get it smaller but control how much [20:44] <c_14> onyx: you could switch from the sequence pattern type to glob pattern type. [20:44] <onyx> ooooh great [20:44] <onyx> I think I saw that before [20:44] <onyx> yeah yesterday there were missing 3 frames [20:45] <onyx> so the rendering stopped [20:45] <onyx> I had to manually duplicate some images and rename them for it to finish rendering [21:38] <bryno> hi! this might seem silly, but... v2.3.3 is considered the latest "stable", right? [21:39] <sacarasc> That's what the topic says. [21:40] <bryno> sorry. i look at the site and see 1.2.8, 2.2.7, and 2.3.3 listed. i don't really know the differences :D [21:42] <llogan> for general users the latest "stable" is considered to be the latest from git master ~98% of the time [21:42] <llogan> the various release numbers are different release branches [21:43] <bryno> that's what i typically went with long ago, before versions were put out more often :S [22:01] <Mavrik> bryno, yes, 2.3.3 is the newest [22:01] <Mavrik> the older versions are there for people who have legacy software relying on those versions [22:02] <bryno> i'm in that spot of deciding which version to go with: 2.3.3 or git master [22:02] <JEEB> just go with master unless something is specifically broken [22:03] <JEEB> or well, nothing says that the release isn't [22:03] <JEEB> since there is no assurance that someone remembers to maintain the release branches with bug fixes [22:03] <JEEB> welcome to open source multimedia [22:03] <JEEB> at least for latest master you can check fate.ffmpeg.org [22:04] <JEEB> and see the test results on various platforms [22:04] <Mavrik> I usually grab the release just to have reproducable builds [22:04] <Mavrik> also I did see backported patches there occasionally ;) [22:04] <JEEB> yes, at times they do get backported [22:04] <JEEB> and you can have reproduce'able ones from master too [22:04] <JEEB> just note the hash [22:04] <JEEB> :) [22:04] <bryno> eheh, thanks :D [22:05] <JEEB> anyways, it's better than it was but it's still somewhat of a mess with releases [22:05] <Mavrik> JEEB, meh [22:05] <Mavrik> I link against a certain version of API so releases help there :) [22:05] <Mavrik> don't have to manually check which hash is the "newest" for an API [22:05] <JEEB> :) [22:06] <Mavrik> but yeah, I'm crazy enough to run production software on OSS :P [22:07] <JEEB> oh that is nothing new :) [22:48] <emulator> when remuxing how can I set the frame rate to the same as when it was decode? Trying to get the example transcoding to give me 1/2 the frames of the original. [22:48] <emulator> *transcoding.c [22:50] <BtbN> you can't do that without re-encoding. [22:51] <BtbN> Or what do you mean? [22:53] <emulator> well, what I mean... is... I want to slow down the frames... and the example actually returns it at 59.94 when originally it was 29.97. I want to be able to do the opposite [22:53] <emulator> I believe it is re-encoding. [22:54] <emulator> It's not a script... It is the example in trunk [22:55] <emulator> I can paste it if you want. [22:55] <emulator> This file: https://www.ffmpeg.org/doxygen/trunk/transcoding_8c_source.html [22:57] <c_14> Do you want to slow down the video or just cut the framerate? [22:58] <emulator> Eventually I would like to know how to do both... [22:59] <c_14> init and use the fps/setpts filters [23:00] <emulator> uhhhh.... I don't know how/what that is... LOL... hold on let me hit the api. [23:02] <emulator> That would be this call? av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx)) [23:05] <c_14> Think so [23:06] <emulator> ok... thanks. [23:06] <emulator> I will look into that later... I goota go. Thanks for everything. [23:23] <Suchiman> applying hqdn3d or not to apply, that is the question [23:30] <rivarun> hello. i have a 30fps video that i'd like to play at half the speed. i've seen some samples of people who upconverted videos to 60fps (and i believe some frame smoothing/generation was done too) to get a smoother animation. is there a way to get a similar result with ffmpeg? [23:31] <rivarun> interpolation is probably the word i'm looking for [00:00] --- Thu Sep 11 2014

1 0