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September 2013

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58 discussions

ffmpeg-devel.log.20130929
by burek 30 Sep '13

30 Sep '13

[00:00] <ubitux> j-b: if you need the video dumps of this course, let me know [00:00] <ubitux> but only if you want them, because :effort: [00:04] <j-b> I'll add the course [00:05] <pippin> j-b: one mostly tend to come up with ad-hoc categorizations that makes sense in the context where one is using the filter :] [00:05] <j-b> pippin: exactly my issue. [00:07] <pippin> you can split according to technical signal processing criteria; and amount of dimensions taken into account - but that would often be less useful for actual use. [00:12] <j-b> ubitux: btw, what is the format of subtitles in wtv files? [00:12] <funman> .docx [00:12] <j-b> I wish [00:12] <ubitux> no idea [00:12] <ubitux> i'm not sure i want to know though [00:13] <ubitux> i see dvb subtitles code [00:13] <ubitux> [~/src/ffmpeg]- grep -i 'codec.*subtitle' libavformat/wtvdec.c st->codec->codec_id = AV_CODEC_ID_DVB_SUBTITLE; [00:13] <ubitux> only see this ^ [00:23] <ubitux> ok, reconstructing properly dvdsub, done. [00:29] <funman> ubitux: seems to be dvbs indeed, but some samples seem to use other formats [00:30] <kierank> "pid" based closed captions sometimes [00:30] <ubitux> funman: dvb teletext then probably [00:31] <funman> AV_CODEC_ID_DVB_TELETEXT [00:31] <funman> dunno if our teletext decoder handles that [00:31] <ubitux> st->codec->codec_id = !ff_guidcmp(subtype, mediasubtype_teletext) ? AV_CODEC_ID_DVB_TELETEXT : AV_CODEC_ID_EIA_608; [00:36] <ubitux> ok so, still on vobsub story [00:36] <ubitux> it seems the cases with timestamps going backward are related to those splitted spu packets [00:37] <ubitux> i think i finally figured out most of the insanity out here [01:39] <BBB> michaelni: works in my emulator (?) very weird [01:40] <michaelni> BBB "make -j12 `make fate-list | grep vp8`" crashed with 1 or 2 randomly here each time [01:43] <michaelni> do you have the patches in some branch somewhere so i can retest the exact same code you tested? [01:45] <BBB> let me try... [01:45] <BBB> https://github.com/rbultje/ffmpeg/tree/vp9 [01:46] <BBB> yeah it doesn't fail here [01:46] <BBB> not sure what's going on [01:53] <michaelni> seems passing here [01:57] <BBB> so why does the patch fail but the tree pass? [01:58] <BBB> should be identical code, minus the vp9 part in it [01:59] <michaelni> i dunno, does it pass for you if you rebase the relevant patch(es) on master ? [02:00] <BBB> let me try [02:02] <BBB> trying now [02:02] <BBB> takes a few minutes [02:06] <michaelni> ive restested the local branch that crashed before, its passing now here too [02:06] <BBB> hm... [02:06] <BBB> odd [02:07] <BBB> it passes here too fwiw [02:07] <BBB> weird [02:12] <michaelni> no crash even after multiple runs [02:18] <michaelni> i guess in absence of better ideas, that means git push [02:20] <michaelni> if you want me to push your rebased version instead of what i have (probably a good idea as mine was affected by that weird crash or it was PEBCAK issue) ... [02:23] <BBB> michaelni: it merged without any offset issues here... let me rebase and make sure [02:23] <BBB> I don't exactly have the latest tree, and I think you inverted some patch order here or there... [02:26] <michaelni> I tried to apply patches ordered so that intermediates are fully working [02:27] <michaelni> also i had a local hotfix for some h264 arm code that failed under qemu but not on real hw apparently but that shouldnt have affected the vp8 tests [02:27] <michaelni> "hotfix" == disabling it [02:28] <BBB> right no that shouldn't matter [02:29] <BBB> let me try to rebase [02:29] <BBB> so the patches are the armneon one and the emu-edge argument change one right? [02:29] <BBB> and then after that ffvp9 [02:32] <michaelni> i only tested emuedge + neon fix [02:34] <BBB> is any of the invidivual files different from my tree? [02:34] <BBB> d<->v [02:37] <BBB> michaelni: https://github.com/rbultje/ffmpeg rebased on latest head (both patches) [02:37] <BBB> michaelni: seems to work for me [02:38] <BBB> (for these 3 files on armneon/emulator) [02:38] <BBB> michaelni: once they are in, I'll rebase ffvp9 and send a new version (if any changes are needed) [02:41] <michaelni> seems the only difference to my tree is one newline and the hotfix [02:43] <BBB> weird [02:43] <BBB> well I'd just push it then [03:11] <cone-323> ffmpeg.git 03Ronald S. Bultje 07master:f574b4da567c: vp8: use 2 registers for dst_stride and src_stride in neon bilin filter. [03:11] <cone-323> ffmpeg.git 03Ronald S. Bultje 07master:face578d56c2: Rewrite emu_edge functions to have separate src/dst_stride arguments. [03:58] <BBB> michaelni: ok, new ffvp9 patch (just with a merge issue in changelog) in my tree on github, branch master [03:58] <BBB> michaelni: I can provide the fate test files if wanted [03:58] <BBB> unless you're hoping for more review [04:14] <michaelni> please provide he fate test files, ill upload them tomorrow (need sleep now ...) [10:08] <ubitux> ok the insanity continues [10:09] <ubitux> it seems the ts in the split packet are actually used to reorder them [10:49] <ubitux> actually no it seems i have to read them by "stream order" [10:49] <ubitux> and ignore ts [11:37] <cone-589> ffmpeg.git 03Carl Eugen Hoyos 07master:d87ff555025e: Remove an incorrect and unneeded assert in the amr demuxer. [12:17] <michaelni> ubitux, do you have further comments on "avcodec/tiff_common: add ff_tadd_bytes_metadata" ? should i change something or ok to push as is ? [12:18] <ubitux> michaelni: no it's fine with me, no further comment [12:18] <michaelni> ok thx [12:34] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:7859e89ff0f9: avcodec/tiff_common: add ff_tadd_bytes_metadata() [12:34] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:e40c96513efb: avcodec/exif: print bytes and undefined types as scalar lists [12:34] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:5c4fc51d9b9f: Revert "fate: Force diff into text mode" [13:09] <BBB> ubitux: can you upload the vp9 fate files to the server? [13:10] <ubitux> i don't think i have write permissions [13:10] <BBB> oh [13:10] <ubitux> i usually ask michael [13:10] <ubitux> less work for me too ;) [13:10] <BBB> ok, I'll send them to him [13:17] <BBB> done [13:17] <BBB> kids-time now [13:20] <BBB> ubitux: how's assmebly going btw? [13:20] <ubitux> sorry didn't have time to continue [13:20] <ubitux> i'm fighting with dvd & dvb subs on my free time... [13:20] <ubitux> it's not fun but i don't like those bugs [13:27] <BBB> doesn't ffmpeg have an infinite number of bugs, just like any other software project? [13:27] <BBB> maybe I'm becoming too corporate :) [13:28] <ubitux> at least one of those is because of me [13:28] <ubitux> so it's in priority [13:32] <Compnn> if you have urls to the samples i can put them on sample repo [13:32] <Compnn> :P [13:33] <luc4> Hi! I wrote this function: http://paste.kde.org/pb44c1418/. It implements a pixel format conversion from yuyv to argb manually or using ffmpeg. What I noticed, and surprised me a little is that the manual conversion is considerably faster than the ffmpeg algorithm. This is pretty weird. Any idea why? Maybe I'm using ffmpeg wrong? Maybe there is a faster algorithm than SWS_FAST_BILINEAR? [13:43] <wm4_> luc4: I've heard more than once that swscale is slower than it should be, but it also could be that swscale is simply doing more work (as in better quality?) than yours, I'd wait to hear from michaelni [13:44] <wm4_> also, wasn't SWS_FAST_BILINEAR broken? [13:45] <luc4> wm4_: sure, that is why I'm asking if I'm using it wrong, or if there is any way to make it faster and explain why the manual computation is faster. [13:46] <luc4> wm4_: also, is that algorithm actually used to convert the pixel format when there is no resize? [13:48] <michaelni> luc4, i dont think sws has a specific routine to do packed yuyv to argb so it converts yuyv first to planar yuv (yuyv isnt very common for video decoders ...) [13:49] <michaelni> if thats the issue adding such unscaled special case shouldnt be hard, if thats a common convertion path for some use case [13:51] <wm4_> michaelni: apple decoders output yuyv, apparently [13:51] <luc4> michaelni: I see, so ffmpeg is inefficient in this case. I'll keep my implementation then. Ok, no problem. Maybe this better path may be added to ffmpeg? I'm using this cause usb cameras tipically use yuv. [13:52] <pigoz> well, vda does. probably other stuff in coremedia does as well. [13:52] <pigoz> (output yuyv) [13:53] <michaelni> also calling sws_getContext() on each frame is not optimal [13:53] <luc4> michaelni: it is static. [13:53] <michaelni> right, sorry, missed that [13:53] <wm4_> michaelni: I have some code that needs to convert images of varying sizes, what to do in that case? [13:55] <michaelni> maybe sws_getCachedContext() could be used in that case [13:56] <wm4_> I mean, the image size can change on every call (I believe sws_getCachedContext would merely detect changes, and then fully reinit) [13:56] <michaelni> yes [13:57] <michaelni> it reinits only when needed instead of always [14:00] <luc4> michaelni: what about adding this to ffmpeg? Might be useful... [14:01] <michaelni> yep thats what i meant by ".... adding such unscaled special case shouldnt be hard, ...." [14:04] <luc4> michaelni: can you point me to where I should add this? [14:05] <michaelni> mainly libswscale/yuv2rgb.c [14:06] <michaelni> it will also need a 1 line change where ff_yuv2rgb_get_func_ptr() is called so its used for yuyv422 [14:09] <luc4> michaelni: thanks! [14:13] <JEEB> luc4, it was a good idea to poke this channel, right? ;) [14:13] <luc4> JEEB: right :-) [14:17] <wm4_> michaelni: it's not really clear what settings sws_getCachedContext checks... for example it mentions that it doesn't check srcFilter and dstFilter; so how much equivalent is it compared to sws_init_context() on arbitrary parameter changes, even when done with av_opt_set_...? [14:17] <luc4> michaelni: what about yuv420 to rgb conversion? Is that taking a long path as well? [14:17] <wm4_> it's also not very orthogonal with sws_init_context() [14:20] <michaelni> luc4, yuv420 (& planar yuv422) use the yuv2rgb code [14:24] <michaelni> wm4_, it checks what you give it as arguments [14:24] <michaelni> that is arguments mismatch context -> reinit [14:25] <wm4_> ok so I could add a check in my code whether only the dimensions change, and then use that function to "update" to sws context [14:28] <wm4_> ideally similar code should go into libswscale directly, but SwsContext really has lots of members... [14:28] <BBB> Compn: I can fw you the email I sent to michael that explains where to get the saples, what's your email? [14:31] <wm4_> michaelni: does sws_getCachedContext really need to free the previous context? I believe sws_init_context() can now do a full reinit [14:31] <wm4_> or is that not the case? [14:31] <michaelni> maybe [14:32] <wm4_> because then one could add a sws_init_cached_context(SwsContext *sws); function [14:32] <luc4> michaelni: ok, that code is pretty complex, maybe I could add a wish to the tracker? [14:32] <wm4_> which internally checks for changes since last reinit [14:34] <wm4_> hm doing that would probably work, but leak memory [14:38] <michaelni> luc4, you certainly can, but code doesnt implement itself so i do prefer patches over wishes :) [14:40] <luc4> michaelni: I understand of course, but I'm not an expert in pixel format conversion. The code is not completely clear to me. [16:08] <wm4_> michaelni: AVFrame currently doesn't carry chroma position info? [16:10] <Daemon404> i dont think anything in ffmpeg gives a crap about it? [16:10] <wm4_> libswscale supports it [16:15] <Daemon404> interesting [16:29] <wm4_> #define SWS_PARAM_DEFAULT 123456 [16:29] <wm4_> sorry what [17:09] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:a0fec71ebda1: avformat/mxfenc: instance number is just 24 bits [17:09] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:28d35dbe9f57: avformat/mpegenc: fix muxrates AVOption maximum value [17:09] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:8c1253c9349c: avformat/mpegenc: check, warn and clip VBV value to within syntactically possible values [17:09] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:f2eb474a5a3d: avformat/mpegenc: check, warn and clip muxrate to syntactical possible values [17:25] <michaelni> BBB, vp9 samples added to fate rsync [17:54] <BBB> michaelni: ty; anything else needed for ffvp9 review? [17:55] <BBB> j-b: you never explained why I received your hug [17:57] <michaelni> BBB: not sure, i think it would be best if ubitux and you review each others code, as you know the code already [17:57] <BBB> ubitux: ok with you? [17:57] <j-b> BBB: for VDD money :) [17:57] <BBB> (I already reviewed ubitux' patches as they were going in) [17:57] <BBB> j-b: ah, you're welcome [17:57] <j-b> BBB: I owe you a beer [17:57] <j-b> or 2 :) [17:58] <j-b> hopefully in NYC [18:00] <BBB> I like beer [18:00] <BBB> sounds good [18:00] <BBB> I imagine vdd was fun? [18:02] <j-b> BBB: I think so. People liked it [18:02] <Daemon404> j-b in NYC??? [18:02] <Daemon404> when is this [18:03] <j-b> Daemon404: I hope early next year [18:03] <Daemon404> j-b, if you let me know when [18:03] <Daemon404> i can use it a an excuse to work from our NYC office [18:03] <Daemon404> for a little while [18:03] <j-b> I would love to live there, actually [18:03] <Daemon404> <.< >>> [18:03] <Daemon404> i love nyc [18:03] <Daemon404> but my SO hates it... [18:03] <Daemon404> so ;) [18:04] <j-b> change SO :) [18:04] <Daemon404> lol [18:04] <Daemon404> well also, the house centipedes in NYC horrify me [18:04] <j-b> ;-) [18:04] <Daemon404> i had them in my manhattan appartment [18:05] <j-b> there are more than inhabitants in NYC? [18:05] <Daemon404> http://en.wikipedia.org/wiki/Scutigera_coleoptrata [18:05] <Daemon404> japanese call 'em gejigeji [18:07] <BBB> house centipedes? [18:07] <BBB> my god where did you live [18:08] <BBB> you know there's acceptable high-end housing (ad affordable) right? [18:08] <BBB> s/ad/and/ [18:08] <Daemon404> Chelsea [18:08] <Daemon404> 19th and 9th [18:08] <BBB> you guys are so willing to accept all horrors and pains just because you don't know any better [18:08] <Daemon404> also because i was on intern salary [18:08] <BBB> go out, see the world, visit 1000 open homes - you'll see you can get better [18:08] <BBB> or if you don't know better, hire someone to explain it to you [18:08] <Daemon404> and i stupidly chose to live in manhatta [18:08] <Daemon404> n [18:08] <BBB> you're not an intern any more [18:08] <BBB> so that's a bad excuse [18:09] <Daemon404> i would prefer brooklyn [18:09] <BBB> j-b: come live in ny, it's awesome [18:09] <Daemon404> somewhere liek park slope [18:09] <BBB> brooklyn is nice also yes [18:09] <Daemon404> or maybe astoria [18:09] <BBB> park slope is expensive though [18:09] <j-b> BBB: I'd like to. [18:09] <BBB> but around there is nice yes [18:09] <Daemon404> BBB, i actually prefer to have a nice largr house [18:09] <BBB> park slope is more for if you have 4-5 kids in school-age (for public schools) [18:10] <Daemon404> 2 floor + basement [18:10] <Daemon404> cant get that in NYC so easil [18:10] <Daemon404> y [18:10] <BBB> Daemon404: oh, then try queens or long island :) [18:10] <Daemon404> heh [18:11] <Daemon404> im actually planning to live in the english countryside [18:11] <Daemon404> (London is a craphole) [18:13] <kierank> only where you live [18:13] <Daemon404> perhaps [18:13] <Daemon404> i do prefer a more rural area [18:14] <Daemon404> and a nice large house [18:14] <Daemon404> suitable for kids [18:14] <BBB> I get the feeling daemon404 is misinformed about city-life because he doesn't know how to "get" it [18:14] <BBB> anyway [18:14] <BBB> battery park and park slope is good for kids [18:15] <Daemon404> heh [18:15] <Daemon404> i lived in park slope for a bit [18:15] <kierank> lol kids [18:15] <Daemon404> i didnt see any house id consider large enoug [18:15] <BBB> I saw doutzen kroes yesterday in battery park [18:16] <BBB> also can someone start reviewing ffvp9 for ubitux/me? [18:16] <BBB> kierank: maybe you? [18:16] <kierank> ok [18:16] <BBB> if you think it looks roughly ok, that's fine, just say that as a review [18:17] <Daemon404> i assume youve done soem minimal fuzz testing? [18:17] <BBB> then at least we have some outside eyes (i.e. not ubitux or me) giving it a look [18:17] <BBB> I'm doing ongoing fuzz testing, along with performance increments [18:17] <BBB> but neither is finished yet [18:18] <Daemon404> ive peeked at it a view times [18:18] <Daemon404> but always decided i didnt have time at the moment to go over 10kloc [18:20] Action: kierank watches gmail struggle with that email size [18:31] <kierank> heh and chrome crashes [18:36] <BBB> kierank: hehe :) use a text editor that has a "save" function ;) [18:37] <kierank> i'm just going to copy and paste stuff [18:38] <Daemon404> BBB, looking at it now [18:38] <Daemon404> give me a few hrs [18:39] <Daemon404> also it made my mail client hang for a minute when hitting reply [18:39] <Daemon404> :D [18:39] <Daemon404> j-b, ping me when you plan to head to NYC perhaps. [18:39] <j-b> sure [18:40] <j-b> if secret plan works, April [18:40] <j-b> if it fails, before :) [18:40] <Daemon404> oic [18:40] <Daemon404> im in Canada for easter [18:40] <Daemon404> so before/after works well [18:42] <kierank> bit of a diego review but meh [18:44] <Daemon404> uint8_t coef[4][2][2][6][6][11] [18:44] <Daemon404> sexy [18:45] <durandal_1707> what about adding structs? [18:46] <j-b> VP9 \o/ [18:49] <Daemon404> yeah im trying to review it right now [18:49] <Daemon404> a few technical issues so far... [18:49] <Daemon404> gonna take me a while [18:56] <durandal_1707> was decoder fuzzed at all? [19:01] <BBB> durandal_1707: a little, not very extensively - it's being worked on (along with performance) [19:02] <BBB> Daemon404: btw all of these dimensions are used, so yes you really need them all :) [19:02] <BBB> and thanks for review [19:13] Action: Daemon404 saves a draft and gets food [19:32] <wm4_> michaelni: vf_scale calls sws_setColorspaceDetails after sws_init_context, is that valid? [19:49] <durandal_1707> wouldn't sws_init_context overwrite what other one set? [19:50] <wm4_> no [19:50] <wm4_> but it does something with it [19:50] <wm4_> (at least handling of special J* formats) [20:19] <cone-589> ffmpeg.git 03Paul B Mahol 07master:5a47f12390d2: doc/demuxers: fix image2 examples [20:37] <michaelni> wm4_, if it works and the docs dont say its not allowed then we probably have to consider it valid as other applications could use it that way too [20:38] <wm4_> michaelni: I think it at least undoes the J* fallback [20:38] <wm4_> c->srcRange |= handle_jpeg(&c->srcFormat); [20:38] <wm4_> and this is the only place handle_jpeg is called [20:38] <wm4_> sws_setColorspaceDetails doesn't do this [20:39] <wm4_> although there's probably only one case where srcRange is read again after that point [20:41] <wm4_> hm or two cases [20:45] <michaelni> i guess that makes it invalid (in at least some corner cases) [21:08] <ubitux> wm4_: ok i think i got it :) [21:08] <wm4_> yay [21:10] <ubitux> i need to cleanup my tree but it should be done in an hour or two [21:26] <cone-589> ffmpeg.git 03Paul B Mahol 07master:61274193e3bf: avfilter/vf_rotate: use vsub for output plane height calculation [21:32] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:4be0c6ed20f2: avcodec/tiff_common: fix bprint max sizes [21:34] <ubitux> michaelni: automatic doesn't require finalize/free, but unlimited does, iirc [21:35] <ubitux> michaelni: also you failed one chunk [21:37] <michaelni> they all already had finalize [21:37] <michaelni> unless i missed one [21:37] <ubitux> ok [21:37] <ubitux> - av_bprint_init(&bp, 10 * count, AV_BPRINT_SIZE_AUTOMATIC); [21:37] <ubitux> + av_bprint_init(&bp, 10 * count, 100 * count); [21:37] <ubitux> please fix this one though [21:37] <ubitux> unless that's what you wanted? [21:38] <ubitux> :) [21:38] <BBB> michaelni: new patch will have one additional vp9 conformance sample [21:38] <BBB> michaelni: so you know to update the fate ref files [21:39] <michaelni> ubitux, i didnt think much about it but the double one doesnt use unlimited because i wanted to prevent really huge numbers from causing huge alloc and strings and metadata [21:39] <michaelni> i can change it to unlimited if preferred [21:40] <ubitux> no it's ok [21:40] <ubitux> i thought it was a typo [21:40] <ubitux> but yeah false alert, forget it [21:40] <michaelni> BBB, do you have a url i can wget ? [21:42] <BBB> vp90-2-03-deltaq.webm [21:42] <BBB> same directory as the other ref files [21:44] <michaelni> BBB uploaded [21:45] <BBB> ty [21:51] <durandal_1707> ubitux, saste: you ok with my dualinput simplification? [21:51] <ubitux> i merely moved code with dualinput [21:52] <ubitux> i didn't have much insight of it [21:52] <ubitux> i'll make some review tonight when i'm done with vobsub [21:52] <ubitux> hopefully in not much time [21:59] <kierank> Paranoialmaniac: what is the point of avcc3 [21:59] <kierank> why can't they just write sps/pps in the video bitstream [22:01] <Paranoialmaniac> kierank: ? avc3 allows conveying parameter sets in-bandly and putting filler data [22:01] <kierank> why do they need the avc3 element? [22:01] <Paranoialmaniac> avc3 is not an element. it's an identifier. [22:01] <kierank> oh [22:02] <Paranoialmaniac> like CODEC FourCC in AVI [22:04] <Paranoialmaniac> avc3 uses the same decoder configuration avcC with avc1 [22:07] <kierank> oh i thought it was an avcc3 element [22:07] <kierank> avcC3 [22:07] <kierank> Paranoialmaniac: what is avc2? [22:08] <Paranoialmaniac> avc2 indicates aggregator and extractor nalu units may be present in the file [22:08] <kierank> ah some scalable thing then [22:09] <Paranoialmaniac> i think it is easy to add support of avc3 in libavcodec [22:09] <Paranoialmaniac> maybe, just adding the identifier [22:14] <Paranoialmaniac> libavcodec already can decode files conveying parameter sets in-bandly even if the stream is signaled as avc1. i had tested a file generated with mp4box which is out of spec obviously [22:21] <ubitux> wm4_: ping [22:22] <ubitux> wm4_: could you try this for me https://github.com/ubitux/FFmpeg/commits/vobsub ? [22:22] <ubitux> it fixes all the issue i have with my samples [22:22] <ubitux> so it will "probably" fix yours [22:42] <cone-589> ffmpeg.git 03Martin Storsjö 07master:9277050e2918: vqf: Make sure sample_rate is set to a valid value [22:42] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:554c22c71091: Merge commit '9277050e2918e0a0df9689721a188a604d886616' [22:48] <cone-589> ffmpeg.git 03Martin Storsjö 07master:68ff9981283a: vqf: Make sure the bitrate is in the valid range [22:48] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:b24f7cf95070: Merge commit '68ff9981283a56c731f00c2ee7901103665092fc' [22:58] <cone-589> ffmpeg.git 03Martin Storsjö 07master:8d07258bb606: avidec: Make sure a packet is large enough before reading its data [22:58] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:3a3ebc7c7dcc: Merge commit '8d07258bb6063d0780ce2d39443d6dc6d8eedc5a' [23:03] <wm4_> ubitux: trying [23:03] <cone-589> ffmpeg.git 03Martin Storsjö 07master:adc09136a4a6: xwma: Avoid division by zero [23:03] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:17049bff64f2: Merge commit 'adc09136a4a63b152630abeacb22c56541eacf60' [23:10] <wm4_> ubitux: appears to work! I see subtitles, and no more length 0 packets [23:10] <ubitux> cool :) [23:10] <ubitux> i'll submit the patchset in a moment, finishing the long explanation [23:21] <cone-589> ffmpeg.git 03Martin Storsjö 07master:59480abce7e4: alac: Do bounds checking of lpc_order read from the bitstream [23:21] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:9e58d46edf95: Merge commit '59480abce7e4238e22b3a4a904a9fe6abf4e4188' [23:29] <ubitux> michaelni: so ok to drop the pascal/delphi faq entry? [23:29] <ubitux> wm4_: patch on ml, hopefully it will be pushed soon [23:29] <cone-589> ffmpeg.git 03Martin Storsjö 07master:28ff439efd23: mvi: Add sanity checking for the audio frame size [23:29] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:2f8418b584d2: Merge commit '28ff439efd2362fb21e1a78610737f2e26a72d8f' [23:29] <ubitux> wm4_: for the backport, i'll wait a bit [23:29] <michaelni> ubitux, yes [23:29] <ubitux> michaelni: k thx [23:32] <ubitux> dvdsub fixed, now i need to figure out the problem with dvb ones [23:32] <ubitux> :'( [23:33] <cone-589> ffmpeg.git 03Clément BSsch 07master:e6273592e442: doc/faq: rework r_frame_rate entry. [23:33] <cone-589> ffmpeg.git 03Clément BSsch 07master:eafb9c52e80c: doc/faq: remove Delphi/Pascal entry. [23:33] <cone-589> ffmpeg.git 03Clément BSsch 07master:2e798c6ce0b4: doc/faq: remove indent in examples. [23:33] <cone-589> ffmpeg.git 03Clément BSsch 07master:565fd3cc1a17: doc/faq: add @command{} in the tool debug entry. [23:37] <cone-589> ffmpeg.git 03Clément BSsch 07master:a89e9ad1c862: swr: fix example code and doxy indent. [23:45] <cone-589> ffmpeg.git 03Martin Storsjö 07master:d07aa3f02b73: riffdec: Add sanity checks for the sample rate [23:45] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:5badcfe2200c: Merge commit 'd07aa3f02b73ab1371c13ac7898338380ca0932b' [23:59] <cone-589> ffmpeg.git 03Martin Storsjö 07master:a9221e39600a: electronicarts: Add more sanity checking for the number of channels [23:59] <cone-589> ffmpeg.git 03Michael Niedermayer 07master:7057cc861d10: Merge commit 'a9221e39600a31ee13e736e9e47743cde23f0280' [00:00] --- Mon Sep 30 2013

1 0

ffmpeg.log.20130929
by burek 30 Sep '13

30 Sep '13

[01:24] <smjd> are you sure FLV doesn't support that? [01:26] <smjd> or is there also a codec called FLV? [01:28] <sacarasc> There is a codec called FLV. [01:31] <smjd> is it Sorenson Spark? [06:48] <elBradford> Does anyone have experience cross compiling ffmpeg with librtmp? [06:50] <elBradford> Everything seems to be going well, however I get "undefined reference" errors in librtmp.o [06:50] <elBradford> Then it stops going well, obviously. [07:37] <taqattack> is there a way command to reduce latency / control packet size in ffmpeg for streaming? [07:45] <EvanDotPro> i am pushing a tcp h264 stream from ffmpeg to a socket server and parsing the NAL units... if i stream it over 127.0.0.1, the NAL units are ~20k-30k bytes each, and the video decodes fine on the other end, but if i try to go over my router / modem where the MTU size is affected, the NAL units are tiny, like 1400-1500 bytes, and the video doesn't seem to want to decode. [07:47] <EvanDotPro> piping to netcat and then over 127.0.0.1 seems to cause the problem too, except that the NAL units seem to be 8191 bytes in that case [08:26] <skyroveRR> Hello, I'd like to display metadata for an audio/video file using ffmpeg, I'm using slackware, I tried http://stackoverflow.com/questions/9464617/retrieving-and-saving-media-meta… but haven't been successful yet, the metadata.txt file is empty. [08:28] <skyroveRR> I'd like to also edit and view the title, comments and other metadata on a particular media file, so what's the command for that? [08:41] <skyroveRR> Hello, I'd like to display metadata for an audio/video file using ffmpeg, I'm using slackware, I tried http://stackoverflow.com/questions/9464617/retrieving-and-saving-media-meta… but haven't been successful yet, the metadata.txt file is empty, and I also want to edit the metadata itself, any ideas? [09:28] <elBradford> Has anyone compiled ffmpeg with librtmp for Android? Looking to pay for help at this point, kind of desperate. [10:22] <BlackBishop> relaxed, burek, sacarasc: mhmm .. same .. tried -preset ultrafast, -vcodec flv | libx264 .. none get past 13 fps :| weird .. [10:27] <BlackBishop> Mhmm, I do have a raspberry pi that is said to have an encoder for mkv/wmv not just decoder .. [10:31] <BlackBishop> actually .. vc1/mpeg2 .. still, good enough .. [11:11] <kms_> there are any key for realtime encoding on low CPU with libtheora? [11:21] <IamTrying> http://paste.fedoraproject.org/42905/38044632/ - why the is giving "could not decode stream" ?? [12:10] <luc4> Hi! I wrote this function: http://paste.kde.org/pb44c1418/. It implements a pixel format conversion from yuyv to argb manually or using ffmpeg. What I noticed, and surprised me a little is that the manual conversion is considerably faster than the ffmpeg algorithm. This is pretty weird. Any idea why? Maybe I'm using ffmpeg wrong? Maybe there is a faster algorithm than SWS_FAST_BILINEAR? [13:17] <luc4> Anyone using ffmpeg for pixel format conversion? [13:18] <zap0> luc4. do you really care? or you have an actual question? [13:19] <JEEB> zap0, he has [13:19] <JEEB> see an hour before [13:19] <luc4> I already asked some hours ago, just trying again to see if someone new connected: Hi! I wrote this function: http://paste.kde.org/pb44c1418/. It implements a pixel format conversion from yuyv to argb manually or using ffmpeg. What I noticed, and surprised me a little is that the manual conversion is considerably faster than the ffmpeg algorithm. This is pretty weird. Any idea why? Maybe I'm using ffmpeg wrong? Maybe there is a faster algorithm than [13:19] <luc4> SWS_FAST_BILINEAR?. [13:19] <JEEB> also you should most probably ask about this on #ffmpeg-devel, although there could be various reasons [13:19] <JEEB> you could check the code in swscale [13:19] <JEEB> for this conversion path [13:20] <zap0> how is it ffmpeg being slower makes it wrong? isn't wrong a measure of the correctness of pixel calculations? [13:20] <JEEB> He is asking if it's possible he's using ffmpeg wrong [13:20] <JEEB> not that ffmpeg is wrong [13:20] <JEEB> not to mention that the compilation options generally don't tend to do too much optimization because that can "get in the way of hand-written assembly optimizations" so if you compile your own code with various optimizations enabled it can end up being faster [13:21] <luc4> Both seems to be correct from what I see. [13:21] <JEEB> I've not looked at that code path myself [13:21] <zap0> ffmpeg would be doing per pixel math. not some lookup table [13:22] <luc4> I can look at the code, but it is unlikely I'll get why it is slower in reasonable time. [13:23] <JEEB> hmm, a look-up table [13:23] <vlt> Hello. I ffmpeg'd an AC3 5.1 stream to wav and got a six channel file. Do you know which channel is which? [13:23] <JEEB> anyways, there can be various reasons [13:23] <JEEB> you're not getting any non-aligned data warnings, right? [13:24] <zap0> vlt, there is no OFFICIAL standard ordering. but variuos configs are "mostly" consistant. [13:25] <luc4> JEEB: I got a warning related to software conversion being run. But I'm not getting it at the moment. [13:25] <zap0> vlt, 6 channels can mean: channels of which there are 6. or it could mean 5.1 or it could mean quad+2-sides. [13:27] <zap0> vlt, if ffmeg was able to detect the 6 channels as meaning "5.1" then it will consistantly use it's 5.1 layout. [13:27] <vlt> zap0: Thsi is what `file` says: ATSC A/52 aka AC-3 aka Dolby Digital stream, 48 kHz,, complete main (CM) 3 front/2 rear, LFE on,, 640 kbit/s reserved Dolby Surround mode [13:28] <zap0> vlt, when playing it back, you will likely have to manually tell the playback device the meaning of this 6 channel file is "5.1" and in 90% of situations it should then work correctly. [13:29] <zap0> vlt, the original file or the WAV file you converted? [13:29] <vlt> zap0: orig [13:29] <zap0> ok, so ffmpeg should comprehend that as meaning 5.1 (3f,2r+LFE) [13:29] <zap0> it will then use 5.1 layout. [13:30] <zap0> but WAV doesn't store the meaning. so the playback device has to be told! [13:30] <zap0> the playback device is only going to see a WAV that has 6 channels. [13:31] <zap0> *TYPICALLY* due to market forces... many playback devices seeing a 6 chanel file will presume 5.1 orientation [13:31] <vlt> zap0: I converted it to that 6 ch wav to be able to edit the audio (in ardour). And now I don't know which channel is which. [13:31] <zap0> vlt, i *i know* sucks, don't it ?!? ;) [13:32] <vlt> zap0: ? [13:32] <zap0> get one of the 5.1 sample AAC files with human talk on each channel. convert it, open it in your app. then listen back, you'll hear what is what. [13:33] <vlt> zap0: Good idea. Thank you. [13:34] <luc4> JEEB: trying in the devel channel. Thans for your advice. [14:46] <funyun> hi. is there anyway to add an image to a movie with ffmpeg? so there's like a thumbnail when it's in itunes? [14:48] <sacarasc> Does iTunes not just take a frame of the movie to use as a thumbnail? [14:49] <funyun> sacarasc: itunes/other apps show a thumbnail if someone has added one. if not it just shows a frame. i'm just wondering if i can add one with ffmpeg [14:49] <sacarasc> I do not know much about meta data adding with ffmpeg... [14:49] <sacarasc> You probably could with MP4box, though. [14:50] <funyun> sacarasc: okay, and i should look under meta data? [14:50] <sacarasc> I would assume so. It would be like album art... [14:51] <funyun> sacarasc: thanks [16:58] <burek> !pastebin BlackBishop [18:32] <zap0> if i generate some 16bit PCM audio, will it be bigendian or little endian? [19:40] <DorjePy> what is the easiest, accurate way learn filters [19:46] <klaxa> read sourcecode? :x [19:51] <BlackBishop> burek: ? [19:52] <durandal_1707> klaxa: why? [19:52] <klaxa> durandal_1707: it's probably the most accurate way to learn how filters work [19:52] <klaxa> unless that's not what he wanted and we misunderstood each other [19:54] <durandal_1707> klaxa: what he wanted? [19:54] <klaxa> <DorjePy> what is the easiest, accurate way learn filters [19:54] <durandal_1707> that could mean how to learn to use filters [19:55] <klaxa> the question is asked rather vaguely [19:55] <durandal_1707> why would i waste time writing filter documentation [20:18] <defaultro> is it possible to crop specific pixel on top and bottom, different size? [20:22] <ztane> hi y'all, I am trying to segment an h264+aac flv stream into individual h264+aac mp4 files, but only the first file works right; [20:23] <ztane> in the past there used to be reset_timestamps option, but it does not seem to be there anymore [20:27] <defaultro> this isn't working, -croptop 88 -cropbottom 88 [20:27] <defaultro> ok [20:29] <defaultro> http://pastebin.com/M2RH2KC3 [20:30] <defaultro> I have 2376x1584 video and I want to remove 187 pixels on top and 62 pixels at the bottom [20:32] <durandal_1707> defaultro: have you read last few lines? [20:32] <defaultro> yeah, use crop filter instead [20:32] <defaultro> isn't cropbottom the cropfilter? [20:33] <defaultro> foudn this, i hope it works - http://ubuntuforums.org/archive/index.php/t-1690664.html [20:35] <defaultro> it works but I am not sure if it's going to remove the pixels properly [20:35] <defaultro> :( it didn't work [20:35] <defaultro> i mean, cropping didn't work [20:36] <durandal_1707> defaultro: read documentation of crop filter [20:36] <defaultro> -vf crop=2376:1584:187:62 [20:36] <defaultro> that didn't do a thing [20:38] <ztane> is it possible to segment a h264/aac video with just vcodec copy, acodec copy? [20:38] <ztane> I still get ever increasing space in the beginning of each file, alas I have to segment the video using raspberry pi, so I cannot re-encode it [20:39] <durandal_1707> defaultro: you set out width and height to same value as input, and expect output to be different .... [20:39] <defaultro> i'm understanding it now :) [20:39] <defaultro> it's working, just need to calculate now [20:39] <defaultro> the first value are the new output dimension [20:40] <defaultro> "Width of the output video. It defaults to iw. This expression is evaluated only once during the filter configuration."his should have been well written as [20:40] <defaultro> Width of the new output video. It defaults to iw. This expression is evaluated only once during the filter configuration. [20:41] <ztane> ah works, I didn't have reset_timestamps 1 :"> [20:44] <defaultro> looks like the new crop filter is not as flexible as croptop and cropbottom. Now I can't specify a specifc height of pixels I want to remove on top and bottom. [20:45] <defaultro> you know what, maybe it did work, 2536:1334:187:62 :) [20:50] <durandal_1707> defaultro: really, you can use expressions to leave filter to calculate it for you [20:52] <defaultro> oh [20:52] <defaultro> that is cool [20:53] <defaultro> btw, when I use -qp 0, the video on youtube sometimes stops [21:13] <defaultro> durandal_1707: just finished reencoding it with -qp 0 and I'm currently uploading to youtube. I'm uploading a 2376x1334 video as compared to my other upload which is 1920x1080. Here is the old one - http://www.youtube.com/watch?v=NSJBYi8RdHs [21:14] <defaultro> brb, going to the gym :) [22:28] <EvanDotPro> can anyone point me to the place in where the x264 NAL unit size is determined based on the MTU or something? [22:37] <EvanDotPro> when i set the output to -f h264 tcp://127.0.0.1:1234 it makes the NAL unit size like 30k bytes on average. [22:38] <EvanDotPro> but when i make the output stdout and pipe to netcat it's like 8k bytes, then if i set it to another IP on my LAN or a remote server, it seems to adjust to the MTU and make them 1440 bytes [22:44] <EvanDotPro> also, i'm a little confused as to if slice-max-size is related to the NAL unit size or not [00:00] --- Mon Sep 30 2013

1 0

ffmpeg-devel.log.20130928
by burek 29 Sep '13

29 Sep '13

[10:57] <cone-168> ffmpeg.git 03Anton Khirnov 07master:38e15df1489d: avframe: note that linesize is not the usable data size [10:57] <cone-168> ffmpeg.git 03Michael Niedermayer 07master:945c7e399af2: Merge commit '38e15df1489d86c016515223ee693e7d0326c56a' [11:10] <cone-168> ffmpeg.git 03Anton Khirnov 07master:0767bfd1994c: lavfi: allow user-provided execute() callbacks [11:10] <cone-168> ffmpeg.git 03Michael Niedermayer 07master:7381d31f2267: Merge remote-tracking branch 'qatar/master' [11:35] <cone-168> ffmpeg.git 03Ronald S. Bultje 07master:c341f734e5f9: Convert multiplier for MV from int to ptrdiff_t. [13:05] <BBB> michaelni: will you also apply the emu_edge patch, or do I need to resend that? [13:06] <BBB> thanks btw :) almost ready for vp9 now [13:08] <michaelni> BBB i wanted after some more testing [13:26] <BBB> michaelni: ok thanks [13:30] <wm4> so we'll get hevc and vp9 at about the same time? interesting... [13:32] <Compn> wm4 : it'll be more interesting to see which one the anime people use [13:33] <Compn> probably depends how many bits each codec can max out [13:33] <wm4> why the anime people? [13:33] <wm4> because they like breaking everything all the time? [13:33] <Compn> because they often choose something that requires a lot of new code in mplayer :P [13:33] <Compn> ordered chapters, 10bit, etc [13:33] <Compn> ass subs [13:33] <durandal_1707> mplayer have none of that [13:34] <wm4> mplayer had 10 bit support before mplayer2 [13:34] <Compn> 10bit should work too... [13:34] <Compn> er i mean ass [13:34] <wm4> mplayer didn't fix most code assuming that 1 Y pixel is 1 byte, though [13:34] Action: Compn needs to wake up [13:38] <michaelni> BBB seems iam getting failures on arm (fate-vp8-test-vector-003 and others), ill double check that it works before [13:39] <michaelni> its mismatchig checksums no crash so far [13:56] <BBB> michaelni: vp8? [13:56] <BBB> michaelni: may be the same issue the c code had [13:56] <michaelni> seems 6 vp8 tests fail (no other failures) [13:57] <michaelni> vectors 3 4 and 7 [14:01] <BBB> that's bilinear again [14:01] <BBB> is that neon or armv6? [14:01] <BBB> it's neon, I see the code is using dst_stride for src again [14:02] <BBB> same issue C code had [14:02] <BBB> let me see if I can fix that... [14:33] <BBB> michaelni: http://pastebin.com/B9Z1gixE [14:55] <michaelni_> BBB seems segfaulting [15:02] <michaelni_> BBB http://pastebin.com/Js6bvK7L [15:14] <BBB> michaelni_: I have the idea that that disassembly isn't quite accurate, can you disassemble all of ff_put_vp8_bilin16_h_neon? [15:15] <BBB> the start offset is probably not at an instruction boundary [15:17] <BBB> oh I probably have to update the stack offsets [15:20] <BBB> michaelni_: http://pastebin.com/2NhTQRvL [15:52] <cone-323> ffmpeg.git 03Michael Bradshaw 07master:f46a3e3d6ec4: libopenjpeg: fix encoding of odd sized subsampled images [15:58] <michaelni_> BBB new patch crashes randomly, gdb doesnt provide very usefull baktrace (only ??) and disassembly says Cannot access memory at address 0xf7f0a70 [16:00] <michaelni_> ill retest with make distclean just to be sure [16:05] <michaelni_> still crashing after distclean [16:26] <BBB> michaelni_: pff... ok I'll check more [16:39] <cone-323> ffmpeg.git 03Paul B Mahol 07master:88c8ed18c972: avfilter/dualinput: remove unused variable [17:39] <cone-323> ffmpeg.git 03Paul B Mahol 07master:07da0854fe98: avformat/vfq: fix deadlock in vqf_read_header() [18:13] <michaelni_> burek, "<durandal_1707> michaelni: those FFmpeg forums nobody use except noobs (expecting help that never comes) so should be removed from contact page" [18:14] <burek> well, feel free to remove it, if you think it doesn't do any help [18:15] <burek> after all, it's a lot easier to criticize than to help, right? :) [18:15] <burek> (considering i didn't see durandal not once on that forum, helping other people) :) [18:16] <michaelni> iam not thining anything, just wanted to make sure you see it as its your forum [18:16] <michaelni> s/thining/thinking/ [18:16] <durandal_1707> burek: that response was caused by recent reply to thread in forum about -af nonexistent and that lavfi should be used .... [18:16] <burek> i didnt understand this last sentence [18:17] <burek> (@michaelni) [18:17] <durandal_1707> thread was from 2012 [18:17] <durandal_1707> and now there is -af [18:19] <michaelni> burek, i am not reading the forums so i cant say anything about them, i just saw durandal_1707 request and noticed you where not online so i just wanted to forward it [18:20] <burek> ok, thank you for that michaelni [18:20] <burek> durandal_1707 [18:20] <burek> if you really want to help people, you should join the forum and help them [18:20] <burek> i don't see any benefit from removing the link to the forum from the contact page [18:21] <durandal_1707> it was not request, it was more like request for status change, because i do not see anybody else except burek and llogan help, and they are not on forum 24h [18:21] <burek> if there was an official ffmpeg forum, i would be more than happy to close mine (and forward the database of posts to the new one, if needed) [18:22] <burek> durandal_1707, but how would removing the link from the contact page help anyone? [18:23] <ubitux> wm4: ok so, back on vobsub debugging: (one - major) problem is that mpeg pes hdr reader is being too smart and factorize the packets [18:24] <ubitux> so sometimes i get one too large packets, and i just can't actually split it to match the vobsub index expected packet size [18:25] <ubitux> i'm still trying to find a sexy way to deal with this [18:26] <wm4> maybe that's the reason mplayer's demux_mpg didn't work with ffmpeg's vobsub [18:37] <durandal_1707> see that fraps report, it take almost 10 days to reply [18:42] <durandal_1707> michaelni: from here: http://ffmpeg.gusari.org/viewtopic.php?f=16&t=1078 vlc can take up to 31 bits in fraps, while maximal supported looks to be 30 [19:08] <michaelni> increasing FF_HUFFMAN_BITS should solve this, best is probably to make it per codec variable as the increase could slow things down [19:09] <michaelni> and having it non constant for get_vlc() also would slow things down [19:12] <ubitux> wm4: btw, you use lavc decoder for dvdsub? [19:12] <wm4> ubitux: yes [19:12] <wm4> I still have the old decoder around... for old things [19:13] <wm4> like the current libav stable release [19:19] <ubitux> ok [19:22] <burek> <@durandal_1707> see that fraps report, it take almost 10 days to reply [19:22] <burek> wouldn't it be more helpful if you would join the forum and help the guy, answering that question [19:47] <durandal_1707> nope, it wouldn't be more helpful [19:47] <durandal_1707> michaelni: aren't you saying that to fix this one need to make things slower? [20:02] <burek> is there a text-only version of histogram filter? [20:03] <durandal_1707> what text only version would do? [20:04] <durandal_1707> short answer no [20:04] <burek> for example if there is a need to calculate the mean value of all the colors in the image [20:04] <burek> text version would help, giving the distribution in a parsable format, which can be used to calc the mean value [20:05] <durandal_1707> hmm, no [20:06] <burek> ok, thanks [20:21] <cone-323> ffmpeg.git 03Michael Niedermayer 07master:a63b479d1191: avformat/utils: factor find_decoder() out [20:44] <cone-323> ffmpeg.git 03Michael Niedermayer 07master:2b834a0605d9: avcodec/huffman: Allow specifying nb_bits to ff_huff_build_tree() [21:05] <burek> just to ask here too, in case people dont read #ffmpeg channel... is there any good page that explains the relation between -maxrate, -minrate and -bufsize, because there are several duplicate questions about it, where people in general lack the understanding of that relation and as a result use them in a wrong way [21:22] <burek> grr.. how to delete a page.. I accidentally put all the first letters in caps.. [21:30] <ubitux> oh fuck, they're split [21:31] <ubitux> why on earth those dvdsub are so complex [21:32] <wm4> because DVDs are a horrible thing [21:34] <ubitux> well [21:34] <ubitux> here is i have a subtitle chunk at a given ts [21:34] <ubitux> but it's not complete, and the rest of that chunk is happening at a LATER timestamp [21:35] <Daemon404> i dream that one day [21:35] <Daemon404> linux will flawlessly play a dvd from 1998 [21:35] <ubitux> and you can't start decoding the first one, since the control sequence is in the next packet [21:35] <ubitux> what. the. hell/ [21:38] <JEEB> that... makes no sense [21:38] <JEEB> are you supposed to just cache it until you get a control sequence? [21:39] <Daemon404> i had the dvd spec at one point [21:39] <Daemon404> lost it though... [21:39] <wm4> admit it, you burned it in dark satanic ritual [21:39] <wm4> +a [21:39] <ubitux> JEEB: yup [21:39] <Daemon404> wm4, it was a pdf [21:39] <ubitux> JEEB: i have no idea why the timestamps are diverging.. [21:39] <Daemon404> so no i didnt burn it... i think [21:40] <Daemon404> ;) [21:40] <wm4> Daemon404: I thought that didn't exist [21:40] <wm4> something about dvd specs only existing in printed form [21:40] <Daemon404> maybe [21:40] <Daemon404> i might be on crack. [21:40] <JEEB> huh [21:40] <JEEB> indeed [21:40] <JEEB> there were "PDFs" [21:40] <JEEB> which were shells that looked like PDFs [21:40] <JEEB> and then you had a binary plugin to adobe reader [21:41] <JEEB> that used a USB key to decrypt the book [21:41] <JEEB> this was only sold from 2007 to 2012 or so? I think [21:41] <ubitux> through an ie6 activex? [21:41] <Daemon404> so you could print screen it? [21:41] <burek> :facepalm: [21:42] <JEEB> http://www.dvdfllc.co.jp/pdf/Guidelines%20for%20the%20DVD%20Book%20Digital%… [21:42] <JEEB> also I love it how the logo of the DVD FLLC consists of seemingly Comic Sans MS [21:43] <Daemon404> 'easily portable' [21:43] <wm4> the fuck [21:43] <ubitux> Daemon404: usb keys are not heavy [21:43] <ubitux> ;)) [21:43] <Daemon404> lol [21:43] <JEEB> :D [21:44] <JEEB> "** IMPORTANT NOTICE ** The sale of DVD Book Digital Edition was terminated on July 31, 2012." [21:44] <JEEB> yeah, it was stopped in 2012 [21:45] <JEEB> http://www.dvdfllc.co.jp/format/f_sbsc.html [21:47] <ubitux> i'm wondering if i should reassemble within dvdsubdec [22:38] <wm4> oh man what's with those patches to force codecs [22:38] <wm4> why is this in libavformat and not in ffmpeg.c? [22:51] <michaelni> av_find_stream_info opens the decoders [22:59] <wm4> hm I see... I thought there was a separation between demuxers and decoders, but that's not really the case anymore [23:01] <michaelni> a sad necessity for all the real world formats [23:03] <michaelni> instead of the passing over AVFormatContext we also could use the AVDIct argument to av_find_stream_info() but it would nt work for streams added after the av_find_stream_info() call that is by the read_packet function used inside av_find_stream_info [23:06] <cone-323> ffmpeg.git 03Matthieu Bouron 07master:c461265a769e: lavf/mxfdec: add j2k codec to intra only codecs [23:43] <j-b> Is there a good doc around, describing categories of video filters (deinterlacers, smoothers, sharpeners, ...) ? [23:53] <ubitux> j-b: filters are not organized into categories, except "audio", "video" and "multimedia" (mixed) [23:54] <ubitux> and they're described here: http://ffmpeg.org/ffmpeg-filters.html [23:54] <j-b> no, I am looking for something wider than FFmpeg [23:54] <ubitux> oh, ok sorry [23:54] <j-b> a bit more theory, probably [23:54] <ubitux> there was a nice coursera course a while ago [23:55] <ubitux> but it was some specific filters though, nothing about interlacing for instance [23:55] <j-b> ubitux: the best I got was http://www.aquilinestudios.org/avsfilters/index.html [23:55] <j-b> but that's a bit poor [23:55] <ubitux> is it for you? [23:56] <j-b> not really [23:56] <j-b> I need to teach a bit to students [23:57] <ubitux> https://www.coursera.org/course/images this one was quite interesting [23:58] <ubitux> deinterlacing/telecining needs a dedicated course IMO [23:58] <ubitux> you can't easily introduce these filters between gamma correction and sharpening [00:00] --- Sun Sep 29 2013

1 0

ffmpeg.log.20130928
by burek 29 Sep '13

29 Sep '13

[00:37] <bffmpeg> skifreak: http://www.ffmpeg.org/download.html search for "static builds", see if you can find a suitable version that was built in the last day or so [00:42] <skifreak> bffmpeg: thanks for the help. In gentoo right now, running v2.0.1. Can easily installed the live ebuild in the package manager, or if you still think it's better, I can install the latest static build in http://ffmpeg.gusari.org/static/64bit/ [00:43] <bffmpeg> skifreak: things change fast here, so if you can quickly build a recent package, do so [00:44] <bffmpeg> if you remain stuck, the devs are very responsive if you can clearly articulate your issue - see http://ffmpeg.org/trac/ffmpeg [00:45] <skifreak> bffmpeg: will do, always glad to see fast progress. Compiling the live version of ffmpeg now [00:45] <skifreak> thanks, I'll definitely look into that. It's been a few months, but I've submitted bug reports before [00:47] <bffmpeg> skifreak: see also http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20quickly%20compile%20libaacplus [00:50] <skifreak> bffmpeg: thanks for all of the links and help. I currently don't have libaacplus installed (wonder if I should...I use ffmpeg primarily for compressing bluray material) [00:50] <JEEBsv> libaacplus shouldn't be recommended any more if it ever was, fdk-aac can do what it can, and more [00:51] <JEEBsv> and it is similarily "nonfree", so it's really in a very, very similar position, just that it can do it all better [00:51] <JEEBsv> (and more, libaacplus is HE-AAC(v2) only [00:51] <bffmpeg> ^^^^^ what jeebs sez [00:51] <skifreak> thanks guys! [00:51] <bffmpeg> jeebs == da shiz [00:52] Action: skifreak is glad he's able to speak to some pros [00:53] <skifreak> JEEBsv: thanks. Currently I'm not compressing bluray audio, usually it doesn't take up much space...though pcm tends to be a little large.... [00:53] <JEEBsv> I just saw libaacplus linked :P [00:53] <skifreak> insight is always appreciated [00:54] <skifreak> compiled the life version of ffmpeg (ffmpeg-9999) and same results as before [00:54] <skifreak> (with the right 3D bluray passthrough file) http://dpaste.com/1398445/ [00:56] <skifreak> should I submit a bug report then, or do either of you want to look at samples of the left (playable even with mplayer2) or right (errors out on analyzation) videos first? [01:02] <bffmpeg> skifreak: "Could not find codec parameters for stream 0 (Unknown: none (HDMV / 0x564D4448)): unknown codec" What's the source of right.m2ts [01:04] <skifreak> bffmpeg: It's a passthrough bluray rip of Immortals 3D (DVDFab) [01:13] <skifreak> bffmpeg: I would really like to test the right file out with another program to see if it's playable, but if ffmpeg/mplayer fails, what other options do I have? [01:13] <bffmpeg> skifreak: not a dvdfab expert, but do you have the option to turn the codec info on/off. i.e. are you sure the codec info is in the m2ts file? [01:14] <skifreak> bffmpeg: accord to mediainfo it is http://dpaste.com/1398454/ [01:15] <skifreak> the left video file according to mediainfo, for comparison http://dpaste.com/1398455/ [01:17] <skifreak> was even wondering if I could force ffmpeg to use a very specific codec, using the exact same one with the same parameters as the left video file. I've had no success on this end so far either [01:18] <skifreak> definitely open to using something else other than DVDFab, but it seems to work the best on anything I throw at it (runs smoothly in wine) [01:18] <bffmpeg> skifreak: mediainfo seems to see it [01:19] <skifreak> that's what I thought, though I'm no expert on ffmpeg and how codecs are recognized [01:19] <bffmpeg> i would say file your original post, with your mediainfo page, as per the trac page. There is an ftp service (see trac) to upload your file [01:19] <skifreak> bffmpeg: cool, definitely will do. Should a 10MB sample of each suffice? [01:19] <bffmpeg> yes [01:21] <skifreak> cool, thanks a lot for your help. I know bluray is a little weird with how it passes 3D, but I'm looking forward to finally compressing passthrough 3D (have been forced to use DVDFab's own parameters and codecs up until now on 3D files, stuck at using a constant bit rate instead of CRF) [01:22] <bffmpeg> give it a crisp title such as : AVC codec in m2ts file not recognized by ffmpeg [01:23] <bffmpeg> if you file correctly the ticket should pop up here [01:23] <skifreak> bffmpeg: thanks, I'm pretty green at submitting bug reports and proper titles and descriptions do take a small amount of time to learn [01:25] <skifreak> will file the bug tonight, need to fill a void in my stomach first and tear apart some muscle fiber first [02:34] <omega_terry> hi there. trying to download a live youtube stream. getting a 403 error. think i'm using the wrong m3u8 file [02:35] <omega_terry> http://www.youtube.com/watch?v=Bht5_9sUg3k&hd=1 [02:36] <omega_terry> http://pastebin.com/qJiWhTN5 [02:38] <omega_terry> that's mine [02:39] <omega_terry> here's a working example [02:39] <omega_terry> http://pastebin.com/XnVBTqKF [02:56] <sine``> whats the syntax for -vc none [02:56] <sine``> its changed since time ago [02:56] <sine``> more shorthand no [02:56] <sine``> now [02:57] <sine``> 0? [03:04] <klaxa> -nv [03:04] <klaxa> *-vn [03:05] <sine``> and what about copy audio [03:34] <klaxa> -c:a copy [04:21] <defaultro> hey guys, how come my local copy of video is very clean using qp=0 but once I uploaded it to youtube, there's tons of image quality lost [04:52] <skifreak> bffmpeg: about to submit the bug. Do you want me to upload the left video file (the one which ffmpeg has no problem analyzing) as well or just the right video file (the problem file)? [05:56] <skifreak> Submitted ticket #3009 (https://trac.ffmpeg.org/ticket/3009) Thanks all for the support [06:30] <Niatross> Will the dca codec ever come out of the "experimental" phase ("-strict -2")? [09:28] <skifreak> I know this was just submitted, and only recently reviewed, but can someone tell me why ticket #3009 (https://trac.ffmpeg.org/ticket/3009) is an enhancement, not a defect? (The priority of wish seems a little low, but that's just me) [10:46] <Willdude123> Is there a way in FFmpeg to keep the aspect ratio but decrease the quality of a video by some percentage? [10:58] <skifreak> Willdude123: I don't know what your end goal is exactly, but i use the crf option through x264 in ffmpeg to accomplish different amounts of change in "perceived quality" [10:59] <Willdude123> skifreak, well basically I have a 720p video, and I want to scale the quality down so that the stream doesn't lag. [11:01] <skifreak> Willdude123: do you mean decrease the resolution? [11:01] <skifreak> or just make the file smaller? [11:01] <Willdude123> Yes, I think I do. Well I want to do anything that I can to make the stream not lag as much due to my slowish internet [11:02] <skifreak> ah, ok, well I'd recommend using the same res for now, and decreasing the quality of the video, though you could do both [11:03] <skifreak> as far as crf goes: example: ffmpeg -i <input> -c:v libx264 -crf <number> <output> explanation and possible crf values: http://mewiki.project357.com/wiki/X264_Settings#crf [11:04] <Willdude123> Okay... [11:04] <Willdude123> I don't completely understand that but I'll work on it. [11:05] <skifreak> crf just changes the perceived quality (example: blurring frames together in action sequences, where sharpness matters less) [11:06] <skifreak> as far as changing the resolution: http://mewiki.project357.com/wiki/X264_Settings#resize [11:07] <Willdude123> Okay. So what possible crf values are there? [11:07] <skifreak> you can pass any x264 options through to ffmpeg the same way crf is above (make sure to put them after -c:v libx264 and before any other stream changes, such as audio, -c:a, or subtitle, -c:s, and most certainly before the output file) [11:08] <skifreak> 0 to 51 http://slhck.info/crf.html [11:08] <Willdude123> Oh to change the res I was using -s [11:08] <skifreak> the lower the number the better the quality [11:09] <skifreak> well, whichever way to want to handle it. I prefer to use x264, seems it's one of the best video codecs out there [11:10] <skifreak> I encode transparent quality 1080p files with a crf value of 18. Transparent being the same perceived quality as the source. [11:12] <skifreak> for kicks, here's an example of the command I use all of the time ffmpeg -i <input> -map 0 -c:v libx264 -vf crop=<crop values> -crf 18 -tune film -preset slower -profile:v high10 <output> [11:12] <skifreak> you probably don't need -vf crop=<crop values>, since you're not cropping [11:13] <skifreak> just a few different settings to make you aware of and play around with [11:17] <Willdude123> Actually, my internet is too crap to stream it even with awful quality [11:17] <skifreak> ok, well, good luck never the less [11:25] <Willdude123> skifreak, thanks. http://imgur.com/QvPNnxy [11:25] <Willdude123> It buffers for about 5 seconds every 15 seconds but nvm, it was an interesting project. [11:27] <skifreak> Willdude123: no problem, sorry it didn't pan out for you. Glad I could be of some help. [11:28] <Willdude123> skifreak, it kinda sometimes works OK. It was more just a project to do than something I actually expected to work. [11:28] <Willdude123> I guess UDP streaming may be a bit faster. [11:29] <skifreak> Willdude123: good to experiment though. I have no clue on streaming of any kind. [11:30] <Willdude123> Cool. I'm working on something that should let me start a torrent downloading in the morning and then watch it during my lessons that happen to involve working with computers. [11:32] <Willdude123> Perhaps I need a VPS with faster interwebz [11:33] <skifreak> well you did say your bandwidth was the limitation. Shame that upstream has such little priority in consumer ISPs. [11:37] <skifreak> Take care Willdude123, I'm going to sleep while watching some Corpse Bride and hope the new bug I submitted is resolved at some point. [11:37] <Willdude123> Night! Thanks a lot for the help. [11:38] <skifreak> No worries, glad I could shed some light. It's nice to be on the other side of the help desk for a change [18:05] <burek> hi :) [18:06] <burek> here is one interesting question imho: "Can ffmpeg detect an overall "lightness" level in a photo?" -> http://ffmpeg.gusari.org/viewtopic.php?f=16&t=1076 [18:07] <durandal_1707> and how would such info be exposed to user? [18:07] <durandal_1707> is that some number in some units? [18:08] <burek> i guess it's an illumination [18:09] <burek> http://en.wikipedia.org/wiki/Lux [18:10] <burek> but since it's a still image, probably some arithmetic middle could be calculated for all pixels [19:05] <burek> is it possible to specify pix_fmt in ffserver.conf [19:06] <burek> or even better is there any way to get the list of all supported options that can be used in ffserver.conf file? [19:07] <burek> i found this, i hope it hasn't changed https://trac.ffmpeg.org/ticket/2954 [19:57] <burek> is there any good page that explains the relation between -maxrate, -minrate and -bufsize, because there are several duplicate questions about it, where people in general lack the understanding of that relation and as a result use them in a wrong way [20:09] <burek> this question is a typical one of that kind http://ffmpeg.gusari.org/viewtopic.php?f=11&t=1063 [21:00] <shevy> hmm got a question [21:00] <shevy> let's assume I have two .mp3 files [21:00] <shevy> both may have ' ' as part of the character name. I am using linux + bash right now [21:00] <shevy> I use this commandline: [21:01] <shevy> ffmpeg -i concat:"foo bar.mp3"\|" bla ble.mp3" -acodec copy output.mp3 [21:01] <shevy> this seems to work, the resulting output.mp3 plays fine. but is this the right syntax to use for concat: ? it looks strange... [21:25] <burek> ShadowJK [21:25] <burek> oops shevy [21:26] <burek> the backslash \ escapes the pipe character, so it's ok [21:26] <shevy> ok [21:26] <burek> concat: will look for the value after the colon : so, it's good :) [22:03] <BlackBishop> any recommended opts to fastly convert from a fullhd mkv(x264/aac) to something flv supported ( @640:480 ) ? [22:06] <BlackBishop> or even 320x240 .. just need it to be fast .. my nvidia ion can decode the mkv fast enough .. but I can't make it encode to flv fast enough .. [22:08] <BlackBishop> -vcodec flv -ac 1 -acodec mp3 -ar 44100 -f flv gets me @ 13fps :| [22:29] <BlackBishop> maybe make vdpau help somehow .. :| [22:51] <relaxed> BlackBishop: no [22:51] <relaxed> a faster cpu will [23:41] <burek> BlackBishop, why not using -vcodec libx264 -f flv [23:45] <sacarasc> Because he wants encoding speed. [23:45] <sacarasc> I suppose -preset ultrafast might work well. [00:00] --- Sun Sep 29 2013

1 0

ffmpeg-devel.log.20130927
by burek 28 Sep '13

28 Sep '13

[00:26] <BBB> is it a known issue that ffplay can'y play video only files? [00:26] <BBB> or rather, it sucks ass at it unless you say -sync video [00:31] <beastd> BBB: report a bug. i think marton would have a look [02:57] <cone-388> ffmpeg.git 03Michael Niedermayer 07release/0.10:20854f9bffd2: avcodec/parser: reset indexes on realloc failure [02:57] <cone-388> ffmpeg.git 03Michael Niedermayer 07release/0.10:e28812439484: avcodec/flashsv: check diff_start/height [02:57] <cone-388> ffmpeg.git 03Michael Niedermayer 07release/0.10:ef8145270f4a: avcodec/rpza: Perform pointer advance and checks before using the pointers [02:57] <cone-388> ffmpeg.git 03Michael Niedermayer 07release/0.10:1a311ad99a57: jpeg2000: check log2_cblk dimensions [02:57] <cone-388> ffmpeg.git 03Michael Niedermayer 07release/0.10:5230f1529ae6: matroska_read_seek: Fix used streams for subtitle index compensation [02:57] <cone-388> ffmpeg.git 03Michael Niedermayer 07release/0.10:9300b1f64e5b: avcodec/pngdsp: fix (un)signed type in end comparission [02:57] <cone-388> ffmpeg.git 03Michael Niedermayer 07release/0.10:a99aff4e4bbe: avcodec/dsputil: fix signedness in sizeof() comparissions [02:57] <cone-388> ffmpeg.git 03Michael Niedermayer 07release/0.10:4a5bb426e26e: avcodec/mjpegdec: Add some sanity checks to ljpeg_decode_rgb_scan() [02:57] <cone-388> ffmpeg.git 03Michael Niedermayer 07release/0.10:f3dc3bef4b40: avcodec/truemotion2: Fix av_freep arguments [02:57] <cone-388> ffmpeg.git 03Michael Niedermayer 07release/0.10:f6057c5a62ea: avcodec/ffv1enc: update buffer check for 16bps [02:57] <cone-388> ffmpeg.git 03Michael Niedermayer 07release/0.10:f66ecdb1b4de: avcodec: add emuedge_linesize_type [02:57] <cone-388> ffmpeg.git 03Michael Niedermayer 07release/0.10:0d1ae06fe95e: update for 0.10.9 [03:10] <cone-388> ffmpeg.git 03Mashiat Sarker Shakkhar 07release/0.10:89d56f3be110: vc1dec: Do not use random pred_flag if motion vector data is skipped [03:26] <cone-388> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n0.10.9': unknown revision or path not in the working tree. [03:26] <cone-388> Use '--' to separate paths from revisions [03:26] <cone-388> refs/tags/n0.10.9:HEAD: avcodec/parser: reset indexes on realloc failure [07:24] <Jerry123> May I ask a question? [07:26] <Jerry123> my ffmpeg code is 2.0.1 version. I use mingw compiler under windows. [07:26] <Jerry123> My configure options are : $ ./configure --prefix=/mingw --disable-yasm --cpu=i686 --enable-static --disab [07:26] <Jerry123> le-shared [07:27] <Jerry123> but that still link the share lib: libiconv-2.dll [07:27] <Jerry123> I want to link the static lib. How can I do it? [07:29] <JEEB> Static is enabled by default, and shared disabled. But that is related to ffmpeg, not its dependencies. Also I feel the easiest way to have whatever link against a static lib instead of a shared one is to not have the dll.a there, just the .a [07:29] <JEEB> dll.a is the library that links against the dll (shared), .a is generally static [07:29] <JEEB> also I have no idea how your toolchain works [07:30] <JEEB> or where this dependency is coming from [07:30] <JEEB> if it's ffmpeg itself then you can most probably also just disable iconv [07:30] <JEEB> if it's not ffmpeg itself then you will have to look somewhere else in your dependencies [07:30] <JEEB> also you really, really do not want to disable yasm [07:31] <Jerry123> Thanks a lot, I will try. [07:31] <JEEB> those are runtime-checked optimizations that range everywhere from MMX to AVX(2), and they /do/ make ffmpeg faster on pretty much every machine [07:32] <Jerry123> I see. [07:36] <TheRyuu> use this toolchain instead of whatever mingw one you're using: http://xhmikosr.1f0.de/tools/MSYS_MinGW-w64_GCC_481_x86-x64_Full.7z [07:36] <TheRyuu> that *only* has static libs [07:36] <TheRyuu> (to link against) [07:37] <Jerry123> OK thanks [07:51] <Jerry123> I delete all *.dll.a files, and then relink, that's wokes well. :) [08:12] <nevcairiel> TheRyuu: yes, it mostly just uses the default things to open files [08:12] <nevcairiel> although addtionally there is also some custom avio logic to open a stream, but thats only used if required [08:14] <TheRyuu> yea, I mostly figured out the workings of it after poking it a bit [09:01] <durandal_1707> wm4: so what you think about replaygain filter? [09:02] <wm4> durandal_1707: is it somewhere? [09:03] <durandal_1707> i posted link here yesterday [09:43] <Jerry123> hi all [09:44] <Jerry123> I use mingw+ffmpeg+sdl. use static sdl lib, sdl lib needed -liconv -lm -luser32 -lgdi32 -lwinmm options. [09:44] <Jerry123> but the config.mak can not included these options. [09:44] <Jerry123> I must add it by hand. [09:51] <durandal_1707> michaelni: if you not gonna relicense noise to lgpl, i will write addgrain filter [09:59] <cone-106> ffmpeg.git 03Martin Storsjö 07master:8b09d917e7dc: tls: Add options for verifying the peer certificate [09:59] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:e6e71fd7b563: Merge commit '8b09d917e7dc7d7f2ace31419f802d4ff518236c' [10:42] <durandal_1707> wm4: tried it? [10:42] <wm4> no [10:42] <durandal_1707> why? [10:42] <wm4> I don't know where it is anyway [10:47] <cone-106> ffmpeg.git 03Martin Storsjö 07master:5c53bf7aaf03: http: Pass options through to the nested protocol [10:47] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:689a1bd917d5: Merge commit '5c53bf7aaf03748464cbf978bffe7ffdb71112b1' [11:38] <durandal_1707> should contrast/gamma be added to hue, or new filter is preffered? [11:39] <cone-106> ffmpeg.git 03Martin Storsjö 07master:705b748e8d86: tls: Add support for listen mode [11:39] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:f31c36e55266: Merge commit '705b748e8d8612385c96428ae36ed0d42a170d93' [11:43] <saste> durandal_1707, depends [11:44] <saste> probably as an extensions, if there is enough code to share with hue [11:45] <cone-106> ffmpeg.git 03Martin Storsjö 07master:1af78b865b66: rtpdec_qt: Add an accidentally removed allocation return value check [11:45] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:47c5c942f823: Merge commit '1af78b865b66acb8020c54262f12aa41ad47e05c' [11:45] <durandal_1707> i pressume it means new filter [11:54] <cone-106> ffmpeg.git 03Martin Storsjö 07master:06ebc0bf9a64: lavf: Allocate arrays with av_realloc if they will be realloced later [11:54] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:328a5b93d3a4: Merge commit '06ebc0bf9a6401733a4ce1310325de19f631819a' [12:19] <cone-106> ffmpeg.git 03Martin Storsjö 07master:d872fb0f7ff2: lavf: Reset the entry count and allocation size variables on av_reallocp failures [12:20] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:72eddc10fa1f: Merge commit 'd872fb0f7ff2ff0ba87f5ccf6a1a55ca2be472c9' [12:48] <cone-106> ffmpeg.git 03Reimar Döffinger 07master:779951cd2c1b: lxf: Support version 1 files [12:48] <cone-106> ffmpeg.git 03Carl Eugen Hoyos 07master:839df90c718d: lxf: Support 16-channel files [12:48] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:e82071e7279a: Merge commit '839df90c718dcab9b9e91ca3c7e73479b3e8103c' [12:56] <Compn> durandal_1707 : i'd like it to have a better name 'hue' but thats just me bikeshedding [12:56] <Compn> i dont know that 'eq' is any better [13:09] <cone-106> ffmpeg.git 03Paul B Mahol 07master:7e52080cb183: pcm: support 24-bit/32-bit little-endian planar [13:09] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:b654aa6beb41: Merge commit '7e52080cb18332cb1da9fe133498750b6b4b0fb3' [13:17] <cone-106> ffmpeg.git 03Paul B Mahol 07master:a529fa2f4dd3: lxf: remove deplanarization hack [13:17] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:fc424f37a37e: Merge commit 'a529fa2f4dd3292ba7d9cd3e6ba76425b1ba7aef' [13:32] <cone-106> ffmpeg.git 03Luca Barbato 07master:5c349ad08a9f: lxf: check the nb_streams instead of relying on padding [13:32] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:d8d235ffe9c1: Merge remote-tracking branch 'qatar/master' [15:02] <Jerry123> I use mingw+ffmpeg+yasm Yasm generate .obj file, but AR needed .o file, how correct it? [15:04] <durandal_1707> saste: how to use sendcmd with X filter, if X filter is used several times in row? [15:05] <cone-106> ffmpeg.git 03Paul B Mahol 07master:4d86a945dfe7: avfilter/af_ladspa: refactor code setting input control ports values [15:05] <cone-106> ffmpeg.git 03Paul B Mahol 07master:110a2f446d54: avfilter/af_ladspa: allow changing controls dynamically [15:05] <cone-106> ffmpeg.git 03Paul B Mahol 07master:191ee4aefbde: avfilter/af_ladspa: set output channel layout for passthrough case [15:15] <michaelni> durandal_1707, i dont mind if you use my code in vf_noise.c 15985 bytes / d0b86b7749de940eb00d3d47cdff71e54ee9e07d under LGPL [15:15] <Jerry123> YASMFLAGS must add a compiler option: --objext=o [15:16] <Jerry123> If -fwin32 , yasm's object's file suffix is .obj [15:18] <durandal_1707> michaelni: is that hash from mplayer? [15:23] <Daemon404> Jerry123, we do not need that [15:23] <Daemon404> we explicitly use -o [15:23] <Daemon404> yasm -f elf -m amd64 -g dwarf2 -I./ -I.// -Pconfig.asm -I libavcodec/x86/ -o libavcodec/x86/hpeldsp.o libavcodec/x86/hpeldsp.asm [15:23] <Daemon404> e.g. like that [15:23] <Daemon404> (this holds true on windows too) [15:24] <Jerry123> I think that's a configure script's bug. [15:24] <Daemon404> i cant reproduce it on windows here. [15:25] <Jerry123> ffmpeg version 2.0.1, configure file, line 3924 [15:25] <Daemon404> im using git HEAD [15:25] <Daemon404> lets see... [15:27] <Jerry123> YASMFLAGS="-f ought to change to YASMFILAGS="--objext=o -f [15:27] <Daemon404> library.mak [15:27] <Daemon404> line 25 [15:27] <Daemon404> where it is actually used [15:27] <Daemon404> $(SUBDIR)x86/%.o: $(SUBDIR)x86/%.asm $(DEPYASM) $(YASMFLAGS) -I $(<D)/ -M -o $@ $< > $(@:.o=.d) $(YASM) $(YASMFLAGS) -I $(<D)/ -o $@ $< [15:27] <Daemon404> -i is *always* passed. [15:28] <Daemon404> er -o [15:28] <Jerry123> I see. [15:28] <Jerry123> but my yasm compiler was generated .obj file [15:28] <Daemon404> what ar you confgre flags [15:29] <Jerry123> that's surprise. [15:29] <Daemon404> and yasm version [15:29] <Jerry123> my yasm is the latest [15:29] <Daemon404> from where [15:29] <Daemon404> which binary [15:29] <Jerry123> yasm 1.2.0 [15:29] <Daemon404> thats not what i asked [15:29] <Jerry123> vsyasm.exe [15:30] <Daemon404> i believe vsyasm is patched. [15:30] <Daemon404> it is not vanilla yasm [15:30] <Jerry123> sorry my English is poor [15:30] <Daemon404> and ou should not be using it [15:30] <Jerry123> I see. [15:30] <Daemon404> http://www.tortall.net/projects/yasm/releases/yasm-1.2.0-win32.exe [15:30] <Jerry123> thanks I will try it. [15:38] <Jerry123> <Daemon404> I thank I find out the reason, my 1.2.0 version's yasm, have no -M option. [15:55] <saste> durandal_1707, you need to use the "unique" filter instance name, something like "parsed_filter_X" [15:55] <saste> that's why we should allow the user to specify a unique name, which will not depend on the position in the parsed graph [15:59] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:df21537dfbb8: avformat/tls: fix {} error for the GNUTLS case [16:41] <michaelni> durandal_1707, its a sha1 from a file [16:49] <cone-106> ffmpeg.git 03Georg Martius 07master:e064d0aaf622: avfilter/vidstabtransform: update optzoom range values for recent vid.stab update. [17:01] <mark4o> durandal_1707: lutyuv can already do gamma, contrast, and brightness adjustments [17:02] <mark4o> although it requires entering formulas currently [17:02] <durandal_1707> command? [17:02] <mark4o> adding options like lutyuv=gamma=0.9:brightness-1.1:contrast=1.2 shouldn't require any change to filter_frame [17:02] <mark4o> -vf lutyuv=y=gammaval(0.5) is listed in the docs [17:03] <mark4o> for gamma [17:08] <durandal_1707> -vf drawgraph ? [17:25] <mark4o> -vf lutyuv=y=clip(val * x + y) for contrast/brightness adjustment [17:27] <mark4o> -vf lutyuv=y=clip(val * 1.5 - 20) [18:04] <durandal_1707> saste: i wonder about doing something like frei0r nervous filter, it cache several frames and than pick random one from the list [18:50] <saste> durandal_1707, why not [18:51] <durandal_1707> saste: another is audio spectrum filter that is like showspectrum but it interprets data differently [18:59] <durandal_1707> saste: about nervous, i'm just not happy with that name [19:09] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:9790a03e67d3: avformat/mxfdec: check sample size more completely [20:25] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:f7bea731d955: avcodec/mlpdec: Fail if the input is too small [20:44] <durandal_1707> michaelni: the hash you put into tickets are invalid [20:44] <durandal_1707> invalid hashes are worse than no hash at all [20:46] <nevcairiel> i just checked the truehd one, and it has the right hash [20:48] <durandal_1707> mxf one is wrong [20:57] <cone-106> ffmpeg.git 03Paul B Mahol 07master:b890198b669d: avfilter: add av_cold to init() [20:58] <cone-106> ffmpeg.git 03Paul B Mahol 07master:c2d19430290d: fate: add phase test [21:07] <michaelni> durandal_1707, hash fixed [21:27] <durandal_1707> michaelni: those FFmpeg forums nobody use except noobs (expecting help that never comes) so should be removed from contact page [21:58] <Mista-D> Can someone point to a location of default analyzedurtion. dealing with an older customized binary that won't accept "-analyzeduraiton 50M", so I want to increase the default in libav. [22:03] <durandal_1707> libavformat/options_table.h [22:26] <Mista-D> @durandal_1707: Thank you. It worked perfectly. [23:20] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:3e0c78bac63c: avcodec/imc: Check flcoeffs5 [00:00] --- Sat Sep 28 2013

1 0

ffmpeg.log.20130927
by burek 28 Sep '13

28 Sep '13

[00:56] <captainastro> hello everyone...please excuse any protocol violations, but it's been many years since I used IRC [00:57] <captainastro> I've been working with point to point streaming and I've had some success with UDP, but it's flakey - choppy, drop frames, freezes, etc. [00:57] <klaxa> well it's UDP [00:57] <captainastro> I've tried using TCP and RTSP, but I am not getting anywhere [00:57] <klaxa> it's unreliable [00:57] <captainastro> yea, udp [00:58] <captainastro> here's what works: sending side [00:58] <captainastro> ffmpeg -v debug -fflags nobuffer -f v4l2 -i /dev/video0 -c:v libx264 -tune zerolatency -b:v 100k -pix_fmt yuv420p -c:a libmp3lame -f mpegts udp://trajan.local:1234?pkt_size=188?buffer_size=65535 [00:58] <captainastro> ffplay -fflags nobuffer -i udp://localhost:1234?listen?pkt_size=188?buffer_size=65535 [00:58] <captainastro> on the receiving side [00:59] <captainastro> when I've substituted TCP, for instance (sending) ffmpeg -v debug -fflags nobuffer -f v4l2 -i /dev/video0 -c:v libx264 -tune zerolatency -b:v 100k -pix_fmt yuv420p -c:a libmp3lame -f mpegts tcp://trajan.local:1234 [00:59] <captainastro> (receiving) ffplay -fflags nobuffer -i tcp://localhost:1234?listen [01:00] <captainastro> it just hangs. Same if I do something like rtsp://localhost:1234/live.sdp [01:00] <captainastro> wiresharc sees the sending machine send out packets to port 1234, but there is no response. [01:00] <captainastro> I'm stumped [01:01] <captainastro> I've tried just about everything I can think of, and I think I've looked at everything that google can dig up. By the ffmpeg docs and the streaming guide both should be working. [01:03] <captainastro> has anyone had any success with point to point streaming with anything other than udp? [01:30] <trem0r> Hey guys. I'm trying to create some music on hold. It needs to be in 16kbps PCM linear. I've got an existing wav file that I need to convert. how can I go about that? [01:30] <trem0r> when I put -ab <desired rate> the bitrate never changes [01:35] <klaxa> PCM is raw [01:35] <klaxa> the bitrate depends on the format [01:37] <trem0r> Can you explain that to me? I'm not really an audio guy =) [01:39] <klaxa> http://en.wikipedia.org/wiki/Pulse-code_modulation [01:39] <captainastro> You need to encode the audio into something like mp3. The encoding compresses the stream and lowers the quality when you specify the bitrate [01:40] <captainastro> PCM streams at whatever rate it was captured at [01:40] <trem0r> I see. [01:40] <trem0r> Thank you [01:43] <trem0r> Funny [01:43] <trem0r> Just converted it to mp3..lowered the bitrate [01:43] <trem0r> when I convert it back to a wav file, the bitrate is like 700kbps [01:44] <klaxa> pcm is raw [01:45] <trem0r> So are you saying that this isn't possible? [01:45] <klaxa> yes [01:45] <klaxa> well [01:45] <klaxa> no [01:45] <klaxa> it is possible, but it will sound bad [01:45] <trem0r> Yeah [01:45] <klaxa> because 16 kbps is very little [01:45] <trem0r> It's for a music on hold system [01:46] <trem0r> Pretty annoying requirement [01:46] <klaxa> maybe you are talking about 16 khz? [01:46] <trem0r> Here's the requirement from our voip host. Perhaps they got it wrong [01:46] <trem0r> v [01:46] <trem0r> Please make sure that the file is a wave file with either 16 kpbs linear PCM encoding or 8kbps PCMU encoding. [01:46] <trem0r> They even mistyped kbps [01:47] <captainastro> you might need to try different encoding technologies - some are better than others [01:47] Action: stump thinks that probably means 8 kHz 16-bit mono [01:47] <trem0r> Lemme test your theory [01:47] <klaxa> yeah 16 kbps is just a weird requirement for pcm [06:25] <t4nk022> When I try to use ffserver + ffmpeg to output a live h.264 rtp stream, I got the problem [06:25] <t4nk022> the timestamp in rtp header may get overflow. [06:26] <t4nk022> since the pkt.pts, pkt.dts in ffmpeg are stored in 64bits length [06:26] <t4nk022> timestamp field in RTP header is stored in 32bits length [06:26] <t4nk022> it may get overflow when long time test. does any one have idea about how to fix the issue? [06:33] <skifreak> having a problem with ffmpeg recognizing the codec for a right bluray, passthrough, video file. mediainfo shows very similar information between the right and left video files, with only the left containing audio. Tried increasing probesize and analyzeduration to no avail. ffmpeg output: http://dpaste.com/1397661/ mediainfo output: http://dpaste.com/1397663/ I have a sample file (10 MB) of each ready on dropbox if needed [06:49] <SirCmpwn> I have an ass file and an mkv file [06:49] <SirCmpwn> I want to include the ass file in the mkv, but I still want softsubs [06:49] <SirCmpwn> how? [06:57] <skifreak> Sircmpwn: use the map option. ffmpeg -i <video> -i <subtitle> -map 0:0 -map 1:1 <other options> <output_video> [06:58] <skifreak> maphttps://trac.ffmpeg.org/wiki/How%20to%20use%20-map%20option [06:58] <SirCmpwn> the numbers there make me worry that I'll lose any other streams in the video file [06:58] <skifreak> typo: https://trac.ffmpeg.org/wiki/How%20to%20use%20-map%20option [06:59] <SirCmpwn> oh, I see [06:59] <SirCmpwn> thank you [06:59] <skifreak> np [09:46] <CentRookie> If one changes the x264 source code slightly, turns a few settings off and recompiles it, do I have to recompile ffmpeg as well? [09:47] <JEEB> if you use shared libraries then you just have to make sure both compiles are API/ABI compatible. [09:48] <JEEB> and you can just install/ldconfig and run [09:48] <JEEB> if static, then ffmpeg has to be rebuilt as well [09:48] <CentRookie> ic [09:48] <CentRookie> yes im using shared [09:49] <CentRookie> Is there a way to add a line into the file header, user data ? I would like to add some infos like original resolution [09:49] <CentRookie> and camera settings [09:50] <CentRookie> am looking for the header info in the source code but havent found it yet [09:51] <JEEB> I'd rather put stuff like that into the container's possible freetext metadata fields [09:52] <JEEB> but I think you can use some kinds of SEI messages for it? [09:52] <CentRookie> sei message? [09:52] <JEEB> I think libx264 had support for it, no idea if you can use them via ffmpeg's wrapper [09:58] <durandal_1707> -x264-params [10:01] <CentRookie> how does it help me durandal [10:13] <simonsimcity> I have a mov-file without MOOV atom ... [10:13] <simonsimcity> Anyone having an idea on how to repair it using ffmpeg? Or another app running on mac ;) [10:21] <JEEB> too bad that's the indedx of the file [10:21] <JEEB> *index [10:21] <JEEB> which contains the offsets of everything in the file, basically [10:22] <JEEB> at best you can do some guesswork, and if it's from some specific hardware thing see if it always outputs packets of a similar size [10:22] <JEEB> there are some tools that claim they can do this guesswork automated, but none of them are free as far as I know [10:23] <JEEB> and they naturally do not release their heuristics [10:23] <JEEB> in some cases such things partially work, in other cases they fail badly [10:23] Action: JEEB shrugs [10:24] <simonsimcity> JEEB, I took a review using http://aeroquartet.com/movierepair/repair# and it told me, the movie would be repairable by 99% ... but it would cost up to 200¬ for us to do it ... [10:24] <simonsimcity> Do you think something like this could help? Have you heard of Untrunc before? http://superuser.com/questions/417100/how-to-open-and-repair-an-m4v-or-mp4-… [10:25] <JEEB> oh [10:25] <JEEB> someone actually started doing something open source? [10:25] <JEEB> nice [10:26] <simonsimcity> Oh! sounds good! [10:26] <simonsimcity> I wrote a small program using QT, ffmpeg and libfaad which rebuilds the index and, given a complete video as an example, recreates the moov atom, thus rendering the file playable again. [10:26] <simonsimcity> quote-end [10:27] <JEEB> well, basically good luck [10:29] <simonsimcity> JEEB, I read that f.e. VLC is able to guess all indexes ... but since the video- and audio-codec also is stored in the missing piece, VLC can't do much. [10:30] <simonsimcity> Is it possible to just add information about the codecs? [10:30] <simonsimcity> I have another file, taken with the same equipment ... [10:31] <durandal_1707> i guess so, but i never compared those bits between files [10:31] <JEEB> well, the problem is that you have to make the offsets match too [10:31] <JEEB> so it will be hex editing and abiding by the specification and so forth [10:32] <JEEB> tools like atomicparsley and L-SMASH's boxdumper probably will come in handy [10:53] <getsome> hey guys, i recently upgraded to a newer ffmpeg version (1.2.3) and now got the problem that generated thumbnails of videos are really huge (thumbnail of 10mb video is 100mb+)... [10:53] <getsome> here is the command i am using [10:53] <getsome> ffmpeg -y -ss 00:00:05 -r 1 -vframes 1 -i <INPUT FILE> -f mjpeg <OUTPUT FILE> [10:54] <getsome> can anyone point me in the right direction to fix this issue? [10:55] <durandal_1707> getsome: upgraded from what? [10:56] <durandal_1707> shouldn't that command generate single file? [10:56] <durandal_1707> what is your source? [10:58] <CentRookie> does ffmpeg support QuickSync from intel? [11:00] <durandal_1707> that question is asked here over and over again [11:00] <durandal_1707> currently not, but there is something quicksync related cooking iirc [11:00] <viric> mittens: ok! [11:02] <CentRookie> too bad [11:03] <CentRookie> thanks though [11:03] <CentRookie> just realized that my server got quicksync [11:15] <getsome> durandal_1707: yes it generates a single image, but it's huge...everything worked fine with FFmpeg version SVN-r26402 [11:20] <durandal_1707> getsome: how big is your source? [11:21] <durandal_1707> i mean resolution of source image [11:21] <durandal_1707> just pastebin whole output [11:23] <getsome> yea sure, just give me a minute [11:27] <getsome> seems like ffmpeg tries to output a video, not sure though: http://pastebin.com/GtZkYK5b [11:29] <durandal_1707> hmm than your -vframes 1 should be after -f mjpeg [11:31] <durandal_1707> getsome: and with 2.0 your command would fail and give big red message [11:36] <getsome> durandal_1707: just tried it with -vframes 1 after the -f and it seems to work now, gonna do some further testing....it's ages ago when this feature was implemented, seems like no one really cared about upgrading the ffmpeg version, still kinda weird that it worked like this with this ancient ffmpeg version [11:36] <getsome> oh and ofc a big thank you for your help [11:52] <caraculo> hi [11:52] <caraculo> does FFMPEG support the IMC4 pixel format? [11:53] <simonsimcity> JEEB, you mentioned L-SMASH's boxdumper ... What is L-SMASH? I can just find some text in ... looks like Japanese ... [11:59] <durandal_1707> caraculo: you have sample? [12:00] <caraculo> http://www.fourcc.org/yuv.php#IMC4 [12:01] <durandal_1707> that is not sample [12:01] <durandal_1707> and without sample it is hard to know code that actually adds support for it is correct [12:03] <caraculo> it is a YUV planar like YV420, but the UV planes are interleaved in a row basis, instead of concatenated. [12:05] <caraculo> something like this in memory: Y_R1 Y_R2 U_R 1 V_R1 [12:06] <durandal_1707> whatever [12:06] <durandal_1707> no sample - no support [12:07] <caraculo> what do you mean exactly by sample? [12:08] <caraculo> look, at FOURCC you have the pixel format described in great detail, ar shall we all just spean in "C"? [12:09] <durandal_1707> well i will just ignore you [12:10] <caraculo> yes, with so many typos in my last sentence I'd do the same. [12:17] <durandal_1707> and uv planes are not interleaved at all [12:18] <caraculo> ? [12:23] <saste> caraculo, do you have a file (sample) using that format? [12:23] <durandal_1707> do you want IMC support in swscale or you just need to decode it? [12:25] <caraculo> I should state my problem better: I need to encode a buffer which is in IMC4 format [12:28] <durandal11707> and where would you send that buffer? [12:31] <caraculo> to disk, basicaly encode it (a sequence of those) and then just save em into a file [12:31] <durandal11707> raw one? [12:31] <saste> caraculo, IMC2 - this looks more like a codec format than a pixel format [12:31] <caraculo> lossless FFv1 [12:32] <saste> in other words it can't be represented through a pixel format [12:32] <durandal11707> saste: it is similar to nv21/12 [12:32] <saste> durandal11707, yes, but you have - one line U, one line V, one line U, ... [12:33] <durandal11707> in nv it is: UVUVUVUVUVUVUVU and here it is UUUUUUUUVVVVVVVVV [12:34] <durandal11707> i see no codec there [12:34] <durandal11707> it could be supported in swscale [12:35] <caraculo> IMC4 is a planar pixel format. [12:36] <caraculo> it is just that the planes are not simply concatenated like in YV420 [12:36] <saste> durandal11707, I don't see how you can code the U/V components with pixdesc.h:AVComponentDescriptor [12:36] <caraculo> now, my input buffer is in such format, I just do not want to transform it to YV240 before encoding it into FFV1. [12:37] <saste> caraculo, how are your coding skills? [12:38] <saste> it is trivial to write a decoder for such a raw format, check for example v410dec.c [12:39] <saste> but i'm still curious, do you have samples using this format? [12:43] <saste> caraculo, who/what is creating the buffer? why don't you use a more standard "pixel format"? [12:46] <caraculo> the buffer comes from a camera... the drives is spiting frames in such format [12:46] <caraculo> driver* [12:47] <caraculo> so I can access the frame buffer, and I just do not want my application to have too re-format it. So I was wondering if FFMPEG can work witch such an input format. [12:48] <saste> caraculo, so the short answer is: no [12:48] <saste> OTOH it should be really trivial to hack your loop for converting the buffer to any of the formats directly supported by ffmpeg [12:49] <caraculo> thanks saste [12:50] <caraculo> yes, it is trivial andknow how to do it, I will just introduce some"latency" and lines of code to the application [15:23] <lei> hi, I have a file test.mp4, I want to convert it to mpeg2ts(contains h.264 and aac), how to write command line ? [15:25] <lei> hi, fellows. [15:29] <viric> lei: it's good to show that you already did some effort on your side, if you want to receive help :) [15:30] <lei> ffmpeg -i srcfile -y -f mpegts dstfile [15:30] <lei> I asm googleing [15:30] Last message repeated 1 time(s). [15:30] <lei> and if you guys can help me, thanks a lot. [15:32] <lei> hi, viric. [15:33] <lei> ffmpeg -i srcfile -y -f mpeg2ts -acodec aac -vcodec h264 dstfile [15:34] <lei> I don't know if this is right. [15:35] <viric> test it :) [15:40] <lei> C:\tc\1\input>ffmpeg -strict -2 -i video.mp4 -y -f mpegts -acodec aac -vcodec h2 [15:40] <lei> 64 out.ts [15:40] <lei> [aac @ 04bcb980] The encoder 'aac' is experimental but experimental codecs are n [15:40] <lei> ot enabled, add '-strict -2' if you want to use it. [15:42] <lei> I try ffmpeg -strict -2 -i video.mp4 -y -f mpegts -acodec aac -vcodec h264 out.ts [15:42] <lei> but it tells me something wrong. [15:44] <lei> viric [15:45] <viric> ah, well, you may need another aac encoder. [15:53] <blacky> hello [15:54] <blacky> how can i read video input from declink [16:00] <blacky> unknown input format show [16:00] <blacky> how to solve this any idea ? [16:00] <blacky> dshow* [16:03] <zap0> declink? [16:18] <blacky> zap0: decklink [16:18] <blacky> sorry [16:19] <zap0> same question decklink? what is it? [16:21] <zap0> oh.. black magic capture card [16:21] <zap0> they have there own API i think [16:24] <viric> what tool can add tags to webm files? [16:24] <viric> author, title, ... [16:25] <saste> viric, ffmpeg -i ... -codec copy -metadata author=... -metadata title=... out.webm [16:26] <saste> or you can use the ffmeta demuxer [16:26] <viric> ah ok. nothing working on a single file, right? [16:26] <viric> all goes through copying the whole file [16:26] <viric> ffmeta? [16:27] <saste> viric, yes you need to create a new file [16:27] <viric> ok [16:28] <viric> saste: I guess there are some restrictions on those tags. usual tag names, format of date field... how may I know them? [16:29] <saste> viric, dunno, try&fail, or check some docs online [16:29] <viric> ah http://wiki.multimedia.cx/index.php?title=FFmpeg_Metadata#Matroska [16:29] <viric> that may apply. [16:29] <viric> thank you saste ! [16:29] <viric> very helpful [16:30] <saste> viric, you owe me a beer [16:30] <viric> yes [17:11] <voltin> Has anybody had any luck taking frame grabs of a rawvideo (Y160) rtsp stream? For some reason FFMPEG is having problems determining the information about the stream including pixel format and size. When I try to specify either option is complains that the option doesn't exist. Here is a pastebin with 3 attempts and their output with debug enabled: http://pastebin.com/JJdQY91d [17:16] <mittens> viric: did you get vp9 working? [17:16] <viric> I didn't have the oportunity to test yet [17:19] <bunniefoofoo> for CBR encoding mpeg2video, is the bitrate constraint enforced per-frame, per-gop or something else? [17:48] <bunniefoofoo> ok I see it is recalculated every frame [19:20] <Franklin> running "gst-launch filesrc location="/home/franklin/Downloads/test.avi" ! codebin2 ! audioconvert ! audioresample ! autoaudiosink" only gives me audio, video isn't showing up, am I missing something? please help. [21:05] <chrisballinger> hey the files i'm creating with a direct stream copy w/ libavformat play back at 3600 fps instead of 24 fps, does anyone have any clues as to where to look more closely? [21:07] <chrisballinger> i'm not using the command line, im using the libavformat api [21:09] <chrisballinger> but i have some stuff i prepared yesterday [21:10] <klaxa> ah right... sorry, not too familiar with the api [21:10] <chrisballinger> https://gist.github.com/chrisballinger/6733678 [21:10] <chrisballinger> that is the output of ffprobe for the broken file [21:11] <chrisballinger> the main thing is that r_frame_rate=1/1, duration_ts=2999, duration=0.033322 [21:12] <chrisballinger> a proper file looks more like r_frame_rate=24/1 duration_ts=449850 duration=4.998333 [21:12] <chrisballinger> https://github.com/OpenWatch/FFmpegWrapper/blob/master/FFmpegWrapper/FFmpeg… [21:12] <chrisballinger> that's where i do the setup for the direct stream copy [21:50] <kni> hey guys [21:50] <kni> I'm trying to set stuff up to I can stream to an external server [21:50] <kni> and I'm confused as to how I configure it [21:51] <kni> is the "ACL allow" stuff for the input feeds or the actual output stream? [22:19] <bunniefoofoo> hi all, looking for some way to get bitrate "more constant", that is to say I have a 5.5 Mbps MPEG2 stream and some of the gops are as high as 7.5 Mbps even though I have bitrate tolerance set to 0.5 mbps [22:19] <bunniefoofoo> I am using the standard approach max_bitrate==min_bitrate==bitrate and setting rc_buffer_size [22:20] <bunniefoofoo> I tried setting bitrate_tolerance to the minimum possible value and ended up with corrupted p-frames (or so it seems) [22:28] <Niatross> Will the dca codec ever come out of the "experimental" phase ("-strict -2")? [23:01] <shevy> is there a way to apply some kind of audio de-noise filter through ffmpeg? on a .mp3 file [23:05] <klaxa> i don't think so, you can use sox though [23:19] <shevy> nooooo! [23:19] <shevy> :( [23:19] <shevy> I hate sox [23:20] <klaxa> how can you hate sox? [23:20] <shevy> I once modified mortal kombat theme song, years ago with sox [23:20] <shevy> I kept that .mp3 file for years... I removed it only a few years ago. I used to think it had a great effect, but it sucked! [23:21] <shevy> the time before that I was using virtualdub + avisynth. that was beautiful [23:21] <shevy> that's the only thing I miss from my windows days [23:22] <klaxa> wait, so because you messed up with the configuration sox is bad? [23:22] <chrisballinger> Here is a stackoverflowized version of my question if that helps: http://stackoverflow.com/questions/19060253/files-created-with-a-direct-str… [23:22] <klaxa> you can also just use audacity if you have a GUI available [23:22] <shevy> klaxa configuration? no. it's a completely useless app [23:22] <klaxa> wow [23:22] <klaxa> seriously? [23:23] <shevy> yes [23:23] <klaxa> and what makes you think so? [23:23] <shevy> klaxa I used it [23:23] <klaxa> sox is a great open source project [23:23] <klaxa> like what... once? [23:23] <shevy> klaxa like many times [23:23] <klaxa> so you can't use it and therefore it's bad? [23:23] <shevy> ffmpeg + mplayer have things well covered [23:23] <klaxa> then don't touch ffmpeg [23:24] <shevy> why [23:24] <klaxa> don't touch any software whatsoever [23:24] <shevy> what does ffmpeg have to do with sox? [23:24] <shevy> ffmpeg is very, very good [23:24] <klaxa> what does you not being able to use the program have to do with its quality? [23:24] <shevy> klaxa if something is shit, it's shit [23:24] <klaxa> you are not providing any evidence [23:24] <shevy> klaxa I gave you one specific example so far [23:25] <klaxa> >I used to think it had a great effect, but it sucked! [23:25] <klaxa> well what did you do to the file? [23:25] <shevy> yes, the fancy effect [23:25] <shevy> It was some morph effect, some "fade" related action [23:25] <klaxa> actually i don't want to discuss this [23:26] <bffmpeg> http://pastebin.com/hGSZtker ffmpeg generated NTSC-DVD.mpg file chokes dvdauthor - any insights? Primary error is WARN: Discontinuity of nnn in audio channel 0; please remultiplex input. [23:51] <skifreak> having a problem with ffmpeg recognizing the codec for a right bluray, passthrough, video file. mediainfo shows very similar information between the right and left video files, with only the left containing audio. Tried increasing probesize and analyzeduration to no avail. ffmpeg output: http://dpaste.com/1397661/ mediainfo output: http://dpaste.com/1397663/ I have a sample file (10 MB) of each ready on dropbox if needed [00:00] --- Sat Sep 28 2013

1 0

ffmpeg-devel.log.20130926
by burek 27 Sep '13

27 Sep '13

[00:00] <kolosos> michaelni: upload of 720p mp4 version here : http://www.sendspace.com/file/iuejix [00:00] <kolosos> downloaded from Youtube [00:03] <kolosos> oh nice, sendspace wants to download a .exe [00:15] <kolosos> <JEEBsv> http://johnvansickle.com/ffmpeg [00:16] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:75b1b13aff73: vc1: check mb_height validity. [00:16] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:15620c153a35: ogg: Always alloc the private context in vorbis_header [00:16] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:896baaaad85c: segafilm: Error out on impossible packet size [00:16] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:91355bec88ce: h261: check the mtype index [00:16] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:86c169c5b691: dxa: Make sure the reference frame exists [00:16] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:3dff283de11a: ac3: Do not clash with normal AVERROR [00:16] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:37e69e2dee7c: ac3: Clean up the error paths [00:16] <cone-549> ffmpeg.git 03Michael Niedermayer 07release/0.10:c6a4397410ab: Merge commit '37e69e2dee7c5167083bb42d669f73f038111a79' into release/0.10 [00:17] <kolosos> JEEBsv: i tried ./ffmpeg-2.0.1-64bit-static/ffmpeg -i jump720.mp4 -target ntsc-dvd jump720m.mpg same buffer issue [00:26] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:c225c620c664: ac3: Return proper error codes [00:26] <cone-549> ffmpeg.git 03Anton Khirnov 07release/0.10:5773065a7105: pictordec: break out of both decoding loops when y drops below 0 [00:26] <cone-549> ffmpeg.git 03Anton Khirnov 07release/0.10:8d2a86a29055: lavf: avoid integer overflow when estimating bitrate [00:26] <cone-549> ffmpeg.git 03Anton Khirnov 07release/0.10:23f73fc241da: ape demuxer: check for EOF in potentially long loops [00:26] <cone-549> ffmpeg.git 03Martin Storsjö 07release/0.10:068c86728668: matroskadec: Check that .lang was allocated and set before reading it [00:26] <cone-549> ffmpeg.git 03Michael Niedermayer 07release/0.10:7a9af1da3900: mjpegb: Detect changing number of planes in interlaced video [00:26] <cone-549> ffmpeg.git 03Martin Storsjö 07release/0.10:5473d23ece9e: mpegvideo: Avoid 32-bit wrapping of linesize multiplications [00:26] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:b0ca5fef09d1: dv: Add a guard to not overread the ppcm array [00:26] <cone-549> ffmpeg.git 03Michael Niedermayer 07release/0.10:c8ca38567669: Merge commit 'b0ca5fef09d1b1268ea0c8f89bf53cd38aaa85e7' into release/0.10 [00:36] <cone-549> ffmpeg.git 03Anton Khirnov 07release/0.10:9978c24abfbb: lavf: fix the comparison in an overflow check [00:36] <cone-549> ffmpeg.git 03Michael Niedermayer 07release/0.10:d8af960e7eba: Merge remote-tracking branch 'qatar/release/0.8' into release/0.10 [00:46] <kolosos> <kolosos> Here's a video that gives the buffer error: http://www.youtube.com/watch?v=yqGcZ6WBZnY and here is same one (720p flv version) that i downloaded that gives buffer error: http://shell2.reverse.net/~kaht/singlejumps.flv [03:14] <kolosos> Result of uname -a ; wget shell2.reverse.net/~kaht/singlejumps.flv ; ffmpeg -i singlejumps.flv -target ntsc-dvd singlejumps.mpg Check http://http://pastebin.com/mgfKmD7M [03:15] <kolosos> ^^^^^^ buffer underflow result of ffmpeg on certain youtube videos such as http://www.youtube.com/watch?v=yqGcZ6WBZnY [03:21] <kolosos> GAH the right buffer underflow link is http://pastebin.com/mgfKmD7M [03:27] <Compn> ok [03:27] <Compn> kolosos : what ver ffmpeg ? [03:27] <Compn> 1.2.2 eh [03:28] <cone-549> ffmpeg.git 03Michael Niedermayer 07master:ddd0e23d587e: avformat/mpegenc: improve field names in underflow error message [03:28] <cone-549> ffmpeg.git 03Michael Niedermayer 07master:b63711808619: avformat/mpegenc: change assert to av_assert0 [03:29] <cone-549> ffmpeg.git 03Michael Niedermayer 07master:930406325c29: avformat/mpegenc: Better default for max_delay [03:29] <cone-549> ffmpeg.git 03Michael Niedermayer 07master:32cde9629693: avformat/mpegenc: Better heuristic for ordering packets [03:29] <michaelni> kolosos, should be fixed [03:34] <kolosos> michaelni: fixed where? [03:34] <kolosos> git? [03:35] <kolosos> release/0.10:d8af960e7eba: ?? [03:36] <kolosos> sheesh, this could easily make ffmpeg the most responsive software updates i have ever seen [03:42] <michaelni> kolosos, ffmpeg master:HEAD [03:45] <kolosos> oh i see, there's a separate git on ffmpeg.org [10:13] <durandal_1707> anybody working on xll dts extension? [10:14] <nevcairiel> not that I am aware, its apparently quite annoying to support [10:18] <durandal_1707> apparently? [10:42] <cone-843> ffmpeg.git 03Kostya Shishkov 07master:4370f65be8d7: g2meet: Respect cursor_stride properly everywhere [10:42] <cone-843> ffmpeg.git 03Michael Niedermayer 07master:1906e00030b3: Merge remote-tracking branch 'qatar/master' [11:21] <nevcairiel> Daemon404: ever since i updated my fate box to your make, i have one fate run which *always* segfaults during building, at different spots though, but only this one, and everytime [11:32] <Daemon404> nevcairiel, no idea [11:32] <Daemon404> sounds like a job server problem [11:33] <Daemon404> go go make [11:33] <nevcairiel> looks exactly like the old crash that happened rather randomly, except that now it seems to happen more often :p [11:33] <Daemon404> nice. [11:33] <durandal11707> faulty hardware? [11:33] <Daemon404> nah i used to get it randomly too [11:34] <nevcairiel> i reverted my fate box to the old make, lets see if it just passes agian [12:27] <motowidlo> hi [12:27] <motowidlo> anyone can help regarding use of ffmplayer on multimonitor environment ? [12:32] <durandal11707> motowidlo: what is ffmplayer? [12:32] <BBB> j-b: what did I do? [12:32] <BBB> j-b: *hug* back btw :) [12:33] <motowidlo> i mean ffplay [12:37] <Daemon404> probably vp9 [12:37] <Daemon404> motowidlo, afaik nobody actually uses ffplay to do anything other than basic testing [12:39] <motowidlo> Daemon404, looking at the source code, too, so I think. :) [13:05] <Compn> motowidlo : mplayer supports multi monitor stuff with -xineramascreen option [13:14] <motowidlo> Compn, MS/win world ? [13:27] <Compn> for windows, mplayer can use the -vo directx:display or :adapter to select the screen [13:27] <Compn> or maybe can just stretch it if you extend desktop [13:27] <Compn> i forgot which option. you should ask in #mplayer or read the manual [13:28] <motowidlo> ok, thx, i'll check it [14:20] <cone-63> ffmpeg.git 03Michael Niedermayer 07master:cd05c4060a2f: avformat/mpegenc: minor simplification / use AV_WB16() [14:21] <Daemon404> i thought direct use of AV_RB/WB was discouraged [14:24] <ubitux> aligned/unaligned? [14:30] <Daemon404> ubitux, safe bytereader [14:30] <Daemon404> probably 'too slow' or something [14:50] <iive> huh? [15:04] <durandal11707> shouldn't AV_WB do same what replaced code does? [15:07] <cone-63> ffmpeg.git 03Michael Niedermayer 07master:e02495372240: avformat/wavdec: check smv_frames_per_jpeg [15:12] <nevcairiel> Daemon404: well muxing seems to be less prone to such issues, since you control the buffer and the values you write, not sure if there really is much that uses a "safe byte writer" [15:12] <Daemon404> true [15:13] <nevcairiel> AV_WB is at least in many many places [15:13] <nevcairiel> +used [15:14] <ubitux> any idea where the dvb subtitles are standardized? [15:15] <Daemon404> ... i assue at dvb [15:15] <Daemon404> :V [15:15] <Daemon404> http://www.dvb.org/standards [15:16] <ubitux> ah cool it's open great [15:16] <ubitux> thx Daemon404 [15:18] <ubitux> the subtitles paper doesn't seem to cover much the muxing though [15:19] <Daemon404> ofc [15:19] <nevcairiel> the dvb blue book there has details on the PES format in mpegts, if thats what you need [15:20] <ubitux> i'd like to understand wtf is wrong with subtitles muxing into mpeg-ts in ffmpeg [15:20] <ubitux> (#2989) [15:20] <ubitux> it might have work at some point actually [15:20] <nevcairiel> look at the bluebook then, it might have the info [15:20] <ubitux> ok [15:20] <ubitux> thanks [15:20] <nevcairiel> syntax tables of the structure in PES etc [15:22] <ubitux> there are a bunch of them [15:50] <ubitux> erm afaict ffmpeg was never really able to mux those subs [15:50] <ubitux> at least not with that kind of input [15:50] <ubitux> i wonder in what situation this was supposed to work [16:59] <ubitux> what is the old equivalent of ffmpeg -i in.ts -c copy -map 0:0 -map 0:5 out.ts, assuming 0:0 video and 0:5 dvbsub? [17:00] <ubitux> -c copy -scodec copy -vcodec copy, but what about the rest? [17:00] <Compn> you mean the map isnt working correctly ? [17:02] <ubitux> yes [17:02] <ubitux> i probably need some -new* or sth [17:04] <Compn> what ffmpeg ver ? [17:04] <ubitux> 0.7.x [17:04] <ubitux> Number of stream maps must match number of output streams [17:04] <ubitux> http://pastie.org/8357515 [17:04] <Compn> i think you want just the one -map 5:5 [17:05] <ubitux> huh? [17:06] <Compn> you want one ts file output with video at 0 and dvsub at 5 ? [17:06] <ubitux> yes [17:06] <ubitux> i want to pick 0:0 and 0:5, stream copy them [17:06] <Compn> i dont even remember if ts muxer had subtitle copy back then [17:07] <ubitux> ow.. [17:07] <Compn> ehe [17:17] <ubitux> michaelni: 04704a7881d94f8fc4c0e9abe609c98cb573776f [17:18] <ubitux> michaelni: who am i supposed to contact, except you? [17:18] <ubitux> mail from the frontpage? [17:24] <michaelni> ubitux, i dont know :( [17:24] <michaelni> just checke my inbox [17:25] <michaelni> but i cant find anything so i dunno who sent me that link [17:35] <ubitux> Compn: it seems old ffmpeg has stream copy, so it should work [17:35] <ubitux> but that's the problem here afaict [17:35] <ubitux> i just have no idea how the cmd line is supposed to look like :( [17:37] <durandal11707> remove dead links ASAP [17:40] <ubitux> anyway, when i try -newsubtitle/-newvideo, i get a "At least one output file must be specified" [17:40] <ubitux> as if it wasn't detecting the output anymore [17:40] Action: ubitux wonder what the hell is going on [17:42] <ubitux> http://pastie.org/8357610 [17:42] <ubitux> any idea? [17:45] <Armada> Is there anyone available to help us solve this regression? http://trac.ffmpeg.org/ticket/2962 [17:47] <durandal11707> ubitux: why you use so old software? [17:47] <ubitux> durandal11707: i want to check if the problem i have is a regression [17:47] <ubitux> it sounds really weird that no one ever tested dvb muxing in mpegts [17:47] <ubitux> so at some point it might have work [17:48] <ubitux> and here i'm just trying to achieve a remuxing with older versions [17:49] <durandal11707> if it ever worked it was by accident [17:49] <ubitux> well there are some muxing code [17:50] <ubitux> dvb seems to be only available in mpegts, so it's really weird if it never worked [17:50] <durandal11707> no coverage - no fate [17:50] <ubitux> they would have wrote a dvb sub encoder if thye weren't able to mux them [17:53] <cone-63> ffmpeg.git 03Michael Niedermayer 07master:5dc6c0ea9ee7: avformat/mpegts: check avio_tell() return code before using it [17:53] <durandal11707> anyone need this: https://github.com/richardpl/FFmpeg/tree/rgain ? [17:54] <ubitux> ah, an alternative to ebu? :) [17:55] <Armada> michaelni: thanks! that fixes our blocker [18:05] <ubitux> im not even able to ffmpeg -i in.ts -vcodec copy -map 0.0 out.ts [18:06] <ubitux> -newvideo doesn't help ("Number of stream maps must match number of output streams" "At least one output file must be specified" if i add it) [18:06] <ubitux> archeology is hard :( [18:06] <durandal11707> you lose more time than fixing issue itself [18:07] <ubitux> probably [18:07] <ubitux> but now i'm curious about that specific problem more than the original issue [18:11] <ubitux> oh, it seems -sn -an actually helps [18:14] <ubitux> ok so this was what i was looking for: ./ffmpeg -y -i in.ts -vcodec copy -scodec copy -an -map 0.0 -map 0.5 -y out.ts [18:14] <ubitux> and... 0.7 has the proble [18:14] <ubitux> problem [18:15] <durandal11707> who is admin on ffmpeg-user ? [18:35] <durandal11707> so anyone needs replaygain filter or not? [18:36] <wm4> wut [18:36] <wm4> yes here [18:36] <wm4> no idea how to test it though [19:47] <rcombs> is there any way to access all input file streams from an avfilter? [19:50] <rcombs> I've got a video filter here, and I need to access the attachments on the input file as soon as they're available [21:04] <TheRyuu> nevcairiel: does LAVFilter just use the standard ffmpeg stuff to open the files for playback? [21:04] <Daemon404> what do you mean [21:05] <Compn> Daemon404 : he means lavf demuxers [21:06] <Compn> TheRyuu : are you talking about libavfilter or some other 3rd party project which has nothing to do with ffmpeg development ? [21:06] <Daemon404> lav [21:07] <Daemon404> as n nevcairiel's thing [21:07] <Compn> annoying another related name to confuse everyone [21:07] <TheRyuu> https://github.com/Nevcairiel/LAVFilters <---this [22:01] <cone-63> ffmpeg.git 03Michael Niedermayer 07master:f31011e9abfb: avcodec/parser: reset indexes on realloc failure [00:00] --- Fri Sep 27 2013

1 0

ffmpeg.log.20130926
by burek 27 Sep '13

27 Sep '13

[00:18] <kolosos> JEEBsv: see -devel [02:17] <goiken> Im trying to get ffserver to work. I can manage to get ffserver running without any complaints. When I try to connect a source, however, it appears that Im not sending any data. Heres the details of what I did: On the remote host (monoceres.uberspace.de) i ran `ffserver -d -f etc/ffserver.conf`. My config is here: http://dimde.monoceres.uberspace.de/ffserver.conf [02:17] <goiken> On my local machine I then ran `./ffmpeg/ffmpeg -f v4l2 -s 320x240 -r 25 -i /dev/video0 http://monoceres.uberspace.de:63894/feed1.ffm`. When I do this, my webcams lamp turns on and the server outputs something about an incomming connection. When I replace /dev/video0 with a file, I face the same problem: Requesting 'http://monoceres.uberspace.de:63894/live.mpg' with vlc (from my local machine) gives me an empty stream. [02:30] <elBradford> I'm trying to statically build ffmpeg with librtmp for android [02:31] <elBradford> I've built rtmpdump, and have a directory with librtmp.a and librtmp.so [02:32] <elBradford> When I build I get errors such as "/tmp/FFMPEG/bin/../lib/gcc/arm-linux-androideabi/4.6/../../../../arm-linux-androideabi/bin/ld: libavformat/librtmp.o: in function rtmp_get_file_handle:libavformat/librtmp.c:190: error: undefined reference to 'RTMP_Socket'" [02:45] <elBradford> It actually looks like I might be missing header files... ? [03:04] <elBradford> Does anyone know why ffmpeg would claim it can't find librtmp when pkg-config reports that it's available? [03:06] <iive> elBradford: check the config*.log file [03:21] <elBradford> Thanks iive - getting a warning: WARNING: arm-linux-androideabi-pkg-config not found, library detection may fail. It's looking for a platform-specific pkg-config? [03:22] <elBradford> Other than that, config.log looks OK... [03:27] <iive> elBradford: are you doing native build or cross ? [03:27] <iive> anyway... gtg. n8 ppl. [03:32] <elBradford> I'm looking for help cross compiling ffmpeg with Android with librtmp. I'm offering a cash bounty. I'm following the following script to build: https://github.com/yixia/FFmpeg-Android - if you're interested email me at b.law [at] outlook [dot] com [03:46] <elBradford> To be more specific, I'd pay$50-$100 for someone to provide working examples of using the script I linked, but with librtmp support. Get in touch if you're interested! Bradford [03:55] <sisco> hi guys how you doing [03:57] <sisco> i would like to know what's the best command line for converting hls to rtmp , because it seems to me (im newbie i know ) to get it work as well its always cutting and freezing but when i play the m3u8 with flash player it's smooth so plz any advice ? for a good command? [04:02] <sisco> hellooooow folks [04:02] <sisco> any one here can help? i posted my question [04:02] <sisco> plz any advice [09:48] <lishkay> How can the time scale (sampling interval) of audio be changed without changing the sample values? [10:00] <roger21> hey [10:00] <roger21> is there a simple way to remove all the metadata when transcoding [10:00] <roger21> ? [10:00] <roger21> like title, lang etc [10:52] <kode54> bah, he left [10:55] <lishkay> Since sample rate is a metadata, I presume changing it involves some functions to handle metadata. [11:20] <roger21> i transcode a 16/9 file to 720x480 and set the ar to 19:9 but my player display it in 720:480 (by default) [11:20] <roger21> while other 720:480 files are displayed in 16:9 by default [11:20] <roger21> what did i miseed? [11:20] <roger21> 16:9* [11:22] <roger21> is there something else to set [11:32] <excalibas> Hello, I want to compile ffmpeg on Ubuntu applying this patch: https://github.com/original-male/FFmpeg/commit/d02509d04d396a98646ca81e9ba3… [11:32] <excalibas> But I am a bit of a noob. Can someone please help? [11:33] <viric> that depends on the kind of help you need [11:33] <excalibas> Do I need to uninstall ffmpeg first? If I try Ubuntu wants to uninstall lots of other software :( [11:34] <viric> I don't think so [11:34] <excalibas> Thanks viric. Next question. How do I apply the patch? [11:35] <viric> curl URL | patch -p1 [11:35] <excalibas> This is inside ffmpeg source folder? [11:35] <viric> yes [11:36] <excalibas> And last question (I hope) How do I compile with vorbis and h264 support? [11:38] <viric> check for libvorbis and libx264 flags, with "configure --help" [11:38] <spaam> if you want to encode it [11:38] <spaam> you can decode vorbis and h264 by default [11:39] <excalibas> I want to encode. My quest is to make a screencast with audio coming from jack [11:40] <excalibas> I found out it is not so easy but I have hopes to make it work by compiling ffmpeg [11:40] <excalibas> Thank you [11:46] <excalibas> ok, problem 1, I for vorbis I see --enable-libvorbis but for h264 I just find an option to enable decoding [11:47] <excalibas> Do I need to install something? [11:48] <spaam> as i said . you dont need to enable vorbis or h264 if you want to decode it.. if you want to encode it. you need to do what viric said [11:49] <spaam> 113803 < viric> check for libvorbis and libx264 flags, with "configure --help" [11:49] <excalibas> spaam, I think I need to encode [11:50] <excalibas> With configure -- help I see --enable-libvorbis but for h264 I just find an option to enable decoding [11:51] <excalibas> --enable-libstagefright-h264 enable H.264 decoding via libstagefright [no] [11:51] <JEEB> if you need H.264 encoding you need libx264 [11:51] <JEEB> so --enable-gpl and --enable-libx264 [11:51] <JEEB> this of course means you need libx264 as well [11:52] <excalibas> :) Many Thnaks [11:58] <excalibas> I think it is working. one more question. If I do make install it installs over the ubuntu instalation or I dont make install and use from this folder? [12:02] <JEEB> the default prefix should be /usr/local so it shouldn't overwrite the libav ffmpeg binary installed by the repositories, but if you want to specify specifically where you want the stuff to go, you can configure with --prefix=/your/prefix/yo/ [12:03] <highgod> Hi, I want to ask a question that, I have got the sps and pps information from the input video, and I want to use the sps and pps information to init the libx264 encoder so that I can get the same sps and pps, how can I implement it? I know how to set the profile and level but how about the other parameters? I don't konw [12:03] <highgod> Can someone give anyhelp? [12:03] <excalibas> JEEB, And If I dont make install and use from the ffmpeg folder it also works? [12:05] <JEEB> yup [12:23] <excalibas> Oh, When I try ffmpeg -fflags +genpts+igndts -f x11grab -vsync 0 -r 30 -s 1440x900 -i :0.0+0,0 -vcodec h264 -f jack -ac 2 -r:a 48000 -i screencast -acodec pcm_s16le -r:v 30 -vsync 2 -async 1 -map 0:0,1,0 -map 1:0 -preset ultrafast -qp 0 aaa.mkv It says: Option framerate not found. [12:24] <excalibas> Any ideas? [12:36] <excalibas> Ok, I think I was trying with ffmpeg from Ubuntu [12:48] <lishkay> Can FFMPEG take 2 input videos and create an output video with one of the inputs above the other? How? I wish to do this to visually compare 2 inputs. [13:14] <durandal11707> lishkay: see docs for overlay filter [14:54] <lishkay> Thanks for pointing out which section (overlay filter). [15:24] <diroots> hi there [15:24] <spaam> everyone hide! diroots is here [15:24] <spaam> :D [15:24] <spaam> diroots: wzup? :) [15:25] <diroots> hi spaam [15:25] <diroots> ...encoding problems :p [15:25] <diroots> when encoding from DVCPRO_HD to H264, I have a strange behavior : quality is very ugly for the 10-15 first seconds, then, for all files, quality is as expected. Do one of you already had this problem? [15:26] <diroots> is this because of one-pass encoding? [15:27] <lishkay> durandal11707: Are you sure? I have been unable to find any option to offset or relatively position the video so that one is above the other. When I write "above", I do not mean "in front of". [15:27] <diroots> maybe it's an already reported bug, or am I the first to meet such a strange thing? [15:28] <spaam> diroots: have you tried some other codec? :) [15:29] <diroots> spaam: I need to solve with these codecs... [15:31] <spaam> diroots: if you try with som other codec we can see if its a x264 problem or DVCPRO_HD problem [15:31] <spaam> maybe the decoder is wrong [15:32] <diroots> so I can only test to encode to another codec, as I will always have DVCPRO_HD files as source [15:34] <diroots> spaam: which compressed codec could I test, which would be equivalent to x264? [15:35] <klaxa> it doesn't need to be equivalent [15:35] <klaxa> just use whatever, libtheora or mpeg4 [15:39] <durandal11707> lishkay: well you use comination of overlay and pad filter.... [15:40] <durandal11707> *combination [15:49] <smjd> diroots: what settings are you using? [15:51] <diroots> smjd: -vcodec libx264 -pix_fmt yuv420p -b 6900k -profile main -coder 0 -g 24 -keyint_min 2 -x264opts colorprim=bt709:transfer=bt709:colormatrix=bt709:ref=2 -sc_threshold 0 -b_strategy 0 -refs 2 -bf 0 -ac 2 -ab 320k -ar 48k [15:51] <diroots> spaam: klaxa : I tried to encode to flv, then with mpeg4, and it seems that there is still the problem dring the first 10seconds [15:53] <spaam> diroots: okey. then it might be a problem with the decoder or the file it self :) [15:53] <diroots> spaam: would this sound like it's the DVCPRO_HD decoder? [15:53] <diroots> ok [15:54] <diroots> this is with all files given... they did not notice it, as only some of their files have movements during these first seconds [15:54] <smjd> how does the input file look with ffplay? [15:55] <diroots> I don't know as it is remote. I call the person to ask how it looks, and she looks with quicktime pro (she's on a mac, and files are sent through final cut pro. [15:57] <diroots> and as the ffmpeg version installed is 6 month old, maybe by updating it to latest version, the dvcpro decoder might have been improved... right? [15:59] <diroots> smjd: I gave you my settings, and these are used within a single pass conversion. Do you think I can get improved conversion, is using a 2 pass encoding session? [15:59] <diroots> or it would not change anything? [16:02] <smjd> I don't really know enough details about H264 to say much to that [16:08] <smjd> I'm not a professional, so I don't know about the limitations of specialized hardware, so presets are mostly enough for me [16:13] <smjd> but from what I know about ABR, it seems logical for two-pass to fix too little bitrate allocated to the first few seconds [16:15] <diroots> yes, my first try will be to try 2 pass encoding, then if I get the same result, i'll try to upgrade my ffmpeg, in case the is an improvement in decoding DVCPRO [16:17] <mister_zombie> Hi, I use ffmpeg to capture images from mp4 videos, I wondered if there was any way to make those jpeg images interlaced (aka progressive jpeg) [16:17] <mister_zombie> ? [17:47] <smjd> mister_zombie, but you might be able to convert the jpg files losslessly to the progressive format with jpegtran [17:48] <smjd> mister_zombie: maybe not, but you can convert the jpg files losslessly to the progressive format with jpegtran* [17:48] <elBradford> I'm looking for help cross compiling ffmpeg with Android with librtmp. I'm offering a cash bounty. I'm following the following script to build: https://github.com/yixia/FFmpeg-Android - if you're interested email me at b.law [at] outlook [dot] com [17:49] <mister_zombie> smjd: And it seems I have jpegtran installed on my server already! Nice! [18:00] <defaultro> hey guys, this video was made by ffmpeg last night :) http://www.youtube.com/watch?v=NSJBYi8RdHs&hd=1 watch it in 1080p [18:01] <durandal11707> yes and? what it have? [18:02] <defaultro> is that for me durandal11707 ? [18:03] <smjd> it has an intro spanning 45% of the video [18:03] <defaultro> yeah, I think I put too much [18:03] <defaultro> it's my first try [18:03] <defaultro> i need to learn the accepted ways of making videos [18:04] <defaultro> recommended ways [18:12] <elBradford> vid looks good defaultro [18:40] <viric> I never know how much kbps to target [18:50] <defaultro> thanks elBradford :) [19:15] <smjd> viric -b:v `echo FILESIZE_IN_MB*137/MINUTES|bc`k [19:16] <smjd> replace FILESIZE_IN_MB and MINUTES with something [19:16] <viric> 137? [19:16] <smjd> 8192/60 [19:16] <viric> ah ok [19:17] <viric> well, I don't have a definite file size in mind. [19:18] <viric> does anyone use mlt here? [19:22] <smjd> maybe MB*140000/MINUTES would be a bit more accurate, [19:22] <smjd> what's mlt? [19:23] <viric> melt [19:23] <viric> it makes it easy to concatenate videos with fadein/fadout and overlap [19:45] <lishkay> I'm merging with overlay 2 video streams of different frame rates. They are desynchronizing. How can I keep them synchronized in the output? [20:41] <lishkay> I have found the fps filter. That might do it. [20:51] <lishkay> No, it does not. [20:58] <hendry> hi, I'm trying to make a video with lots of images. e.g. ffmpeg -i img%04d.webp -s 1440x1080 out.avi, however I want to show the time in the top right corner. How can I do that? watermark? [21:00] <chrisballinger> hey if I can't use sizeof(AVStream) how can I malloc an array of AVStreams? [21:02] <lishkay> https://ffmpeg.org/trac/ffmpeg/wiki/FilteringGuideBurnt in Timecode [21:02] <lishkay> hendry: https://ffmpeg.org/trac/ffmpeg/wiki/FilteringGuide | Burnt in Timecode [21:23] <denysonique> ALSA buffer xrun. [21:23] Last message repeated 2 time(s). [21:23] <denysonique> I get when: avconv -f alsa -ac 2 -i hw:0,0 -f x11grab -r 30 -s 1280x800 -i :0 -vcodec huffyuv -acodec pcm_s16le -threads 1 video.mkv [21:33] <hendry> denysonique: in my experience that means your CPU isn't beefy enough [21:33] <hendry> denysonique: have a look at https://github.com/kaihendry/recordmydesktop2.0/blob/master/x11capture [22:00] <trem0r> Hey everyone. I have a wav file that needs to be converted into another wav with 16 kpbs linear PCM encoding [22:01] <trem0r> How can I accomplish this? [22:10] <denysonique> hendry: I have i5 [22:12] <denysonique> hendry: also, don't forget to add support for ALSA input without pulseaudio [22:12] <denysonique> not everyone is on n00buntu. [22:13] <denysonique> lol, hendry I can hear you thinkpad keyboard. [22:14] <viric> Can I add any 'id3tags' or so to webm? [22:14] <viric> metainformation [22:16] <lishkay> Using -r on the input does at least approximately what i want. [22:22] <hendry> denysonique: i use archlinux. pulseaudio is the future [22:22] <hendry> denysonique: sadly [22:24] <denysonique> hendry: on my gentoo there is no need to bloat my system with pulseaudio. ;> [22:33] <hendry> denysonique: i really thought the way you did... see previous versions. But pulseaudio is the only sane way to do complex recordings with say application audio output and microphone output [22:34] <denysonique> hendry: in this case yes [22:39] <andrew_> how do i get ffmpeg to write to the same file it read from? can i just do ffmpeg -i file.mp3 file.mp3 [22:40] <viric> you can't do that [22:40] <viric> well, you should not [22:40] <viric> that will probably destroy your file. [22:41] <andrew_> basically i have a bunch of music thats missing a tag and i want to just add that metadata [22:41] <andrew_> im not doing any transcoding [22:41] <viric> but I think ffmpeg can't "modify" a file. From some, will create others. [22:44] <andrew_> like i want to preserve the filename [22:45] <andrew_> would something like this work: "for file in *; do mv "$file" tmp; ffmpeg -i temp -metadata disc=1 "$file"; done [22:57] <Kirito> There's no alternative for -t that tells it to "end" at a specific time is there? [22:57] <viric> Kirito: -to [22:58] <Kirito> Ahh, thank you, that makes my life so much easier [22:59] <viric> it is a bit recent. :) [23:00] <Kirito> That explains why I couldn't find anything last time I looked.. ages ago at least. Thank you >< [23:03] <viric> you are welcome [23:04] <chrisballinger> hey all [23:04] <chrisballinger> when manually doing a bitstream copy with libavformat all of my video pts/dts are 0 [23:12] <viric> is it possible to encode to VP9? [23:13] <viric> encoding with libvpx gives vp8 [23:56] <mittens> try libvpx-vp9 [23:56] <chrisballinger> how do you get the proper PTS/DTS of a h.264 video stream AVPacket in an mp4 container? [00:00] --- Fri Sep 27 2013

1 0

ffmpeg-devel.log.20130925
by burek 26 Sep '13

26 Sep '13

[00:08] <BBB> michaelni__: that's a very sad solution, I'm trying to slowly update the whole tree to use ptrdiff_t where appropriate... [00:08] <BBB> michaelni__: I really do think it makes more sense longer-term, as much as it's kind of a pain in the transition period [00:08] <BBB> x264 did it too, they profit every day [00:08] <BBB> and yay I finally have real internet [00:12] <Skyler_> x264 uses intptr_t (I have no opinion on the ptrdiff vs intptr thing though) [00:12] <BBB> I know :) [00:12] <Skyler_> ah, okay [00:12] <BBB> my point was you're not using int and suffer horribly every day [00:15] <iive> BBB: you had not google fiber until now? [00:15] <BBB> I don't think new york city has google fiber [00:16] <BBB> (and yes I don't live in california anymore, me+family moved back) [00:16] <iive> :( [00:22] <kierank> BBB: did you leave google [00:22] <BBB> yes [00:22] <Compn> shh dont say it :D [00:24] <Compn> the g word [00:24] <Compn> BBB : neat, now you can tell us all of the secrets :P [00:25] <gnafu> Ooh. [00:28] <iive> BBB: did you start working of ffvp9 before or after you left google? [00:31] <cone-906> ffmpeg.git 03Michael Niedermayer 07master:c88ac1e0233f: avcodec/g2meet: fix regression with rgb cursors [00:33] <gnafu> And what are you going to do now (aside from going to Didney Worl)? [00:36] <Compn> curious if he likes cali or ny better :P [00:39] <BBB> Compn: ny ofcourse [00:39] <iive> moving from one state to the other is no cheap or easy endeavor. [00:40] <BBB> iive: I did ffvp9 while I was on leave between jobs, I did some work while at google and on leave I was officially still employed by google but it was sort of my private time [00:40] <BBB> I finished a few bugs after starting my new job, but not a lot [00:40] <BBB> gnafu: see linkedin ;) I work for http://www.twosigma.com/ now [00:41] <Compn> now to make xvp9 encoder and make bank ? [00:41] <Compn> too many makes [00:41] <Compn> hmm [00:42] <iive> so you are now in micro-transaction business? [00:42] <kierank> BBB: heh you gave in like our old x264 gsoc student [00:43] Action: kierank did an internship at one of these places a few years ago [00:43] <iive> or how was the name.. of these automated stock trading systems ... [00:43] <iive> high frequency trading. [00:50] <BBB> kierank: ? [00:50] <kierank> one of our old gsoc students went into a firm like yours [00:50] <cone-906> ffmpeg.git 03Michael Niedermayer 07master:bb4b041df351: avcodec/mpegvideo_enc: check qmin/qmax [00:50] <BBB> ah [00:50] <BBB> also I don't do high frequency trading [00:50] <BBB> it's a hedge fund primarily, not a hft firm [00:54] <iive> i see. [00:55] <BBB> kierank: which student? :) [00:55] <kierank> dylan [00:56] <kierank> i forget his surname [00:56] <kierank> wrote weightp [00:56] <BBB> ah, cool [00:59] <Compn> BBB : lemme know good stock tips :) [00:59] <Compn> if you find any [00:59] <BBB> you know I'm not allowed to do that :) [00:59] <Compn> even if you hear about them in a newspaper or tv broadcast? [01:00] <BBB> how would I, you, or regulators, know the difference? [01:00] <Compn> not asking you to do any illegal insider trading :P [01:00] <BBB> safer I don't say anything :) [01:01] Action: Compn facepalms [01:02] <kierank> there were a lot of strict rules when i interned as well [01:02] <kierank> it was taken very seriously [01:03] <Compn> so is that health company google is starting [01:03] <Compn> is that going to be a public company under a different stock name ? [01:03] <Compn> i wonder if that would be a good investment [01:20] <iive> Compn: i'll tell you a secret. stocks picked at random give same returns as these picked by skillful brokers. [01:21] <iive> and maybe a little better [01:23] <Compn> thats why i'm aksing for tips, from people in the technology field [01:23] <Compn> new startups etc [01:28] <iive> Compn: i guess you've read this before - http://www.newscientist.com/article/mg21228354.500-revealed--the-capitalist… ? [01:29] <iive> the stock market is basically a casino and the odds are always in favor of the bank :) [02:18] <cone-906> ffmpeg.git 03Michael Niedermayer 07master:65bf9a44d7b0: avcodec/pngdec: check for stream end in png_decode_idat() [02:39] <Compn> iive : yes i know. :( [02:39] <Compn> another research paper using zzuf fuzzer :) [02:43] <Compn> hmm [02:43] <Compn> 60gb / 50kb/s ?\\ [02:44] <Compn> argh enter key broken [02:44] <Compn> 13 days to download all samples repo? [02:45] <iive> Compn: i guess you are not with google fiber either :| [02:47] <Compn> 50kb/s is reccomended speed in the readme [02:50] <Compn> hmm what was the fuzz project that gave ffmpeg all of those automatically generated fuzz crash reports? [02:51] <cone-906> ffmpeg.git 03Michael Niedermayer 07release/0.5:fde0b7d91c9c: avcodec/rpza: Perform pointer advance and checks before using the pointers [02:51] <cone-906> ffmpeg.git 03Michael Niedermayer 07release/0.5:31f9e849a88b: matroska_read_seek: Fix used streams for subtitle index compensation [02:51] <cone-906> ffmpeg.git 03Michael Niedermayer 07release/0.5:e7484d54252d: avcodec/dsputil: fix signedness in sizeof() comparissions [02:51] <cone-906> ffmpeg.git 03Michael Niedermayer 07release/0.5:617a9eedc654: avcodec/ffv1enc: update buffer check for 16bps [02:51] <iive> oh.. [02:55] <cone-906> ffmpeg.git 03Michael Niedermayer 07release/0.5:b012da4019e3: update for 0.5.13 [02:57] <iive> cu have fun [02:58] <Compn> hmmmm [02:58] <Compn> wait, gentoo has a test flag for libav that includes video samples? [02:59] <Compn> we dont own the copyright on our samples repo... [02:59] <Compn> seems odd to see a distro distributing copyrighted video clips :) [03:00] <Compn> if indeed i am reading this correctly [03:13] <cone-906> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n0.5.13': unknown revision or path not in the working tree. [03:13] <cone-906> Use '--' to separate paths from revisions [03:13] <cone-906> refs/tags/n0.5.13:HEAD: avcodec/pngdec: check for stream end in png_decode_idat() [03:17] <Compn> ooo broke the cone [06:18] <Compn> noooooo, cone is down [09:58] <cone-867> ffmpeg.git 03Anton Khirnov 07master:668643b9239c: matroskadec: check av_strdup() when setting defaults [09:58] <cone-867> ffmpeg.git 03Michael Niedermayer 07master:57070d9892ad: Merge commit '668643b9239c70b777aea322eab411ebc960d9a0' [10:12] <cone-867> ffmpeg.git 03Anton Khirnov 07master:e880418660c8: cabac: remove write-only h264_mps_state[] [10:12] <cone-867> ffmpeg.git 03Michael Niedermayer 07master:b0b66269f422: Merge commit 'e880418660c80e2f12a123e131975cdb6b12cd13' [10:41] <cone-867> ffmpeg.git 03Anton Khirnov 07master:cab8c5f8e140: h264: do not reinitialize the global cabac tables at each slice header [10:41] <cone-867> ffmpeg.git 03Michael Niedermayer 07master:4a89868c5f9d: Merge commit 'cab8c5f8e140c96ba3725ab709d823abfd1e31a5' [11:13] <cone-867> ffmpeg.git 03Anton Khirnov 07master:2725f2d40315: doc/filters: fix an option name in the unsharp docs [11:13] <cone-867> ffmpeg.git 03Anton Khirnov 07master:5f4b1b1cbd06: lavc doxy: document that avcodec_flush_buffers() invalidates decoded frames [11:13] <cone-867> ffmpeg.git 03Michael Niedermayer 07master:9b3d4258d9f3: Merge commit '2725f2d40315b56f17c5dffe39dda7d94786302d' [11:13] <cone-867> ffmpeg.git 03Michael Niedermayer 07master:c3c6999ea1c3: Merge remote-tracking branch 'qatar/master' [13:57] <wm4> at which point did request_channel_layout actually become reliable enough so that request_channels (now deprecated) is not needed anymore? [13:58] <nevcairiel> has it actually? :p [13:58] <wm4> dunno, currently I just set both [13:58] <nevcairiel> request_channels has been deprecated ever since i started paying attention like 3 years ago [13:58] <wm4> meh [14:07] <cbsrobot> can anyone share "SMPTE 428-1: D-Cinema Distribution Master - Image Characteristics" ? [16:02] <xlinkz0> avutil/timestamp.h still isn't c++ compatible :( [16:04] <Daemon404> xlinkz0, which bit [16:04] <Daemon404> and are you using a proper extern "C" {} [16:05] <Daemon404> ah... i see compound literals [16:06] <wm4> this specific use of compound literals is probably a bit too clever [16:15] <xlinkz0> i did use extern "C" [16:15] <xlinkz0> .. [16:20] <j-b> BBB: *hug* [16:20] <Daemon404> wm4, well yeah [16:20] <Daemon404> it's a define [16:20] <Daemon404> which will be used directly in the c++ [16:21] <ubitux> j-b: btw, if you had reported #8935, i would have explained to you the problem, and afaict the problem is from vlc [16:21] <ubitux> do you want an explanation? [16:22] <j-b> ubitux: yes [16:22] <ubitux> j-b: in ffmpeg sense, AV_CODEC_ID_TEXT is verbatim text, no markup, SUBRIP is text with markup (and SRT is the old broken layouts of packets with timestamps in) [16:23] <ubitux> matroska.c: [16:23] <ubitux> {"S_TEXT/UTF8" , AV_CODEC_ID_SUBRIP}, [16:23] <ubitux> {"S_TEXT/UTF8" , AV_CODEC_ID_TEXT}, [16:23] <ubitux> {"S_TEXT/UTF8" , AV_CODEC_ID_SRT}, [16:23] <ubitux> (sorry extra spaces) [16:23] <ubitux> what i read from the ticket [16:23] <ubitux> VLC only supports "TEXT" [16:23] <ubitux> it should support SUBRIP as well, [16:23] <ubitux> because in Matroska, "TEXT" are actually text with markup [16:23] <ubitux> there is no distinction between markup and no markup [16:24] <ubitux> markup is assumed (subrip markup, aka <i> <b> etc) [16:24] <ubitux> in ffmpeg, we should NOT change SUBRIP to TEXT [16:24] <ubitux> because our decoders will not interpret the markup anymore [16:25] <j-b> So? [16:25] <ubitux> (the packet payload layout will likely be the same ) [16:25] <j-b> S_TEXT/UTF8 is VLC_CODEC_SUBT [16:25] <ubitux> j-b: so... you need to support AV_CODEC_ID_SUBRIP, as consider it TEXT, likely [16:25] <ubitux> this is how i understand the issue at least [16:25] <ubitux> anyway, if you think there is a problem with ffmpeg, tell me [16:26] <j-b> no, you just proved my point, thanks. [16:26] <ubitux> which one? [16:26] <wm4> yeah, TEXT needs to be for "real" plain text [16:27] <ubitux> in matroska sense you can consider it subrip anyway [16:27] <wm4> AV_CODEC_ID_SRT is the old codec ID for subrip with timestamps-in-packet-data, right? [16:27] <ubitux> yes [16:27] <wm4> libav still uses that [16:27] <ubitux> yes [16:27] Action: wm4 has a terrible hack to deal with that [16:27] <j-b> that we should have our own demuxer. [16:27] <ubitux> the packet are sane now [16:28] <ubitux> you get clean subrip/text packet now [16:28] <ubitux> not the crafted packets anymore [16:28] <ubitux> j-b: want the commit hash where i explain all of this? [16:28] <ubitux> i believe it was in APIChanges, and even the Changelog [16:29] <ubitux> mmh maybe that one was just about ASS [16:29] <ubitux> subrip is even older [16:29] <ubitux> but it was in APIChanges as well it seems [16:30] <ubitux> (look for SUBRIP in the APIChanges) [16:41] <xlinkz0> how can I seek to the next keyframe using the API ? [16:41] <Daemon404> pray [16:42] <xlinkz0> mkay [16:42] <xlinkz0> to whom? [16:42] <xlinkz0> the ops? [16:42] <wm4> maybe try to seek to PTS + 1/timebase [16:43] <xlinkz0> on second thought i can afford to just read_frames [16:43] <wm4> then it should seek forward, as long as you don't use AVSEEK_FLAG_BACKWARD [16:43] <cone-549> ffmpeg.git 03Michael Niedermayer 07master:ab6fbe820628: avcodec/cabac: remove h264_lps_state [16:44] <wm4> hm, that reminds me: does AV_PKT_FLAG_KEY mean you can use this packet and all following ones to resume decoding after you've reset the decoder? [16:44] <xlinkz0> where are the flags defined? [16:44] <wm4> which ones [16:45] <xlinkz0> the seek ones [16:45] <wm4> or in other words, does AV_PKT_FLAG_KEY always coincide with seek points in the seek index? [16:45] <wm4> xlinkz0: avformat.h [16:45] <xlinkz0> thanks [16:45] <xlinkz0> so i should use av_seek_frame and not avformat_seek_file ? [16:46] <wm4> both should work [16:46] <wm4> I have no clue which you should use [16:46] <wm4> and both suck [16:46] <xlinkz0> is there anything in this entire project you guys are happy with? :\ [16:46] <wm4> no [16:46] <xlinkz0> :( [16:46] <wm4> some subtitle formats can be seeked only with avformat_seek_file() btw. [16:48] <Daemon404> [15:43] <+wm4> then it should seek forward, as long as you don't use AVSEEK_FLAG_BACKWARD [16:48] <Daemon404> on some formats [16:48] <wm4> yeah [16:48] <wm4> so unreliable [16:48] <Daemon404> you cant rely on jack without indexing first [16:49] <wm4> is there a way to make libavformat index a whole file? [16:49] <Daemon404> no [16:49] <Daemon404> i use ffms2 for this [16:49] <Daemon404> ;) [16:49] <wm4> lol ok [16:49] <michaelni__> ubitux, about the delphi headers did you try to mail the guy who wrote them ? [16:50] <ubitux> nope [16:50] <ubitux> i thought it was just a relic of the past entry [16:50] <ubitux> and didn't thought much about it [16:50] <Daemon404> it is [16:50] <Daemon404> i think delphi has much better support for c stuff now [16:50] <Daemon404> (yes it still exists... somehow) [16:52] <michaelni__> iam ok with removing the link but the guy who submited it originally should be notified that his page disappeared, he might not be aware ... [16:52] <Plorkyeran> +wm4> or in other words, does AV_PKT_FLAG_KEY always coincide with seek points in the seek index? <-- no [16:52] <xlinkz0> so tried seeking after getting some frames and apparently it doesn't work :\ [16:52] <Plorkyeran> it does for most formats [16:52] <wm4> Plorkyeran: heh, just as expected [16:52] <Plorkyeran> but it's set on the wrong frames for PIR h264 [16:52] <ubitux> michaelni__: ok i'll mail him [16:52] <michaelni__> ubitux, thx [16:53] <TheFluff> xlinkz0: how many different formats do you want to be able to deal with [16:53] <xlinkz0> mp4 h264 only [16:53] <TheFluff> if the answer is "all of them", you have a nontrivial problem on your hands [16:53] <xlinkz0> i'd be happy navigating an mp4 [16:53] <TheFluff> then it shouldn't be that hard [16:54] <xlinkz0> ok so i seeked once, got a frame with av_read_frame and decoded the frame [16:54] <xlinkz0> then i seeked again, got another frame the same way [16:55] <xlinkz0> but the second frame is the one after the first one, at the first seek point [16:55] <xlinkz0> do i need to reset something if I want to seek again? [16:55] <TheFluff> did you flush your buffers? [16:55] <TheFluff> as in, call avcodec_flush_buffers() [16:55] <TheFluff> after the seek [16:56] <xlinkz0> the second one? [16:56] <TheFluff> you should flush after each seek, iirc [16:56] <xlinkz0> i didn't, will do now [16:56] <TheFluff> actually I probably shouldn't give you advice on this because I can't remember shit and it's been ages since I did anything with avformat [16:57] <xlinkz0> thanks ! it works [16:57] <wm4> I really wonder why ffms2 isn't part of ffmpeg [16:58] <xlinkz0> you should always give advice on ffmpeg, or noone else will and the person asking will be lost in oblivion :( [16:58] <xlinkz0> sadly there are no examples with seeking [17:00] <TheFluff> wm4: it's written in C++ and thus unclean [17:00] <Daemon404> too high level [17:00] <Daemon404> too usable [17:00] <TheFluff> but I agree that for a lot of use cases it would be nice to have an API that works something like that [17:01] <Plorkyeran> it's like 5k lines of code counting the avs and vapoursynth plugins [17:01] <Daemon404> getframe() > writing that goddamn packet loop for the Nth time [17:01] <Plorkyeran> writing something similar in C as part of ffmpeg should not be that big of a deal [17:01] <Plorkyeran> (but I have no interest in doing it because fuck writing C) [17:04] <pippin> why seek out cancer if you dislike having a cold? [17:05] <xlinkz0> is my getframe loop any good? http://codepad.org/930B0Qvl [17:05] <TheFluff> yes please try starting a c/c++ holy war with the lamest troll I've seen all week [17:10] <wm4> xlinkz0: why it the code block doing video decoding duplicated? [17:11] <xlinkz0> the one at the end is for decoding cached frames [17:11] <xlinkz0> i should put that in a separate function indeed [17:11] <Paranoialmaniac> xlinkz0: av_read_frame may return EAGAIN (which takes negative value). don't confuse with EOF [17:11] <wm4> what's EAGAIN for? [17:12] <Paranoialmaniac> "there is no data available right now, try again later" [17:12] <wm4> libavdevice stuff? [17:13] <Paranoialmaniac> dunno. but i often get EAGAIN when reading some flv files. [17:14] <xlinkz0> thanks [18:33] <cone-549> ffmpeg.git 03Michael Niedermayer 07master:45f0623ae3fa: avcodec/lagarith: check scaled probabilities [20:05] <durandal_1707> have the emu edge issue been fixed? [20:17] <cone-549> ffmpeg.git 03Michael Niedermayer 07master:c99d27287d9b: avcodec/wmaprodec: check that there are input bits left in decode_subframe_length() [20:58] <wm4> haha the next group of fringe bugs [21:01] Action: durandal_1707 here goes food [21:06] <durandal_1707> michaelni: ^ is bug in parser [21:49] <cone-549> ffmpeg.git 03Paul B Mahol 07master:f5498ef38daa: avcodec/flicvideo: fix infinite loops [22:01] <cone-549> ffmpeg.git 03Michael Niedermayer 07master:851a6e2f1aee: avcodec/wmalosslessdec: Fix return code for invalid buffer sizes [22:55] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:ba5dfc25ee0a: indeo4: Do not access missing reference MV [22:55] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:06c52faef27e: indeo4: Check the quantization matrix index [22:55] <cone-549> ffmpeg.git 03Michael Niedermayer 07release/0.10:91ad27e8f5fb: Merge commit '06c52faef27e5bded4ceda7e6d1541f9fb20e84c' into release/0.10 [22:56] <JEEB> avformat/mpegenc.c:remove_decoded_packets() -- what does the buffer underflow error actually mean? [22:56] <JEEB> is it a VBV underflow, aka bits are going out too fast? or something else [23:02] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:609345cd5e2a: indeo4: Validate scantable dimension [23:02] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:e2dcb8208e8f: indeo5: return proper error codes [23:02] <cone-549> ffmpeg.git 03Michael Niedermayer 07release/0.10:3308b443f934: Merge commit 'e2dcb8208e8f6cffef58a85127765047f5ef8868' into release/0.10 [23:16] <kolosos> http://pastebin.com/2A5ardCU <=== i get this strange buffer underflow error on several input formats of a youtube video - 480.flv, 720.mp4, 720.webm, 480.webm, etc. [23:18] <kolosos> then dvdauthor confirms as follows : WARN: Discontinuity of 71 in audio channel 0; please remultiplex input [23:34] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:36921fcdd361: indeo: Reject impossible FRAMETYPE_NULL [23:34] <cone-549> ffmpeg.git 03Martin Storsjö 07release/0.10:729143e2d27d: ac3dec: Don't consume more data than the actual input packet size [23:34] <cone-549> ffmpeg.git 03Martin Storsjö 07release/0.10:a593d2e92e14: mov: Do not allow updating the time scale after it has been set [23:34] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:0d24adbe8d8e: dsicinav: Bound-check the source buffer when needed [23:34] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:246e0e2c994f: alsdec: Fix the clipping range [23:35] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:8006716f2155: xl: Make sure the width is valid [23:35] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:9c779b5dd0e8: bink: Bound check the quantization matrix. [23:35] <cone-549> ffmpeg.git 03Luca Barbato 07release/0.10:54e03863691d: vc1: check the source buffer in vc1_mc functions [23:35] <cone-549> ffmpeg.git 03Michael Niedermayer 07release/0.10:210a437e105f: Merge commit '54e03863691dcae73260f70108b3731b70773e7c' into release/0.10 [23:36] <kolosos> <cone-549> ffmpeg.git Martin Storsjö release/0.10:729143e2d27d: ac3dec: Don't consume more data than the actual input packet size [23:37] Action: kolosos wonders if that could be related to http://pastebin.com/2A5ardCU [23:37] <JEEBsv> not related to you :P [23:37] <kolosos> meh ok [23:37] <JEEBsv> the encoders seem to be working just fine for you, it's just the mpegenc muxer erroring out :P [23:38] <JEEBsv> also do feel free to grab a static build of a newer ffmpeg version and see if that does the same thing [23:38] Action: kolosos shudders [23:38] <JEEBsv> fflogger> Static FFmpeg builds - 1) for 3.x+ kernels: http://ffmpeg.gusari.org/static/ or 2) for 2.6.32+ kernels: [23:39] <JEEBsv> http://johnvansickle.com/ffmpeg [23:39] <JEEBsv> a static binary you can just run from some directory :P [23:39] <michaelni> kolosos, for VBV errors try change max/min bitrate, vbv size, also try higher quality encoding settings like trellis, mbd 2, they can help make the video fit better in the buffer [23:39] <JEEBsv> michaelni: that seems to be a muxer VBV error, not a video encoder error, funny enough :s [23:40] <JEEBsv> I tried to look at the code in mpegenc.c [23:40] <JEEBsv> but I just couldn't read what exactly it was failing at [23:40] <michaelni> hmm [23:40] <JEEBsv> @ remove_decoded_packets() [23:41] <michaelni> when the muxer fails, first question is, is the encoder producing vbv compliant output ? [23:42] <JEEBsv> anyways, if I recall correctly the -target setting does set the VBV constraints at least [23:42] <JEEBsv> it also changes rate and resizes the clip [23:43] <JEEBsv> also it is correct to think of that error as a VBV underflow? [23:45] <JEEBsv> kolosos: anyways do try with a static build [23:45] <JEEBsv> that would at least tell that the thing still happens [23:45] <JEEBsv> the 1.2.x branch is by now pretty old [23:46] <kolosos> <michaelni> when the muxer fails, first question is, is the encoder producing vbv compliant output ? [23:46] <kolosos> any way to check that? [23:49] <kolosos> Here's the video: http://www.youtube.com/watch?v=yqGcZ6WBZnY [23:50] <michaelni> probably better if you upload the flv somewhere, who knows if 2 people downloading from youtube get the exact same file [23:50] <michaelni> also probably a good idea to open a ticket on trac [23:51] <kolosos> ok - learning a lot, thought this morning i will "just quickly download a vid and burn to a dvd" [23:52] <kolosos> that was 6 hours ago :) [23:54] <kolosos> i basically tried all 6 the higher-res formats available for download using the firefox plugin "1-click Youtube Video Downloader 2.1.5" [23:55] <kolosos> i'm off to the rink for practice but will be baaaak [00:00] --- Thu Sep 26 2013

1 0

ffmpeg.log.20130925
by burek 26 Sep '13

26 Sep '13

[00:08] <BallsDeep> I am trying two different commands to transcode DTS to AC-3. [00:08] <BallsDeep> Note: [00:08] <BallsDeep> I know I can use one "ffmpeg" command to get the end result but I want to try the "dcadec" command. [00:09] <BallsDeep> Commands I am using: [00:09] <BallsDeep> dcadec -o float filename.dts > filename.raw [00:09] <BallsDeep> ffmpeg -ar 48k -ac 6 -f f32le -i filename.raw filename.ac3 [00:09] <BallsDeep> The end result: The AC-3 track is created and it sounds like shit. [00:11] <BallsDeep> Any ideas? [00:38] <LithosLaptop> have you tried adding a bitrate parameter? [00:38] <LithosLaptop> -ab 384k [00:43] <BallsDeep> yes, 448k and 640k. same problem occurs (static in the background and the dialogue sounds like a bunch of chipmunks talking) [00:44] <LithosLaptop> oh ok, maybe ffmpeg is incorrectly picking up the input file's sampling rate [00:45] <LithosLaptop> oh [00:45] <LithosLaptop> nm [00:45] <LithosLaptop> I see you specify it [00:46] <LithosLaptop> are you sure the raw file is sampled at 48k? [00:48] <LithosLaptop> also make sure each sample really is f32le [00:57] <BallsDeep> When I use the dcadec command, I don't know what "-o float" is creating for a raw file. I'm assuming "f32le". How would I know? After all, it's a raw file so how could I tell? Is there a piece of software out there that can tell me what a raw file contains? [01:01] <Jan-> is the argument "-ac 2" still the preferred way to just dump two audio channels? [01:01] <Jan-> Assuming we had four to begin with, that is? [04:11] <Zutara> Hi. I'm trying to batch convert MKV's to AVI's but when I run the command it asks me if I want to replace the MKV's. What am i doing wrong? Pastebin: http://pastebin.com/t7rbBufY [04:13] <sacarasc> avconv help is in #libav. [04:16] <Zutara> I get the same output when I do ffmpeg. [04:17] <sacarasc> Does it also say that it is deprecated? Because that's avconv too. [04:17] <Zutara> Yeah. [04:23] <Zutara> Why is it asking me to overwrite it? [04:58] <grepper> Zutara: for this you should ask in #bash. You can't use wildcards the way you are using them. [04:59] <grepper> you could use a loop: for vid in *.mkv; do avconv "$vid" ... [04:59] <grepper> anyway, neither an ffmpeg or avconv question really [05:00] <Zutara> Okie dokie. :) [05:10] <grepper> Zutara: for vid in *.mkv; do avconv "$vid" -vcodec copy -acodec copy -y "converted-$vid" ; done [05:11] <Zutara> I had that sent to me in #libav. What do I put inside the quotes? [05:11] <grepper> what quotes [05:11] <Zutara> in the command you just sent. [05:11] <Zutara> Or do I leave the quotes out/ [05:11] <Zutara> *? [05:11] <klaxa> i suggest you read some stuff about bash [05:11] <grepper> type it exactly as shown [05:11] <grepper> heh [05:12] <klaxa> $vid in this case is a variable [05:12] <klaxa> which gets expanded to its value when interpreted by bash [05:12] <klaxa> the values are the files that match *.mkv in terms of bash expansion [05:13] <defaultro> how come, when I use -crf 0, the video gets some macro blocking. That's why I just always use -crf 15 but I'm not sure if this is the highest quality [05:14] <klaxa> bad source? [05:14] <klaxa> weird filters? [05:14] <klaxa> :x [05:14] <defaultro> k [05:15] <klaxa> -crf 0 should produce lossless, at least if we are talking about libx264 [05:15] <klaxa> maybe ask in #x264 they might know more [05:17] <defaultro> yup, but everytime i use -crf 0, video will definitely get macro blocking [05:17] <defaultro> k [05:20] <theholyduck> defaultro, well then, either something is wrong with your ffmpeg, or ffmpeg doesnt support it right [05:20] <theholyduck> i dunno [05:20] <theholyduck> i almost always use x264 directly [05:20] <defaultro> got it [05:20] <defaultro> i've never tried x264 directly [05:20] <elBradford> HI! Looking for some assistance with compiling a static build of ffmpeg and librtmp for Android. Does anyone have experience in this area? [05:20] <defaultro> i wasn't aware we can do that :) [05:26] <elBradford> I just need some help with the ffmpeg build finding the rtmpdump library [05:28] <defaultro> hey folks, is cqp an old option? my ffmpeg doesn't know it [05:31] <klaxa> defaultro: try -qp 0 instead [05:32] <defaultro> yup, it's running [05:32] <defaultro> found it here, http://trac.ffmpeg.org/wiki/x264EncodingGuide#LosslessH.264 [05:32] <defaultro> does it matter if my extension is either mkv or mp4? [05:32] <klaxa> it changes the container [05:33] <defaultro> but quality is the same right? [05:33] <klaxa> matroska is probably the most sophisticated container [05:33] <defaultro> k [05:33] <klaxa> yeah since the video stream is independent from the container [05:33] <defaultro> what;s the reason for choosing a container? [05:34] <klaxa> compatibility vs. features [05:34] <defaultro> is it because of the number of supported video players? [05:34] <defaultro> got it [05:34] <defaultro> mp4 is famous [05:34] <defaultro> and has more support. Am i right? [05:34] <klaxa> mkv is too, you can put anything into mkv [05:34] <defaultro> k [05:34] <klaxa> i'd think so [05:34] <defaultro> that's great info i learned tonight :) [05:35] <defaultro> wow, my bitrate is 239932 kb/s, lol [05:35] <klaxa> well it's lossless [05:36] <klaxa> wait what? [05:36] <defaultro> not sure how youtube will treat this kinds of upload [05:36] <klaxa> 240 mbps? [05:36] <defaultro> yes, it's lossless [05:36] <defaultro> yeah [05:36] <defaultro> that's what ffmpeg -i wrote [05:36] <klaxa> well youtube should be able to deal with it [05:36] <klaxa> because a) they use ffmpeg themselves [05:36] <defaultro> k [05:36] <klaxa> and b) i uploaded lossless videos and they could deal with it [05:36] <defaultro> i just hate youtube because they reencode it again [05:37] <defaultro> our video loses the nice quality of the image [05:38] <elBradford> Is this the right room to ask questions regarding building ffmpeg? [05:38] <defaultro> it is ok elBradford [05:38] <defaultro> if someone knows, they will help you [05:39] <elBradford> Thanks defaultro [05:39] <defaultro> but remember in irc, you have to be patient :) [05:39] <klaxa> elBradford: get the sources for librtmp, compile static, get sources for ffmpeg, compile static and link against static librtmp [05:39] <defaultro> sometimes you won't even get an answer [05:39] <klaxa> did that make sense to you? [05:40] <elBradford> klaxa - it seems like ffmpeg is looking at pkg config for librtmp - if I include it using a -I flag, will that work? [05:40] <defaultro> i'm reencoding a 60fps video and I'm using -qp 0. Bitrate is showing 487210 kbits/s [05:40] <klaxa> do you have a static version of librtmp? [05:41] <elBradford> Yeah, I can build that. [05:41] <elBradford> I mean, I have built that already [05:41] <klaxa> you will need static versions of all libraries you want to include [05:42] <klaxa> did you read the compilation guide? [05:42] <klaxa> https://trac.ffmpeg.org/wiki/CompilationGuide [05:43] <elBradford> I'll read through that, I hadn't seen that. I'm doing it against the Android toolchain, but that will be helpful. I'll come back if I get stuck. Thanks Klaxa [08:24] <yusari> Hi all. I'm looking for a CLI command to convert a 23.98fps mov file to 24fps without losing any quality. Can anyone help? [08:25] <relaxed> why is that necessary? [08:27] <yusari> Hi relaxed. It's for digital cinema conversion. [08:30] <relaxed> what format does it need to be in? [08:31] <yusari> input - 23.98fps mov, output 24fps mov [08:34] <relaxed> ffmpeg -i input.mov -c:v libx264 -crf 15 -r 24 -c:a pcm_s16le output.mov [08:36] <yusari> Thanks a lot, relaxed! [08:36] <yusari> would the same command work if I wanted to convert a 25fps mov file to a 24fps mov file as well? [08:37] <relaxed> yes, ffmpeg will duplicate or drop frames to achieve your desired frame rate. [08:38] <yusari> and it will do so for both the image as well as audio? [08:40] <relaxed> I'm not sure about the audio. [08:40] <relaxed> It should maintain sync if that's your concern. [08:46] <yusari> wouldnt the libx264 option re-encode the video, thus losing some of its quality? [08:47] <yusari> my input file is apple prores 442, and I'd like to keep the output file the same [08:48] <relaxed> you will have to re-encode to change the frame rate. [08:50] <yusari> no way to change the frame rate without re-encoding, thus losing quality? [08:50] <relaxed> No [08:51] <relaxed> ffmpeg prores enocoder only supports yuv422p10le [08:51] <relaxed> did you mean 422? [08:52] <yusari> yes, im sorry [08:52] <yusari> I meant 422 [08:53] <relaxed> then remove "-c:v libx264 -crf 15" and use -c:v prores [08:53] <yusari> ffmpeg -i input.mov -c:v prores -r 24 -c:a pcm_s16le output.mov [08:54] <yusari> so that would change the existing frame rate (whether its 23.98 or 25) to 24fps [08:55] <yusari> wrt audio, I require 24bps for cinema assets, so I would use pcm_s24le instead, correct? [08:55] <relaxed> yeah, there's more info prores here https://trac.ffmpeg.org/wiki/vfxEncodingGuide [08:55] <relaxed> for audio try -c:a copy [08:58] <yusari> would -c:a copy change the audio tempo as well tosync with the new image frame rate? [08:58] <relaxed> no, it would copy the stream [08:58] <yusari> or would you suggest I use something like Audacity for that? [08:59] <yusari> right, so ffmpeg can't be used to change the tempo of the audio to match the changed frame rate of the image [09:02] <relaxed> maybe there's a filter for it, check "ffmpeg -filters | less" [09:11] <yusari> thers's an atempo switch to adjust tempo [12:26] <aleray> hi, I have a bunch a hd MOV files which I would like to concatenate without reencoding (to then convert to webm or ogg). How can I do this? The machine I will use for that is a mac [12:43] <aleray> I tried: ffmpeg -i "concat:input1.mpg|input2.mpg|input3.mpg" -c copy output.mpg but th equality is altered [12:43] <aleray> the quality [12:43] <aleray> I would like to not reencode it [12:44] <viric> that can be done with very specific input formats, isn't it? I guess. [12:47] <aleray> viric, it seems to be mpeg 2 inside the mov [12:48] <viric> I don't know the details [12:53] <aleray> viric, would an output of ffprobe or so help? [12:56] <viric> aleray: no, I don't know the details of ffmpeg :) [14:06] <chrrbss> can anyone help me with itoffset? I'm overlaying a bunch of images onto a video and want to offset the appearance of the image by xxx seconds [14:06] <chrrbss> currently got: ffmpeg -i ./../../../sourcevideo/test.mov -f image2 -itoffset 0:0:20 -i %2d.png -s 1280x720 -strict -2 -filter_complex 'overlay=0:0;' combined.mp4 [14:10] <chrrbss> sorry; pastebin is here: http://pastebin.com/h7nMQhNT [14:15] <aleray> back [14:16] <chrrbss> and pastebin with output: http://pastebin.com/z7Jwxm1D [14:16] <aleray> viric, I was away, I may have miss your answer if any [14:16] <chrrbss> sorry, dont use irc that much :) [14:20] <aleray> "ffmpeg -i file.mov" gives me http://dpaste.com/1395871 [14:20] <aleray> how can I do to concatenate several of those mov files [14:20] <aleray> ? [14:25] <viric> no answer [14:29] <Bombo> aleray: i did that with mencoder once [14:29] <Bombo> but i found this https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20%28join,%20merge%29%2… [14:29] <aleray> Bombo, I tried but somehow the file quality is reduced [14:29] <aleray> it seems to reencode the thing [14:30] <Bombo> aleray: you tried what exactly? [14:30] <Bombo> did you chose 'copy' as codec? [14:30] <aleray> Bombo, ffmpeg -f concat -i 1.MOV 2.MOV -c copy output.mpg [14:31] <aleray> Bombo, and "ffmpeg -i "concat|1.MOV|2.MOV" output.mpg" [14:31] <Bombo> that would need to reencode from h264 to mpeg2 [14:31] <Bombo> try output.mov [14:32] <aleray> Bombo, ok [14:32] <aleray> the first command or the second? [14:32] <aleray> boiled_sugar, [14:32] <aleray> oups [14:32] <Bombo> try both [14:33] <Bombo> for lossless copy, stay with the same codec/container [14:34] <aleray> Bombo, could you write down the exact command you would try? [14:38] <Bombo> ffmpeg -i "concat:1.mov|2.mov" -c copy output.mov [14:39] <Bombo> try this [14:39] <aleray> Bombo, [14:39] <aleray> ok [14:39] <aleray> thanks [14:39] <Bombo> does it work? i didn't try ;) [14:40] <aleray> Bombo, I think this is what I'm running now, but it takes time so I suspect it reencodes the video [14:40] <aleray> I'm waiting for it to be done to see how it looks like [14:40] <aleray> 5 minutes to go or so [14:41] <Bombo> or this one with demuxing: ffmpeg -f concat -i <(find . -name '*.mov' -printf "file '%p'\n") -c copy output.mov [15:11] <aleray> Bombo,seems to be working with ~/bin/ffmpeg -f concat -i files.txt -c copy output6.mov [15:12] <chrrbss> in case anyone is interested; i found the solution by moving itoffset in front of the input of the images like so: http://pastebin.com/ZedxVnY3 [15:23] <aleray> Bombo, it turned out that some lecture we have recorded overlaps (end of lecture A and beginning of lecrue B on the same file). Is it possible to concat only part of a file? [15:43] <ayaka> I want to embed a live stream into web, could ffsever do it and which output format shall I choice [15:45] <Bombo> aleray: not sure, but you could cut the movs before cat [15:47] <Bombo> aleray: copy from timestamp to timestamp... with -ss 00:00:00 (start time) and -t 00:01:00 (duration, 1min) [18:44] <version2beta> I am having issues getting closed captions (608/708) out of ffmpeg using ccextractor when the filter chain includes yadif. Everything works fine when yadif isn't in the chain, but with it, ccextractor complains about "Error: Broken AVC stream - forbidden_zero_bit not zero ..." [18:46] <version2beta> my ffmpeg command is basically `ffmpeg -i udp://239.10.10.1:1234 -an -sn -map 0:v -c copy -f mpegts pipe:3 -map 0:v -vf yadif=0:-1:1,scale=720:-1 -c:v mjpeg -r 1 -f image2pipe pipe:1` [18:47] <version2beta> Both pipes are streamed into Nodejs where code either pipes it back out to ccextractor (pipe:3) or splits out jpegs and sends them off to a better life. [18:48] <version2beta> btw, the error occurs only about once every three minutes, but consistently with yadif and not yet once without yadif. [19:35] <Niatross> I want to transcode a DTS track to AC-3. I want to do it in three steps. [19:35] <Niatross> BTW: I know you can do a DTS to AC-3 transcode with one "ffmpeg" command, but I want to break it down into 3 steps (DTS to RAW&&.RAW to WAV&&.WAV to AC-3) [19:36] <Niatross> Are the following steps correct: [19:36] <Niatross> 1 ) DTS to RAW: [19:36] <Niatross> ffmpeg -i input.dts -f f32le output.raw [19:36] <Niatross> 2) RAW to WAV: [19:36] <Niatross> ffmpeg -ar 48k -ac 6 -f f32le -i input.raw -acodec pcm_f32le output.wav [19:36] <Niatross> or [19:37] <Niatross> ffmpeg -ar 48k -ac 6 -f f32le -i input.raw -acodec pcm_s32le output.wav [19:37] <Niatross> 3) WAV to AC-3: [19:37] <Niatross> ffmpeg -i input.wav -ab 640k output.ac3 [19:37] <Niatross> My questions are: [19:37] <Niatross> Which command in Step 2 should I use (should I keep it floating-point all the way until it reaches the AC-3 encoder or should I switch it to signed-integer)? [19:37] <Niatross> Should I keep every input and output 32-bit (instead of 16-bit) before it reaches the AC-3 encoder? [20:45] <Niatross> anyone? [20:48] <durandal_1707> Niatross: keep it float [20:49] <durandal_1707> ffmpeg -h encoder=ac3 [20:50] <iive> btw, why raw? wav pcm IS raw. [20:51] <Niatross> i want no header on it [21:11] <Niatross> member:durandal_1707 On step 2, should I leave "-acodec pcm_f32le" or let it default to 16 bit? What I mean by this is: Should I keep every input and output 32-bit before it reaches the AC-3 encoder?)? [21:13] <durandal_1707> Niatross: converting float to int is pointless as int would be converted back to float [21:13] <durandal_1707> same apply to 32bit to non32bit [21:15] <durandal_1707> if you do not care at all about audio quality and transcoding speed (the pointless 3 commands that already points out) you can do whatever you like [21:15] <Niatross> ok, so I'll leave the "-acodec pcm_f32le", so it doesn't default to 16-bit. [21:15] <durandal_1707> if you want [21:16] <Niatross> well maybe it wouldn't default to 16 bit, since it will take the "-f f32le" from the raw input [21:18] <durandal_1707> nope it will default to 16 [21:19] <Niatross> ok, so it's best to leave the "-acodec pcm_f32le" to keep it 32-bit until the ac-3 encoder gets a hold of it [21:20] <durandal_1707> yes [21:20] <Niatross> thanks for helping me!!!!!!!!!!!! [22:14] <kolosos> http://pastebin.com/2A5ardCU "buffer underflow" error in converting youtube[480].flv video to ntsc-dvd .mpg [22:14] <kolosos> Any tips? [22:19] <kolosos> (h264 -> mpeg2video) (aac -> ac3) [dvd @ 0x9b3720] buffer underflow i=1 bufi=1550 size=1792 [22:20] <viric> kolosos: it can be a pts problem [22:20] <viric> is it really a so simple conversion? [22:20] <kolosos> viric: http://pastebin.com/2A5ardCU [22:21] <viric> ahm looks like so, yes [22:21] <viric> no idea. [22:23] <kolosos> viric, thanks for the help. i'll try a different youtube video & see what happens [22:25] <kolosos> going to try jump480.webm instead of jump480.flv [22:27] <JEEB> kolosos, buffer underflows mean that the mpeg-2 encoder wasn't able to keep the rate within the constraints set [22:29] <JEEB> at least that's what VBV underflow means, not exactly sure if this is the same thing :P [22:35] <kolosos> JEEB - isn't -target NTSC-DVD supposed to take care of that? [22:36] <JEEB> it should just set the limit [22:36] <JEEB> if the encoder cannot cope with it... [22:36] <JEEB> :P [22:38] <JEEB> ugh [22:38] <JEEB> mpegenc.c [22:38] <JEEB> it's a weird mix of things [22:40] <kolosos> JEEB: for instance, the .flv video was AAC at 44100 Hz and the .mpg output was AC2 at 48000 Hz. Is ffmpeg telling me that it can't figure out how to upsample? [22:40] <JEEB> no [22:40] <JEEB> the warning is coming from mpegenc.c [22:40] <JEEB> or wait... mpegenc.c is from libavformat [22:40] <JEEB> asdf [22:40] <JEEB> it's a muxer [22:41] <JEEB> so it's saying that a buffer underflow happens during muxing [22:42] <JEEB> but I'm still not sure if it's saying that packets are too big or too small... [22:42] <JEEB> if(stream->buffer_index < pkt_desc->size || [22:42] <JEEB> stream->predecode_packet == stream->premux_packet){ [22:42] <JEEB> buffer index is smaller than the size of the packet? [22:42] <JEEB> or [22:42] <JEEB> predecode_packet equals premux_packet [22:43] <JEEB> sorry I can't make heads or tails of this :P [22:43] <JEEB> but if it doesn't error out [22:43] <JEEB> I guess it's OK? [22:43] <kolosos> np, i was hoping for an answer like "you are doing something totally idiotic, use the -XYZ switch" [22:45] <kolosos> next I'm going to try the 720p MP4 encoded youtube vid [22:46] <JEEB> anyways, the actual encode seems to be going on OK, but the muxer then is telling that there are buffer underflows [22:46] <JEEB> muxer puts streams into a container (in this case, MPEG-PS), encoders take in raw data and create streams [22:49] <kolosos> guessing that it's the 44100 vs 48000 audio, can i create a -target NTSC-DVD with an AC2 rate of 44100 s/s ? [22:49] <JEEB> it's not that [22:50] <JEEB> the audio rate has to be 48kHz in DVDs [22:50] <kolosos> ah ok [22:50] <kolosos> so you think it's a real software issue [22:50] <kolosos> some internal buffer not being ready [22:50] <JEEB> no [22:50] <kolosos> ^_^ [22:51] <kolosos> the "muxer" is not happy... why [22:51] <JEEB> I don't know what exactly it's trying to say, but it doesn't happen in the video/audio encoders, it's just warning you that the packetizing (?) buffer is either underused or overused, I just can't read from the code which it is :P [22:51] <JEEB> also as far as I can see, it's not an error [22:51] <JEEB> just a warning [22:51] <kolosos> OH [22:51] <kolosos> OOOOH [22:52] <kolosos> ok let me see if i can play the output... [22:52] <JEEB> it should play just fine methinks [22:54] <kolosos> ffprobe jump480m.mpg looks good : [22:54] <kolosos> mpeg2video yuv420p, 720x480 [SAR 32:27 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Audio: ac3, 48000 Hz, stereo, fltp, 448 kb/s [22:56] <JEEB> also, d'oh... it is an AV_LOG_ERROR :/ then why did you get multiple of them... [22:56] Action: JEEB pokes the -devel channel [22:57] <kolosos> mpeg2video 720x480 plays perfectly in totem [22:57] <JEEB> yes, the encoding doesn't fail as such, the video and audio should be just fine and the packets written should be just fine [22:59] <kolosos> ok, in that case let me try do dvdauthor from the mpeg [22:59] <JEEB> but if it's an error then some packets didn't get written... or what the flying fuck happened [22:59] <JEEB> join #ffmpeg-devel to see if I get a response [22:59] <JEEB> I'm going to sleep soon :P [22:59] <JEEB> <JEEB> avformat/mpegenc.c:remove_decoded_packets() -- what does the buffer underflow error actually mean? [22:59] <JEEB> <JEEB> is it a VBV underflow, aka bits are going out too fast? or something else [23:00] <JEEB> I asked this [23:02] <kolosos> dvdauthor seems to agree: lots of WARN: Discontinuity of 71 in audio channel 0; please remultiplex input. [23:03] <JEEB> there's basically a counter in MPEG-TS (and I think in MPEG-PS as well) [23:03] <JEEB> in each packet [23:03] <JEEB> so that just means that there was a discontinuity of X packets in there :P [23:03] <kolosos> ah, so smart software can see there are samples missing [23:04] <JEEB> In MPEG-TS it was just 0-15 so not sure if you could count a discontinuity of 71 :D [23:05] <kolosos> but, again, just a warning (i hope) so i SHOULD have a playable DVD. any idea how to test that in fedora? [23:05] <JEEB> no, as I just said it is AV_LOG_ERROR [23:05] <JEEB> I thought it was a warning because you got it multiple times [23:06] Action: kolosos doesn't know what AV_LOG_ERROR is and how do you know that? [23:06] <JEEB> av_log(ctx, AV_LOG_ERROR, [23:06] <JEEB> "buffer underflow i=%d bufi=%d size=%d\n", [23:06] <JEEB> i, stream->buffer_index, pkt_desc->size); [23:06] <JEEB> that's the message you're getting [23:06] <JEEB> although it only breaks from that loop [23:07] <JEEB> it doesn't return nonzero or whatever [23:07] <JEEB> argh [23:07] <JEEB> anyways, join #ffmpeg-devel and see if anyone answers what I asked and possibly you then ask extra questions :P [23:13] <kolosos> JEEB: thanks for help [23:35] <kolosos> JEEB - it seems to be related to the input file - I tried another youtube vid which did not have the same problem [23:35] <JEEBsv> uhh [23:35] <JEEBsv> after you decode the input [23:35] <JEEBsv> it's all raw data in both cases [00:00] --- Thu Sep 26 2013

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