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July 2014

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ffmpeg-devel.log.20140725
by burek 26 Jul '14

26 Jul '14

[03:07] <BBB> kurosu_: lol :) [05:05] <Zeranoe> Getting a compile error with the latest FFmpeg "libavutil/bprint.h:35:48: error: types may not be defined in 'sizeof' expressions" [05:43] <cone-868> ffmpeg.git 03Nicolas George 07master:04bc37007224: lavf/avio: do not include bprint.h. [05:44] <michaelni> Zeranoe, maybe fixed [06:31] <jamrial> ubitux: aac-al_sbr_hq_cm_48_5.1 seems to be failing in all of your gcc 4.9.1 FATE slots [06:32] <jamrial> can't reproduce it here with mingw gcc 4.9.1 [06:54] <nevcairiel> Speaking of, I should see about updating my mingw stations over the weekend [12:50] <cone-21> ffmpeg.git 03Diego Biurrun 07master:060e4a9e0627: dct/rdft: Remove duplicate typedefs for context structs [12:50] <cone-21> ffmpeg.git 03Michael Niedermayer 07master:937eaba2a115: Merge commit '060e4a9e0627adcda9101dd68ec7f22b4ce22b5b' [13:06] <cone-21> ffmpeg.git 03Diego Biurrun 07master:7fb993d338d8: qpeldsp: Mark source pointer in qpel_mc_func function pointer const [13:06] <cone-21> ffmpeg.git 03Michael Niedermayer 07master:2904d052b7ca: Merge commit '7fb993d338d88f2f62e0a358b6c9f3eb9a3a08ac' [13:33] <cone-21> ffmpeg.git 03Timothy Gu 07master:065fa6687236: RELEASE_NOTES: fix WebVTT decoder typo (-> encoder) [13:33] <cone-21> ffmpeg.git 03Michael Niedermayer 07release/2.3:84642ec8798a: avformat/dv: implement fallback in dv_extract_pack() [13:33] <cone-21> ffmpeg.git 03Andreas Cadhalpun 07release/2.3:0f04e2741e26: libavformat/img2dec.c: Add a long_name to the piped image demuxers [13:33] <cone-21> ffmpeg.git 03Michael Niedermayer 07release/2.3:9cb45f6ad224: doc/APIchanges: mention FF_INPUT_BUFFER_PADDING_SIZE chage [13:33] <cone-21> ffmpeg.git 03Timothy Gu 07release/2.3:489d066d498a: RELEASE_NOTES: fix WebVTT decoder typo (-> encoder) [17:51] <wm4> does ffmpeg still support BeOS [17:52] <wm4> I just found out why ffmpeg tries to deal with the assumption that libc error codes can be negative, even though C specifies it's not [17:52] <wm4> and it's because of... BeOS [17:53] <ubitux> i think we support haiku [17:53] <ubitux> not sure if it applies for it as well [17:54] <wm4> yeah, since 2010 only Haiku support [17:55] <wm4> and C99 requires that error codes are strictly positives [17:55] <wm4> but I don't know how broken Haiku is [17:55] <ubitux> we had a fate instance [17:55] <ubitux> but it died a while ago [19:09] <kierank> might be worth asking on this side: [19:09] <kierank> when you get formats like ac3, adts, latm which have multiple packets concatenated [19:10] <kierank> how are teh timestamps generated [19:10] <kierank> and where [20:31] <llogan> he actaully is referring to a "hash brown" and not some oddball project name or whatever. weird. [20:42] <michaelni> ubitux, haiku died ?! [20:46] <michaelni> haiku didnt die, seem ffmpeg crashed on it and it was stuck on a GUI blah blah crashed whatever dialog [20:49] <ubitux> it was dead for a long long while :P [20:49] <ubitux> so you just resurect it? [20:51] <cone-217> ffmpeg.git 03Diego Biurrun 07master:165e9df19567: fft-test: Pass the right struct members instead of casting [20:51] <cone-217> ffmpeg.git 03Michael Niedermayer 07master:aadf5222da51: Merge commit '165e9df19567ec0b6abee1ee2c26027e6d7aa7bf' [20:53] <Compn> kierank : container pts dts or do you mean raw format ? [20:53] <Compn> kierank : if raw, packetizer code i'd assume [20:54] <kierank> parser apparently [20:55] <michaelni> ubitux, i just killed fate on it and restarted it, we will see if thats enough or if tehres an actual issue [21:37] <cone-217> ffmpeg.git 03Michael Niedermayer 07master:24327706e173: avutil/opt: ensure the right buffer is used in set_string_number() [23:48] <mraulet> michaelni: new hevc conformance bitstreams https://dl.dropboxusercontent.com/u/588600/hevc_bitstreams.zip [23:53] <michaelni> mraulet, is the random file extension intended (.bit.bin vs. .bit) ? [23:53] <michaelni> should i upload them like that ? [23:55] <mraulet> oops the renaming of the extensions was not working correctly [00:00] --- Sat Jul 26 2014

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ffmpeg.log.20140725
by burek 26 Jul '14

26 Jul '14

[01:26] <kippi> hey [01:26] <kippi> should this work? use these two filters? -vf "select='gt(scene, 0.01)',showinfo" -af silencedetect=n=-10dB:d=20 [01:30] <c_14> it should [01:30] <Hello71> !tias [01:30] <c_14> Those are 3 filters though. [02:31] <troy_s> michaelni: Please reconsider the YCbCr conversion bug. [02:32] <troy_s> michaelni: The PNG values are 100% correct, assuming whatever viewer you use dumps them as 1:1 RGB. [02:32] <troy_s> michaelni: They had to be encoded from raw data as gamma 1.0 because otherwise automatic color transforms will occur. [02:33] <michaelni> troy_s, i checked in multiple viewers the pngs are identical except +-1 [02:33] <troy_s> (And given that the values are YCbCr and not RGB, this would be entirely problematic [02:33] <michaelni> try gimp [02:33] <troy_s> Yes but the +- 1 is the issue. [02:33] <troy_s> The rounding shouldn't be happening from what I can see, unless bit shifting is mangling? [02:35] <michaelni> if its a +-1 rounding issue, why dont you mention this ? [02:35] <michaelni> also why is the difference png totally wrong ? [02:35] <michaelni> it shows differences in areas that are 100% identical [02:36] <michaelni> its really hard to debug something with such low quality information [02:38] <michaelni> troy_s, also if you argue gamma=1 is correct why does only one of the 2 pngs have gamma=1 ? [02:38] <troy_s> michaelni: Those are only tags. [02:38] <troy_s> michaelni: You can ignore them. [02:38] <troy_s> michaelni: GIMP for example, should. [02:39] <michaelni> but they affect how the png is displayed in some software [02:39] <troy_s> michaelni: So will plain RGB in some software as it tries to color manage! [02:39] <troy_s> michaelni: There is no format that I can give you aside from a float EXR that will interpret the values 110% correct [02:40] <troy_s> And if I do that, then I am at the whim of the software reading the EXR correctly as well as the fact we add more conversions in [02:40] <troy_s> (Unsigned integer to float) [02:40] <troy_s> So A) The quality of information is absolutely as precise as I can assert to being correct. [02:41] <troy_s> B) there is no raw data wrapper that I can do better than PNG in terms of accessibility. Tags may or may not be abided by for some software, but the raw channel values are accurate in those files. [02:42] <troy_s> C) The pure transform of the RGB image into YCbCr is as high quality as possible based off of a 32 bit float EXR with correct SMPTE ranges and passed through a matrix +offset transform in float, then rounded. [02:43] <troy_s> So in terms of data values, we have exactly six to examine, all of which are 1:1. RGB (YCbCr) in the high quality source, and RGB in the dest. [02:44] <troy_s> (Nine if you wish to roll the input RGB image through the transform logic in FFMPEG.) [02:47] <michaelni> troy_s, lets start again from square 1, what bug is this about ? [02:47] <michaelni> you opened 2 bug reports [02:47] <troy_s> 1) RGB to YCbCr transforms using 709 (two scenarios) yield incorrect results. [02:47] <troy_s> (Likely due to rounding.) [02:47] <troy_s> Fair? [02:48] <troy_s> (At 8 bit is the initial testbed 444) [02:48] <troy_s> I discarded the first bug on your advice to reduce the first round of issues to single points. [02:49] <troy_s> (The longer target here is to try and fix YCbCr to RGB and vice versa at a higher level, but again I took your advice to break it down further.) [02:50] <troy_s> (FF still struggles with the transforms, despite being 1000x better since the 601 hard coded values were dropped.) [02:50] <michaelni> you opened 3 bug reports, one you closed, what is the difference between the other 2 ? [02:50] <troy_s> One is for studio range, the other full range. [02:51] <troy_s> (The first changed from testing the high level, to reducing it to very repeatable steps that are strictly FF's domain) [02:52] <troy_s> (The errors will accumulate and make it impossible for us to figure out exactly where FF can be repaired or improved.) [03:05] <michaelni> troy_s, should i delete the difference pngs from the tickets ? they dont represent the difference, so noone else wastes time from looking at them [03:07] <michaelni> troy_s, or is the other diff png correct ? (i checked just the one from 3801) [03:07] <troy_s> They both are [03:07] <troy_s> They show which blocks are different in the SMPTE patterns [03:08] <troy_s> In the case of studio range tests, which slivers of the grad are deviating. [03:08] <troy_s> (It is a binary diff basically) [03:08] <troy_s> (Both of them) [03:08] <troy_s> michaelni: Cooking, but will answer any and all questions as I can. [03:10] <michaelni> well, the one from 3801 certainly is not correct in the sense that multiple areas which are error free are displayed in different colors, i would have expected a color difference PNG to be 50% gray where there is no error not various different colors [03:11] <michaelni> also the png is described by: "A visual difference between the 8 bit SMPTE input versus theoretical and FFMPEG YCbCr output." [03:11] <troy_s> Anything red is a difference [03:11] <troy_s> Yes. [03:11] <michaelni> yeah iam starting to realize this now [03:12] <troy_s> From 8 bit correctly encoded PNG [03:12] <troy_s> Converted to YCbCr theoreticals [03:14] <michaelni> actual color difference would have been significantly more usefull [03:14] <troy_s> michaelni: Sorry. That can probably be possible via Compare in IM [03:14] <troy_s> michaelni: But I had trouble enough asserting no channel mangling. [03:15] <troy_s> michaelni: My apologies. I will work toward better compares for the next round. [03:15] <michaelni> i can do it in gimp, i am just a bit unhappy about having to reverse engeneer what the bug is about [03:16] <troy_s> michaelni: My issue. I thought I was communicating effectively. [03:16] <troy_s> michaelni: I could have just dumped the two RGB channel images, but that too probably would have been suboptimal. [03:22] <michaelni> troy_s, can you add the expressions you used to generate the correct ycbcr data to the tickets ? [03:30] <troy_s> michaelni: Absolutely. After dinner. [03:30] <michaelni> ok, thanks, no hurry [03:30] <troy_s> michaelni: The basic YCbCr transform is a matrix and offsets [03:30] <troy_s> (I even calculated the coefficients off of the original primaries and white points) [03:31] <troy_s> michaelni: (_And_ emailed Poynton about them, which he replied) [03:47] <troy_s> michaelni: The ODS is worth looking at [03:48] <troy_s> michaelni: (in the original report) as it shows a perfect theoretical breakdown line by line, generated off of nothing but the coefficients themselves as input data (only coefficients and RGB values as input) [03:49] <troy_s> michaelni: It generates the YPbPr (0..1 luma and -0.5 to 0.5 chroma) [03:49] <troy_s> michaelni: And builds off of that to get to both studio and full range data [03:49] Action: michaelni feels like it will be easier to reverse engeneer the equation from the data [03:49] <troy_s> (I included 601 and 240M in it as toggles as well, so that the other cases can be tested.) [03:50] <troy_s> It isn't [03:50] <troy_s> The equation for YPbPr is dead simple actually [03:50] <michaelni> no, you can write it down ? [03:50] <troy_s> Yep [03:50] <troy_s> But it really is a matrix [03:50] <troy_s> (With three offsets) [03:50] <michaelni> you can write it as matrix i dont mind at all [03:51] <troy_s> Sure. That matrix is in the ODS if you care. [03:51] <troy_s> (IIRC) [03:51] <michaelni> i dont think so [03:51] <michaelni> IIRC theres some idealized thing there [03:51] <michaelni> but this is about rounding +-1 [03:51] <michaelni> so we need to get Z256 -> Z256 not R->R [03:52] <michaelni> theres more than one way to round, not to mention dither and noise shaping [03:53] <troy_s> michaelni: Right then. Let me check my files. I can get an ODS quick [03:54] <troy_s> michaelni: I have a hunch that those minor precision errors are accumulating and making the drift quite significant at the tail end. [03:54] <troy_s> (FFMBC obviously had the same issues, but I chose to bring it upstream because FFMPEG now has a much more robust chain in place) [03:55] <troy_s> (Aside from the sRGB hard coded for XYZ transforms, that no one will care about until 2020 is all over. [04:14] <troy_s> michaelni: Who is coding the colorimetry aspect in swscale for XYZ? [04:14] <troy_s> michaelni: I would love to chat with them. [04:16] <troy_s> (I don't believe it is the AV folks is it?) [04:25] <michaelni> i think the xyz code had 3 or 4 contributors [04:26] <Hello71> oh hey, an op. haven't seen one of those in here in... uh. [04:26] <troy_s> michaelni: Hrm. Anyone lead point on it currently? [04:27] <michaelni> i think noone of them touched the code after their contribution [04:27] <troy_s> Drat. [04:27] <troy_s> That's a hugely important chunk of code WRT 2020 and other formats that are upon us. [04:28] <troy_s> (For example, transforming from 2020 to a standard sRGB display, or a wide gamut, etc. (Likely means a kick in the ass to ffplay to gut or fix the OpenGL code, not that I think that it is even possible thanks to chipset vendors bogging it up.) [04:29] <troy_s> michaelni: If anyone who is familiar with the ffmpeg base better than I is capable and interested, I am more than willing to help carve it out. My colorimetry side should be able to at least see breakages. [04:32] <michaelni> troy_s, iam happy to help with awnsering questions about swscale, also if you want to take over maintainership of the xyz code thats welcome as well [04:33] <troy_s> michaelni: Might take me a while to try and get familiar with it, plus god knows how many horrible questions (of the stupid sort) I'd end up nagging you with. [04:33] <michaelni> yes iam a bit affraid of that ;) [04:34] <michaelni> but i think your knowledge about that colormetry & xyz stuff is probably better than mine [04:35] <michaelni> and i have too much things i already maintain so i cant give xyz and surrounding code as much attention as it should get [04:37] <michaelni> and git grep xyz libswscale/ as well as git grep XYZ libswscale/ [04:37] <michaelni> should explain how it all works [04:40] <troy_s> Hrm. git grep... [04:40] <troy_s> michaelni: First thing first [04:41] <troy_s> michaelni: How difficult is it to add a -primaries option to swscale? [04:41] <troy_s> michaelni: I say primaries rather than coefficients because A) we can glean the luma coefficients from the primaries, and B) the primaries would be forward looking and cover every possible variation on 2020. [04:43] <michaelni> "adding" options is easy, doing something with them may or may not be easy [04:43] <michaelni> see libswscale/options.c [04:43] <troy_s> michaelni: I think there is enough sort of hint work in place (I need not deal with the bitshift nightmare, just worry about the numerical float values.) [05:25] <michaelni> troy_s, btw about the rgb->ycbcr, see rgb24ToUV_c & rgb24ToY_c & chrRangeToJpeg_c() & lumRangeToJpeg_c() [05:25] <troy_s> Erf. [05:25] <troy_s> Let me decode that. [05:26] <troy_s> What am I looking at? [05:26] <michaelni> troy_s, also you can use -cpuflags 0 to disable all asm/SIMD to make sure it uses the C code (which is easier for debuging) [05:26] <troy_s> Why are the two named differently? [05:26] <troy_s> Oh... so I can just pass that on the CLI? [05:26] <michaelni> -cpuflags 0, yes [05:26] <troy_s> My problem is that my ability to look at the gnarly bitshifts is very opaque to me [05:27] <troy_s> I can handle float values and even unsigned ints at times, but the shifting makes my head go bonkers. [05:27] <michaelni> well its fixed point no magic really, float would be too slow and too unpredictable for regression tests [05:27] <troy_s> michaelni: Two questions then, A) If there is a rounding issue it is in one of those four functions for studio and full respectively and B) where are the coefficient tables located now? [05:28] <troy_s> michaelni: Speaking of which, having a regression test that tests the YCbCr chain would be excellent. An image or even just values to known values (like the 25 odd in the SMPTE test) [05:30] <michaelni> input_rgb2yuv_table [05:30] <rcombs> ghuuuuuuuu, just tried to use ffmpeg for a testing application on Ubuntu, failing to remember that it's actually avconv [05:30] <rcombs> if I wanted avconv, I'd have asked for it [05:31] <troy_s> rcombs: Nightmare. [05:31] <rcombs> yeah [05:35] Action: michaelni falls asleep, troy_s ill look tomorrow if you had more questions [05:36] <troy_s> michaelni: I'll attach the ODS for the calculations as per your request. [05:36] <troy_s> michaelni: And we will go from there. [05:36] <troy_s> michaelni: Hopefully someone might have some tips on how to sort out the rounding issue. [05:36] <troy_s> (or whatever it is) [05:36] <troy_s> Then I'll tackle 601 and 240M just to make sure it's working correctly. [05:36] <troy_s> And then onto decoding. Yuck. [06:25] <t4nk949> Hello [13:45] <vklimkov> looking for a person familiar with ffmpeg on android. ping if interested [13:47] <Mavrik> I strongly suggest you ask a concrete question. [13:58] <vklimkov> Marvik: it's not about question. suggesting kind of job [18:40] <NeedFFMpegHelp> Hi, I need help with running FFMpeg with my Osprey card. [18:40] <NeedFFMpegHelp> http://pastebin.com/wrnANGkR [18:40] <NeedFFMpegHelp> ^^ That is my output [18:40] <NeedFFMpegHelp> ^^ And the command [18:51] <NeedFFMpegHelp> Does anyone have any idea's? [18:54] <sfan5> "Input/output error" isn't really helpful, but that's not your fault [18:58] <NeedFFMpegHelp> sfan5, Ok, is there something I can do to get a more helpful message? [18:58] <sfan5> probably not [18:59] <NeedFFMpegHelp> sfan5 - Thank you... If it where you, what would be the next step? [18:59] <sfan5> google the problem if not already done [18:59] <NeedFFMpegHelp> lol [18:59] <NeedFFMpegHelp> I see. [19:00] <NeedFFMpegHelp> So, it appears to be a crossbar issue, where FFMpeg doesn't support crossbar devices. [19:01] <NeedFFMpegHelp> I was hoping that someone would have a work around, or if FFMpeg plans on adding support for it? [19:04] <NeedFFMpegHelp> As there are people who agree that it would add a great deal of value to the project: http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=15&t=889 [20:02] <llogan> NeedFFMpegHelp: submit a feature request ticket on the bug tracker if there isn't one already. [20:42] <MarcelvanLeeuwen> Hi! [20:43] <MarcelvanLeeuwen> is it possible to encode a dts-ma to dts? [20:43] <BigArah> What's Up #FFMPEG!? [20:43] <BigArah> relaxed you up in here? [20:44] <c_14> MarcelvanLeeuwen: ffmpeg -i file -c:a dts -c:v copy outfile [20:44] <MarcelvanLeeuwen> okay thanks going to test [20:45] <BigArah> ffmpeg -i /path/file.mp4 2>&1 | grep "Duration"; [20:45] <BigArah> how would I be able to save that variable and use it again? [20:47] <llogan> BigArah: https://trac.ffmpeg.org/wiki/FFprobeTips [20:48] <llogan> shows a "better" way to get the duration. how you use the duration is up to you. [20:48] <BigArah> ok, I can run that from php then and pass it back [20:48] <BigArah> $duration = shell_exec(ffprobe...); [20:49] <BigArah> How do you feel about that llogan ? [20:50] <llogan> i know nothing of PHP, but the example shows eval usage which may be what you are looking for. [20:50] <BigArah> yeah [20:50] <BigArah> thanks man [20:50] <BigArah> I appreciate that [20:52] <NeedFFMpegHelp> llogan: thanks for the reply... I'll do that. What do you think the likely hood of that feature being added? [20:53] <llogan> NeedFFMpegHelp: hard to say. depends on developer interest. unless you supply a patch to ffmpeg-devel mailing list to implement it. [20:54] <llogan> or you can try placing a bounty. we have a bountysource account if you prefer using that service. [20:54] <NeedFFMpegHelp> Does that bounty source account accept BitCoin? [20:55] <llogan> i think so [20:55] <trn> Hello, would anyone be willing to help me with what is probably an extremely noobish issue? [20:55] <NeedFFMpegHelp> Very nice.. Can you give me the details? I'd like to look into that. [20:55] <trn> I've complete info with question and all relevant info in pastebin at http://goo.gl/m3p9Gs [20:56] <llogan> NeedFFMpegHelp: i think you just make a ticket on the FFmpeg bug tracker, then wait about 15 minutes, then it will appear bountysource. you then tell it how much of a bounty you're willing to spend to get it implemented. [20:56] <trn> Introduction info at the top, nearly unmolested output (only removed irrelevant repeating data so it's 1000 lines instead of 100000), and I hope I've woreded the question correctly. :) [20:57] <trn> I'm going to be using the ffmpeg libraries, but want to know what I'm doing in the cli before converting to C. (I've also never used ffmpeg before, so please excuse my ignorance.) [20:57] <NeedFFMpegHelp> llogan: Thank you, I see that BountySource takes BitCoin, so that is perfect. Thanks so much. [20:58] <trn> Also I hope it's enough info, this is just a few lines of a isolated troublesome function in a prototype application thats several thousand lines of code running across several hundred machines. [20:58] <llogan> NeedFFMpegHelp: let me know how you like or dislike it. i don't think anyone has used it yet, but the FFmpeg account hasn't been there for very long. [20:59] <trn> So before I embarass myself on the mailing list, anyone want to take a look? [21:01] <llogan> trn: that's a lot of info. if you ask on mailing list you can omit everything above line 728. [21:02] <llogan> except for a few sentences explaining the issue [21:02] <trn> Everyone who asked questions on the list that I looked at was almost always asked for more information, so I'm just trying to be complete. [21:03] <llogan> yes, i understand, and users often do the opposite. [21:03] <Primer> Is ffmpeg now avconv? [21:03] <sacarasc> No. [21:03] <sacarasc> avconv is from a fork. [21:03] <sacarasc> ffmpeg uses ffmpeg. [21:03] <Primer> So why are distros now using avconv? [21:04] <trn> And relevant output is difficult to isolate when they ask for "complete" output when I have multiple ffmpeg processes running and sending output and accepting input via different network and IPC mechanisms. [21:04] <Primer> Sorry if this is a sore subject [21:04] <sacarasc> Because they don't ship ffmpeg, but a form called libav. [21:04] <sacarasc> *fork [21:04] <trn> llogan: 999 - 1020 is my issue :) [21:04] <Primer> but I needed to splice a video and have been doing that with ffmpeg for a long time, but my newly installed Linux mint 17 has no ffmpeg [21:04] <llogan> because the Debian maintainer (who was formerly a FFmpeg developer) switched to the fork, and then forced his decision to the users. [21:05] <Primer> wow [21:05] <llogan> You can download a static build. http://ffmpeg.org/download.html [21:05] <llogan> or compile http://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [21:05] <sacarasc> Or build yourself: https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [21:05] <Primer> lovely, thanks [21:06] <Primer> hopefully this will work, whereas avconv is failing at this: avconv --ss 00:05:32 -t 30 -i file.mov clip.mov [21:07] <Primer> it seeks to the correct location, but the duration isn't enforced, it just goes to the end [21:07] <llogan> trn: i can't really look at it now, but if you post on ML maybe I (or someone else) can. make sure to provide your details in the message itself instead of a link to pastebin [21:07] <llogan> also see https://trac.ffmpeg.org/wiki/MailingListEtiquette if you're feeling nervous about the ML. [21:08] <Primer> nope, it also just goes to the end [21:08] <Primer> So any thoughts? ffmpeg -y -ss 00:07:06 -t 30 -i REC_0004.MOV -c:v copy -c:a copy clip.mov [21:08] <trn> llogan: I will do so. I'm more nervious about posting on the mailing list because this is part of a large application and I don't really want an archived record forever available of what an idiot I am. :) [21:09] <Primer> http://pastie.org/9420798 [21:09] <llogan> it's not idiotic. and if you don't want a record just use a fake name. [21:10] <llogan> Primer: can you provide the input? [21:10] <Primer> I did [21:10] <Primer> err [21:10] <Primer> you mean the file? [21:10] <Primer> it's huge :) [21:11] <Primer> -rwxr-xr-x 1 daniel daniel 1.1G Jul 25 11:18 REC_0004.MOV [21:11] <trn> llogan: Thanks, but if my boss knows I wasted 2 days when all I needed was "-set_magic_bs 1" or "#define MAGIC_BS 1" I'm going to lose prestige in the office. [21:11] <Primer> If I were to upload that from here, my co-workers would be pissed [21:12] <trn> I'll try to do something tho. [21:12] <llogan> Primer: is the issue only with this particular input? [21:12] <BigArah> hey llogan I got it to work with shell_exec() and the old grep command passing it as a value [21:12] <BigArah> thanks for your help [21:13] <Primer> llogan: I've not made clips from videos in a very long time. I suppose I can test with some other input. [21:13] <Primer> http://mobius-actioncam.com/ is what made the video, if you're curious [21:13] <trn> Also one unrelated quick questions before I go read the source code.... is the ffmpeg-bundled NUT container handling considered inferior or superior to linking mplayer/ffmpeg git libnut? [21:14] <trn> Because I'm using NUT rather extensively to pipe around arbritrary multimedia data both IPC and around a cluster. [21:17] <NeedFFMpegHelp> ! [21:19] <trn> My only other option for intra-application cluster transport and IPC is MPEG-2 Part 1 transport streams, which seems a lot more complex. [21:19] <Primer> llogan: I'll upload the source file when I get a chance and get back to you. Thanks for your interest in this matter. [21:19] <trn> Because some parts of the application may cut, splice, and later concatenate streams without parsing them, and NUT passed all the tests. [22:16] <FrEaKmAn_> hi all.. [22:16] <sfan5> hello [22:16] <FrEaKmAn_> if I want to convert multiple files to mp4 [22:16] <FrEaKmAn_> and then concat those files [22:16] <FrEaKmAn_> is it good to have same framerate? [22:16] <FrEaKmAn_> convert to same fm? [22:16] <sacarasc> I think it won't work without the same frame rate. [22:17] <FrEaKmAn_> ok.. [22:18] <FrEaKmAn_> so for vcodec I will use libx264 and for audio aac [22:18] <FrEaKmAn_> but what I should define for framerate? [22:19] <sacarasc> What different ones do you have? [22:19] <FrEaKmAn_> I don't know.. users upload them [22:19] <FrEaKmAn_> and then I convert and concat [22:25] <sacarasc> Do you know where they might be sourcing from? Will all the videos they want combined be from the same source? [22:25] <trn> FrEaKmAn_: I am working under similar circumstances and we deal with an internal 'standard' of 720p 25fps and convert everything to that, upscaling smaller resolution streams, upscaling larger ones, and then sync'ing the video to the audio. [22:26] <FrEaKmAn_> trn: how do you sync video with audio? [22:26] <FrEaKmAn_> sacarasc: not really [22:26] <trn> FrEaKmAn_: Usually we get 24, 29.97, and 60 in the wild for input. [22:27] <trn> FrEaKmAn_: Right now by letting vsync handle it. [22:27] <FrEaKmAn_> what do you use for parameter? [22:28] <trn> 25 fps just seemed like a good intermediate. I'm in no position to tell you really, I just started using the ffmpeg stuff this week. [22:28] <trn> And we're using the libav* ffmpeg libraries directly. [22:28] <FrEaKmAn_> ok [22:31] <trn> I know async format is depreceated, but I would *assume* something like: -async 1 -vsync 2 -r:v 25 for the output would work. [22:33] <trn> Where you could probably then concat the produced outputs. [22:34] <trn> I would assume you could just use multiple -i $in1 -i $in2 etc and do it all in one swoop, unless you have some situation that makes that impractical. [22:36] <FrEaKmAn_> trn: thanks [22:53] <zybi1> hi [22:54] <zybi1> how to re-encode a 70 mbit mkv-file to a about 35 mbit mov file please? [22:59] <trn> Quick licensing question ... If I'm linking to ffmpeg libraries on an application that runs only on my server and the servers are under control of our organization, and ffmpeg is compiled --enable-gpl --enable-nonfree, but we are not distributing anything, the server software dynamically linking to the compiled ffmpeg can remain propriatary, correct? [23:00] <trn> Because there is no distribution of the server source in binary or source format, only server output, which I assume can't be reasonably considered a derivative work :) [23:04] <JEEB> well, due to the --enable-nonfree flag you wouldn't be able to distribute binaries of that configuration anyways, as it bars you from doing it (licenses being incompatible) [23:06] <JEEB> with regards to other points, follow the license(s) involved. IANAL. Generally if you are not distributing any binaries and not dealing with the A* gnu licenses, you should be OK. [23:06] <trn> That was what I thought too. :)_ [23:10] <Primer> llogan: I was able to limit the recording time by specifying -t after -i [23:12] <llogan> Primer: i'm not sure why before -i did not work as expected, although the behavior of -t differs depending on location [23:12] <Primer> right [23:12] <Primer> the docs is what lead me to try -t after -i :) [23:12] <Primer> See? PEOPLE SOMETIMES READ THE DOCS! [23:13] <llogan> zybi1: https://trac.ffmpeg.org/wiki/Encode/H.264#twopass this should give you a general idea [23:13] <YaMoonSun> I downloaded an .mp4 and I want to extract the .aac codec from it and contain it via .m4a or make a precise encode via .mp3 - I keep running into errors. http://i.imgur.com/7gdVVhU.png [23:15] <llogan> ffmpeg -i input -map 0:a -codec:a copy output.m4a OR ffmpeg -i input -vn -codec:a copy output.m4a OR ffmpeg -i input -c:a libmp3lame -q:a 4 output.mp3 [23:17] <YaMoonSun> I used the -strict - 2 or whatever and it preceeded to give me an error. Did I just have the command wrong, or? [23:17] <YaMoonSun> Tyvm btw [23:18] <zybi1> thanks llogan [23:19] <llogan> YaMoonSun: option placement matters. you're attempting to apply -strict to the input (to the decoder). [23:21] <YaMoonSun> I thought placing commands prior to -i would affect thee entire encode and not just the input - Every time I read out whole directory paths I give myself a headache. I need to start replacing them with variables to simplify this. [23:22] <llogan> ffmpeg [options for input] -i input [options for output] output [23:22] <YaMoonSun> So the strict command goes where? [23:23] <YaMoonSun> Why can I not open This Is A File.mp4, but I can open thisisafile.mp4? [23:24] <llogan> -strict, in your case, is an output option. [23:24] <llogan> i don't know why you can't open the file. you haven't provided any context [23:24] <llogan> but you don't need to use strict because your input is already AAC, so you don't need to re-encode [23:25] <YaMoonSun> So my entire command was wrong for the function that I was attempting. I was trying to encode when I should have been trying to extract. [23:25] <YaMoonSun> Damn [23:26] <trn> llogan: Maybe you can help me without reading my 'novel'. :) [23:27] <trn> Is there some good explanation anywhere of PTS/DTS/timebase/etc. And how that applies to either container format and/or data pakcets in the streams in the container. [23:28] <trn> It took forever to wrap my head around the fact that both the stream and the container can specify aspect ratios and they might not always be the same. :) I'm a slow learner. [23:30] <trn> But now for the player side of things, I determine screen aspect ratio, and can properly zoom into a 16:9 stream inside a 4:3 container on a 16:9 output to avoid having big black borders ... [23:30] <trn> since it would otherwise show both letterboxed and pillarboxed. [23:37] <YaMoonSun> That .m4a command worked nice, but the .mp3 reduced the bitrate from 192kbps to 141kbps [23:45] <c_14> YaMoonSun: judging by your question, you probably want to use -c:a copy instead of -acodec aac [23:46] <c_14> Also, when converting to mp3 you'll probably want to define a bitrate or quality. [23:47] <YaMoonSun> Cheers - I'm trying to figure out how to edit with Audacity without losing quality now =/ [23:48] <sacarasc> If the output is lossy anywhere along the line, you've already lost quality. [23:48] <c_14> You'll always lose quality when encoding from a lossless codec to a lossless codec. [23:48] <c_14> s/lossless/lossy [23:49] <c_14> No idea why I said lossless there... [23:49] <trn> Question on prototyping using the ffmpeg cli vs. using the libraries directly. [23:49] <trn> In my application I'm reading from buffered sockets and using O_NDELAY. [23:50] <trn> Is there any way to replicate O_NDELAY behavior when using a Unix named pipe as ffmpeg input? [23:51] <YaMoonSun> The m4a is flawless (considering the source), but unable to be opened in audacity, so I don't know what to do if I want to crop parts of the audio out without losing quality. [23:52] <c_14> You can crop directly during the extraction from the mp4. [23:52] <c_14> Or while converting to mp3. [23:53] <YaMoonSun> Within ffmpeg? =0 [23:53] <c_14> https://trac.ffmpeg.org/wiki/Seeking%20with%20FFmpeg [23:55] <YaMoonSun> I don't know if I should be excited or scared. The program is amazing, but pretty complex. Going to take weeks to get the commands memorized. [00:00] --- Sat Jul 26 2014

1 0

ffmpeg-devel.log.20140724
by burek 25 Jul '14

25 Jul '14

[00:15] <J_Darnley> What pixel format are you giving as input? [00:15] <J_Darnley> Green sounds like there are chroma planes set to 0. [00:16] <Orix> AV_PIX_FMT_YUV420P [00:18] <J_Darnley> Then either use a grayscale one or set the chroma to 128 [00:21] <ubitux> smarter: i think you can extract the planes with something like ffmpeg -i ... out%d.Y or something [00:21] <Orix> i wanted to avoid passing the grayscale format into the swscale but that seems to work [00:22] <smarter> ubitux: hmm [00:28] <Orix> is there a way to check a flag for grayscale? I see in Rad Tools it says "File type: grayscale" [00:28] <Orix> anyway to check that with ffmpeg? [00:29] <J_Darnley> Do you mean "check a file"? [00:30] <J_Darnley> ffprobe and ffmpeg can both print info about files. [00:30] <J_Darnley> They should both print the pixel format found in a file [00:30] <J_Darnley> They won't check content though. [00:31] <Orix> Using the API, not using ffprobe or ffmpeg. The api is reporting the pixel format AV_PIX_FMT_YUV420P but I get a green layer on top of the video. If I force it to use AV_PIX_FMT_GRAY8 as the source format it looks fine (but the other videos become gray scale) [00:35] <J_Darnley> In that case I don't know. [00:45] <iive> Orix: what is the original codec used by the file? [00:45] <iive> is it possible that you are not using correctly the chroma planes and somehow zero them. [01:09] <Orix> iive: the codec is bink and its reporting as AV_PIX_FMT_YUV420P [01:11] <iive> Orix: do you get green if you play it with ffplay? [01:11] <Orix> let me try [01:11] <Orix> Yes, I get green with ffplay [01:13] <cone-109> ffmpeg.git 03Diego Biurrun 07master:c3c96deb5f8c: fft-test: Check memory allocations [01:13] <cone-109> ffmpeg.git 03Michael Niedermayer 07master:074bb7515e33: Merge commit 'c3c96deb5f8cbbdb700ba97920ceedddacb5dcb9' [01:13] <cone-109> ffmpeg.git 03Michael Niedermayer 07master:629e5e28f70a: avcodec/fft-test: fix memory alloc checks [01:14] <iive> then it might be codec problem, kshishkov is the person who knows most about bink, he is however libav developer, if he is ineterested in bink samples at all. [01:14] <iive> do you have other files that have same issue? [01:14] <Orix> nope, just that one [01:15] <Orix> and the only difference i can see is Rad Tools tells me that format is Grayscale while the other is Color [01:16] <iive> if I remember correctly, libavcodec image allocations fills chroma samples with 0x80, specifically to give gray image instead of green one. [01:17] <iive> it might be bug, or improper codec handling. [01:17] <Orix> i'm pretty fresh to ffmpeg so i haven't gone digging into the implementation yet [01:17] <Orix> makes me feel better that ffplay shows it as green as well [01:17] <iive> do you have bink encoder? something that could produce small sample file? [01:18] <Orix> hmm, no not off hand. I can look at generating one though [01:19] <iive> well, in any case it would be good idea to fill bugreport and provide sample, to ffmpeg or/and libav [01:21] <michaelni> yes, please open a bug report at trac.ffmpeg.org [01:21] <Orix> alright, will do. i'll see about getting a sample file [01:22] <michaelni> thanks [01:35] <kierank> any suggestions about how to make the opus parser special case ts input [01:43] <cone-109> ffmpeg.git 03Diego Biurrun 07master:6133d65861f2: fft-test: K&R formatting cosmetics [01:44] <cone-109> ffmpeg.git 03Michael Niedermayer 07master:02ea384abd8e: Merge commit '6133d65861f265d429ade53e910fe877c95a1615' [01:51] <BBB> what is max_transform_hierarchy_depth_intra/inter in the sps? [01:51] <BBB> I mean, I can see what the code does, but its all crypto-nonsense to me; what is the english-language meaning of that element? [02:00] <cone-109> ffmpeg.git 03Diego Biurrun 07master:d0de7330b973: fft-test: Pass correct struct members to (i)dct functions [02:00] <cone-109> ffmpeg.git 03Michael Niedermayer 07master:ab0f78ea7777: Merge commit 'd0de7330b973b354e2ceb0b99fd545b868b0a9c4' [02:16] <llogan> git bisecting. did understand why it didn't pinpoint regression after many bisects. realized i was running ffmpeg instead of ./ffmpeg. beer desire increased. [02:16] <llogan> *didn't [02:22] <cone-109> ffmpeg.git 03Diego Biurrun 07master:ce2e858f5b34: http: K&R formatting cosmetics [02:22] <cone-109> ffmpeg.git 03Michael Niedermayer 07master:8b59ab1af0dd: Merge commit 'ce2e858f5b3416c2d54f7f8c14e901f75c48b785' [05:22] <cone-109> ffmpeg.git 03Lukasz Marek 07master:97392553656a: lavfi/src_movie: remove frame from priv context [05:22] <cone-109> ffmpeg.git 03Lukasz Marek 07master:f7539562522b: lavfi/src_movie: minor simplifications [08:05] <kurosu_> BBB: that's the log2 scale between a intra/inter CU and the transform blocks it can contain [08:06] <kurosu_> another view is, there's a quadtree of transform syntax (last coeff, cbp, etc), and it's its level [08:06] <kurosu_> the partitioning of the coding block may have an impact but I don't recall well [08:07] <kurosu_> I'd say it's like the 8x8 transform flag in avc except it can go deeper [08:07] <kurosu_> regarding the threading api, I don't know [08:08] <kurosu_> big blocks are probably required to need it, but ffvp9 is so much faster (and those probabilities being reused between frames) that I wonder [08:09] <kurosu_> I would have thought it could be reused there, but I didn't plan to perform a study on how reusable it is [08:10] <mraulet> might be good to encode a sequence with max ctb size == 16 [08:10] <kurosu_> mostly, you told me you had tested something similar in ffvp9 and it didn't help - and I wanted to see for myself [08:11] <kurosu_> mraulet, hi [08:11] <mraulet> hi [08:11] <kurosu_> I know you have another threading for shvc, didn't quite check how compatible they were [08:12] <kurosu_> but it works in openhevc and the few sequences I tested [08:12] <mraulet> it is from my point of view [08:12] <kurosu_> (they were main) [08:12] <mraulet> I havent tested it but I dont see any compatibilty issue with what you propose [08:13] <kurosu_> you were more concerned about threading over slices/tiles and over frames [08:13] <mraulet> yes [08:13] <kurosu_> the 2 being mixed could not play well [08:13] <mraulet> but this is also what we have in shvc [08:13] <mraulet> we do have 2 decoders running in parallel [08:14] <kurosu_> unfortunately, neither ffmpeg nor the test sequences I had can help evaluate it [08:14] <mraulet> using the FB technology to start as early as possible the upper layer [08:14] <mraulet> me too [08:14] <mraulet> people are using big blocks :-) [08:14] <mraulet> it helps in compression efficiency [08:15] <kurosu_> anyway, that api mostly works with high threading - not much of a difference on dual cores [08:16] <mraulet> I think I will port it into openhevc and see what it gives on samples I have [08:17] <kurosu_> beware that you introduced and used a thread_frame_ctx in lieue of a thrad_ctx [08:17] <kurosu_> needs to be changed in the new api code [08:20] <kurosu_> I was wondering if a 2-steps api could help - seeing how the following line lags at least one CTB after the current one [08:20] <kurosu_> but that's mostly hevc, so I didn't bother [08:22] <kurosu_> other question I asked myself: how reusable was x265 asm (I'm thinking of MC), except for the gpl they have instead of lgpl [08:24] <mraulet> the most interesting part from x265 are idcts and sao asm :-) [08:25] <kurosu_> it seems they only have up to 8x8 ? [08:25] <kurosu_> dct16/dct32 are the most consuming ones iirc [08:28] <mraulet> yes they are [08:29] <mraulet> especially idct32 [08:29] <mraulet> you need a compromise in between unrolling the code and putting loops [13:03] <BBB> kurosu_: I guess I see, if e.g. the CU size is 32x32, then this says how many levels deeper the TU can go (e.g. if its only 32x32 or 16x16, its 1); that would then mean that for a 8x8 CU, both 8x8 and 4x4 TUs are available, right? [15:27] <cone-739> ffmpeg.git 03Carl Eugen Hoyos 07master:badcd3da81ac: Print a warning instead of requesting a sample for an invalid exif tag. [16:32] <cone-739> ffmpeg.git 03Timothy Gu 07master:c508adb22366: hevc: propagate error code from set_sps() [18:26] <cone-739> ffmpeg.git 03Carl Eugen Hoyos 07master:ef21b3d38e61: Ignore xing number of frames if the file is larger than indicated. [18:26] <cone-739> ffmpeg.git 03Michael Niedermayer 07master:183bbc2af1dc: Merge remote-tracking branch 'cehoyos/master' [18:26] <cone-739> ffmpeg.git 03Michael Niedermayer 07master:3c6e220a8c2d: avformat/matroskaenc: Factor mkv_start_new_cluster() out [18:27] <cone-739> ffmpeg.git 03Michael Niedermayer 07master:7923aa0fba6b: avformat/matroskaenc: Start new cluster if relative timestamp could not otherwise be stored [19:04] <cone-739> ffmpeg.git 03Nidhi Makhijani 07master:436ced244fad: mpegenccontext: Remove unused opaque pointer [19:04] <cone-739> ffmpeg.git 03Michael Niedermayer 07master:ce8e27e0ed2d: Merge commit '436ced244fadcde2c0b925627920e84b25482542' [19:43] <kurosu_> BBB: yes, for 32x32, given a minimal transform size of 4, it can go up to 3 [19:43] <kurosu_> and of course is at least 1 for 64x64 [19:45] <kurosu_> I think TUs are not allowed to cross PUs, so anything besides 2Nx2N also forces 1 at least iirc [20:11] <kurosu_> BBB: in any case, maybe you can ask on #x265 too - I suppose you no longer jeopardize the soul of your firstborn by doing so ;) [20:17] <ubitux> Timothy_Gu: you didn't fix WebVTT decoder into encoder in the Release Notes. [20:17] <ubitux> i believe i pointed that out [21:34] <Timothy_Gu> ubitux: oh shoot. I didn't read your mail completely. You can commit a fix for this right now. [21:34] <ubitux> i prefer if you fix it yourself [21:35] <ubitux> and in the branch in particular because we link to it [23:41] <cone-135> ffmpeg.git 03Michael Niedermayer 07master:d44b8f0a4724: avformat/matroskaenc: print a warning when the relative timestamp wouldnt fit in 16bit [23:41] <cone-135> ffmpeg.git 03Michael Niedermayer 07master:383a04a12773: avformat/mux: keep AVPacket fields consistent in ff_write_chained() [23:41] <cone-135> ffmpeg.git 03Michael Niedermayer 07master:351b22caae65: avformat/mux: support re-interleaving packets in ff_write_chained() [23:41] <cone-135> ffmpeg.git 03Michael Niedermayer 07master:66ae994c544a: avformat/segment: re-interleave packets if needed [00:00] --- Fri Jul 25 2014

1 0

ffmpeg.log.20140724
by burek 25 Jul '14

25 Jul '14

[00:02] <hilacha> someone have knowledge about the g723.1 codec? i need to record it passsively, but it appears like the g723.1 stream inside the packets are invalid using ffmpeg to decompress it :( [00:02] <hilacha> the packets are 24bytes length like in the specification... [01:29] <relaxed> llogan: thanks for looking into it [02:19] <saratoga> does ffmpeg have a reasonably optimized routine for unpacking 12 bit pixels/samples into 16 or 32 bit? [03:34] <t4nk996> Hi I am trying to use ffmpeg library to seek an audio file (DASH FMP4 AAC MED). The avformat_seek_frame and avformat_seek_file are somehow failing to do so. I checked online and it seems like there have similar bugs in this context but I wasn't able to find an exact solution. Can someone help me with the same ? [04:04] <troy_s> Anyone getting the types may not be defined in 'sizeof' expressions in C++ mixed code? [04:05] <troy_s> (bprint) [04:10] <troy_s> Is fflogger a bot? [04:12] <troy_s> http://pastebin.com/C88vFp91 [04:12] <troy_s> Seems silly pasting compiler output. [04:31] <t4nk996> With reference to seeking using ffmpeg: I have got my audio in place but my video shows fray frames initially and starts back from the beginning once the file is decoded though audio is correct. Can anyone tell me how to let decoder not produce the starting frames for the video once it loops back. Can I flush these frames .. ? [04:31] <t4nk996> *gray [04:40] <troy_s> t4nk996: Hrm. Are those bi-directionally encoded frames I wonder? [04:41] <t4nk996> troy_s : Yep! [04:41] <troy_s> t4nk996: Hard to work around that no? The decoder needs to decode up to the point that it can decode those frames. [04:42] <t4nk996> Which means once it reaches to end it starts decoding the starting frames ? Right ? [04:42] <troy_s> t4nk996: I'm not an expert on B frames, but it can be an arbitrary number of frames. [04:43] <troy_s> t4nk996: So if you had five frames, and the bidirectionals were up to 3, you'd need to hit frame four, and then frames 1, 2, and 3 can be decoded correctly. [04:43] <troy_s> t4nk996: If you have the encoding in your control, maybe you can control where the Bs begin. [04:44] <t4nk996> Oh you are right .. I can actually identify the keyframes and tell th deocder to just seek to that frame and after that do a linear skipping [04:44] <t4nk996> But what I am worried about is the intial frames coming up after the end .. [04:45] <t4nk996> Cz I don't see end frames to depend on the initial frames [04:45] <t4nk996> Don't have much idea though.. [04:48] <troy_s> t4nk996: Explain? [04:48] <troy_s> t4nk996: Following our example, you are saying the three frames end up being decoded at the very end? [04:49] <t4nk996> So I have a seperate audio and video files which I am decoding using separate ffmpeg decoders. I want both of them to seek 30 secs in each file. [04:50] <t4nk996> So now when I see the output both the audio and video starts at 30 initially [04:50] <troy_s> Ah. So the chances that you dive in at the 30 second mark and land on a series of B frames, correct? [04:50] <t4nk996> But once the video comes to the end. The video deocder is producing the initial frames [04:50] <troy_s> Yes. [04:50] <t4nk996> Yep [04:50] <troy_s> Because that's one of the only ways it makes sense. [04:51] <t4nk996> you are right ! [04:51] <troy_s> This always happens with bidirectional frames. [04:51] <t4nk996> but why the inital frames on end [04:51] <troy_s> There's a little known factoid that when you decode frame by frame that there are often extra frames at the end, and you have to loop until it's empty. [04:51] <troy_s> Because it is the only way to decode efficiently. For example, if you dive in and your first three are bi-directional, you can't decode those until the fourth say. [04:52] <troy_s> But at the fourth, you are decoding a whole frame, so what do you do with the frame? [04:52] <troy_s> If you toss it and offset it, now your whole thing is out of whack by three frames. [04:52] <troy_s> (your fourth frame becomes your first if you do this.) [04:52] <troy_s> So that too is problematic. [04:52] <troy_s> Make sense? [04:52] <t4nk996> yep ! [04:53] <t4nk996> It makes a lot of sense. I get your point ! [04:53] <t4nk996> Thats so true ! [04:53] <troy_s> t4nk996: This happens even when decoding from the beginning. https://github.com/chelyaev/ffmpeg-tutorial/issues/7 [04:54] <t4nk996> Right ! Okay so can I decode DASH audio using AVSEEK_FLAG_FRAME [04:55] <t4nk996> It looks like the header file says that it might fail for some demures [04:55] <t4nk996> I mean frame by frame [04:55] <troy_s> t4nk996: Actually that's an option... to decode up to your 30 second marker. [04:55] <troy_s> I still think the bi-directional frame issue can catch you out though, as there's no way to _not_ skip proper decoding order. [04:57] <t4nk996> Oh Right .. I'll try the frame by frame option and try to find a way out for not landing on B frame. Thanks a lot for helping. I'll definitely let you know if I get through this :) [04:57] <troy_s> t4nk996: I'm not sure that bi-predictive frames can necessarily be solved to be honest. [04:57] <troy_s> t4nk996: Simply because it is a case of "I cannot build this frame without a frame somewhere after it." [04:58] <t4nk996> But keyframes should have dependencies right ? [04:58] <troy_s> t4nk996: Which means that every single call to mustering a frame will fail until it gets to the frame that can decode the whole frame. So the question is ultimately what to do with those skipped beats. [04:58] <t4nk996> So I can actually land there ! and do linear seeking henceforth [04:58] <t4nk996> ^no [04:59] <t4nk996> *I mean key frames should have no dependencies [04:59] <troy_s> Ah. So you mean seek to the first picture frame near your 30 second marker? [04:59] <troy_s> That would work I think. [04:59] <t4nk996> Yep ! [04:59] <troy_s> At the cost of being somewhat potentially irregular. [05:00] <troy_s> (Doubly so if the encode in questoin is a long take with no motion where no such P frames are likely to be placed under heavier compression.) [05:00] <troy_s> There simply isn't a super clean way of dealing with it I fear. [05:01] <t4nk996> Haha ! True ! :) [05:01] <t4nk996> Thanks a lot though. It was a a lot of help ! [05:01] <troy_s> Glad I could help. One of the few things I have bumped into trying to get frame precise decoding. [09:23] <luc4> Hello! If I wanted to convert a video from 1080p to 1080i, should I simply transcode adding -flags +ildct+ilme? Is this correct? [09:29] <dduvnjak> i'm doing a mix of several input audio files into single one, or more precisely several background files + one main one [09:30] <dduvnjak> ffmpeg.exe" -y %LIST% -i "Avicii-WakeMeUp.mp3" -filter_complex "amix=inputs=%size%" -ar 44100 -ac 2 -q:a 1 "mix.mp3" [09:30] <dduvnjak> i did a volume normalization of each of them [09:30] <dduvnjak> however, when i mix the, the output file's volume drops significantly [09:30] <dduvnjak> is there way this can be avoided of fixed? [09:50] <zenderz> hi all [09:51] <zenderz> anyone have any experience with the smooth streaming format? seems the muxer has recently been added to ffmpeg. but there are no docs yet [09:54] <sfan5> zenderz: would be more helpful if you could state your exact problem [09:56] <zenderz> using the command : [09:57] <zenderz> ffmpeg -y -i testinput.mpg -c copy -b 10000K -f smoothstreaming output [09:57] <zenderz> seems to generate the manifest and video/audio fragments [09:58] <zenderz> however it does not play on any reference smooth streaming players. And I want to understand the command options so i can customise the output [10:00] <sfan5> zenderz: command options: http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavformat/smoothstreaminge… [10:00] <sfan5> (after the static const AVOption options[] = {= [10:00] <sfan5> (after the static const AVOption options[] = {) [10:02] <zenderz> thanks. Having a look now [10:04] <ubitux> ffmpeg -h muxer=smoothstreaming [10:34] <zafu> hi, is there a video stabilisation filter in ffmpeg? [10:35] <zafu> or what is my best option to stabilize a jerky smartphone video? [10:38] <ubitux> yes, vidstab [10:43] <zafu> is it a utility? can't find it on debian [10:44] <ubitux> http://ffmpeg.org/ffmpeg-filters.html#vidstabdetect-1 [10:44] <ubitux> http://ffmpeg.org/ffmpeg-filters.html#vidstabtransform [10:44] <zafu> thanks [10:55] <zafu> I don't get why there is an input and output file at the vidstabdetect stage [10:56] <zafu> where it's only detection for the second stage, should I throw out the output video from that stage? [10:58] <zafu> ah, "dummy.avi" :) now I get it [11:12] <luc4> Hello! Anyone who knows how I can converto from 1080p to 1080i? [11:13] <Mavrik> uh [11:13] <Mavrik> you really shouldn't :) [11:14] <Mavrik> using interlaced video filter and setting x264opts to tff=1 should do it tho [11:24] <ubitux> zafu: should probably be replaced with -f null - instead [11:24] <ubitux> to avoid a pointless encode & muxing [11:24] <ubitux> it will be faster [11:25] <ubitux> the dummy.avi is because there is a show=1 [11:26] <zafu> what use is the show=1 ? [11:27] <zafu> in any case it works very well, impressive [11:31] <luc4> Mavrik: I need it for development. [11:32] <Mavrik> luc4, I suggest you find an interlaced video then, doing progressive -> interlaced conversion isn't something you ever want to do and it usually does not result in same thing you get from a camera [11:34] <luc4> Mavrik: actually I'm interested in performance of decoding. Not very interested in the image itself. Do you think using a converted video will lead to different results? Cause I need a few samples at different resolutions so it would be simpler to convert as needed. [11:35] <Mavrik> luc4, yes. [11:36] <Mavrik> alot of encoders will just cop-out and give you progressive video tagged as interlaced [11:36] <Mavrik> you'll also get two fields at the same PTS time out of that [11:36] <Mavrik> get a proper 1080i60 video and rescale it to what you want [11:37] <luc4> Mavrik: so rescaling is ok and can I also remux to something different without affecting the result? [11:37] <Mavrik> yes, as long as you rescale and reencode while keeping fields as they are [11:37] <Mavrik> (scale filter has a flag for that, encoders as well) [16:20] <nicholaswyoung> Can ffmpeg output flac to a pipe? [16:21] <Fjorgynn> pipe? [16:22] <Fjorgynn> it can output flac to a file [16:22] <sacarasc> ffmpeg -i input.wav -c:a flac -f flac - [16:22] <Fjorgynn> what are you pipeing to? [16:22] <sacarasc> Or: mkfifo ilikecheese && ffmpeg -i input.wav -c:a flac -f flac ilikecheesee [16:23] <Fjorgynn> sacarasc: never heard of mkfifo before :o [16:24] <sacarasc> Named pipes, woo! [16:25] <Fjorgynn> :D [16:39] <kaotiko> hi [17:11] <luc4> Mavrik: sorry, I'm having a hard time trying to find out the correct option to preserve interlaced streams. If I have an interlaced video, and transcode to lower the bitrate, do I need to add an option or will ffmpeg preserve that? [17:13] <c_14> luc4: As long as you don't add a deinterlacing filter it should remain interlaced. [17:13] <luc4> c_14: so progressive -> progressive and interlaced -> interlaced means I don't need to add anything? Thanks! [17:14] <c_14> ye, it should usually just work [17:14] <luc4> thanks! [17:15] <klaxa|work> does interlaced have any advantages over progressive? [17:16] <c_14> On screens that can interlace (crts) it increases the perceived framerate. [17:16] <c_14> (without using extra bandwidth) [17:17] <klaxa|work> ah [17:17] <klaxa|work> cool [17:21] <c_14> On screens that can't interlace (pretty much everything else) it gives you funky lines. (without using extra bandwidth) [17:28] <vklimkov> looking for a person hands on ffmpeg and android. if interested ping me privately for details [17:43] <Peter_Occ> When ffmpeg adds a timestamp to a video stream, it starts with a time that's six seconds ahead of the real start time, The time then freezes for six seconds until the video catches up to the timestamp, and proceeds with the correct timestamp. Has anyone ever found a way to get the timestamp to start with the correct time? I thought maybe I could have no timestamp and given the starting... [17:43] <Peter_Occ> ...time, add it to the video after the capture is complete. Is that something ffmpeg can do? [17:56] <Peter_Occ> Ok will do, [18:04] <Peter_Occ> Here is the command line and output http://pastie.org/9417882 [18:07] <c_14> And that gives you the strange timestamps? [18:07] <c_14> Also, note of advice: You really shouldn't run ffmpeg as root. [18:09] <Peter_Occ> Yes here is the video produced by that command http://geotonics.com/captures/testingA.mp4 [18:12] <c_14> try ffmpeg -f mjpeg -r 10 -ss 6 -i [foobar] [18:27] <Fjorgynn> 99 [18:31] <Peter_Occ> c_14, That eliminated the freeze, and the timestamp is accurate for the entire video, only now the video doesn't start for 6 seconds after the command is sent. [18:33] <c_14> yeah, I'm assuming the stream does some sort of weird buffering. ie it sends the first frame and then starts filling the buffer and while the buffer is filling it keeps sending the first frame [18:37] <Peter_Occ> In the example I linked to, its not sending the first frame for six seconds. It sends the video in real time so the video itself is not frozen, its just the timestamp thats frozen. [18:49] <c_14> If seeking 6 seconds into the stream fixes the timestamp issues, I don't think the problem is on ffmpeg's side. [18:53] <c_14> Have you tried the same command with a file source? [18:54] <Peter_Occ> So are you saying that you or anyone else have ever been able to get it working properly? You can start the video immediately and the timestamp is accurate? [18:54] <Peter_Occ> [18:56] <c_14> I've never heard of anyone having issues, but if you give me the command again I can try it myself. [18:59] <Peter_Occ> ffmpeg -f mjpeg -r 10 -i http://user:pass@192.168.0.7/video/mjpg.cgi -vf "drawtext=/usr/share/fonts/dejavu/DejaVuSans-Bold.ttf:text='%{localtime\:%D %T}': fontcolor=red@.8: x=7: y=10" -t 00:00:30 -r 10 /var/www/html/captures/testingE.mp4 [19:03] <c_14> Works fine for me. [19:03] <c_14> With a file source that is. [19:04] <Peter_Occ> I tried using a file instead of the camera, but it just dies. There must be some other change I have to make to the command line. http://pastie.org/9417993 [19:04] <c_14> Can ffplay play testingE.mp4? [19:04] <c_14> If it can try adding -probesize 1G -analyzeduration 1G [19:09] <Peter_Occ> I don't have ffplay, but I have no problem playing the videos in Firefoxj. I added -probesize 1G -analyzeduration 1G to the command line but the output is the same. date && ffmpeg -f mjpeg -r 10 -probesize 1G -analyzeduration 1G -i /var/www/html/captures/testingE.mp4 -vf "drawtext=/usr/share/fonts/dejavu/DejaVuSans-Bold.ttf:text='%{localtime\:%D %T}': fontcolor=red@.8: x=7: y=10" -t... [19:09] <Peter_Occ> ...00:00:30 -r 10 /var/www/html/captures/testingEb.mp4 [19:10] <c_14> try adding -s widthxheight as an input option, where width and height detail the width and height of the video [19:20] <saratoga> jhMikeS: opps, just realized sd_mutex is a simple typo for sd_mtx [19:20] <saratoga> sorry about the kind of clueless comment on gerrit [19:20] <saratoga> opps, wrong window [19:31] <Peter_Occ> That didn't help either. I started over with a command that I know works and I was able to copy the file, but now the timestamp is only changing 1 second every 5 seconds. Is there an adjustment for that? In this video, the second timestamp is the stamp from the copy. http://geotonics.com/captures/testingEg.mp4 [19:32] <Peter_Occ> ffmpeg -i /var/www/html/captures/testingE.mp4 -vf "drawtext=/usr/share/fonts/dejavu/DejaVuSans-Bold.ttf:text='%{localtime\:%D %T}': fontcolor=red@.8: x=7: y=30" -t 00:00:30 -r 10 -y -async 1 /var/www/html/captures/testingEg.mp4 [19:34] <c_14> Can you try it with a current static build? [19:51] <Peter_Occ> I'm not sure if I can do that or not. I got ffmpeg using YUM with the fusion depository. Will I be able to switch back? [19:52] <c_14> ye [19:52] <c_14> just grab the tar [19:52] <c_14> unpack it and use ./ffmpeg instead of ffmpeg [19:52] <c_14> or /path/to/unpacked/ffmpeg [options] [19:56] <Peter_Occ> Where would be a good place to put it? I'm on Fedora 20 [19:57] <c_14> you can put it somewhere like /usr/local/bin or $HOME/bin, it doesn't matter as long as you're not overwriting things. [20:09] <Peter_Occ> I used the static build to copy the same file and the results are the same- the timestamp is 5 times too slow. There must be a reason for that. [20:24] <c_14> Ok, this is weird. If I use your command on one of my files it works fine. Timestamp updates each second etc. If I use it on testingEg.mp4 the timer updates way too slow. [20:31] <Peter_Occ> I wonder if that's because that file is a copy. The timestamp is not slow in the original file. Try this file http://geotonics.com/captures/testingE.mp4 [20:38] <c_14> Same thing. [20:39] <c_14> I even took one of my videos dropped the fps down to 10 and tried using drawtext with that and it worked fine. [20:39] <c_14> I'm running out of possible causes. [20:42] <Peter_Occ> It seems like the slow timestamp isn't caused by anything that happens when the copy is made - whatever is causing it is in the video when its captured. [20:46] <Peter_Occ> I'm starting a new capture every two minutes,. I think I can actually solve my original problem if I start the video 30 seconds early and add the 30 second seek with -ss 30. That should start the video at the exact time with the correct timestamp, [20:47] <c_14> As long as it works, I guess? [20:52] <Peter_Occ> It would require running 2 processes at a time. In other words, start a video at 1:30 minutes, seek 20 seconds, have it run another 2 minutes until 3 minutes, and start another at 2:30 minutes, have it run until 5 minutes. [20:53] <Peter_Occ> It would require running 2 processes at a time. In other words, start a video at 1:30 minutes, seek 30 seconds, have it run another 2 minutes until 3 minutes, and start another at 3:30 minutes, have it run until 5 minutes. [20:54] <Peter_Occ> Is that even possible ? What happens if you run ffmpeg twice from a script? [20:55] <Peter_Occ> or maybe 2 different scripts [20:55] <c_14> As long as they're not using the same output file it should be fine. [20:57] <Peter_Occ> Ok, thanks for all the help. [21:07] <t4nk057> Hey troy_s you are around ? [21:07] <troy_s> t4nk057: Go [21:07] <troy_s> But only if you can tell me how the hell the luminance coefficients were generated for 601. [21:08] <t4nk057> Hi ! So I tried the frame seeking for video [21:08] <t4nk057> for which I use av_rescale ! [21:08] <t4nk057> to get the frame_offset ! I need to seek to [21:09] <troy_s> t4nk057: And how did that go? [21:09] <t4nk057> which looks something like av_rescale(time_in_ms, time_base.den, time_base.num) where time_base are for the stream [21:09] <troy_s> (never seeked based on P frames etc before.) [21:10] <t4nk057> the issue is it gives negative values for some time_in_ms [21:10] <t4nk057> something like 30000 -> gives neg while 30030 doesn't [21:11] <t4nk057> Nope I haven't been able to seek it [21:11] <troy_s> Erf. [21:12] <troy_s> Have you thought about using the DTS values? [21:12] <troy_s> The PTS values could work too possibly, but DTS might be more straightforward. [21:13] <t4nk057> I guess dts cannot handle h.264 [21:13] <troy_s> ??? [21:13] <troy_s> Doesn't every codec need a decoding time stamp? [21:13] <t4nk057> http://libav-users.943685.n4.nabble.com/Frame-accurate-seeking-on-H-264-vid… [21:14] <t4nk057> Umm ! I read something this morning. I am trying to find a link to it [21:15] <t4nk057> https://code.google.com/p/qtffmpegwrapper/issues/detail?id=18 [21:15] <t4nk057> Here ! [21:15] <t4nk057> Not that I understood this completely ! [21:15] <troy_s> t4nk057: This sort of approach might work http://libav-users.943685.n4.nabble.com/Frame-accurate-seeking-on-H-264-vid… [21:16] <t4nk057> Right ! So the only things is that it doesn't converts the time to a frame number offset for most of the cases [21:16] <troy_s> t4nk057: Quite familiar with the Blender decoding, which shifted in recent years to a timecode sort of base. [21:16] <t4nk057> where time is the time of keyframe in the video [21:18] <troy_s> t4nk057: You've read this part regarding the nuances of PTS / DTS yes? http://dranger.com/ffmpeg/tutorial05.html [21:19] <troy_s> t4nk057: Note how it states "May not work either" an awful lot. :) [21:22] <t4nk057> Reading .. [21:23] <troy_s> t4nk057: (Side note, this is one of the reasons NLEs for offline editing generally do an 'ingest media' step at the beginning and convert to a suitable offline format such as DNxHD or ProRes. DNxHD was designed specifically to tackle the frame accuracy required in delivering an edit decision list. ProRes of course was basically a knock off of it.) [21:34] <troy_s> t4nk057: Any progress? [22:25] <t4nk057> No troy_s [22:25] <t4nk057> It still goes to the beginning [22:25] <troy_s> t4nk057: Bad luck. [22:25] <troy_s> t4nk057: Have you tried the PTS trick? [22:26] <t4nk057> The link you gave me said that it should have a flush queue function vefore flusing internal buffer [22:26] <t4nk057> I didn't really get the pts thing [22:26] <troy_s> Hrm. [22:26] <t4nk057> so I wrote a flush method [22:26] <troy_s> If (and that's a big if) the decoding time stamp is correctly grabbed, I think you could use that. [22:26] <troy_s> Or PTS. [22:26] <t4nk057> which takes the AvPacketList and frees it [00:00] --- Fri Jul 25 2014

1 0

ffmpeg-devel.log.20140723
by burek 24 Jul '14

24 Jul '14

[00:26] <ePirat> how to add a codec? [00:26] <ePirat> *protocol [00:28] <cone-382> ffmpeg.git 03Diego Biurrun 07master:57f1b1dcc77a: fft-test: Drop unnecessary #ifdefs around header includes [00:28] <cone-382> ffmpeg.git 03Michael Niedermayer 07master:77304cf3523c: avcodec: dct/rdft only support float mode, skip their contents for fixed point [00:28] <cone-382> ffmpeg.git 03Michael Niedermayer 07master:9eeedf876998: Merge commit '57f1b1dcc77a93c2a5c503d4e47fe2f567cf9db5' [00:34] <cone-382> ffmpeg.git 03Diego Biurrun 07master:5be7d8597d02: fft-test: Comment some #endifs [00:34] <cone-382> ffmpeg.git 03Michael Niedermayer 07master:9515d7c629cd: Merge commit '5be7d8597d02bcaba2de2503c0e59cf865c746dc' [00:34] <Compn> ePirat : probably copy a current protocol and change it from there [00:38] <wm4> ePirat: didn't you just send a patch that adds a protocol [00:38] <ePirat> already figured out [00:39] <ePirat> i just meant how to add it in terms of in which files to reference it [00:39] <ePirat> so that it compiles [00:45] <Compn> yes i thought thats what you wanted [00:45] <Compn> configure and probably libavformat/allformats [00:45] <Compn> maybe some others [01:05] <llogan> michaelni: are you envisioning the wiki projects page to be open to all, or marked as read-only and users continue asking on devel so potential projects can be reviewed? [01:05] <llogan> laziness says to be open, but nitiness says to review to help reduce crap [01:09] <relaxed> I nominate llogan as wiki sheriff [01:19] <michaelni> llogan, imo we shouldnt try to fix an issue before tere is one [01:20] <michaelni> that is why restrict the access to a wiki page before anyone tries to do something nasty [01:20] <michaelni> or rather why add more work on our sholders before we need to [01:21] <ePirat> ubitux, done https://ffmpeg.org/pipermail/ffmpeg-devel/2014-July/160172.html [01:22] <cone-382> ffmpeg.git 03Martin Storsjö 07master:06a1d1323bc8: configure: Extend the neon intrinsics check [01:23] <cone-382> ffmpeg.git 03Michael Niedermayer 07master:7e5d75d53f9b: Merge commit '06a1d1323bc8afd03f085ae98d9b74e431b3534e' [01:24] <wm4> so I'm seeing this debian guy on the mailing list [01:25] <wm4> apparently he's working on ffmpeg packages? [01:25] <wm4> no such packages in debian unstable or experimental, though [01:27] <wm4> ePirat: your patch lacks some files, I think [01:30] <ePirat> fail [01:30] <ePirat> forgot to add the actual protocol& [01:35] <iive> wm4: last time i've heard about debian packages, they were waiting approval from ftp masters. [01:36] <ePirat> wm4, fixed https://ffmpeg.org/pipermail/ffmpeg-devel/2014-July/160173.html [01:36] <llogan> If you grew a beard during the whole Debian process, by the time it was accepted you would look like Compn [01:37] <J_Darnley> ePirat: It looks like you've sent it to the wrong project. This is FFmpeg not Libav [01:45] <ePirat> J_Darnley, it was requested that I send it to ffmpeg too [01:46] <llogan> ePirat: he's referring to the "This file is part of Libav." [01:47] <llogan> etc, etc, etc [01:48] <ePirat> llogan, anything else then the license/header that I should change? [01:49] <llogan> just look at the source of the other stuff using the same license in libavformat [01:49] <llogan> for examples [01:55] <J_Darnley> Also: the micro version starts at 100 in ffmpeg [01:55] <J_Darnley> and there's a cosmetic alignment issue with your addition to the makefile. [01:56] <ePirat> ? [01:56] <wm4> I think it's bullshit to make contributors prepare two patches [01:56] <wm4> one for ffmpeg and one for libav [01:57] <J_Darnley> I agree. [01:57] <wm4> the way it works is that libav will apply it, and ffmpeg will merge the libav commit [01:57] <wm4> so there's absolutely no need to make the contributor send a ffmpeg specific patch [01:57] <wm4> the value in having the contributor send a patch to ffmpeg at all is so that ffmpeg can review the patch [01:57] <wm4> and give suggestions for improvement [01:58] <wm4> and review can be done on a patch even if it contains the evil "Libav" string [01:59] <ePirat> J_Darnley, whats wrong in the makefile? [02:00] <J_Darnley> I'll send an email [02:01] <ePirat> oh i think i found it, I indented with tabs accidentally [02:01] <J_Darnley> OMG! TABS! THEY'RE DEVIL CHARACTERS! [02:02] <wm4> yes they are [02:03] <llogan> ePirat: tools/patcheck will discover those for you [02:03] <wm4> they have the ability to incite endless bikesheds and flames [02:04] <J_Darnley> I don't even like them in a word processor. [02:06] <ePirat> does patcheck expects a patch file? [02:06] <ePirat> I get somewhat messy output [02:07] <J_Darnley> A patch ot diff, yes [02:08] <J_Darnley> *or [02:08] <J_Darnley> You can also pipe into it. [02:13] <ePirat> J_Darnley, fixed I think https://ffmpeg.org/pipermail/ffmpeg-devel/2014-July/160177.html [02:16] <J_Darnley> I don't see any more obvious problems. [02:46] <cone-382> ffmpeg.git 03Michael Niedermayer 07master:a61325798a9a: Changelog: add "version <next>:" so people dont add new things to the last release [03:04] <Orix> hey guys, I'm trying to get the bink codec to load: http://pastebin.com/azvFGhLJ but I keep getting the following error: [bindvideo @ <memory>] Extradata missing or too short [03:08] <michaelni> ePirat, ive posted a few minor commets, btw, how can i test it ? [03:10] <michaelni> Orix, you need a demuxer which will set extradata from the file header(s) [03:11] <Orix> michaelni: would you be able to elaborate or link me to the documentation? [03:11] <Orix> i've been going off of https://ffmpeg.org/doxygen/trunk/api-example_8c-source.html [03:12] <michaelni> Orix, doc/examples/demuxing_decoding.c might work [03:13] <michaelni> or well, ffplay.c /ffmpeg but they are big [03:13] <Orix> yeah, tried to dig at them a bit [03:13] <Orix> i'll check out demuxing [03:13] <Orix> thanks [03:47] <jamrial> [cpuflags @ 000000000031f188] [Eval @ 000000000031e960] Invalid chars 'now' at the end of expression '3dnow' [03:47] <jamrial> [cpuflags @ 000000000031f188] Unable to parse option value "3dnow" [03:48] <jamrial> Failed to set value 'mmx+mmxext+3dnow' for option 'cpuflags': Error number -22 occurred [03:48] <jamrial> http://fate.ffmpeg.org/history.cgi?slot=x86_64-athlon64-gentoo-gcc [03:49] <jamrial> http://fate.ffmpeg.org/history.cgi?slot=x86_32-athlon64-gentoo-gcc [03:49] <jamrial> http://fate.ffmpeg.org/history.cgi?slot=x86_32-athlon64-gentoo-gcc-3dnow [04:27] <jamrial> 265dadbe introduced this regression [08:00] <ubitux> llogan: you should tweet about the recent release [08:02] <ubitux> (michaelni: i'll test the icecast thing so please don't apply in a hurry ;)) [08:07] <jamrial> speaking of recent release, it needs to be added to trac [08:50] <ePirat> michaelni, you need to have an icecast server or alternatively just something that will accept a PUT request [08:51] <ePirat> michaelni, and then just do icecast://username:pass@mycoolserver:8000/mountpoint.mp3 or so [08:56] <cone-382> ffmpeg.git 03Michael Niedermayer 07master:bbc5adfb9428: avutil/opt: Fix parsing regression with constants starting with numbers [08:57] <michaelni> ePirat, ok, thx [08:57] <ePirat> how do I use av_freep? [08:59] <michaelni> av_free(x) -> av_freep(&x) [08:59] <michaelni> jamrial, 2.3 added [09:01] <ePirat> michaelni, thanks [09:02] <ePirat> michaelni, and I wonder if there is a function that could replace my cat_header stuff [09:14] <michaelni> ePirat, av_bprint_init+ n * av_bprintf+av_bprint_finalize or something based on AV_DICT_APPEND but cat_header() is fine too [09:15] <ePirat> ok [09:15] <ePirat> is it fine to freep it inside the function and return NULL? [09:19] <ePirat> michaelni, if I just use one av_log it looks ugly: [09:19] <ePirat> [icecast @ 0x7fc63be015c0] Streaming ogg but appropriated content type NOT set! [09:19] <ePirat> Set it with -content_type application/ogg [09:20] <ePirat> the second missing the icecast @ & part, causing it to look weird [09:21] <ePirat> and the line will get too long and it's awful to split it to two lines [09:25] <michaelni> ePirat, hmm, ok sure, leave it as 2 calls then [09:26] <ePirat> would work if I somehow could suppress the icecast @ & part tough [09:26] <michaelni> that would be a NULL as first parameter but its probably better to know where a message came from [09:26] <ePirat> ok [10:33] <cone-382> ffmpeg.git 03Carl Eugen Hoyos 07master:c75fdee74717: avcodec/x86/hevc_deblock: Fix compilation with nasm. [10:51] <cone-382> ffmpeg.git 03Vittorio Giovara 07release/2.2:68fd80ee1ca2: g2meet: allow size changes within original sizes [10:51] <cone-382> ffmpeg.git 03Michael Niedermayer 07release/2.2:3e78f8689167: Merge commit '68fd80ee1ca22c39b6ef4e6641b5b2e0d4d89a14' into release/2.2 [10:51] <cone-382> ffmpeg.git 03Alessandro Ghedini 07release/2.2:4ddac7199b24: vc1: Do not return an error when skipping b frames [14:26] <cone-382> ffmpeg.git 03Diego Biurrun 07master:2b6ab3a2bd7e: mpegvideo: Move QUANT_BIAS_SHIFT define to the only place it is used [14:26] <cone-382> ffmpeg.git 03Michael Niedermayer 07master:d0cb6caf691c: Merge commit '2b6ab3a2bd7e7cee5e7a55dd2e48b8feb4a826bb' [14:36] <cone-382> ffmpeg.git 03Diego Biurrun 07master:20ec0daaa1cd: fft-test: Make global exptab variable static [14:36] <cone-382> ffmpeg.git 03Michael Niedermayer 07master:54256482cbc0: Merge commit '20ec0daaa1cdd3996ac6ed0198de961129474044' [18:05] <cone-109> ffmpeg.git 03Michael Niedermayer 07master:1bf371b24ebd: avcodec/h264: factor is_extra() out [18:05] <cone-109> ffmpeg.git 03Michael Niedermayer 07master:601c2388549f: avcodec/h264: support AV_PKT_DATA_NEW_EXTRADATA [18:05] <cone-109> ffmpeg.git 03Michael Niedermayer 07master:01b236b70438: avformat/flvdec: use AV_PKT_DATA_NEW_EXTRADATA for h264 too [18:57] <cone-109> ffmpeg.git 03Andrey Utkin 07master:8a4c0866dc7d: avio: Introduce avio_read_to_bprint(avioctx, bp, max_size) [18:57] <cone-109> ffmpeg.git 03Andrey Utkin 07master:b6a6459a24c6: lavd/lavfi: work with non-mappable files for graph_file option [19:45] <kurosu_> If I had a new threading in lavc/thread.h, what do I need to do documentation-wise ? [19:45] <kurosu_> is it a public api? (ie api bump?) [19:53] <jamrial> that header is not installed, and all the functions have the ff_ prefix, so it's internal to lavc [20:01] <kurosu_> I see there's a doc/multithreading.txt [20:39] <cone-109> ffmpeg.git 03Nidhi Makhijani 07master:eac77fcd56fc: mpegenccontext: Remove unused dct_unquantize_h261 function pointers [20:39] <cone-109> ffmpeg.git 03Michael Niedermayer 07master:97d2b8d6f5a4: Merge commit 'eac77fcd56fc2a3391f0d86faf54302afb368ff7' [20:46] <cone-109> ffmpeg.git 03Diego Biurrun 07master:d445c865f87e: fft-test: Reduce variable scope where sensibly possible [20:46] <cone-109> ffmpeg.git 03Michael Niedermayer 07master:09f5b3750f2d: Merge commit 'd445c865f87e7f13230e4dac3f77f5af21da95cb' [20:53] <cone-109> ffmpeg.git 03Diego Biurrun 07master:7288b3458507: fft-test: Drop unnecessary pointer indirection for context structs [20:53] <cone-109> ffmpeg.git 03Michael Niedermayer 07master:088875b6b855: Merge commit '7288b345850792430302a8f85a4b29140b770497' [21:04] <cone-109> ffmpeg.git 03Christophe Gisquet 07master:ca081217cda8: hevcdsp: change types of SAO parameters [21:42] <cone-109> ffmpeg.git 03Christophe Gisquet 07master:373677f95816: hevc_mvs: avoid deriving tmvp in amvp [22:18] <llogan> ubitux: done [22:18] <ubitux> thx [22:19] <llogan> damn, i should have added an image from mandelbrot filter [22:25] <cone-109> ffmpeg.git 03Christophe Gisquet 07master:670b7f203ad2: x86: hevcdsp: align [23:35] <BBB> kurosu_: so & Im wondering how & portable this new threading function really is [23:35] <BBB> kurosu_: do you epect other codecs to use it? [23:35] <BBB> kurosu_: like, realistically, not theoretically [23:52] <Orix> hey guys, i have a decent movie loader rendering right now, but I'm currently experiencing an "issue". I noticed that one of the movies i'm trying to render is grayscale, how do i properly convert that into RGB32? I currently see a green overlay [23:54] <michaelni> Orix, libswscale or libavfilters scale filter (which uses libswscale) [23:55] <michaelni> or well, if you just need gray ->rgb32 you can also write your own converter [23:55] <Orix> ah i see, and in swscale swap the pixel format for the source depending on what the file is? [23:55] <michaelni> yes [23:55] <Orix> makes sense [23:55] <Orix> thanks [23:59] <Orix> imageConvertContext = sws_getCachedContext(nullptr, mCodecContext->width, mCodecContext->height, mCodecContext->pix_fmt, mWidth, mHeight, AV_PIX_FMT_RGB32, SWS_FAST_BILINEAR | SWS_CPU_CAPS_MMX2, nullptr, nullptr, nullptr); [23:59] <Orix> that is what I currently have, but I'm still seeing the green overlay on the grayscale video [00:00] --- Thu Jul 24 2014

1 0

ffmpeg.log.20140723
by burek 24 Jul '14

24 Jul '14

[00:59] <relaxed> daala-person: ffmpeg -h encoder=libx264 [01:11] <daala-person> relaxed: Thanks. [06:24] <slowguy> hi [06:30] <slowguy> i need to make an application which can take a live video input and dump to a file..with a button to tell the app to skip the advertisements [06:30] <slowguy> so that when the button is pressed then video is not dumped to file [06:31] <slowguy> can someone give me pointers..how can i do that either by an existing program or a method to make a new program [08:19] <relaxed> llogan: my build on the download page now points to burek's builds [12:06] <jchillerup> Hi, I'm trying to stream my microphone over RTP, but I want to reduce the internal buffer to a minimum (i.e. I want near-live streaming of the microphone input). Right now I use this command to do what I want, but there is over a second's worth of buffer: ffmpeg -f alsa -i hw:0,0 -acodec libmp3lame -ab 256k -ac 1 -f rtp rtp://234.5.5.5:1234 [12:15] <jchillerup> Oh, turned out to be on the receiver side [13:27] <luc4> Hello! I know DVB-T is typically an mpeg2 ts with h264 in some resolution like 1080i or similar. Can I transcode a test video to such a format with ffmpeg to test with my video player? [13:28] <luc4> In particular I would like to know what is the proper way to create the interlaced video. The conversion to ts and h264 is pretty simple. [13:41] <kippi> I am using, -vf "select='gt(scene, 0)',showinfo" -f null - is there away to only show when theres movement [14:36] <Enverex> When using concat, if you're going from one format to another, say 3 AAC files concatting to a single AAC file, does it reencode it? [14:36] <sacarasc> Depends what options you use. [14:36] <Enverex> e.g. ffmpeg -i "concat:wotw1.m4a|wotw2.m4a|wotw3.m4a" -c copy "wotwfykk.m4a" [14:36] <sacarasc> That shouldn't, because you used -c copy. [14:38] <Enverex> The reason I asked is because I saw this, which made me think it was reencoding - "size= 270535kB time=06:28:56.65 bitrate= 95.0kbits/s, video:0kB audio:266606kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.473666%" - that said, it only took a few seconds so it couldn't have been reencoding it, it would have taken ages to do 6 and a half hours :P [14:40] <Enverex> Cheers sacarasc [16:42] <nypenick> hey ffmpeg gyus [16:42] <nypenick> I'm using the ffmpeg lib for decoding audio [16:43] <nypenick> I have the problem that av_seek_frame seems to be imprecise [16:44] <nypenick> for example I want to seek to timestamp 23000 but after seeking the timestamp is at 23400 [16:45] <nypenick> has anyone an idea what I'm doing wrong? [17:22] <sfan5> nypenick: precise seeking is probably jut not possible, you may need to find the earliest seekable point before where you want to seek and decode everything until you get where you want to end up [17:22] <sfan5> what I just said may be or may not be helpful [17:23] <sfan5> I know almost nothing about using libav* [17:32] <nypenick> sfan5: so av_seek_frame doesn't do that automatically? [17:32] <sfan5> looks like it doesn't [17:33] <nypenick> but how can I find the earliest seekable point? [18:16] <joselo> Hi there [18:17] <joselo> I'm new using ffmpeg and iŽm having some trouble when trying to save to a file a live stream [18:18] <joselo> ok im on it [18:19] <joselo> http://pastebin.com/Nxb5b1E6 [18:20] <joselo> there it is [18:21] <joselo> the problem is that the save process stops always at 1minute 4 second [18:21] <joselo> so I always end up with a mp4 file with that duration [18:23] <c_14> Does the same thing happen if you get rid of the -t option? [18:28] <joselo> yeah the same result [18:28] <joselo> also tried with -to [18:30] <c_14> Is that ' live=1' part of the url? [18:30] <joselo> no [18:31] <joselo> also tried with out that param [18:31] <joselo> just with "rtsp://192.168.0.102:554/live.sdp" [18:31] <c_14> Have you tried using it with -codec copy? [18:32] <joselo> no, I read a bit the docs but I wasn't able to get it to work [18:33] <c_14> Eh, try -c copy and use something like mkv as output just for testing purposes. I'm guessing mp4 doesn't like amr_nb as an audio format. [18:36] <joselo> ok ill try that out [18:39] <Mavrik> hmm, do we have any comparisons of quality of libswscale resample algorithms? [19:50] <jdolan> is there a convenient way to compare av_gettime() against a packet's pts? [20:33] <zenderz> hi all [20:35] <zenderz> anyone had any experience with generating smooth streaming format content. I see the format has recently been added [20:39] <zenderz> looking at using something like: #ffmpeg -i testinput.mpg -c copy -f smoothstreaming output [20:39] <zenderz> but there is nothing in the docs on how to use -f smoothstreaming [20:40] <c_14> Check the docs for ismv, I'm pretty sure that's what the format's called. [20:41] <c_14> https://ffmpeg.org/ffmpeg-formats.html#mov_002c-mp4_002c-ismv [20:41] <zenderz> ya ismv is the file format. but smoothstreaming also generates the manifest files [20:42] <zenderz> i think -f smoothstreaming is only newly added [20:43] <c_14> Ah, ye. [20:43] <c_14> https://ffmpeg.org/ffmpeg-formats.html#smoothstreaming [20:44] <c_14> No examples yet though... [20:44] <Katharsis> i'm using this for recording my desktop: [20:44] <Katharsis> ffmpeg -f x11grab -s 1280x768 -r 25 -qscale 1 -i :0.0 file.mp4 [20:44] <Katharsis> i don't know why there is no sound [20:44] <Katharsis> any ideas? [20:44] <c_14> Because you never added sound? [20:45] <c_14> You'll need to add an alsa or pulse device. [20:45] <c_14> https://ffmpeg.org/ffmpeg-devices.html#alsa-1 [20:45] <c_14> https://ffmpeg.org/ffmpeg-devices.html#pulse [20:46] <Katharsis> you mean -f alsa ? [20:46] <Katharsis> what's the diff between them? [20:46] <c_14> Pulse and alsa? [20:47] <c_14> Pulse is an abstraction layer on top of alsa that lets you do fun things sometimes. [20:49] <Katharsis> encoder 'aac' is experimental and might produce bad results. [20:49] <Katharsis> Add '-strict experimental' if you want to use it. [20:49] <Katharsis> i got this exception [20:51] <c_14> If you want to use aac, the internal encoder is usually fine with decent bitrates, just add -strict experimental. You can also use one of the alternate encoders or a different codec for audio. [20:51] <zenderz> c_14. any idea (short of reading the source code) where i can get some examples on smooth streaming use? [21:00] <Katharsis> c_14: ok, i used `ffmpeg -f alsa -i default -f x11grab -s 1280x768 -r 25 -qscale 1 -i :0.0 -strict experimental filename.mp4` and it works [21:00] <Katharsis> am i used the best quality (sound / video) option in here? [21:01] <c_14> zenderz: Well, ffmpeg -i file -strict -2 -b:v bitrate -f smoothstreaming out; seems to create a directory called out with a Manifest file, and a bunch of fragments inside of some QualityLevels folders, but I can't tell you what to do with them since I don't understand the format. You could always wait here to see if somebody understands the format better or you could send an email to the ffmpeg-user ml to ask [21:01] <c_14> around there. [21:02] <c_14> Katharsis: That'll default to whatever settings are default for mp4, which usually aren't that great. You might want to pick codecs and bitrates by hand. [21:02] <c_14> Maybe look at: https://trac.ffmpeg.org/wiki/Encode/H.264 [21:06] <zenderz> thanks c_14. just FYI. the manifest points to the different file chunks which can be of diferent rates for adaptive streaming (much like hls). It seems to be generating everything correctly for me but im getting some errors which i dont understand. ill wait around for a bit. Thanks again for you help [21:07] <iRonNCSU> what exactly is the "start" value for an input video? why shouldn't it always be 0? Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'camVid60s.mp4': [21:07] <iRonNCSU> Metadata: [21:08] <iRonNCSU> major_brand : isom [21:08] <iRonNCSU> minor_version : 512 [21:08] <iRonNCSU> compatible_brands: isomiso2avc1mp41 [21:08] <iRonNCSU> encoder : Lavf55.33.100 [21:08] <iRonNCSU> Duration: 00:01:00.07, start: 0.072562, bitrate: 422 kb/s [21:08] <iRonNCSU> Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 320x240, 289 kb/s, 29.97 fps, 29.97 tbr, 11988 tbn, 59.94 tbc (default) [21:08] <c_14> Next time pastebin please. [21:08] <iRonNCSU> oh sorry [21:09] <iRonNCSU> http://pastebin.com/uwUcgKyT [21:10] <iRonNCSU> basically, i'm trying to get 2 videos, ostensibly of the same framerate, to the exact same length, but the offset on this "start" on one of the videos always throws off the lengths, even if I use the same -t and -ss values [21:10] <c_14> Are you reencoding when you're cutting? [21:12] <iRonNCSU> yes [21:15] <c_14> I'm not entirely sure, but it might have something to do with certain encoders adding buffers of nothing to the beginning of the stream. [21:15] <c_14> Might want to wait for someone who knows more though. [21:16] <iRonNCSU> in the pastbin for example, that was the file (originally an FLV) convereted to mpeg4, it was supposed to be exactly 1 minute long, but has that "start" value. the orignal flv doesn't have this value. [21:23] <Katharsis> c_14: thx for your tip and time [21:27] <hilacha> hi all. i'm developing a alcatel voip call recoder based in g723.1 codec inside h.323. i've identified (by the headers and the bytes count by packet) that it's 6.3kbit/s. i've extracted it from the RTP stream (like i do with the g729 codec) but i can't convert it with ffmpeg nor asterisk. To convert the stream to wav, there is need to specify additional options to ffmpeg to convert it? [21:32] <iRonNCSU> so i've narrowed my issue down to the audio track. Even using async 1 it doesn't match up in durations. i MUST reencode the audio to make the times match (which is what adds that start: value), but the problem is that the audio is alredy 32kbps, so reencoding it sounds horrible [22:04] <llogan> relaxed: i squawked about that a few days ago, but it should be fixed now. i would have fixed it myself but i was still travelling. [22:20] <nexusIL> Hi I'm recording from a stream, But for different times I run the ffmpeg command I get diffrent order of stream for the channel. I'm trying to understand where this is coming from. [22:26] <Tox1city> Hey Guys, I'm having some trouble compiling FFMpeg, I keep getting the ERROR: x265 not found [22:26] <llogan> Tox1city: did you install x265 first? [22:26] <Tox1city> Yes [22:26] <llogan> does the tail of config.log give you any clues? [22:31] <nexusIL> any ideas? [22:35] <llogan> nexusIL: you can use a pastebin site to show the first ~50 and last ~50 lines of config.log [22:36] <llogan> also, how did you install x265? [22:37] <Tox1city> Thanks [22:37] <llogan> nexusIL: sorry, that was for Tox1city [22:37] <nexusIL> K [22:37] <Tox1city> It seems I was missing x265.pc in my PKGconfig [22:38] <llogan> i've seen that once before when the user attempted to install specific components of x265 and omitted x265.pc for some reason [22:40] <llogan> i mean he was attempting to manually install specific files. [22:41] <Tox1city> Okay [22:42] <Tox1city> To be honest, I'm not an advanced linux user but the x265 command did tell me x265 was installed [22:42] <Tox1city> It seems to be compiling well now [22:44] <llogan> nexusIL: you should have asked on superuser, not stackoverflow since SO is for programming questions only [22:50] <nexusIL> oh I see, I didn't know where to ask that [22:54] <nexusIL> Anyway I think it has something to do with it not recognized as a program [23:04] <nexusIL> how can I add demux capabilties to ffmpeg? I want to demux mp2 [23:07] <llogan> demuxing usually means you want to take something out of a container, but mp2 in ffmpegese refers to the mp2 audio decoder or encoder [23:11] <nexusIL> Ok, So I'm looking on adding decoding ability to ffmpeg for mp2, I have encoding as `ffmpeg -formats` says [23:13] <llogan> ffmpeg -decoders | grep mp2 [23:14] <llogan> ffmpeg can natively decode mp2 audio without any special steps needed [23:15] <nexusIL> ah nice, so what's the formats refers to? [23:15] <JEEB> demuxers/muxers [23:15] <JEEB> lavf stuff [23:16] <nexusIL> ok tnx [00:00] --- Thu Jul 24 2014

1 0

ffmpeg-devel.log.20140722
by burek 23 Jul '14

23 Jul '14

[01:27] <michaelni> Anssi, did you see / do you have time to review "0718 10:57 Nicolas Martyan ( 16K) [FFmpeg-devel] [PATCH */7] avformat/hlsenc" ? [02:07] <BBB> kurosu: oh wow, thats cool, so the scan order is the only difference? Im trying to figure out how to abuse this in vp9 :-p [02:32] <cone-216> ffmpeg.git 03Dale Curtis 07master:42f3bc407265: avformat/adtsenc: Optionally enable ID3 in ADTS. [02:32] <cone-216> ffmpeg.git 03Dale Curtis 07master:70f7006eb311: avformat/mov: Pass through iTunSMPB from MOV. [02:38] <cone-216> ffmpeg.git 03Jan Gerber 07master:fc27e7272e6f: mpegts: do not export empty language tags [02:39] <cone-216> ffmpeg.git 03Michael Niedermayer 07master:c7168d1a22a6: Merge commit 'fc27e7272e6f1a7053812f8a15b1b7098de44c71' [02:44] <cone-216> ffmpeg.git 03Luca Barbato 07master:b396bbad100a: configure: Do not add extralibs to avresample .pc file [02:44] <cone-216> ffmpeg.git 03Michael Niedermayer 07master:7f1543d46cd8: Merge commit 'b396bbad100a7493691d09b8dceba91e3cd28e2e' [02:50] <cone-216> ffmpeg.git 03Luca Barbato 07master:bb0babd7054b: build: Support executable only ldflags [02:50] <cone-216> ffmpeg.git 03Michael Niedermayer 07master:dac096cf39d2: Merge commit 'bb0babd7054bed7edfd9f4d6b20cdba864de1830' [02:59] <cone-216> ffmpeg.git 03Luca Barbato 07master:28df1d24112c: http: Provide an option to override the HTTP method [02:59] <cone-216> ffmpeg.git 03Michael Niedermayer 07master:e621e6cac6dd: Merge commit '28df1d24112c6ad0763985df2faeeb198cfbad69' [03:05] <cone-216> ffmpeg.git 03Luca Barbato 07master:389380c27915: http: Do move the class instantiation in the conditional block [03:05] <cone-216> ffmpeg.git 03Michael Niedermayer 07master:ab7c67905d1a: Merge commit '389380c27915b0505fed538cd54c035c891fabd9' [03:34] <cone-216> ffmpeg.git 03Janne Grunau 07master:d45ebd4876ab: configure: add support for neon intrinsics [03:34] <cone-216> ffmpeg.git 03Michael Niedermayer 07master:135b1a3f2c79: Merge commit 'd45ebd4876ab8fc07736a644de07e1b1d11a8e5d' [04:46] <cone-216> ffmpeg.git 03James Yu 07master:16e665228621: aarch64: NEON intrinsics dct_unquantize_h263. [04:46] <cone-216> ffmpeg.git 03Michael Niedermayer 07master:c933f2c4cca3: Merge commit '16e6652286213014c28a9c3af70435e767d28638' [05:21] <cone-216> ffmpeg.git 03Andreas Cadhalpun 07master:135ea609cebb: libavformat/img2dec.c: Add a long_name to the piped image demuxers [07:29] <ubitux> ePirat: can you send a ffmpeg version of the icecast protocol patch, i'm interested in this [07:29] <ubitux> ? [12:33] <mraulet> michaelni: patch ok https://ffmpeg.org/pipermail/ffmpeg-devel/2014-July/160076.html [12:33] <mraulet> I need to check the other one [12:51] <cone-381> ffmpeg.git 03Ben Avison 07master:6869612f5c7d: arm: Macroize the test for 'setend' CPU instruction support [12:51] <cone-381> ffmpeg.git 03Michael Niedermayer 07master:7cdb3b2b79d7: Merge commit '6869612f5c7d4d2f20f69a5658328a761deadb1c' [12:59] <cone-381> ffmpeg.git 03Diego Biurrun 07master:78670fbf07bd: imgconvert: cosmetics: Reshuffle defines to reduce ifdeffery [12:59] <cone-381> ffmpeg.git 03Michael Niedermayer 07master:4ec1cb24978e: Merge commit '78670fbf07bd03479073d4218440de4914304680' [13:06] <cone-381> ffmpeg.git 03Mickaël Raulet 07master:fa0506e4136b: hevc/rext: update boundary strength for chroma deblocking filter [13:29] <cone-381> ffmpeg.git 03Mickaël Raulet 07master:627c044f50da: hevc/rext: fixing rdpcm and scaling list for Range Extension [13:54] <cone-381> ffmpeg.git 03Stefano Sabatini 07master:713157484a10: lavf/segment: do not allow to create segments with no key-frames [13:54] <cone-381> ffmpeg.git 03Stefano Sabatini 07master:44071b0d252f: lavf/segment: sanitize segment end time in case last packet do not have a defined duration [14:04] <cone-381> ffmpeg.git 03Mickaël Raulet 07master:37822a566fc8: hevc/rext: remove warning for cross component prediction [14:05] <cone-381> ffmpeg.git 03Mickaël Raulet 07master:062b9f2cdb89: fate/hevc: update with new Rext bitstreams [14:09] <ubitux> <j-b> ubitux: depends, if yo provide the function or not // sorry for the delay; what do you mean? mpv is able to do that [14:09] <ubitux> j-b: like this: https://github.com/mpv-player/mpv/blob/master/common/av_log.c#L191 [14:15] <ePirat> ubitux, I will [14:16] <ubitux> ePirat: thank you :) [14:17] <cone-381> ffmpeg.git 03Mickaël Raulet 07master:560b0253ff2b: hevc/mvs: simplify derive_temporal_colocated_mvs [14:17] <cone-381> ffmpeg.git 03Mickaël Raulet 07master:e9bb45ba7905: hevc/mvs: remove same prediction block [14:21] <cone-381> ffmpeg.git 03Mickaël Raulet 07master:c4d63cd230b7: hevc/mvs: simplifying derive_spatial_merge_candidates function [14:28] <kierank> mraulet: are you going to ibc [14:28] <mraulet> it is planned to be there [14:29] <kierank> remind me to introduce you to the guys from the uk equivalent of 4ever [14:29] <kierank> thiera [14:29] <kierank> thira I mean [14:29] <mraulet> I recently disrupt my Achille tendon& So depending on how I can walk back I would be there [14:29] <mraulet> ok [14:30] <kierank> ibc is not a good place if you can't walk [14:30] <mraulet> I know it :-) [14:30] <mraulet> I have reduced the trip to one day [14:30] <mraulet> anyway the project will be there again [14:31] <kierank> we will be running an OSS event on saturday [14:31] <mraulet> Plan to be there only on sunday [14:33] <kierank> are you going to vdd? [14:33] <mraulet> it is not planned, too many travels already [15:12] <cone-381> ffmpeg.git 03Mickaël Raulet 07master:3ad046083119: hevc/mvs: code cleaning [15:19] <cone-381> ffmpeg.git 03Mickaël Raulet 07master:cd714afe8136: hevc/mvs: more cleanups [15:53] <ePirat> ubitux, in ffmpeg there is also no way to get the container format infos in a protocol, right? [15:53] <ubitux> yeah i don't think so [15:58] <cone-381> ffmpeg.git 03Mickaël Raulet 07master:5f08b417b4c0: hevc/mvs: cosmetics [16:05] <cone-381> ffmpeg.git 03Anton Khirnov 07master:7a4cf671170d: hevc: cleaning up SSE2 and SSSE3 deblocking filters [16:26] <cone-381> ffmpeg.git 03Anton Khirnov 07master:500a0394d57a: avcodec/x86/hevc_deblock: add %ifs to avoid "do nothing instructions" [16:26] <cone-381> ffmpeg.git 03Anton Khirnov 07master:dc69247de421: avcodec/x86/hevc_deblock: use of paddw instead of psllw [16:26] <cone-381> ffmpeg.git 03Anton Khirnov 07master:e8581b17a8d2: avcodec/x86/hevc_deblock: use test instead of cmp 0 [16:26] <cone-381> ffmpeg.git 03Anton Khirnov 07master:b435043abb36: hevc: cleanups in SSE2 and SSSE3 loop filters, use fewer instructions [16:26] <cone-381> ffmpeg.git 03Anton Khirnov 07master:ae2f048fd757: avcodec/x86/hevc_deblock: cosmetics [16:41] <cone-381> ffmpeg.git 03Anton Khirnov 07master:d7e162d46b4a: hevcdsp: remove an unneeded variable in the loop filter beta0 and beta1 will always be the same within a CU [16:41] <cone-381> ffmpeg.git 03Michael Niedermayer 07master:ca6b33b8bd5e: avcodec/x86/hevcdsp_init: Fix "warning: assignment from incompatible pointer type" [17:11] <ePirat> ubitux, my patch requires the possibility to change HTTP method to PUT [17:11] <ePirat> else it would be pointless since connecting would fail everytime [17:12] <ubitux> is that a problem? [17:13] <ubitux> i mean, is that something you haven't already solved? [17:14] <ePirat> it is just not possible as I already said. this requires changes to the http.c and addition of a new option, would prefer if someone adds this who is a bit more familiar with the code [17:17] <ubitux> are you refering to "[PATCH] http: Provide an option to set the write method"? [17:17] <ePirat> yes [17:18] <ubitux> this commit will be picked from libav, there is no need to write it differently [17:18] <ePirat> ok [17:19] <ubitux> you can mention it when you send your patch if you don't want to also send that patch [17:36] <michaelni> our http already supports setting the method to put since yesterday [17:45] <cone-381> ffmpeg.git 03Muhammad Faiz 07master:4852a88dfbda: avutil/opt: set_string_number(): remove unneeded copy [17:45] <cone-381> ffmpeg.git 03Michael Niedermayer 07master:539e72727f12: avutil/opt: simplify set_string_number() [18:08] <cone-381> ffmpeg.git 03PrzemysBaw Sobala 07master:c68098ba4a14: avcodec/mjpegdec: add pix_fmt: 0x14121200 [18:21] <cone-381> ffmpeg.git 03Christophe Gisquet 07master:ed248e7f70d9: hevc: wait proper position for tmvp [18:52] <cone-381> ffmpeg.git 03Carl Eugen Hoyos 07master:050d79b6ddaf: Autodetect big-endian tiff files. [18:52] <cone-381> ffmpeg.git 03Michael Niedermayer 07master:bc15b5238f99: Merge remote-tracking branch 'cehoyos/master' [22:09] <cone-382> ffmpeg.git 03Luca Barbato 07master:7bdd2ff68259: http: Use a constant for the supported header size [22:09] <cone-382> ffmpeg.git 03Michael Niedermayer 07master:016cca450474: Merge commit '7bdd2ff6825951f7a6a6008303acfce7c2a63532' [22:29] <cone-382> ffmpeg.git 03Diego Biurrun 07master:67bc1ba5d759: libxvid: Drop PPC-specific CPU detection hack [22:29] <cone-382> ffmpeg.git 03Michael Niedermayer 07master:013172ae9f1a: Merge commit '67bc1ba5d75953d136bfa02ce6c0a27e9fd9dac3' [22:38] <cone-382> ffmpeg.git 03Diego Biurrun 07master:eba2233b58c2: build: Add define for SIMD extensions requiring 16-byte aligned buffers [22:38] <cone-382> ffmpeg.git 03Michael Niedermayer 07master:1df2f7b6a88e: Merge commit 'eba2233b58c2c4b468c58287d6537b2f1188a8cd' [23:27] <cone-382> ffmpeg.git 03Diego Biurrun 07master:11c7155cce50: dct/rdft: Add missing typedefs for context structs [23:27] <cone-382> ffmpeg.git 03Michael Niedermayer 07master:005e81d6cff4: Merge commit '11c7155cce50971a38e6cebd39a3ba5a54645fae' [23:29] <ePirat> michaelni, how do I set it? [23:34] <michaelni> "-method PUT" might work, didnt try [23:34] <michaelni> or whatever method you want [00:00] --- Wed Jul 23 2014

1 0

ffmpeg.log.20140722
by burek 23 Jul '14

23 Jul '14

[01:49] <a46> vlc won't stream the video of an flv correctly but piping it through avconv works. What are some good arguments to pass to avconv to make this more efficient? [01:54] <c_14> avconv support is in #libav [01:54] <c_14> but you'll probably want something like -codec copy [04:22] <joselo> Hi I'm having an issue when using ffmpeg (i'm quite new with it). Im running this command: http://pastebin.com/nQLs69R3 And the issue is that the process terminates always at 01:04 [10:14] <termos> does ffmpeg have support for trimming down the mpegts overhead? There's about an 18% overhead on my hls stream that comes from mpegts header data [10:51] <Carlos92> Hello, is it possible to show in the ffserver status (stat.html) in the colum "File" under the "Connection Status" the full requestet Filename? As example "test.flv?hereissomeextrainfo", all the stuff behind a ? is ignored in the status output, it shows only "test.flv".. [10:53] <Carlos92> I need it, because I am using a proxy befor the ffserver and I need some extra info about the client, so I was thinking I put some extra info behind the file with "?" [10:54] <Carlos92> as example: "https://mystreamingsrv/test.flv?z1hs81msn" [12:37] <a46> when I encode to mpeg, static elements e.g. unchanging text, in the video move around and blur in different directions like a motion blur when they should be unchanging. What is that? [12:41] <Keshl_> Depends on the video codec and such. It could be many things. Some stuff is better at encoding static elements than others. [12:54] <a46> Keshl_: the source is an h264 flv and the dest is -f mpeg (mpeg1video (yuv420p Lavf53.21.1)) [12:56] <sacarasc> Why mpeg 1? [13:08] <a46> I'm trying to stream it and other formats give an error [13:08] <a46> what else could I use sacarasc ? [13:09] <sfan5> mpeg4 for example [13:10] <a46> [mp4 @ 0x9bc94c0] muxer does not support non seekable output [13:30] <Mavrik> why are you trying to stream video with container that's not made for streaming? [13:32] <spaam> Mavrik: tell him what formats he should use instead ! :D [13:32] <Mavrik> I just came so I have no idea what is he trying to do. [13:32] <Mavrik> But streaming video with a format that has to have a full index in the header... won't work. :P [13:36] <pk__> hi [13:37] <pk__> i am giving this command ffmpeg -y -i /home/Carol.avi -map 0:0 [13:37] <pk__> -pix_fmt yuv420p -f rawvideo /tmp/vvv -map 0:1 -f u16le -acodec [13:37] <pk__> pcm_s16le -ac 2 -ar 44100 /tmp/aaa [13:38] <pk__> sorry let me pastebin it [13:39] <pk__> http://ur1.ca/ht0us [13:39] <pk__> if i do the two maps in two commands then they work [13:39] <pk__> but two maps in a single command do not :( [13:40] <pk__> can someone help [13:41] <pk__> i want to crate two named pipes and feed them audio and video respectively [13:45] <sfan5> I'm pretty sure ffmpeg does not support multiple output files [13:46] <pk__> sfan5: then what can i do? any other method? [13:47] <sfan5> https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs [13:47] <pk__> okay let me see [13:58] <a46> Mavrik: I'm trying to stream an h264 flv. mpeg works but it has motion blur on stuff that should remain static. someone suggested mp4 so I tried that and it doesn't work for the reason you mentioned, what could I do instead? [13:59] <jnvsor> pk__: Sounds like you just want to run 2 separate commands, one after the other [14:00] <Mavrik> a46, flv is also a header-heavy format [14:01] <Mavrik> which means your stream won't work unless your client is connected while header is sent... are you using a streaming server? [14:01] <pk__> jnvsor: but then wont there be any sync problem? [14:02] <jnvsor> pk__: It looks like you're trying to split the streams into separate files - if they aren't properly timestamped you'll have a sync problem anyway [14:02] <a46> Mavrik: flv is the source file. I'm trying to stream it so I don't have to download it to watch it. I'm piping it to VLC at the moment. [14:02] <pk__> jnvsor: u are right :) [14:03] <Mavrik> a46, is there a reason why don't you give it directly to VLC? [14:04] <a46> Mavrik: VLC can't handle the flv, it only streams the audio [14:04] <Mavrik> that's... very unlikely [14:04] <a46> Mavrik: I tried getting that to work for a long time first [14:04] <jnvsor> a46: Then it's the video codec that's the problem, not the container [14:05] <a46> if I try to stream a sample h264 file it works, if I try to stream this flv with the same settings the receiving end doesn't get a video stream [14:05] <a46> (with vlc) [14:06] <Mavrik> can you give an url? [14:06] <jnvsor> or ffprobe the file and pastebin it for us [14:07] <a46> can you access this? http://store59.media52.justin.tv/archives/2014-7-20/live_user_tomcr00s3_140… [14:07] <jnvsor> I'm getting video on vlc [14:08] <jnvsor> And ffprobe looks fine too, might just be your vlc that's the problem [14:09] <a46> yes, I've installed nightly to try to get it to work and various codecs, not sure what else to do [14:10] <jnvsor> See if you can access it with ffplay [14:12] <a46> I only have cli on the server I'm trying to stream it from. I'll try streaming it with ff and watching it with ffplay on my machine? [14:15] <a46> to clarify what I'm trying to do: I have too slow internet to stream the flv directly from the http source so I've downloaded it on a server and am trying to stream it, once that works hopefully I can tweak the quality settings to reduce the bandwith required [14:16] <a46> on the server I can't get vlc to send the video stream so I resorted to converting it to mpeg with ffmpeg and piping it to vlc which works but the quality is abysmal [14:19] <Baluse2> hello [14:27] <Mavrik> a46, ffplay on your url works perfectly fine [14:28] <Mavrik> except huge amount of h.264 errors O.o [14:29] <a46> "stream 0 codec frame rate differs from container frame rate: 6000.00 (6000/1) -> 30.00 (30/1)" [14:29] <a46> I bet that's the issue vlc has with it [14:30] <Baluse2> I converted an mp3 to adpcm_ms and time has tripled in players! But plays fine [14:38] <a46> ok this + ffplay works: ffmpeg -i input.flv -acodec copy -vcodec copy -f flv tcp://10.0.0.1:1234?listen [14:39] <a46> with all the h264 errors like you mentioned [17:08] <drsprite> Hey there. I'm trying ffmpeg out for a timelapse I'm working on. Right now I have a test timelapse that's only 3 seconds, but the file is 350mb. That's quite large for 3 seconds. My final product will be about 5 minutes and I fear it'll be huge. Any thoughts on how to optimize the file size? The command I used to create it is here: http://pastebin.com/TQnjYvjn [17:16] <sfan5> drsprite: can you paste the output of "ffprobe timelapse.avi"? [17:17] <drsprite> ah, I suppose it's 7 seconds. Here's the output. http://pastebin.com/gufe7pwz [17:17] <drsprite> I'm still learning as I go [17:19] <krullie> The stream has a bitrate of 40000. You should tell the encoder to limit to a certain amount using -b:v 9M or some other number [17:20] <drsprite> Okay, i'll read up on that a bit [18:09] <slowguy> i want to create an app which can take an mpeg stream from network and write it to a file [18:10] <slowguy> challenge is that it should have a button ..as long as this button is pressed (like during advertisements) it should not feed video to the file during this time [18:10] <slowguy> can somebody suggest me an approach? [19:00] <a46> capture network traffic with winpcap [19:00] <a46> parse for mpeg [19:00] <a46> write to file unless button pressed [19:07] <Hello71> generic video question: playing h264 video and playing without a valid reference frame produces green video [19:07] <Hello71> why is it green? [19:08] <JEEB> most probably decoder-specific, unless you have enough data for the decoder to do some things. Anyways, green sounds like the default values for the YCbCr picture [19:09] <Hello71> I was figuring something along the colorspace lines (green intensity). [19:10] <Hello71> vp8 skips to the next keyframe with "[ffmpeg/video] vp8: Discarding interframe without a prior keyframe!" [19:10] <Hello71> obviously I'm not relying on this behavior for anything, just wondering why it would be green [19:12] <Hello71> theora kinda does the same thing, but it has less intraframe predictors so mostly "clean" [19:13] <Hello71> it's also green though [19:13] <JEEB> well, I don't remember what values the YCbCr planes get filled with, but I wouldn't be surprised if it was something that became green :P [19:13] <Hello71> hm, I guess that might even be player-specific [19:13] <Mavrik> green is usually the color used by overlay keying as well [19:14] <JEEB> more like decoder specific [19:14] <JEEB> since the decoder is giving you the YCbCr planes [19:14] <Mavrik> e.g. your OS draws overlay area green when compositing desktop and GPU then draws image from HW overlay on top of the green area [19:14] <JEEB> and then the player just grabs that and converts to RGB and shows it on the screen :P [19:14] <Hello71> possibly, I'm using vaapi for both decoding and rendering [19:15] <Hello71> ah, using --hwdec=no --vo=x11 makes it grey instead [19:16] <JEEB> that sounds like lavc decoding getting used, and yes -- grey does sound a bell [19:16] <JEEB> (I think that's 128 or so for all planes?) [19:16] <Hello71> yeah, --vo=vaapi --hwdec=vaapi gives green [19:16] <Hello71> everything else is grey [19:16] <JEEB> the rendering method really shouldn't matter, as it's the decode that produces the YCbCr image [19:16] <JEEB> as I said [19:17] <JEEB> *decoder [19:18] <Hello71> wait, no, it's just the decoder. you're right [19:20] <Hello71> https://happinessforme.com/~alex/shot0001.jpg --hwdec=no, https://happinessforme.com/~alex/shot0002.jpg --hwdec=vaapi [19:20] <Hello71> intel hd graphics... 3000? i think [19:20] <JEEB> anyways, it's just the initial value of the planes before decoding [19:20] <JEEB> nothing special [19:20] <JEEB> lavc uses 128 (?) or so for all planes [19:21] <JEEB> vaapi uses something that ends up being green, I guess? [19:21] <Hello71> probably an internal buffer in the graphics device [19:21] <Hello71> I don't know. [19:22] <Hello71> I think it's pretty cool how it shows all the inter-frame prediction the encoders do. [19:24] <BigArah> Hey everyone [19:24] <BigArah> working on trying to get mp4s encoded properly so they'll play in html5 editors [19:25] <BigArah> I mean embedded html5 players like JW Player [19:25] <BigArah> I have an mp4 I generated that when I use the jwplayer it only plays back the music [19:25] <BigArah> but if I upload it to youtube, then download it back from youtube, and put that file in the jwplayer it does both the audio and the video [19:25] <BigArah> any ideas on that? [19:27] <relaxed> youtube encodes everything you upload, I believe [19:27] <BigArah> yeah [19:27] <BigArah> and they compress it well too [19:27] <BigArah> i uploaded a 6.5MB video and downloaded a 2.7MB version that was proper [19:28] <relaxed> pastebin your command and all console output of a video jwplayer won't play. [19:31] <BigArah> ok on the way [19:31] <BigArah> http://pastebin.com/fDskBxZ0 [19:41] <relaxed> that's not what I asked for. [19:46] <relaxed> BigArah: pastebin the console output from the command [19:54] <BigArah> I fixed it [19:54] <BigArah> let me give you code [19:54] <BigArah> ffmpeg -i input.mp4 -vcodec libx264 -pix_fmt yuv420p -profile:v baseline -preset slower -crf 18 -vf "scale=trunc(in_w/2)*2:trunc(in_h/2)*2" output.mp4 [20:10] <BigArah> hey relaxed [20:10] <BigArah> Do you know how to overlay a png image on a video? like a logo/watermark sort of thing? [20:13] <relaxed> look at -filter_complex in the man page or website [20:40] <daala-person> Hello. [20:41] <daala-person> I'm having an issue building ffmpeg. [20:41] <daala-person> I'm using Linux Mint, and its "ffmpeg" is really avconv, so I have to build ffmpeg myself. [20:43] <daala-person> I'm trying to build ffmpeg according to the instructions on https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu , installing all the extra codecs execpt for libfdk-aac. [20:44] <daala-person> Everything's fine until I try to run ffmpeg's ./configure (I can paste the command I used if you want), when I get: [20:44] <daala-person> ERROR: libvpx decoder version must be >=0.9.1 [20:44] <sfan5> the libvpx mint has is too old [20:44] <sfan5> if you don't need it disable it [20:45] <daala-person> Hello, sfan5. [20:45] <daala-person> I would do that, except I'm using libvpx 1.3.0, just downloaded today. [20:45] <daala-person> Yes, I need libvpx. [20:46] <sfan5> what does apt-cache show libvpx say? [20:46] <daala-person> $ apt-cache show libvpx [20:46] <daala-person> N: Unable to locate package libvpx [20:46] <daala-person> E: No packages found [20:46] <daala-person> It seems that mint doesn't even package libvpx. [20:46] <sfan5> try apt-cache show libvpx1 [20:46] <relaxed> dpkg -l "libvpx*" | grep ^ii [20:47] <daala-person> dpkg -l "libvpx*" | grep ^ii [20:47] <daala-person> ii libvpx1 1.0.0-1 VP8 video codec (shared library) [20:47] <sfan5> it seems to be 1.0.0 not 1.3.0 [20:47] <relaxed> you need to point ffmpeg's configure to the newer version with --extra-cflags="-I/path/to/prefix/include" --extra-ldflags="-L/path/to/prefix/lib" [20:47] <relaxed> if you did install 1.3.0, that is [20:48] <daala-person> I think I know what's happening. I must have 1.0.0 installed to my system directories, but I have 1.3.0 built in my home directory. [20:48] <relaxed> where "/path/to/prefix" is "$HOME/ffmpeg_build" in your case [20:49] <relaxed> you can have both installed [20:49] <daala-person> Thank you. [20:49] <daala-person> I'll give that a try. [20:49] <llogan> the compile guide shows how to do this. i'm not sure if you referred to that or not [20:50] <daala-person> I did everything according to the compile guide. [20:50] <daala-person> Except I used different directory names. [20:51] <relaxed> did you change the cflags path to match these changes? [20:51] <daala-person> I'm doing that right now. [20:56] <daala-person> I think that libvpx may have failed to leave its libraries in the right place or something. If I list the files in the directories you told my to point configure to: [20:56] <daala-person> $ ls /home/thomas/tmp/ffmpeg-build/lib/ [20:56] <daala-person> libopus.a libopus.la libx264.a libyasm.a pkgconfig [20:56] <daala-person> $ ls /home/thomas/tmp/ffmpeg-build/include/ [20:56] <daala-person> libyasm libyasm.h libyasm-stdint.h opus x264_config.h x264.h [20:57] <daala-person> Nothing about libvpx. Is that correct? [20:58] <voip1__> hello guys [20:58] <voip1__> i have troble with running ffmpeg in daemon mode [20:58] <voip1__> for example ffmpeg -i rtmp://1.1.1.1:1935/stream-sd -preset fast -acodec aac -strict experimental -ar 44100 -ac 2 -b:a 96k -vcodec libx264 -b:v 900k -f flv rtmp://2.2.2.2:1935/live/ex & [20:58] <relaxed> daala-person: why don't you just follow the guide to a T until you get the hang of it. [20:59] <daala-person> Okay, I'll do that and see if it works. [20:59] <relaxed> or use the same install prefix, cflags, and ldflags for everything... [21:00] <relaxed> voip1__: ffmpeg .... -f flv rtmp://2.2.2.2:1935/live/ex </dev/null 2>ffmpeg.log & [21:01] <voip1__> relaxed, </dev/null 2>ffmpeg.log nessesary part [21:01] <voip1__> ? [21:02] <llogan> ffmpeg has a "daemon mode" if you use -d option as first option. [21:02] <voip1__> i also tryed with -d same problem [21:02] <relaxed> llogan: fancy [21:03] <llogan> i've never tried it [21:03] <relaxed> voip1__: ffmpeg expects keyboard input, so that's why you need </dev/null [21:04] <voip1__> good its works with </dev/null ! [21:05] <relaxed> 2>ffmpeg.log sends the console output to a log file. use 2>/dev/null if you don't need a log. [21:05] <Peter_Occ> /msg NickServ identify meltdown [21:06] <voip1__> yaeh i dont neeed see output ! [21:06] <relaxed> Peter_Occ: I think you just exposed your password :( [21:06] <Peter_Occ> yeah but I did it correctly, so I don't get it. [21:06] <relaxed> there was a space before /msg [21:07] <voip1__> brillinat [21:07] <voip1__> thank you relaxed [21:07] <relaxed> you're welcome [21:07] <Peter_Occ> Ok thanks [21:09] <voip1__> relaxed, one more cuestion [21:09] <voip1__> some times ffmpeg terminatde whe disapear incoming stream [21:10] <voip1__> di you know any good script for re runing [21:13] <relaxed> forever? [21:15] <relaxed> while :; do ffmpeg -i blah output; done [21:15] <llogan> Peter_Occ: better to run that stuff in a non-channel buffer [21:16] <Peter_Occ> OK [21:16] <relaxed> voip1__: for backgrounding ffmpeg you'll need to remove the ";" before "done" [21:17] <voip1__> ok, thank you ! [21:19] <Peter_Occ> Has anyone noticed that the ffmpeg timestamp start with the wrong timestamp? For example, here is a two minute video. http://geotonics.com/testimages/1405956001.mp4 The video starts at the correct time, but it starts with a timestamp 6 seconds ahead. The timestamp is then frozen for six seconds. When the video gets to the time already indicated in the timestamp, the timestamp then resumes... [21:19] <Peter_Occ> ...with the correct time. Is this a known problem? [21:24] <BigArah> Hey Peter_Occ [21:24] <BigArah> I know I have run commands where they tell me the time stamp might be messed up [21:24] <BigArah> but it never happens [21:24] <BigArah> relaxed [21:24] <BigArah> I tried your idea [21:24] <BigArah> http://pastebin.com/LfmmUUJH [21:24] <BigArah> here is the output, #epicfail [21:24] <BigArah> probably something stupid I did [21:26] <relaxed> BigArah: I don't think you can use -filter_complex with -vf [21:30] <relaxed> BigArah: Do me a favor and grab a recent version of my build. The one you have was compiled with gcc-4.9.0 which has issues. [21:30] <BigArah> man I suck way too bad at this to do that, lol [21:30] <BigArah> I am PHP / MySQL [21:30] <BigArah> and I suck at that too, lololol [21:32] <relaxed> download http://johnvansickle.com/ffmpeg/builds/ffmpeg-git-20140722-64bit-static.tar… and replace the ffmpeg on your system with the one in this archive. [21:32] <BigArah> right, I know what you mean [21:32] <Peter_Occ> BigArah, the timestamp is definitely wrong for the first six seconds. In the example I linked to, the timestamp is frozen for the first six seconds. [21:32] <BigArah> that doesn't mean I am capable of it! [21:33] <BigArah> hey relaxed, tell me if I can do it on this command [21:40] <daala-person> FFmpeg is building right now. No errors, but a few warnings about the code. [21:40] <daala-person> Building it in the directory I'd like should be trivial. [21:41] <daala-person> Thanks, everyone. [23:24] <daala-person> I have a somewhat stupid question. I couldn't find this anywhere online. Is there a way to list the private options for a codec? [00:00] --- Wed Jul 23 2014

1 0

ffmpeg-devel.log.20140721
by burek 22 Jul '14

22 Jul '14

[00:01] <cone-390> ffmpeg.git 03Andreas Cadhalpun 07master:e59ce54412f7: doc: fix two spelling errors [01:26] <thardin> any ffmpeg devs going to europython? [02:01] <cone-390> ffmpeg.git 03Diego Biurrun 07master:4de8b60684ce: idct: Move arm-specific declarations to a header in the arm directory [02:01] <cone-390> ffmpeg.git 03Michael Niedermayer 07master:110420aac01d: Merge commit '4de8b60684ce13dff3e3d372dae4f49b9e53f755' [02:12] <cone-390> ffmpeg.git 03Carl Eugen Hoyos 07master:b31d76e45fc3: tiff: support reading gray+alpha at 8 bits [02:12] <cone-390> ffmpeg.git 03Michael Niedermayer 07master:08b597b1f8f2: Merge commit 'b31d76e45fc3c6529dd7109e721676f3ec376d00' [02:22] <cone-390> ffmpeg.git 03Ronald S. Bultje 07master:c39059bea3ad: h264: Fix direct temporal mvs for bottom-field-first poc order [02:22] <cone-390> ffmpeg.git 03Michael Niedermayer 07master:dc021a0218e1: Merge commit 'c39059bea3adebcd888571d1181db215eee54495' [02:34] <cone-390> ffmpeg.git 03Vittorio Giovara 07master:819bb3e81c72: fate: add h264-direct-bff test [02:34] <cone-390> ffmpeg.git 03Michael Niedermayer 07master:ec59f027ac74: Merge commit '819bb3e81c72939ae1c1a4e5fc3dfdf3e98df074' [03:25] <cone-390> ffmpeg.git 03Michael Niedermayer 07master:437da3e349c2: avfilter/vf_owdenoise: use av_malloc_array() [03:25] <cone-390> ffmpeg.git 03Michael Niedermayer 07master:71e515c180a5: avformat/sapenc: Use av_mallocz_array() [04:22] <cone-390> ffmpeg.git 03Muhammad Faiz 07master:ce385c869c0d: avfilter/biquads: avoid clipping when using floating type [07:49] <anshul> in get_bits function why are we doing av_assert2(n>0 && n<=25); [07:49] <anshul> why not only av_assert2(n<=25); [07:55] <nevcairiel> Because <= 0 is bad? [09:22] <anshul> ok i was looking for condition when this <= 0 arise, anyway leave it will find out myself [09:27] <nevcairiel> assert2 is usually disabled anyway [12:55] <michaelni> db0company, as you are maintaining our webpage, you should have op here [13:18] <cone-601> ffmpeg.git 03Janne Grunau 07master:a9f3f5fadb57: Revert "tiff: support reading gray+alpha at 8 bits" [13:18] <cone-601> ffmpeg.git 03Michael Niedermayer 07master:5b998dae597a: Merge commit 'a9f3f5fadb57bae3f3ff0be69e56b2c6014f2513' [13:51] <cone-601> ffmpeg.git 03Diego Biurrun 07master:81b9bf319226: dct-test: Move arch-specific bits into arch-specific subdirectories [13:51] <cone-601> ffmpeg.git 03Michael Niedermayer 07master:d986c414dea1: Merge commit '81b9bf319226fe03436c80aaa8a2c91767cab7ce' [13:57] <Compn> harumph [13:57] <Compn> new website looks great [13:57] <Compn> i will make some small tweaks and send patches. [13:58] <Compn> e.g. putting packages above source on download page [13:58] <Compn> putting OS icons on the download buttons [13:58] <Compn> btw ffmpeg website looks like mplayer website colorscheme now :D [14:00] <cone-601> ffmpeg.git 03Nidhi Makhijani 07master:c3ec963d0d0f: mjpeg: return proper error code [14:00] <cone-601> ffmpeg.git 03Michael Niedermayer 07master:c05dbb8744f5: Merge commit 'c3ec963d0d0fac2d817f7fbce2e89ed46eeba550' [14:01] <Compn> oh, the os icons show up with javascript [14:01] <Compn> but the icons dont show up if js is disabled [14:01] <Compn> i see. [14:02] <Compn> did we ever get the copyright on the website design ? [14:02] <Compn> db0company : is your design under CC or gpl license ? [14:07] <cone-601> ffmpeg.git 03Michael Niedermayer 07master:40541716661e: avcodec/dct-test: remove fixed array sizes [14:17] <ubitux> still no tweet about 2.3 and the new website? [14:20] <db0company> I just tweeted it o/ [14:20] <ubitux> db0company: we have a twitter/ffmpeg [14:22] <db0company> I know, I mentioned it @FFmpeg [14:32] <kierank> new site looks nice [14:39] <wm4> typical hipster design [14:39] <wm4> with a download button covering half of the screen, literally [14:40] <wm4> but I guess the only effect that needs to be achieved is looking modern [14:40] <db0company> I like hipster design :) [14:41] <db0company> I also tried to make it look accessible for newbies even though I haven't been asked to do so [14:47] <Compn> i like giant button for newbies [14:47] <Compn> :P [14:47] <Compn> ubitux : i think logan is twitter guy, and he said he is without internets [14:48] <ubitux> yes it's llogan [14:48] <Compn> i'm also happy its not-libav design anymore :P [14:48] <Compn> db0company : what license is the site ? i forgot... [14:54] <db0company> there were no license when I started editing it and I'm not sure if I can just add one myself, I don't own most of the code [14:55] <db0company> I usually use apache2 for some reason but I don't think it would fit here [14:57] <Compn> oh ok [14:57] <Compn> so its based on the old website code ? [14:58] <Compn> thats ok :) [14:58] <Compn> the old author gave ffmpeg permission to use that :) [14:58] <Compn> i just mean your parts are under some kind of open source, thats fine with me [15:21] <cone-601> ffmpeg.git 03Muhammad Faiz 07master:4927c0a9880b: avutil/opt.c: fix bug when parsing option [16:31] <ubitux> j-b: is it possible to get/print the exact libav* version vlc is built against? [16:31] <ubitux> mmh, and the one used at runtime [17:01] <cone-601> ffmpeg.git 03Michael Niedermayer 07master:2c5c7f108926: avutil/opt: remove dead code from set_string_number() [17:01] <cone-601> ffmpeg.git 03Michael Niedermayer 07master:88efc013b10c: avutil: move QP2LAMBDA constant to eval [17:01] <cone-601> ffmpeg.git 03Michael Niedermayer 07master:265dadbe7de5: avutil/opt: Support general expressions involving AVOption constants in set_string_number() [17:04] <db0company> Compn: but I think it's a good idea to add a license [17:44] <Compn> db0company : yes, we've had issues with copyrights in the past ... :\ [18:08] <db0company> do you think that'd be a problem if I add one? considering i'm not the only contributor and I can't really reach all of them one by one to ask [18:19] <j-b> ubitux: depends, if yo provide the function or not [18:19] <Compn> db0company: yeah, i realize that. maybe just a notice that any of your work is licensed under whichever license you choose [18:20] <Compn> i put all my stuff in public domain. i never really wrote it down anywhere except a mail or two on the mailing list [19:51] <kurosu> BBB: there's a benefit to using x/y reporting [19:51] <kurosu> but that's actually less trivial that one may have thought [19:51] <kurosu> https://github.com/kurosu/ffmpeg/commits/hevc_mt [19:51] <kurosu> around 2% on my dualcore (seems to run 3 or 4 threads?) [19:52] <kurosu> more than 5% on the samples I tested with on a 6-cores [19:55] <kurosu> maybe not the best design, but the raster_scan / increment is required [19:55] <kurosu> first because of the way dbf and sao are ran, second because of wpp/slices [21:51] <cone-227> ffmpeg.git 03Diego Biurrun 07master:bd499d9af668: build: Conditionally build and test iirfilter [21:51] <cone-227> ffmpeg.git 03Michael Niedermayer 07master:94c4ad27dc63: doc/APIchanges: mention FF_INPUT_BUFFER_PADDING_SIZE chage [21:51] <cone-227> ffmpeg.git 03Michael Niedermayer 07master:79df91bfc403: Merge commit 'bd499d9af668aef979ec9f3f3215b8dd508c7ec1' [22:21] <cone-227> ffmpeg.git 03Diego Biurrun 07master:16b7328058fa: build: Conditionally build and run DCT test program [22:22] <cone-227> ffmpeg.git 03Michael Niedermayer 07master:9a436ae30f10: Merge commit '16b7328058fa600d5158c84d9cc621a134eb88bc' [23:05] <cone-227> ffmpeg.git 03Diego Biurrun 07master:2ce53880bfeb: build: Conditionally compile all avcodec test programs [23:05] <cone-227> ffmpeg.git 03Michael Niedermayer 07master:16c66f3151cf: Merge commit '2ce53880bfeb431decdd2fc82298c0aff823ab47' [23:23] <cone-227> ffmpeg.git 03Katerina Barone-Adesi 07master:4d2d0595cf9d: fate: Split audio- and video-related RealMedia targets [23:23] <cone-227> ffmpeg.git 03Michael Niedermayer 07master:c75c594e5a73: Merge commit '4d2d0595cf9dd199c35aefef450952d209a56573' [00:00] --- Tue Jul 22 2014

1 0

ffmpeg.log.20140721
by burek 22 Jul '14

22 Jul '14

[00:05] <ac_slater> hey all, I want to make a ffmpeg plugin or something for my metadata stream. I have a custom binary format that I want to associate with some AV data. Is there a plugins guide or a 'extending ffmpeg' guide [00:14] <c_14> phingers: try using -framerate instead of -r on the input side, also h264 does not look like a valid idct according to the manpages. [00:29] <phingers> I was on an old server, I switched out to a new rhel6 and the code is working much better [00:30] <phingers> This is working well now - ffmpeg -r 1/10 -i foo%03d.png -i audio.wav -r 32 -c:v libx264 -strict experimental -b:a 192k -pix_fmt yuv420p -shortest -vf drawtext="fontsize=30:fontcolor=white@0.9:fontfile=FreeSerif.ttf:shadowy=1:shadowx=1:text='I am getting the hang of ffmpeg\!':x=(w-text_w)/2:y=(h-text_h-line_h)/2" output.mp4 [00:30] <phingers> But I want to control when the text appears and disappears [00:30] <phingers> start_number and n don't seem to be working [00:32] <phingers> plus I have three images, and it seems to be only picking up 2 [00:33] <c_14> try switching from -r in the input to -framerate, using -r can cause weird things to happen [00:37] <c_14> And judging by the examples from the manpage you might be able to control when the text appears using the enable= expcession [00:37] <phingers> actually the opposite happened adding in framerate and dropping r caused the filesize to go from 1706623 to 353171 - it lost the audio and skips to 40 seconds in when I hit play [00:39] <phingers> http://pastie.org/9408017 [00:40] <c_14> LIt's telling me that pastie does not exist [00:40] <phingers> http://pastie.org/9408019 [00:40] <phingers> is the other one [00:40] <c_14> ah, now it does [00:41] <phingers> http://pastie.org/9408021 [00:41] <c_14> The input framerate option, not the output one [00:41] <phingers> the only difference is the r vs framerate [00:41] <c_14> ffmpeg -framerate 1/10 [00:41] <phingers> oh, sorry [00:43] <phingers> that works, I have 6 images though, and it is only picking up the first 5 [00:44] <c_14> Ok, let's start by updating ffmpeg... 0.10.11 is rather how do I put it, dated. [00:45] <c_14> try grabbing a static build [00:49] <phingers> http://pastie.org/9408032 [00:49] <phingers> I updated ffmpeg - same issue though [00:49] <phingers> playing with expression [00:52] <c_14> Ok, you don't need -strict experimental if you're going to be using libvo_aacenc (though the builtin might be better, use -c:a aac if you want that one). The next thing I'd try is either removing the filter and see if that gives you all the frames and/or removing the audio/the -shortest and see if that helps. [00:52] <c_14> If all that doesn't help then I remember somebody else having similar issues, but I can't remember how I ended up helping him... [00:54] <phingers> audio is 4 mins long - so shortest is def needed [00:54] <phingers> I'm all for simplifying the command. [00:54] <phingers> I'm going to add in a text to speech web service call, any recommendations? [00:55] <phingers> Do you mess with ffmpeg for a living? [00:55] <c_14> Mainly just for fun. [00:55] <c_14> What do you mean 'add in a text to speech web service call'? [00:55] <phingers> I want to overlay custom text on top of the audio track [00:56] <phingers> can I town down the one audio track to be background music? [00:56] <phingers> tone [00:57] <c_14> You can cut the audio into three sections and then fade/lower the volume of the middle section. [00:59] <phingers> I just want the whole audio track to run quieter [00:59] <phingers> like I do with text, opacity 0.2 [00:59] <c_14> Just use the volume filter. [01:05] <phingers> okay, I think the -af worked, how do I add in the second audio file at 100% volume? [01:06] <c_14> you'll probably have to switch to a filter_complex something along the lines of '[0:a]volume:foo[aquiet];[aquiet][1:a]concat:n=2:a=1:v=0[a]' -map '[a]' [01:07] <phingers> it wont let me add two audio streams I dont think [01:07] <c_14> hmm? [01:07] <phingers> I put both and only the audio comes, not the speak [01:08] <c_14> commandline? [01:08] <phingers> http://pastie.org/9408058 [01:09] <phingers> does it serialize the audio? [01:10] <c_14> your current commandline doesn't concatenate the audio, it'll create an output file with two audio streams. [01:10] <c_14> Which is why I said you'll probably need a filter_complex with the concat in it. [01:13] <phingers> nailed it. Wow [01:13] <phingers> amazing, who knew ffmpeg was so powerful [01:14] <phingers> so I have them combined into one output, but still need to lower the first one's volume [01:14] <c_14> ye, look at the example I gave you. [01:15] <phingers> you'll have to break it down for me, it seems complex. [01:15] <phingers> I have -filter_complex amix=inputs=2:duration=first [01:17] <c_14> You wanted to mix them, ok. Then [0:a]volume=foo[quiet] <- this'll lower the volume of the audio stream of the first input file and "store" it in quiet, then [quiet][1:a]amix=foo[a] <- this'll take [quiet] and the audio stream of the second input file, mix it and store it in [a], connect those two with a ';' in the filter complex and then add -map '[a]' [01:19] <phingers> http://pastie.org/9408071 [01:20] <phingers> must be missing something simple [01:20] <phingers> explanation makes sense [01:20] <c_14> Ehh, you weren't supposed to take that foo literally. [01:20] <phingers> lol [01:21] <c_14> and that amix= part is off as well [01:21] <c_14> Just to get it straight, do you want to mix the tracks or concatenate them? [01:21] <phingers> mix [01:21] <phingers> one is the elevator music in the background, the other is the "doctor" speaking instructions [01:22] <phingers> what do I replace foo with then? [01:23] <phingers> somehow I name my inputs? [01:23] <c_14> '[1:a]volume:<options>[aquiet];[aquiet][2:a]amix:<options>[a];[0:v]drawtext=<options>[v]' -map '[a]' -map '[v]' [01:23] <c_14> https://ffmpeg.org/ffmpeg-filters.html#volume [01:23] <c_14> Look at the documentation of each of the filters on what to replace <options> with. [01:24] <c_14> And yes, you'll need to put the drawtext filter into the filter_complex because I'm pretty sure you can't have a filter_complex and a -vf in the same commandline. One of them gets ignored. That's how it is with multiple -vfs anyway. [01:28] <phingers> I dont see any options for volume [01:29] <c_14> volume=volume=0.5 will have the volume for example [01:30] <phingers> http://pastie.org/9408097 [01:30] <phingers> close [01:31] <c_14> Your text= has quotations, you'll need to escape them. [01:31] <c_14> try escaping them once, if that doesn't work, try twice etc etc. It can be a bit tricky sometimes [01:32] <phingers> the single or double quote? [01:32] <phingers> single escaped twice gives same error [01:33] <c_14> The single [01:34] <c_14> You might be able to replace the single quotes around the text with double quotes. [01:34] <phingers> I have double quotes encompassing the drawtext? [01:34] <phingers> those arent needed? [01:35] <c_14> you can probably get rid of them, those are usually only for escaping on the shell level, the single quotes around the whole filtergraph should take care of that. [01:36] <phingers> dropping the double, and adding it in around the text did it [01:37] <phingers> crazy stuff - its working. [01:38] <phingers> You are quite the master at this. I owe you a beer. [01:38] <phingers> but it isnt taking the shortest bit [01:40] <phingers> I want it to be as long as the slides are. [01:41] <c_14> try placing the -shortest before the map? it should work though... [01:41] <c_14> That or switch to duration=shortest in the amix options. [01:41] <c_14> That could be interfering. [01:44] <phingers> the shortest on the audio portion stops after the talking sound is done [01:44] <phingers> it needs to be mapped to the length of the video somehow [01:44] <c_14> ahh [01:45] <phingers> almost need another complex filter to map those two for shortest? [01:46] <c_14> The -shortest should be working though... You could always just add a -t output option and forcibly halt if you know how long the stream should be. [01:47] <phingers> tried shortest before and after the maps [01:47] <phingers> still taking the full audio length [01:47] <c_14> I'm gonna try testing something on my machine. [01:49] <phingers> I can send you my files if you'd like [01:50] <c_14> Sure, that would help. [01:55] <phingers> emailed [01:56] <phingers> bounced too big. [01:57] <c_14> Mhm, well I think I might have found the cause. If I don't apply the drawtext filter -shortest works, so I'm seeing if I can find a workaround for taht. [01:59] <phingers> I have to walk the dogs, back as soon as their bladders are empty [02:02] <c_14> kk [02:04] <onyx> hey guys, anyone with experience running ffmpeg using cron? [02:05] <onyx> I cant get it to work... I been researching all day [02:07] <onyx> Im running in root cron tab [02:07] <onyx> this doesnt work http://pastebin.com/YsLjzQsR [02:08] <c_14> onyx: running ffmpeg as root is usually considered overkill/bad practice, I'll look at it in a sec though [02:08] <onyx> root is the only user i have so far thats why [02:08] <c_14> phingers: As soon as you add the video into the filter_complex even if you just pass it through the null filter -shortest appears to lose effect. I don't know why, but to me this is probably a bug and I'd report it as such. In your case since you know the output video is supposed to be 25 seconds you can just add -t 25 as an output option and that'll limit it. [02:09] <c_14> onyx: Does that command work if you run it in your shell? [02:09] <phingers> onyx: it should run in a cron the same as a command line more or less [02:09] <phingers> onyx: specify full paths in cron [02:09] <c_14> ^That too. [02:10] <onyx> I read that full path thing but I dont understand what that means [02:10] <c_14> /path/to/ffmpeg [02:10] <phingers> no .. [02:10] <onyx> I tried creating a record.sh file and running that [02:10] <onyx> http://pastebin.com/XSw8JRKC [02:11] <onyx> if I execute that record.sh it works [02:11] <onyx> but in the cront job doesnt work [02:11] <phingers> onyx: /usr/bin/ffmpeg parameters /pathto/var/www/vod/video.mp4 [02:11] <phingers> you need to specify full path for ffmpeg and the output file [02:11] <phingers> which ffmpeg on your system [02:12] <onyx> ok one sec [02:12] <phingers> assuming you use the default one for the system [02:12] <onyx> is showing this path /usr/local/bin/ffmpeg [02:12] <phingers> k so put that in your cron [02:13] <phingers> and specify full path to output [02:13] <onyx> full path for output file will be /var/www.vod/video.mp4 [02:14] <phingers> c_14: where is the best place to report the bug? [02:14] <onyx> sorry this path /var/www/vod/video.mp4 [02:14] <c_14> https://ffmpeg.org/bugreports.html [02:14] <c_14> The important part is near the bottom. [02:23] <phingers> https://trac.ffmpeg.org/ticket/3789#ticket [02:24] <phingers> what other video creation tools do you know about? [02:25] <phingers> basically, I need to have different text fade in and out on each slide, and even stick a video clip in between the slides, with various audio clips as well. [02:25] <phingers> it almost seems like I would need to run multiple ffmpeg commands to build up the video needed [02:27] <c_14> Hmm, either that or look at some sort of actual full-on video editing software. Some movie editing thing or something. Don't know of any good ones off the top of my head though. [02:28] <phingers> needs to be dynamic running off the server [02:28] <phingers> only other one I know of is https://developer.stupeflix.com/documentation/ [02:28] <c_14> Your best bet might be to programatically combine several ffmpeg commands. [02:28] <phingers> but their support stinks [02:29] <phingers> slide one, slide one text, slide one audio - save [02:29] <phingers> rinse repeat [02:29] <phingers> combine them all with an background audio into one big file? [02:30] <c_14> something like that, sure [02:32] <onyx> ok, it works now [02:32] <onyx> the only issue is that when I add the `date +%d-%H` to the file name it doesnt work [02:33] <c_14> I'm guessing cron doesn't like that. [02:33] <c_14> you could always /bin/bash -c 'ffmpeg [foobar]' [02:33] <onyx> lost me there [02:34] <c_14> `date +%d-%H` is shell expansion, cron probably won't do that [02:34] <c_14> so fix it by spawning a shell using cron and running the command there [02:35] <Riviera> nah, it's not that [02:35] <phingers> look at this example: touch "/tmp/$(date +%d-%m-%Y).sql" [02:35] <Riviera> cron does start a shell, so the backticks won't be a problem (don't use backticks, though, use $()) [02:35] <Riviera> the issue is that % is special for many cron variants [02:35] <phingers> ^exactly [02:35] <Riviera> use \% instead of % [02:35] <c_14> aah, did not know that [02:35] <c_14> My cron commands tend not to have '%'s in them. [02:36] <Riviera> "Percent-signs (%) in the comand, unless escaped with backslash (\), will be changed into newline characters [..]" from "man 5 crontab" on this system, which uses Vixie cron. [02:37] <onyx> ok so I changed it to `date +\%d-\%H` [02:37] <c_14> Hmm, yeah here too. Good to know. [02:38] <phingers> no ticks [02:38] <phingers> parens [02:38] <phingers> with a dollar sign [02:38] <phingers> see the example above [02:39] <onyx> oh so $(date +\%d-\%H) [02:39] <onyx> ^like that? [02:39] <phingers> try it out [02:39] <Riviera> (Also it's typically better to use a date format in filenames that sorts lexicographically the same as chronologically, maybe better %Y-%m-%d.) [02:41] <onyx> ok, waiting for cron to kick in [02:43] <phingers> okay, my attempt at the first slides are good, how do I pad it with some space at the begining or end of the audio if I want? [02:43] <c_14> apad will pad the end [02:43] <c_14> you can use an aevalsrc=0 and a concat to pad the front [02:46] <onyx> found this [02:46] <onyx> You can not use % in the command area. They will need to be escaped and if used with command substitution like the date command you can put it in backticks. Ex. `date +\%Y-\%m-\%d`. Or use bash's command substituion $() [02:52] <onyx> that did it, cron is working now [02:52] <onyx> thank you so much guys! [02:52] <onyx> really appreciate it [02:53] <onyx> ffmpeg ftw! [02:54] <Riviera> cool [02:55] <Riviera> $() is a feature of POSIX sh, not exclusive to bash, though. [02:58] <Hello71> `` is posix too, right? [02:59] <Riviera> yup, but older than that [02:59] <phingers> I have three videos created, now I just mash them together with an audio track? [03:00] <c_14> yep [03:00] <phingers> seems to easy [03:00] <phingers> black screen with audio only [03:02] <phingers> http://pastie.org/9408208 [03:02] <c_14> You'll probably want to actually concat the videos [03:03] <c_14> -filter_complex '[0:v][1:v][2:v]concat=n=3:v=1:a=0[v]' -map '[v]' [03:04] <phingers> I want to keep those video's audio though, and add the new sound in as a low background music [03:05] <c_14> Are we back at the mixing stage? [03:05] <phingers> lol [03:05] <c_14> -filter_complex '[0][1][2]concat=n=3:v=1:a=1[v][a];[a][3]amix[out]' -map '[v]' -map '[out]' [03:05] <c_14> or something [03:05] <c_14> ah, wait [03:05] <c_14> forgot the volume filter [03:05] <c_14> but you get the jist [03:08] <phingers> it isnt accepting shortest again [03:08] <phingers> same issue I'd guess [03:08] <c_14> I feel like we've been here before. [03:08] <c_14> Yeah probably. [03:16] <phingers> its mixing the audio and videos up [03:17] <c_14> with the concat? [03:17] <c_14> you might need to specify them seperately [03:17] <c_14> ie [0:v][0:a] etc [03:18] <phingers> I'm double checking the raw files. [03:20] <phingers> http://pastie.org/9408240 [03:20] <phingers> its not picking up audio from this clip generation [03:21] <c_14> Have you tried playing/ffprobing the file to check it has audio? [03:21] <phingers> hmm... its maybe just the sublime embeded player [03:21] <phingers> direct in browser it has audio [03:21] <phingers> thats weird. [03:25] <phingers> what did you mean by separately? Sorry I think my brain is turning to mush now with these complex_filters [03:26] <c_14> If the command is erroring out you can try specifying the inputs for the concat filter seperately, ie instead of specifying the whole file with [0] specify the video and audio streams with [0:a][0:v] [03:28] <phingers> it showed me clip 3 video with the first video's audio [03:28] <phingers> then video 2, and clip 1's audio [03:28] <phingers> then video 2, and clip 2 audio [03:29] <c_14> ohlol, try what I said with the seperately, that might fix it i hope. [03:29] <phingers> http://pastie.org/9408257 [03:31] <c_14> And that isn't working? [03:31] <c_14> As in messing up the audio/video order? [03:31] <phingers> never even see clip 3 video [03:32] <phingers> audio is coming out 2,1,3 [03:32] <phingers> video is coming out 3,2----- [03:34] <phingers> I give up for now. ahhhh! I appreciate your help c_14 [03:34] <phingers> catch you later all. [03:34] <c_14> np, still don't know why that wouldn't work though [03:38] <c_14> The one thing that it might be is if the audio and video lengths of the three input videos don't match up that it might concat a bit weirdly, you could see if that might be the issue by using the apad filter in the commands you're using to generate those 3 videos to pad the audio to the length of the corresponding video track. [08:56] <the_s3m> Hey guys, I would like to use ffmpeg on Windows Phone 8 but I couldn't find any good solution to make it work on Windows Phone, so I would like some developer feedback from someone who knows what needs to be done in order to run ffmpeg on windows phone and how much work would it take to make this work. [09:50] <JEEBsv> the_s3m: VLC team was doing work regarding it; and FFmpeg now builds with visual c 2013's compiler [09:51] <JEEBsv> so you could see how #videolan builds its VLC for WinRT [09:53] <the_s3m> thanks for info [09:53] <the_s3m> so vlc already has the working arm build? [09:54] <the_s3m> or is it still in development? [10:53] <Mavrik> yes, VLC has a working ARM build for awhile now [11:26] <cdhris> Hello, is there any way to automatically include the filters such as -bsf h264_mp4toannexb if needed without importing it into the command? [11:28] <cdhris> c_14: do you know that? :P [11:34] <Mavrik> no. [11:35] <Mavrik> but that exact filter won't do anything if it cannot [11:35] <cdhris> i think when i include it in the streams i shouldn't [11:35] <cdhris> it doesnt work [11:38] <cdhris> Yes, confirmed, It doesn' [11:38] <cdhris> it doesn't work when it don't need it [12:22] <cdhris> "Stream #0:2[0x102]: Unknown: none ([6][0][0][0] / 0x0006)" [12:23] <cdhris> Is there any way to stip all the unknown codecs? [12:23] <cdhris> cause i have problem with this, if i stream has this one and i use -map 0 it will try to include this as well [12:23] <cdhris> and it will fail [12:23] <cdhris> strip* [12:23] <sfan5> -map 0:v -map 0:a -map 0:s [12:23] <sfan5> that will only select all known video, audio or sub streams [12:25] <cdhris> Can i ask a noobish question? [12:25] <cdhris> what data codec exactly is [12:26] <sfan5> that questions does not make any sense to me, can you rephrase it? [12:27] <cdhris> I mean in some rtmp streams [12:27] <cdhris> there is a Data codec as well that you can copy with -dcodec copy [12:27] <cdhris> what is the data codec exactly [12:27] <sfan5> hm [12:27] <sfan5> I didn't even know there is a -dcodec option [12:28] <sfan5> I'm probably the wrong person to ask then [12:31] <cdhris> The command you gave me does work [12:31] <cdhris> the only issue [12:32] <cdhris> is that if there is a stream that has only audio /video for example and if i write -map 0:s it will fail saying that there is n subtitles to map. My point is to have a static command [12:32] <cdhris> that can work on all streams [12:32] <cdhris> and strip the unknown codecs [13:07] <Diogo> ./ffmpeg -i rtmp://IP/live/STREAMA -i rtmp://IP/live/STREAMB -filter_complex '[0:0] [0:1] ] [1:2] [2:0] [2:1] [2:2] concat=n=3:v=1:a=2 [v] [a1] [a2]' -map '[v]' -map '[a1]' -map '[a2]' -f flv rtmp://STREAMFINAL [13:08] <Diogo> i want to to a fallback system if the first stream failed put the second stream without breaks the streamfinal? [13:08] <Diogo> this is possible? [14:01] <Giepher> hello which version ffmpeg support -filter_complex? I rum it on ffmpeg 0.7 display ffmpeg unrecognized option 'filter_complex' [14:05] <spaam> 0.7 is old :S [14:06] <cdhris> Is there any way to strip ONLY the unknown Codecs from Input Stream? [14:07] <cdhris> Like this => Stream #0:2[0x102]: Unknown: none ([6][0][0][0] / 0x0006)" [14:36] <cdhris> Hello again [14:37] <cdhris> [mpegts @ 0x2c5c3e0] DTS 4130328530 < 4130663717 out of order [mpegts @ 0x2c5c3e0] DTS 4130751384 < 4131134930 out of order [mpegts @ 0x2c5c3e0] PES packet size mismatch [mpegts @ 0x2d79fa0] AAC bitstream not in ADTS format and extradata missing av_interleaved_write_frame(): Invalid data found when processing input [14:37] <cdhris> what does this means :( [15:37] <mustaf> Hi, on the download page of ffmpeg, when I choose Linux, there is section for static builds, "Linux Static Builds" both links lead to the same location. [15:38] <mustaf> On the old download page there where two. [15:41] <c_14> Yeah, that's been reported on the ml. Should be fixed soon". [15:42] <mustaf> OK [15:43] <c_14> This is the link if you're looking for it: http://johnvansickle.com/ffmpeg/ [15:44] <c_14> If you see anything else off with the new website don't hesitate to report it. [15:48] <mustaf> OK, I'll, nice website by the way. [16:15] <cdhris> c_14: my friend [16:15] <cdhris> do you know what this error means [16:15] <cdhris> AAC bitstream not in ADTS format and extradata missing av_interleaved_write_frame(): Invalid data found when processing input [16:17] <cdhris> http://pastebin.com/SaRJ85jH [16:18] <cdhris> btw +bitexact etc same if i dont write them [16:18] <cdhris> Even the most simpkle command like this: ffmpeg -y -hide_banner -i "http://localhost:5211" -c copy -map 0 -f mpegts /tmp/test.ts [16:18] <cdhris> fails and shows me the error above [16:23] <cdhris> if i transcoded it [16:23] <cdhris> it works but i want to copy the audio not transcode it [16:24] <c_14> Try adding -flags +global_header [16:24] <c_14> If that doesn't work there's an open bug report that has a similar issue: https://trac.ffmpeg.org/ticket/279 [16:26] <cdhris> it doesnt work [16:26] <cdhris> but that report is 3 years ago [16:27] <cdhris> and i am using ffmpeg version 2.3 with all the latest libs etc [16:27] <cdhris> tried with ffmpeg 2.2.4 same [16:28] <c_14> Yeah, but it's never been marked as closed, maybe nobody got around to it. [16:28] <c_14> Do you get the same issue if you mux to say matroska? [16:29] <cdhris> what is the command [16:29] <c_14> ffmpeg -i file -c copy -map 0 -f matroska /tmp/test.mkv [16:30] <cdhris> i think it works with that [16:30] <cdhris> so its bug with ts [17:13] <Katharsis> hi guys [17:14] <Katharsis> my client CS6 adobe doesn't recognize .mp4 file which was recorded on Debian 7.5 [17:14] <Katharsis> i tried a lot of combination, but it faild [17:19] <c_14> Can you ffprobe the mp4? [17:20] <Katharsis> c_14: i tried but it returns empty file [17:21] <c_14> Is the file actually correct? What's the file size, can you play it? [17:21] <Katharsis> I can play it with VLC on linux without any problems [17:21] <Katharsis> it has ~5mb [17:21] <Fjorgynn> i see [17:22] <c_14> And ffprobe says empty file? [17:22] <Katharsis> yeap [17:22] <c_14> What does something like mediainfo say? [17:22] <Katharsis> i used it: ffprobe -v quiet -print_format json -show_format -show_streams test.mp4 > test.json [17:22] <c_14> What version of FFmpeg do you have? [17:22] <Katharsis> the newest form debian 7.5 repository [17:23] <c_14> That'll be libav, see #libav for support or download a static build/compile from source. [17:24] <Katharsis> what's about with this libav? [17:24] <Katharsis> should i paste you the results of ffmpeg -i test.mp4 ? [17:24] <c_14> You can put in on a pastebin, but I'm pretty sure you're not using FFmpeg, but Libav. [17:25] <Katharsis> http://pastebin.com/S4jLNKQ3 [17:26] <c_14> According to that partial output, the file looks fine. Can you play it with ffplay? [17:27] <Katharsis> yes I can [17:27] <c_14> Wait, so ffprobe file says empty file, but ffmpeg -i file gives you output? [17:27] <Katharsis> yeap [17:27] <Katharsis> ;) [17:27] <c_14> Can you pastebin the full output of ffprobe file? [17:28] <Katharsis> there is no output as you can see [17:28] <Katharsis> i used for it this prompt: [17:28] <Katharsis> ffprobe -v quiet -print_format json -show_format -show_streams test.mp4 > test.json [17:28] <Katharsis> test.json is empty [17:29] <c_14> ffrobe -v quiet won't output anything unless it throws errors [17:30] <Katharsis> ffprobe test_25ftp.mp4 without args works [17:30] <Katharsis> http://pastebin.com/8cScNyKr [17:31] <c_14> try ffprobe -hide_banner -print_format json -show_format -show_streams test.mp4 &> test.json [17:31] <c_14> that should give you your json. [17:31] <Katharsis> yes, it has [17:31] <c_14> get rid of the & and it should be valid json. [17:31] <Katharsis> but the source has "Unrecognized option 'hide_banner' Failed to set value '-print_format' for option 'hide_banner'" [17:33] <c_14> That's probably because you're using Libav. Pastebin the full output of `ffmpeg' [17:34] <Katharsis> on this laptop i has diffrent linux so my result will not be the same [17:34] <c_14> But coming back to your original issue. The file looks correct, and if ffplay can play it the problem seems to be on CS6es side. [17:35] <c_14> Especially since both vlc and ffmpeg can play it, the party that can't is usually at fault. [18:33] <jnvsor> Anyone on here knowledgeable about audio desync? [18:47] <Insayne> hello, does ffmpeg support EPG data reading from RTSP streams ? [19:34] <Fjorgynn> So I found the problem [20:01] <Baluse> Hello [20:01] <Baluse> Does anyone know where a-law codec is in FFmpeg library? [20:05] <jnvsor> ffmpeg -codecs | grep alaw [20:05] <jnvsor> I you have D you can decode, if you have E you can encode [20:05] <jnvsor> If not your ffmpeg is compiled without the codec support [20:08] <Baluse> i got the source [20:09] <Baluse> I know that it is supported I am looking for sauce/source [20:14] <c_14> Presumably in libavcodec/pcm.c [20:14] <Baluse> yep found it:) [20:14] <Baluse> it seems nice collection of codecs, there are a hell lot of them [20:17] <c_14> Well, there are a whole lot of variations on pcm. [20:21] <Baluse> yep adpcm is worse [20:22] <Baluse> However opus etc are using external lib [20:22] <Baluse> But what would be the difference if it was embeded ? [20:23] <c_14> hmm? [20:25] <Baluse> https://www.ffmpeg.org/general.html#Audio-Codecs [20:26] <Baluse> here is says E means that support is provided through an external library. [20:30] <jnvsor> How do I record from pulseaudio without audio desync? [20:30] <c_14> ah yes. The main difference between internal codecs and external libraries is that with the internal codecs you don't have to worry about licensing issues or making sure everything works across version updates and the users don't have to install the library to use the codec. [21:43] <toso> Hi there. I'm trying to record an analogic video+audio from a pci card. I'm having trouble with the audio. I can use alsa and record the audio with arecord.. but the same does not apply with ffmpeg -f alsa -i ... [21:43] <toso> This is a simillar problem from ml: https://lists.ffmpeg.org/pipermail/ffmpeg-user/2013-March/014217.html [21:43] <Fjorgynn> i see [21:45] <toso> sacarasc: sorry. http://pastebin.com/H3ejanbc [21:49] <toso> with the ffmpeg output: http://pastebin.com/KkBJtFuZ -- thanks for any help [21:50] <c_14> Try adding -map 1 [21:52] <toso> c_14: man... thank you so much. [21:54] <c_14> np [22:39] <ac_slater_> hey all, is it possible to have an h264 stream in an mpegts file/container/stream? [22:40] <c_14> yes [22:40] <sacarasc> That's how a lot of TV is transmitted. [22:40] <ac_slater_> that's what I thought [22:41] <ac_slater_> also, I just posted to the libav-user mailing list, but I might as well ask on here. I'd interesting in muxing a custom binary datastream into a mpegts file/container. Does ffmpeg support this? [22:41] <ac_slater_> (not the application itself, but libav*) [22:42] <ac_slater_> s/I'd/I'm [22:42] <ac_slater_> s/interesting/interested [22:42] <ac_slater_> (talking on the phone and typing isnt easy sometimes) [22:47] <ac_slater_> my idea is to mux together Video+data into a mpegts context, and deliver it via rtp (and to a container) [22:47] <ac_slater_> the mpegts files in the source tree are a little large and I havent figured out how they all connect yet. Any advice is appreciated. [00:00] --- Tue Jul 22 2014

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