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August 2016

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ffmpeg-devel.log.20160810
by burek 11 Aug '16

11 Aug '16

[02:55:45 CEST] <jamrial_> is trac down for anyone else? [03:00:25 CEST] <michaelni> jamrial_, seems working again, dont ask me what the issue was, i dont know [03:07:12 CEST] <jamrial_> michaelni: yeah, works now, thanks [07:58:20 CEST] <kd_dev> Hello!, I had 2 questions: [07:58:39 CEST] <kd_dev> 1. Is some kind og demosaicing algorithm implemented as filters in ffmpeg? [07:58:54 CEST] <kd_dev> 2. If not, any ideas on how to proceed about doing so? [07:59:07 CEST] <kd_dev> found some discussion here: https://ffmpeg.org/pipermail/ffmpeg-devel/2010-July/090493.html [09:49:12 CEST] <jsebechlebsky> Hi, I'm trying to make a mingw build of FFmpeg to test failing fifo muxer test, but I can't make configure detect pthreads... Does someone have experiece with mingw+pthreads build? [10:03:49 CEST] <cone-311> ffmpeg 03Paul B Mahol 07master:4506f911858b: avfilter/vf_remap: >8 bit support [10:19:59 CEST] <Timothy_Gu> jsebechlebsky: mingw or mingw-w64? [10:20:32 CEST] <Timothy_Gu> the former is very old (i686-pc-mingw32). don't use it [10:20:52 CEST] <Timothy_Gu> mingw-w64 bundles pthreads methinks [10:20:58 CEST] <jsebechlebsky> it's mingw-w64, but I've just figured it out... [10:21:17 CEST] <jsebechlebsky> --disable-w32threads [10:22:22 CEST] <jsebechlebsky> enabled pthreads... I've originally supposed pthreads are preferred [11:38:15 CEST] <cone-311> ffmpeg 03Paul B Mahol 07master:80262d8c86e9: avfilter/avf_showwaves: add sqrt and cbrt amplitude scalers [12:58:05 CEST] <robUx4> nevcairiel: DXVA2 doesn't support FF_PROFILE_H264_HIGH_10 ? [12:58:45 CEST] <robUx4> H.264 variable-length decoder, film grain technology has P010 output so I suppose it should [12:59:02 CEST] <nevcairiel> there is no h264 decoder with 10-bit support [13:00:05 CEST] <robUx4> you mean hardware decoder [13:00:10 CEST] <robUx4> +? [13:00:17 CEST] <nevcairiel> of course, thats what we're talking about isnt it? :) [13:00:24 CEST] <robUx4> I have some hi10 samples that decode [13:00:25 CEST] <robUx4> yes [13:00:33 CEST] <robUx4> why the P010 output then ? [13:00:54 CEST] <nevcairiel> never seen a device that offers that [13:01:54 CEST] <robUx4> the Xbox One does [13:01:58 CEST] <nevcairiel> which devices even offer FGT? that feature never really took off [13:02:12 CEST] <nevcairiel> maybe the 10-bit output is related to FGT somehow [13:02:38 CEST] <robUx4> possibly [13:04:49 CEST] <robUx4> here's a log from the Xbox (not the S) http://pastebin.com/aMQ1GTCX [13:06:13 CEST] <nevcairiel> or AMD gave every device P010 output for some reason, who knows [13:07:44 CEST] <nevcairiel> in any case the specification for those devices clearly states the profiles which are supported, and main10 is not among them [13:08:01 CEST] <robUx4> is there an easy to force the HW decoding with hi10 sources ? [13:08:08 CEST] <robUx4> or it's just too different ? [13:10:07 CEST] <nevcairiel> might be able to trick avcodec into offering that [13:43:46 CEST] <Compn> nevcairiel : i thought there was some nvidia cards that did 10bit h264 [13:43:50 CEST] <Compn> but probably i am wrong [13:46:01 CEST] <Compn> oh no, guess i'm wrong [13:46:07 CEST] <Compn> bunch of software decoding going on haha [13:46:59 CEST] <durandal_1707> you always wrong [13:52:04 CEST] <BtbN> Pascal should support that though? [13:52:22 CEST] <nevcairiel> what makes you thin that, h264 support hasnt changed for years [13:53:37 CEST] <BtbN> So it supports 10bit HEVC decoding, but not h264? oO [13:54:06 CEST] <nevcairiel> indeed [13:54:27 CEST] <nevcairiel> codecs are quite different, assuming one means the other is not how it works .D [13:54:56 CEST] <BtbN> would have expected that they share a lot of the decoding hardware, so if they add 10bit, it would work for both. [14:03:13 CEST] <Compn> probably the 8bit h264 chips are cheap [14:03:20 CEST] <Compn> while the 10bit h264 chips cost another nickel [14:04:18 CEST] <nevcairiel> its part of the main GPU chip so its integrated into there somewhere, i dont know if they source those logic circuits from somewhere else or develop them tho [14:05:10 CEST] <nevcairiel> but there is really no point for them in264 10-bit support, there is literally zero commercial content [14:09:00 CEST] <robUx4> AVCHD is not 10 bits ? [14:09:30 CEST] <nevcairiel> isnt that blu-ray compatible video [14:09:45 CEST] <robUx4> ah no AVCHD is 8 bits [14:23:25 CEST] <durandal_1707> lot of love in ffmpeg-user lately [15:35:31 CEST] <robUx4> nevcairiel: for the record hi10p decoding doesn't work, neither in P010 nor NV12 [18:32:15 CEST] <omerjerk> Hi [18:32:25 CEST] <omerjerk> I've added alsenc here - https://github.com/omerjerk/FFmpeg/blob/alsenc_first_patch/libavformat/alse… [18:32:36 CEST] <omerjerk> Still I'm not able to encode to out.als [18:32:39 CEST] <omerjerk> for e.g. [18:32:57 CEST] <omerjerk> ffmpeg -i in.wav -c:a als out.als gives error [18:33:13 CEST] <omerjerk> Error = Could not write header for output file #0 (incorrect codec parameters ?): Operation not permitted [18:33:38 CEST] <omerjerk> Is it even possible to output into .als file or not? [18:39:10 CEST] <durandal_1707> omerjerk: only if you wrote als muxer [18:40:38 CEST] <durandal_1707> write into mp4 [18:41:48 CEST] <cone-931> ffmpeg 03Burt P 07master:16ea0bca1473: avfiltergraph.c: restore disabling of auto conversions [18:43:40 CEST] <durandal_1707> so we gonna remove hacks which blocks codecpar transformation? [18:45:30 CEST] <jamrial_> can that be done without negatively affecting output files? [18:46:00 CEST] <durandal_1707> hacks are for invalid files anyway [18:47:30 CEST] <jamrial_> poke michaelni, afaik he added most of those hacks and he probably has example use cases or files [18:51:00 CEST] <durandal_1707> michaelni: see above? [18:57:30 CEST] <michaelni> anything i should test ? but i dont have a too good feeling with ffmpeg->codecpar to be "just working" [19:16:54 CEST] <jamrial_> michaelni: in ffmpeg.c transcode_init(), the stream copy section does some crazy calculations only for avi/mp4 output using avctx->ticks_per_frame and avctx->time_base that with codecpar will not be possible [19:17:41 CEST] <jamrial_> not sure how you could test aside from dropping that chunk when converting ffmpeg to codecpar and seeing how it ends up muxing that kind of file [19:28:15 CEST] <michaelni> speaking about ticks_per_frame, theres also [FFmpeg-devel] [PATCH 2/2] doc/examples/muxing: Fix packet duration [21:02:48 CEST] <durandal_170> omerjerk: still around? [21:02:58 CEST] <omerjerk> yes. [21:03:07 CEST] <omerjerk> There's some bug in the ALS muxer code. [21:03:13 CEST] <omerjerk> I'm trying to fix that. [21:03:35 CEST] <durandal_170> what about into mp4 muxer? [21:03:44 CEST] <omerjerk> that works perfect. [21:03:56 CEST] <durandal_170> nice [21:13:59 CEST] <BtbN> uhm, that sounds kinda expected. [21:14:44 CEST] <durandal_170> but it works with mencoder I think [21:18:43 CEST] <iive> i hope these are for different avi files [21:19:06 CEST] <durandal_170> :) job for Carl [00:00:00 CEST] --- Thu Aug 11 2016

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ffmpeg.log.20160810
by burek 11 Aug '16

11 Aug '16

[00:17:29 CEST] <bencc> kepstin: ok, so no need for gpu [00:18:18 CEST] <bencc> any specific features in a cpu? clock-speed, number of cores...? [00:18:37 CEST] <bencc> for example here I see a large number of servers and has no idea what to pick [00:18:38 CEST] <bencc> https://securedservers.com/bare-metal-dedicated-servers [00:19:20 CEST] <kepstin> More is better? You'll really have to just try and see to find what specs you'll need to meet your specific requirements [00:24:17 CEST] <DHE> usually, but high-core count CPUs run slower. so you can't max out both at the same time. [00:27:25 CEST] <bencc> kepstin: I don't have specific requirements :) [00:27:41 CEST] <bencc> just transcoding vp8 to h264 with 720p and standard quality [00:28:25 CEST] <bencc> is it better to have one machine with many CPUs or several smaller machines? [00:38:44 CEST] <mrelcee> i have a 2012 core i7 desktop processor in a machine. quad core.. it tears up the frames and asks for seconds... running freebsd.. [00:40:04 CEST] <mrelcee> i imagine a newer core with ddr4 would be impressive [00:40:45 CEST] <mrelcee> but i paid almost nothing for it.. it's a hand me down from someone who upgraded [00:41:20 CEST] <DHE> I have an e5-2698 v4. holy hell this thing is a beast [01:19:37 CEST] <cas_> Hi, I am using the segment muxer with segment_atclocktime and strftime for recoring an audio stream but the filenames are not exact on the segment_time duration (I use 5 seconds and get e.g. 00.ts, 09.ts, 15.ts, 19.ts...) [01:20:18 CEST] <cas_> is there a way how to stick to the offset? [03:05:05 CEST] <jookiyaya> what is beter 720p or 1080i [05:18:12 CEST] <Spring> hey durandal_1707, remember that ffplay bug we talked about? Turns out if you omit -an and use atrim it prevents the slowndown. [05:21:49 CEST] <Spring> which is strange considering -an should disable audio altogether, but for whatever reason sync'ing both trims with audio enabled fixes it. [05:22:20 CEST] <Spring> tried adding -an + atrim but the slowdown occurs again [05:22:54 CEST] <Spring> so I may look into a volume adjustment filter if there's one to simulate mute [05:24:12 CEST] <Spring> yup, volume=0 works fine, nice! [06:50:42 CEST] <antiPoP> how can I add a watermark proportional to the video dimensions (which are variable)? [08:19:33 CEST] <wallbroken> how to set preset in ffmpeg? [09:14:38 CEST] <Spring> wow, I'm stunned at how low the bitrate can go with a high CRF yet still be far higher quality than higher max bitrates [09:16:01 CEST] <Spring> most examples I've seen of vpx set some max bitrate on any given crf but after numerous tests going as high as CRF 35 on a mostly still produces superb results [09:16:18 CEST] <Spring> *still scenes. [12:34:35 CEST] <jookiyaya> would i see improvement in upgrading from 1333mhz ddr3 to 2133mhz ddr3 [12:36:42 CEST] <DHE> if you're bottlenecked on memory, yes. tools like perf can help measure that [12:38:13 CEST] <jookiyaya> perf? [12:38:52 CEST] <DHE> it's a linux tool that ships with the kernel. tons of benchmarking features using x86-specific features [12:39:54 CEST] <jookiyaya> i am using windows [12:42:54 CEST] <Spring> can someone take a squiz at this output and see if there's something odd, as the output size and bitrate is identical whether I set the max bitrate to 1M or 10M. http://pastebin.com/06LY91Wz [12:43:32 CEST] <jkqxz> jookiyaya: Not many processors run DDR3 above 1600MT/s (some 1866MT/s) by default. You would need to have an overclockable chipset to see any value from 2133MT/s DDR3 over slower types. [12:44:11 CEST] <Spring> there aren't any errors but the bitrate doesn't make sense [12:44:31 CEST] <jookiyaya> jkqxz i see; do you know anything about this bottleneck app [12:46:13 CEST] <jookiyaya> dhe wouldn't that app only show bottleneck if ram is running out [12:47:14 CEST] <Spring> I should mention the bitrate in the output is 20-21Mbps [12:47:14 CEST] <jkqxz> Spring: Maybe you're permanently hitting qmax? [12:48:34 CEST] <Spring> jkqxz, could you explain a little? [12:50:13 CEST] <jookiyaya> jkqxz even if i have 2133MT ram, if i use 2133MT setting, it's still called overclocking? [12:51:10 CEST] <jkqxz> Spring: qmax sets an upper bound on the quantisation level - if you set it low with a low bitrate target then you may just end up encoding all frames at the bound as the target can't be met at all. Try increasing it. [12:52:14 CEST] <jkqxz> jookiyaya: Yes. (You would be overclocking the memory controller, not the memory itself.) [12:52:29 CEST] <jookiyaya> jkqxz i see [12:55:17 CEST] <Spring> jkqxz, hmm. Didn't realize that would occur, expected the max bitrate cap to take precedence. Raised the qmax and am within the desired bitrate, though quality obviously took a hit. [12:58:45 CEST] <jkqxz> qmax is probably something you should only set in weird cases (you absolutely require some minimum quality level and are prepared to violate the bitrate constraints to achieve that). [14:28:34 CEST] <pagios> hello [14:28:36 CEST] <pagios> <pagios> File is missing 'moov' atom can i reencoder this mp4? [14:28:36 CEST] <pagios> <pagios> hello [14:28:36 CEST] <pagios> * georgios has quit (Ping timeout: 240 seconds) [14:28:36 CEST] <pagios> <pagios> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1983b20] Could not find codec parameters (Video: h264, 480x640) [14:28:36 CEST] <pagios> <pagios> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x1983b20] Could not find codec parameters (Audio: aac, 44100 Hz, stereo, s16) [14:28:38 CEST] <pagios> <pagios> how can i resolve this [14:29:56 CEST] <durandal_1707> was already asked before, see irc logs [14:30:25 CEST] <pagios> wherE?> [14:30:53 CEST] <durandal_1707> ffmpeg web page [14:35:40 CEST] <Spring> qmax really has the final say, doesn't it. Tested a CRF 35 with -b:v 0 (to enable CRF in vp9) and qmax 35. Very high quality output at 15.3MB. [14:36:06 CEST] <Spring> Next I tested CRF 33 with -b:v 20M, qmax 40. Worse quality, 27.9MB. [14:36:29 CEST] <durandal_1707> pagios: untrunc something [14:39:47 CEST] <Spring> wish there were a more consistent quality across the same settings. Since I get better quality on other clips using the latter settings. [14:39:59 CEST] <kepstin> Spring: yep, it pretty much works like: CRF says "hmm, this only needs around qp 37 to hit this quality" then qmax comes along and goes "oh no you don't, qp 35 it is!" [14:41:00 CEST] <kepstin> Spring: for most cases, you should be adjusting only crf to control quality and not setting qmax/qmin at all... [14:44:11 CEST] <pagios> durandal_1707, cant find [14:44:17 CEST] <pagios> can that moov be recovered? [14:45:07 CEST] <kepstin> pagios: in general no, but iirc there's a couple services around that try to guess the encoder used on the original video and make up a moov that might work. [14:45:31 CEST] <Spring> last time I encountered a bad moov atom error I used Mp4box which fixed it [14:45:46 CEST] <pagios> worked on linux? [14:46:05 CEST] <Spring> not sure if Mp4box is on nix [14:46:09 CEST] <bencoh> it is [14:46:10 CEST] <pagios> i am running headless [14:46:36 CEST] <bencoh> https://gpac.wp.mines-telecom.fr/mp4box/ [14:46:55 CEST] <Spring> in my case the command I found was: MP4Box -add fixed.mp4 -isma sample.mp4 [14:47:14 CEST] <Spring> however yours may be a different error [14:48:00 CEST] <pagios> its weird coz it was a recorded livestream [14:48:23 CEST] <soulshock> which SDI cards do ffmpeg support? any other than Blackmagic DecLink? [14:49:21 CEST] <kepstin> pagios: recorded how? if you were converting it to mp4 locally, maybe your encoder was killed before it had a chance to write the moov at the end of the file? [14:50:06 CEST] <BtbN> ffmpeg supports everything that has a propper driver. [14:50:59 CEST] <kepstin> if it works with v4l (linux) or dshow (windows), ffmpeg should be able to use it. decklink is kind of a special case because they use their own special apis. [14:52:17 CEST] <BtbN> Doesn't Mageware have SDI capture cards? [14:52:29 CEST] <BtbN> *Magewell [14:52:48 CEST] <BtbN> Yeah, they do: http://www.magewell.com/usb-capture-sdi [14:54:18 CEST] <kepstin> heh, that's just a usb video class (uvc) device, it doesn't even need a special driver [14:54:43 CEST] <bencoh> the question is which timestamp does it returns ... [14:55:03 CEST] <bencoh> -s [14:56:09 CEST] <BtbN> kepstin, are you sure? They have a special kernel module that advertises v4l2+alsa support. [14:56:48 CEST] <kepstin> BtbN: i suspect they have a special driver that allows controlling some of the advanced features, but they also list uvc support [14:56:58 CEST] <Spring> kepstin, I find the problem is that using CRF is unpredictable with filesizes. Looking at the settings above I find it strange how lowering the CRF and raising the qmax yet keeping the max bitrate made the filesize almost twice as large and worse quality. [14:57:24 CEST] <bencoh> Spring: pretty simple though [14:57:27 CEST] <kepstin> Spring: crf is intentionally unpredictable with filesizes. [14:57:32 CEST] <BtbN> kepstin, that would be quite interesting. Does UVC support audio? [14:57:49 CEST] <kepstin> BtbN: most uvc webcams i've seen have built-in mics, so I assume so [14:58:02 CEST] <BtbN> well, but they use a seperate driver for that? [14:58:32 CEST] <furq> Spring: is that actually a problem [14:58:33 CEST] <Spring> basically I get settings right on one clip only find they need tweaking in the next, yet they both have such little movement. [14:58:38 CEST] <kepstin> Spring: if you want predictable filesize, use a 2-pass bitrate encode. if you want constant quality, and have the encoder use as many bits as needed for that, use crf without max bitrate or qmax. [14:58:40 CEST] <BtbN> Also, http://www.magewell.com/usb-capture-sdi/tech-specs says it's v4l2, not uvc [14:59:00 CEST] <BtbN> hm, but i guess the uvc driver exposes it via v4l2 [14:59:08 CEST] <furq> the assumption behind crf is that filesize doesn't usually matter [14:59:12 CEST] <bencoh> BtbN: yup [14:59:14 CEST] <kepstin> BtbN: 'v4l2' is the linux video api. 'uvc' is the usb standard for how to transmit video over usb. [14:59:23 CEST] <BtbN> That's actually quite nice then. [14:59:25 CEST] <furq> and that it's generally a bad way of specifying quality, which is what was standard back in the bad old days of xvid [14:59:30 CEST] <BtbN> The HDMI USB card works in the same way. [14:59:30 CEST] <kepstin> BtbN: the linux kernel comes with a v4l2 driver that supports uvc devices [15:00:42 CEST] <Spring> reasons for needing a certain filesize: file host limits, reasons for needing quality: people who will be viewing the clips critically. Trying to find some sweet spot basically. [15:00:50 CEST] <bencoh> Spring: you lowered crf and increased qmax, which basically means that "complex" scenes will take more space, while "simple" ones will be "optimized out" (ie encoded with less bits) [15:01:05 CEST] <kepstin> Spring: if you want the max quality for a given filesize, do a 2-pass vbr encode [15:01:08 CEST] <Spring> but it was worse quality [15:01:18 CEST] <bencoh> Spring: perceptually [15:01:37 CEST] <Spring> all my encodes are two pass fwiw [15:01:42 CEST] <Spring> in vpx [15:02:10 CEST] <kepstin> sure, but if you want max quality for a given filesize, don't use crf. just let the vbr ratecontrol handle it [15:02:35 CEST] <kepstin> if you're encoding to a max filesize, using crf will only ever reduce quality. [15:03:24 CEST] <kepstin> with libvpx using crf and maxbitrate, the purpose of crf is to set a "max quality" and then make the file smaller if using the full bitrate would cause the quality to be too high [15:03:44 CEST] <Spring> the thing is that I've tried setting for eg: a 10M cap and it wasn't the best quality for the filesize. [15:04:07 CEST] <Spring> if it was I'd be content [15:05:00 CEST] <kepstin> well, if doing a 2-pass encode with libvpx with '-b:v 10M' and no qrf or qp related options doesn't provide best quality for the given file size, then that's a bug in libvpx [15:05:40 CEST] <kepstin> i honestly wouldn't be surprised, libvpx is mostly developed for realtime streaming by google nowadays, so they probably haven't put much attention into that :/ [15:08:23 CEST] <kepstin> btw, how are you determining that "it wasn't the best quality"? measuring with ssim or something, or just looking at the video? [15:09:21 CEST] <Spring> taking a PNG frame capture using ffmpeg at the exact timestamp of each encode, then brightening them all in Photoshop using overlaid adjustment curves to bring out the details. [15:09:37 CEST] <kepstin> well, that's about the worst possible way to do that :/ [15:10:00 CEST] <Spring> what's the problem? [15:10:08 CEST] <kepstin> the psy optimizations in the codec explicitly try not to save details that aren't visible to the human eye [15:10:49 CEST] <Spring> the artifacts are visible though, it's quite noticeable, the curves just make it easier to compare. [15:10:52 CEST] <kepstin> so brightening them and using adjustment curves is rather counter to the goals of the codec; you're just comparing meaningless background noise patterns [15:11:33 CEST] <Spring> I should also mention again these are all nearly static scenes, so details will be stared at [15:11:45 CEST] <Spring> fixed camera, little movement [15:11:45 CEST] <kepstin> and comparing sill frames by eye doesn't work well, since the psy optimizations usually make use of the eye's motion sensitivity, and comparing particular still frames will have variations depending on keyframe positioning etc [15:18:14 CEST] <kepstin> If you want to compare video frames with brightness or contrast adjustments, do them before encoding :-) [15:23:33 CEST] <Spring> like for example from one clip: -b:v 10M = 43.5 MB (no crf or qmin/max) vs -crf 35 -b:v 0 qmax 35 = 15.7MB. The latter is marginally better quality than the former. [16:16:56 CEST] <kepstin> Spring: in one scene perhaps. The vbr encode tries to distribute bitrate over the entire file so that it looks roughly constant quality, while the qmax setting will cause some scenes to be better quality at the expense of others. [16:17:41 CEST] <kepstin> but yeah, that's an awfully big difference in file size. I suspect there's some issues in libvpx's 2-pass ratecontrol :/ [16:18:02 CEST] <kepstin> assuming the quality on the second file is comparable to the first [16:22:19 CEST] <Spring> yeah, the quality on the second is exceptional. Problem is it's inconsistent, so [16:24:46 CEST] <Spring> one clip will be 15.7MB another 30MB. And on the 30MB clip a different setting works better. So it's kind of frustrating finding some middle ground. [16:25:27 CEST] <Spring> especially where foliage is involved, since compression is highly noticeable [17:04:29 CEST] <moparisthebest> I need to capture a stream to the filesystem and at the same time stream it via http to possibly multiple clients, it looks like ffserver can do the streaming, but it writes an .ffm to disk too? [17:05:09 CEST] <moparisthebest> what I want to avoid is writing the .ffm to disk and also curl'ing or whatever the stream to the same disk, is there any way to do that? [17:05:17 CEST] <moparisthebest> or maybe I'm using the wrong tools or something? not sure [17:09:51 CEST] <kepstin> Spring: the quality on the second is inconsistent *because* you are using qmax. If instead of using qmax you simply decreased crf, the quality should be more consistent (and of course the file larger, since consistently good quality needs more bits) [17:47:06 CEST] <DHE> moparisthebest: HTTP-based serving is usually done with something like MPEG-DASH or HLS for live streams. ffserver is on the chopping block and should be avoided if at all possible [17:50:29 CEST] <moparisthebest> DHE: ah thanks, yea I plan to keep this system for years so I don't want to base it on something soon-to-be-killed :) [17:51:08 CEST] <DHE> if at all possible, I think you're best off using a standard HTTP server (apache, nginx, whatever) and have ffmpeg operate in the serving directory running the protocol of your choosing. [17:52:33 CEST] <moparisthebest> so what I've got is an computer with a camera I want to do all the encoding on and a wifi link to the main server with storage disks [17:52:53 CEST] <moparisthebest> since the wifi link is slow, I don't want to serve multiple clients over that, I want to serve multiple clients from the main server, but not have it transcode anything... [17:52:57 CEST] <furq> if you want to be able to send the stream to a remote server then use ffmpeg on the client and nginx-rtmp on the server [17:54:00 CEST] <furq> you can set it up to serve hls or dash and save local copies [17:54:57 CEST] <moparisthebest> nginx is what I already use on that server so that sounds like a good option [17:55:15 CEST] <furq> you'll need to recompile it for the rtmp module [17:55:43 CEST] <furq> i prefer to have a separate binary for that [17:55:50 CEST] <moparisthebest> doesn't nginx do dynamic modules now? or recompiling is fine meh [17:55:57 CEST] <furq> maybe [17:56:00 CEST] <furq> it's been a while since i touched it [17:56:11 CEST] <moparisthebest> I haven't tried but I read that someplace [17:58:18 CEST] <DHE> 1.9.x does have dynamic modules, for some value of x. [17:58:38 CEST] <DHE> x=11 [17:58:50 CEST] <furq> the latest commit on the rtmp module is "fixed rtmp as a dynamic module" [17:58:52 CEST] <moparisthebest> hmm but hls/mpeg-dash looks like I'll have to transcode on the server running nginx right? [17:58:53 CEST] <furq> so that looks promising [17:59:05 CEST] <furq> and no, flv to hls/dash is just remuxing [17:59:40 CEST] <DHE> moparisthebest: no, but you'll have to output to hls/mpeg-dash compatible codecs [17:59:52 CEST] <moparisthebest> is h264 one? that's all I can do :/ [17:59:56 CEST] <furq> rtmp only really supports h264 and aac anyway [18:00:01 CEST] <furq> all the other usable codecs are trash [18:00:02 CEST] <moparisthebest> ah good [18:00:30 CEST] <DHE> those are the only codecs officially permitted by HLS as well [18:00:38 CEST] <furq> yeah [18:01:10 CEST] <furq> if you don't need web browser support (without flash) then rtmp itself is the best choice [18:01:32 CEST] <furq> hls and dash aren't exactly great [18:02:13 CEST] <moparisthebest> I want 1) record to filesystem in whatever format 2) view in html5 browsers with no flash, so phones work and stuff [18:02:22 CEST] <furq> yeah you can do that [18:02:33 CEST] <furq> with hls or dash, that is [18:02:37 CEST] <moparisthebest> I'm basically rolling my own security camera system since all the pre-built stuff is crap security-wise [18:03:01 CEST] <moparisthebest> the 'camera' is a raspberry pi 3 hehe, about $100 each with everything needed [18:03:10 CEST] <furq> hls/dash have got pretty high latency [18:03:29 CEST] <furq> i'm not sure how low you can get it with short segments [18:03:59 CEST] <furq> it's at least one and practically usually 2-3 segments of latency with hls, which is normally 10-30 seconds [18:04:09 CEST] <furq> idk how usable it'll be with one-second segments [18:06:00 CEST] <moparisthebest> hmm yea I was hoping for ~4 seconds or so [18:06:05 CEST] <moparisthebest> on the lan anyway [18:08:42 CEST] <furq> even if you serve hls you can still view the stream over rtmp if you need solid latency [18:08:51 CEST] <furq> but yeah that needs flash or a real video player [18:09:21 CEST] <moparisthebest> yea maybe I use a real player on the monitor screen I plan on setting up [18:09:30 CEST] <moparisthebest> and if I'm looking at it on a phone or over the internet, latency is fine [18:15:19 CEST] <Mysoft> does libav have options to skip frames? [18:16:19 CEST] <Mysoft> i think that is more codec specific... but i wonder if anyone knows [18:17:15 CEST] <Mysoft> that when skipping frames on x264 the codec will keep track of interframe blocks [18:17:32 CEST] <Mysoft> so that only the required chain of blocks will be required [18:18:15 CEST] <Mysoft> or if i must do that using some lower level operations? :) [18:21:44 CEST] <bencoh> Mysoft: it does [18:22:22 CEST] <bencoh> but depending on what you're looking for it might not be "enough" for you [18:23:24 CEST] <Mysoft> just wanted to avoid decoding unecessary blocks when skipping frames yeah :) [18:23:26 CEST] <bencoh> when I selecting I frames only (or IDR? dont remember) it happened to us that we didnt receive any keyframes for some streams, because there were no IDR [18:23:49 CEST] <bencoh> (that's something we played with over 2 years ago, so I dont remember every details) [18:23:52 CEST] <bencoh> -s [18:24:47 CEST] <Mysoft> i see... i will be using it to show a video preview of what would be passed to other devices (from my accelerated stream SERVER) [18:25:35 CEST] <bencoh> yeah I suspected that :) [18:26:07 CEST] <bencoh> (here it was to display thumbnails for previews ... like what you can get on *tube players) [18:26:24 CEST] <Mysoft> i see yeah :) [18:26:34 CEST] <Mysoft> i found nice that "PotPlayer" had this option too [18:26:45 CEST] <Mysoft> first time i see a lightweight player on windows with such support [18:26:45 CEST] <Mysoft> hehe [18:27:01 CEST] <bencoh> so yeah depending on which frames you drop you might not get expected result on some streams [18:27:29 CEST] <Mysoft> i'm also not using swscale because even as fallback it's too slow [18:27:38 CEST] <Mysoft> i.e. compared to a jitted sw scaler :) [18:28:58 CEST] <Mysoft> because when you will use a scaler like that (that you always have the same input/output size) (or that it rarely changes) [18:29:11 CEST] <Mysoft> creating dynamic code that will hardcode the math for the scale speed up it considerable [18:29:29 CEST] <Mysoft> altough you have to implement it for each platform [18:29:43 CEST] <Mysoft> so i'm putting some considerable effort to make this as nice as possible :] [18:30:44 CEST] <bencoh> Mysoft: hmm ... doesn't swscaler already have some kind of JIT? [18:35:19 CEST] <bencoh> hmm maybe not, nevermind [18:36:51 CEST] <drv> there is a tiny amount of runtime generated code in swscale (see lumMmxextFilterCode/chrMmxextFilterCode) [18:48:11 CEST] <moparisthebest> can I set the quality value for -c:v h264_omx some way? or does it only support a static framerate ? [18:50:11 CEST] <moparisthebest> or, where would I find that kind of info? ffmpeg --help full didn't really show me anything of help I think [18:53:46 CEST] <furq> -h encoder=h264_omx [18:53:50 CEST] <furq> although that doesn't contain anything useful [18:54:05 CEST] <furq> it looks like the only quality-related param you can change is the bitrate [19:12:46 CEST] <the_gamer> how to convert this? http://www.pasteall.org/73860 it plays in vlv and mpv, how to convert it with ffmpeg? [19:14:08 CEST] <Mysoft> the_gamer you have option to specify INPUT format [19:14:19 CEST] <Mysoft> like the input sample rate... and etc... [19:14:24 CEST] <Mysoft> that are invalid on that .avi header [19:16:22 CEST] <the_gamer> Mysoft, could you give me the syntax to do this? [19:21:31 CEST] <the_gamer> ffmpeg -c:v avrn -an -i PICT0001.AVI -vf scale=640:480 -an katze.mpg [19:21:35 CEST] <the_gamer> still the same [19:21:49 CEST] <the_gamer> giving the codec and telling it to ignore the audio doesn't work [19:22:43 CEST] <the_gamer> would be happy for any hint [19:28:45 CEST] <Mysoft> tried -ar before -i ? [19:29:05 CEST] <Mysoft> or -r:a whatever :P [19:29:57 CEST] <the_gamer> ffmpeg -c:v avrn -ar 22050 -r:v 30 -i PICT0001.AVI -vf scale=640:480 -an katze.mpg [19:30:06 CEST] <the_gamer> still throws invalid sample rate: 0 [19:30:21 CEST] <the_gamer> but i don't even want audio! i gave -an [19:30:33 CEST] <Mysoft> well it has to parses the header [19:30:35 CEST] <Mysoft> if the header is invalid [19:30:40 CEST] <Mysoft> is likely it's garbage [19:30:48 CEST] <Mysoft> so it stops before it goes even worse [19:31:02 CEST] <Mysoft> as if the header is too invalid... it wouldnt be able to get the video either [19:31:09 CEST] <the_gamer> video is playable with mplayer and i want to correct it so that it works everywhere. thought ffmpeg could do it? [19:31:25 CEST] <Mysoft> maybe if you increase the loglevel [19:31:34 CEST] <the_gamer> ffmpeg doesn't seem to get the video but i want the video only [19:31:37 CEST] <Mysoft> it would help.. but i never tried that level of correctness with ffmpeg :) [19:33:59 CEST] <the_gamer> http://www.pasteall.org/73866 [19:34:56 CEST] <the_gamer> doesn't look more helpful to me [19:35:56 CEST] <Mysoft> hum... [19:36:06 CEST] <Mysoft> on the full help... there's the av lib options [19:36:15 CEST] <Mysoft> maybe there is something to deal with such errors [19:38:39 CEST] <Mysoft> maybe... [19:38:50 CEST] <Mysoft> -err_detect ignore_err [19:43:05 CEST] <the_gamer> nope [19:46:33 CEST] <Mysoft> you know that is weird... [19:47:11 CEST] <Mysoft> everytime i see that error on google [19:47:14 CEST] <Mysoft> it's like "Invalid sample rate 0, using input rate 48000" [19:47:48 CEST] <Mysoft> and -ar was suppose to have fixed that [19:48:18 CEST] <Mysoft> the_gamer can i get a sample rate [19:48:20 CEST] <Mysoft> so i can test here? [19:48:30 CEST] <Mysoft> *get a sample video [19:48:32 CEST] <the_gamer> mencoder can do it! [19:48:40 CEST] <the_gamer> i don't get ffmpeg to fix it [19:49:06 CEST] <the_gamer> wait a second, you can [20:27:03 CEST] <eemgr> How would I go about converting a .mp4 or .mov to a .swf with the highest quality? [20:28:06 CEST] <Mysoft> i think .swf is just a container for a flash plugin that contains a vp9 stream? [20:28:19 CEST] <kepstin> well, usually h264 video nowadays [20:28:37 CEST] <furq> it was never vp9 was it [20:28:47 CEST] <Mysoft> .flv is vp9 [20:28:55 CEST] <Mysoft> and is the most common format on .swf built-in stuff [20:28:57 CEST] <Mysoft> as far i know [20:28:59 CEST] <kepstin> but yeah, it's an adobe flash file that optionally contains embedded video and or audio streams [20:29:03 CEST] <kepstin> Mysoft: you mean vp6? [20:29:10 CEST] <Mysoft> hu [20:29:11 CEST] <furq> yeah [20:29:12 CEST] <Mysoft> lol yes :P [20:29:50 CEST] <Mysoft> but with a h264 embedded on the swf... odds is that you can just copy the video as is [20:29:53 CEST] <kepstin> like, I think you have to have some flash code (actionscript?) in a swf to act as a player, it's not just a video format on its own [20:30:03 CEST] <furq> although ffmpeg defaults to spark/flv1 [20:30:15 CEST] <Mysoft> well yes the swf is the player [20:30:16 CEST] <kepstin> but yeah, assuming your mp4/mov is h264 you can probably just use the video as-is [20:30:18 CEST] <Mysoft> with the stream builtin :P [20:30:27 CEST] <eemgr> it converst just fine with your typical ffmpeg -i in.mov out.swf [20:30:31 CEST] <furq> ffmpeg has an swf muxer so hopefully that does the right thing [20:30:44 CEST] <eemgr> but ideally i'd obtain a higher quality, it appears to be really degraded after the conversion [20:30:45 CEST] <furq> eemgr: if it's a mov then you can probably just copy the source streams [20:30:48 CEST] <Mysoft> -acodec copy -vcodec copy [20:30:49 CEST] <Mysoft> :] [20:30:51 CEST] <furq> add -c copy before out.swf [20:31:01 CEST] <kepstin> i wonder if ffmpeg's 'swf' muxer maybe just makes a flv file [20:31:08 CEST] <eemgr> i'll give it a shot, one min [20:32:33 CEST] <eemgr> hmm [20:33:48 CEST] <eemgr> "Output file #0 does not contain any stream" from [20:33:49 CEST] <eemgr> ffmpeg Stopmotion_end_slides_500x500_FA.mov -acodec copy -vcodec copy out.swf [20:34:04 CEST] <eemgr> sure [20:34:28 CEST] <kepstin> that 'does not contain any stream' message usually means that some message printed earlier explains /why/ there's no stream [20:34:29 CEST] <eemgr> https://gist.github.com/Michels10/cd0173652ba2f029a6091d35513b7c32 [20:34:31 CEST] <the_gamer> Mysoft, here you are, sorry for the delay https://drive.google.com/file/d/0B2OzieSeNyaYLTd3eWVndjBVY3M/view?usp=shari… [20:34:59 CEST] <furq> where's the rest of the output [20:35:00 CEST] <kepstin> eemgr: you forgot to put a -i before your input file [20:35:04 CEST] <furq> oh [20:35:04 CEST] <eemgr> oops [20:35:06 CEST] <furq> that's whewre [20:35:27 CEST] <blindwolf> Evening [20:35:30 CEST] <Mysoft> the_gamer ouch 149mb sample :P [20:35:42 CEST] <eemgr> https://gist.github.com/Michels10/cd0173652ba2f029a6091d35513b7c32 [20:35:54 CEST] <kepstin> "SWF muxer only supports VP6, FLV1 and MJPEG" fun [20:36:00 CEST] <furq> nice [20:36:02 CEST] <eemgr> hm [20:36:05 CEST] <kepstin> you're probably better off using the flv muxer [20:36:08 CEST] <the_gamer> Mysoft, i would make it smaller& but mencoder only does the first picture as i have seen& and i can't cut it without fixing it :( [20:36:17 CEST] <kepstin> and then using external tools to embed the file into a player swf [20:36:21 CEST] <furq> well at least now we know it isn't just creating an flv [20:36:35 CEST] <Mysoft> the_gamer you could just used a hexeditor to kill it [20:36:43 CEST] <Mysoft> but yeah if there's headers at the end it would be a problem :P [20:36:48 CEST] <Mysoft> no problem :) [20:36:54 CEST] <kepstin> or if you're using an external flash player, it can probably just play your original mp4 file with no changes... [20:37:00 CEST] <bencoh> yeah it's really swf, not some flv variant [20:37:12 CEST] <eemgr> i've got an flv of the vid as well [20:37:50 CEST] <bencoh> I wonder if it plays properly with gnash :] [20:38:15 CEST] <kepstin> looks like it's using the old flash version, so probably would work with gnash, yeah [20:38:19 CEST] <the_gamer> Mysoft, please tell me when you have it so i can delete it [20:39:11 CEST] <eemgr> ffmpeg -i in.mov -acodec mp3 -vcodec flv1 out.swf worked but again with mediocre quality [20:39:45 CEST] <the_gamer> -qscale 0? [20:39:57 CEST] <eemgr> is that the default? [20:40:40 CEST] <kepstin> eemgr: the flv1/mp3 codecs are the default when using swf output [20:41:00 CEST] <eemgr> ah, i see [20:41:17 CEST] <kepstin> but yeah, you're probably better off not using it. With the old codecs supported, you're gonna get pretty big files for decent looking output [20:42:37 CEST] <eemgr> qscale 0 provided an acceptable quality, actually [20:42:52 CEST] <eemgr> just need to get it to around 2.2mb [20:42:58 CEST] <eemgr> is that the -b option? [20:46:07 CEST] <Mysoft> got it the_gamer [20:46:15 CEST] <the_gamer> thank you [20:47:27 CEST] <Mysoft> lol even windows media player 6 [20:47:34 CEST] <Mysoft> can play that video [20:47:41 CEST] <Mysoft> really weird that ffmpeg is struggling with it [20:49:59 CEST] <the_gamer> Mysoft, i got many of those, the cat's camera always does that format so i would love to know how to deal with it [20:50:07 CEST] <eemgr> thanks for all of your help guys [20:50:13 CEST] <the_gamer> if ffmpeg can't do it i have to use mencoder. better than nothing [20:50:18 CEST] <Mysoft> the_gamer yeah let's see what's wrong :) [21:01:46 CEST] <durandal_170> the_gamer: report issue to bug tracker [21:08:05 CEST] <Mysoft> :] [21:08:12 CEST] <Mysoft> got it [21:08:18 CEST] <Mysoft> let's see if it will generate valid output [21:10:10 CEST] <the_gamer> durandal_170, did it& site loads forever atm [21:11:18 CEST] <the_gamer> Mysoft, i am curious about more information :) [21:11:32 CEST] <Mysoft> ffmpeg -f mjpeg -an -i PICT0001.AVI output.mp4 [21:11:58 CEST] <Mysoft> or [21:12:18 CEST] <the_gamer> durandal_170, i get 530, tracker seems to be down now? [21:12:24 CEST] <Mysoft> ffmpeg -f mjpeg -an -i PICT0001.AVI -vcodec copy output.avi [21:12:25 CEST] <the_gamer> 503 sorry [21:12:34 CEST] <Mysoft> to just fix the .avi container [21:14:06 CEST] <bencoh> what does "file" says on this "avi", btw? [21:15:00 CEST] <the_gamer> PICT0001.AVI: RIFF (little-endian) data, AVI, 640 x 480, 30.00 fps, video: Motion JPEG, audio: 0 Hz) [21:16:04 CEST] <the_gamer> ticket is created: https://trac.ffmpeg.org/ticket/5770#ticket [21:16:22 CEST] <bencoh> then -f mjpeg wont work. -c:v mjpeg, maybe [21:16:25 CEST] <bencoh> (before -i) [21:16:26 CEST] <durandal_170> you created 4 tickets [21:16:51 CEST] <Mysoft> bencoh it just did [21:16:59 CEST] <bencoh> ? [21:17:04 CEST] <the_gamer> durandal_170, oh, that was the error i guess& let me see if i can delete 3 [21:17:12 CEST] <Mysoft> ffmpeg -f mjpeg -an -i PICT0001.AVI -vcodec copy output.avi [21:17:17 CEST] <Mysoft> ^ i actually tried that [21:17:19 CEST] <Mysoft> before pasting it here [21:17:36 CEST] <bencoh> -f mjpeg works on an AVI file? [21:17:37 CEST] <bencoh> wtf [21:17:46 CEST] <Mysoft> well i assume it may be like mp3 [21:17:53 CEST] <bencoh> huh? [21:18:00 CEST] <durandal_170> can work if there is parser [21:18:14 CEST] <Mysoft> where it will just look for the tag [21:18:25 CEST] <Mysoft> so you can have garbage at the end of a .mp3 file [21:18:29 CEST] <Mysoft> but all players will play just fine [21:18:31 CEST] <bencoh> ah [21:18:34 CEST] <Mysoft> *at the START [21:18:42 CEST] <durandal_170> the_gamer:stop creating new tickets [21:19:02 CEST] <bencoh> if mjpeg data is stored in a single RIFF chunk, maybe [21:19:29 CEST] <Mysoft> yeah digital cameras [21:19:32 CEST] <Mysoft> tend to do that :P [21:19:40 CEST] <the_gamer> durandal_170, i am closing atm [21:19:42 CEST] <Mysoft> they just issue an almost fixed broken header [21:19:47 CEST] <Mysoft> and then spam jpegs :P [21:20:17 CEST] <the_gamer> durandal_170, got 503, don't know if something was wrong with my connection/browser or the server [21:20:27 CEST] <the_gamer> durandal_170, was not on purpose, i am sorry [21:20:49 CEST] <durandal_170> np [21:21:26 CEST] <the_gamer> only got them to close, can't see a way to delete them :( [21:21:49 CEST] <bencoh> that was taken care of already :) [21:22:16 CEST] <the_gamer> bencoh, my duplicate tickets? [21:22:20 CEST] <bencoh> ah, nevermind [21:22:21 CEST] <Mysoft> so bencoh i assume mjpeg will just look for the jpeg header [21:22:24 CEST] <Mysoft> for every frame :) [21:22:36 CEST] <Mysoft> just like mp3 do for each audio frame [21:23:18 CEST] <the_gamer> Mysoft, works, thank you [21:23:18 CEST] <durandal_170> mp3 is bad example [21:23:27 CEST] <Mysoft> why? [21:23:31 CEST] <Mysoft> they are both mpeg O_o [21:24:12 CEST] <durandal_170> it does not separate frames like jpeg.. [21:24:19 CEST] <Mysoft> yes it does [21:24:35 CEST] <Mysoft> mp3 is split in "audio frames" [21:26:18 CEST] <Mysoft> that's how a .mpg file does it's audio... [21:26:27 CEST] <Mysoft> altough .mpg unlike mjpeg has intraframes :P [21:27:04 CEST] <durandal_170> compare mjpeg parser code with mp3 one [21:30:57 CEST] <Mysoft> heh well i don't know about how specifically ffmpeg does it but [21:31:29 CEST] <Mysoft> "An MP3 file is made up of MP3 frames, which consist of a header and a data block. This sequence of frames is called an elementary stream. " [21:31:31 CEST] <Mysoft> "Motion JPEG (M-JPEG or MJPEG) is a video compression format in which each video frame or interlaced field of a digital video sequence is compressed separately as a JPEG image." [21:31:37 CEST] <Mysoft> to me that means they follow the same idea [21:32:23 CEST] <durandal_170> yes.... [00:00:00 CEST] --- Thu Aug 11 2016

1 0

ffmpeg-devel.log.20160809
by burek 10 Aug '16

10 Aug '16

[00:08:43 CEST] <iive> i don't see patch for that commit in the maillist [00:09:32 CEST] <iive> so this cannot be old patch that have been committed by somebody else. [00:11:11 CEST] <durandal_1707> your point being? [00:11:24 CEST] <llogan> he simply wants to know where the patch came from [00:15:24 CEST] <durandal_1707> live mplayer fork irc channel [00:18:05 CEST] <iive> the point is, that this commit seems to be directly from Derek and he had left ffmpeg. [00:18:54 CEST] <nevcairiel> git records who pushed a commit separate from who authored it, you should check it out [00:19:34 CEST] <durandal_1707> If Elvis can why can't Derek? [00:20:05 CEST] <iive> durandal_1707: never heard of fork named live mplayer before. [00:20:31 CEST] <durandal_1707> its #mpv [00:21:38 CEST] <iive> so you are the committer. fine. [00:23:39 CEST] <iive> nevcairiel: has that always been the case. I remember having issue with finding out the committer, some years ago. [00:24:47 CEST] <nevcairiel> ever since I used git, so at least a couple years now, cant speak for "always" [00:25:01 CEST] <iive> :) [00:25:14 CEST] <durandal_1707> svn days? [00:38:45 CEST] <iive> svn doesn't have an author, only committer. And it seems that both fields are present in initial git versions. [00:38:56 CEST] <iive> well, my bad. [01:44:49 CEST] <jya> is there a way with ffmpeg to remux a mp4 and either add a proper edit list or rework the pts so that the first sample starts at 0 ? [01:55:55 CEST] <llogan> that's a question for #ffmpeg [02:03:34 CEST] <jya> llogan: thought that one day someone will interject that :) [02:18:42 CEST] <cone-371> ffmpeg 03Derek Buitenhuis 07n3.1.2:HEAD: docs/filters: Fix parameter names for colorspace filter [04:19:15 CEST] <hpj_> hi, guys! since it doesn't seem to have been reported in the irc channels and the mailing list, i'm here to break the bad news: one of the ffmpeg contributors died last week [04:22:11 CEST] <hpj_> LEI Xiaohua, doctoral candidate at the Communications University of China in beijing, age 25, died on 4 august on the university campus [04:25:51 CEST] <hpj_> he had worked through the previous night and was found dead in the morning near the elevator in the lab building [04:26:58 CEST] <hpj_> cause of death was presumably exhaustion, no suspicion of foul play, he had been generally healthy and even a half-marathon runner [04:29:27 CEST] <hpj_> despite his youth, lei had established himself as a highly respected expert in the chinese language world of a/v developers, writing numerous tutorials and always eager to help other developers with their problems [04:32:48 CEST] <hpj_> his award-winning blog on a/v development has 4.5 million visits and is considered a reference guide in the chinese a/v devel world [04:33:31 CEST] <hpj_> here is a newspaper report on this death (in chinese): http://edu.qq.com/a/20160804/003711.htm [09:29:46 CEST] <Timothy_Gu> hpj_: Not to be insensitive, but I don't see his name mentioned in the log of any FFmpeg commits. Nevertheless, I just read some of the blog posts on FFmpeg and those were helpful indeed, especially for people whose Chinese is better than English. As a compatriot and another contributor to FFmpeg, I'd like to think he is now in a better place. [09:45:11 CEST] <hpj_> yes, i guess his work was more around ffmpeg than within the project itself: http://leixiaohua1020.github.io/#ffmpeg-development-examples [10:52:27 CEST] <Compn> hpj_ : wow , thats crazy. although unexpected for a 25 year old to die from exhaustion doing computer work [10:52:36 CEST] <Compn> thanks for sharing [11:39:08 CEST] <hpj_> Compn: in fact, there have been a series of high-profile sudden deaths (heart attack and alike) in the chinese it industry in recent times [11:42:37 CEST] <hpj_> not to mention cases of sudden deaths of chinese gamers every other day, often having sat in internet cafes for days on end (chinese gamers often prefer to play in internet cafes instead of at home because of the low latency internet connection there) [12:19:59 CEST] <Compn> hpj_ : i'm aware that this happens in china. but i dont think its common occurance here in usa. [12:20:41 CEST] <Compn> cultural differences in dealing with long working hours and exhaustion [12:36:04 CEST] <cone-118> ffmpeg 03Paul B Mahol 07master:011837285ffc: doc/filters: add one more lutyuv example [12:46:35 CEST] <cone-118> ffmpeg 03Michael Niedermayer 07master:369ed11e3c8a: avfilter/drawutils: Fix single plane with alpha [13:41:50 CEST] <ubitux> we are doing something bad in drawtext [13:42:07 CEST] <ubitux> fenv.h use of feclearexcept()/fetestexcept() [13:42:17 CEST] <ubitux> unless i'm mistaken, these are not thread safe [13:42:33 CEST] <ubitux> this was introduced in dec87454 [13:43:04 CEST] <ubitux> it's the only place where we use such include [13:43:18 CEST] <ubitux> andrey_utkin ^ [13:46:44 CEST] <rcombs> it's thread-safe as of C11, but we're not that [13:46:45 CEST] <ubitux> or maybe they actually are [13:47:01 CEST] <rcombs> probably thread-safe on real implementations but& still yucky [13:47:25 CEST] <ubitux> does "thread-safe" implies "not global state"? [13:47:34 CEST] <andrey_utkin> ubitux, i wonder what we could do with that. seems we can only drop this code - just to convert silently and hope for better [13:48:07 CEST] <ubitux> that's what we do everywhere, isn't it? [13:48:16 CEST] <rcombs> ubitux: as of C11/C++11 each thread must maintain its own floating-point exception mask [13:48:31 CEST] <andrey_utkin> i don't remember clearly why i added it. either my own error-handling paranoia, or i was pointed to this during patch review [13:48:46 CEST] <rcombs> previously it's unspecified, and probably worked the same way since iirc that's what x86 and ARM actually do [13:49:25 CEST] <andrey_utkin> ubitux, i think since you've found this you want to send the patch yourself? [13:49:38 CEST] <rcombs> why not just check if it's > INT_MAX or < INT_MIN though? [13:49:45 CEST] <rcombs> (or whatever the macros for the relevant type are) [13:50:03 CEST] <andrey_utkin> what "it" (in "it's")? [13:50:07 CEST] <rcombs> plus isnan I guess [13:50:21 CEST] <rcombs> these checks are around an assignment from a float to an integer [13:50:40 CEST] <rcombs> double->int, now that I check [13:50:53 CEST] <rcombs> should be fine to just check if it's out of range or NaN, yeah? [13:51:04 CEST] <ubitux> andrey_utkin: i wasn't planing to send a patch :p [13:51:17 CEST] <ubitux> just reading stuff about fp and came accross this code in ffmpeg [13:51:42 CEST] <andrey_utkin> i think checks on range of INT_MAX/MIN are pointless because they apparently involve same conversion as the present one [13:52:12 CEST] <andrey_utkin> check on nan sounds reasnable [13:55:23 CEST] <rcombs> andrey_utkin: the comparison would be done between doubles [13:56:22 CEST] <andrey_utkin> rcombs, in "if (res < INT_MAX)" and such? [13:56:39 CEST] <rcombs> yeah, INT_MAX would be promoted to double there, no? [13:57:24 CEST] <andrey_utkin> i thought otherwise for no reason, need to check [13:58:58 CEST] <andrey_utkin> according to this, you rcombs are right https://www.eskimo.com/~scs/cclass/int/sx4cb.html [14:01:01 CEST] <rcombs> also that check has more problems [14:01:38 CEST] Action: rcombs double-checks something [14:04:05 CEST] <rcombs> OK so that'd be CVTTSD2SI [14:04:21 CEST] <rcombs> the check looks for FE_INVALID|FE_OVERFLOW|FE_UNDERFLOW [14:04:42 CEST] <rcombs> OVERFLOW and UNDERFLOW are impossible here [14:04:48 CEST] <rcombs> but FE_INEXACT is possible and not checked for [14:05:55 CEST] <rcombs> (maybe it's different on x87? I dunno) [14:11:47 CEST] <rcombs> and even if this was all technically correct and thread-safe, it's pretty rude to change the floating-point exception mask in library code unless you're on a dedicated thread [14:28:39 CEST] <andrey_utkin> thanks for pointing this all out, i've put this on my schedule and hopefully i'll sanitize this soon. ubitux, rcombs - please give your emails so that i CC you when i send a patcg [14:29:00 CEST] <rcombs> we're on the list anyway [14:29:06 CEST] <ubitux> no need to cc :) [16:57:15 CEST] <durandal_1707> can I get comments for fillgaps filter? [19:54:33 CEST] <coralisk> rekt [22:00:40 CEST] <jamrial_> for whoever wants to try converting ffmpeg to codecpar, http://pastebin.com/YKcEFcQ4 combines all five relevant commits that were noop'd during merges [22:02:12 CEST] <jamrial_> should hopefully be more helpful than looking only at the first one, which still kept some avstream->codec usage in place [22:03:51 CEST] <durandal_1707> whats so hard about it? [22:06:18 CEST] <jamrial_> last i checked there were some format specific hacks that used avctx fields not present in codecpar [22:52:02 CEST] <rcombs> I'd still like some way to export arbitrary extra fields from decoders or parsers [22:52:37 CEST] <rcombs> that gets exposed on the AVStream (so you can get it out of find_stream_info instead of running the decoder yourself) [23:05:12 CEST] <j-b> 1m [00:00:00 CEST] --- Wed Aug 10 2016

1 0

ffmpeg.log.20160809
by burek 10 Aug '16

10 Aug '16

[00:05:47 CEST] <bray90820> llogan: Thanks [00:10:00 CEST] <bray90820> licson: I ran it with a WMA2 file in the directory but it said "*.wmv: No such file or directory" [00:10:54 CEST] <llogan> did you navigate to the directory first before running the command. did you modify the command at all? are there any *.wmv files in the directory? [00:13:18 CEST] <bray90820> licson: I did [00:13:18 CEST] <bray90820> http://pastebin.com/raw/Xu8VXmhm [00:13:34 CEST] <bray90820> I did not modify it [00:13:51 CEST] <bray90820> And all the files are AVI [00:15:19 CEST] <bray90820> licson: I hd to change WMA to AVI [00:15:25 CEST] <bray90820> I got it working now thanks [00:15:43 CEST] <bray90820> But Could I make it check any file type? [00:16:06 CEST] <ozette> complete output of mkv > m3u8 : https://paste.fedoraproject.org/404572/06945361/ [00:17:09 CEST] <llogan> bray90820: that's more of a bash question than ffprobe: for f in *; [00:17:33 CEST] <bray90820> Alright [00:17:34 CEST] <bray90820> Thanks [00:18:41 CEST] <ozette> my question.. why does the playlist start at out194.ts [00:20:36 CEST] <ozette> it's as if out.m3u8 is overwritten each 5 segments [00:20:56 CEST] <ozette> and finally i end up with a playlist which starts at the last 5 [01:12:03 CEST] <llogan> ozette: -hls_list_size default is 5. [01:12:13 CEST] <llogan> add "-hls_list_size 0" [01:12:58 CEST] <llogan> http://ffmpeg.org/ffmpeg-formats.html#hls-1 [01:22:42 CEST] <ozette> llogan: ah, thanks, now you say.. i recall reading about that some time ago [01:22:57 CEST] <ozette> have to wonder why the default is 5 though [02:03:48 CEST] <jya> is there a way with ffmpeg to remux a mp4 and either add a proper edit list or rework the pts so that the first sample starts at 0 ? [02:05:14 CEST] <c_14> the second should be done automatically [02:07:15 CEST] <c_14> though you can try explicitly passing -start_at_zero or whatever it was called [02:08:25 CEST] <llogan> -avoid_negative_ts make_zero [02:21:22 CEST] <jya> c_14: I already have a mp4, whatever google used to generate those samples, the first sample is either 0.083333 or 0.0666667 [02:21:39 CEST] <jya> I'm trying to make them all start at the same time (don't really mind if it isn't 0). [02:23:09 CEST] <jya> c_14: tried ffmpeg -i test.mp4 -start_at_zero -vcodec copy -acodec copy test2.mp4, this doesn't appear to change the first cts [02:23:38 CEST] <c_14> add -avoid_negative_ts make_zero [02:24:09 CEST] <jya> weird argument for a mp4 :) [02:25:58 CEST] <jya> c_14: nope.. still starting at a non-0 value (it has an audio track which does start at 0 if that makes a difference) [02:26:16 CEST] <c_14> it does [02:26:27 CEST] <c_14> ffmpeg shifts all tracks equally [02:26:43 CEST] <c_14> So if the tracks don't start at the same time it'll shift the one that starts first to 0 [02:27:02 CEST] <jya> hmmm.. any way to force it ? don't mind if a/v sync is slightly off [02:27:49 CEST] <jya> the other issue is that my original samples are fragmented mp4, and I'd like to keep them that way... [02:28:16 CEST] <c_14> fragmented how? there's a bunch of fragmenting options for the mp4 muxer [02:28:46 CEST] <c_14> https://ffmpeg.org/ffmpeg-formats.html#mov_002c-mp4_002c-ismv [02:28:58 CEST] <jya> c_14: it's those files there: https://github.com/w3c/web-platform-tests/tree/master/media-source/mp4 [02:30:17 CEST] <jya> i'm trying to fix the files because right now, the MSE w3c tests checks that when you add those files, the buffered range (the intersection of all tracks) starts at 0. but this is incorrect as audio starts at 0 but video at a different time. The test works because blink incorrectly use dts in place of pts [02:31:09 CEST] <c_14> There's frag_keyframe, frag_duration and frag_size. You'll have to figure out which of those is right for the fragmentation (maybe check with boxdumper or something) [02:31:48 CEST] <jya> will do thanks. those options above still do not work right. [02:31:51 CEST] <c_14> As to the offset, either find the offset for the files and then seek to it (discarding part of the audio) or extract the audio and video to separate files (so the timestamps start at 0 for both) and then combine them again [02:32:49 CEST] <jya> by default this file https://github.com/w3c/web-platform-tests/blob/master/media-source/mp4/test… has a first sample of 0.16667 (according to ffprobe) after doing ffmpeg -i test-v-128k-320x240-24fps-8kfr.mp4 -start_at_zero -avoid_negative_ts make_zero -vcodec copy -acodec copy test2.mp4, now the first sample has a time of [02:32:49 CEST] <jya> 0.083008 [02:36:38 CEST] <c_14> Getting rid of the avoid_negative_ts option seems to make it start at 0.0000 [02:44:21 CEST] <jya> c_14: ah yes indeed.. awesome [03:39:34 CEST] <KDDLB> hm [03:39:48 CEST] <KDDLB> how do I change (not convert) the framerate on a video file? [03:40:23 CEST] <KDDLB> e.g.: I have a 240FPS video, but its actually 30FPS [03:43:07 CEST] <relaxed> KDDLB: man ffmpeg|less +/-r\\[ [03:45:26 CEST] <KDDLB> there we go [03:45:32 CEST] <KDDLB> I need to reencode it, it looks like [03:45:38 CEST] <KDDLB> I did -c:v copy and it didnt work [03:45:57 CEST] <relaxed> whole command? [03:46:18 CEST] <KDDLB> ffmpeg -r 240 -i Downloads/VID_20160808_213636.mp4 video.mp4 [03:46:40 CEST] <KDDLB> I tried with & ffmpeg -r 240 -i Downloads/VID_20160808_213636.mp4 -c:v copy video.mp4 [03:52:54 CEST] <darkliight> Hi, just starting to play with ffmpeg ... when I 'ffprobe -i bluray:.', I'm told there are '7 usable playlists, and then continues by choosing the longest playlist .... is there a way to just have ffprobe spit back a list of usable playlists and stop after that? [03:53:28 CEST] <relaxed> KDDLB: you may have to re-encode [03:53:39 CEST] <KDDLB> re-encoding works [06:01:43 CEST] <budric> hi, can anyone shed some light on how to get ffmpeg to stream video to kodi? Do I need to startup ffserver and do ffmpeg -> ffserver -> kodi, or is there an easier way? [08:34:12 CEST] <the_cuckoo> hi - was wondering if there's any high level documentation concerning avfilter? I've had some success with it, but i'm having a problem with yadif/send_field and a progressive input... can paste code if anyone wants to review it (based on the ffmpeg sample), but docs would be fine [09:17:19 CEST] <ahoo> it is possible to change the framerate of a video without re-encoding it? [09:17:40 CEST] <ahoo> i wouldn't have thought that was possible. [09:18:13 CEST] <ahoo> KDDLB: i don't think you can. [09:18:45 CEST] <furq> sure you can [09:18:48 CEST] <furq> not with ffmpeg though [09:19:24 CEST] <furq> dedicated mp4/mkv muxers (mp4box, l-smash, mkvtoolnix) can do it with h264 streams [09:19:36 CEST] <furq> it probably works with other formats/codecs but i've never had cause to find out [09:19:51 CEST] <ahoo> so the framerate can be specified in the container and the stream is then played back at that rate? [09:20:15 CEST] <furq> something like that [09:20:15 CEST] <ahoo> oh never mind. [09:20:26 CEST] <Mavrik> There's no specified farmerate for H.264 [09:20:34 CEST] <Mavrik> Video has to be retimestamped and audio adjusted as well. [09:20:38 CEST] <furq> ^ [09:20:50 CEST] <furq> if it was as simple as setting a metadata flag then ffmpeg would be able to do it [09:22:15 CEST] <ahoo> so [09:22:23 CEST] <ahoo> it isn't possible without re-encoding. [09:22:30 CEST] <furq> yes it is [09:22:38 CEST] <ahoo> we just concluded it is not [09:22:42 CEST] <ahoo> didn't we? [09:22:42 CEST] <furq> no we didn't [09:22:45 CEST] <ahoo> ok [09:22:50 CEST] <ahoo> i see :3 [09:23:29 CEST] <ahoo> maybe different wording helps [09:23:49 CEST] <ahoo> how about "it isn't possible without re-processing the streams" [09:24:05 CEST] <furq> i guess [09:24:21 CEST] <furq> but the resulting video will be untouched (other than the playback speed) [09:24:47 CEST] <ahoo> oh [09:24:51 CEST] <ahoo> is that the case? [09:25:00 CEST] <furq> yeah [09:25:01 CEST] <ahoo> then i understand. it is. [09:25:06 CEST] <furq> you just need to rewrite the timestamps [09:25:17 CEST] <ahoo> and the timestamps are in the container? [09:25:26 CEST] <furq> that's my understanding of it [09:25:28 CEST] <furq> with h.264 at least [09:25:29 CEST] <ahoo> i see [09:26:25 CEST] <furq> you will probably need to reencode the audio though [09:28:24 CEST] <ahoo> hehe, or tolerate the reverb [09:28:43 CEST] <ahoo> phasing/ [09:40:34 CEST] <durandal_1707> the_cuckoo: could you pastebin code? [09:41:03 CEST] <the_cuckoo> durandal_1707: sure - gimme a couple of minutes while i rip out the specifics [09:42:05 CEST] <durandal_1707> also what's actual problem? [09:42:31 CEST] <the_cuckoo> when the input is progressive, i don't get anything out until i flush - and then it's only the last frame [09:42:53 CEST] <the_cuckoo> when the input is interlaced, i get the correct result - twice the output [09:46:37 CEST] <the_cuckoo> durandal_1707: http://pastebin.ca/3679427 [09:48:03 CEST] <the_cuckoo> the diagnostic on line 151 shows that all frames are presented and the one line 130 shows that i get EGAIN repeatedly - until i flush, then i get a frame [09:48:17 CEST] <the_cuckoo> (being the last one) [09:49:03 CEST] <the_cuckoo> i can work around it specifically for the yadif filter, but this is more of a generic filter which wraps any avfilter chain [09:55:08 CEST] <the_cuckoo> furq: no, you don't need to re-encode the audio - but otherwise, yes, video has to be [09:56:43 CEST] <the_cuckoo> audio isn't aligned to frame rate - the number of samples per audio frame is normally something like 1024 (aac) or 1152 (mp2/mp3), or it can be variable (vorbis) [09:57:28 CEST] <durandal_1707> the_cuckoo: did you try to set option to deint only interlaced? [09:58:27 CEST] <the_cuckoo> durandal_1707: yup - result[ L"mode" ] = L"send_field"; result[ L"parity" ] = L"auto"; result[ L"deint" ] = L"interlaced"; [09:59:26 CEST] <the_cuckoo> ffmpeg doesn't have the problem - just my wrapper [09:59:49 CEST] <durandal_1707> you set pts? [10:00:21 CEST] <the_cuckoo> av_frame->pts = frame->get_position(); - line 166 [10:01:05 CEST] <the_cuckoo> they're 0 based and increments in steps of 1 (being the ratio of the frame rate) [10:01:50 CEST] <the_cuckoo> timebase stuff being specified on line 32 [10:28:06 CEST] <durandal_1707> the_cuckoo: you used code from examples directory as template? [10:28:44 CEST] <the_cuckoo> yup - some of it is lifted verbatim and is pretty much unchanged (should find the comments match iirc) [10:32:46 CEST] <durandal_1707> the example works here just fine with yadif, also your pastebin have only calling of get frame in flushing [10:33:15 CEST] <the_cuckoo> ah - may have missed something in the cutting [10:34:04 CEST] <the_cuckoo> hmm - yeah - can add that to the paste if i can edit - checking [10:35:03 CEST] <the_cuckoo> hmmph - couldn't edit - http://pastebin.ca/3679490 [10:35:30 CEST] <the_cuckoo> added the main entry point to the class - do_fetch - at line 179 [10:35:56 CEST] <the_cuckoo> it exposes more internal plumbing details which may seem confusing (but are largely irrelevant) [10:37:24 CEST] <the_cuckoo> (the code is lgpl btw - haven't pushed it to a public repo yet, but will do shortly) [10:38:54 CEST] <the_cuckoo> the image is extracted on line 252 [10:40:13 CEST] <durandal_1707> I only see that you don't handle eagain from filtergraph [10:40:40 CEST] <durandal_1707> in that case you need to feed next frame [10:40:58 CEST] <the_cuckoo> we do - line 259 [10:41:08 CEST] <durandal_1707> and try again to receive frame [10:41:19 CEST] <the_cuckoo> yup - all in the loop [10:42:07 CEST] <Batman_> Hello, can anyone help me with this problem: http://stackoverflow.com/questions/38827978/ffmeg-dumping-a-rtsp-streaming-… ? Thank you! [10:42:54 CEST] <durandal_1707> yes but I nowhere see that you feed graph with another frame [10:44:05 CEST] <the_cuckoo> ah :) - right - that's in the callback... line 139 - this is automatically inveked from the fetch on line 238 [10:44:17 CEST] <the_cuckoo> invoked even [10:45:02 CEST] <the_cuckoo> was trying to hide some of the more confusing details there :) - the routing is complicated because the frame holds audio and video - we need to split the audio up to the resultant frame rate) [10:46:03 CEST] <the_cuckoo> the frame being the frame_type_ptr - not the avframe of course [10:52:09 CEST] <durandal_1707> in loop at line 230 you must ensure that buffersrc will receive frame when buffersink returns eagain [10:53:36 CEST] <durandal_1707> using callback is confusing, add logs to see when you give and receive AVframe [10:55:14 CEST] <the_cuckoo> yeah - i know - it's a little bit warped :) - i couldn't find a better way to approach it - but for sure, the frames are delivered - diagnostic on line 151 shows that during the collection of frame 0, the entire input is presented [10:56:04 CEST] <the_cuckoo> the do/while on 230 doesn't restrict the number of frames given to you guys at all [10:56:23 CEST] <the_cuckoo> it just keeps on going until eof is received [10:58:46 CEST] <the_cuckoo> it's curious - does exactly the right thing with an interlaced input [11:00:40 CEST] <durandal_1707> what happens with all progressive input? [11:01:49 CEST] <the_cuckoo> well, just the problem i'm facing - the loop pulls through every frame from the input during the first frame - hits eof - flushes the context - receives the last progressive image [11:02:19 CEST] <the_cuckoo> doesn't matter how long the input is [11:02:45 CEST] <the_cuckoo> send_frame does the right thing afaict - passthrough of the input [11:04:05 CEST] <the_cuckoo> oh - curious... [11:04:15 CEST] <the_cuckoo> the result of that is 50 fps [11:04:59 CEST] <the_cuckoo> that can't be right... [11:06:16 CEST] <the_cuckoo> fps_out_ = av_buffersink_get_frame_rate( buffersink_ctx_ ); <- line 98 [11:06:48 CEST] <the_cuckoo> oh - we can't know that there can we? [11:07:23 CEST] <the_cuckoo> you haven't seen anything from the input yet - you don't know if it's interlaced or progressive.. [11:09:34 CEST] <the_cuckoo> we do of course :) [11:10:46 CEST] <the_cuckoo> umm - i wonder - perhaps the quick and dirty hack of detecting the progressive/yadif combination is the best route here - ie: just set it up to completely bypass your stuff [11:11:04 CEST] <the_cuckoo> it breaks some stuff for us, but nothing which is critical at this point [11:13:34 CEST] <the_cuckoo> still - doesn't explain why this works: ffmpeg -y -v 0 -i out.ts -vf yadif=send_field:auto:interlaced out2.ts (where out.ts is a 250 frame progressive input at 25fps - out2.ts is also a 250 frame progressive output at 25 fps) [11:20:51 CEST] <the_cuckoo> is there some kind of reconfiguration going on internally in this case which we can detect? [11:40:11 CEST] <the_cuckoo> durandal_1707: have to say - i really appreciated that you tried to help me out here :) - thinking i've zoned in on the problem - an incorrect solution will suffice for the time being (hopefully) - would still love to get to the bottom of it though :) [11:45:33 CEST] <durandal_1707> the_cuckoo: you can just not feed it with progressive frames [11:49:38 CEST] <the_cuckoo> yeah - but like i said - it's not a specific wrapper for yadif :) - it's supposed to expose all your video filters [11:50:38 CEST] <the_cuckoo> that makes it a tad trickier to filter out yadif from the avfilter graph set up [11:50:50 CEST] <the_cuckoo> (and it could get it wrong) [11:54:00 CEST] <the_cuckoo> but to be clear, that will be my path ahead for the time being [12:06:52 CEST] <the_cuckoo> Batman_: hmm - [rtsp @ 0x1f18340] UDP timeout, retrying with TCP would seem to be the problem [12:08:10 CEST] <the_cuckoo> i'd probably test with the udp:// protocol myself (thus bypassing a lot of stuff in the process), but it certainly looks curious [12:09:35 CEST] <the_cuckoo> https://ffmpeg.org/pipermail/ffmpeg-user/2014-June/022086.html <- may help? [12:11:54 CEST] <Spring> Does libvpx have any options specific to it that aren't supported by h.264? [12:16:41 CEST] <Spring> found that when ffplay's -loop is set to 0, and the trim is short (eg a few seconds long) upon looping the second time it slows down the playback. Is this normal? [12:16:43 CEST] <the_cuckoo> Spring: yes - loads - ffmpeg -h full | less and search of libvpx [12:17:17 CEST] <Spring> the_cuckoo, thanks for the tip [12:19:17 CEST] <Spring> hmm, nothing returns for libvpx in the full output [12:19:43 CEST] <Spring> oh, probably because the buffer isn't long enough :p [12:20:36 CEST] <the_cuckoo> :D - hence less [12:20:41 CEST] <the_cuckoo> or more [12:30:32 CEST] <Spring> it's strange, as sometimes ffplay loops without slow down while other times not. For example I can reliably reproduce the issue when looping a portion of a clip near the beginning [12:30:56 CEST] <Spring> but looping later in the clips the slow down isn't present [12:36:47 CEST] <Spring> here's a captured WebM of the bug: https://a.uguu.se/pzzEqQOFmL88_ffplayloopslowdownbug%5BTrailers-Umbrellasof… [12:37:23 CEST] <Spring> excuse the long filename, but it shows the issue. Loop from '5' seconds to '8' seconds of the trailer. [12:43:04 CEST] <Spring> pastebin of the ffplay command: http://pastebin.com/Fc0bSAXR [13:21:08 CEST] <omegaenigma> anyone familiar with mixed resolution video overlay opacity settings? [13:43:28 CEST] <Batman_> the_cuckoo: Thanks for the tip, I have to restart my PC, I'll try what you pointed [13:46:27 CEST] <xeche_> Hello people. Anyone around to help me make sense of using hevc_qsv from command line? ffmpeg -codecs reports hevc_qsv available both for encoding and decoding, but actually using it gives me an error "[hevc_qsv @ 02659ac0] Could not load the requested plugin: 2fca99749fdb49aeb121a5b63ef568f7" [13:46:58 CEST] <xeche_> I'm running it on an i7 Skylake, so I don't think there should be anything wrong in the way of hardware support. [13:49:05 CEST] <BtbN> Are you on Linux? [13:49:23 CEST] <xeche_> BtbN: Windows. :) [13:50:33 CEST] <xeche_> I'm looking in the IntelSWTools folder, which comes with the Intel Media SDK installer. It has a folder with various DLLs, as well as a couple of subfolders that look very much like they're named after plugin UIDs [13:51:57 CEST] <xeche_> However, the UID name I gave in the above error message is not a folder. So am I correct in assuming that ffmpeg, even if it is prebuilt with qsv support, and the Intel SDK DLLs are not enough to enable the QSV support? [13:52:49 CEST] <BtbN> Must be some driver issue with Intel, yes. No idea how to deal with it, Intel Drivers and especially QSV are a giant mess. [13:53:20 CEST] <BtbN> Are you sure your hardware even supports hevc encoding? [13:55:55 CEST] <xeche_> BtbN: According to Intel's documentation, yes. Looks like QSV is supported on any Intel CPU new than 2nd or 3rd generation [13:55:58 CEST] <xeche_> That's before Haswell. [13:56:07 CEST] <BtbN> QSV, yes. [13:56:10 CEST] <BtbN> But not HEVC. [13:56:34 CEST] <BtbN> Try encoding h264 instead, if it works, you know what's the issue. [13:56:44 CEST] <xeche_> Alright then. :) [13:56:46 CEST] <xeche_> Thanks. [13:57:08 CEST] <well0ne> hey [13:57:45 CEST] <well0ne> i'm trying to concenate h264+mp3 streams (single episodes) , to send em to a rtmp server, any codec settings are the same, they are properly encoded [13:58:08 CEST] <well0ne> but when im streaming them and they change to the next file i'll alwys get Non-monotonous DTS in output stream when the file is starting [13:58:24 CEST] <well0ne> as i said i properly encoded them ( i think so) [13:58:28 CEST] <well0ne> how can i prevent that [13:59:13 CEST] <xeche_> BtbN: Thanks for the input, sir. You are indeed right. h264_qsv did work. [14:00:46 CEST] <BtbN> well0ne, of course you get a non-monotonous DTS. You just append an entirely new stream, whos timestamps start from the beginning again. [14:01:23 CEST] <well0ne> okay, sure makes sense... [14:01:49 CEST] <well0ne> is there any option to fix it, to get sure the timestamps are new generated? [14:02:05 CEST] <well0ne> or is a re-encoding needed for that [14:02:13 CEST] <BtbN> not without transcoding. [14:02:18 CEST] <well0ne> damn [14:02:43 CEST] <BtbN> It shouldn't be too much of a problem though, or does anything cease working because of it? [14:03:36 CEST] <well0ne> no it seems fine to me, i had some sync issues before, but now as my framerates codecs and attributes are all the same, it seems to work, im just getting those messages when changing to the next file [14:04:43 CEST] <xeche_> BtbN: Looking up some more documentation, Skylake CPUs should support both encoding and decoding in hardware. However, it does specify HEVC 8-bit decode / encode. [14:04:52 CEST] <well0ne> but im confused, i stream series [14:05:05 CEST] <well0ne> i have a series, where i just copy the streams to the rtmp server [14:05:08 CEST] <well0ne> same codes [14:05:13 CEST] <xeche_> BtbN: I'm not entirely sure what 8-bit refers to in this context. Bits per plane? [14:05:16 CEST] <well0ne> but i dont get the dts messages there [14:06:35 CEST] <well0ne> http://pastebin.com/btuGyTdu [14:06:39 CEST] <well0ne> have a look here [14:06:57 CEST] <well0ne> the codes and attributes should be the same, so i dont know what this causes [14:09:44 CEST] <well0ne> i thought maybe this is happening of different input lengths of audio/video, but i used -shortent [15:05:45 CEST] <k_sze> I just tried to strip the audio stream from a video recorded with iPhone 6s Plus. [15:06:26 CEST] <k_sze> I used `-map 0 -map -0:a:0 -copy_unknown`, because there are two streams besides the video and audio streams, of unknown codec. [15:07:00 CEST] <k_sze> But I still got this image: "[mov @ 0x7fb17d811800] Unknown hldr_type for mebx / 0x7862656D, writing dummy values" [15:07:54 CEST] <k_sze> Does anybody know what that's about? [15:19:49 CEST] <the_cuckoo> why are you trying to preserve the other streams? isn't it enough to just use ffmpeg -i input.mp4 -acodec copy output.aac [15:20:04 CEST] <furq> he's trying to remove the audio stream [15:20:38 CEST] <the_cuckoo> k - then -vodec copy -an output.mp4 [15:20:46 CEST] <k_sze> s/image/message/ [15:21:42 CEST] <the_cuckoo> s/vodec/vcodec/ [15:26:27 CEST] <furq> mebx appears to be some kind of nonstandard apple extension (if you can imagine such a thing) [15:26:33 CEST] <furq> you can probably ignore it [15:31:18 CEST] <the_cuckoo> https://developer.apple.com/library/mac/documentation/QuickTime/QTFF/QTFFCh… <- gets a mention here - Timed Metadata Sample Description [15:31:55 CEST] <the_cuckoo> not clarifying it much for me though [15:34:32 CEST] <ritsuka> that track contains the gps location recorded with the video [15:34:45 CEST] <the_cuckoo> ah - cool [15:39:20 CEST] <lethalwp> hello, i was trying to play an hevc video with mpv, but it didn't want to hardware accellerate it. After some digging, it's ffmpeg that doesn't hwaccel YUV420P10LE; but it does for YUV420P. I've recompiled by adding the 10LE and it seems to work fine for me. Is there a reason hevc 10bit hwaccel isn't enabled? [15:52:12 CEST] <rmoorelxr> is there such thing as an INFO frame? or is that for sure some third party weirdness i'm encountering [16:00:44 CEST] <jkqxz> Infoframes are a thing in CEA-861 (and therefore HDMI). What is the context? [16:05:33 CEST] <rmoorelxr> receiving "raw h264 stream" that has I, P frames, and INFO frames [16:05:45 CEST] <rmoorelxr> getting data and frame defs from third party [16:06:04 CEST] <rmoorelxr> when I try to stream/write a file the result is looks scrambled [16:07:02 CEST] <rmoorelxr> I get frame timestamps with the byte array but I don't know how to manually turn them into useable video [16:07:30 CEST] <rmoorelxr> when I use provided libraries I can write to a file and it works, but I'm trying to avoid going to disk [16:09:47 CEST] <jkqxz> That sounds like some third-party weirdness. What do the data look like? (An annex B bytestream, for example.) [16:15:18 CEST] <rmoorelxr> INFO PPPPPPPPPP INFO PPPP I PPPPP INFO etc [16:16:31 CEST] <xeche_> Hello again, sirs (maybe madames). Has anyone used av_image_alloc with PIX_FMT_QSV before? [16:18:37 CEST] <jkqxz> xeche_: Don't. Make an AVHWFramesContext and allocate from that. (See ffmpeg_qsv.c.) [16:20:28 CEST] <xeche_> jkqxz: Thank you. Does this go for both encoding and decoding? [16:20:31 CEST] <jkqxz> Or actually not that either because it hasn't merged from libav yet. You have to make the surfaces yourself and them put them in the frames manually (setting buf[0] and data[3]). [16:21:18 CEST] <xeche_> jkqxz: Alright, so the utility functions for working with encoder input may or may not be the right thing to use for QSV ? [16:21:42 CEST] <xeche_> I'm usinga nightly build from about a week ago, so it's version 3.1.1 I believe. [16:21:55 CEST] <xeche_> *using a [16:22:24 CEST] <jkqxz> Yes. Hardware surfaces are all special and require weird handling. The hwcontext stuff makes that more consistent, but it is not merged to ffmpeg for libmfx/qsv. [16:23:27 CEST] <xeche_> jkqxz: I see. I can handle the manual memory management, I think, but the sws_scale functionality might be a bit of a hassle. [16:25:27 CEST] <jkqxz> swscale has no support at all for hardware surfaces. You may be able to memory map them and make it work, but that's all up to you to do. [16:27:51 CEST] <rmoorelxr> corrupted macroblocks in my garbage vid: http://pastebin.com/5Z42Q8rj [16:30:36 CEST] <the_cuckoo> durandal_1707: (after some distractions) i've got the passthrough stuff working just fine :) [16:32:07 CEST] <the_cuckoo> one other question regarding the avfilter api - if i want to key frame an effect (like say, have the video rotate for the first or last 25 frames), how would i achieve that? could i somehow specify the rotation for each frame or is there another way to do it? [16:33:16 CEST] <durandal_1707> if video dimensions change, you generally can't [16:33:33 CEST] <the_cuckoo> input video will remain the same [16:33:59 CEST] <the_cuckoo> the output - that can change - i'll composite it on to a fixed background if necessary [16:34:09 CEST] <durandal_1707> you will need to recreate filtergraph to be safe [16:34:55 CEST] <the_cuckoo> k - so you can't set up animatable keys in the avfilter graph spec either? [16:35:22 CEST] <durandal_1707> but if size is same there are commands and enable thing option [16:35:54 CEST] <kepstin> some filters support either changing (some) parameters at runtime via messages, and some let you use expressions in parameters (rotate does, for example) - and you can write animations via the expression syntax [16:36:18 CEST] <the_cuckoo> kepstin: ah - both sound good [16:37:13 CEST] Action: kepstin notes that the rotate filter has fixed output size, and will clip the frame if you don't pick an output size big enough to hold the rotated image. [16:37:15 CEST] <the_cuckoo> could i be lazy and ask for an example of the expression thing? (i'll chase the messages thing in the api) [16:37:34 CEST] <the_cuckoo> yeah - thought it did that [16:37:41 CEST] <furq> the_cuckoo: http://ffmpeg.org/ffmpeg-utils.html#Expression-Evaluation [16:37:55 CEST] <kepstin> there's also some examples on the rotate filter specifically: https://www.ffmpeg.org/ffmpeg-filters.html#rotate [16:38:06 CEST] <the_cuckoo> excellent - thanks guys [16:38:21 CEST] Action: kepstin once made a video that just tilted back and forth every couple seconds, almost made him seasick [16:38:32 CEST] <the_cuckoo> :) [16:41:25 CEST] <Spring> can anyone reproduce this? http://pastebin.com/Fc0bSAXR [16:43:32 CEST] <Spring> (ffplay playback slowdown after the first loop) for me this has affected every video I've tried. I've written down various start/end time combos that display the slowdown, and those that don't between 5-20 seconds. [16:43:38 CEST] <durandal_1707> Spring: what happens? [16:43:59 CEST] <Spring> durandal_1707, https://a.uguu.se/pzzEqQOFmL88_ffplayloopslowdownbug%5BTrailers-Umbrellasof… [16:44:06 CEST] <Spring> ^ video capture of issue [16:50:00 CEST] <durandal_1707> Spring: so it happens only with loop option? [16:51:05 CEST] <Spring> durandal_1707, yes, it doesn't display the slow down upon the first playback [16:53:17 CEST] <durandal_1707> probably some kind of bug, you could use loop filter as alternative [16:53:40 CEST] <durandal_1707> for short durations :) [16:56:47 CEST] <xeche_> jkqxz: So how would I go about creating the surfaces. I mean what does the FFMPEG API actually do for me, where HW acceleration is concerned. At this point it seems like almost nothing. [16:57:10 CEST] <Spring> durandal_1707, does the loop filter require setting its own start point? Just using loop=100 doesn't appear to do anything [16:58:32 CEST] <durandal_1707> start point is frame number, should be 0 [16:59:33 CEST] <the_cuckoo> thanks kepstin and durandal_1707 - yeah - that does the rotation thing nicely in my wrapper - nice that it correctly handles scrubbing/reverse play too [17:01:01 CEST] <durandal_1707> Spring: duration is in frames to, set it to some enough big number [17:01:22 CEST] <durandal_1707> I mean size [17:01:56 CEST] Action: the_cuckoo will need to look into the messages thing tomorrow [17:01:59 CEST] <xeche_> jkqxz: I guess a more fruitful endeavor would be to actually find some example of how QSV / HEVC can be used with FFMpeg. [17:02:03 CEST] <jkqxz> xeche_: Yes, lavc is a pretty thin wrapper here. The useful part is that the AVPacket side of it is in common with lavf, so you can use the encoded data there. The AVFrame side is not currently very useful, though there is more stuff around for it pending merge (lavfi support, notably). [17:02:33 CEST] <xeche_> jkqxz: Alright. So am I right in thinking this is why I need the libmfx wrapper? [17:02:47 CEST] <xeche_> i.e. i need to work with Intel specific structs, types, etc [17:03:13 CEST] <xeche_> http://ffmpeg.org/pipermail/ffmpeg-cvslog/2015-March/088245.html <- closes thing I've found to an example [17:03:15 CEST] <Spring> durandal_1707, thanks a bunch, got it working now. BTW what did you mean 'for short durations', the loop filter isn't ideal for longer loops? [17:03:19 CEST] <jkqxz> To deal with the frames in hardware surfaces, yes. [17:03:45 CEST] <kepstin> Spring: it buffers the entire section of video to loop as raw video in ram, so yeah. [17:03:47 CEST] <durandal_1707> Spring: it keeps everything in memory [17:05:08 CEST] <Spring> hmm, may need to test on some high bitrate 1440p footage to make sure this is going to work. [17:06:15 CEST] <jA_cOp> Hi. I'm using -map_metadata and the ffmetadata file format to add some tags to an OGG/Opus file, and it looks like a single "KEY=VALUE" string is capped at 1023 (perhaps followed by a single byte null terminator to make 1024) bytes? Does anyone know if this is really the case? [17:06:22 CEST] <Spring> almost a GB, welp [17:06:49 CEST] <jkqxz> xeche_: Not sure about the H.265 support. You need to do something with a special plugin for it (for licensing reasons, maybe?), but I don't know how that works at all. [17:07:01 CEST] <jA_cOp> I'm trying to add album art in the METADATA_BLOCK_PICTURE format so this limit is a real blocker :( [17:07:37 CEST] <xeche_> jkqxz: Okay. Thanks for all the help. I'll try to figure something out. :) [17:07:57 CEST] <xeche_> If anybody knows of an example on how to work with QSV and FFMpeg, I'd be very grateful if you let me know. [17:09:24 CEST] <jA_cOp> just found this: https://trac.ffmpeg.org/ticket/4833 I guess that's really the case then... [17:09:29 CEST] <Spring> 1.3GB for 10 loops :/ [17:09:38 CEST] <Spring> I'm fine but I wonder about others [17:09:56 CEST] <furq> jA_cOp: why not use something like mutagen [17:10:07 CEST] <furq> using ffmetadata doesn't seem very robust in general [17:10:36 CEST] <Spring> does the bug tracker get much attention for things like this? May end up submitting it anyway but wondering if it would realistically be noticed. [17:11:16 CEST] <jkqxz> xeche_: Use vaapi instead :P (The libmfx implementation in ffmpeg being named "qsv" is unhelpful, there are other ways to access the quick sync hardware.) [17:11:46 CEST] <furq> jkqxz: i'm pretty sure they said they were on windows [17:12:22 CEST] <jkqxz> Hence the ":P". [17:12:40 CEST] <jA_cOp> furq, the Python library? I was hoping to keep everything in one shell script (and one standalone CLI program to generate the METADATA_BLOCK_PICTURE format for a given file), but thanks, there are few CLI utilities that support adding metadata to OGG/Vorbis so I might use that! So far I've only found kid3-cli to work [17:13:28 CEST] <durandal_1707> Spring: number of loops doesn't really matters, size does [17:14:28 CEST] <durandal_1707> report bug anyway, I might look at it [17:14:30 CEST] <furq> jA_cOp: ogg flac isn't widely supported from what i can tell [17:14:39 CEST] <furq> not even libflac supports adding metadata to ogg flac [17:14:48 CEST] <furq> mutagen supports it though which is nice [17:15:23 CEST] <jA_cOp> Sorry, I meant OGG/Opus the second time too, but yeah, the situation is pretty crappy for Opus as well! [17:15:42 CEST] <kepstin> the mutagen api is pretty nice for writing tiny little python scripts to do things, but they don't really provide a useful command-line tool :/ [17:15:58 CEST] <furq> oh nvm you said opus, not flac [17:17:14 CEST] <xeche_> well, at least I can create a QSV hardware context and assign it to the AVCodecContext [17:17:41 CEST] <xeche_> not that it gets me much further, but there's an av_qsv_alloc_context in the libmfx stuff, somewhere. If anyone's interested. [17:23:08 CEST] <Spring> 'Submission rejected as potential spam' :D [17:23:24 CEST] <Spring> 90.82% probability I'm a bot lol [17:26:56 CEST] <Spring> tried three times to register. Is there any way around this? [17:27:27 CEST] <Spring> on the trac.ffmpeg.org that is [17:28:35 CEST] <durandal_1707> Spring: hmm, with stream_loop only way to do this is in 2 pass [17:28:47 CEST] <durandal_1707> with ffmpeg [17:29:37 CEST] <durandal_1707> You cut relevant part with trim and than use stream loop in resulted file [17:29:55 CEST] <Spring> was using the loop only for the ffplay preview, rather than the video but thanks for the heads up [17:30:08 CEST] <Spring> *rather than the encode [17:30:11 CEST] <durandal_1707> lavfi doesn't supports seeking right now :( [17:41:35 CEST] <Spring> also is stream_loop documented any place? couldn't find it in the filters docs [17:42:01 CEST] <durandal_1707> its ffmpeg input option [17:42:28 CEST] <Spring> oh, right of course [18:20:17 CEST] <Spring> is there any benefit to lower CRF values if the max bitrate is set to less than the CRF would go in the first place? Or should they both be set to roughly the same ballpark quality? [18:22:43 CEST] <Spring> as I see examples of this places where the CRF will be far lower than the max bitrate, which is confusing [18:25:06 CEST] <peg> api usage question: I want to decode from PCM_ALAW and then encode into MP3, the problem is that the frame_size in unknown. How one can do you it ? what is the size chuncks of raw data I should give the decoder ? [18:26:33 CEST] <peg> superdump: any idea ? [18:50:02 CEST] <the_cuckoo> peg: i would have thought the number of samples for mp3 encoding would be known (1152 isn't it?) - when decoding, you just need to fill up a buffer, as soon as you have enough samples to encode, you take them out of the buffer and squirt them into the encoder [19:11:10 CEST] <peg> first I have to fill a AVPacket with data to decode (from PCM_ALAW). [19:11:36 CEST] <peg> and the question is what the size of this packet should be [19:11:59 CEST] <kepstin> Spring: with x264, for the same final file size, a crf encode and a 2-pass vbr encode will be the same quality. 1-pass vbr mode is quite a bit different, and using vbv controls (max bitrate, buffer size) changes things a bit. [19:12:57 CEST] <kepstin> Spring: the use of crf with vbv controls is a bit weird, but it's mostly useless when you're encoding many files and some will be smaller, some larger, but you want to make sure they have a max limit on bitrate. [19:13:13 CEST] <kepstin> s/mostly useless/mostly useful/ [19:14:42 CEST] <kepstin> (are you looking at libvpx encoding? it's a bit different there, but what I said above still mostly applies) [19:16:40 CEST] <kepstin> with libvpx doing vp8, you're required to specify bitrate with crf mode, and it's treated as a "maximum" value, and iirc using 2-pass is recommended even when using crf mode. [19:19:06 CEST] <Spring> yeah, was coming from using vpx usage [19:19:23 CEST] <Spring> I know 2 pass is practically a necessity there [19:20:15 CEST] <Spring> but wondered if it made any real difference to have say CRF 8 with max bitrate of 10M if CRF 8 with no cap is like 32M bitrate [19:21:30 CEST] <kepstin> in that case it should be identical to doing a vbr encode at 10M (there might be some local differences if parts of the video are harder/easier to encode) [19:22:05 CEST] <kepstin> the idea of libvpx's crf mode is "make files that have this bitrate, but make the files smaller if they don't need that much bitrate to reach this min quality level" [19:22:43 CEST] <Spring> is there some ballpark table of equivalent CRF value bitrates? [19:23:12 CEST] <kepstin> i dunno, maybe on average, but it's really completely dependent on video content [19:23:27 CEST] <kepstin> so you have to test on a representitive sample of the videos you plan to encode [19:23:39 CEST] <Spring> I see, best to experiment I suppose. [19:23:51 CEST] <kepstin> (this mode is really best used as a "set it and forget it" option when encoding many videos with the same settings) [19:24:19 CEST] <kepstin> it was designed for use encoding youtube videos, really :) [19:51:17 CEST] <kirked> in a video with a 4:3 SAR & 4:3 FAR, and 16:9 DAR, what's the proper command line parameter to extract a frame using DAR instead of FAR? (frame image is currently squished) [19:51:25 CEST] <the_cuckoo> peg: iirc, you get the duration/number of samples from the avframe which is returned [19:51:55 CEST] <peg> the_cuckoo: how ? [19:52:25 CEST] <the_cuckoo> av_frame_->nb_samples [19:53:47 CEST] <the_cuckoo> you'll also need to know the number of channels, the type of the audio (pcm16 presumably - 2 bytes) and the interleaving [20:07:11 CEST] <the_cuckoo> interleaving is either packed or planar (packed for a stereo source is like the first sample is the first sample of the first channel [call it c0s0], next is c1s0, then c0s1, c1s1, etc - while planar is like c0s0, c0s1, .... c1s0, c1s1, ... etc - if that makes sense :)) [20:08:36 CEST] <the_cuckoo> channels is derived from av_frame->layout (there's a set of enums for mono, stero, 2.1 etc) and you'll need to map those to number of channels [20:09:16 CEST] <Cryp71c> Not familiar w/ ffmpeg, working with some legacy code. Transcoding a video file using this set of options: ` -i /mnt/media/temp/af0c37c4777847699c4591bedbd8ba01/b2898100fb034dd88045a3c731f4e054 -vf "transpose=2, format=yuv420p" -metadata:s:v rotate=0 -codec:v libx264 -f mp4 -y /mnt/media/temp/af0c37c4777847699c4591bedbd8ba01/d6cc50b786594e0687869e6d2af87d5f` [20:09:53 CEST] <Cryp71c> Works alright for small videos, but larger videos that are ~10 minutes long and ~250 mb are taking upwards of 50 minutes to transcode on our servers, seems a bit high? [20:10:33 CEST] <rmoorelxr> its bc of your long pathnames [20:12:32 CEST] <jkqxz> You haven't supplied any options (like bitrate or quality) to the libx264 encode there. I've no idea what the defaults are, but they probably aren't what you want. [20:15:21 CEST] <the_cuckoo> Cryp71c: i'd suggest -threads auto [20:15:40 CEST] <the_cuckoo> but there is probably more tuning you can do [20:16:28 CEST] <bencoh> or just change the preset ;) [20:16:55 CEST] <bencoh> default for x264 is medium iirc (not sure though) ... which can be quite slow [20:17:31 CEST] <furq> -threads auto is the default anyway [20:17:41 CEST] <the_cuckoo> ah - k :) [20:18:11 CEST] <the_cuckoo> hmm - nah - it says it's 1 [20:18:53 CEST] <furq> it's the default for x264 [20:19:00 CEST] <the_cuckoo> ah - gotcha [20:24:34 CEST] <s00b4u> Hi, I am trying to use FATE but not able to [20:25:13 CEST] <s00b4u> When I try this command: make fate SAMPLES=fate-suite/ [20:25:27 CEST] <s00b4u> I get error: "make: *** No rule to make target `fate'. Stop." [20:25:50 CEST] <s00b4u> Can anyone suggest possible reasons [20:25:51 CEST] <s00b4u> Can anyone suggest possible reasons? [20:26:12 CEST] <durandal_1707> from what directory? [20:26:31 CEST] <s00b4u> I tried from top level source directory [20:26:34 CEST] <s00b4u> ffmpeg [20:27:05 CEST] <s00b4u> then also from ffmpeg_sources [20:27:10 CEST] <s00b4u> same error [20:27:16 CEST] <durandal_1707> and how you got source? [20:27:30 CEST] <s00b4u> clone from Github [20:27:33 CEST] <the_cuckoo> have you ran configure already? [20:27:43 CEST] <s00b4u> yes, installation is fine [20:27:51 CEST] <s00b4u> I can transcode videos [20:28:35 CEST] <s00b4u> I think I should mention about the environment on which I am using FFMPEG [20:28:47 CEST] <sofakng> does anybody know what is required to cross-compile d3d11va support on linux? [20:29:07 CEST] <s00b4u> I am using it on "Bash on Ubuntu on Windows" [20:29:18 CEST] <durandal_1707> omg [20:29:36 CEST] <s00b4u> ?? [20:30:23 CEST] <the_cuckoo> i was wondering how well that would work myself :) - kinda cool that you got an ffmpeg build from it [20:32:01 CEST] <durandal_1707> try just : make fate [20:32:01 CEST] <s00b4u> Yes, this a kind of new platform [20:32:25 CEST] <s00b4u> I just tried.. but same error [20:32:48 CEST] <the_cuckoo> grep fate Makefile [20:32:49 CEST] <s00b4u> It's actually feels like "real" linux [20:33:44 CEST] <the_cuckoo> it's sandboxed isn't it? can you start native programs from it? [20:34:21 CEST] <s00b4u> Its a bash shell running on top of windows kernel [20:34:44 CEST] <s00b4u> so, the bash system calls are being translated according to windows kernel [20:35:03 CEST] <durandal_1707> well something is obviously wrong [20:35:41 CEST] <s00b4u> When I try "grep fate makefile", I get "No such directory or file" [20:35:51 CEST] <the_cuckoo> upper case M [20:36:41 CEST] <s00b4u> with upper case also [20:36:44 CEST] <s00b4u> same issue [20:37:21 CEST] <the_cuckoo> guessing your probably need to run configure again - and if that file's not there, then you're in the wrong directory [20:37:36 CEST] <s00b4u> ok, I will try that now [20:37:46 CEST] <Spring> does drawtext require setting a font? It says the default is 'Sans' but I don't have a font by that name installed. [20:38:46 CEST] <bencoh> "so, the bash system calls are being translated according to windows kernel" err [20:40:23 CEST] <Spring> nvm, thought it was the path. [20:42:58 CEST] <s00b4u> @bencoh: https://blogs.msdn.microsoft.com/wsl/2016/04/22/windows-subsystem-for-linux… [20:43:36 CEST] <s00b4u> What I mentioned earlier was like fitting an elephant in a cup [20:44:12 CEST] <s00b4u> but in one line, its actually what i mentioned earlier [20:44:32 CEST] <s00b4u> I quote from the blog: " By placing unmodified Linux binaries in Pico processes we enable Linux system calls to be directed into the Windows kernel." [20:58:32 CEST] <kepstin> Spring: i'm pretty sure the drawtext stuff uses fontconfig, where 'Sans' is set up as an alias that'll pick some reasonable sans-serif font that you have installed. [21:11:29 CEST] <Cryp71c> jkqxz, regarding your suggestion on picking values for bitrate or quality for libx264, this page seems to have info on what I need? https://trac.ffmpeg.org/wiki/Encode/H.264 [21:11:35 CEST] <Cryp71c> wanted to make sure before I read the whole thing [21:11:59 CEST] <furq> you can probably stop reading at the end of the crf section [21:12:03 CEST] <Cryp71c> bencoh, when you had mentioned preset earlier were you referring to some of the default values that are mentioned on that page ^ ? [21:12:28 CEST] <furq> Cryp71c: http://dev.beandog.org/x264_preset_reference.html [21:13:54 CEST] <the_cuckoo> furq: re: threads - wouldn't -threads auto before the -i also apply to the input decode? (i don't use ffmpeg command line that often - more interested in the libs - but i thought that was how it worked anyway) - could benefit Cryp71c if the bottleneck is in the decode? [21:14:34 CEST] <Cryp71c> furq, thanks, are crf / two pass abr mutually exclusive? The phrasing suggests they are two possible rate control modes and you would typically choose between them? [21:14:40 CEST] <furq> i don't think multithreaded decoding works with h264 [21:14:46 CEST] <furq> Cryp71c: yes [21:14:53 CEST] <furq> and two-pass will naturally take twice as long [21:15:03 CEST] <furq> for little or no benefit with x264 [21:15:35 CEST] <Cryp71c> Which one is used by default? I'm comparing each of the two examples with the command this legacy code is using and can't tell? [21:15:41 CEST] <Cryp71c> (cant tell which one we're using) [21:15:46 CEST] <furq> the defaults are -crf 23 -preset medium [21:15:54 CEST] <furq> which are usually pretty reasonable [21:16:02 CEST] <the_cuckoo> furq: yeah - threads work with h264 [21:16:28 CEST] <Cryp71c> Does 10 minutes / 250 Mb taking 50 minutes to transcode sound typical? [21:16:33 CEST] <furq> no [21:16:43 CEST] <furq> what cpu is that [21:16:46 CEST] <Cryp71c> 1 sec [21:16:59 CEST] <kepstin> Cryp71c: without knowing anything about what computer you're using to encode, can't say anything about whether that's typical. [21:17:00 CEST] <furq> also what resolution is the input video [21:19:00 CEST] <furq> kepstin: i would say that's pretty atypical for something that was described as a server [21:19:09 CEST] <furq> unless the input is 4k or something [21:19:40 CEST] <kepstin> hey, "server" doesn't mean much, it could be a 10yr old sparc or something ;) [21:19:50 CEST] <Cryp71c> This is on an Azure VM, getting the specs from our devops guy [21:19:55 CEST] <furq> 10 year old sparcs are pretty atypical though [21:20:05 CEST] <furq> at least i hope so [21:20:20 CEST] <kepstin> hmm, VM. So you're probably dealing with low core count and maybe shared scheduling :/ [21:20:23 CEST] <Cryp71c> and the video is 1920x1080, 30fps [21:20:29 CEST] <furq> is the cpu contended [21:20:39 CEST] <furq> that will obviously slow things down a lot [21:21:57 CEST] <Cryp71c> No its not supposed to be on this tier of VM hosting [21:25:16 CEST] <jookiyaya> BIG NEWS: amd destroys intel http://imgur.com/a/EJlNF [21:25:52 CEST] <rmoorelxr> is that safe to click on [21:26:06 CEST] <furq> christ not this again [21:26:47 CEST] <kepstin> rmoorelxr: unless you'll get in trouble at work for having some meaningless graphs on your screen, yeah. [21:27:47 CEST] <furq> rmoorelxr: it turns out a cpu with hashing extensions is faster at sha1 hashing than some cpus without hashing extensions [21:27:54 CEST] <furq> imagine that [21:27:56 CEST] <rmoorelxr> lol [21:28:49 CEST] <jookiyaya> rmoorelxr yes it is safe [21:28:51 CEST] <jookiyaya> it's just a picture [21:29:38 CEST] <furq> you might as well link a list of cpus ranked by TDP with the 9590 at the top [21:29:40 CEST] <kepstin> as far as I can tell, there's no DRM keys hidden in that picture, so the cops can't pick you up for DMCA violations for downloading it. [21:30:16 CEST] <furq> i know someone who owns a 9590 ;_; [21:30:38 CEST] <furq> the crying face is inaccurate because it's really funny [21:31:23 CEST] <furq> he bought a 9590 and crossfire 7990s and then bought a chinese 500W psu to run it all [21:31:33 CEST] <furq> and then had the audacity to complain when it exploded [21:31:56 CEST] <rmoorelxr> haha and those 500W are really just aspirational [21:43:08 CEST] <iive> jookiyaya: i asked you to stop with that bulshit. [21:43:55 CEST] <iive> furq: actually somebody claimed that the test is doctored, because there are intel cpu with sha1 extension and its speed is like it is using pure software calculation. [21:44:22 CEST] <furq> that wouldn't surprise me [21:44:45 CEST] <furq> although even if it is remarkably fast at sha1, who cares [21:44:45 CEST] <jookiyaya> iive who said that [21:44:52 CEST] <iive> yeh, and that was told to jookiyaya the first time he posted it. [21:45:08 CEST] <iive> and he had posted same link at least 2 times since. [21:45:36 CEST] <iive> so, i'm tempted to ban him as spammer. [21:45:52 CEST] <jookiyaya> iive techincally it's different link [21:46:09 CEST] <jookiyaya> i changed the image site [21:46:12 CEST] <furq> iive: i think he just told you to do it [21:48:20 CEST] <iive> 2016-08-07 14:55:04 (+0300)<jkqxz> jookiyaya: Zen adds the hardware instructions for hashing functions, so obviously it can do them at close to the memory line rate. (Skylake and Goldmont have them too - presumably that benchmark is deliberately not using them on the 6700K to make the results more amusingly misleading.) [21:48:22 CEST] <jkqxz> I thought that was rigged initially, but on further investigation Skylake actually doesn't have the SHA extensions. Apollo Lake (Goldmont, the low power core for cheap devices) will have them very soon, though, and should therefore come at or close to the top of that table. [21:54:14 CEST] <iive> ok, good to know. [22:03:50 CEST] <jookiyaya> they should add extension so it does x265/x265 encoding better [22:04:16 CEST] <jookiyaya> or is that too hard to do [22:32:03 CEST] <kepstin> jookiyaya: there was some stuff in avx2 that helped x264 a bit, but I dunno about x265 [22:32:15 CEST] <jookiyaya> oh [22:32:17 CEST] <jookiyaya> what is avx2 [22:32:41 CEST] <kepstin> of course, the xen 'apu' (integrated graphics) processors will probably include a hardware h265 encoder, but that's completely distinct from x265. [22:33:38 CEST] <jookiyaya> then what software would it help with [22:33:54 CEST] <jookiyaya> if it doesn't assist the x265 [22:40:26 CEST] <furq> kepstin: summit ridge isn't an apu [22:40:50 CEST] <furq> it is also a really wonky name [22:41:15 CEST] <kepstin> sure, but they are gonna release an apu with xen cores at some point. [22:41:46 CEST] <furq> i preferred it when they named their uarches after racetracks [22:42:00 CEST] <kepstin> i liked it back when they named them after horse breeds. [22:43:11 CEST] <furq> i liked it when you could unlock the multiplier with a pencil [22:43:47 CEST] <furq> instead of unlocking the multiplier by paying intel an extra £100 [23:26:20 CEST] <bencc> how to choose a good cpu for transcoding vp8 to h264? [23:26:31 CEST] <bencc> are there features to check? [23:27:10 CEST] <bencc> is it important to have a graphics card or on-board gpu? [23:44:56 CEST] <kepstin> bencc: basically no graphics cards support decoding vp8, so you need cpu for that. And in most cases, using a cpu encoder for h264 will provide better results than a gpu/dedicated hardware encoder [23:45:27 CEST] <kepstin> so the general recommendation is to use a reasonably modern intel chip, ideally with multiple cores, and don't worry about gpu at all. [23:46:38 CEST] <kepstin> hyperthreading seems to help with x264, fwiw. [00:00:00 CEST] --- Wed Aug 10 2016

1 0

ffmpeg-devel.log.20160808
by burek 09 Aug '16

09 Aug '16

[00:50:29 CEST] <cone-654> ffmpeg 03Michael Niedermayer 07master:74314f1f5f9e: avcodec/ffv1: template functions to allow data types different from int16_t [00:50:30 CEST] <cone-654> ffmpeg 03Michael Niedermayer 07master:ce2217b25ecc: avcodec/ffv1: add AV_PIX_FMT_GBRP16 support [06:42:41 CEST] <durandal_1707> how to call filter which injects frames into timestamp gaps? [09:48:55 CEST] <durandal_1707> so how to name it? [10:14:54 CEST] <j-b> 'morning [10:24:40 CEST] <BtbN> durandal_1707, gapdetector? [10:28:03 CEST] <durandal_1707> BtbN: its not detector, its frame/samples injector [10:28:42 CEST] <BtbN> So... gapinjector? [10:29:17 CEST] <nevcairiel> that sounds like it injects gaps [10:29:33 CEST] <BtbN> gapfiller? [10:30:17 CEST] <durandal_1707> injector? :) [10:53:26 CEST] <iive> how about using word "insertion" instead? [10:54:11 CEST] <iive> e.g. gap insertion. [11:57:10 CEST] <BtbN> Trying to create a 170GB mp4 file is apparently not a good idea. [11:57:32 CEST] <Compn> thats a good recipe for OOM [11:57:32 CEST] <BtbN> "Error writing trailer of out.mp4: Cannot allocate memory" [11:57:57 CEST] <bencoh> I wonder how much space you'd need to allocate in memory for the atom :) [11:57:58 CEST] <Compn> do you need to recover it because of missing moov ? [11:58:26 CEST] <BtbN> Well, i still have all the input data to convert again, just a day of re-muxing lost. [11:58:51 CEST] <BtbN> But why does it need so much memory to write the moov atom? [11:58:54 CEST] <durandal_1707> how much memory you have? [11:58:57 CEST] <BtbN> 64GB [11:59:03 CEST] <bencoh> :] [11:59:28 CEST] <bencoh> either it pre-allocates structures, or there is a leak there [12:01:17 CEST] <wbs> BtbN: because it needs to write the byte offset to every single packet in the file [12:01:45 CEST] <BtbN> So mp4 is just pretty much useless for files that huge? [12:02:06 CEST] <nevcairiel> pretty much [12:02:10 CEST] <bencoh> you could try fragmented mp4 if you really need it, but... [12:02:10 CEST] <Mavrik> Yup. [12:02:14 CEST] <wbs> pretty much. if you write fragmented files, they won't have a single index in one location, but you won't need to keep that much in memory [12:02:28 CEST] <BtbN> So lets hope mkv works. [12:02:30 CEST] <nevcairiel> i specialized muxer could probably write it in small parts, but the demuxer might then explode the same way [12:03:09 CEST] <wbs> (in practice it probably only used over 32 GB or so, but when growing some arrays by doubling the size, it might have ran out of memory around there) [12:03:19 CEST] <Mavrik> Consider looking at TS as well, it handles these cases better (although with more overhead). [12:03:35 CEST] <nevcairiel> streaming formats like TS of course dont care a bit about the file size [12:03:36 CEST] <BtbN> I want to upload this to YouTube, so whatever works for that is fine. [12:04:03 CEST] <nevcairiel> although TS might get screwy if your 170GB isnt just huge but also long [12:04:08 CEST] <nevcairiel> wrapping timestamps are annoying [12:04:22 CEST] <BtbN> It's 7 a day long stream. [12:04:24 CEST] <Mavrik> Yeah, those crashes around 26 hrs or so :) [12:04:33 CEST] <bencoh> then definitely wrapping :) [12:05:05 CEST] <bencoh> BtbN: err .... will you be able to upload 170go to youtube anyway? [12:05:15 CEST] <BtbN> I uploaded larger files already. [12:05:18 CEST] <bencoh> oh, okay [12:05:28 CEST] <bencoh> what did you use then? [12:05:33 CEST] <BtbN> flv [12:05:45 CEST] <BtbN> But they weren't this long, only large. [12:07:17 CEST] <rcombs> yeah try matroska [12:07:55 CEST] <nevcairiel> if you wanted to be absolutely safe could turn off the cue index in matroska, but it only writes video keyframes anyway so unless its all intra its probably better [12:08:09 CEST] <rcombs> for 170GB, maybe it is [12:08:17 CEST] <BtbN> Currently this is a mess of 110k 4-8 second long flv files. I wonder if a plain stupid concat of them would work. [12:08:51 CEST] <rcombs> oh over 5 days that's not that high bitrate though, is it [12:09:15 CEST] <BtbN> It was x264 in crf mode, at 20. [12:09:36 CEST] <BtbN> The bitrate varied between less than 1k to over 15k, depending on the content. [12:10:16 CEST] <bencoh> BtbN: depending on how you split those I'd suspect a/v desync, but .... [12:10:43 CEST] <BtbN> It was split by nginx-rtmp, and so far it seems to have done a decent job. [12:10:49 CEST] <bencoh> ah okay [12:10:52 CEST] <bencoh> mux-split then [12:11:01 CEST] <bencoh> not transcoded [12:12:06 CEST] <BtbN> https://www.youtube.com/watch?v=4to9gvRsp2A that's a segment from the very end of it, a/v sync seems flawless. [16:54:02 CEST] <ubitux> hi [16:54:12 CEST] <ubitux> jamrial: Timothy_Gu: thanks for the merges [16:54:30 CEST] <ubitux> michaelni (or anyone else); please do not push the edit list thing from youtube [16:54:52 CEST] <nevcairiel> comment on the ML for better visibility [16:54:56 CEST] <ubitux> i'd like to have a look because it looks like it's using an approach we dismissed already, and it might be causing fundamental issues [16:55:20 CEST] <nevcairiel> just a hint that you want to review it more thoroughly should stop it from being pushed prematurely [16:55:29 CEST] <ubitux> nevcairiel: i haven't look yet, i'm just giving me a delay, 'cause i just went back from holidays and i have a heavy week incoming [16:55:37 CEST] <ubitux> ok [16:57:19 CEST] <ubitux> but i want to test it first before making any comment [16:57:21 CEST] <ubitux> anyway [16:57:50 CEST] <nevcairiel> well you dont have to comment the patch, just that everyone knows you want to review [17:52:04 CEST] <cone-371> ffmpeg 03Burt P 07master:b2b659b17df2: af_hdcd: Add analyze mode [17:59:02 CEST] <durandal_1707> kierank: is there way to detect timestamp gaps? [18:23:56 CEST] <cone-371> ffmpeg 03Burt P 07master:dbd7a84c8141: af_hdcd: Don't warn if converting from AV_SAMPLE_FMT_S16P [18:35:02 CEST] <kierank> durandal_1707: ? [18:36:22 CEST] <durandal_1707> kierank: due to missing frames that do not decode correctly [18:43:31 CEST] <cone-371> ffmpeg 03Hendrik Leppkes 07master:3bf142c77337: cmdutils: remove the current working directory from the DLL search path on win32 [18:43:32 CEST] <cone-371> ffmpeg 03Michael Niedermayer 07master:6aa39080ccea: avcodec/rawdec: Fix palette handling with changing palettes [18:43:33 CEST] <cone-371> ffmpeg 03Michael Niedermayer 07master:65298a192a92: avformat/id3v2: Mark variable as unused to avoid "set but not used" warning [21:13:04 CEST] <omerjerk> Hi [21:13:19 CEST] <omerjerk> Is there any way to get the actual number of samples in an audio frame? [21:13:53 CEST] <omerjerk> For e.g. the last frame might not span the whole frame_size. [21:14:06 CEST] <omerjerk> I'm not sure if I explained the issue properly. :/ [21:15:48 CEST] <BtbN> divide the size by sample size? [21:16:03 CEST] <omerjerk> yeah so how will I get to know the sample size? [21:18:48 CEST] <omerjerk> I need to know the total number of samples in the whole audio file. [21:18:56 CEST] <omerjerk> That'll solve the whole problem. [21:20:46 CEST] <omerjerk> or the number of remaining samples may be? i'm not sure. [21:26:05 CEST] <BtbN> you know the format, so you know the size of a single sample. [21:31:01 CEST] <omerjerk> yes i do know the size of a single sample [21:31:09 CEST] <omerjerk> but that's not what I want. [21:31:53 CEST] <omerjerk> the last frame is not supposed to always have avctx->frame_size samples in it. [21:32:38 CEST] <omerjerk> so I need to know the actual samples left/total number of samples/actual number of samples in the current frame. [21:32:42 CEST] <omerjerk> Anything will do. [21:34:50 CEST] <BtbN> So you have an AVFrame, and want to know how many samples are in there? [21:36:32 CEST] <omerjerk> yes [21:36:42 CEST] <BtbN> And what's wrong with nb_samples? [21:36:58 CEST] <omerjerk> oh. let me see. :| [21:37:23 CEST] <omerjerk> I completely forgot about that. [21:37:28 CEST] <omerjerk> let me just test. [21:42:40 CEST] <cone-371> ffmpeg 03Michael Niedermayer 07release/3.1:a75a7feebd42: avformat/mov: Enable mp3 parsing if a packet needs it [21:42:41 CEST] <cone-371> ffmpeg 03Michael Niedermayer 07release/3.1:e160064d39d5: avcodec/raw: Fix decoding of ilacetest.mov [21:42:42 CEST] <cone-371> ffmpeg 03Michael Niedermayer 07release/3.1:19d2921bbfec: avcodec/rawdec: Fix palette handling with changing palettes [21:42:43 CEST] <cone-371> ffmpeg 03Hendrik Leppkes 07release/3.1:9745c5ebf873: cmdutils: remove the current working directory from the DLL search path on win32 [21:43:12 CEST] <omerjerk> BtbN, thanks a lot. It worked. :) [22:16:15 CEST] <kevr> Hello. [22:16:40 CEST] <kevr> I'm trying to extract an image from a frame via libav in C, is there a good resource of examples i could read up on? [22:17:04 CEST] <cone-371> ffmpeg 03Michael Niedermayer 07release/3.1:4275b27a2300: Update for 3.1.2 [22:52:44 CEST] <Chloe> why does af_atempo.c have a 0.5 to 2.0 limit on the tempo change? [22:59:58 CEST] <durandal_1707> because of underlaying algo [23:40:27 CEST] <cone-371> ffmpeg 03Derek Buitenhuis 07master:26695aedc200: docs/filters: Fix parameter names for colorspace filter [23:55:13 CEST] <iive> where this commit came from? [23:55:43 CEST] <durandal_1707> parallel universe [23:58:25 CEST] <iive> i don't think these actually exists [23:59:37 CEST] <durandal_1707> what you need? [00:00:00 CEST] --- Tue Aug 9 2016

1 0

ffmpeg.log.20160808
by burek 09 Aug '16

09 Aug '16

[00:01:28 CEST] <DHE> Prelude2004c: you might be able to use `perf` to benchmark what's taking so long... are you overloading the GPU? (I'm guessing you're using one) [00:02:39 CEST] <DHE> if it's overloading the GPU, perf won't help much. if you overload the CPU this might find out why [00:05:04 CEST] <durandal_1707> Prelude2004c: show full command and output [00:12:44 CEST] <Prelude2004c> ehy guys.. sorry for the delay [00:12:48 CEST] <Prelude2004c> overloading GPU no,, 60% [00:14:36 CEST] <Prelude2004c> here is everything.. command is at the bottom [00:14:38 CEST] <Prelude2004c> http://pastebin.com/raw/txA4cMsP [00:15:05 CEST] <Prelude2004c> the -map option '[vout]' kills it somehow [00:31:03 CEST] <durandal_1707> Versions differs somehow, compiled ffmpeg vs linked libs [00:32:12 CEST] <durandal_1707> also make sure yasm is enabled [00:32:55 CEST] <durandal_1707> yadif and scale is otherwise slow [00:44:50 CEST] <Prelude2004c> ya yasm is on [00:44:56 CEST] <Prelude2004c> looks like format=nv12 is a problem [00:45:00 CEST] <Prelude2004c> when i use format=yuv420p it works bettter [01:11:51 CEST] <devster31> hi, I'm trying to skip the first half a second of a clip, this is my command: ffmpeg -ss 00:00:00.50 -i in.mp4 -vcodec copy -acodec copy out.mp4 but the output is identical to the original [01:12:54 CEST] <thebombzen> devster31: you can't truncate the video with -codec copy [01:13:15 CEST] <thebombzen> if it copies the stream then it won't reencode it, so you'll get the same thing [01:13:55 CEST] <devster31> ok, so what should I use? I looked here for the command: https://trac.ffmpeg.org/wiki/Seeking [02:53:22 CEST] <jookiyaya> big news: amd destroys intel http://static2.gamespot.com/uploads/scale_super/460/4601080/3046688-8311357… [03:13:14 CEST] <wallbroken> x265 has the same compatibility as x264 ? [03:14:16 CEST] <DHE> jookiyaya: third time it's been mentioned [03:14:33 CEST] <DHE> wallbroken: ? [03:14:45 CEST] <jookiyaya> DHE i see, 7 more times to go [03:17:03 CEST] <wallbroken> DHE, i don't know if is better to encode a video in x264 or x267 [03:17:07 CEST] <wallbroken> *x265 [03:17:54 CEST] <jookiyaya> use x265 [03:17:59 CEST] <jookiyaya> better encoder [03:18:17 CEST] <wallbroken> how to enable it in ffmpeg? [03:18:33 CEST] <jookiyaya> i use handbrake so i don't know [03:18:38 CEST] <DHE> they're different codecs. you can't compare them [03:18:53 CEST] <wallbroken> DHE, different in what? [03:18:54 CEST] <DHE> that is, you can't really compare x264 and x265, they make incompatible bitstreams [03:19:07 CEST] <DHE> anymore than you can compare Snow and XviD [03:19:11 CEST] <jookiyaya> use x265 ; it's better [03:19:28 CEST] <jookiyaya> xvid is old and sucks [03:19:43 CEST] <wallbroken> ffmpeg does not support x265 natively [03:19:44 CEST] <wallbroken> why? [03:19:52 CEST] <jookiyaya> i thought it did [03:19:56 CEST] <DHE> xvid, h264, h265 are like a sequence of generations [03:20:14 CEST] <DHE> wallbroken: not natively for encoding. x265 is a third party library that provides an encoder ffmpeg uses [03:20:21 CEST] <DHE> there is a native decoder [03:22:23 CEST] <wallbroken> why of third party? [03:33:55 CEST] <jookiyaya> wallbroken good question, no diea [05:06:19 CEST] <wallbroken> lower crf means bigger file? [05:12:55 CEST] <kepstin> lower crf means higher quality, which assuming you're using the same settings elsewhere (e.g. preset) means bigger file, yes. [06:01:46 CEST] <k_sze[work]> Is there a way to tell ffmpeg to not silently skip over corrupt frames? [06:02:50 CEST] <k_sze[work]> I have seen this happen with some .nut files that contain corrupt frames (because they were written over an SMB connection and the ethernet connection was unstable) [06:03:15 CEST] <k_sze[work]> Instead of silently skipping over corrupt frames, I would like ffmpeg to give me a black frame. [06:03:22 CEST] <k_sze[work]> (or any kind of blank frame) [06:21:12 CEST] <durandal_1707> k_sze[work]: sounds like filter task [06:22:12 CEST] <durandal_1707> is it CFR or VFR? [06:22:37 CEST] <k_sze[work]> constant frame rate [06:26:32 CEST] <Anonaly> hi, I need some help please [06:27:35 CEST] <Anonaly> I have recovered files from a nearly dead hard drive, and the recovery software was able to recover a pile of .mov files. [06:27:55 CEST] <Anonaly> However the .mov files do not play in any video player (mplayer, vlc, windows media player) [06:28:04 CEST] <Anonaly> havent tried quicktime yet [06:28:46 CEST] <Anonaly> ffplay tells me this [06:28:55 CEST] <Anonaly> moov atom not found0/0 [06:28:55 CEST] <Anonaly> ./LostFile_Mov_9865114.mov: Invalid data found when processing input [06:29:47 CEST] <durandal_1707> there's utility to readd missing moov atom [06:30:09 CEST] <durandal_1707> with varying success [06:30:12 CEST] <Anonaly> ok thats what led me here ! [06:30:29 CEST] <Anonaly> apparently ffmpeg can do it, and I have the latest version installed [06:31:09 CEST] <durandal_1707> nope, ffmpeg doesn't recreate missing moov atom [06:31:14 CEST] <furq> Anonaly: http://vcg.isti.cnr.it/~ponchio/untrunc.php [06:31:26 CEST] <furq> that's the only oss tool i know of [06:32:09 CEST] <furq> you'll need a working video from the same source [06:32:22 CEST] <Anonaly> yeah thats what this other program I saw required [06:32:48 CEST] <Anonaly> you have to pay for the software and then you need a video from the same recording device [06:33:01 CEST] <Anonaly> I don't have that video recorder, the owner of the HD does, [06:33:06 CEST] <Anonaly> I guess he will have to do that [06:33:24 CEST] <Anonaly> or get him to make a small video and send it to me [06:34:26 CEST] <Anonaly> http://www.remorepair.com/windows/hd-video-repair-utility.html [06:34:31 CEST] <Anonaly> that one [06:34:40 CEST] <furq> http://slydiman.me/eng/mmedia/recover_mp4.htm [06:34:42 CEST] <furq> there's that as well [06:34:47 CEST] <furq> which has presets for common cameras [06:35:10 CEST] <Anonaly> ok but that is for mp4 [06:35:14 CEST] <Anonaly> this is .mov [06:35:20 CEST] <furq> This utility can recover video and audio streams from unfinalized MP4/MOV/3GP files without (or empty) header. [06:35:28 CEST] <furq> mp4 and mov are more or less the same [06:35:32 CEST] <Anonaly> ok [06:35:41 CEST] <Anonaly> thanks [06:35:46 CEST] <Anonaly> ! [07:28:17 CEST] <Anonaly> furq, that link you gave me I don't see how it has presets for common cameras [07:28:22 CEST] <Anonaly> http://slydiman.me/eng/mmedia/recover_mp4.htm [07:28:37 CEST] <Anonaly> or I don't know how to use them [07:28:42 CEST] <Anonaly> one or the other [07:29:12 CEST] <Anonaly> when I tried it, it wanted a good condition video file from the camcorder [07:32:32 CEST] <Anonaly> furq, I am blind sorry [07:32:37 CEST] <Anonaly> forget all that [09:47:07 CEST] <k_sze[work]> hmm, looks like I can't specify a negative angle with -metadata:s:v:0 rotate [09:48:04 CEST] <durandal_1707> you don't need negative angle [09:48:11 CEST] <k_sze[work]> yes, I realize that.\ [09:48:33 CEST] <k_sze[work]> I had to retry a few times to understand what was going on. [09:49:14 CEST] <k_sze[work]> ffmpeg didn't give me an error message for negative angle though. It would have helped if it did. [10:18:17 CEST] <Spring> is there any way of setting a minimum ffplay window size? I only see the ability to set a forced window size [10:18:34 CEST] <Spring> that way the titlebar isn't crunched into itself at smaller sizes [10:18:39 CEST] <Spring> *wouldn't be [10:24:53 CEST] <Spring> also a shame there's no -to option, since that would allow trim values to be previewed [10:56:26 CEST] <henrikx> Hello [10:56:28 CEST] <henrikx> ? [10:56:45 CEST] <henrikx> Could I have a little bit of help? [10:57:14 CEST] <henrikx> (command used at bottom) http://puu.sh/qtLxZ/74fb900cbb.png [10:57:20 CEST] <henrikx> 'nuff said [10:57:23 CEST] <henrikx> what am I doing wrong? [11:10:53 CEST] <furq> henrikx: get rid of -f h264 [11:19:35 CEST] <henrikx> @furq Now this http://puu.sh/qtMlW/ffa3ebd849.png [11:21:37 CEST] <xochilpili> hi all [11:21:45 CEST] <henrikx> yo [11:21:47 CEST] <xochilpili> how can i install ffmpeg in freebsd? [11:21:56 CEST] <furq> pkg install ffmpeg [11:22:10 CEST] <xochilpili> i mean, i know how to, but my question is something like: full codecs support [11:22:26 CEST] <xochilpili> reading this : http://dduvnjak.pw/ffmpeg-install-on-freebsd/ [11:23:11 CEST] <furq> that's more or less how you do it [11:23:34 CEST] <Mandevil> henrikx: Resample the audio with -ra [11:23:36 CEST] <furq> although if you don't have the pkgng version installed then that'll compile all the dependencies, which you probably don't want [11:23:41 CEST] <furq> henrikx: -c:a aac [11:23:42 CEST] <xochilpili> but i cant compile theora [11:23:53 CEST] <furq> there's no reason to do that separately [11:24:08 CEST] <furq> libtheora is packaged [11:24:10 CEST] <xochilpili> furq, so just directly to ffmpeg port ? [11:24:19 CEST] <furq> right [11:24:25 CEST] <furq> there's no reason to use a different version of gcc either [11:24:35 CEST] <furq> and if you do then 4.6 isn't the newest any more [11:24:36 CEST] <xochilpili> ffmpeg26/ ? [11:24:45 CEST] <furq> /usr/ports/multimedia/ffmpeg [11:24:49 CEST] <henrikx> @furq whats the last message u sent? [11:24:51 CEST] <furq> it's stuck on 2.8.6 though [11:24:59 CEST] <furq> henrikx: add -c:a aac before the output file [11:25:05 CEST] <henrikx> ok [11:25:31 CEST] <xochilpili> furq, i have ffmpeg26 and i have portsnap updated [11:25:49 CEST] <xochilpili> is that correct? [11:26:09 CEST] <henrikx> Thanks! now it works [11:26:17 CEST] <furq> no [11:26:21 CEST] <furq> ffmpeg26 is an old version [11:26:35 CEST] <furq> http://vpaste.net/guHzS [11:28:10 CEST] <xochilpili> furq, what deps do i need ? [11:28:23 CEST] <furq> the line below "make config" installs all the deps [11:28:24 CEST] <henrikx> is there any way to pause the playback? [11:28:42 CEST] <xochilpili> furq, i just want to use this server as a transcoder using PHP-FFMpeg framework [11:29:07 CEST] <xochilpili> in order to use html video tag, ogg, mp4 and webm file formats [11:29:09 CEST] <furq> the port makefile will automatically install all the dependencies from source, which takes forever [11:29:35 CEST] <furq> it's faster to run make build-depends-list and then install all those with pkgng [11:29:39 CEST] <furq> which is what that script does [11:31:03 CEST] <xochilpili> furq, the script show me the ncurse window to select the deps; is that correct? [11:31:07 CEST] <furq> yes [11:31:12 CEST] <furq> that's what make config is [11:31:22 CEST] <xochilpili> then, what should i have to select ? [11:31:28 CEST] Action: xochilpili :P [11:31:29 CEST] <furq> whichever codecs you want [11:31:45 CEST] <furq> presumably theora, x264, vpx and fdk-aac [11:31:52 CEST] <xochilpili> wait, codecs for input or codecs for output? [11:31:55 CEST] <furq> output [11:31:58 CEST] <xochilpili> oks [11:32:01 CEST] <furq> oh and opus as well [11:32:05 CEST] <furq> and vorbis [11:32:13 CEST] <furq> i'm pretty sure those are all selected by default though [11:32:17 CEST] <furq> except fdk-aac [11:33:15 CEST] <xochilpili> yeah, and opus [11:33:27 CEST] <xochilpili> fdk-acc and opus aren't by default [11:34:48 CEST] <xochilpili> so, this is installing binaries, but that's not recommended by official doc, right? [11:34:58 CEST] <furq> by which official doc [11:35:11 CEST] <henrikx> @furq is there any playback control in ffmpeg [11:35:18 CEST] <henrikx> ? [11:35:22 CEST] <furq> not that i know of [11:35:28 CEST] <henrikx> hm [11:35:37 CEST] <xochilpili> furq, never mind... [11:36:55 CEST] <xochilpili> furq, now it's installed [11:37:02 CEST] <xochilpili> ffmpeg -protocols ? [11:37:19 CEST] <henrikx> @furq can you start it at a timestep? [11:37:24 CEST] <henrikx> stamp* [11:37:28 CEST] <furq> -t 01:23:$5 [11:37:33 CEST] <furq> :45 [11:37:38 CEST] <xochilpili> furq, http://vpaste.net/zQvYu [11:37:56 CEST] <furq> ? [11:38:09 CEST] <xochilpili> furq, what? what's wrong? [11:38:14 CEST] <furq> i don't know [11:38:20 CEST] <furq> i thought you were telling me something was wrong [11:38:47 CEST] <xochilpili> no, i just put ffmpeg -protocols > vpaste [11:39:01 CEST] <bencoh> codecs show in -codecs [11:41:30 CEST] <henrikx> @furq oh thanks [13:04:03 CEST] <wallbroken> furq, is true that ffmpeg for x265 uses a third party library for encoding? [13:05:20 CEST] <ritsuka> yes, it uses x265 [13:05:24 CEST] <Mavrik> It uses x265 library for encoding :P [13:05:37 CEST] <ritsuka> the format is called h.265 or hevc [13:05:56 CEST] <wallbroken> but for x264, ffmpeg has is native library [13:06:06 CEST] <furq> no [13:06:06 CEST] <ritsuka> no [13:06:08 CEST] <bencoh> no [13:06:10 CEST] <furq> no [13:06:12 CEST] <ritsuka> no [13:06:23 CEST] <furq> yes [13:06:23 CEST] <Mavrik> x264 is also a native library [13:06:30 CEST] <Mavrik> neither x265 or x264 are formats :P [13:07:08 CEST] <Guest_83747> Allah is doing [13:07:10 CEST] <furq> did you not believe DHE when he told you this eight hours ago [13:07:14 CEST] <Guest_83747> sun is not doing Allah is doing [13:07:20 CEST] <Guest_83747> moon is not doing Allah is doing [13:07:22 CEST] <furq> oh hey allah is doing guy is back [13:07:27 CEST] <ritsuka> sigh& [13:07:39 CEST] <furq> he must have made it through every channel on freenode [13:13:29 CEST] <wallbroken> why to build in x265 i need another version of ffmpeg? [13:14:11 CEST] <iive> wallbroken: it might need it for the command line program. if you need only the library, you should be able to disable it. [13:17:06 CEST] <furq> aren't you on debian [13:17:23 CEST] <iive> Guest_83747: if you think that somebody would convert or recognize yours god glorry, just because you spam his name here, you are complete idiot. Quite the contrary. [13:17:25 CEST] <wallbroken> i'm on windows 10 [13:17:37 CEST] <xochilpili> furq, remember me? [13:17:41 CEST] <xochilpili> ffmpeg -y -i /media/Movies/Che.el.Argentino.avi -threads 12 -vcodec libx264 -acodec libfaac -b:v 1000k -refs 6 -coder 1 -sc_threshold 40 -flags +loop -me_range 16 -subq 7 -i_qfactor 0.71 -qcomp 0.6 -qdiff 4 -trellis 1 -b:a 128k -pass 1 -passlogfile /tmp/ffmpeg-passes57a869adf2f22cnoki/pass-57a869adf2ffe tmp/export-webm.mp4 [13:17:47 CEST] <xochilpili> Unknown encoder 'libfaac' [13:17:50 CEST] <xochilpili> which means? [13:17:53 CEST] <iive> Guest_83747: if you want to give bad name of your religion and spread hate, then you are doing great. [13:17:59 CEST] <furq> you built with libfdk-aac [13:18:01 CEST] <xochilpili> i remember that we install this faac right ? [13:18:04 CEST] <furq> no [13:18:10 CEST] <furq> faac sucks [13:18:20 CEST] <xochilpili> damn [13:18:29 CEST] <xochilpili> im using this https://github.com/PHP-FFMpeg/PHP-FFMpeg [13:18:38 CEST] <furq> -c:a libfdk_aac [13:18:39 CEST] <xochilpili> and this options are default [13:19:00 CEST] <xochilpili> let me change that in the framework! [13:19:07 CEST] <furq> most of those defaults seem worthless or bad [13:19:17 CEST] <xochilpili> really? [13:19:29 CEST] <xochilpili> damn [13:19:31 CEST] <furq> most of them are the defaults for -preset medium [13:19:38 CEST] <furq> which is the default preset anyway [13:19:50 CEST] <xochilpili> do you know some php framework using ffmpeg that you recommend ? [13:20:01 CEST] <furq> i assume either this predates -preset or it's just an accreation of cargo-cult stackoverflow answers [13:20:07 CEST] <furq> accretion [13:20:20 CEST] <furq> and no, i try to pretend that php doesn't exist [13:20:34 CEST] <xochilpili> lol [13:20:43 CEST] <xochilpili> why? [13:21:04 CEST] <bencoh> :] [13:21:19 CEST] <furq> otherwise i have vivid nightmares about rasmus lerdorf and my primary school headmistress chasing me in a go-kart [13:21:32 CEST] <bencoh> :D [13:24:41 CEST] <furq> aww [13:24:44 CEST] <furq> he's my favourite spammer [13:24:57 CEST] <furq> he's much better than that guy who posts holocaust denier videos [13:25:39 CEST] <xochilpili> furq, how can i encode a video to use it with ios ? [13:25:44 CEST] <xochilpili> i mean which codec ? [13:25:47 CEST] <Spring> can the trim filter be set in ffplay? Tried -vf "trim=value:value" but it didn't work [13:25:48 CEST] <furq> x264 [13:26:13 CEST] <furq> if you need support for really old iOS devices then you need to use -profile baseline [13:26:33 CEST] <xochilpili> libx264 and libfdk-aac ? [13:26:36 CEST] <furq> yeah [13:26:49 CEST] <furq> https://developer.apple.com/library/ios/documentation/NetworkingInternet/Co… [13:27:22 CEST] <xochilpili> Audio: aac (LC) Video: h264 [13:27:34 CEST] <xochilpili> i have this video converted and i cant see it on ios [13:27:48 CEST] <xochilpili> i have no older device, iphone se [13:28:20 CEST] <xochilpili> oh! [13:28:23 CEST] <xochilpili> it works!!! [13:28:28 CEST] <furq> you're welcome [13:28:29 CEST] <xochilpili> god damn! [13:28:34 CEST] <xochilpili> thanks a lot furq :D :D :D [13:29:36 CEST] <wallbroken> not clear about x265 [13:29:49 CEST] <wallbroken> why, to encode in x265 i need another version of ffmpeg? [13:30:35 CEST] <bencoh> x265 (the commandline tool) might use ffmpeg (or rather the libav* libraries) to mux its output [13:30:47 CEST] <bencoh> and it might depend on a recent version [13:30:54 CEST] <furq> are you asking why you need ffmpeg to compile x265 or why your ffmpeg binary doesn't support x265 [13:32:25 CEST] <furq> or are you asking why you need to rebuild ffmpeg to add support for an encoder [13:33:06 CEST] <wallbroken> furq, to encode in x264 i just download ffmpeg zaranoe, and give the right command [13:33:16 CEST] <wallbroken> wy to encode in x265 that's not possible? [13:33:20 CEST] <wallbroken> that's all [13:33:25 CEST] <xochilpili> furq, do you know how can i ffmpeg srt to webvTT ? [13:33:44 CEST] <furq> wallbroken: it is possible [13:33:54 CEST] <furq> the zeranoe builds have x265 support [13:34:08 CEST] <furq> xochilpili: presumably -i src.srt -c:s webvtt out.vtt [13:34:21 CEST] <furq> i'm not sure if you even need -c:s webvtt [13:34:36 CEST] <xochilpili> furq, thanks! checking [13:35:19 CEST] <wallbroken> furq, my command for x264 is: ffmpeg.exe -i input.vob -b:a 128k out.mp4 [13:35:30 CEST] <wallbroken> for x265 how to modify it? [13:35:35 CEST] <furq> -c:v libx265 [13:35:48 CEST] <wallbroken> just add it? [13:36:02 CEST] <wallbroken> ffmpeg.exe -i input.vob -c:v libx265 -b:a 128k out.mp4 [13:36:10 CEST] <furq> yes [13:36:20 CEST] <wallbroken> and it's compatible with apple devices? [13:36:23 CEST] <wallbroken> like ipad ? [13:36:26 CEST] <furq> no [13:36:27 CEST] <bencoh> ... [13:36:34 CEST] <wallbroken> mmm why? [13:36:40 CEST] <furq> do i look like tim cook [13:38:56 CEST] <xochilpili> furq, if im converting a video from avi to mp4 using those codecs, and i have the srt for that original video, then webvtt will match with the movie? [13:39:05 CEST] <furq> i should hope so [13:39:25 CEST] <xochilpili> why wouldnt ? [13:40:08 CEST] <furq> presumably it'll just copy the timestamps from the srt [13:40:20 CEST] <furq> so if you're filtering the video in some way that changes the length, that'll screw it up [15:29:06 CEST] <xochilpili> how can i know clip duration with ffmpeg? [15:30:02 CEST] <xochilpili> ok, ffprobe [15:39:20 CEST] <rmoorelxr> if im receiving a stream of "h264" bytes (from third party so using air-quotes) can i somehow pipe these to ffmpeg and have it create a stream? [15:41:18 CEST] <furq> rmoorelxr: -f h264 -i pipe: [15:43:49 CEST] <rmoorelxr> ok cool. so what kind of pipe will do this? no experience with this stuff [15:44:41 CEST] <furq> that will take input on stdin [15:44:47 CEST] <furq> so mysource | ffmpeg -i pipe: [15:46:02 CEST] <rmoorelxr> great thanks [15:46:37 CEST] <BtbN> a raw h264 stream is lacking some important information though, like a framerate/timestamp for example. [15:53:36 CEST] <xochilpili> furq, is it possible to stream http with ffmpeg ? [15:53:50 CEST] <rmoorelxr> i get a buffer, a length, a frametype, and a frameStamp which im assuming is the timestamp [15:53:54 CEST] <xochilpili> in order to catch video from <video html5 tag ? [15:54:27 CEST] <furq> xochilpili: https://www.ffmpeg.org/ffmpeg-formats.html#hls-1 [15:54:47 CEST] <furq> and https://github.com/dailymotion/hls.js/ [16:03:04 CEST] <Spring> hmm, well that seemed to resolve itself, the trim filter for ffplay is working now [16:28:53 CEST] <xochilpili> furq, thanks! [17:14:06 CEST] <henrikx> what does this mean? [17:14:06 CEST] <henrikx> [flv @ 04e27900] Failed to update header with correct duration. [flv @ 04e27900] Failed to update header with correct filesize. [17:14:24 CEST] <henrikx> How to fix? [17:14:48 CEST] <henrikx> need to fix? [17:40:45 CEST] <Pixel> Hi, question: ffmpeg support only srt subtitile ? or txt is to supported ? [17:50:02 CEST] <c_14> ffmpeg supports srt, webvtt, ass, mov_text and other formats [17:59:06 CEST] <Pixel> other formats that txt ? [17:59:54 CEST] <c_14> The ones I listed are all text formats [18:13:46 CEST] <Pixel> ok thanks [20:03:27 CEST] <greasemonkey_> Hi! Can someone please help me with ffserver and a h264 transcode conf, perhaps a link to a tutorial? [20:10:24 CEST] <htoetl> how to add audio noise to a video in ffmpeg2 ? [20:11:25 CEST] <henrikx> http://puu.sh/qubbL/aacf5336fb.png as u can see it says speed 1.3x. How do I make it just 1? [20:11:44 CEST] <Mavrik> greasemonkey_, ffserver is dead and unmaintained, use some other streaming server. [20:11:51 CEST] <furq> henrikx: add -re before -i [20:12:04 CEST] <henrikx> thanks [20:13:39 CEST] <ahoo> may i ask a question regarding the windows binaries?+ [20:14:05 CEST] <ahoo> i'll just do it [20:14:15 CEST] <ahoo> so [20:14:19 CEST] <ahoo> i want to grab my screen [20:14:33 CEST] <ahoo> according to the wiki the command line is <what it is> [20:14:41 CEST] <ahoo> i'm getting an error: [20:14:43 CEST] <ahoo> Could not enumerate video devices (or none found). [20:14:51 CEST] <ahoo> i understand i have to install a codec first [20:14:55 CEST] <ahoo> ...for this to work [20:15:10 CEST] <ahoo> is that correct? what would be an x264 codec that works with ffmpeg? [20:15:19 CEST] <furq> are you using -f dshow [20:15:22 CEST] <ahoo> yes [20:15:30 CEST] <furq> https://github.com/rdp/screen-capture-recorder-to-video-windows-free [20:15:35 CEST] <furq> you need to install that (or something similar) [20:16:01 CEST] <furq> alternatively you can use -f gdigrab [20:16:16 CEST] <furq> dshow is faster iirc [20:34:08 CEST] <ahoo> furq: are you a wizard? [20:34:12 CEST] <ahoo> furq: tyvm [20:47:50 CEST] <greasemonkey_> Mavrik, what do you suggest? OBS [20:53:45 CEST] <squarebracket> is there a way to print the AFD information for a frame? I've only been able to figure out how to print that the information is present in the side_data [20:56:54 CEST] <markizano> what's the difference between framerate and fps ? (I'm thinking velocity vs speed?) [20:58:05 CEST] <BtbN> the length of the word. [20:58:05 CEST] <furq> they're the same thing [20:58:59 CEST] <markizano> >.< [20:59:12 CEST] Action: markizano sed -r 's/-vf framerate/-vf fps/g' [21:01:26 CEST] <furq> the difference between the filters is mentioned in the docs [21:01:34 CEST] <furq> https://ffmpeg.org/ffmpeg-filters.html#fps-1 and https://ffmpeg.org/ffmpeg-filters.html#framerate [21:02:07 CEST] <BlackThorneDK> hi, does anyone here know, if ffmpeg can be made tolerant of 'Dequeued v4l2 buffer contains xx bytes, but yy were expected.'. ffplay seems to just keep streaming, but ffmpeg stops recording after a glitch... [21:03:47 CEST] <BlackThorneDK> I got two Magewell XI100DUSB-HDMI connected, but it dequeues when I switch HDMI input source on the fly. I would like to use ffmpeg for both mixing the video feeds and streaming on to other sources, while saving to disk. It works, but only if I don't change the source while streaming. [21:05:43 CEST] <BtbN> ffmpeg and ffplay use the exact same code for handling v4l2. [21:06:46 CEST] <BlackThorneDK> yeah, but the code returns an error, when a partial frame is recieved. ffplay just prints the error, and keeps on going. ffmpeg freezes the stream, and continues to grap audio only [21:43:25 CEST] <rmoorelxr> is there any hope of extracting usable video from a "raw h264" byte stream [21:43:59 CEST] <rmoorelxr> i get frame timestamps independently [22:16:02 CEST] <kevr> Herro. [22:16:31 CEST] <Fjorgynn_> je [22:22:21 CEST] <rmoorelxr> is there such a thing as an INFO frame [22:22:29 CEST] <rmoorelxr> or is that some third party weirdness [22:42:53 CEST] <pgorley_> hi, i'm using ffmpeg 2.8 from the ubuntu 16.04 repos [22:43:17 CEST] <pgorley_> i set AVCodecContext->get_format to my own function, but it's never called [22:43:47 CEST] <pgorley_> any tips on what i could do to figure out why that is? [23:24:09 CEST] <Dresk|Dev> So we're trying to build ffmpeg natively with MSVC2015 so we can avoid using the dreaded MSVCRT.dll, and link against a true modern CRT, but obviously we lose all inline ASM and ASM files. Has anyone had success building on Win32/64 with YASM and MSVC for full optimizations and native runtimes? [23:30:57 CEST] <Bray90820_> Can someone tell me what this error means? [23:30:57 CEST] <Bray90820_> http://pastebin.com/raw/A8AV7MrQ [23:33:28 CEST] <Mavrik> bray90820, you're trying to put WMA video into MP4 container. [23:33:37 CEST] <Mavrik> MP4 container does not support WMA video streams. [23:33:46 CEST] <bray90820> Aaahhh it's the Stupid WMA error again [23:34:06 CEST] <Mavrik> I'm guessing you want to convert to something actually playable by most players :P [23:34:12 CEST] <Mavrik> So you'll have to reencode both video and audio. [23:34:24 CEST] <bray90820> What would be the code to do that? [23:34:36 CEST] <Mavrik> remove "-c copy" [23:34:56 CEST] <Mavrik> replace it with "-codec:v libx264 -codec:a aac -b:a 128k" [23:34:59 CEST] <Mavrik> should be close enough. [23:35:35 CEST] <bray90820> Thanks that seems to be working [23:35:54 CEST] <bray90820> How could you tell it was WMA [23:36:19 CEST] <Mavrik> It says so in the output :P [23:36:33 CEST] <Mavrik> After Input #0 it prints what the input file has: [23:36:34 CEST] <Mavrik> Stream #0:0(eng): Audio: wmapro (b[1][0][0] / 0x0162), 44100 Hz, stereo, fltp, 128 kb/s [23:36:34 CEST] <Mavrik> Stream #0:1(eng): Video: wmv3 (Main) (WMV3 / 0x33564D57), yuv420p, 352x288, 722 kb/s, 25 fps, 25 tbr, 1k tbn, 1k tbc [23:36:49 CEST] <Mavrik> And after Output #0 it prints what you're trying to create :) [23:37:15 CEST] <Mavrik> Also, it's actually WMV video and WMA audio :) [23:37:27 CEST] <bray90820> Aaahhh thanks [23:38:07 CEST] <bray90820> Is there any way to search for whatever videos in a directory have WMA audio? [23:39:55 CEST] <llogan> bray90820: ffprobe -loglevel error -select_streams a -show_entries stream=codec_name -of default=nw=1 input.wmv [23:40:18 CEST] <llogan> will show wmav2 or similar [23:40:56 CEST] <llogan> key word being "wma" [23:41:20 CEST] <llogan> or codec_long_name if you prefer "Windows Media Audio foo" [23:42:56 CEST] <bray90820> Well That code forThat's not really what i meant [23:43:17 CEST] <bray90820> I wanted it to check each file in the directory and output which files had wha audio [23:44:21 CEST] <llogan> you can use the command i gave you to do that via scripting of your choice [23:44:56 CEST] <bray90820> it seems that with your code you need to input the name of a file and tells you weather it has WMA audio [23:45:34 CEST] <bray90820> Maybe I'm wrong [23:45:56 CEST] <llogan> yes. and in linux i would use it with a bash for loop or with find for it to use the ffprobe on each file in a directory [23:46:42 CEST] <bray90820> If I was similar with bash enough I would do that but I'm not [23:47:58 CEST] <llogan> for f in *.wmv; do echo -n "$f: "; ffprobe -loglevel error -select_streams a -show_entries stream=codec_name -of default=nw=1:nk=1 "$f"; done [23:48:20 CEST] <ozette> i tried to transcode the sintel movie (.mkv) to .m3u8 (for HLS), but the playlist doesn't look right, and i've got a bunch of .vtt files which only say "WEBVTT" [23:48:37 CEST] <ozette> one moment i'll provide the command [23:49:35 CEST] <ozette> essentially this: ffmpeg -i sintel.mkv out.m3u8 [23:50:18 CEST] <ozette> the playlist: http://paste.debian.net/787528/ [23:50:54 CEST] <llogan> you need to show the complete console output [23:50:57 CEST] <llogan> from the command [23:51:31 CEST] <ozette> ok [23:53:51 CEST] <ozette> running the command again to capture the output.. [23:54:54 CEST] <ozette> anyway, maybe i don't understand the format correctly, but i expect the playlist to contain a reference to all segments, (about 200) [00:00:00 CEST] --- Tue Aug 9 2016

1 0

ffmpeg-devel.log.20160807
by burek 08 Aug '16

08 Aug '16

[04:41:39 CEST] <cone-642> ffmpeg 03Chris Moeller 07master:7387506b0693: avformat: read id3v2 comment tags [05:46:34 CEST] <Timothy_Gu> atomnuker: you mentioned the fast coder has been tested a lot, but the patch hasn't been merged into master yet, right? [05:47:22 CEST] <Timothy_Gu> Also if you could add the 6-7 times info into the commit message it'd be great [05:47:49 CEST] <Timothy_Gu> and nice work :) [08:39:02 CEST] <bp0> Can I push that hdcd analyze_mode patch? https://patchwork.ffmpeg.org/patch/110/ [08:39:21 CEST] <bp0> What is the normal time to wait for comments? [08:49:59 CEST] <Timothy_Gu> bp0: the hdcd_ana_mode_t typedef isn't really needed right? Either way, the _t suffix is supposed to be reserved, so an int could be used instead [08:51:01 CEST] <bp0> its not anymore, I guess, it was an enum [08:51:12 CEST] <bp0> but i changed to use #defines [08:51:46 CEST] <bp0> I will change to int [08:52:51 CEST] <Timothy_Gu> !!(ctlret == HDCD_TG_MISMATCH) the double !! isn't needed [08:54:24 CEST] <bp0> !! removed [08:56:08 CEST] <bp0> next? :) [11:21:13 CEST] <cone-654> ffmpeg 03Vitaly Ostrosablin 07master:90ddcaaf2de7: avfilter: add crystalizer audio filter [11:23:46 CEST] <cone-654> ffmpeg 03Paul B Mahol 07master:81fcd9126544: doc/filters: fix typo in extrastereo docs [14:27:59 CEST] <BtbN> Can someone get rid of that broken chatbot/troll over in #ffmpeg? [14:32:38 CEST] <durandal_1707> das bot? [15:09:32 CEST] <iive> BtbN: which one? [15:11:51 CEST] <BtbN> jookiyaya. Kind of too late now though. Also hard to tell if it's just a troll. Seriously suspected a chatbot for a while. [15:12:18 CEST] <iive> more like a troll, he poasted same link in few other channels [15:12:24 CEST] <iive> but got ingored there [15:54:54 CEST] <cone-654> ffmpeg 03Paul B Mahol 07master:1562a4c4f3eb: doc/filters: reorganize some entries, move scale_npp to right section [16:10:00 CEST] <durandal_1707> 2 filters left for 300th one [16:49:41 CEST] <kierank> shame the api is useless [16:50:14 CEST] <kierank> still willing to fund a decent api [16:54:16 CEST] <Chloe> What's useless about it? [16:54:42 CEST] <nevcairiel> its only useless for his usecase [17:01:17 CEST] <durandal_1707> kierank: I will look at that [17:01:33 CEST] <kierank> nevcairiel: it's not [17:01:49 CEST] <kierank> you can count the number of api users of lavfi on one hand [17:02:26 CEST] <nevcairiel> and i seriously doubt they have the same reasoning as you [17:02:32 CEST] <kierank> any realtime playback applicaton (vlc) can't use lavfi [17:02:36 CEST] <kierank> because it buffers arbitrarily [17:03:18 CEST] <kierank> it's too "designed" around ffmpeg.c and files [17:03:44 CEST] <nevcairiel> it doesnt guarantee any certain buffering, but thats not the same as buffering arbitrarily, because in reality its not going to just buffer all sorts of random shit unless you use something dumb like the loop or reverse filters [17:04:01 CEST] <kierank> in my email i gave a perfectly reasonable use-case where it buffered arbitrarily [17:04:17 CEST] <kierank> you have no idea what the latency is and whether it's fixed or variabble [17:05:01 CEST] <kierank> again not a problem if you are a file [17:05:08 CEST] <durandal_1707> its for overlay two frames... [17:05:36 CEST] <kierank> overlaying subtitles on a video is not a hard demand [17:05:42 CEST] <nevcairiel> there is only so many options you have when merging two streams, either you buffer or you drop [17:06:35 CEST] <kierank> the next problem is undefined latency in a filter chain overall [17:07:38 CEST] <nevcairiel> a new API isnt going to fix that, not witout making the filtergraph internally extremely inflexible [17:10:57 CEST] <durandal_1707> well filtergraph as concept in lavfi is very limited [17:11:50 CEST] <kierank> i don't particularly care what's going inside the filtergraph just so long as after n milliseconds I'll have output [17:12:47 CEST] <kierank> non-file playback is not a hard problem [17:15:20 CEST] <kierank> maybe we should have a session on it at vdd [17:16:06 CEST] <bencoh> that's not just "kierank's usecase", that's actually pretty much every "broadcast" (live) usecase, afaict [17:16:17 CEST] <kierank> it's any live playback case [17:16:26 CEST] <kierank> vlc would have the same problem [17:16:38 CEST] <bencoh> (and one of the reason I didn't use it as well - not the only one, though) [17:16:58 CEST] <kierank> bencoh: what was the other reason? [17:18:07 CEST] <bencoh> avframe handling/referencing but that was quite some time ago (and maybe it'd have been worked out given more time) [17:18:26 CEST] <kierank> I think this is sorted now [17:20:10 CEST] <durandal_1707> it was nightmare before.. [17:20:55 CEST] <bencoh> durandal_1707: that's what it looked like to me, yeah. haven't had a look at all that stuff for over a year now [17:36:54 CEST] <cone-654> ffmpeg 03Michael Niedermayer 07master:bbec14de3126: avcodec/raw: Fix decoding of ilacetest.mov [19:19:48 CEST] <cone-654> ffmpeg 03Yuli Khodorkovskiy 07master:f0adb99d068e: avcodec/qsvdec: Fix null dereferences in the qsv decoder [20:21:59 CEST] <cone-654> ffmpeg 03Paul B Mahol 07master:571f33de0aa8: avfilter/avf_showspectrum: improve dB scale description [20:42:49 CEST] <Compn> thats great though that edit list patch is posted :) [20:44:55 CEST] <Compn> only a couple hundred lines too [20:45:08 CEST] <Compn> impressive for such a feature people requested for years [23:08:45 CEST] <Prelude2004c> hey guys.. nobody really talking in ffmpeg .. question that might be simple [23:08:46 CEST] <Prelude2004c> hey guys.. good day ... i am looking for some help here with something silly that i can't figure out... so i am splitting an input like " -filter_complex '[0:v]yadif,format=nv12,split=3[vout1][vout2][vout3]" and the output can't keep up with the input and its running at less than 1x... ok so to debug i did -filter_complex '[0:v]yadif,format=nv12,split=1[out1] and it did the same thing..only a single output . same problem. now [23:08:46 CEST] <Prelude2004c> when i took out the ,split=1[out1] -map '[out1]', everything went ok and it was 1x .. any ideas ? [23:56:15 CEST] <cone-654> ffmpeg 03Paul B Mahol 07master:22d13e4290c8: avfilter/af_dynaudnorm: use better check for infinite loop [00:00:00 CEST] --- Mon Aug 8 2016

1 0

ffmpeg.log.20160807
by burek 08 Aug '16

08 Aug '16

[02:31:25 CEST] <ajf> I want to convert a video file from h.264 to MJPEG, just for the heck of it. How do I tell ffmpeg to maintain the source's bitrate? [02:32:05 CEST] <c_14> you can't [02:32:36 CEST] <c_14> you could technically check the size of the input file, do math and get the bitrate and tell ffmpeg to use that [02:32:39 CEST] <c_14> But it won't be the same quality [02:32:46 CEST] <ajf> oh sure [02:32:54 CEST] <c_14> And unless the input is constant bitrate, it won't match anyway [02:32:58 CEST] <ajf> right [02:33:26 CEST] <c_14> If this is just random testing whatever, check the output of ffprobe if it gives you a bitrate on the input, then subtract the audio stream and round down a bit [02:34:15 CEST] <ajf> hmm [02:34:26 CEST] <ajf> does -maxbitrate apply to the entire stream, or just video? [02:35:45 CEST] <c_14> Do you mean -maxrate ? [02:35:48 CEST] <c_14> There's no -maxbitrate [02:35:51 CEST] <ajf> probably [02:36:18 CEST] <c_14> I think it only applies to the video stream, I'm not sure though. [02:36:40 CEST] <ajf> oh wait I think I want -b:v [02:36:46 CEST] <ajf> and that sets video bitrate specifically [02:38:09 CEST] <c_14> yes [02:40:36 CEST] <c_14> Though I'm not sure the mjpeg encoder has a cbr mode... [02:41:26 CEST] <ajf> hah, yep [02:41:35 CEST] <ajf> I think it's completely ignored me [02:41:37 CEST] <ajf> :D [02:42:34 CEST] <c_14> you might be able to use -q or -qp to set a quantizer and play around with that until it's about the same size [02:46:49 CEST] <ajf> yeah. it looks like the mjpeg encoder really doesn't like being told what bitrate to use [02:47:04 CEST] <ajf> which does make sense in fairness [02:52:16 CEST] <ajf> huh, mjpeg doesn't like -qp [02:54:12 CEST] <c_14> -q seems to work [03:02:09 CEST] <ajf> it looks like mjpeg is really stubborn [03:02:10 CEST] <ajf> :< [03:02:55 CEST] <ajf> I mean I'd happily tell it to encode everything as JPEGs of quality 10 (extreme quantisation) [03:02:59 CEST] <ajf> but there's no such option [03:04:32 CEST] <c_14> set -q:v to 31 [03:04:43 CEST] <c_14> I think that's the lower end of the range [03:04:52 CEST] <ajf> yeah [03:04:56 CEST] <ajf> it is also the default [03:27:33 CEST] <ajf> thanks for your attempts to help me [05:03:56 CEST] <Spring> is there a bulletproof syntax for filter_complex commands? [05:04:02 CEST] <Spring> using the command line [05:04:47 CEST] <Spring> I've escaped both colons and commas (not that they even need to be escaped but I've tried everything) and keep getting errors [05:05:39 CEST] <Spring> the comment in parenthesis above referring to the commas that is [05:10:16 CEST] <durandal_1707> using script files [05:10:50 CEST] <Spring> durandal_1707, I'm using batch [05:11:27 CEST] <Spring> ffmpeg keeps detecting the entire line as the arguments for a single command [05:11:49 CEST] <durandal_1707> I mean ffmpeg option [05:12:55 CEST] <Spring> is there a doc page on this, nothing is showing up in a search [05:16:30 CEST] <durandal_1707> -filter-complex-script? [05:20:15 CEST] <Spring> this? https://trac.ffmpeg.org/wiki/FilteringGuide#Scriptingyourcommandlineparamet… [05:20:19 CEST] <Spring> can't find anything else [05:21:12 CEST] <wallbroken> is possible to convert from pcm to aac ? [05:21:18 CEST] <durandal_1707> its not in wiki, :( [05:22:06 CEST] <durandal_1707> Spring: search ffmpeg docs on ffmpeg.org page [05:22:39 CEST] <durandal_1707> wallbroken: yes, its possible [05:22:48 CEST] <wallbroken> how? [05:22:59 CEST] <Spring> woah, literally has a single line about it mentioned nowhere else [05:23:10 CEST] <Spring> -filter_script [05:23:32 CEST] <Spring> er, rather, -filter_complex_script [05:24:11 CEST] <durandal_1707> wallbroken: add -c:a aac [05:25:28 CEST] <wallbroken> durandal_1707, it's a little harder, the PCM stream is in an .avi file [05:26:27 CEST] <durandal_1707> wallbroken: what container? [05:26:38 CEST] <wallbroken> .avi [05:26:47 CEST] <durandal_1707> for output? [05:27:21 CEST] <wallbroken> for output must be .aac format [05:27:29 CEST] <wallbroken> the input is a pcm in a .avi file [05:28:28 CEST] <durandal_1707> than just: ffmpeg -i in.AVI out.aac for start [05:29:09 CEST] <wallbroken> https://www.dropbox.com/s/5evj7aijndgvzt6/ok.txt?dl=0 [05:29:13 CEST] <wallbroken> this is the input file [05:29:23 CEST] <wallbroken> can you tell me the entire command? [05:29:27 CEST] <furq> he just did [05:29:47 CEST] <wallbroken> but i need to tell the bitrate [05:29:58 CEST] <furq> -b:a 128k [05:30:21 CEST] <Spring> alternative question, can the trim filter be configured to cut a certain part of a video according to the original timestamp, then that trimmed portion speed adjusted in a single command? [05:30:36 CEST] <Spring> *single filtergraph line [05:30:44 CEST] <wallbroken> " ffmpeg -i in.AVI -b:a 128k out.aac" that's all? [05:31:20 CEST] <furq> yes [05:31:39 CEST] <Spring> even using seconds instead of the full timestamp (which even escaped gives errors) doesn't trim the video correctly before applying the speed filter [05:32:01 CEST] <wallbroken> furq, PCM it's lossy? [05:33:19 CEST] <durandal_1707> Spring: you do not escape them.when using script option [05:34:14 CEST] <Spring> I kind of need the batch script to be one file though, and as mentioned even using seconds doesn't produce the right output (eg: using '64' instead of 1:04) [05:35:27 CEST] <Spring> eg http://pastebin.com/7ukXsxMd [05:36:01 CEST] <Spring> this produces a 3:48 video clip of a static frame [05:37:20 CEST] <Spring> or a /super/ slow video it's hard to tell [05:37:34 CEST] <durandal_1707> perhaps you give args for exact frame number [05:38:06 CEST] <furq> don't you need atrim as well as trim [05:38:36 CEST] <Spring> yes but since trim was causing trouble I wanted to figure out one at a time [05:39:27 CEST] <Spring> I also need timestamps rather than frame number, since that's what is used for the input initially [05:41:09 CEST] <Spring> I guess I'm just completely puzzled by the trim filter, since the others work fine without it, whereas '-ss blah to blah' works but is affected by the speed filter so it outputs the wrong portion [05:42:04 CEST] <Spring> hence why I needed to see if the trim filter could be forced to trim the video before applying the speed filter [05:42:29 CEST] <durandal_1707> Spring: what ffmpeg version? [05:43:02 CEST] <durandal_1707> perhaps you need to reset video pts to 0 [05:43:22 CEST] <Spring> durandal_1707, version N-80801-gc0cb53c [05:43:32 CEST] <durandal_1707> not just speed it up [05:44:50 CEST] <durandal_1707> because ffmpeg will try to keep audio and video in sync [05:45:29 CEST] <durandal_1707> and trim doesn't reset pts [05:48:28 CEST] <furq> Spring: -vf trim=30:40,setpts=2*(PTS-STARTPTS) -af atrim=30:40,atempo=0.500 [05:48:29 CEST] <furq> that works for me [05:49:48 CEST] <durandal_1707> Yeah [05:49:57 CEST] <Spring> I'm actually wondering if it's interpreting 2.000 as '2000' not '2.0' [05:50:28 CEST] <furq> it isn't [05:50:34 CEST] <furq> you need to subtract STARTPTS or it won't work [05:50:45 CEST] <furq> or reset the pts in some other way, but that seems easiest [05:52:13 CEST] <furq> the command you pasted will work but the video will start playing at 2:08 [05:53:01 CEST] <wallbroken> durandal_1707, it does not work [05:53:09 CEST] <wallbroken> the .avi file has two streams in [05:53:43 CEST] <wallbroken> please take a look at those mediainfo input file: https://www.dropbox.com/s/5evj7aijndgvzt6/ok.txt?dl=0 [05:54:08 CEST] <wallbroken> the play of the output file does not produce anything [05:54:28 CEST] <wallbroken> the file lenght is 50 mins, but it does nor reproduce anything [05:54:41 CEST] <durandal_1707> wallbroken: what's output of ffmpeg command? [05:55:44 CEST] <durandal_1707> mediainfo is useless to me, better use ffprobe.... [05:56:02 CEST] <wallbroken> https://www.dropbox.com/s/py68bqqu8bnp5qa/out.txt?dl=0 [05:57:06 CEST] <Spring> furq, you don't know how much you relieved me with that answer, thanks a TON [05:57:17 CEST] <furq> Spring: -vf trim=00\\:00\\:30:00\\:00\\:40 [05:57:20 CEST] <furq> that also works for me fwiw [05:57:58 CEST] <furq> although it's probably easier to just convert to seconds rather than figure out what escaping level you're at [05:58:10 CEST] <furq> it is nice to know that it technically does accept duration specs though [05:59:01 CEST] <Spring> hmm, may try testing the escaped shorthand [05:59:09 CEST] <wallbroken> durandal_1707, you saw it? [05:59:23 CEST] <wallbroken> *seen [05:59:37 CEST] <furq> wallbroken: that looks like it's working fine [05:59:40 CEST] <furq> what exactly is the problem [05:59:48 CEST] <durandal_1707> yes, what you use for playback? [06:00:20 CEST] <wallbroken> furq, open it with vlc [06:00:36 CEST] <wallbroken> play, the playback is about 4 seconds, with no sound [06:00:39 CEST] <wallbroken> then stops it [06:00:52 CEST] <wallbroken> but the duration of all the track is about 50 minutes [06:00:58 CEST] <wallbroken> and in vlc is also reported [06:01:37 CEST] <Spring> does the shorthand trim=timecode interpret a single timecode as inherently representing the start (without an end trim)? [06:02:09 CEST] <wallbroken> furq, wait, now it seem to work but the duration is continuosly changing [06:02:10 CEST] <furq> The defaults are such that all the input is kept. So it is possible to set e.g. just the end values to keep everything before the specified time. [06:03:24 CEST] <furq> wallbroken: have you tried playing it in something better than vlc [06:03:51 CEST] <wallbroken> furq, vlc is the masterpice of playback software [06:03:58 CEST] <wallbroken> *masterpiece [06:04:38 CEST] <furq> i'll take your word for it [06:04:53 CEST] <durandal_1707> lol [06:05:27 CEST] <wallbroken> furq, that's all people said to me: "vlc playpacks also rocks" [06:06:23 CEST] <furq> there's no accounting for taste [06:06:45 CEST] <furq> try it in ffplay, mpv, fb2k or something competent [06:07:17 CEST] <wallbroken> i don't know, but which software should be better? windows media player? quicktime player? [06:07:31 CEST] <furq> mpv or mpc-hc on windows [06:09:53 CEST] <wallbroken> somebody can tell me once for all, if is preferred ffmpeg 32bit or 64? [06:10:48 CEST] <durandal_1707> 64 on 64 bit, 32 bit otherwise [06:11:17 CEST] <wallbroken> 32 bit encoded video could be not compatible with some platform? [06:11:18 CEST] <Spring> I use PotPlayer [06:11:34 CEST] <furq> wallbroken: no [06:11:35 CEST] <furq> that would be insane [06:11:46 CEST] <wallbroken> what? [06:12:37 CEST] <durandal_1707> Isn't potplayer license violator? [06:13:19 CEST] <wallbroken> another thing: i want to convert a vob video in x264 with quality factor: 23, how to do it? [06:13:41 CEST] <furq> -i src.vob out.mkv [06:13:45 CEST] <Spring> not sure, heard something like that. Best player I've used on Windows though. [06:13:58 CEST] <wallbroken> furq, and about the quality factor [06:14:00 CEST] <wallbroken> and the preset? [06:14:12 CEST] <furq> the default crf is 23 [06:14:19 CEST] <wallbroken> and the preset? [06:14:22 CEST] <furq> medium [06:14:25 CEST] <wallbroken> if i want to use placebo? [06:14:29 CEST] <furq> -preset placebo [06:14:37 CEST] <furq> note that i am not recommending preset placebo [06:14:54 CEST] <wallbroken> just for the high time needed? [06:15:08 CEST] <furq> the clue's in the name [06:15:41 CEST] <furq> if you want to spend a million years encoding then at least use x265 and get some benefit out of it [06:16:30 CEST] <wallbroken> x265? [06:16:36 CEST] <wallbroken> is better than x264 ? [06:16:44 CEST] <furq> it's one better than x264 [06:17:01 CEST] <wallbroken> ok, and how to use it? [06:17:53 CEST] <furq> https://trac.ffmpeg.org/wiki/Encode/H.265 [06:18:22 CEST] <wallbroken> and between mp4 and mkv [06:18:27 CEST] <wallbroken> which one is better? [06:18:57 CEST] <wallbroken> something buggy in aac of ffmpeg [06:19:05 CEST] <wallbroken> i tried opening the same file with groove music [06:19:13 CEST] <wallbroken> and the wrong duration time here too [06:21:23 CEST] <wallbroken> let me try doing it in mp3 [06:48:37 CEST] <flai> Hey guys, I want to encode a video for use in a HTML5-<video /> element. Currently I'm trying to use mp4 with h264/aac. The problem is that the video-file isn't working, and I can't figure out why. Details: (Picture, Commandline, ...): https://gist.github.com/moritzuehling/3a08559340910721079811069768f751 [06:49:12 CEST] <flai> Any help and pointer on why my encoded video isn't working and what I can do to make it work would be very appreciated! [08:52:56 CEST] <hollabackatcha> can ffsever be used for live streaming ? right now it's just reloading inputed videos from the beginning... it's not live [09:16:34 CEST] <Spring> can the atempo filter be stacked for slow mo, as well? Using atempo=.500,atempo=.500 gives an error [09:23:28 CEST] <c_14> Spring: it should work (it works here) [09:24:40 CEST] <Spring> c_14, did you test using the -filter:a option? [09:24:54 CEST] <c_14> I used -af (which should be the same thing) [09:25:11 CEST] <Spring> I'm adding mine within -filter_complex [09:25:41 CEST] <c_14> Works with a filter_complex as well [09:28:57 CEST] <Spring> c_14, http://pastebin.com/DsnLTESy [09:30:05 CEST] <Spring> ffmpeg crashes as soon as the encode begins [09:32:22 CEST] <c_14> The problem is the atrim filter [09:32:26 CEST] <c_14> If you remove that it works [09:32:43 CEST] <Spring> it works if the second atempo isn't there [09:33:10 CEST] <c_14> not on my system [09:33:33 CEST] <Spring> actually, you're right, wtf [09:34:23 CEST] <c_14> eh [09:34:34 CEST] <furq> that works here [09:34:38 CEST] <c_14> The atrim filter doesn't accept timestamps in HH:MM:SS.MSS notation [09:34:46 CEST] <furq> oh [09:34:58 CEST] <c_14> hmm, no it should [09:34:58 CEST] <Spring> WUT [09:35:06 CEST] <furq> -af atrim=0,atempo=0.5,atempo=0.5 [09:35:08 CEST] <c_14> the escaping just isn't working [09:35:08 CEST] <furq> that works for me [09:35:39 CEST] <furq> Spring: fwiw if those are the only filters you're using then there's no need to use filter_complex [09:35:44 CEST] <c_14> adding single quotes around the timestamp in addition to the backslashes works for me [09:35:50 CEST] <c_14> "[0:a]atrim='00\:00\:00.000',atempo=.5,atempo=.5[a]" [09:35:52 CEST] <c_14> ^like that [09:36:59 CEST] <c_14> The colons weren't escaped correctly, so the atrim filter never produced output which caused the assertion in the atempo filter [09:38:19 CEST] <furq> also it's a waste of time to run that in the first pass [09:38:28 CEST] <furq> just use -an to ignore the audio [09:39:06 CEST] <c_14> Ye, for pass 1 you can just ignore the audio since that isn't taken into account for the 2pass data generation [09:39:24 CEST] <Spring> I had -an originally, can't recall why I removed it, possibly while encountering some other error [09:40:44 CEST] <Spring> and yup, the single quotes fixed the error, cheers! [09:41:31 CEST] <Spring> is there a technical reason why atempo can't go below .5 or above 2.0 in one run? [09:47:31 CEST] <LJHSLDJHSDLJH> I'm trying to reduce video files size using the below command line and it's taking forever to complete and the amount of reduced file is size is very disappointing. Can anyone help me with better syntax maybe! [09:47:35 CEST] <LJHSLDJHSDLJH> for x in *.mp4;do ffmpeg -i "$x" -c:v libx264 -crf 24 -b:v 1M -c:a aac -strict -2 "./x/$x";done [09:48:53 CEST] <cbsrobot> LJHSLDJHSDLJH: If possible update the ffmpeg version [09:49:22 CEST] <cbsrobot> aac encoder got a lot better lately and is not experimental anymore [09:50:08 CEST] <cbsrobot> so you can drop the " -strict -2" [09:50:33 CEST] <cbsrobot> for the video drop the " -b:v 1M", just play with the -crf value [09:51:27 CEST] <cbsrobot> I suggest try values between 27 (high compression, small filesize) and 18 (smaller compression, bigger filesize) [09:59:50 CEST] <LJHSLDJHSDLJH> hello cbsrobot : ffmpeg version 2.8.6-1ubuntu2 Copyright (c) 2000-2016 the FFmpeg developers [09:59:57 CEST] <LJHSLDJHSDLJH> do I need to upgrade that? [10:01:44 CEST] <cbsrobot> LJHSLDJHSDLJH: I do not remember the exact version the aac encoder was updated, but drop the "-strict -2" and if it still works then you should be good to go [10:02:01 CEST] <furq> it was upgraded in 3.0 [10:02:13 CEST] <furq> but if the source files are mp4 then there's probably no point reencoding the audio [10:02:24 CEST] <furq> just use -c:a copy [10:05:09 CEST] <LJHSLDJHSDLJH> alright, now it's working again .. I'll wait and see the results , thanks [10:05:11 CEST] <LJHSLDJHSDLJH> for y in *.mp4;do ffmpeg -i "$y" -c:v libx264 -crf 20 -c:a copy "./y/$y";done [10:21:51 CEST] <xochilpili> hi all [10:22:13 CEST] <xochilpili> im having this trouble, i want to use vod (video on demand) using vlc [10:22:32 CEST] <xochilpili> i have use: cvlc /media/Movies/Che.el.Argentino.avi --sout '#transcode{vcodec=libx264,vb=2000,scale=0,channels=2,samplerate=44100}:std{access=http{mime=video/mp4},mux=mp4,dst=:8080/cam1}' -vvv [10:22:42 CEST] <xochilpili> and this is the full output: http://pastebin.com/HEC4GSyj [10:22:53 CEST] <xochilpili> please any help i will appreciate :D [10:26:25 CEST] <cbsrobot> xochilpili: maybe ask in #videolan [10:26:35 CEST] <xochilpili> im in there [10:26:38 CEST] <xochilpili> nothing... [10:26:59 CEST] <xochilpili> " I also dont know what your problem could be ;) " [10:27:04 CEST] <xochilpili> they said [11:50:28 CEST] <LJHSLDJHSDLJH> using the below is too slow as well and moreover it exceeded the original file size [11:50:31 CEST] <LJHSLDJHSDLJH> for y in *.mp4;do ffmpeg -i "$y" -c:v libx264 -crf 20 -c:a copy "./y/$y";done [11:51:27 CEST] <LJHSLDJHSDLJH> using this is quickly BUT it converted 762mb file into 2.4gb file size [11:51:28 CEST] <LJHSLDJHSDLJH> for z in *.mp4;do ffmpeg -i "$z" -c:v libx264 -preset ultrafast -crf 20 -c:a copy "./z/$z";done [11:52:03 CEST] <LJHSLDJHSDLJH> therefore ffmpeg conversion is considered crap at my side until further change happen [11:52:19 CEST] <LJHSLDJHSDLJH> I'll have to redownload the whole thing again [12:56:01 CEST] <flai> Hey guys, I want to encode a video for use in a HTML5-<video /> element. Currently I'm trying to use mp4 with h264/aac. The problem is that the video-file isn't working, and I can't figure out why. Details: (Picture, Commandline, ...): https://gist.github.com/moritzuehling/3a08559340910721079811069768f751 [12:56:03 CEST] <flai> Any help and pointer on why my encoded video isn't working and what I can do to make it work would be very appreciated! [12:56:14 CEST] <flai> (I hope it was okay to bump after 6 hours) [13:21:15 CEST] <BtbN> Both files work perfectly fine for me. [13:21:46 CEST] <BtbN> Whatever Browser that is, I'd blame it for refusing to play the second file. [13:24:48 CEST] <furq> flai: pastebin the output of ffprobe -show_streams and/or mediainfo [13:37:54 CEST] <Spring> is there a calculation for determining atempo values larger than 2.0? [13:38:29 CEST] <jookiyaya> big news: zen destroys intel http://static2.gamespot.com/uploads/scale_super/460/4601080/3046688-8311357… [13:41:23 CEST] <flai> furq: https://gist.github.com/moritzuehling/e105d2dd67402bbcf12f15eecdc6897b [13:42:42 CEST] <furq> maybe try without -level 3.1 [13:42:54 CEST] <furq> i would have assumed ffmpeg would just ignore it for 720p60 but evidently not [13:47:03 CEST] <flai> furq: Encoding... [13:54:50 CEST] <flai> furq: Not working :( https://gist.github.com/moritzuehling/60e83cdfbec40a10db33a5a8b5f9e531 [13:55:00 CEST] <jkqxz> jookiyaya: Zen adds the hardware instructions for hashing functions, so obviously it can do them at close to the memory line rate. (Skylake and Goldmont have them too - presumably that benchmark is deliberately not using them on the 6700K to make the results more amusingly misleading.) [13:57:58 CEST] <BtbN> I also have a hard time figuring out what is even compared there. Some MB/s value. of what? Seems rigged to me. Specially as the AMD machine has the fastest RAM of all of them installed. [13:58:00 CEST] <flai> Maybe it helps: The source-file is recorded with OBS + NVENC... [13:58:42 CEST] <furq> BtbN: it's SHA1 hashing speed [13:59:01 CEST] <furq> so yeah, pretty much worthless [13:59:22 CEST] <BtbN> Then it's even more stupid to give the AMD machine the fastest RAM [14:00:52 CEST] <jookiyaya> #5 on that list has ddr4 too [14:00:58 CEST] <furq> it's actually pretty smart to do that because it makes it easier to produce a screenshot where it's at the top of the list [14:02:06 CEST] <furq> jookiyaya: the ram speed doesn't really matter, sha1 hashing speed is a worthless metric [14:02:19 CEST] <furq> especially when some of those cpus have extensions for it and some don't [14:02:54 CEST] <furq> i have to suspect i'm being overly charitable by even explaining this [14:03:29 CEST] <jookiyaya> then basically, any benchmark score is useless info to you [14:03:38 CEST] <jookiyaya> since it's only one aspect [14:03:49 CEST] <BtbN> The score of a pointless benchmark is indeed useless, yes. [14:04:24 CEST] <BtbN> No CPU that would be a "worthy opponent" in that benchmark is in the list. [14:04:33 CEST] <jkqxz> In this channel, we would probably be more interested in a benchmark of x264 performance... [14:04:45 CEST] <BtbN> Like, a i7-6700k or i7-6900k [14:05:30 CEST] <BtbN> Wait, there is a 6700k. [14:05:35 CEST] <jkqxz> There's a 6700K there. Presumably they avoided using the hardware instructions on it in order to ensure the right horse won (it does look like it's in roughly the right place for a software implementation). [14:05:38 CEST] <BtbN> Yeah, it seems indeed rigged then. [14:05:48 CEST] <jookiyaya> jkqxz why still use x264, when there is x265 [14:06:31 CEST] <Spring> 1.7 too fast 1.5 too slow, 1.6 too slow... :/ [14:06:54 CEST] <jookiyaya> spring too fast? [14:07:07 CEST] <jkqxz> x264 has better tuning to occupy the processor as much as possible, so it will be a better benchmark. Also 9001 times as many people use it, which probably makes it more relevant... [14:07:14 CEST] <flai> furq: Any idea where the problem may lay? [14:07:20 CEST] <furq> nope [14:07:32 CEST] <furq> well [14:07:34 CEST] <jookiyaya> spring 1.7 is not too fast [14:07:39 CEST] <furq> if i had to guess i'd say edge is the problem [14:07:43 CEST] <furq> but you can't really do anything about that [14:08:00 CEST] <Spring> jookiyaya, can't find the correct calculation for atempo when stacking values over 2.0 (ie: 200%) [14:08:02 CEST] <jookiyaya> spring is that a typo [14:08:19 CEST] <furq> Spring: what final value do you want [14:08:33 CEST] <jookiyaya> spring, huh? please explain why you said 1.7 = too fast [14:09:34 CEST] <furq> if you want 3.0 then use 2.0 and 1.5 [14:10:21 CEST] <jookiyaya> furq do you have any idea what spring is saying [14:10:27 CEST] <furq> yes [14:10:34 CEST] <Spring> furq, so I'm using an input value (percentage) then calculating the setpts and atempo values from it. However going above 2.0 I can't find the right additional atempo value to add to it [14:10:43 CEST] <furq> i know [14:10:50 CEST] <furq> you were perfectly clear [14:10:57 CEST] <jookiyaya> furq how is 1.7 too fast, i don't get it [14:11:25 CEST] <Spring> was just letting off a little steam while looking for 280% [14:11:38 CEST] <furq> 2.0 and 1.4 [14:14:36 CEST] <Spring> furq, way too slow [14:15:42 CEST] <jookiyaya> spring so only 1.7 is fast? [14:16:21 CEST] <jookiyaya> furq how is 1.7 too fast, i don't get it [14:16:47 CEST] <Spring> there's some spot between 1.6 and 1.7 that seems to be the correct value for atempo at 280% video speed, but I can't tell what the calculation is meant to be [14:17:25 CEST] <furq> i don't get how 2.0 and 1.4 doesn't work [14:18:15 CEST] <Spring> it seems like it would be simple but the results say otherwise [14:19:36 CEST] <jookiyaya> what are you 2 talking about [14:21:06 CEST] <Spring> jookiyaya, atempo (audio tempo filter) has a usable range of 0.5 to 2.0. Any lower or higher and you have stack the filters to achieve the desired speed (eg: over 200%/2.0 or under 50%/0.5) [14:21:55 CEST] <jookiyaya> spring english please, example "encoding is too slow" picture looks crappy [14:22:08 CEST] <jookiyaya> audio and video doesn't sync/match [14:23:20 CEST] <Spring> basically audio isn't the correct speed for the video, so they don't match [14:23:45 CEST] <jookiyaya> spring for which x265 versions? [14:24:34 CEST] <Spring> it's not about a particular codec [14:24:35 CEST] <jookiyaya> i've been using x265 since 1.5, how come i never had this problem [14:24:43 CEST] <jookiyaya> i've been using x265 since 1.5 to 1.9, how come i never had this problem [14:25:08 CEST] <jookiyaya> spring then why did you say this? <Spring> 1.7 too fast 1.5 too slow, 1.6 too slow [14:25:39 CEST] <jookiyaya> 1.7 magically became too fast for you? are you serious [14:26:04 CEST] <jookiyaya> furq any idea [14:26:26 CEST] <jookiyaya> dv any idea what spring is talking about [14:26:39 CEST] <jookiyaya> fahadash any idea what spring is talking about [14:26:46 CEST] <jookiyaya> durandal_1707 any idea what spring is talking about [14:26:52 CEST] <jookiyaya> ozette any idea what spring is talking about [14:27:05 CEST] <Spring> ...I interjected at an inopportune moment, pay my issue no mind [14:27:17 CEST] <jookiyaya> spring no you said it [14:27:23 CEST] <jookiyaya> i want my explanation [14:27:37 CEST] <Spring> I already explained it like twice, what don't you understand? [14:27:49 CEST] <jookiyaya> you didn't explain in english [14:28:15 CEST] <jookiyaya> that most people would understand [14:29:00 CEST] <jookiyaya> <Spring> 1.7 too fast 1.5 too slow, 1.6 too slow ; 1.7 magically became too fast for you? explain [14:29:12 CEST] <durandal_1707> it can't match, he is not using interpolation [14:29:41 CEST] <jookiyaya> durandal_1707 any idea what spring is saying [14:30:40 CEST] <jookiyaya> [05:05] <jookiyaya> jkqxz why still use x264, when there is x265 [14:30:41 CEST] <jookiyaya> [05:06] <Spring> 1.7 too fast 1.5 too slow, 1.6 too slow... :/ [14:30:41 CEST] <jookiyaya> [05:06] <jookiyaya> spring too fast? [14:30:41 CEST] <jookiyaya> [05:06] <jkqxz> x264 has better tuning to occupy the processor as much as possible, so it will be a better benchmark. Also 9001 times as many people use it, which probably makes it more relevant... [14:31:19 CEST] <jookiyaya> 05:07] <Spring> jookiyaya, can't find the correct calculation for atempo when stacking values over 2.0 (ie: 200%) [14:31:19 CEST] <jookiyaya> [05:07] <jookiyaya> spring is that a typo [14:31:19 CEST] <jookiyaya> [05:08] <furq> Spring: what final value do you want [14:31:19 CEST] <jookiyaya> [05:08] <jookiyaya> spring, huh? please explain why you said 1.7 = too fast [14:31:19 CEST] <jookiyaya> [05:09] <furq> if you want 3.0 then use 2.0 and 1.5 [14:31:19 CEST] <jookiyaya> [05:10] <jookiyaya> furq do you have any idea what spring is saying [14:31:56 CEST] <durandal_1707> He is attempting to slow video, IIRC [14:32:35 CEST] <jookiyaya> spring wait, when you said too fast/slow , you are not talking about encoding speed? [14:33:33 CEST] <durandal_1707> no [14:33:45 CEST] <jookiyaya> oh [14:34:13 CEST] <jookiyaya> what does it mean "too slow" then [14:36:06 CEST] <durandal_1707> that video is out of sync I guess [14:36:39 CEST] <Spring> video speed, anyway you can PM me if you'd like to continue the questions. Found that I set the multiplier too low for the batch script though (as it doesn't support decimal numbers), which may explain why the setpts value is off. [14:37:07 CEST] <Spring> well, not natively decimal, I convert them [14:40:15 CEST] <Spring> every number after the decimal is stripped so I have to multiply by like 100,000 to preserve enough of the value [14:57:47 CEST] <flai> furq: Apparently it was the AAC. The encoded aac from OBS isn' [14:58:07 CEST] <flai> doesn't work, if I reencode with -c:a aac it works [14:58:11 CEST] <flai> Thanks for your help! [16:40:50 CEST] <yangxuan8282_> hi [16:42:24 CEST] <yangxuan8282_> when I building ffmpeg in the chroot, it says: ERROR: mmal not found. [16:45:09 CEST] <yangxuan8282_> anybody face same issues? [16:59:34 CEST] <furq> yangxuan8282_: check config.log [16:59:58 CEST] <furq> https://github.com/FFmpeg/FFmpeg/blob/master/configure#L5766 [17:03:07 CEST] <yangxuan8282_> furq thanks for reply, the config.log is too long, I don't know check which part [17:03:13 CEST] <furq> the very end [17:14:32 CEST] <viric> Hello ffmpeg [17:14:48 CEST] <durandal_1707> Hello [17:14:48 CEST] <viric> is there anything I can use to adjust audio sync&drift between tracks? [17:15:10 CEST] <viric> other than adelay [17:15:52 CEST] <viric> I mean - I want help finding out the adelay parameter and speed adjustment :) [17:16:16 CEST] <viric> (atempo) [17:18:06 CEST] <durandal_1707> depends what tracks have [17:18:50 CEST] <viric> they are independant; one from a videocamera, the other from a dictophone [17:19:00 CEST] <viric> (MTS file + WMA file) [17:19:37 CEST] <durandal_1707> and how they differ? [17:19:52 CEST] <viric> could be 1 minute [17:20:06 CEST] <viric> the drift of the wma may be 0.1s per hour [17:20:25 CEST] <viric> maybe 0.5. 0.1 wouldn't be very much noticeable [17:20:57 CEST] <viric> But the point is that I don't know anything else than testing adelay values with ffplay [17:21:03 CEST] <viric> and checking manually the sync [17:22:06 CEST] <durandal_1707> adelay is for constant difference [17:23:01 CEST] <viric> adelay+atempo, I use [17:23:23 CEST] <viric> but I don't want to *calculate* the values by manual brute force [17:24:26 CEST] <durandal_1707> There is some kind of sound which indicate relative time [17:24:30 CEST] <durandal_1707> ? [17:24:36 CEST] <viric> it is speech [17:24:38 CEST] <viric> I did not clap hands [17:24:56 CEST] <viric> both tracks quite have the same speech; simply one is clearer [17:25:30 CEST] <viric> I tried using kdenlive's "audio reference" and alignment, but it crashes. :) [17:25:42 CEST] <viric> (and it is only about delay, not drift) [17:27:12 CEST] <durandal_1707> Hmm, silencedetect? [17:27:50 CEST] <durandal_1707> it will show when silence ended [17:28:08 CEST] <durandal_1707> but nothing for drift [17:28:56 CEST] <viric> I just feel that I'm missing something relevant; how can people compose tracks using multiple microphones? [17:29:06 CEST] <durandal_1707> unless voice have beats and someone write beat detector [17:29:57 CEST] <viric> audacity has something... http://manual.audacityteam.org/man/audio_alignment_tips.html [17:30:05 CEST] <durandal_1707> viric: compensationdelay filter if you know distance [17:30:25 CEST] <viric> what is compensationdelay? [17:30:48 CEST] <durandal_1707> calf plugin ported to lavfi [17:34:50 CEST] <viric> hm interesting [17:34:56 CEST] <viric> well, I'm not really mixing the audio though [17:35:03 CEST] <viric> I want dictaphone track aligned with video [17:36:37 CEST] <durandal_1707> yea, I guess load tracks in audacity and see how much to shift [17:37:18 CEST] <viric> Another thing I have to do most often is to switch the microphone during a talk of two people [17:37:24 CEST] <viric> imagine each has its own good microphone [17:37:47 CEST] <viric> I'm trying with kdenlive.. but it is a hell to do that. Maybe I just don't know a term for it to google for... [17:38:11 CEST] <bencoh> audacity is probably handy for that yeah [17:38:16 CEST] <bencoh> you'll just "see" it [17:38:21 CEST] <bencoh> (the waves) [17:38:37 CEST] <viric> aha [17:38:52 CEST] <viric> I should try. [17:39:09 CEST] <viric> So if I extract the audio from the mts and load it in audacity... the startpts would be correct once I mix them with video again? [17:40:46 CEST] <durandal_1707> no you would need to insert samples or remove them... [17:41:27 CEST] <viric> I'm a bit disturbed by this mts specificalities. Looking at ffprobe -show_streams I see for stream 0 (video): [17:41:30 CEST] <viric> start_pts=95400 [17:41:30 CEST] <viric> start_time=1.060000 [17:41:37 CEST] <viric> And for the audio track of the mts: [17:41:42 CEST] <viric> start_pts=91800 [17:41:42 CEST] <viric> start_time=1.020000 [17:42:38 CEST] <durandal_1707> that is another issue [17:42:51 CEST] <viric> :) [17:44:07 CEST] <viric> so if I extract the audio into a wav, I'd lose the start_time/pts, isn't it? [17:44:36 CEST] <viric> and direct remix would have a 1s a/v misalignment [17:45:59 CEST] <viric> Moreover, I have *two* mts that are meant to be concatenated. One has start_time=1.06, the other start_time=7019.74 [17:46:30 CEST] <viric> how can I use ffmpeg so the two files look like a single recording in input? [17:47:56 CEST] <durandal_1707> there's concat demuxer and filter [17:48:33 CEST] <viric> but will that pay attention to those start_time/start_pts? [17:51:10 CEST] <durandal_1707> I guess, at least demuxer [17:53:17 CEST] <viric> kdenlive makes a short quirk on file switch, while concatenating [18:35:52 CEST] <yangxuan8282_> @furq I read the config.log, the last part is: fatal error: interface/mmal/mmal.h: No such file or directory [18:37:43 CEST] <yangxuan8282_> then I copy the mmal directory to the chroot: sudo cp -a /opt/vc/* /home/alarm/chroot/alarm/opt/ , but still got same issues [19:42:10 CEST] <ozette> what's the mimetype of hls segmented videos .m3u(8)? [19:46:10 CEST] <c_14> Apparently application/vnd.apple.mpegurl [19:46:14 CEST] <c_14> https://www.ietf.org/id/draft-pantos-http-live-streaming-19.txt section 10 [19:47:13 CEST] <ozette> hmm [19:47:27 CEST] <ozette> what about application/x-mpegURL ? [19:47:35 CEST] <ozette> i assume both will work [19:49:17 CEST] <c_14> According to the draft that's not valid. [19:50:38 CEST] <c_14> It's either application/vnd.apple.mpegurl or audio/mpegurl [19:52:56 CEST] <c_14> There may, however, be legacy reasons why that would be supported by some applications. [19:55:24 CEST] <ozette> hm ok [20:17:14 CEST] <jookiyaya> big news: zen destroys intel http://static2.gamespot.com/uploads/scale_super/460/4601080/3046688-8311357… [20:20:42 CEST] <Chloe> There are three types of lies; lies, damned lies, and benchmarks. [20:21:16 CEST] <jookiyaya> so benchmark scores mean nothing? [20:22:23 CEST] <c_14> Never trust any statistic you didn't manipulate yourself. [20:23:37 CEST] <Chloe> jookiyaya: they mean nothing, and you probably shouldnt post them for a third time. [20:24:08 CEST] <jookiyaya> chloe of course they mean something [20:31:32 CEST] <iive> jookiyaya: you already discussed that news here [20:32:13 CEST] <jookiyaya> iive i don't even remember you ever commenting on it [20:39:18 CEST] <DHE> amd's been behind intel for a long time now, how'd they catch up so quick? [20:42:45 CEST] <jookiyaya> i wouldn't say quick though [20:43:12 CEST] <iive> DHE: look the discussion from 6-7 hours ago. [20:53:18 CEST] <DHE> iive: will do later, gotta find if I have an ffmpeg bug or if I've screwed something up... [21:34:26 CEST] <DHE> av_interleave_write_frame() is supposed to buffer packets from all streams such that 'ffprobe -show_frames' would show all frames in strictly increasing DTS order to the whole file, not just each independent stream. is that right? [21:40:39 CEST] <DHE> the reason I ask is I need output that is so "cleaned" but ffmpeg isn't give it to me [22:00:30 CEST] <DHE> am I wrong? [22:37:46 CEST] <chris_> Hi! Can someone please help me with h264_nvenc? [22:38:04 CEST] <DHE> that's not how it works. state your question [22:39:12 CEST] <chris_> I get this error when using the encoder: [h264_nvenc @ 0x260a180] Failed to create nvenc instance: invalid version (15) [22:40:27 CEST] <DHE> you might be using an old (or new?) version of the SDK [22:40:51 CEST] <DHE> or an old/new version of ffmpeg. either way there was a version bump in the last few months [22:41:40 CEST] <chris_> It's supposed to be v 7.0. [22:42:07 CEST] <chris_> Just compiled today. [22:42:40 CEST] <chris_> with latest sources. [22:43:04 CEST] <viric> chris_: the nvenc headers you used may not match your GL driver version [22:43:45 CEST] <chris_> Ok. But it still compiles, though? [22:44:05 CEST] <DHE> it'll compile, but not run [22:44:32 CEST] <viric> chris_: the headers only describe the lib that will be used at runtime (your nvidia driver) [22:44:44 CEST] <viric> you are responsible to match the headers with the nvidia driver you run [22:45:34 CEST] <chris_> That makes sense, but sure how to accomplis that. I'm just a noob on coding. [22:45:41 CEST] <chris_> not sure [22:45:51 CEST] <viric> if you used the latest headers, just update your driver [22:51:10 CEST] <chris_> My nvidia driver is v 361.42. Is ther an "apt-get install nvidia-latest" that get the latest driver? [22:55:34 CEST] <viric> that sounds debianish. No idea. [22:57:09 CEST] <DHE> no, that's how nvidia numbers their drivers. [22:57:19 CEST] <DHE> though it is a little out of date now [23:04:28 CEST] <Prelude2004c> hey guys.. good day ... i am looking for some help here with something silly that i can't figure out... so i am splitting an input like " -filter_complex '[0:v]yadif,format=nv12,split=3[vout1][vout2][vout3]" and the output can't keep up with the input and its running at less than 1x... ok so to debug i did -filter_complex '[0:v]yadif,format=nv12,split=1[out1] and it did the same thing..only a single output . same problem. now when i [23:04:28 CEST] <Prelude2004c> took out the ,split=1[out1] -map '[out1]', everything went ok and it was 1x .. any ideas ? [23:17:26 CEST] <theTroy> Hi! I am trying to encode with hevc_nvenc codec - it works fine and does use the nvidia card. Unfortunately the resulting video is identified as 8-bit uncompressed RGB by ubuntu and is not played back by totem/vlc at all? [23:20:47 CEST] <Prelude2004c> theTroy, try nvenc_hevc .. it may be defaulting to a different encoder because you have it backwards [23:21:42 CEST] <theTroy> Prelude2004c: sorry I did the right way round - written it wrong here. My command line is : ffmpeg -i INPUT -c:v nvenc_hevc -profile high444p -pixel_format yuv444p -preset default -b:v 3000k -c:a copy OUTPUT [23:21:42 CEST] <theTroy> ]]] [23:21:50 CEST] <theTroy> apologies for the brackets [23:22:02 CEST] <Prelude2004c> that is the output format you set ? [23:22:07 CEST] <Prelude2004c> the -f [23:22:20 CEST] <theTroy> I dont have -f doh! thanks! [23:23:10 CEST] <theTroy> Prelude2004c: although here they do not suggest to use -f https://trac.ffmpeg.org/wiki/Encode/H.264 [23:23:26 CEST] <Prelude2004c> maybe... it was a suggestion :) [23:23:39 CEST] <theTroy> right, encoding now - we will see, the output was set as .avi [23:23:58 CEST] <Prelude2004c> maybe its not detecting the .avi [23:24:04 CEST] <Prelude2004c> so its picking some other default [23:24:12 CEST] <DHE> you don't want avi. ever [23:26:00 CEST] <Prelude2004c> DHE, you know this stuff well.... [23:26:04 CEST] <Prelude2004c> hey guys.. good day ... i am looking for some help here with something silly that i can't figure out... so i am splitting an input like " -filter_complex '[0:v]yadif,format=nv12,split=3[vout1][vout2][vout3]" and the output can't keep up with the input and its running at less than 1x... ok so to debug i did -filter_complex '[0:v]yadif,format=nv12,split=1[out1] and it did the same thing..only a single output . same problem. now when i [23:26:04 CEST] <Prelude2004c> took out the ,split=1[out1] -map '[out1]', everything went ok and it was 1x .. any ideas ? [23:26:09 CEST] <Prelude2004c> i am stuck on that one :( [23:26:37 CEST] <Prelude2004c> wonder if some bug i the code.. why would split=x[] actually slow down the output ( or input ) [23:26:40 CEST] <Prelude2004c> even when =1 [23:27:29 CEST] <DHE> you got multithreading enabled? [23:27:42 CEST] <Prelude2004c> yup.. -threads 0 [23:27:49 CEST] <Prelude2004c> you mean on the ffmpeg itself? [23:28:08 CEST] <Prelude2004c> --enable-pthreads yes i do [23:28:53 CEST] <Prelude2004c> strange thing is.. even 1 input causes an issue... when i don't use the split , works perfectly [23:29:07 CEST] <Prelude2004c> problem is i have to output 3 different bit rates [23:32:19 CEST] <theTroy> So yeah nvenc_hevc shows up as 24 bits RGB (RV24) in vlc when played back [23:32:25 CEST] <theTroy> and it actually plays back no video whatsoever [23:32:46 CEST] <theTroy> if I encode with non nvenc h265 with libav - it does play it back [23:34:22 CEST] <Prelude2004c> sorry " theTroy" i am still learning too [23:34:59 CEST] <theTroy> yeah np Prelude2004c hopefully someone else would be able to help [00:00:00 CEST] --- Mon Aug 8 2016

1 0

ffmpeg-devel.log.20160806
by burek 07 Aug '16

07 Aug '16

[01:04:14 CEST] <cone-954> ffmpeg 03Michael Niedermayer 07release/3.1:67f421fd77ec: ffplay: Fix invalid array index [01:04:15 CEST] <cone-954> ffmpeg 03Michael Niedermayer 07release/3.1:7c9ee83d2f30: avformat/oggdec: Fix integer overflow with invalid pts [01:04:16 CEST] <cone-954> ffmpeg 03Burt P 07release/3.1:43407bde3e47: avfilter/af_hdcd: small fix in af_hdcd.c where gain was not being adjusted for "attenuate slowly" [01:10:39 CEST] <bp0> cool. [02:26:08 CEST] <cone-954> ffmpeg 03Michael Niedermayer 07master:c1bfeda5a346: avcodec/ffv1enc: Fix assertion failure with non zero bits per sample [04:07:27 CEST] <cone-954> ffmpeg 03Michael Niedermayer 07release/3.1:54d48c8e901d: avcodec/ffv1enc: Fix assertion failure with non zero bits per sample [04:07:28 CEST] <cone-954> ffmpeg 03Anssi Hannula 07release/3.1:456cf87de934: avformat/hls: Fix regression with ranged media segments [04:07:29 CEST] <cone-954> ffmpeg 03Anssi Hannula 07release/3.1:3586c6868703: avformat/hls: Sync starting segment across variants on live streams [04:07:30 CEST] <cone-954> ffmpeg 03Anssi Hannula 07release/3.1:309fa24f361f: avformat/hls: Use an array instead of stream offset for stream mapping [07:42:34 CEST] <UltraBloxX> Hello. I would like to contribute an audio filter that I've ported into libavfilter. [08:50:42 CEST] <rcombs> UltraBloxX: send to the mailing list [08:50:57 CEST] <rcombs> use git format-patch or git send-email [18:08:26 CEST] <bp0> michaelni, hello. git push origin master --dry-run, resulted in: fatal: remote error: access denied or repository not exported: /ffmpeg [18:09:17 CEST] <michaelni> bp0, whats the url of the repo you use ? [18:09:45 CEST] <michaelni> its in .git/config [18:10:18 CEST] <bp0> url = git://source.ffmpeg.org/ffmpeg [18:10:32 CEST] <michaelni> git:// is always read only, [18:10:45 CEST] <michaelni> url = git@source.ffmpeg.org:ffmpeg should work [18:12:08 CEST] <bp0> Everything up-to-date [18:14:19 CEST] <bp0> I guess, now I can really do some damage :) [18:14:33 CEST] <bp0> so if I wanted to push that small patch converting INT to BOOL [18:14:48 CEST] <bp0> I must first copy that patch from my work branch to master [18:14:49 CEST] <bp0> then push [18:15:04 CEST] <jamrial> you're better keeping the git:// address for url and adding the ssh one as a separate pushURL line [18:15:53 CEST] <michaelni> bp0 "git cherry-pick [that hash number thing of a commit] " will pick a patch into your current branch [18:16:52 CEST] <michaelni> bp0, if you are unsure, i can push the patch too [18:17:25 CEST] <bp0> well, I would like to, for the experience [18:17:55 CEST] <bp0> so now, I've cherry picked it, and git push origin master --dry-run lists it. [18:18:23 CEST] <bp0> I can remove --dry-run then? [18:18:37 CEST] <michaelni> you can check with "git log -p [whatever dry run listed]" what it is that would be pushed [18:18:51 CEST] <michaelni> always good to double check that [18:19:17 CEST] <michaelni> git log of the hash range should list the single commit you cherry picked [18:19:42 CEST] <bp0> indeed, it is the same [18:19:52 CEST] <michaelni> dry run should have only listed one range (otherwise you would be pushing multiple branches) [18:20:12 CEST] <michaelni> if that all matches up you can run it without dry run [18:20:41 CEST] <cone-003> ffmpeg 03Burt P 07master:2ce985c049bc: af_hdcd: convert AVOptions from INT to BOOL [18:21:13 CEST] <bp0> That's a bit scary. [18:21:21 CEST] <bp0> Does everything look ok? [18:21:50 CEST] <michaelni> bp0, yes all looks fine [18:21:51 CEST] <bp0> Do I now reply to myself in the ML: "applied thx"? [18:22:13 CEST] <michaelni> yes [18:23:46 CEST] <bp0> grand [18:23:50 CEST] <bp0> :) [18:23:57 CEST] <bp0> Thank you [18:24:18 CEST] <michaelni> np [18:24:41 CEST] <bp0> jamrial, made the change you mentioned, seemed to work [18:24:54 CEST] <bp0> what is the advantage? [18:25:22 CEST] <BtbN> you can pull without unlocking your key. [18:25:46 CEST] <bp0> ah [18:52:31 CEST] <jamrial> michaelni: could you test https://github.com/jamrial/FFmpeg/tree/master ? [18:53:58 CEST] <jamrial> also check if you're ok with the new error message. it's a mix of the one in libav and the one we currently have [18:55:06 CEST] <jamrial> didn't want to keep the libav one as is because ours has some extra information theirs doesn't [18:59:47 CEST] <jamrial> for that matter, the commit after this one would essentially be a noop because we still haven't ported ffmpeg to codecpar [19:00:56 CEST] <jamrial> then the following two seem complex after a quick glance (our code differs way too much), so i'll leave them in more capable hands [19:55:34 CEST] <michaelni> jamrial, seems working fine [19:57:44 CEST] <jamrial> michaelni: ok, thanks [19:58:15 CEST] <cone-003> ffmpeg 03Anton Khirnov 07master:90944ee3ab79: avconv: refactor selecting an encoder [19:58:16 CEST] <cone-003> ffmpeg 03James Almer 07master:61da882cea45: Merge commit '90944ee3ab79081845ea1bd97eea475031ce0842' [21:17:23 CEST] <durandal_1707> we should remove bunch of seters and geters functions now that libav api compatibility is not enforced [21:17:55 CEST] <durandal_1707> and move such fields where they belong [21:18:34 CEST] <jamrial> durandal_1707: yes, that's scheduled for the next major bump [21:19:13 CEST] <durandal_1707> good [21:19:58 CEST] <jamrial> the set/getters can't be removed right away since they are public api, but will be deprecated and given the usual ~2 years before they are axed [22:47:05 CEST] <cone-003> ffmpeg 03Michael Niedermayer 07master:803c058a6f0c: avformat/mov: Enable mp3 parsing if a packet needs it [23:30:28 CEST] <cone-003> ffmpeg 03James Almer 07master:d950279cbfed: avcodec/ttadsp: cosmetics [00:00:00 CEST] --- Sun Aug 7 2016

1 0

ffmpeg.log.20160806
by burek 07 Aug '16

07 Aug '16

[01:14:12 CEST] <kode54> how do I instruct configure to disable all video codecs? [01:15:19 CEST] <drv> I don't think there is a way to specifically filter by codec type [01:15:35 CEST] <drv> (assuming you are asking about how to disable all video but not audio) [01:16:13 CEST] <kode54> I have --disable-decoders and explicitly enable a handful of codecs, but it still enables h264 decoding [01:18:54 CEST] <drv> I tried --disable-everything --enable-decoder=aac and I only get aac, no h264 [01:19:00 CEST] <drv> maybe one of the other codecs you enable selects h264? [01:22:43 CEST] <drv> if you are not disabling parsers, you probably get the h264 parser selecting h264 decoder automatically [01:35:24 CEST] <kode54> drv, does mov or mp4 container imply video codecs? [01:35:54 CEST] <kode54> I can't disable mpeg4video or h263 parsers, or h263/h264/mpeg1video/mpeg2video/mpeg4 decoders [01:37:14 CEST] <drv> you'll have to brave the maze of configure to find out for sure :) [01:39:11 CEST] <mrjoker> what is the difference between choosing "main" as profile and "none" as profile on x265 [02:21:39 CEST] <jpleau> with libavformat, I have an AVStream *, nb_frames is set to 0... this is a .mkv file, video stream is h264, audio (x2) are ac3. Do I need to do something special to be able to access the actual number of frames? [02:45:47 CEST] <jpleau> I used the duration from AVFormatContext and multiplied by the average framerate, seems to give a good enough number for what I want to do [03:01:45 CEST] <kepstin> jpleau: with most formats, you actually have to demux the file and actually count the number of frames you get :/ [04:17:17 CEST] <jpleau> kepstin: hm ok. I'll see if I can do that easily, maybe I can get around just looping with av_read_frame and incrementing (I'm still new to those libraries, going through examples and doxygen) [04:53:04 CEST] <jpleau> yay that seems to work, just needed to check packet.stream_index for the video stream \o/ [09:04:00 CEST] <Justice> Hi guys. I'm trying to ONLY remux audio back into a video stream. I am currently using the following command: ffmpeg -i audio.mp3 -i video.mp4 -c copy final_video.mp4 Is this correct or is there a better way to do it? [09:10:10 CEST] <Justice> Also, as a secondary question to that: the video file is in .MOV format and the audio file is .WAV (16-bit signed PCM). Does that command (or any better suggestion) mux everything together losslessly? [10:47:13 CEST] <hollabackatcha> so, ffmpeg cannot loop over playlists ? like ffmpeg -re -f concat -i cont.txt -c copy -f flv rtmp://localhost:1935/live/test but not exiting after it reaches end of playlist [12:54:18 CEST] <Spring> is there a dummy's guide to filter mapping? [12:54:57 CEST] <Spring> I have two filters set up currently and want to add a third but I'm not sure precisely how I should do it [12:55:11 CEST] <Spring> keep getting errors about it [13:03:51 CEST] <Spring> okay, seemingly making progress [13:04:15 CEST] <xochilpili> hi all [13:05:43 CEST] <xochilpili> i dont know where to ask this : i have a multiple videos, and i want to shared them using html5 in a webpage, in my library are many different extensions like: mp4, avi, mkv and so on, i have found and install adobe media server, but i dont understand if this will let me to do what i meant to [13:06:45 CEST] <xochilpili> so, is there a way to "transcode" "in the air" in order to users watch videos depending on device? using html5 tags or something [13:06:56 CEST] <xochilpili> im really lost in this task [14:19:45 CEST] <DHE> Spring: can you show a specific example? [14:20:29 CEST] <Spring> DHE, I think my original mistake was using colons/semicolons when it needed to be a comma [14:20:49 CEST] <DHE> xochilpili: on-the-fly transcoding is possible, but tends to not scale too well if you expect traffic. mp4 will likely be best supported, at which point it's just a question of what codec [14:20:57 CEST] <Spring> currently just working on converting integers to decimal for a batch script [14:21:01 CEST] <DHE> Spring: sounds right. semicolons for complex filtergraphs, commas most of the time otherwise [14:22:01 CEST] <Spring> do you know if the atempo filter accepts numbers like 1.250 and .750, or does it have to be strictly like the examples here: https://ffmpeg.org/ffmpeg-filters.html#atempo [14:22:53 CEST] <xochilpili> DHE, i dont get it, this is too weird , i have installed adobe media server and he expects a f4v file , where am i supposed to get it if i have almost a 500 gb of movies, tvshows and many homemade videos in different formats and codecs [14:22:59 CEST] <DHE> is says 0.5 to 2.0, so anything in between goes [14:23:02 CEST] <furq> Spring: it should work fine [14:23:06 CEST] <xochilpili> how does plex doit ? [14:24:11 CEST] <furq> plex transcodes everything [14:24:39 CEST] <DHE> times like this I love my e5-2690+ CPUs [14:25:00 CEST] <furq> flash media server is probably a bad choice for this [14:25:20 CEST] <furq> you'd need to remux/transcode everything to flv yourself afaik [14:26:55 CEST] <furq> and at that point you might as well just use http [14:27:19 CEST] <DHE> especially if you said HTML5 based video [14:27:25 CEST] <DHE> (which he did) [14:27:41 CEST] <furq> oh [14:27:47 CEST] <furq> does fms do that nowadays [14:28:09 CEST] <furq> apparently it does hls now [14:29:06 CEST] <DHE> unfortunately HLS is only natively supported by apple and apparently some mobile platforms. unless you get that javascript filter [14:30:01 CEST] <furq> xochilpili: https://helpx.adobe.com/adobe-media-server/dev/configure-dynamic-streaming-… [14:30:07 CEST] <furq> step 1: "encode media files" [14:30:19 CEST] <furq> so yeah this probably isn't what you want [14:31:01 CEST] <xochilpili> is it possible to transcode on the fly with apache? [14:31:24 CEST] <DHE> it can be done, but transcoding is hard on the CPU. how many users do you expect to see at a time? [14:31:25 CEST] <Spring> btw why is the atempo limited to that range? [14:31:40 CEST] <xochilpili> how does plex doit? icant believe that i have spent all night searching and reading and im not able to find a solution to this [14:31:47 CEST] <furq> why not just use plex [14:31:59 CEST] <xochilpili> DHE, not too much people [14:32:11 CEST] <Spring> introduces problems if you need to change the speed outside of that range, since it requires duplicating filters and working out the difference [14:32:19 CEST] <DHE> xochilpili: plex is targeted at small numbers of users. maybe 1 or 2 at a time [14:32:43 CEST] <xochilpili> furq, i cant use authentication on plex, and the library is a mess [14:33:04 CEST] <xochilpili> i cant search by genre or "organize library as i want" [14:33:52 CEST] <furq> it's open source isn't it [14:34:39 CEST] <xochilpili> is not [14:37:49 CEST] <xochilpili> i cant believe this [15:01:11 CEST] <kritzikratzi> hi! does the resampler have an option for nearest neighbor scaling? i tried setting linear_interp=1 before initializing the context, but that didnt change anything. [15:02:12 CEST] <kritzikratzi> basically i want the silliest possible resampler where a rectangle signal remains rectangular when i up/downsample (preferrably it wouldnt even change into a trapezoid, it would still be a harsh rectangle) [15:02:51 CEST] <kritzikratzi> ps. im working talking about the swresample library, not options for ffmpeg.exe [15:13:14 CEST] <furq> Kei_N: -sws_flags neighbor [15:13:19 CEST] <furq> oh nvm [15:14:37 CEST] <furq> also i thought irssi's nick completion was smarter than that [15:18:26 CEST] <furq> kritzikratzi: https://www.ffmpeg.org/doxygen/3.1/structSwsContext.html#a4bad42bdd38e916f0… [15:18:37 CEST] <furq> SWS_POINT should be the same as -sws_flags neighbor [15:20:55 CEST] <kritzikratzi> furq: i meant swresample, not swscale [15:21:36 CEST] <kritzikratzi> in the meantime i just messed with all the params, setting filter_size to low value seems to do what i want av_opt_set_int(swr_context192, "filter_size", 2, 0); [16:16:16 CEST] <intracube> hi, is there a video filter that can combine X number of input frames? [16:17:58 CEST] <intracube> I'm trying to do a timelapse/speedup effect and looking for something better than dropping frames, since that will give a stuttery result [16:33:27 CEST] <Spring> so I've run into a problem [16:33:55 CEST] <Spring> when combing start and end times for an output using -ss to [16:34:21 CEST] <Spring> along with changing the speed of the video it gets the wrong timestamp [16:35:20 CEST] <Spring> the speed seems to be affecting the resulting timestamp, as if rather than trimming at 1:04 of the original video it's trimming 1:04 of the stretched video, even though the filters come after the timestamp commands [16:35:37 CEST] <Spring> or something like that, it's wildly different [16:36:59 CEST] <Spring> oh snap, this is because I'm changing the PTS isn't it? [16:39:28 CEST] <Spring> how do I compensate for this? I need the video to be trimmed first [16:41:46 CEST] <mosb3rg> sup fellas, so i have been working towards using a cookie to dump a live feed, now this technique works fine on my local linux machine. but when i take it to my server or windows vps so i tried linux and windows ffmpeg latest builds. only the local linux machine is able to decrypt the key based on the cookie. [16:42:24 CEST] <mosb3rg> so i tried checking to see if the cookies differ on each system, even though they do.. each cookie works for me locally here on linux and i can dump and relay fine. no decryption issue, but on the servers or anywhere else i am trying i see error 400 [16:43:08 CEST] <mosb3rg> so i assumed, it perhaps was some type of blocking of the hostmask or user agent, or anything like this ip restrictions... but im not the primary account holder, also the video plays fine on the intented website in each location. [16:43:57 CEST] <mosb3rg> so it only refuses to decrypt and throws the 400 error when i try it elsewhere.. im met without alot of practical reasoning why this keeps happening. i tried custom user agent it made no difference, and at the same respect, it didnt matter locally here on linux it just worked.. [16:47:23 CEST] <Spring> is the trim filter identical to the -ss and to commands? [17:27:43 CEST] <Spring> is anyone active to check the correctness of a command for me? [17:28:21 CEST] <Spring> http://pastebin.com/K94iHm4g [17:29:05 CEST] <Spring> when using that ffmpeg throws an error: 'No such filter: ''. Error initializing complex filters. Invalid argument.' [17:36:15 CEST] <durandal_1707> remove last comma [17:41:11 CEST] <Spring> durandal_1707, thanks, didn't notice that. However after removing the comma I'm now encountering an error mid encode 'Failed to initialize encoder: Invalid parameter. Additional information: [libvpx-vp9 @ 0000000001f0e460] Additional information: rc_twopass_stats_in.buf not set.' [17:41:49 CEST] <Spring> script was working fine before trying to add the trim filter [17:44:26 CEST] <DHE> sounds like we need to see the WHOLE commandline [17:53:06 CEST] <Spring> DHE, http://pastebin.com/sLzqn8L7 [17:55:02 CEST] <DHE> your 2pass logfile is the input file? uhh... [17:56:10 CEST] <Spring> DHE, the parameters at the top had to be manually added since my batch script hides them [17:56:34 CEST] <Spring> all the values are from variables, so the second pass input is actually a variable [17:57:32 CEST] <Spring> the log file is the name of the input not the actual input [17:57:46 CEST] <Spring> so it outputs to test.mp4.log [17:57:56 CEST] <Spring> not test.mp4 [18:01:39 CEST] <Spring> okay, so tested using seconds only values [18:01:46 CEST] <Spring> which /did/ work [18:02:20 CEST] <Spring> though it's still an incorrect result. Is trim choking on the colons in the timestamps? [18:15:14 CEST] <Spring> according to the docs the trim filter start and end are 'expressed as time duration specifications' yet every example I've seen in search results uses a different syntax [18:15:56 CEST] <Spring> (a syntax without colons) [18:16:03 CEST] <furq> you might need to escape the colons [18:33:17 CEST] <Spring> furq, so tried escaping the colons with 1-5 backslashes as per docs, and an error states: no such filter: ''. Error initializing complex filters. Invalid argument. Frustrating. [18:44:39 CEST] <Spring> if I escape both the colons and commas I get this error, which states it thinks the following scale command is part of the trim command https://a.uguu.se/Nx14kUpFavCX.png [18:44:55 CEST] <Spring> using a single backslash [18:46:58 CEST] <Spring> *sigh* all I wanted to really fix is being able to trim the video at the correct timestamps /before/ the setpts filter is applied, so as to cut the correct parts of the video [18:51:35 CEST] <wallbroken> guys, do you know x264vfw ? [18:53:58 CEST] <durandal_1707> who still use vfm? [18:54:43 CEST] <wallbroken> who uses windows [18:57:10 CEST] <durandal_1707> use dshow [18:57:45 CEST] <JEEB> that would work if his lulzy capture device supports dshow or vfw-over-dshow [18:58:02 CEST] <JEEB> because that's the only reason I can see using VFW at this point. capture devices that only are available through VFW or something [19:03:38 CEST] <wallbroken> JEEB, i needed to encode a .VOB in mkv, so i needed h264 and i used virtualdub, but virtualdub supports x264vfw [19:04:15 CEST] <wallbroken> so i used virtualdub+x264vfw, what do you think about? [19:04:36 CEST] <wallbroken> and also, added a plugin to virtualdub to support reading of vobs [19:04:50 CEST] <durandal_1707> it works? [19:05:02 CEST] <wallbroken> when i did it, in 2013, yes [19:05:08 CEST] <wallbroken> now i need to do it again [19:05:51 CEST] <durandal_1707> why you use virtualdub? [19:06:23 CEST] <wallbroken> because is a masterpiece on windows old school users [19:07:08 CEST] <durandal_1707> what filters do you use? [19:07:28 CEST] <wallbroken> when i needed it, only resize [19:08:06 CEST] <wallbroken> now in windows 10 i have ubuntu bash, so i could use something in linux environment, if it's better [19:09:10 CEST] <durandal_1707> well virtualdub works under wine good enough for me when doing RE [19:09:53 CEST] <furq> i'm sure i've heard of some other software which can convert from vob to mkv [19:09:57 CEST] <furq> if only i could remember what it was called [19:11:35 CEST] <durandal_1707> mencoder [19:12:49 CEST] <wallbroken> furq ffmpeg do it? [19:12:55 CEST] <furq> of course it can [19:13:03 CEST] <wallbroken> you can set quality factor? [19:13:15 CEST] <furq> of course you can [19:13:33 CEST] <wallbroken> and between ffmpeg and vdub, which one is better? [19:13:55 CEST] <wallbroken> for "better" i mean, better quality on output file [19:13:58 CEST] <furq> well only one of those can actually export to mkv [19:14:31 CEST] <durandal_1707> quality is encoder job... [19:15:56 CEST] <furq> i can't believe virtualdub is still going [19:16:11 CEST] <furq> it must be at least ten years since i last used it [19:16:22 CEST] <furq> since it became apparent that it was never going to support any decent muxers [19:16:25 CEST] <wallbroken> furq, what wrong with that? [19:16:38 CEST] <furq> it only exports to avi and it uses vfw codecs [19:44:12 CEST] <wallbroken> i have a dvd to convert, and this dvd has some video menu [19:44:23 CEST] <wallbroken> to be selected by dvd player remote controller [19:44:45 CEST] <wallbroken> is like a chapter [19:44:57 CEST] <wallbroken> is there some way to save it into the video file? [19:54:52 CEST] <wallbroken> furq, but internal plugins like vfw can export in mkv, mp4 and many others [19:55:01 CEST] <wallbroken> then you need to remux with audio [20:12:32 CEST] <Justice> Hey all. Was in yesterday and asked a question but had to go so I will re-ask: I'm trying to ONLY remux audio back into a video stream. I am currently using the following command: ffmpeg -i audio.mp3 -i video.mp4 -c copy final_video.mp4 Is this correct or is there a better way to do it? [20:46:12 CEST] <c_14> You'll probably want explicit mappings, in your case -map 0:a -map 1:v -map 1:s [21:34:59 CEST] <Justice> c_14: is mapping needed when there is only 1 audio stream and 1 video stream? [21:36:02 CEST] <c_14> no\ [21:36:17 CEST] <c_14> ffmpeg will by default map 1 video, 1 audio and 1 subtitle stream [21:39:48 CEST] <Justice> ok awesome. and that command muxes them together losslessly? [21:40:10 CEST] <c_14> as long as you keep -c copy, yes [21:40:57 CEST] <Justice> last question for now, when i start the muxing process it says that to not use a specific codec or something b/c it is defunct [21:41:09 CEST] <Justice> do i have to worry about that error message [21:41:27 CEST] <c_14> hmm? [21:41:33 CEST] <c_14> What's the exact message [21:41:36 CEST] <Justice> sec ill get the exact msg [21:43:26 CEST] <Justice> Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead [21:47:09 CEST] <Justice> i'll be back in a bit, c_14 if i'm not here just pm me [21:48:06 CEST] <c_14> Justice: ignore that [21:48:12 CEST] <c_14> It's ffmpeg-internal [21:48:17 CEST] <c_14> Will be fixed eventuall" [21:48:19 CEST] <c_14> Shouldn't break anothyng [21:48:23 CEST] <c_14> +y [22:18:41 CEST] <wallbroken> i've created a 600 mb mkv file from a 3.2 gb vob [22:19:00 CEST] <wallbroken> 3.2 vob file is fast in changing position in timeline [22:19:07 CEST] <wallbroken> mkv is veryslow [22:19:24 CEST] <wallbroken> 5 - 10 seconds when i'm lucky [22:19:31 CEST] <wallbroken> vlc crash when i'm unlucky [22:29:49 CEST] <thebombzen> I was told that "nut" is a container that can hold anything that FFmpeg can produce. is there any downside to using "nut" as a temporary output format? [22:45:19 CEST] <durandal_1707> may not be supported by others [22:47:08 CEST] <DHE> for a temporary format the concern is information is not lost. like that the timestamps don't lose any resolution [23:13:18 CEST] <Justice> ok thanks c_14 [23:18:52 CEST] <Justice> omg i feel like such a nuisance....does ffmpeg automatically use all cpu cores and threads by default or do I have to enable that? [23:19:14 CEST] <soulshock> it will use them all by default [23:19:38 CEST] <Justice> ok perfect. should be all set then (: [23:23:25 CEST] <soulshock> Carbots er for cool [23:23:31 CEST] <soulshock> doh wrong chat [00:00:00 CEST] --- Sun Aug 7 2016

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