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April 2016

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ffmpeg-devel.log.20160419
by burek 20 Apr '16

20 Apr '16

[00:04:35 CEST] <pfelt> ubitux: just found where it processes teh commands libavfilter/avfiltergraph.c:avfilter_graph_send_command() looks like it'll work to pass to all filters with the same name so i can use the instance parameter to set it [00:04:54 CEST] <pfelt> only other question is. is there a way to send commands from an external process? i haven't found a way yet [00:06:36 CEST] <durandal_1707> z filter something [00:06:50 CEST] <durandal_1707> need another lib [00:11:00 CEST] Action: pfelt 's google foo isn't strong today [00:19:40 CEST] <pfelt> durandal_1707: zmq filter [00:49:38 CEST] <pfelt> is there any rhyme to the name of a filter? (f* vs vf* vs af* etc) [01:25:01 CEST] <Compn> pfelt : Video Filter v Audio Filter [01:25:06 CEST] <Compn> vf v af [01:25:29 CEST] <Compn> if thats what you were asking :D [08:02:07 CEST] <kurosu> the wma lossless samples are 81KB zipped. Should I upload them somewhere, or is sending them to the mailing list ok ? (I guess the zip would be stripped from the mail) [08:23:53 CEST] <ubitux> h [08:23:57 CEST] <ubitux> oups [08:35:55 CEST] <rcombs> so apparently AVFoundation sometimes decides to give us a 24-bit signed integer low-aligned (in 32-bit) stream [08:36:03 CEST] <rcombs> as opposed to the usual 32-bit float [08:36:31 CEST] <rcombs> no idea why, and I've been poking at supporting it with no luck [08:36:49 CEST] <rcombs> seems to be random [08:37:06 CEST] <rcombs> same device switches between giving 24-bit aligned int and 32-bit float [08:37:17 CEST] <rcombs> (not during a single session, of course) [09:31:57 CEST] <nevcairiel> sounds like a reliale api [10:47:57 CEST] <JEEB> heh, coded_width/height case :/ [10:52:19 CEST] <j-b> 'morning [11:04:40 CEST] <rcombs> now I'm curious what it uses coded_width/height for [11:09:31 CEST] <nevcairiel> nothing of consequence it appears [11:11:41 CEST] <nevcairiel> it does however check if its not null for some reason [11:11:56 CEST] <nevcairiel> anyway exporting it in the compat block does no harm [11:15:44 CEST] <wm4> is it a good idea to parse h264 extradata (sps in particular) on init to check the stream's profile? [11:16:26 CEST] <nevcairiel> i do that [11:17:42 CEST] <wm4> the fact that libavformat attempts to extract extradata even from annex b streams makes this attractive [11:22:24 CEST] <wm4> some decoders return EAGAIN [11:22:34 CEST] <wm4> this conflicts with the new decode API [11:22:39 CEST] <wm4> bleh [11:44:15 CEST] <mateo`> i'm wondering ... is the old bsf api still available ? (i just saw the audiotoolbox related patch) [11:46:12 CEST] <wm4> it's deprecated, which means it'll still work for a while [11:46:48 CEST] <nevcairiel> but any internal usage in ffmpeg libraries or tools should be updated rather sooner than later [11:47:11 CEST] Action: wm4 wonders if there's any API ffmpeg.c uses that is NOT deprecated [11:48:49 CEST] <mateo`> wm4: ok, i'm asking because the mediacodec decoder uses it, and i don't want it to be broken. [11:49:25 CEST] <wm4> you should update it [11:49:38 CEST] <mateo`> I don't have time to do the porting right now, but it's planned for something within 2~3 weeks [11:50:30 CEST] <wm4> that should be fine [11:50:31 CEST] <mateo`> it's not too far away in time ? [11:50:38 CEST] <mateo`> ok [12:14:54 CEST] <rcombs> moving to the new API is pretty simple; I only had trouble because I had to screw around with extradata and because I uncovered a couple semi-related issues at the same time [12:15:23 CEST] <rcombs> though it could be tricky if using a filter that isn't 1:1 [12:15:26 CEST] <nevcairiel> since it uses extradata it should make that easier than the old compat hacks with avctx [12:15:46 CEST] <nevcairiel> since its native codecpar now [12:16:19 CEST] <rcombs> I mean in audiotoolbox [12:16:44 CEST] <nevcairiel> ah well [12:20:28 CEST] <rcombs> for autobsf it'll be awkward around 1:N bsfs, if those end up existing [12:20:41 CEST] <nevcairiel> we only have a N:1 bsf so far [12:20:51 CEST] <nevcairiel> the old api didnt really allow for 1:N, so =p [12:21:12 CEST] <rcombs> guess I'll assume they don't exist for now [12:21:23 CEST] <rcombs> what's N:1? [12:21:29 CEST] <nevcairiel> vp9 superframe [12:21:31 CEST] <rcombs> the VP9 one? [12:21:32 CEST] <rcombs> ah [12:21:44 CEST] <nevcairiel> it sometimes eats multiple frames and then packs them into one [12:21:55 CEST] <rcombs> we packed B-frames now? [12:22:04 CEST] <nevcairiel> vp9 doesnt have b frames [12:22:23 CEST] <rcombs> </bad_joke> [12:22:41 CEST] <nevcairiel> speaking about b-frames, the mpeg4 bframe unpack bsf could in theory be implemented as 1:N [12:22:48 CEST] <rcombs> also, wait, VP9 doesn't have B-frames? [12:22:50 CEST] <nevcairiel> but because its old, it somehow does that with internal buffering [12:23:18 CEST] <nevcairiel> rcombs: not in the classic (mpeg) definition of a B frame, no [12:23:50 CEST] <rcombs> I assume it has frame reordering [12:23:53 CEST] <nevcairiel> no [12:23:58 CEST] <nevcairiel> thats the shit it wanted to avoid [12:24:11 CEST] <rcombs> huh, what's it do to compensate [12:24:13 CEST] <wm4> wait what, really [12:24:30 CEST] <wm4> no timestamp reordering fun and such? [12:24:32 CEST] <rcombs> just picks up efficiency gains elsewhere? [12:24:52 CEST] <nevcairiel> let me see if i can find BBBs explanation from some weeks ago [12:24:56 CEST] <rcombs> I guess that explains all my WTFs looking at the lavc/libvpx files and wondering where all the timestamp handling is [12:25:55 CEST] <rcombs> wm4: "where's the fun in that?" [12:26:14 CEST] <wm4> seriously [12:26:45 CEST] <wm4> so I guess these "golden frames" (or wtf it was) act as b-frames, and since they're never output, no user-visible reordering is needed? [12:28:24 CEST] <atomnuker> I think B-frames were patented from MPEG2 days until like 2 or so years ago [12:28:38 CEST] <rcombs> lovely [12:31:37 CEST] <nevcairiel> http://forum.doom9.org/showthread.php?p=1756682#post1756682 [12:31:39 CEST] <nevcairiel> found it [12:32:57 CEST] <nevcairiel> but yeah it comes down to using golden frames/altrefs if it really helps efficiency, and then add a inter frame full of skip blocks in the time position where the output is supposed to be [12:33:02 CEST] <nevcairiel> that avoids the headache of re-ordering [12:34:08 CEST] <rcombs> I'd imagine that adds a little overhead, but it's worth it in the lack of pulling out hair [12:34:44 CEST] <nevcairiel> the advantage is that you could pre-process those invisible ref frames to be better references [12:35:44 CEST] <rcombs> ah, nice [12:37:17 CEST] <rcombs> so who remembers Perian [12:38:58 CEST] <wm4> wasn't it dshow on osx [12:39:06 CEST] <wm4> ffdshow I mean [12:39:14 CEST] <nevcairiel> kind of, the devs gave it up a few years back [12:41:12 CEST] <rcombs> so perian provided an ac3 decoder pre-10.11 (when the OS started shipping with it) [12:41:26 CEST] <rcombs> and I guess it breaks with audiotoolboxdec [12:42:09 CEST] <rcombs> so I was poking at explicitly asking for the Apple decoder in audiotoolboxdec (there's an API to pass in a "manufacturer") [12:42:33 CEST] <rcombs> but apparently they advertise the decoders as manufacturer=appl just like the official ones [12:43:10 CEST] <wm4> lol! [12:43:21 CEST] <rcombs> so eh whatever [12:43:33 CEST] <nevcairiel> tell people that they screwed themself [12:43:33 CEST] <nevcairiel> :D [12:43:35 CEST] Action: rcombs tells people to uninstall it [12:43:56 CEST] <rcombs> or, well, the ac3 one is marked properly, actually [12:44:08 CEST] <rcombs> but all the others (mp1, mp2, etc) are marked "appl" [12:44:46 CEST] <rcombs> and I don't feel particularly inclined to add the handling for this if it's not reliable [12:45:41 CEST] <rcombs> meanwhile, the eac3 decoder actually supports 7.1 (unlike the lavc one) [12:46:18 CEST] <nevcairiel> why is ordinary 7.1 not supported anyway, or are we talking about the dependent stream business [12:46:25 CEST] <rcombs> the dependent stream stuff [12:46:29 CEST] <durandal_1707> Improve lavc [12:46:31 CEST] <rcombs> there's no "ordinary" 7.1 afaik [12:46:42 CEST] <rcombs> always involves dependent streams [12:47:02 CEST] <rcombs> but you have to tell it the channel count ahead-of-time, and I'm using lavc's ac3 header parser to determine that [12:47:22 CEST] <rcombs> if you tell it there are 6 channels then it'll decode the 5.1 mix and give you that [12:47:32 CEST] <rcombs> and doesn't tell you there are actually 8 in any way [12:48:00 CEST] <nevcairiel> from what I can tell eac3 should support 7.1 without magic [12:48:30 CEST] <rcombs> I'm not 100% sure on it, but I know all my samples use dependent substreams to code 7.1 [12:48:35 CEST] <wm4> what about those ac3+eac3 in the same stream nightmares? [12:48:42 CEST] <nevcairiel> probably because they are all from Blu-ray [12:48:47 CEST] <wm4> lol [12:48:47 CEST] <rcombs> which makes some sense since it's supposed to be backwards-compatible to some degree [12:49:37 CEST] <rcombs> like, devices are supposed to be able to strip the dependent substreams and convert the header to a standard ac3 one and shove that in an SPDIF packet [12:50:01 CEST] <nevcairiel> HD DVD had eac3 as a fully mandatory format, there may be HDDVD EVO samples with eac3 7.1 [12:50:16 CEST] <nevcairiel> blu-ray made like no formats mandatory other than core ac3 and dts [12:50:37 CEST] <nevcairiel> hence the backwards compatible hackery for eac3 7.1 [12:51:00 CEST] <rcombs> poking around the AudioToolbox binary, I found there's apparently a function that parses an (E)AC3 frame and returns the channel count and sampling rate [12:51:06 CEST] <rcombs> but it's private API because ?????? [12:51:38 CEST] <nevcairiel> from what I can tell, supporting this in lavc wouldnt be insanely hard either, all the coding tools seem to be present mostly [12:51:49 CEST] <rcombs> nevcairiel: I don't mean the BD thing with an AC3 frame and then an EAC3 dependent substream frame [12:51:52 CEST] <nevcairiel> but its so rare in commercial content that personally I just dont care [12:52:21 CEST] <rcombs> I mean devices are supposed to be able to repack an EAC3 frame into AC3 [12:52:24 CEST] <nevcairiel> rcombs: ac3 and eac3 have slightly different bitstream syntax, modifying an eac3 frame to be ac3 sounds silly otherwise [12:53:10 CEST] <rcombs> "Dolby Digital Plus is a functional superset of Dolby Digital, and decoders include a mandatory component that directly converts (without decoding and re-encoding) the Dolby Digital Plus bitstream to a Dolby Digital bitstream (operating at 640kbit/s) for carriage via legacy S/PDIF connections (including S/PDIF over HDMI) to external decoders (e.g. AVRs, etc.)." [12:53:46 CEST] <nevcairiel> eac3 can even have higher btirates than that, must be some arcane magic to get rid of some [12:53:57 CEST] <rcombs> it's just dropping the dependent substreams [12:54:08 CEST] <rcombs> and repacking in the ac3 bitstream format [12:54:08 CEST] <nevcairiel> you can have higher bitrates in 5.1 as well [12:55:02 CEST] <wm4> "S/PDIF over HDMI" haha [12:55:08 CEST] <wm4> I like the sound of that [12:55:09 CEST] Action: rcombs shrugs [12:55:30 CEST] <wm4> users tend to dislike if passthrough is called spdif, for whatever reasons [12:55:55 CEST] <rcombs> muh initialisms [12:56:07 CEST] <rcombs> very scary [12:57:10 CEST] <rcombs> nevcairiel: but yeah, I took a look at doing dependent substream decoding for eac3 a while back [12:57:22 CEST] <rcombs> saw that it'd have to involve some awkward logic to remap channels [12:57:39 CEST] <wm4> does AT not output planar? [12:57:42 CEST] <nevcairiel> at least we have planar channels now, that makes such things miles easier [12:57:44 CEST] <rcombs> like, you'd decode one frame, then move the channel data around, then decode the substream [12:57:56 CEST] <nevcairiel> you can just flip pointers [12:57:59 CEST] <rcombs> yeah, definitely easier that way [12:58:12 CEST] <rcombs> and then I decided I didn't care enough to keep going [12:58:18 CEST] <rcombs> since literally 0 people have requested this from me [12:58:55 CEST] <rcombs> which is less than the number of people who have requested support for DTS-in-FLAC [12:59:01 CEST] <rcombs> (1 person) [12:59:35 CEST] <wm4> who created such a file? the ISIS? [12:59:41 CEST] <Daemon404> 1:58] <+rcombs> and then I decided I didn't care enough to keep going <-- this speaks to me on a spiritual level [13:00:15 CEST] <rcombs> ran out of ~effort~ [13:00:17 CEST] <nevcairiel> i tell people to go screw themself if they come with shit like dts in flac [13:00:26 CEST] <rcombs> me too [13:00:31 CEST] <nevcairiel> they should be happy that i support dts in wav [13:01:24 CEST] <rcombs> DTS in FLAC: take a DTS CD, read it like a CDDA (i.e. treat the data as if it were PCM audio), and encode it to FLAC [13:01:29 CEST] <Daemon404> isnt dts in wav more or less apdif? [13:01:33 CEST] <nevcairiel> no [13:01:35 CEST] <Daemon404> spdif* [13:01:36 CEST] <nevcairiel> its just dts in wav [13:01:43 CEST] <nevcairiel> no spdif packing [13:01:49 CEST] <Daemon404> oh. [13:01:57 CEST] <nevcairiel> its from dts audio cds [13:01:57 CEST] <rcombs> playback involves decoding the FLAC, then putting the FLAC decoder's output through a DTS decoder [13:02:22 CEST] <Daemon404> ..... [13:02:27 CEST] <Daemon404> sounds japanese [13:02:36 CEST] <Daemon404> like their magic aac to wav thing [13:02:38 CEST] <rcombs> apparently this is all so you can tag the FLAC files [13:02:52 CEST] <Daemon404> because mka doesnt exis [13:02:57 CEST] <Daemon404> t [13:03:03 CEST] <rcombs> Daemon404: most people call an "aac to wav thing" an "aac decoder" [13:03:13 CEST] <Daemon404> hah. but no. [13:03:18 CEST] <Daemon404> this is named fakeaacwav [13:03:21 CEST] <Daemon404> no. really. [13:03:39 CEST] <rcombs> what, it crams an AAC bitstream in WAV as if it were PCM? [13:03:47 CEST] <Daemon404> nope [13:04:02 CEST] <Daemon404> it does some weird conversion to wav such that it can losslessly transform back [13:04:03 CEST] <rcombs> explain [13:04:05 CEST] <Daemon404> pcm in wav. [13:04:16 CEST] <Daemon404> so it cam ne cut with avisynth [13:04:23 CEST] <Daemon404> amd converted back losslessly to aac [13:04:29 CEST] <nevcairiel> that sounds bizzare [13:04:33 CEST] <Daemon404> i can dig up the program when im home [13:04:43 CEST] <Daemon404> keeb knows what im talking about [13:04:43 CEST] <nevcairiel> it pads it with zeros to some pcm bitrate for cutting? [13:04:46 CEST] <Daemon404> jeeb even [13:04:58 CEST] <Daemon404> i dont know what it does besides magic [13:10:15 CEST] <JEEB> aac2wav, yeah. I remember that thing [13:10:48 CEST] <wm4> it could use spdif framing? [13:10:57 CEST] <JEEB> it basically packages AAC into PCM so you can cut it as if it's raw PCM [13:11:04 CEST] <JEEB> (within video frame boundaries at least) [13:11:23 CEST] <JEEB> so you could wavsource() it into avisynth and trim() [13:11:40 CEST] <JEEB> and then extract the AAC track once again with the same tool so you can mux it into a container [13:15:13 CEST] <Daemon404> http://aviutl.info/fakeaacwav-faw/ [13:15:18 CEST] <Daemon404> it is apparently still in use. [13:16:54 CEST] <Daemon404> ... and so is avoutl [13:17:01 CEST] <Daemon404> aviutl* [13:23:15 CEST] <JEEB> ugh, not at my e-mail client but can someone tell cehoyos in the PGS thread that since we now know how to handle it correctly, he shouldn't commit a correct-only-for-SD solution :V [13:23:38 CEST] <JEEB> since the YCbCr-to-RGB function is supposedly only correct for BT.601 [13:26:45 CEST] <rcombs> apparently adding support for non-PCM to avisynth would've been too hard? [13:26:54 CEST] <wm4> JEEB: send a mail to the ML with those lines [13:27:08 CEST] <wm4> rcombs: avisynth works on uncompressed data only [13:27:27 CEST] <rcombs> and evidently this would have been difficult to change [13:28:57 CEST] <JEEB> wm4: not doing top-posting seems to be problematic with my phone [13:29:05 CEST] <JEEB> and that's the only mail client I have on hand atm [13:29:15 CEST] <JEEB> but OK, I will try [13:30:40 CEST] <wm4> JEEB: no, I meant "sent" [13:31:03 CEST] <wm4> as in I've sent a mail [13:48:31 CEST] <JEEB> ah [13:48:33 CEST] <JEEB> dänks [14:39:58 CEST] <Daemon404> nevcairiel, wtf is chromium using coded_* for [14:40:11 CEST] <nevcairiel> dunno [14:40:16 CEST] <nevcairiel> but it checks if its not zero [14:40:33 CEST] <Daemon404> wut [14:41:05 CEST] <nevcairiel> we could just fill it in the legacy compat thing again and be done with it, its not really a big deal [14:41:35 CEST] <Daemon404> sure [14:41:38 CEST] <Daemon404> but they ought to fix it [14:41:46 CEST] <Daemon404> also i read their "system ffmpeg" patch [14:41:51 CEST] <Daemon404> wtf are they doing with AV*Internal [14:41:52 CEST] <nevcairiel> once they want to use codecpar they will need to anyway [14:52:29 CEST] <atomnuker> holy crap I managed to make clang crash [14:52:48 CEST] <atomnuker> it's asking me with all caps "PLEASE" to send a backtrace [14:52:57 CEST] <wm4> poor thing [14:57:03 CEST] <Daemon404> [FFmpeg-devel] FFMpeg/FFServer P2P streaming between ubuntu based embedded device and smartphone [14:57:11 CEST] <Daemon404> another example of ffserver's existence doing more harm than good [15:02:00 CEST] <wm4> yeah [15:02:16 CEST] <wm4> but it will be used as proof that users still have interest in ffserver and that it shouldn't be deleted [15:42:05 CEST] <cone-016> ffmpeg 03Martin Storsjö 07master:933dec0e29ec: file: Add an option for following a file that is being written [15:42:06 CEST] <cone-016> ffmpeg 03Derek Buitenhuis 07master:90eb24996913: Merge commit '933dec0e29ec4d2cb83474279a6c52d62fdb7310' [15:43:55 CEST] <cone-016> ffmpeg 03Lukasz Marek 07master:8833f1508b7b: opt: Add const to av_opt_next [15:43:56 CEST] <cone-016> ffmpeg 03Derek Buitenhuis 07master:c54d835e2f18: Merge commit '8833f1508b7b6afc3172a8017934a7a54428c686' [20:20:45 CEST] <cone-016> ffmpeg 03Michael Niedermayer 07master:564b4591bbe2: opt: Add av_opt_copy() [20:20:46 CEST] <cone-016> ffmpeg 03Derek Buitenhuis 07master:e344646c808c: Merge commit '564b4591bbe223bdc5f36a1131eaef103f23f5d0' [20:38:50 CEST] <tmm1_> what's the procedure for getting libav patches into ffmpeg? [20:39:15 CEST] <JEEB> tmm1: usually things get merged (esp. now that TEP2 has been merged) [20:39:26 CEST] <Daemon404> wait for them to be merged, or send them rebase on ffmpeg if you are impatient [20:39:35 CEST] <JEEB> pretty much that [20:39:37 CEST] <Daemon404> there is a merge backlog at the moment due to a very large merge a while back. [20:41:52 CEST] <tmm1> okay, if i want to submit to the list i can just cherry-pick onto master and fix any conflicts right? [20:41:59 CEST] <pfelt> speaking as a total noob here&.. could someone walk me through the TS stuff? i'm working on a modification to a filter and it's sorta working. now i'm getting "dropping frame x from stream y at ts z" which, if i'm understanding TS right, is expected as i'm not resetting the TS [20:42:06 CEST] <Daemon404> tmm1, and of course confirm it works [20:42:12 CEST] <Compn> and doesnt break fate.... [20:42:24 CEST] <Daemon404> that is part of "works" [20:42:35 CEST] <tmm1> cheers [20:42:37 CEST] <Compn> well "works 4 me" and "works" ... [20:42:53 CEST] <Daemon404> bug clsoed as STFUIMDREPPER [20:43:01 CEST] <Compn> i mean "feature works" v "works fate testing" [20:43:03 CEST] <Compn> nevermind. [20:43:08 CEST] <Daemon404> i know. [20:43:38 CEST] <durandal_1707> anyone to backport few commits for me? [21:48:45 CEST] <Angus> If there's no header for https://ffmpeg.org/doxygen/3.0/vdpau__h264_8c_source.html [21:49:10 CEST] <Angus> how can you call av_register_hwaccel(&ff_h264_vdpau_hwaccel) on it from external code? [21:51:15 CEST] <wm4> you can't [21:51:42 CEST] <wm4> hwaccels are registered like other codecs etc. [21:52:23 CEST] <Angus> av_register_all registers the codecs nicely [21:52:40 CEST] <Angus> but it doesn't seem like it calls av_register_hwaccel at all [21:53:10 CEST] <fritsch> works for us (in kodi) with ffmpeg 3.0 [21:53:38 CEST] <Angus> oh, totally missed a function call [21:53:42 CEST] <wm4> Angus: it does, maybe it's just not compiled in [21:53:42 CEST] <Angus> that solves the mystry. [21:53:50 CEST] <wm4> or that [22:07:37 CEST] <rcombs> Angus: by building as a static library and doing `extern AVHWAccel ff_h264_vdpau_hwaccel;` :P [22:07:49 CEST] <rcombs> (Perian did this for decoders for some reason) [22:08:14 CEST] <wm4> wat [22:11:55 CEST] <Angus> rcombs, D: [22:12:24 CEST] <rcombs> I didn't say it was a good idea [22:12:45 CEST] <atomnuker> oh hey VP9 finally got a spec officially published [22:12:58 CEST] <atomnuker> http://www.webmproject.org/vp9/ [22:13:05 CEST] <Angus> I did briefly try that, but realized very quicky that I'd end up including a significant part of FFMpeg into the program I'm building. [22:15:41 CEST] <Angus> https://ffmpeg.org/pipermail/ffmpeg-user/2012-July/008125.html [22:15:48 CEST] <Angus> is this still applicable? [22:16:36 CEST] <Angus> since https://trac.ffmpeg.org/wiki/HWAccelIntro [22:16:43 CEST] <Angus> seem to indicate otherwise [22:17:03 CEST] <JEEB> decoders and hwaccels are kind of different [22:17:14 CEST] <JEEB> one can use a decoder without external initialization or such [22:17:33 CEST] <JEEB> one cannot use a hwaccel without some sort of special poking to set the ground [22:18:43 CEST] <Angus> hmmm [22:18:45 CEST] <Angus> I see [22:18:47 CEST] <JEEB> ffmpeg (the cli app) has some support for the hwaccels in avcodec [22:19:12 CEST] <JEEB> but in general they require special code so they're not around in the cli apps [22:19:49 CEST] <Angus> it does look like this exists: [22:19:49 CEST] <Angus> https://www.ffmpeg.org/doxygen/3.0/ffmpeg__vdpau_8c_source.html [22:21:03 CEST] <wm4> yes, it demonstrates most basic API usage [22:21:43 CEST] <wm4> Angus: it got slightly simpler recently: http://git.videolan.org/?p=ffmpeg.git;a=blob;f=ffmpeg_vdpau.c [22:21:52 CEST] <wm4> (but uses API not in the 3.0 release, or any release) [22:22:38 CEST] <Angus> hmmm [22:24:14 CEST] <Angus> that file seems to be identical to the one found in 3.0's / folder [22:24:55 CEST] <wm4> consider getting a pair of new eyes [22:25:09 CEST] <wm4> look e.g. at vdpau_get_buffer [22:27:49 CEST] <Angus> ah, found my issue [22:27:50 CEST] <Angus> https://github.com/FFmpeg/FFmpeg/blob/master/ffmpeg_vdpau.c [22:27:52 CEST] <Angus> is not 3.0 [22:28:16 CEST] <wm4> yes, releases are just outdated snapshots that nobody cares about [22:28:26 CEST] <wm4> so all branches etc. point not to the release [22:30:40 CEST] <Angus> I'm may be suffering from the side effects of corperate branch management. [22:32:59 CEST] <fritsch> wm4: lol [22:34:14 CEST] <wm4> fritsch: it's true [22:34:29 CEST] <wm4> on the other hand, master is usually remarkably stable [22:34:45 CEST] <wm4> we don't even allow incompatible API changes etc. [22:39:58 CEST] <fritsch> wm4: yeah kodi is always on latest stable ... when filing bugs this one guy always says: reproduce with master ... [22:40:12 CEST] <fritsch> which is not funny then, if filing bugs for stable releases is not wanted / allowed [22:40:43 CEST] <wm4> lol [22:40:58 CEST] <wm4> cehoyos has his own policies anyway [22:41:32 CEST] <fritsch> for example last time with the aligned buffers [22:41:42 CEST] <fritsch> that was in no way reproducable with ffmpeg command line [22:41:54 CEST] <fritsch> so hard to file according to his rules [22:42:23 CEST] <wm4> "rules" [22:42:23 CEST] <fritsch> but btw. concerning these aligned buffers. Somehow the docu needs to make clear: if you supply user buffers, make sure that x, y, z [22:42:28 CEST] <fritsch> which is not documented [22:42:32 CEST] <wm4> I rarely report bugs because of that [22:42:51 CEST] <fritsch> we report all of them and when users find new ones ... we let the mreport it [22:42:52 CEST] <wm4> if I care I try to fix it myself or talk to someone who knows the code in question [22:43:08 CEST] <fritsch> yeah, that's the short cut, but developers don't always have time [22:43:20 CEST] <fritsch> and our bugs are often quite special [23:55:37 CEST] <Angus> wm4: you tested this here: http://git.videolan.org/?p=ffmpeg.git;a=commit;h=6b706ce85fa56564986211b99d… [23:56:00 CEST] <Angus> does vdpau_retrieve_data manage to get the decoded data back from GPU? [00:00:00 CEST] --- Wed Apr 20 2016

1 0

ffmpeg.log.20160419
by burek 20 Apr '16

20 Apr '16

[00:46:52 CEST] <Guest20457> guys, anywhere with examples i can learn more about codecs , bitrates and encoding using ffmpeg , its working but seems i have pixel issues, i need to understand the process better, any good documentations around? [00:48:43 CEST] <llogan> Guest20457: you mean anything other than the official documentation and the wiki? [00:49:04 CEST] <Guest20457> yep [00:49:20 CEST] <Guest20457> i can see alot about commands , how to stream and restream [00:51:30 CEST] <Guest20457> but for example, if i choose to stream h264 over udp , then recieve and encode and restream over rtmp or any other, what parameters should i worry about , from the source to destination [00:52:33 CEST] <Guest20457> i read about the latency at he wiki, still reading, but i guess there are to many explanations wish i do not feel i understand pretty well [00:55:27 CEST] <Guest20457> for example, i am using this at the capture pc side ( win 7 32) , ffmpeg -f dshow -i video="webcam" - video_size 1280x960 -framerate 18 -codec:v libx264 -pix_fmt yuv420p -qp 0 -preset fast -f mpegts udp://ip-of-local-server:1500 [00:56:07 CEST] <Guest20457> at the local server side i am playing with ffplay ( linux debian wheezy ) ffplay -i udp://ip-of-source:1500 [00:56:53 CEST] <Guest20457> but still having issues with pixel gettin malformed sometime, however i added -pix_fmt option for compatibility [00:58:35 CEST] <Guest20457> now if i get the video with bad pixel and shapes, as shown on ffplay, should i adjust something from the source side , were the capture is, or i can do it at the server sid ( fix, encode, stream ) ? [01:20:39 CEST] <DHE> as a rule udp:// is best suited for fixed LAN links with matched speeds, or for receiving only. ffmpeg will just spew UDP packets and the risk of packet loss is actually quite high [01:21:06 CEST] <DHE> start by using TCP instead and see if that helps [01:22:36 CEST] <Guest20457> thank you will do [01:33:43 CEST] <Guest20457> is ffplay reliable for such test? or its better to record with ffmpeg and then play to check? [01:43:52 CEST] <DHE> files on disk are always reliable, but usually a pain for streaming [01:45:05 CEST] <Guest20457> i guesse the best way is to know the source file/capture device verry well to optimise for best streaming, there is no general rule for this [01:47:28 CEST] <Guest20457> seems network issue, using same pc does work alot better [03:28:40 CEST] <FlyingJester_> How would I specify an offset to begin an input file at in the output? [03:29:08 CEST] <FlyingJester_> Not like -ss, but the opposite. Let's say I want an audio file to start 10 seconds in to the output file. [03:30:51 CEST] <furq> FlyingJester_: -itsoffset [03:31:15 CEST] <FlyingJester_> Ah, thanks, furq [03:37:24 CEST] <FlyingJester_> I can use values like "1:23" as time offsets, right? [03:41:40 CEST] <FlyingJester_> Let's say I have an audio file `file1.ogg' and a video file `file2.mp4', and I want the audio from file1 to start 4 minutes and 45 seconds into the output file. What's wrong with this: https://gist.github.com/FlyingJester/6b9996f7cffaf0b1024e438b5966e08a [03:41:51 CEST] <FlyingJester_> I get the audio from file1.ogg at the very start of the output file. [03:55:29 CEST] <furq> FlyingJester_: iirc you need to set a negative itsoffset on the video file [03:55:38 CEST] <furq> or use a positive ss on the audio file [03:56:02 CEST] <FlyingJester_> Hmm. Okay, I'll try that. [03:56:18 CEST] <FlyingJester_> Although setting ss on the audio file sounds like not what I want. [03:56:20 CEST] <furq> you can also replace -t 5:40 with -shortest [03:56:30 CEST] <FlyingJester_> I want the audio to come in after 4:45 of video. [03:58:50 CEST] <FlyingJester_> The -t in that should clip a few seconds off the audio before the video ends, which is what I want. [03:59:21 CEST] <furq> nvm then [03:59:44 CEST] <FlyingJester_> Setting a negative itsoffset just makes the video start in the middle. [04:03:09 CEST] <furq> oh fun [04:03:11 CEST] <furq> https://trac.ffmpeg.org/ticket/1349#comment:6 [04:03:51 CEST] <FlyingJester_> Well, at least it [04:03:58 CEST] <FlyingJester_> Well, at least it's not just me. [04:04:15 CEST] <FlyingJester_> I can just prepend some silcence to the audio track in audacity or something. [04:04:34 CEST] <furq> https://ffmpeg.org/ffmpeg-filters.html#adelay [04:05:28 CEST] <furq> i guess `-af adelay=285000|285000` should work [05:37:38 CEST] <johnnny22-afk> How can I make ffmpeg render a frame better, I often get a gray'ish image [05:38:59 CEST] <johnnny22-afk> on hevc content that is. [05:51:00 CEST] <pfelt> evening all. i'm looking at the movie filter and it looks like it only pulls video content from the specified file. if i want both video and audio content do i have to use both movie and amovie? [05:53:21 CEST] <pfelt> nm. i read further down in the docs. /me hangs his head in shame [11:56:07 CEST] <morissette> Any thoughts on how to convert this file to mp4 or something usable? https://bpaste.net/show/e5b0672633e2 [12:02:21 CEST] <Snoo20664> ffmpeg -i raw_video.svr libx264 out.mp4 [12:04:03 CEST] <furq> morissette: svr? [12:04:24 CEST] <morissette> furq, i have no idea; the only context I have is it is from a DVR [12:04:28 CEST] <morissette> Snoo20664, trying that [12:04:53 CEST] <morissette> Snoo20664, raw_video.svr: Invalid data found when processing input [12:05:20 CEST] <Snoo20664> yeah didn't think it would work heh [12:06:02 CEST] <Snoo20664> open raw_video.svr with mediainfo, click view -> text, and then copy the text and paste it to pastebin [12:06:10 CEST] <Snoo20664> or bpaste or whatever [12:06:26 CEST] <morissette> kk [12:07:16 CEST] <morissette> https://bpaste.net/show/b9294a3bc52a [12:07:24 CEST] <morissette> Thats a bloody useful tool [12:09:50 CEST] <morissette> So thoughts? [12:12:23 CEST] <morissette> Here is the file also if that helps for actual testing purposes: https://s3.amazonaws.com/observint/raw_video.svr [12:13:53 CEST] <furq> $ file raw_video.svr [12:13:53 CEST] <furq> raw_video.svr: data [12:13:55 CEST] <furq> well that's useful [12:16:44 CEST] <morissette> yeah thats what i started with a couple hours ago [12:16:45 CEST] <morissette> haha [12:16:59 CEST] <morissette> its raw bait car data [12:17:23 CEST] <morissette> and i was given one night to integrate into our existing product and convert to something playable :) [12:17:36 CEST] <morissette> b/c developers are "magic" [12:17:54 CEST] <morissette> sad thing is when I downloaded from dropbox on my iphone [12:17:59 CEST] <morissette> it works in vlc app [12:18:01 CEST] <morissette> no issue [12:18:22 CEST] <morissette> but vlc on linux says "missing header" [12:22:06 CEST] <furq> huh [12:22:15 CEST] <furq> vlc 2.0.5 plays the video, vlc 2.2 plays a silent audio track [12:22:24 CEST] <furq> neither of them say what container it is [12:23:17 CEST] <morissette> https://bpaste.net/show/caf6a76a568f [12:23:21 CEST] <morissette> hmmm [12:24:10 CEST] <Snoo20664> https://forum.videolan.org/viewtopic.php?t=23305 [12:24:23 CEST] <Snoo20664> people having the same issues a decade ago with no solution heh [12:24:30 CEST] <Snoo20664> everything i tried failed :Z [12:26:03 CEST] <morissette> :( [12:26:24 CEST] <morissette> ...must not break down and implement a windows solution... [12:28:17 CEST] <morissette> looks like i get to learn windows server 2012 in 2 hours and then implement some sort of microservice on it to transcode [12:28:46 CEST] <morissette> but its all clicking so shouldn't be too hard [12:29:03 CEST] <morissette> outside of figuring out how to make python run as a daemon on windows [12:30:31 CEST] <morissette> hmmm [12:30:32 CEST] <morissette> hopefuly [12:30:35 CEST] <morissette> http://stackoverflow.com/questions/1798429/any-open-source-fast-h263-decode… [12:36:58 CEST] <morissette> from ffmpeg -codecs: DEV.L. h263 H.263 / H.263-1996, H.263+ / H.263-1998 / H.263 version 2 [12:37:33 CEST] <furq> morissette: http://s000.tinyupload.com/index.php?file_id=82058814924573083559 [12:38:10 CEST] <morissette> hmmm [12:38:11 CEST] <morissette> .ts [12:38:26 CEST] <furq> it's not actually .ts, i remuxed it with vlc 2.0.5 [12:38:38 CEST] <furq> vlc raw_video.svr --sout file/ts://raw_video.ts [12:38:38 CEST] <morissette> [mharris@mori Downloads]$ file raw_video.ts [12:38:38 CEST] <morissette> raw_video.ts: MPEG transport stream data [12:38:44 CEST] <furq> well yeah the remuxed file is ts [12:38:47 CEST] <furq> i meant the svr isn't [12:39:48 CEST] <morissette> missing audio [12:39:54 CEST] <morissette> but video is better than nothing [12:40:08 CEST] <morissette> plus we keep the original for chain of evidence bs [12:40:35 CEST] <furq> http://sprunge.us/gKFD [12:40:44 CEST] <furq> i am reasonably certain that ffmpeg doesn't know how to demux it [12:41:02 CEST] <morissette> fair enough well I really appreciate it [12:41:16 CEST] <morissette> this will be a much more automate-able solution [12:41:29 CEST] <morissette> vs hacked together windows / freeware / conversion [12:42:11 CEST] <furq> idk you could probably get a raise on the back of being forced to install TURBOSOFT VIDEO CONVERTER ULTRA 7 [12:42:39 CEST] <furq> trial version, obviously [12:42:44 CEST] <morissette> haha no raises for me [12:43:13 CEST] <morissette> already got like an eighth of a million [12:43:31 CEST] <morissette> maybe a raise if i can get the audio and timestamp working in the new few hours [12:43:49 CEST] <morissette> i have this svr and a wmv file [12:44:04 CEST] <morissette> the wmv plays with timestamp and audio [12:44:26 CEST] <morissette> which i imagine they want both when viewing online when not a empty boring bait car vid [12:44:50 CEST] <morissette> you think the timestamp is another vid stream? [14:08:31 CEST] <andrey_utkin> Could please anybody look at this - concat-ing H.264 MP4 files encoded with "-x264opts stitchable=1" (option to ensure concat-ability) gives broken video: https://gist.github.com/andrey-utkin/1cc892661dc3216de7d4c6ff5f21ef9d [14:45:30 CEST] <andrey_utkin> fflogger !pb andrey_utkin [14:54:31 CEST] <otila> with latest x265 (r11428) and latest ffmpeg git, crf option does not have any effect, even though I get "x265 [info]: Rate Control / qCompress : CRF-15.0 / 0.60" [14:54:59 CEST] <otila> I get same file output with crf 15 and 25, except that x265 info line shows different crf [16:05:03 CEST] <DHE> When converting surround sound to stereo, I find the volume to be rather low. Besides inserting a volume filter, is there something that can help? I'm using the swr_convert() API. Everything is working fine otherwise [16:05:29 CEST] <DHE> google suggests messing with mixing coefficients, is that really the best option? [16:06:14 CEST] <andrey_utkin> Could please anybody look at this - concat-ing H.264 MP4 files encoded with "-x264opts stitchable=1" (option to ensure concat-ability) gives broken video: https://gist.github.com/andrey-utkin/1cc892661dc3216de7d4c6ff5f21ef9d [16:11:31 CEST] <durandal_1707> DHE: for headphones? [16:12:11 CEST] <durandal_1707> there is dynaudnorm filter [16:34:07 CEST] <DHE> durandal_1707: just stereo in general. I'm comparing to a commercial product that does similar and the audio is significantly louder [16:34:55 CEST] <kepstin> well, if it's louder, there's a good chance it might clip :/ [16:35:04 CEST] <durandal_1707> yea it depends how your matrix is [16:35:31 CEST] <kepstin> this is one of those 'there's no single definitive good way to do it' things. [16:35:56 CEST] <DHE> I have no control over the commercial thing. ffmpeg I'm just using swr_* functions based on the examples and whatever defaults for my channel layout conversion request [16:39:48 CEST] <kepstin> well, if you can figure out what the commercial thing is doing, you could certainly replicate it by using custom mixing coefficients [16:50:01 CEST] <DHE> kepstin: it's a piece of hardware, I can't unfortunately [16:50:25 CEST] <PacketPusha> When I am defining a file output mask for frames such as frame_%d.jpg how can I make it apply leading zeros? [16:50:44 CEST] <PacketPusha> Right now I get frame_1.jpg ..... up to frame_10.jpg [16:51:13 CEST] <PacketPusha> But then some software pulls frame_1.jpg, frame_10.jpg, frame_11.jpg in that order. [16:52:15 CEST] <furq> frame_%03d.jpg [16:52:21 CEST] <PacketPusha> Nice, thanks. [16:52:24 CEST] <furq> man sprintf [16:52:31 CEST] <PacketPusha> Ok, will do. [17:26:32 CEST] <Prelude2004c> hello everyone good day [17:27:23 CEST] <Prelude2004c> can someone help me with something... sooo strnage [17:27:25 CEST] <Prelude2004c> http://pastebin.com/350NtDyc [17:27:30 CEST] <Prelude2004c> the audio keeps getting out of sync [17:27:44 CEST] <Prelude2004c> not right away but over time it does that.. can anyone assist me on how to try and keep the audio within sync [17:45:11 CEST] <Ximon> Hi, what's the correct way of escaping special characters in filenames when using the tee pseudo-mixer? [19:00:52 CEST] <maziar> what is this error :"library.mak:30: recipe for target 'libavcodec/x86/imdct36.o' failed" [19:02:00 CEST] <BtbN> Not the error. [19:02:21 CEST] <maziar> BtbN look at this http://pastebin.com/nMwsHYk1 [19:02:40 CEST] <BtbN> yes, that looks like the error. [19:03:57 CEST] <maziar> i want to compile ffmpeg on debian jessie, and i follow this tutorial : https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu ,,, but why i get this error , and how to fix it ? [19:07:15 CEST] <maziar> i want to compile ffmpeg on debian jessie, and i follow this tutorial : https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu ,,, but why i get this error: http://pastebin.com/nMwsHYk1 , and how to fix it ? [19:10:24 CEST] <llogan> maziar: never seen that one before. how do i duplicate the issue? did you modify the steps in the guide at all? [19:10:39 CEST] <maziar> yes [19:11:05 CEST] <maziar> llogan can you help me [19:11:18 CEST] <llogan> maybe [19:11:31 CEST] <maziar> llogan please kindly help me [19:11:37 CEST] <llogan> i'm trying to [19:52:23 CEST] <maziar> i need help about this issue [19:53:05 CEST] <llogan> maziar: i was wating for you to answer my questions [19:53:34 CEST] <maziar> llogan sorry, i don't see your question, [19:54:25 CEST] <llogan> "<llogan>... how do i duplicate the issue? did you modify the steps in the guide at all?" [19:55:38 CEST] <maziar> i did every step on that guid, but a question what you mean by :"how do i duplicate the issue? " i don't understand it [19:59:56 CEST] <llogan> what do i need to do to get the same error? [20:09:19 CEST] <maziar> llogan compile it, step by step on debian jessie [20:14:35 CEST] <maziar> llogan i will try it again [20:36:13 CEST] <blackfreighter> Hi, people. Newby here, trying to install ffmpeg in Fedora 23. RPM Repo seems empty, already ran dnf install outputs "Cannot download ffmpeg-2.8.5-1.fc23.x86_64.rpm: All mirrors were tried". Downloaded the tarball instead, but I'm not quite sure how I should go about the installation. [20:36:52 CEST] <llogan> blackfreighter: just download a compiled binary here http://johnvansickle.com/ffmpeg/ [20:38:38 CEST] <blackfreighter> llogan: thank you! Will this install ffmpeg or just let me use it's capabilities? [20:38:49 CEST] <blackfreighter> its* [20:39:13 CEST] <llogan> you don't need to install it. just run it like any other executable. or point your scripts to it [20:48:27 CEST] <llogan> maziar: i can't duplicate the issue [20:49:16 CEST] <llogan> compiles fine, other than the x265 cmake command no longer finding a compiled yasm [20:49:17 CEST] <maziar> llogan let me finish it [20:49:58 CEST] <llogan> this is on Debian 8.4.0 x86_64 [21:17:19 CEST] <mike_> evening all [21:19:26 CEST] <Guest66826> i have been looking for a way to stream raw data from pc 1 to pc 2 , where all the work will be done, via udp, i need to do this to minimise cpu utilisation to the minimum , how can i do that [21:23:00 CEST] <Guest66826> iv been using ffmpeg -f dshow -i INPUT -codec:v libx264 -f mpegts udp://local_ip_pc_2 , its working but i need something like this ffmpeg -f dshow -i INPUT -c:v copy udp://local_ip_pc_2 , so the question is how can i feed this to another ffmpeg process, decode then encode to desired stream as final output? [21:25:46 CEST] <Guest66826> should i worry about the raw format of the device, to decode it properly from pc2? wish means if i have different devices i should use different decoders? like for this one its a webcam and outputs rawvideo bgr24 , and ffmpeg says encoder Lavc57.34.100 [21:26:33 CEST] <maziar> llogan can i change "PATH=" PATH="/bin:$PATH" cmake -G "Unix Makefiles" -DCMAKE_INSTALL_PREFIX="$HOME/ffmpeg_build" -DENABLE_SHARED:bool=off ../../source " insist to :"PATH="$HOME/bin:$PATH" cmake -G "Unix Makefiles" -DCMAKE_INSTALL_PREFIX="$HOME/ffmpeg_build" -DENABLE_SHARED:bool=off ../../source" [21:30:33 CEST] <maziar> llogan ping [21:30:37 CEST] <relaxed> maziar: what are you trying to do? [21:30:51 CEST] <relaxed> different bin and install path? [21:31:21 CEST] <maziar> relaxed i want to install ffmpeg. path make it : /root/bin/ i want /bin [21:33:04 CEST] <maziar> relaxed is it posible ? [21:33:34 CEST] <relaxed> "cmake -LA ../../source" will show you all the options [21:33:57 CEST] <relaxed> cmake -LA ../../source | grep INSTALL [21:34:48 CEST] <llogan> maziar: install yasm from repo, then you can remove "PATH=" PATH="/bin:$PATH". that was intended for users who want to compile the newer yasm instead [21:35:30 CEST] <relaxed> maziar: do you see the right option now? [21:39:57 CEST] <maziar> llogan ok, let me tesrt [21:42:36 CEST] <Guest66826> llogan ping [21:44:13 CEST] <llogan> Guest66826: huh? [21:55:41 CEST] <Guest66826> can you help with feeding rawvideo to ffmpeg? [22:19:06 CEST] <maziar> llogan ping [22:19:16 CEST] <maziar> llogan this is last output [22:19:20 CEST] <maziar> llogan this is last output : LD ffserver_g [22:19:21 CEST] <maziar> CP ffserver [22:19:23 CEST] <maziar> STRIP ffserver [22:22:35 CEST] <pfelt> could someone explain to me what pts=PTS-STARTPTS does? i'm assuming that PTS is encoded in the video stream (in my case an mpegts) and that STARTPTS is the current wall time on the machine? [22:22:58 CEST] <pfelt> Guest11324: what is it you need? [22:24:39 CEST] <kepstin> pfelt: in the setpts filter, 'STARTPTS' is the pts value of the first frame that the filter sees [22:25:07 CEST] <kepstin> so that setpts=pts=PTS-STARTPTS sets the first frame's pts to be '0', and then keeps the relative frame timing for later frames. [22:25:36 CEST] <kepstin> so, e.g. frames with pts 15, 17, 20 turn into 0, 2, 5 [22:28:19 CEST] <pfelt> are those actual timestamps, or just relative to each other? (eg: not time 2, or 5, but really display frame pts=3 right after frame pts=2 and before frame pts=4) [22:28:46 CEST] <Mavrik> PTS (presentation timestamps) are actually timestamps in the stream timebase :) [22:28:50 CEST] <Mavrik> E.g. they're not in seconds [22:28:55 CEST] <Mavrik> And they don't necessarily start at 0 [22:29:36 CEST] <Guest66826> hi pfelt [22:30:43 CEST] <pfelt> Mavrik: but those timestamps don't really have anything to do with "current time". how could they given that my mpegts was recorded hours ago and playback now works. (am i following that right?) [22:30:44 CEST] <Guest66826> i want to stream from pc 1 over udp without any encoding, max bitrate , to pc 2 , and there do the encoding [22:31:33 CEST] <Mavrik> pfelt, no they don't have anything to do with "current time" - essentially they tell the player at what time to display a frame (or audio sample) [22:31:48 CEST] <pfelt> Guest11324: "ffmpeg -i INPUT -vcodec rawvideo -f mpegts udp://ip:port" work for you? [22:31:48 CEST] <Mavrik> STARTPTS isn't wall time [22:31:52 CEST] <maziar> what is this error for : Unknown encoder 'libfaac' [22:31:53 CEST] <maziar> " [22:31:53 CEST] <Mavrik> it's the PTS of the first frame (start) [22:32:00 CEST] <Mavrik> in most videos it should be 0 [22:32:07 CEST] <Mavrik> But it's not always if they're streams or cut somewhere. [22:32:23 CEST] <pfelt> Mavrik: ok. yeah. [22:32:28 CEST] <pfelt> i think i'm following you now [22:32:33 CEST] <pfelt> (you and kepstin) [22:33:02 CEST] <Mavrik> So in short, they're just relative to each other [22:33:14 CEST] <pfelt> and the setpts filter does its thing on *every* frame basically [22:33:14 CEST] <Mavrik> er. [22:33:18 CEST] <Mavrik> That came out wrong I guess. [22:33:47 CEST] <Mavrik> pfelt, yeah, what your "setpts=pts=PTS-STARTPTS" does it just makes sure that first frame of your video will have PTS of 0 [22:33:53 CEST] <Mavrik> and all others will then follow [22:34:03 CEST] <Mavrik> instead of starting at some other number [22:34:19 CEST] <Guest66826> pfelt, thats what i use at pc 1 , but how do i recieve as input from pc 2, shouldint it be ffmpeg -u udp://ip:port ... [22:34:33 CEST] <Guest66826> sorry -i instead of -u [22:34:41 CEST] <pfelt> Guest11324: yeah. that should be right [22:34:59 CEST] <pfelt> don't forget, firewalls can get in the way of stuff like this [22:35:02 CEST] <Guest66826> without any decoding as its a raw video right? [22:35:15 CEST] <Guest66826> its a dummy switch , local network [22:35:37 CEST] <pfelt> Guest66826: machines can run iptables or windows firewall [22:35:53 CEST] <maziar> pfelt do you know this error : Unknown encoder 'libfaac' ",,,, and how to fix ? [22:35:58 CEST] <pfelt> Guest66826: but yeah. you should be able to read it in fine with a -i [22:36:14 CEST] <pfelt> maziar: sounds like you don't have libfaac enabled in your ffmpeg compile [22:36:35 CEST] <maziar> pfelt, how should i do it ? [22:36:42 CEST] <Mavrik> Don't. [22:36:47 CEST] <Mavrik> Use one of the non-shitty AAC encoders. [22:36:54 CEST] <pfelt> hehe [22:36:57 CEST] <Mavrik> ffmpeg -codecs, see which one you have. [22:37:36 CEST] <Guest66826> yes all disabled, however its saying failed to open or configure filtergraph, and ffprobe unsupported codec ... [22:37:47 CEST] <pfelt> one last quetion relating to setpts. it runs the formula PTS-STARTPTS on the frame's pts right? (eg: it doesn't just set up a counter then increment the counter after the first frame) [22:38:15 CEST] <maziar> pfelt please kindly help me [22:38:20 CEST] <Mavrik> pfelt, yes. [22:38:24 CEST] <pfelt> Guest66826: you'd need to post the command and output somewhere in a pastebin for someone to help [22:38:41 CEST] <Guest66826> but if i encode using libx264 it works, wish means no firewall/network issue, thats why i was thinking about mabe it needs to be decoded first before playing [22:38:59 CEST] <pfelt> Mavrik: ah. kk. that's my problem then. (i'm making modifications to the movie filter and it's erroring. i knew it was something i was missing in my head) [22:39:55 CEST] <pfelt> maziar: you'll need to either read up on rebuilding ffmpeg for your platform, or do as Mavrik suggested and change your command to use a different codec [22:40:44 CEST] <maziar> pfelt what is the best encoder for audio? [22:41:04 CEST] <pfelt> that's a pretty subjective question. it depends on what you are doing and what you need [22:41:22 CEST] <pfelt> currently i'm using "-an" in all my encoding and it works perfectly [22:41:49 CEST] <pfelt> provides 100% lossy compression. but then again, i *need* to strip audio out [22:43:15 CEST] <maziar> pfelt i have some movie, and i want best quality, for it which is better ? [22:44:00 CEST] <Guest66826> pfelt, command from pc1 "pastbin.com/raw/Ee0buKh6" command at pc 2 "pastbin.com/raw/Gkwp7b4N" [22:46:04 CEST] <pfelt> maziar: sorry, friend. i can't answer that question for you. you'll need to look at what codecs you have in your toolbox and research which will be the best for your application [22:46:58 CEST] <maziar> pfelt actually i want best command for convert MP4 to m3u8, do you have suggestion ? [22:47:20 CEST] <pfelt> Guest66826: hmm. not sure on that myself. does it help if you change the -f to matroska instead of mpegts? [22:47:28 CEST] <pfelt> maziar: i don't :( [22:49:08 CEST] <Guest66826> pfelt, its giving some erros, il search and try to fix them using matroska and see what happens, thank you [22:50:48 CEST] <pfelt> :) sorry i couldn't me more help [22:51:25 CEST] <maziar> pfelt i also get this error : Unknown encoder 'libvo_aacenc' [22:51:48 CEST] <pfelt> i know there are some issues with mpegts in ffmpeg not being quite standard unless you specify a bunch of other parameters (enet frames sizes aren't 1316). not sure if that's your problem tho [22:52:06 CEST] <pfelt> maziar: i believe it was suggested to use -codecs to see what you have available to you [22:53:21 CEST] <Guest66826> pfelt, thats a good start :) [22:53:57 CEST] <maziar> pfelt , would you please tell me by sample command ? [22:54:20 CEST] <pfelt> ffmpeg -codecs [22:54:28 CEST] <pfelt> maziar: ^^ [22:56:38 CEST] <maziar> pfelt http://pastebin.com/nwp5jWgh [23:01:25 CEST] <kepstin> maziar: now you scroll down to the line that says 'aac' and look over to the list of encoders on the right. [23:01:48 CEST] <kepstin> in this case, you have libfdk_aac, which is probably a good choice. [23:02:14 CEST] <maziar> kepstin libavutil 55. 22.100 / 55. 22.100 [23:02:15 CEST] <maziar> libavcodec 57. 35.100 / 57. 35.100 [23:02:16 CEST] <maziar> libavformat 57. 34.103 / 57. 34.103 [23:02:17 CEST] <maziar> libavdevice 57. 0.101 / 57. 0.101 [23:02:18 CEST] <maziar> libavfilter 6. 44.100 / 6. 44.100 [23:02:19 CEST] <maziar> libswscale 4. 1.100 / 4. 1.100 [23:02:20 CEST] <maziar> libswresample 2. 0.101 / 2. 0.101 [23:02:21 CEST] <maziar> libpostproc 54. 0.100 / 54. 0.100 [23:02:55 CEST] <maziar> kepstin i just want to create m3u8 from a MP4, how should i do, i'm really tired to do it [23:04:38 CEST] <kepstin> you mean you want to make segments for an HLS stream? m3u8 is just a playlist :) [23:05:31 CEST] <Mavrik> *facepalm* [23:07:17 CEST] <kepstin> maziar: any idea about what you want to use for segment lengths, bitrate, etc? [23:07:42 CEST] <maziar> segment is 10 second and bitrate is 480kbps [23:08:39 CEST] <kepstin> assuming you've already set up the rest of the command (it would be helpful if you show us your existing command line), it should be as simple as adding '-c:a copy' (if the input has aac audio), or '-c:a libfdk_aac' to fix your error. [23:29:15 CEST] <maziar> kepstin what is this error : [NULL @ 0x4b8c980] Unable to find a suitable output format for '32' [23:29:15 CEST] <maziar> 32: Invalid argument [23:29:51 CEST] <maziar> kepstin what is this error : [NULL @ 0x4b8c980] Unable to find a suitable output format for '32' 32: Invalid argument [23:30:10 CEST] <kepstin> that error almost certaily means that you made a typo on the command line. [23:30:29 CEST] <kepstin> so please follow the instructions that the helpful bot has given you. [23:35:19 CEST] <maziar> i want to create m3u8 from mp4 for test ffpmeg a sample , a test, how can i do it [23:36:37 CEST] <llogan> repeating what you want to do while ignoring requests for aditional information is the best way to get help. also, make sure to provide only the errors with no context. [23:46:33 CEST] <pfelt> ok. another time question. what is the timebase? (ref: https://ffmpeg.org/ffmpeg-filters.html#setpts_002c-asetpts) [23:49:50 CEST] <kepstin> pfelt: timebase is a fraction that you multiply the pts by to get the time in seconds. [23:50:22 CEST] <kepstin> so, e.g. ntsc video might have a timebase of 1001/30000, then frame pts is 0, 1, 2, 3 [23:52:27 CEST] <kepstin> typically the timebase will be the inverse of the framerate, so the pts values are nice, but that's not always the case. [23:55:32 CEST] <kepstin> I think mkv files use a timebase of 1/1000 and the pts values are just milliseconds. [00:00:00 CEST] --- Wed Apr 20 2016

1 0

ffmpeg-devel.log.20160418
by burek 19 Apr '16

19 Apr '16

[00:01:17 CEST] <atomnuker> rcombs: got kinda working PCE support for the aac encoder, though the decoder's not happy about paired channels for some reason [00:01:27 CEST] <rcombs> \o/ [00:01:37 CEST] <atomnuker> do you have any files I can test to see what I'm doing wrong [00:02:17 CEST] <atomnuker> preferably with some common standard layout not in the AAC spec so I can see what the correct order should be [00:06:04 CEST] <rcombs> try https://www2.iis.fraunhofer.de/AAC/7.1auditionOutLeader_v2_rtb.mp4 [00:06:16 CEST] <rcombs> uses PCE to signal BD 7.1 [00:06:45 CEST] Action: wm4 adds to fucked up samples collection [00:06:49 CEST] <JEEB> :D [00:10:15 CEST] <rcombs> in theory that configuration should be valid, and you'd think the fraunhofer guys would know how to encode AAC properly [00:13:48 CEST] <rcombs> allegedly, ISO/IEC 14496-3 2009 PDAM 4 will specify channel configuration 12 as BD 7.1, but I don't have a sample for that [00:19:40 CEST] <cone-795> ffmpeg 03Carl Eugen Hoyos 07master:db7d0d6e7c43: lavc/fic: Cosmetics, fix a typo. [01:00:56 CEST] <cone-795> ffmpeg 03Michael Niedermayer 07master:76d0209db482: avcodec/intrax8: Remove duplicated chunk from ba5bcf96124a4933eef170dfe7955809d8d54a64 [02:56:30 CEST] <cone-795> ffmpeg 03Michael Niedermayer 07master:81064795034e: avcodec/bitstream_filter: Fix initializing options from the argument string [02:56:31 CEST] <cone-795> ffmpeg 03Michael Niedermayer 07master:57fc93ecb2ce: avcodec/remove_extradata_bsf: Add back 'k' and 'e' options [02:56:32 CEST] <cone-795> ffmpeg 03Michael Niedermayer 07master:ce18e48aec5d: avcodec/dump_extradata_bsf: Add back 'k' and 'e' options [10:19:25 CEST] <j-b> 'morning [10:37:58 CEST] <wm4> how to make anyone hate (L)GPL [13:35:28 CEST] <petrurares> hi there! I have a doubt, I want to create a test for mss1 codec. I have a sample video, I update microsoft.mak and when I do " make -k -j6 fate-mss2 SAMPLES=../samples" the program don't cover any line of libavcodec/mss2.c [13:35:33 CEST] <petrurares> Any idea? [13:36:26 CEST] <petrurares> * change mss2 to mss1 in all the words [13:37:25 CEST] <Compn> did you add fate-mss1 to the fate makefile ? [13:37:43 CEST] <Compn> petrurares [13:37:57 CEST] <petrurares> yes [13:38:56 CEST] <petrurares> I add the following lines: [13:39:02 CEST] <petrurares> FATE_MSS1 += fate-mss1-pal [13:39:11 CEST] <petrurares> fate-mss1-pal: CMD = framecrc -i $(TARGET_SAMPLES)/mss1/screen_codec.wmv [13:39:16 CEST] <petrurares> FATE_SAMPLES_AVCONV-$(call DEMDEC, ASF, MSS1) += $(FATE_MSS1) [13:39:21 CEST] <petrurares> fate-mss1: $(FATE_MSS1) [13:41:28 CEST] <petrurares> when I want to view the cover of fate-mss2 test I get the same resoults more or less [13:41:42 CEST] <Compn> you're saying the fate test runs , but the code coverage is bad ? [13:41:53 CEST] <petrurares> yes [13:43:59 CEST] <petrurares> I don't know if I am running the tests wrong or lcov doesn't work as expected [13:44:38 CEST] <Compn> i'm not sure about lcov either , wait for someone who knows things :) [13:45:01 CEST] <petrurares> ok, thanks :) [13:47:21 CEST] <wm4> did anyone try to use libressl with ffmpeg yet? [13:47:27 CEST] <wm4> via openssl API compat? [13:57:33 CEST] <fritsch> wm4: openelec [13:57:36 CEST] <fritsch> wm4: and libreelec [13:57:41 CEST] <fritsch> the patch needed is minimal [13:58:27 CEST] <fritsch> https://github.com/OpenELEC/OpenELEC.tv/blob/master/packages/multimedia/ffm… <- seems, even more trivial [13:58:56 CEST] <fritsch> so if libressl is in place all fine [13:58:57 CEST] <wm4> haha [13:59:01 CEST] <wm4> thanks [13:59:42 CEST] <fritsch> https://github.com/OpenELEC/OpenELEC.tv/blob/openelec-6.0/packages/multimed… <- it was more work in older versions it seems [14:00:13 CEST] <wm4> that patch just introduces a different define and consequently requires changing all the code [14:00:19 CEST] <fritsch> jep [14:01:08 CEST] <nevcairiel> someone enlighten me, how can you fork openssl and then put the fork under a more liberal license [14:01:20 CEST] <wm4> nevcairiel: sure seems strange [14:01:30 CEST] <fritsch> i have no idea to be honest how that worked [14:03:59 CEST] <nevcairiel> i find it interesting that its not really obvious which license its actually under, the website didnt tell me, nor does the source have a license or copying file or something that seems obvious [14:06:30 CEST] <nevcairiel> sounds to me like its still apache 1.0 [14:06:37 CEST] <nevcairiel> ie. not actually more "libre" [14:08:36 CEST] <nevcairiel> ie. the openelec patch would seem to be problematic [14:10:25 CEST] <Daemon404> fritsch, that patch is uh... [14:10:25 CEST] <Daemon404> uh. [14:39:35 CEST] <fritsch> Daemon404: ultra hot? :-) [14:40:04 CEST] <JEEB> isn't it just superhot now? ;) [14:40:38 CEST] <fritsch> uber hot [16:00:56 CEST] <cone-042> ffmpeg 03Diego Biurrun 07master:d3044cf37fcd: opt-test: Move some variable declarations to avoid block braces [16:00:57 CEST] <cone-042> ffmpeg 03Diego Biurrun 07master:0d2fcdb1c5c9: opt-test: Merge struct declaration and initialization [16:00:58 CEST] <cone-042> ffmpeg 03Derek Buitenhuis 07master:1be5509fbbc6: Merge commit '0d2fcdb1c5c9e844c232e5429130727121990d0e' [16:01:08 CEST] <Daemon404> wm4, ping [16:01:14 CEST] <wm4> yes? [16:01:17 CEST] <Daemon404> lavf: use new decode API [16:01:24 CEST] <Daemon404> these 4 commits [16:01:27 CEST] <Daemon404> what do, re: merge [16:01:55 CEST] <wm4> the ones which make ffmpeg use it will break fate [16:02:06 CEST] <wm4> at least the decoding one, anyway [16:02:29 CEST] <wm4> for the others I have rebased patches on the ML, but let me check first [16:02:32 CEST] <Daemon404> so whay do i do? do i skip all 4? do you have ffmpeg versions? [16:06:12 CEST] <wm4> Daemon404: the first is "lavc: introduce a new decoding/encoding API with decoupled input/output" right? [16:06:17 CEST] <Daemon404> yes [16:06:40 CEST] <wm4> "[PATCH v2 3/8] lavc: introduce a new decoding/encoding API with decoupled input/output" should be the ffmpeg version of this [16:07:00 CEST] <wm4> then skip the ffmpeg ones [16:07:02 CEST] <Daemon404> as it been ok'd [16:07:06 CEST] <Daemon404> has* [16:07:30 CEST] <wm4> not explicitly, but there hasn't been rejections either [16:07:38 CEST] <wm4> the main problem was with the ffmpeg.c patches [16:08:14 CEST] <wm4> and "lavf: use new decode API" apparently has problems, so just skip it for now [16:09:47 CEST] <Daemon404> so apply that single patch and skip the other 3? [16:09:52 CEST] <Daemon404> or rather, all 4. [16:10:35 CEST] <wm4> I'm fine with applying the first patch [16:10:40 CEST] <wm4> the rest will cause fate failures [16:10:59 CEST] <Daemon404> right [16:11:01 CEST] <Daemon404> so apply "[PATCH v2 3/8] lavc: introduce a new decoding/encoding API with decoupled input/output" [16:11:04 CEST] <Daemon404> and skip all 4 merges [16:11:07 CEST] <wm4> michaelni_: any immediate opinion? [16:11:23 CEST] <wm4> well maybe I should just resent them [16:11:25 CEST] <wm4> so skip all 4 [16:11:34 CEST] <wm4> it's getting too messy [16:11:48 CEST] <wm4> (Libav might have applied additional changes to the first patch) [16:12:03 CEST] <Daemon404> lol.. ok [16:12:18 CEST] <Daemon404> ill just skip all four and leave getting them in ffmpeg to you [16:12:23 CEST] <Daemon404> although teh thread seems kind of dead [16:12:36 CEST] <wm4> I gave up on trying to fix the ffmpeg.c failures [16:12:50 CEST] <wm4> they're only because of how ffmpeg tries to set dts on drain packets [16:13:01 CEST] <Daemon404> what about the lavf patch [16:13:15 CEST] <wm4> it's said to cause hangs in fate [16:13:20 CEST] <Daemon404> O_o [16:13:33 CEST] <wm4> apparently I missed this reply and so didn't investigate this at all [16:14:13 CEST] <cone-042> ffmpeg 03wm4 07master:05f66706d182: lavc: introduce a new decoding/encoding API with decoupled input/output [16:14:14 CEST] <cone-042> ffmpeg 03wm4 07master:84aea95e31e5: avconv: use new decode API [16:14:15 CEST] <cone-042> ffmpeg 03wm4 07master:35846d93e023: avconv: use new encode API [16:14:16 CEST] <cone-042> ffmpeg 03wm4 07master:8bc4accc37ab: lavf: use new decode API [16:14:17 CEST] <cone-042> ffmpeg 03Derek Buitenhuis 07master:528af33b6f45: Merge commit '8bc4accc37ab047d2fd85d672c577b39dfc918e1' [16:14:29 CEST] <Daemon404> wm4, it would be good to push the new api at least [16:14:35 CEST] <Daemon404> sooner rather than later [16:14:47 CEST] <wm4> I guess so [16:15:19 CEST] <Daemon404> jkqxz, https://git.libav.org/?p=libav.git;a=commit;h=98114d70e48caf871b0fe9b8e5bf8… [16:15:23 CEST] <Daemon404> do you expect any issue merging this [16:15:29 CEST] <Daemon404> or the nvidia filter from anton after it [16:15:35 CEST] <nevcairiel> .. its a new file? :) [16:15:43 CEST] <Daemon404> it is but lavfi scares me [16:15:51 CEST] <Daemon404> and i cant test vaapi [16:23:21 CEST] <Daemon404> nevcairiel, is there anything special to merging lavfi filters? [16:23:24 CEST] <Daemon404> or is it safe [16:23:41 CEST] <nevcairiel> just make sure to update the license of new files, otherwise there should be nothing special [16:23:51 CEST] <Daemon404> i did [16:23:53 CEST] <Daemon404> k [16:24:02 CEST] <cone-042> ffmpeg 03Mark Thompson 07master:98114d70e48c: lavf: VAAPI scale filter [16:24:03 CEST] <cone-042> ffmpeg 03Derek Buitenhuis 07master:34245eccaf99: Merge commit '98114d70e48caf871b0fe9b8e5bf8ebd989b845d' [16:24:19 CEST] <Daemon404> lavfi: add an NVIDIA NPP-based scaling filter [16:24:22 CEST] <Daemon404> wat about this [16:24:29 CEST] <Daemon404> i know our nvidia stuff is different [16:25:20 CEST] <nevcairiel> dunno [16:25:29 CEST] <Daemon404> is it BtbN i ask? [16:25:34 CEST] <nevcairiel> dont remember which part of the cuda configure changes we took and which we didnt [16:25:41 CEST] <Daemon404> right. [16:26:07 CEST] <BtbN> Only the nvenc encoder is different iirc. The rest should be like in libav, as it was merged [16:26:23 CEST] <Daemon404> so that filter is ok to merge? [16:26:46 CEST] <nevcairiel> i think we already have the other cuda hwupload filter which uses linked cuda, so its probably not any different to that [16:27:08 CEST] <BtbN> It should be, the cuda infrastructure is identical. [16:27:16 CEST] <Daemon404> ok. [16:34:33 CEST] <cone-042> ffmpeg 03Anton Khirnov 07master:8a02a8031ef4: lavfi: add an NVIDIA NPP-based scaling filter [16:34:34 CEST] <cone-042> ffmpeg 03Derek Buitenhuis 07master:94e5f0922b72: Merge commit '8a02a8031ef4f98faf5647f0e01a8922247bf748' [16:36:26 CEST] <cone-042> ffmpeg 03Martin Storsjö 07master:d44f3e405950: avio: Apply avoptions on the URLContext itself as well [16:36:27 CEST] <cone-042> ffmpeg 03Derek Buitenhuis 07master:4eef36a4f6db: Merge commit 'd44f3e4059506a182f59218b1e967d42b01e097c' [16:48:41 CEST] <cone-042> ffmpeg 03Andrey Utkin 07master:ccea588f8319: avio: Add an option 'rw_timeout' [16:48:42 CEST] <cone-042> ffmpeg 03Derek Buitenhuis 07master:299d4f9428c2: Merge commit 'ccea588f831906084b8c8235222920e6984beb72' [16:48:58 CEST] <wm4> jesus christ that Libav discussion about using assert [16:49:10 CEST] <nevcairiel> luca just doesnt get what they are good for [16:49:11 CEST] <nevcairiel> never has [16:49:13 CEST] <nevcairiel> never will [16:49:30 CEST] <Daemon404> ... im pretty sure i had a huge argument with him about assert abotu 2 years ago [16:53:23 CEST] <Daemon404> Timothy_Gu, is it normal for fatebeta to not have the ability to sort by a column [16:54:48 CEST] <nevcairiel> Daemon404: works for me, except the comment column which wouldnt make all that much sense either way [16:56:36 CEST] <Angus> If I want to use the AVHWAccel ff_h264_vdpau_hwaccel found in... [16:56:37 CEST] <Angus> https://ffmpeg.org/doxygen/2.8/vdpau__h264_8c_source.html [16:56:39 CEST] <Daemon404> it isnt even clickable for me nevcairiel [16:56:44 CEST] <Daemon404> on fatebeta.f.o [16:56:56 CEST] <Angus> is the only thing I need to do av_register_hwaccel? [16:57:43 CEST] <Angus> as it seems decode_nal_units calls into setup_hwaccel -> find_hwaccel [16:58:59 CEST] <michaelni_> ubitux, is there any magic for lcov and the src/ directory ? i remember there was some issue with it on coverage.ffmpeg.com and petru seems to have some problem too with some files not showing up that shuld, but maybe its unrelated [16:59:02 CEST] <BtbN> no, you have to setup vdpau yourself and handle the vdpau output frames [17:00:32 CEST] <Angus> is there any examples of the vdpau setup process? [17:01:15 CEST] <ubitux> michaelni_: yes but Andreas fixed it afaik [17:01:36 CEST] <ubitux> e740c3fb90c02802bc121e9f5a3cf58615b4cea1 i suppose [17:02:05 CEST] <ubitux> it was a few months ago, ask him if he didn't rebased since then (not likely but well..) [17:02:18 CEST] <ubitux> it fixed fate for me [17:04:08 CEST] <ubitux> wm4: yeah every single time there is a patch containing an assert luca tries to push his rejection of it [17:04:22 CEST] <ubitux> it's always the same endless debate :p [17:12:43 CEST] <nevcairiel> does anyone know if avcodec just doesnt support mono aac or something? I have a sample from a user thats supposedly mono, but avcodec decodes it as stereo. some other stream probers seem to say its mono, but i dont know their accuracy [17:13:06 CEST] <Daemon404> are you sure it isnt dual mono [17:13:09 CEST] <Daemon404> is it from japan? [17:13:25 CEST] <Daemon404> or is it SBR? [17:13:36 CEST] <nevcairiel> is SBR yes, but why would that force stereo [17:16:29 CEST] <Daemon404> i assume it's HE then [17:16:40 CEST] <Daemon404> and thus uses parametric stereo [17:16:49 CEST] <nevcairiel> no, its only HE, not HEv2 [17:16:52 CEST] <nevcairiel> HEv2 is PS [17:17:43 CEST] <Daemon404> ok [17:17:49 CEST] <atomnuker> nevcairiel: PCEs can cause some funny behaviour as I've learned, might be that [17:18:16 CEST] <nevcairiel> debugging in the aac code is apparently impossible due to the f'ing template dealy [17:19:04 CEST] <nevcairiel> well lets see if it uses PCE [17:19:07 CEST] <durandal_1707> anyone want to backport something for me? [17:21:49 CEST] <nevcairiel> the extradata has no PCE and says channel config 1, which is mono [17:23:19 CEST] <nevcairiel> wait, there is some crazy ass code that fro some reason assumes there is PS when SBR is used [17:23:34 CEST] <nevcairiel> why would it assume that [17:25:32 CEST] <jkqxz> Daemon404: Nothing funny in those. It's the next one with (avconv|ffmpeg)_vaapi.c that might do something nasty (and also makes it possible to actually test stuff). [17:28:06 CEST] <nevcairiel> atomnuker: do you know how this line makes sense? http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/mpeg4audio.c;h=18… .. why is AAC_LC considered HE-AACv2? [17:33:25 CEST] <atomnuker> nevcairiel: c->object_type != AOT_AAC_LC makes no sense [17:36:18 CEST] <cone-042> ffmpeg 03Petru Rares Sincraian 07master:849e55e58ecc: fate: Add test for mss1 codec [17:40:10 CEST] <nevcairiel> all those profiles are build upon the LC profile, no clue what the check is really supposed to say [17:40:19 CEST] <nevcairiel> its been like that since PS support was implemented [17:41:12 CEST] <atomnuker> might be a typo since the comment says HEv2 [17:45:53 CEST] <nevcairiel> the real problem starts in decode_ga_specific_config, where an implicit PS (-1) gest turned into PS=1 on mono streams with SBR [17:46:41 CEST] <atomnuker> ah, yeah, forgot the sample was AAC HE [17:48:39 CEST] <nevcairiel> this whole functionality seems to have gone mostly unchanged, everything ends up at the original HEv2 commit :) [17:48:47 CEST] <Daemon404> nice [17:48:53 CEST] <Daemon404> who even encodes he v1 btw [17:48:56 CEST] <nevcairiel> guess HE mono is rare [17:49:03 CEST] <Daemon404> ive never seen it in the wild [17:49:19 CEST] <nevcairiel> for when your signal is mono and you dont need PS? [17:49:53 CEST] <Daemon404> sounds rare [17:50:01 CEST] <atomnuker> few do, but those who do know PS can reduce quality vs just SBR for a given bitrate [17:50:09 CEST] <nevcairiel> well sure mono audio isnt encountered that often [17:50:29 CEST] <atomnuker> will take a look at the specs tonight to see the correct way to signal v2 [17:50:41 CEST] <nevcairiel> but ffmpeg even sets the profile to aac-he, not v2, so at some point it figures out there is no PS [17:50:53 CEST] <nevcairiel> but it doesnt undo the stereo-fication [17:51:11 CEST] <Daemon404> nevcairiel, ... does it happen pre-codecpar? [17:51:14 CEST] <Daemon404> <_<; [17:51:41 CEST] <nevcairiel> yes [17:51:47 CEST] <nevcairiel> the code i'm testing is pre [17:51:51 CEST] <Daemon404> ok [17:51:56 CEST] <Daemon404> phew. [17:52:27 CEST] <wm4> is it legitimate to use AVOption to communicate information from a decoder to the API user? [17:52:29 CEST] <nevcairiel> even outputs stereo audio for some reason, yet claims its aac-he only =p [17:53:23 CEST] <Daemon404> wm4, shouldnt that be sidedata [17:53:24 CEST] <Daemon404> or metadata [17:53:55 CEST] <nevcairiel> using avoption as information export seems fishy [17:54:07 CEST] <nevcairiel> although i'm sure we have it somewhere already [17:54:29 CEST] <wm4> Daemon404: it's not per-frame data, it's only set on init [17:54:41 CEST] <wm4> anyway, doesn't matter too much [17:54:54 CEST] <nevcairiel> what kind of info is it [17:55:50 CEST] <Daemon404> wm4, wouldnt it be stream-level sidedata then [17:55:58 CEST] <wm4> no [17:56:14 CEST] <wm4> nevcairiel: from a system decoder wrapper, whether it actually uses hwaccel [17:56:54 CEST] <nevcairiel> i suppose avoption is fine in that case [17:57:01 CEST] <nevcairiel> even if a bit iffy [17:57:05 CEST] <Daemon404> still seems like something the api user shouldnt need to know [17:57:11 CEST] <Daemon404> except for ricing [17:57:15 CEST] <nevcairiel> need maybe not, but maybe want [18:26:56 CEST] <nevcairiel> atomnuker: looks like the behavior of our decoder is "correct" to some degree, the specification considers this type of behavior appropriate [18:30:02 CEST] <nevcairiel> 1.6.6.3 in the spec [19:21:45 CEST] <wm4> lol [19:22:39 CEST] <wm4> heh chromium is accessing private fields (though that's no the cause) [19:44:13 CEST] <nevcairiel> even included internal.h =p [19:49:23 CEST] <kurosu> Is there something better to use that pastebin for posting chunks of code? I think I saw people using something else, don't know why ? [19:49:45 CEST] <nevcairiel> usually personal preference, some like github's gist, some use any other paste site [19:52:26 CEST] <kurosu> jamrial, before: http://pastebin.com/UQdq5xsT [19:53:38 CEST] <kurosu> after: http://pastebin.com/qGKBi1eQ [19:55:50 CEST] <atomnuker> I have aliases for sprunge.us for diffs and 0x0.st for everything else [21:59:21 CEST] <pfelt> afternoon all. i'm playing with writing my own filter and am struggling just a little bit. are there any docs on how to create a new filter? http://wiki.multimedia.cx/?title=FFmpeg_filter_HOWTO is the best i've found, and it seems to be dated? [22:01:31 CEST] <jkqxz> Take an existing filter with the same input/output properties as the one you want to create and modify it. [22:02:56 CEST] <pfelt> heh. that's what i thought the answer would be [22:03:24 CEST] <pfelt> i'm missing something somewhere. i segfault on options initialization. i'll keep digging [22:04:55 CEST] <pfelt> i think it has to do with zero'ing my private context right [22:05:44 CEST] <atomnuker> make sure your AVClass is the first field in your private context [22:05:47 CEST] <jkqxz> The private context it makes for you is zero initialised (to be priv_size). [22:07:40 CEST] <pfelt> hmm. then i'm not sure why set_string() is segfaulting on the free [22:08:36 CEST] <ubitux> pfelt: doc/writing_filters.txt [23:09:23 CEST] <Daemon404> https://trac.ffmpeg.org/ticket/5451 [23:09:35 CEST] <Daemon404> reported by "misc' and cc'd to me? [23:10:10 CEST] <Daemon404> i guess cause im the merge commit gut [23:10:12 CEST] <Daemon404> guy [23:10:16 CEST] <Daemon404> ill check tomorrow i suppose [23:14:47 CEST] <kurosu> trying to create a wmaudio sample from matroska to wma, aframes seem to generate several warnings about DTS - wma muxer issue or PEBKAC ? in the later case, does anyone have a better suggestion ? [23:18:17 CEST] <cone-316> ffmpeg 03Petru Rares Sincraian 07master:f25367f4b41c: fate: Add test for 012v codec [23:23:49 CEST] <pfelt> not sure if this is the right place to ask, but what's the difference between using -i and a filter that's a source? [23:24:30 CEST] <wm4> you mena moviesrc or whatever that was? [23:24:36 CEST] <wm4> *mean [23:24:59 CEST] <pfelt> the filter i'm playing with create will be a video source filter [23:25:16 CEST] <pfelt> just wondering why we can do filters that are sources and the -i (which seems to me to also be a source) [23:25:27 CEST] <pfelt> and yeah, it's kinda based on movie/amovie [23:25:32 CEST] <nevcairiel> -i is for plain files [23:25:47 CEST] <nevcairiel> sources are more generic, they could also open a file if they want to [23:25:53 CEST] <nevcairiel> but they can also entirely generate the signal [23:25:55 CEST] <nevcairiel> like testsrc [23:26:14 CEST] <wm4> and yes ffmpeg.c can use them [23:31:07 CEST] <durandal_1707> michaelni_: why is there 2 factor in tak parser ? [23:31:59 CEST] <durandal_1707> it breaks demuxing of very small frames at end of stream [23:42:25 CEST] <pfelt> this may be a filtering question, but is there any good way to swap input streams while the command is running? kinda like streamselect but with actually opening a new input file? [23:43:03 CEST] <nevcairiel> well your filter can do whatever, although changing stream parameters like frame size etc can be problematic [23:43:32 CEST] <pfelt> yeah. that's what my filter was intending to do, i just wanted to make sure that it didn't already exist somewhere [23:43:44 CEST] <pfelt> the folks in the user channel didn't seem to think it did [23:44:53 CEST] <ubitux> lavfi has pre-opened stream [23:45:11 CEST] <ubitux> i hope you're not trying to make a streamselect that actually opens files :p [23:45:30 CEST] <pfelt> either that or a movie source that allows for changin the source to a new file [23:45:51 CEST] <ubitux> :/ [23:46:06 CEST] <pfelt> here's my use case: [23:46:23 CEST] <nevcairiel> cant you just open multiple files at once and switch through them as needed, instead of opening them on demand [23:46:28 CEST] <pfelt> i've got a grid (built with *stack) and i want to change what one of the grid items is displaying without killing the whole thing [23:46:34 CEST] <ubitux> the closest sane option is see for you is to add command input to movie filter [23:46:44 CEST] <ubitux> to live switch the source [23:47:21 CEST] <pfelt> ubitux: yeah. i started there actually [23:47:39 CEST] <pfelt> i segfault when i try to switch, i think because of how movie builds it's pads, but i could be wrong there [23:48:15 CEST] <pfelt> this leads to another question path. process_command(). the command stops at the first filter that has that name right? [23:48:37 CEST] <pfelt> so if i have multiple movie filters installed, how could i send a command to the second instance of the filter? [23:48:55 CEST] <ubitux> mmh i don't remember the details [23:48:59 CEST] <nevcairiel> i think you can name them [23:49:11 CEST] <ubitux> try the command mode in ffmpeg ('c') [23:49:27 CEST] <ubitux> and send a ping or something [23:49:37 CEST] <ubitux> there might be an help to help you target a specific one [23:50:36 CEST] <pfelt> i'm trying to grok this code along with writing a new filter too. in my test filter i've set an instance name which i could use to determine if "this instance" should process the command or let it move on to another instanc [23:56:17 CEST] <michaelni_> durandal_1707, i have no idea [23:58:11 CEST] <cone-316> ffmpeg 03Paul B Mahol 07master:13406b6124a5: avcodec/tak_parser: fix parsing of streams with bunch of small frames at end [00:00:00 CEST] --- Tue Apr 19 2016

1 0

ffmpeg.log.20160418
by burek 19 Apr '16

19 Apr '16

[00:34:06 CEST] <TAFB2> I want ffmpeg to stream a short video in a loop but insert random .mp3 audio songs, possible? [00:46:52 CEST] <andrey_utkin_> what is the exact upper estimation of h264 bitstream inflation at applying of startcode emulation prevention? dst_size = src_size * 5 / 4? [02:03:03 CEST] <spirou> are there an easy way to remove a part of a song in the middle with ffmpeg? [02:04:26 CEST] <spirou> if I would like to cut away 05:27-06:32 from a file [02:14:37 CEST] <andrey_utkin_> spirou, you could make two cuts and then concat them. Or maybe use aselect filter if there's such one (just guessing, haven't checked actual doc) [02:19:30 CEST] <prelude2004c_Zzz> hello .. http://pastebin.com/tVi1BhQU .. can anyone help? [02:19:48 CEST] <prelude2004c_Zzz> still trying to get an answer on this problem [02:20:09 CEST] <furq> that paste has expired [02:20:17 CEST] <J_Darnley> Have you even visited that link? [02:22:41 CEST] <spirou> andrey_utkin_: ok I guess it would be ffmpeg -t 05:27 -i in.mp3 -ss 06:32 -i in.mp3 out.mp3 [02:23:06 CEST] <J_Darnley> no [02:23:19 CEST] <J_Darnley> that will have two streams [02:23:23 CEST] <prelude2004c_Zzz> oh sorry :( [02:23:27 CEST] <J_Darnley> you will need something to concat them [02:23:36 CEST] <prelude2004c_Zzz> let me see if i can get another going .. [02:24:46 CEST] <spirou> J_Darnley: uhmm.. I never done that... I guess some flag that tells that input2 should continue after input1 ? [02:25:11 CEST] <J_Darnley> no, I'm thinking of a concat filter [02:26:30 CEST] <prelude2004c_Zzz> hey guys.. ok see this paste [02:26:30 CEST] <prelude2004c_Zzz> http://pastebin.com/REQMtykq [02:26:45 CEST] <J_Darnley> > in 60 min [02:27:00 CEST] <J_Darnley> Lats hope someone doesn't turn up in 61 minutes trying to help [02:27:05 CEST] <furq> prelude2004c_Zzz: " [02:27:10 CEST] <furq> "-map i:0x1114 -map i:0x1114" [02:27:30 CEST] <prelude2004c_Zzz> oh i always do that .. my bad :) [02:28:17 CEST] <furq> also the second ffmpeg will only include one audio track unless you use -map 0 [02:28:27 CEST] <prelude2004c_Zzz> oh yes.. i did that because the source contains only 1 5.1 audio channel.. so i am splitting that into 2 channels and then one is going to be stereo and the other i leave as 5.1 [02:28:54 CEST] <prelude2004c_Zzz> furq .. do you mean on the second one i just run -map 0 and it will work ? [02:29:14 CEST] <furq> i don't know, but i know it won't work without that [02:29:44 CEST] <prelude2004c_Zzz> oh.. do i put that on ffmpeg -map 0 |pipe > ffmpeg or do i put it input | ffmpeg -i - .... -map 0 [02:29:50 CEST] <furq> the second one [02:29:55 CEST] <prelude2004c_Zzz> ok cool let me try it [02:32:02 CEST] <prelude2004c_Zzz> hum.. You ( furq ) .. you are good you [02:32:09 CEST] <furq> i am good me [02:33:01 CEST] <prelude2004c_Zzz> :) last question .. since i am nearly there [02:33:03 CEST] <prelude2004c_Zzz> "-c:a:0 aac -b:a:0 128k -ac 2 -c:a:1 aac -b:a:1 256k -ac 6" [02:33:21 CEST] <prelude2004c_Zzz> the source is both 5.2 .. i set -ac2 for the first one.. and ac6 for the second.. now they both encode at 5.1 [02:33:47 CEST] <prelude2004c_Zzz> how do i separate it so the first can be set to stereo and the second will be 5.2 [02:33:51 CEST] <prelude2004c_Zzz> 5.1 rather [02:34:04 CEST] <furq> -ac:0 and -ac:1 [02:34:13 CEST] <furq> or -channels:a:0 if you want to be modern [02:34:24 CEST] <prelude2004c_Zzz> alright.. ok.. makes sense... logic prevails as always [02:34:43 CEST] <prelude2004c_Zzz> let me get with the times .. -channels:a:0 it is :P [02:35:52 CEST] <prelude2004c_Zzz> ehhhh channels:a:0 unrecognized option :P [02:37:12 CEST] <furq> i guess that one hasn't been modernised then [02:39:02 CEST] <prelude2004c_Zzz> ya... also back to -ac:0 seems to still stay on 5.1 [02:39:09 CEST] <prelude2004c_Zzz> doesn't convert to 2 ( stereo ) [02:39:58 CEST] <prelude2004c_Zzz> "-c:a:0 aac -b:a:0 128k -ac:0 2 -c:a:1 aac -b:a:1 256k -ac:1 6" [02:42:27 CEST] <snakeryslug> as someone who is ignorant of the details of av files, i was suprised that a video that's only 1 picture throughout was 300MB. [02:42:37 CEST] <prelude2004c_Zzz> seems that the 128K and 256k works [02:42:59 CEST] <prelude2004c_Zzz> just not the channels ( ac ) [02:51:47 CEST] <spirou> ok I suppose I have to cut beginning or end to two files that then be concatenated with ffmpeg -i concat:"in1.mp3|in2.mp3" -c copy out.mp3 [02:57:05 CEST] <Snoo20664> you can just copy /b audio files [02:57:26 CEST] <Snoo20664> copy /b in1.mp3 + in2.mp3 out.mp3 [02:58:08 CEST] <spirou> wel... I'm in linux but doesn't a mp3 have some header [02:58:12 CEST] <furq> that's exactly what the concat protol does, and you can't do that with mp3 [02:58:20 CEST] <furq> because of the id3 headers [02:58:38 CEST] <furq> s/protol/protocol [02:59:01 CEST] <spirou> ok.. so to make a good concatenation of the two mp'3 I need another longer way to write it? [02:59:04 CEST] <prelude2004c_Zzz> furq .. any other ideas ? "-c:a:0 aac -ac:0 2 -b:a:0 128k -c:a:1 aac -ac:1 6 -b:a:1 256k" << --- this seams to convert the first audio bit rate to 128 and the second to 256 but both are set to 5.1 instead of 2.0 for the first audio and 5.1 for the second [02:59:17 CEST] <furq> spirou: https://trac.ffmpeg.org/wiki/Concatenate#demuxer [02:59:46 CEST] <prelude2004c_Zzz> last piece of my puzzle ( at least that i know of right now :) ) [03:00:31 CEST] <spirou> ok I must make a text file with one line per mp3 file then [03:00:38 CEST] <furq> prelude2004c_Zzz: sorry, that should be -ac:a:0 [03:00:59 CEST] <furq> because apparently you might want to set -ac on a video track [03:01:00 CEST] <prelude2004c_Zzz> oh.. sorry my bad.. i was trying to look at documention .. didn't notice that too :) [03:01:05 CEST] <prelude2004c_Zzz> oh ic [03:02:15 CEST] <prelude2004c_Zzz> YAY !!!!! [03:02:21 CEST] <prelude2004c_Zzz> 2 beers for furq [03:04:55 CEST] <spirou> when I try ffmpeg -f concat -i file.txt -c copy out.mp3 it spews a ton of "[mp3 @ 0xaeb19c0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 24196608 >= 24195814" upon me... [03:06:18 CEST] <spirou> I guess I don't have to worry about? [03:28:07 CEST] <snakeryslug> can anyone explain why a video that's one picture, if left out, is 300MB? i thought compression algs would make it much smaller. [03:28:48 CEST] <spirou> If I understand right the one-liner to remove a part in the middle of a file is ffmpeg -i in.mp3 -filter_complex "trim=duration=05:27[a]; trim=start=06:32[b],setpts=PTS-STARTPTS[d]; [a][b]concat[c]" -map [c] out.mp3 [03:29:21 CEST] <furq> spirou: that'll reencode [03:29:30 CEST] <spirou> snakeryslug: how do you mean? a video that is only one frame long? [03:30:40 CEST] <spirou> furq: oh... I don't know where to put a -c copy in there though... [03:31:05 CEST] <furq> you can't use filters without reencoding [03:31:18 CEST] <spirou> ow :'( [03:31:33 CEST] <spirou> ok it was a great idea at least :D [03:31:47 CEST] <snakeryslug> spirou: it's one of those videos where it's just the same picture through the whole thing and the important part is the audio. i don't know how to measure frames, but when i extraced the audio only, the file size went down by 300MB. [03:32:53 CEST] <spirou> snakeryslug: ok, must have been a pretty bad way of store that frame yeah [03:33:13 CEST] <spirou> you didn't reencode the sound by mistake though? [03:34:04 CEST] <snakeryslug> i don't think so. i did `ffmpeg -i .\input.mp4 -codec copy -vn output.m4a` [03:49:34 CEST] <spirou> snakeryslug: are you sure the picture is actually still, they don't do something stupid as slowly slightly zoom out the picture or something? [03:50:02 CEST] <TAFB2> I want ffmpeg to stream a short video in a loop but insert random .mp3 audio songs, possible? [03:51:04 CEST] <spirou> TAFB2: so video from one file, and audio from multiple other mp3 files? [03:51:30 CEST] <furq> TAFB2: not with ffmpeg itself but you could pipe the audio in somehow [03:52:04 CEST] <TAFB2> got a pipe example? :) [03:52:15 CEST] <kepstin> could even just use a music player on shuffle and capture an alsa loop device or pulseaudio monitor device. [03:52:26 CEST] <spirou> Any idea what the "non monotonically increasing dts to muxer" meant btw? [03:52:51 CEST] <furq> spirou: does the file play back correctly [03:53:23 CEST] <spirou> furq: yeah no problem [03:53:52 CEST] <spirou> nothing I noticed when listen at least [03:56:04 CEST] <spirou> (that doesn't mean it is not wrong somehow anyway though) [04:00:07 CEST] <spirou> dts means time stamp if I understand right, I suppose it don't like the timestamps on the second half of the song [04:00:24 CEST] <furq> i'm not really sure how that works in mp3 [04:01:01 CEST] <spirou> Perhaps ffmpeg -ss 06:32.5 -i in.mp3 in2.mp3 is not right way to cut a beginning? [04:21:33 CEST] <snakeryslug> spirou: not 100% sure. i guess i'd have to use ffprobe to find out (right?) [04:23:29 CEST] <spirou> snakeryslug: yeah that would be a good idea, it would show how many kb/s the video stream have [04:34:48 CEST] <PacketPusha> I can's quite seem to get this figured out, I am using "ffmpeg -y -threads 8 -i "video.mp4" -an -sn -vsync 0 -vframes 10 -s 854x480 -vcodec mjpeg -pix_fmt yuvj420p -b:v 2000 -bt 20M "test_%d.jpg" -ss 10" so far but I want to skip the first 10 seconds and then every 30 seconds get 10 frames as images. [04:34:49 CEST] <PacketPusha> Suggestions? [04:35:48 CEST] <furq> PacketPusha: https://ffmpeg.org/ffmpeg-filters.html#select_002c-aselect [04:37:32 CEST] <furq> something like select='lt(mod(t\,750)\,10)' should work for 25fps video [04:37:46 CEST] <furq> replace 750 with fps * 30 [04:37:47 CEST] <spirou> shouldn't the -ss 10 be before the -i in that? [04:37:57 CEST] <furq> it can be before or after [04:38:03 CEST] <furq> although you don't need it if you use select [04:38:28 CEST] <furq> or -vframes [04:38:38 CEST] <spirou> aha [04:39:48 CEST] <furq> er [04:39:58 CEST] <furq> select='lt(mod(n\,750)\,10)' [04:40:34 CEST] <furq> will select the first 10 frames of every 750 [04:40:39 CEST] <PacketPusha> Interesting, I'll test this examples out [04:40:59 CEST] <PacketPusha> Is there a way to do it based on time? [04:41:10 CEST] <furq> n is frames, t is time [04:41:22 CEST] <PacketPusha> Since I don't control the video sources and their fps [04:41:28 CEST] <PacketPusha> Ohh, so it does [04:41:45 CEST] <furq> that won't work with time if you want 10 frames though [04:42:13 CEST] <furq> there's probably some more-complicated select invocation which will do it [04:43:38 CEST] <snakeryslug> part [04:47:51 CEST] <furq> actually [04:48:20 CEST] <furq> select='between(mod(n\,750)\,251,260)' [04:48:20 CEST] <Xen0_> j [04:48:33 CEST] <furq> will skip the first 10 seconds as well [04:48:54 CEST] <furq> you're probably better off just getting the framerate with ffprobe [04:55:34 CEST] <PacketPusha> ffprobe? [04:55:58 CEST] <PacketPusha> Oh, I see [04:56:03 CEST] <PacketPusha> And then using that info? [04:56:08 CEST] <PacketPusha> *use [04:58:08 CEST] <furq> ffprobe -v error -of csv -select_streams v:0 -show_entries stream=r_frame_rate foo.mkv | cut -d, -f2 [04:58:30 CEST] <furq> and then yeah just use that * 30 [05:02:49 CEST] <damontoo> I'm trying to stream video from a pi camera module (h264) using ffserver to stream (webm) but I haven't been able to find anything that works. I sort of got it working but with FPS at less than 2, and latency around like 20 seconds. Any idea how to do this correctly? [05:03:22 CEST] <furq> ffserver isn't very good, and also i doubt the rpi will be able to encode vp8 or vp9 at >2fps [05:03:44 CEST] <damontoo> hmm [05:04:01 CEST] <furq> the rpi does have a hardware h.264 encoder but you can't use it from ffmpeg [05:04:04 CEST] <furq> you can use it from gstreamer [05:04:22 CEST] <damontoo> I was usiung raspivid and piping the output into ffmpeg [05:04:36 CEST] <damontoo> using* [05:05:25 CEST] <furq> that works, but the webm transcode will be running on the cpu which is hopeless [05:05:46 CEST] <furq> i'd probably pipe it into ffmpeg and have it send the h.264 stream to an rtmp or rtsp server or something similar [05:05:47 CEST] <damontoo> so is vp8/9 much more intensive than say, h264 to an mjpeg stream? [05:06:03 CEST] <furq> libvpx is much slower than x264 [05:06:09 CEST] <damontoo> because mjpeg people get pretty high fps. Like 15-30 [05:06:11 CEST] <furq> but i wouldn't do x264 encoding on the rpi if i could avoid it [05:06:48 CEST] <furq> stick with the hardware encoder or just send a raw stream to a machine which can encode it properly [05:07:28 CEST] <duncan_donuts> Hi, I am trying to use the hash muxer, but am getting not a suitable format error [05:07:42 CEST] <duncan_donuts> md5 muxer works though [05:07:50 CEST] <duncan_donuts> version 2.7.2 Ubuntu server [05:07:54 CEST] <duncan_donuts> Any clues? [05:08:06 CEST] <damontoo> Can't. This is for a remote ipcam with opencv integration. Which people have working with < 150ms latency and fairly good FPS.. but by using rtsp/netcat so it doesn't stream to a browser [05:08:58 CEST] <furq> duncan_donuts: i think hash was added since 2.7.2 [05:09:23 CEST] <furq> damontoo: if you need browser support then rtmp might work for you [05:09:28 CEST] <furq> you'll need flash though [05:10:05 CEST] <furq> https://github.com/arut/nginx-rtmp-module [05:10:08 CEST] <furq> that's a good free rtmp server [05:10:45 CEST] <damontoo> I was trying to avoid plugins hoping for something that worked with HTML5, which is why I wanted webm. Buuut, I guess flash is acceptable. Why can't flash just die. :\ [05:11:07 CEST] <furq> well that module will also do hls and dash, but those introduce a lot of latency [05:11:28 CEST] <furq> if you need <5sec latency then those are pretty much worthless [05:11:43 CEST] <damontoo> hell no. I'm fine at like < 500ms [05:11:47 CEST] <furq> and you'll need a lot of tuning to get it close to 5 seconds [05:11:58 CEST] <damontoo> oooh [05:12:01 CEST] <damontoo> 5 seconds [05:12:04 CEST] <furq> yeah [05:12:24 CEST] <furq> i've never really needed <10sec latency so maybe it's possible to tune them lower [05:12:33 CEST] <furq> but they're always going to be worse than a proper streaming protocol [05:13:33 CEST] <damontoo> yeah the latency isn't as bad as the frame rate. OpenCV will be reading license plates and tracking speed so better framerates are needed more [05:14:13 CEST] <damontoo> I've seen demos of OpenALPR running on the pi and it works well [05:14:20 CEST] <furq> well i assume raspivid uses the hardware encoder, so the framerate should be ok [05:14:39 CEST] <furq> you'd just pipe the output to ffmpeg and have it send it directly to the server without transcoding [05:14:48 CEST] <furq> or use gstreamer directly [05:15:34 CEST] <damontoo> mmk. Thanks. I'll research those options [05:16:15 CEST] <furq> html5 streaming is still a mess [05:20:09 CEST] <duncan_donuts> furq: ok thanks, that would certainly explain it [05:20:23 CEST] <furq> duncan_donuts: http://johnvansickle.com/ffmpeg/ [05:21:11 CEST] <duncan_donuts> furq: brilliant, thanks! [05:34:55 CEST] <Threads> found encoding_tool changes writing application was wondering do you have one for writing library [06:08:43 CEST] <ferdna> helo... i am in need of help [06:09:53 CEST] <ferdna> i need to stream a webcam to several computers [06:10:06 CEST] <ferdna> the webcam supports rtsp [06:45:35 CEST] <PacketPusha> furq: [06:45:39 CEST] <PacketPusha> ffmpeg -y -threads 8 -i "video.mp4" -an -sn -vsync 0 -vf select='lt(mod(n\,'$(( 30 * (`ffprobe -v error -of csv -select_streams v:0 -show_entries stream=r_frame_rate video.mp4 | cut -d, -f2`)))')\,10)' -s 854x480 -vcodec mjpeg -pix_fmt yuvj420p -b:v 2000 -bt 20M "test_%d.jpg" [06:47:41 CEST] <PacketPusha> Thanks again, that worked [06:48:57 CEST] <ferdna> good night [07:22:25 CEST] <johnnny22-afk> what's the best way to convert pgmyuv to jpeg ? [10:32:12 CEST] <realies> I wonder why I get multiple containers when ffprobing a clip... Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '\server\media\amb.mp4': [10:32:35 CEST] <realies> does that mean that the file has multiple wrappers on it? [10:42:10 CEST] <furq> realies: https://www.ffmpeg.org/ffmpeg-formats.html#mov_002fmp4_002f3gp_002fQuickTime [10:42:21 CEST] <furq> those are all different names for the same container [10:42:54 CEST] <realies> furq, thanks [11:04:09 CEST] <yzT> what's really the Common Encryption (CENC)? I know it's a standard for working with different DRM systems, but what is it really is? An independent server which handles the requests? Or something more "transparent" like the EME and MSE in HTML5 [11:09:45 CEST] <JEEB> it's a way of having your proprietary field in the manifest for how you handle the keycrap (and the w3c clearkey one which can have just the URL to the key), and a standard way of handling the encryption of the streams [11:10:08 CEST] <JEEB> that way you can share the key between DRM systems and just have the metadata for each of them in the manifest [11:10:49 CEST] <JEEB> it specifically doesn't specify how you handle the key stuff, each system handles that by themselves from the metadata provided in the manifest (and sometimes in the ISOBMFF segments) [11:11:26 CEST] <yzT> but, where it is processed? in the same server which serves the manifest? [11:11:31 CEST] <yzT> or is it something different [11:11:33 CEST] <Snoo20664> ot [11:11:41 CEST] <Snoo20664> it's normally on a diff server [11:11:53 CEST] <Snoo20664> but the manifest/m3u8 points to the key directly [11:11:56 CEST] <JEEB> CENC doesn't specify anything related to that [11:12:09 CEST] <JEEB> it just specifies the way different systems can get their information [11:12:12 CEST] <JEEB> from the manifest [11:12:42 CEST] <JEEB> CENC is just the spec for how the DASH side of things is handled [11:14:36 CEST] <yzT> so, in the connection workflow (starting from a user point of view), the CENC is done before the user retrieves the key, right? [11:19:33 CEST] <yzT> what do you think of this diagram --> http://i68.tinypic.com/157nrjs.png [14:49:16 CEST] <TheCycoONE> question about the av_dup_packet deprecation. After av_read_stream is it necessary to call av_packet_ref (where we were calling av_dup_packet). [15:03:40 CEST] <TheCycoONE> or I guess -- when is pkt->buff NULL after av_read_frame [15:04:04 CEST] <DHE> I see no such function in my copy of ffmpeg (based on git) [15:04:10 CEST] <DHE> that is av_read_stream [15:04:56 CEST] <TheCycoONE> s/stream/frame [15:05:02 CEST] <TheCycoONE> sorry [15:06:52 CEST] <DHE> once you have an AVPacket from av_read_frame it's yours. you would use av_packet_ref if you need a second independent copy of the packet. internally av_packet_ref uses an atomic reference counting system where possible [15:07:53 CEST] <TheCycoONE> ok, I was worried about "If pkt->buf is NULL, then the packet is valid until the next av_read_frame() or until avformat_close_input()." [15:08:16 CEST] <TheCycoONE> I know the old av_dup_packet was specifically to create a buffer when it was null, presumably to handle that case. [15:26:07 CEST] <TheCycoONE> DHE, is that true even if I'm queuing the packet potentially doing more calls to av_read_frame before it's decoded? [15:26:54 CEST] <TheCycoONE> or should I treat that as a new reference and unref both now [15:27:40 CEST] <TheCycoONE> https://github.com/CorsixTH/CorsixTH/blob/master/CorsixTH/Src/th_movie.cpp#… if you want code [15:28:05 CEST] <TheCycoONE> ^ is called from https://github.com/CorsixTH/CorsixTH/blob/master/CorsixTH/Src/th_movie.cpp#… [15:29:26 CEST] <DHE> TheCycoONE: the packet is yours once it comes out of av_read_frame. it's up to you to unref and/or free it (as the case may be) when you're done with it [15:30:21 CEST] <TheCycoONE> DHE, what does that note in the av_read_frame documentation about being valid until the next av_read_frame or avformat_close_input mean? [16:01:09 CEST] <DHE> TheCycoONE: I admit I'm still writing my first app myself. I'd want to double-check the specifics [16:51:32 CEST] <TheCycoONE> DHE, ah ok [16:51:54 CEST] <TheCycoONE> I'll double check everything with valgrind [18:21:15 CEST] <momomo> what do you think would be the best approach to push in advertisement into a hls stream ? [18:22:06 CEST] <momomo> i am thinking of manipulating the playlist but that would require me to do so all the time .. and update the media sequence field [18:22:12 CEST] <momomo> for all future playlists [20:55:34 CEST] <onezuff> trying to shorten videos by shaving one second off the end [20:55:49 CEST] <onezuff> ffmpeg -ss 01 -i clip.mp4 -vcodec copy -acodec copy shoter.mp4 is making videos of the same length. how can i shave off just 1 second? [20:56:14 CEST] <kepstin> onezuff: that would shave 1 second off the start (if it worked) [20:56:36 CEST] <kepstin> note that -ss isn't very accurate when using copy mode, because it has to start on a keyframe. [20:56:38 CEST] <onezuff> video is still coming out the same length, any ideas? [20:57:04 CEST] <kepstin> if you want to cut off the end of the video, use -t with a time 1s shorter than the video length - that *will* work in copy mode [20:57:27 CEST] <onezuff> is there a way to do this without calculating all the video lengths upfront? [20:57:29 CEST] <onezuff> :( [20:58:01 CEST] <kepstin> nope, but you can use e.g. 'ffprobe' to get the video length and script it. [20:58:15 CEST] <onezuff> well thats a relief at least lol [21:00:03 CEST] <onezuff> 239.552000 O___O [21:01:26 CEST] <onezuff> kepstin, if i shave it from the front will that just be easier? [21:01:47 CEST] <kepstin> shaving off the front will require re-encoding, unless there happens to be a keyframe in the right spot [21:02:09 CEST] <onezuff> my scripting skills are not good, dang [21:02:24 CEST] <kepstin> that's why it didn't seem to work when you tried it - it started at the closest keyframe before the time you requested, which just happened to be the start of the video. [21:03:17 CEST] <onezuff> found a flag that made the time readable - 0:03:59 [21:03:18 CEST] <kepstin> if you're ok with re-encoding, you can easily trim off the start with precision. [21:03:33 CEST] <onezuff> i dont want to re-encode, lots and lots of videos [21:03:50 CEST] <kepstin> well, if you're scripting, you want the machine-readable time rather than human-readable :) [21:04:00 CEST] <onezuff> oh i see [21:04:45 CEST] <kepstin> that way you just take the '239.552000', which is a time in seconds, subtrack 1, and then use '-t 238.552' as an option to ffmpeg. [21:05:19 CEST] <onezuff> i know this probably painfully simple but if i can figure this out i have just saved about 30 hours of work [21:05:33 CEST] <onezuff> i have over 500 videos i need to run this on [21:05:39 CEST] <onezuff> lol [21:08:31 CEST] <onezuff> for some reason its shaving 10 seconds off with that [21:16:26 CEST] <DHE> When converting surround sound to stereo, I find the volume to be rather low. Besides inserting a volume filter, is there something that can help? I'm using the swr_convert() API. Everything is working fine otherwise [21:21:58 CEST] <zamba> i need help with a/v sync issues when capturing from a capture card and alsa [21:22:04 CEST] <t4nk869> hey guys can anyone help me to export all frames (without knowing framerate) in a lossless format (bmp)? [21:22:28 CEST] <zamba> [alsa @ 0x500f840] ALSA buffer xrun [21:22:33 CEST] <zamba> Past duration 0.665962 too large [21:22:44 CEST] <zamba> are just some of the errors i'm getting [21:23:27 CEST] <kepstin> t4nk869: https://www.ffmpeg.org/ffmpeg-formats.html#image2-2 [21:24:07 CEST] <kepstin> if you don't include any -r or -frames options, it'll just do all frames. [21:25:19 CEST] <t4nk869> kepstin thank you but what about format I just add .bmp instead of .jpg? [21:25:29 CEST] <kepstin> yeah [21:25:49 CEST] <t4nk869> nice :) I will try it now [21:29:41 CEST] <t4nk869> I think something is wrong, it skips frames [21:29:43 CEST] <t4nk869> frame= 396 fps= 13 q=-0.0 size=N/A time=00:00:16.51 bitrate=N/A speed=0.528x frame= 403 fps= 13 q=-0.0 size=N/A time=00:00:16.80 bitrate=N/A speed=0.525x [21:30:54 CEST] <t4nk869> this is the command I used ffmpeg -i "source.mkv" "img\'bmp-%03d.bmp" [21:35:07 CEST] <geermain> Hi everyone, i've recently heard about framemd5, and I was trying to get some resources on how a frame checksum was computed. Anyone has relevant links about that? Thanks! [21:35:13 CEST] <geermain> http://ffmpeg.org/ffmpeg-formats.html#framemd5 [21:39:08 CEST] <Synal> Hi, is there a way of making ffmpeg continue a job that stoped mid way through from a previous time? [21:39:43 CEST] <kepstin> geermain: see the 'framehash' docs that are referenced there. It basically just takes the raw format audio/video and does a hash over the buffer. [21:40:47 CEST] <kepstin> (or if you manually specify a codec, it'll do the encoded packets of the codec instead...) [21:41:27 CEST] <t4nk869> kepstin sorry to bother you but I tried many options but it always skips frames [21:41:57 CEST] <llogan> Synal: if you mean stopped by suspending it in linux, then just run "fg" to unsuspend and return to foreground [21:42:03 CEST] <kepstin> t4nk869: what do you mean, skips frames? what's the ffmpeg output you see? [21:42:21 CEST] <t4nk869> frame= 396 fps= 13 q=-0.0 size=N/A time=00:00:16.51 bitrate=N/A speed=0.528x frame= 403 fps= 13 q=-0.0 size=N/A time=00:00:16.80 bitrate=N/A speed=0.525x [21:42:32 CEST] <t4nk869> as you see it says frame 396 and then frame 403 [21:42:39 CEST] <geermain> Thanks @kepstin i'll have a look [21:42:46 CEST] <Synal> llogan: No, I mean it was set to encode a load of files, but my system rebooted. Is it possible to continue ffmpeg from where it left off? [21:43:00 CEST] <kepstin> t4nk869: that line doesn't update every frame, that's normal. [21:43:06 CEST] <llogan> Synal: you didn't specifiy, so I guessed. [21:43:18 CEST] <Synal> llogan: its alright, I should have been clearer [21:43:21 CEST] <t4nk869> kepstin oh I see, thank you [21:43:33 CEST] <llogan> Synal: unfortunately, the answer is probably "no" [21:44:22 CEST] <Synal> llogan: okay, thanks for the help. Looks like I'm leaving my machine running overnight tonight then :) [21:53:00 CEST] <llogan> Synal: was it a bash for loop or something? or just one long encoding? [21:53:38 CEST] <nadermx> Can I run a double pass to update the empty moov_atoms [21:53:53 CEST] <Synal> It was a bash for loop [21:59:57 CEST] <llogan> i suppose you can remove the finished files and then re-run it. make sure the file that it was halted on is redone because the output will be incomplete or corrupted [22:03:52 CEST] <Synal> llogan: Yeah, I'm thinking of doing that. [22:09:38 CEST] <benlieb> I'm making videos for distribution online. What is the most common video/audio format? [22:09:44 CEST] <benlieb> I'm not a pro... [22:11:32 CEST] <benlieb> I run a dance education website. Dance instructors that want me to host their older DVDs in digital format send me DVDs. When importing dvds with handbreak, the audio is AAC (core audio) is this common? Will it run on most setups? [22:16:09 CEST] <TheCycoONE> handbreak picks reasonable defaults, aac is common. [22:16:30 CEST] <TheCycoONE> I assume you picked the web profile [22:17:22 CEST] <kepstin> benlieb: yeah, the format that you'll want to use for best compatibility is aac audio, h.264 video in an mp4 file. [22:18:52 CEST] <benlieb> It's also by default using an m4v extension. FOLLOWERSSPECIAL.m4v but says format will be mp4 [22:19:00 CEST] <benlieb> should I change the extension to mp4? [22:20:28 CEST] <kepstin> doesn't really matter, but I think mp4 is probably more common? [22:22:06 CEST] <TheCycoONE> your other option is to upload to youtube and embed the youtube video - then they take care of all the transcoding [22:22:25 CEST] <TheCycoONE> and selecting appropriate formats for whoever is watching [22:22:48 CEST] <mike_> hello everyone [22:23:11 CEST] <Guest20457> can someone help with ffmpeg to ffserver streaming? [22:23:25 CEST] <Guest20457> unable to get the hang of it [22:24:10 CEST] <Guest20457> should the video codec off ffmpeg be the same in ffserver? [22:25:00 CEST] <JEEB> unless you think you can become an ffserver maintainer, run away from it from it [22:25:08 CEST] <JEEB> there are actual streaming servers [22:25:16 CEST] <JEEB> and ffmpeg itself can be used for a lot [22:26:05 CEST] <Guest20457> i know , just playing around with it , i need to build my own streaming server , could it be that hard :D [22:26:31 CEST] <Mavrik> ... [22:26:44 CEST] <JEEB> ffserver will literally get removed unless someone actually wants to maintain it and bring it to the 21st century [22:26:53 CEST] <JEEB> because it's full of ugly [22:27:00 CEST] <JEEB> and nobody knows it [22:27:29 CEST] <Guest20457> what i am trying to do is the following, stream raw video from webcam , from 1slow pc ( hence i wont do transcoding in it ) , recieve via ffserver [22:27:40 CEST] <Guest20457> mmm , so this is only for ffmpeg you mean? [22:27:59 CEST] <Guest20457> and you do not advice to use ffserver? [22:28:15 CEST] <JEEB> yes, unless you want to actually start being a maintainer for ffserver, keep away [22:28:34 CEST] <Guest20457> ahan , got it, thank you will keep that in mind [22:28:36 CEST] <JEEB> use icecast or nginx-rtmp or many of the other open source things [22:28:53 CEST] <JEEB> you can feed them with ffmpeg [22:29:06 CEST] <Guest20457> yes, but can they do the same as ffserver? [22:29:18 CEST] <JEEB> depends on what exactly you want from ffserver [22:29:29 CEST] <Guest20457> i tried nginx-rtmp , but i want different thing [22:30:00 CEST] <JEEB> I mean, you can also just push stuff out from the slow machine, take it in with ffmpeg or something else and transcode it, and then feed it to whatever streaming server or spew it out with just ffmpeg [22:31:08 CEST] <Guest20457> i have 6 computers, slow ones, each have 2 capture cards, i want to stream them to a server where all the stranscoding stuff happens and then publish them to wherever, after that i can use either nginx-rtmp and flowplayer [22:31:52 CEST] <Guest20457> ok , so ffmpeg will take the raw input, transcode and re-stream you mean? [22:32:20 CEST] <Guest20457> so rather then running streams inside ffserver i can just use several ffmpeg instances [22:34:01 CEST] <Guest20457> can you guide me to a website or a book to learn more about this? [22:34:22 CEST] <Guest20457> with examples of streaming raw and re-streaming [22:48:30 CEST] <Guest20457> hi again [22:49:15 CEST] <Guest20457> can anyone point me to the right direction to stream uncompressed codec from ffmpeg to ffmpeg using udp [22:49:38 CEST] <Guest20457> and what codecs are supported for that [22:50:01 CEST] <BtbN> you don't want to stream uncompressed video [22:50:23 CEST] <BtbN> that can easily overwhelm even a gigabit link. [22:50:24 CEST] <Guest20457> it will be on local net [22:50:32 CEST] <Guest20457> no multicast just uni [22:50:50 CEST] <BtbN> So? More than 1 Gbit/s is still too much. [22:51:06 CEST] <BtbN> unless you happen to have a 10GbE network [22:51:17 CEST] <Guest20457> i have no choice , will have to adjust the max speed somehow [22:51:31 CEST] <BtbN> why would you have no choice? [22:51:55 CEST] <BtbN> to me this sounds like you are looking for a lossless codec, not uncompressed frames. [22:51:57 CEST] <Guest20457> nop , 1GB wired , but the reason i am doing this cause the source of stream is a slow cpu [22:52:01 CEST] <Mavrik> I very much doubt you have a network capable of eating several raw video streams. [22:52:11 CEST] <Mavrik> And network gear that can saturate 1GB ports. [22:52:19 CEST] <Guest20457> mmm [22:53:17 CEST] <Guest20457> yes al pcs have 1gb ports [22:53:53 CEST] <Guest20457> besides , i could manage to restrict ffmpeg data flow using some parameters right? [22:54:20 CEST] <Guest20457> anyway , what codec supports transport stream [22:55:09 CEST] <BtbN> how would you restrict the data flow? uncompressed is uncompressed. Nothing to adjust, it has the bandwidth it has. [22:55:37 CEST] <Guest20457> yep its 400Kps [22:55:44 CEST] <Guest20457> its a usb low end tv tuner [22:55:51 CEST] <BtbN> 400Kbps? [22:55:57 CEST] <Guest20457> and each pc has 2 [22:56:20 CEST] <BtbN> an uncompressed 720p60 stream is already over 1GBit/s [22:56:43 CEST] <Mavrik> Yeah, 1Gbit won't do. [22:56:49 CEST] <Mavrik> Don't stream uncompressed. [22:56:52 CEST] <Mavrik> Just... don't. [22:57:06 CEST] <BtbN> And if it's a TV tuner, you get an encoded stream as input anyway, so why don't just send that? [22:57:14 CEST] <Guest20457> yes ur probably right [22:58:01 CEST] <Guest20457> it says bgr24 [22:58:07 CEST] <Guest20457> isint that rawvideo [22:58:49 CEST] <TD-Linux> a TV tuner stream will fit over gigabit fine [22:59:01 CEST] <Mavrik> Maybe. [22:59:32 CEST] <TD-Linux> it has to fit over USB 2.0 :) [22:59:46 CEST] <Mavrik> So? [22:59:55 CEST] <Mavrik> It's still a silly thing to do. [22:59:57 CEST] <Guest20457> ill try to encode to something that wont eat much cpu , and get streamed quicly, since in both case i cannot use uncompressed streames for all the inputs [23:00:04 CEST] <TD-Linux> not really. why bother compressing if you don't have to [23:00:16 CEST] <Mavrik> Especially since on even an enterprise 10Gbit network you'll have issues with latency when you saturate it enough. [23:00:24 CEST] <Mavrik> Just... don't. [23:00:34 CEST] <Mavrik> Keep network utilization low to keep streaming working. [23:00:39 CEST] <Mavrik> Use a cheap losless compression. [23:00:48 CEST] <Mavrik> It'll make life easier on all sides. [23:00:51 CEST] <BtbN> TD-Linux, not if you decode it first... [23:01:05 CEST] <TD-Linux> lossless doesn't provide any guarantees about maximum bandwidth so it's not super useful here [23:01:19 CEST] <TD-Linux> BtbN, right but USB TV tuners output uncompressed 4:2:2 usually [23:01:21 CEST] <Guest20457> thats what i though TD-Linux, but for the long run Mavrik is right, since i have 6 pcs with 2 tv tuner each, would need a 1GB port on destination server for each pc, wish i dont have :D [23:01:22 CEST] <BtbN> lossless won't ever saturate a gigabit link [23:01:33 CEST] <BtbN> TD-Linux, USB TV tuners output mpegts with h264/mpeg2 usualy. [23:01:46 CEST] <TD-Linux> maybe a HD TV tuner [23:01:49 CEST] <TD-Linux> I guess that wasn't specified [23:01:53 CEST] Action: TD-Linux assumed analog tv [23:02:05 CEST] <BtbN> I have never heard of an analog usb tuner [23:02:12 CEST] <TD-Linux> wat [23:02:13 CEST] <Mavrik> You do a lot of assumptions and then you have a problem with losless not providing guarantees? :P [23:02:26 CEST] <Guest20457> no , its a cheap empia tv tuner [23:02:28 CEST] <Mavrik> BtbN, I think he means SD DVB [23:02:31 CEST] <Mavrik> or ATSC [23:02:36 CEST] <TD-Linux> um no I mean NTSC [23:02:39 CEST] <Mavrik> Since analog TV is dead. [23:02:40 CEST] <TD-Linux> or PAL [23:02:54 CEST] <Mavrik> And kinda irrelevant for last few years. [23:03:12 CEST] <BtbN> Even with SD DVB, you get encoded mpeg2 video and some eqaually ancient audio codec [23:03:12 CEST] <Guest20457> yes its dead, but i live in Lebanon and thats the only way to re-stream local channels [23:04:02 CEST] <Guest20457> the hardware will definetly get upgraded once the design is in place, and i get the hand of ffmpeg at least the basics [23:04:09 CEST] <Mavrik> Guest20457, anyway, I'd do "ffmpeg -i <grab parameters> -codec:v libx264 -qp 0 -preset ultrafast -f mpegts udp://<target machine>:<target port>" to stream losless h.264 [23:04:24 CEST] <Mavrik> and then grab the input on the other machine with ffmpeg -i udp://127.0.0.1:<port> ... [23:04:39 CEST] <Guest20457> Yes thats the best option for now , thank you [23:04:43 CEST] <Mavrik> Mix in some audio params and should be good enough [23:04:47 CEST] <BtbN> shouldn't it be 0.0.0.0:port? [23:04:54 CEST] <Guest20457> it can [23:04:56 CEST] <Mavrik> Yes, 0.0.0.0 or ip [23:04:58 CEST] <Mavrik> sorry :) [23:05:03 CEST] <Guest20457> :D [23:05:10 CEST] <Guest20457> thanks , already got it [23:05:11 CEST] <Mavrik> Anyway, we use UDP/MPEG-TS transport alot across datacenters and works well [23:05:17 CEST] <Mavrik> If your network works well of course. [23:06:36 CEST] <Guest20457> sure, its a local network, lets worry about output streams once i get this working, but its no big issue, a good server for encoding and a good internet uplink will do the job [23:07:03 CEST] <Guest20457> anyway, thank you all for your help [23:07:17 CEST] <Guest20457> will definitly come back for more :P [00:00:00 CEST] --- Tue Apr 19 2016

1 0

ffmpeg-devel.log.20160417
by burek 18 Apr '16

18 Apr '16

[01:00:14 CEST] <cone-359> ffmpeg 03Rostislav Pehlivanov 07master:77fb7177afd6: vc2enc: don't require interlacing for 1080p50/60 base video formats [01:00:14 CEST] <cone-359> ffmpeg 03Rostislav Pehlivanov 07master:9e138c481306: vc2enc: use 32x16 slices by default [01:36:02 CEST] <cone-359> ffmpeg 03Michael Niedermayer 07master:618032da0d05: fate: add test for Ticket4816 (PNG decoding with alpha) [02:04:49 CEST] <cone-359> ffmpeg 03Michael Niedermayer 07master:487c346d98c0: avcodec/exr: Fix "libavcodec/exr.c:1494:13: warning: ISO C90 forbids mixed declarations and code" [02:33:18 CEST] <cone-359> ffmpeg 03James Almer 07master:b63ba3a51782: avformat/hashenc: add missing avio_flush to hash_write_trailer [02:41:41 CEST] <cone-359> ffmpeg 03Ivan 07master:3a727606c474: avcodec/h264: Fix for H.264 configuration parsing [11:32:37 CEST] <cone-744> ffmpeg 03Paul B Mahol 07master:7a0aee16881d: avcodec/takdec: fix decoding of some sample rates with multichannel coder [13:19:31 CEST] <cone-925> ffmpeg 03Michael Niedermayer 07master:c8d95e5ceb72: avfilter/colorspacedsp_template: Add argument protecting () to avg() [13:34:08 CEST] <Daemon404> "Reproduces a bug to remain consistent with libvpx' behaviour." :( [13:37:52 CEST] <atomnuker> why won't libvpx fix the bug themselves? [13:37:53 CEST] <wm4> just following the spec [13:38:27 CEST] <wm4> atomnuker: maybe it'd break some youtube videos [13:49:48 CEST] Action: Daemon404 wonders if this is the entire OS/2 community on the ML [14:07:42 CEST] <Compn> haha [14:07:47 CEST] <Compn> they are active though [14:08:00 CEST] <Compn> more active than some other archs... [14:22:30 CEST] <BtbN> how is OS/2 even still alive? [14:23:02 CEST] <Daemon404> because korea [14:24:30 CEST] <atomnuker> which korea? [14:27:23 CEST] <Daemon404> the one who is allowed to post on the internet [14:29:02 CEST] <JEEB> Daemon404: at least one of the guys is clearly running it in a VM [14:29:20 CEST] <JEEB> but yes, quite funky that it's still alive there [15:24:05 CEST] <Compn> best korea has no need for internet [15:46:52 CEST] <cone-925> ffmpeg 03Ronald S. Bultje 07master:6135aed0d2b2: vf_scale: support bt2020 in in/out_color_matrix properties. [15:46:53 CEST] <cone-925> ffmpeg 03Ronald S. Bultje 07master:ee729cc0ed0b: vp9: ignore frame_context_index field in key-/intraonly-frames. [18:23:18 CEST] <cone-795> ffmpeg 03Michael Niedermayer 07master:6d160afab2fa: avfilter/vf_readvitc: Use avpriv_frame_get_metadatap() avoid direct access to AVFrame.metadata (to comply to API) [19:06:40 CEST] <Daemon404> nevcairiel, https://github.com/dwbuiten/FFmpeg/commit/fef5e35ae84665ff3a948fedec1154d8a… [19:06:46 CEST] <Daemon404> does this look correct to you [19:06:59 CEST] <Daemon404> (i am uncertain about the last changes from return 1 to 0) [19:07:18 CEST] <nevcairiel> the return value is just normal API returns now,0 or averror [19:07:20 CEST] <Daemon404> er, ignore the commented out bit too [19:07:22 CEST] <Daemon404> im rming that [19:07:26 CEST] <Daemon404> ok [19:07:31 CEST] <nevcairiel> EAGAIN in case the bsf eats a packet [19:07:38 CEST] <nevcairiel> but this one doesnt [19:07:58 CEST] <Daemon404> yes [19:10:35 CEST] <Daemon404> nevcairiel, anything look obviously wrong? [19:11:09 CEST] <nevcairiel> i usually preferred to get rid of buf/buf_size and reference in->data directly, but not like I care [19:11:38 CEST] <Daemon404> i can do that; it's just cosmetics [19:12:14 CEST] <nevcairiel> posted two comments [19:14:24 CEST] <Daemon404> nevcairiel, av_packet_copy_props is called directly after new_pacjet, right? [19:14:32 CEST] <nevcairiel> makes no difference when [19:14:36 CEST] <Daemon404> ok [19:14:42 CEST] <nevcairiel> just after new packet [19:15:22 CEST] <Daemon404> fixes for both comments pushed [19:15:37 CEST] <nevcairiel> posted a third one =p [19:15:57 CEST] <nevcairiel> dont think copy props can fail [19:16:07 CEST] <Daemon404> it can [19:16:18 CEST] <Daemon404> https://www.ffmpeg.org/doxygen/2.4/avpacket_8c_source.html#l00561 [19:16:33 CEST] <Daemon404> at least we have code that checks if it can [19:21:00 CEST] <Daemon404> also fixed [19:21:31 CEST] <Daemon404> im not sure if out must be freed on error or not [19:21:39 CEST] <nevcairiel> it must [19:21:42 CEST] <nevcairiel> you have ownership [19:21:49 CEST] <nevcairiel> well, if you allocated anything [19:21:55 CEST] <nevcairiel> if its untouched then it d oesnt matter [19:22:08 CEST] <Daemon404> all right [19:22:25 CEST] Action: Daemon404 really couldnt care less about learning bsf internals here, just wants to finish it [19:23:20 CEST] <Daemon404> tehre [19:23:50 CEST] <Daemon404> i find it odd that the filter must free the in packet, though [19:30:40 CEST] <nevcairiel> this way it could buffer it if it uses it partially [19:30:43 CEST] <nevcairiel> it takes ownership [19:30:58 CEST] <Daemon404> hmm ok [19:31:02 CEST] <Daemon404> any other issue left? [19:38:08 CEST] <nevcairiel> didnt see anything [19:38:12 CEST] <nevcairiel> but no clue what that thing even does [19:38:12 CEST] <Daemon404> ok [19:48:57 CEST] <cone-795> ffmpeg 03Anton Khirnov 07master:33d18982fa03: lavc: add a new bitstream filtering API [19:48:58 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:af9cac1be175: Merge commit '33d18982fa03feb061c8f744a4f0a9175c1f63ab' [19:49:09 CEST] <Daemon404> rcombs, ^ [19:50:47 CEST] <cone-795> ffmpeg 03Anton Khirnov 07master:4426540f0c3e: avconv: switch to the new BSF API [19:50:48 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:98e3153fa3a1: Merge commit '4426540f0c3ee516662f79d0a6ab5b95503b6611' [19:52:51 CEST] <cone-795> ffmpeg 03Vittorio Giovara 07master:6202e2fede75: indeo4: Rework stream analysis report [19:52:52 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:1117d6f4b122: Merge commit '6202e2fede75df92cbc374a3f7d6893d0c5ac721' [19:53:25 CEST] <cone-795> ffmpeg 03Vittorio Giovara 07master:64250d94b74d: indeo4: Consistently initialize variables [19:53:26 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:d9357332ae9a: Merge commit '64250d94b74d3fd47cc8b1611f48daf6a6ed804a' [19:55:01 CEST] <petrurares> Hi there! I want to save a new frame side data, concretely AVMasteringDisplayMetadata. I tried to put some random data in doc/examples/decoding_encoding.c example but I can't retrieve back such data [19:55:11 CEST] <petrurares> any idea how to do this? I want to make a test but firstly I need to know how to do that. Thanks :) [20:06:54 CEST] <cone-795> ffmpeg 03Vittorio Giovara 07master:709c0f79d803: nuv: Use the correct context for av_image_check_size [20:06:55 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:dff9894acdf8: Merge commit '709c0f79d8032fcf733bfe58e79ca7ff0858c8bc' [20:12:02 CEST] <cone-795> ffmpeg 03Vittorio Giovara 07master:0c6a70873fc6: intrax8: Move error resilience out of intrax8 [20:12:03 CEST] <cone-795> ffmpeg 03Vittorio Giovara 07master:750562549cee: intrax8: Wrap multiline macros in do{}while(0) clauses [20:12:04 CEST] <cone-795> ffmpeg 03Vittorio Giovara 07master:42244ce07a1f: intrax8: Move a comment to the place it corresponds [20:12:05 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:909f5e71001f: Merge commit '0c6a70873fc6e43194b471d112c30823b6c8d0b4' [20:12:06 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:e445b80c741c: Merge commit '750562549ceef268b29b94f6a887d9cf331a8c78' [20:12:07 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:2c3e38f44fd2: Merge commit '42244ce07a1f4f5108ae86d50fe72db785d483ed' [20:14:22 CEST] <cone-795> ffmpeg 03Diego Biurrun 07master:6f5ff559dbd0: intrax8: Adjust printf conversion specifier for sizeof expression [20:14:23 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:7120bff0a326: Merge commit '6f5ff559dbd01fa4ea96bc0a1d2e0c21f2db8d13' [20:28:14 CEST] <cone-795> ffmpeg 03Vittorio Giovara 07master:2ade1cdafb96: intrax8: K&R formatting cosmetics [20:28:15 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:ba5bcf96124a: Merge commit '2ade1cdafb96bf47e77f7ed74731d78a30aae950' [20:30:17 CEST] <cone-795> ffmpeg 03Vittorio Giovara 07master:ad8aa8e6c648: intrax8: Move documentation from implementation to header file [20:30:18 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:c29ac4abb818: Merge commit 'ad8aa8e6c648b61f01b9f106f27b9d4f3d094345' [20:32:11 CEST] <cone-795> ffmpeg 03Vittorio Giovara 07master:0372e73f917e: intrax8: Check and propagate errors from ff_intrax8_common_init [20:32:12 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:f64b53f2bf5f: Merge commit '0372e73f917e72c40b09270f771046fc142be4a7' [20:32:52 CEST] <cone-795> ffmpeg 03Vittorio Giovara 07master:d909f43b5c77: vc1dec: wmv2dec: Validate ff_intrax8_common_init return value [20:32:54 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:61400a1dc751: Merge commit 'd909f43b5c773a73c8d526638744547ba4aa8c59' [20:37:22 CEST] <cone-795> ffmpeg 03Diego Biurrun 07master:8dead2aaca4a: Move const qualifier before type name [20:37:23 CEST] <cone-795> ffmpeg 03Diego Biurrun 07master:48362ceadeb2: doc: Update paths to match new examples location [20:37:24 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:9d4799c247e3: Merge commit '8dead2aaca4aa8b84b77b05745755afb56b7d37a' [20:37:25 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:b0c75963c87d: Merge commit '48362ceadeb2eb5286ae94ef7f9542d990ff7ec7' [20:40:21 CEST] <cone-795> ffmpeg 03Diego Biurrun 07master:30e9ef21cea0: timefilter-test: Only compile timefilter-test if JACK is enabled [20:40:22 CEST] <cone-795> ffmpeg 03Diego Biurrun 07master:65a802401c6c: build: Add component for the SRTP common code [20:40:23 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:9b36f446acbf: Merge commit '30e9ef21cea09fa5e880e979c9f5b39edccbb6f4' [20:40:24 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:8f540b86125a: Merge commit '65a802401c6cc136576bb2e613c0577cbf622aa8' [20:41:06 CEST] <cone-795> ffmpeg 03Diego Biurrun 07master:52385410b7de: des-test: Move a variable declaration to avoid an ifdef [20:41:07 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:4b23a8137fd3: Merge commit '52385410b7de012c2562fb5ba8f07a4e1112c4e7' [20:44:52 CEST] <cone-795> ffmpeg 03Carl Eugen Hoyos 07master:1106a2c5dd7e: lavc/webp: Print a warning for unsupported chunks. [20:46:19 CEST] <cone-795> ffmpeg 03Diego Biurrun 07master:a84713e70d06: parseutils-test: Move some variable declarations to avoid block braces [20:46:20 CEST] <cone-795> ffmpeg 03Derek Buitenhuis 07master:d97a61a8f132: Merge commit 'a84713e70d06238100cb0f867ad6031c272054b0' [21:26:25 CEST] <JEEB> wm4: I love how YUV_TO_RGB1_CCIR et al are completely undocumented unless you go search for the values [21:30:45 CEST] <wm4> they also can't be used without carefully looking what these macros expand to [21:31:45 CEST] <JEEB> yeah [21:32:14 CEST] <cone-795> ffmpeg 03Thilo Borgmann 07master:37a4d3383fb9: lavfi/perspective: Add basic timeline editing. [21:32:31 CEST] <JEEB> which explains why my initial question regarding lack of explanation of those macros [21:32:38 CEST] <JEEB> *-lack [21:32:42 CEST] <JEEB> *-lack of [21:33:01 CEST] <JEEB> didn't get any real reply :P [00:00:00 CEST] --- Mon Apr 18 2016

1 0

ffmpeg.log.20160417
by burek 18 Apr '16

18 Apr '16

[00:06:05 CEST] <__raven> hm i tried some of your hints but i always get errors with input data :( [00:09:09 CEST] <kepstin> __raven: well, first thing to check is whether ffmpeg can read all of your input files individually [01:10:29 CEST] <speak> ffmpeg -i speak-motion.flac -i logo.png -filter_complex "[0:a]showcqt='axisfile=line.png: axis_h=2', format=yuv420p[vid]; [1:0]overlay=x=W-w-10:y=10" -map "[vid]" -map 0:a -codec:v libx264 -crf 18 -preset fast -codec:a aac -strict -2 -b:a 192k speak-motion_nofont.mp4 [01:10:38 CEST] <speak> I get "Cannot find a matching stream for unlabeled input pad 1 on filter Parsed_overlay_2" [01:11:06 CEST] <speak> I've had a bit of hard time finding good examples on filter_complex working on multiple inputs, so I know I'm doing it incorrectly somehow [01:34:40 CEST] <rabelais> I'm trying to bring an overlay in to a video at a specific time and have it fade in. I was able to get the overlay to happen at the appropriate time (e.g., 3 seconds after the start of the video), but I can't seem to also get the fade in to happen as well, it says that fade doesn't support enable. Is there a way to do this? [04:32:54 CEST] <prelude2004c2> hey guys.. good evening.. can someone help me.. as an input i am encoding and every time i rerun things the audio is always jumping around from the mapping .. is it possible to map things correctly on the input before i pipe it to the output for encoding [04:32:56 CEST] <prelude2004c2> http://pastebin.com/Zcj8hs3E [04:40:23 CEST] <prelude2004c2> can anyone help? [04:50:26 CEST] <Snoo20664> whats your line [04:50:31 CEST] <Snoo20664> encoding line* [04:52:13 CEST] <furq> prelude2004c2: which input do you want [04:54:40 CEST] <prelude2004c2> "-map i:0x1011 -map i:0x1013 -map i:0x1015" [04:54:43 CEST] <prelude2004c2> would something like that work ? [04:54:51 CEST] <prelude2004c2> to keep them in order for transcoding ? [04:55:02 CEST] <prelude2004c2> ignore the 0x values.. its from another sample [04:55:37 CEST] <prelude2004c2> real one is " "-map i:0x1211 -map i:0x1216 -map i:0x1215" " [04:55:41 CEST] <prelude2004c2> what do you think ? will that work ? [04:55:44 CEST] <furq> if the pids are always the same then that should work [05:26:24 CEST] <prelude2004c2> another question [05:27:07 CEST] <prelude2004c2> for some reason the " Stream #0:4[0x1215](san): Audio: mp2, 48000 Hz, stereo, s16p, 160 kb/s " is never copied over ? it shows that it has the input but transcoding never shows it [05:27:16 CEST] <prelude2004c2> and i have c:a:0 and c:a:1 set [05:27:22 CEST] <prelude2004c2> weird... i don't see an error [05:27:52 CEST] <furq> don't set those if you're also mapping by pid [05:28:16 CEST] <prelude2004c2> see http://pastebin.com/ZxchqKpn [05:28:32 CEST] <prelude2004c2> but i have to down convert it to different bit rate [05:28:45 CEST] <furq> oh [05:29:01 CEST] <prelude2004c2> like .. i am taking the UDP transport stream.. i am running the -map i:xxxxx to sort the ones i want.. i took one " eng " and one " san " [05:29:32 CEST] <prelude2004c2> then i am | ffmpeg -i - ... etc etc.. and converting from ac3 > aac and mp2 > aac [05:29:45 CEST] <prelude2004c2> the ac3 > aac comes through... the mp2 > aac is completely ignored [05:29:48 CEST] <prelude2004c2> odd. [05:30:04 CEST] <furq> you can just use -map 0 to select all input streams [05:30:34 CEST] <prelude2004c2> i dont want all though. sometimes it has multiple audio streams .. and i only want to select 2 max [05:30:45 CEST] <prelude2004c2> sometimes it comes with like 4-5 audio streams [05:31:12 CEST] <furq> the ffmpeg you're piping to will always have the same 3 streams [05:32:04 CEST] <furq> although i'm not sure why you're piping it at all [05:32:48 CEST] <prelude2004c2> oh why.. it was an old thing actually where i was setting up multiple bit rates and i was tee'ing it off to differnet work [05:32:53 CEST] <prelude2004c2> i guess not really applicable at the moment :) [05:32:59 CEST] <prelude2004c2> but i guess the problem is the same [05:34:09 CEST] <prelude2004c2> the input on the pipe is correct. shows me all the audio i want and video [05:34:24 CEST] <prelude2004c2> all is right.. its just that last encoding stage that is skipping the mp2 but its not telling me why [06:13:40 CEST] <prelude2004c2> hey ffmpeg support cc ? [06:13:51 CEST] <prelude2004c2> i don't seem to have that data too [06:14:05 CEST] <prelude2004c2> using -scodec copy [06:27:19 CEST] <furq> prelude2004c2: -c:s webvtt [06:27:20 CEST] <furq> for hls [06:27:31 CEST] <furq> assuming the ts has text subtitles, otherwise you'll need to convert them manually [06:31:54 CEST] <prelude2004c2> not subtittles.. i want the closed caption data [07:18:25 CEST] <sarmiena_> whats a good place to start learning about low latency streaming? [07:18:32 CEST] <sarmiena_> wheres* [07:47:49 CEST] <Snoo20664> i would just start experimenting sarmiena_ [07:48:05 CEST] <Snoo20664> and then asking an questions you have here as you go [07:48:35 CEST] <sarmiena_> so just use the man pages & start messing around? [07:48:53 CEST] <Snoo20664> yeah pretty much [07:48:57 CEST] <sarmiena_> ok [07:49:18 CEST] <Snoo20664> trial and error is best way to learn imo [07:49:48 CEST] <sarmiena_> so far i got the streaming working from my isight on MBP, but theres massive latency. like 50 seconds [07:50:00 CEST] <furq> what format are you using [07:50:05 CEST] <sarmiena_> sec [07:50:30 CEST] <sarmiena_> http://pastie.org/10800610 [07:50:53 CEST] <sarmiena_> then i run ffmpeg -f avfoundation -video_size 640x480 -framerate 30 -i "FaceTime" http://localhost:8090/feed1.ffm [07:50:58 CEST] <furq> ffserver ;_; [07:51:10 CEST] <sarmiena_> is that a bad thing? [07:51:28 CEST] <furq> ffserver is not very good and on the brink of being abandoned [07:51:37 CEST] <sarmiena_> whats the alternative? [07:51:50 CEST] <furq> i would suggest using HLS, but that's not much use if you want low latency [07:52:08 CEST] <furq> so the best alternative i know of is using an external streaming server [07:52:30 CEST] <sarmiena_> i basically want to do something like& i guess surveilance camera or something? [07:52:37 CEST] <furq> what kind of latency would be acceptable [07:52:42 CEST] <sarmiena_> but that external streaming server gotta be setup somehow [07:52:49 CEST] <sarmiena_> im thinking like 2 seconds [07:53:01 CEST] <furq> yeah HLS and DASH are probably no good then [07:53:30 CEST] <furq> the only thing i've used for streaming is nginx-rtmp, which you might be able to tune to <2 second latency [07:53:38 CEST] <furq> someone else might be able to suggest a better alternative [07:53:47 CEST] <sarmiena_> gotcha. thats interesting [07:54:22 CEST] <sarmiena_> so just point the ffmpeg command as above to nginx-rtmp? [07:54:26 CEST] <furq> yeah [07:54:28 CEST] <furq> https://gist.github.com/fur-q/d7028f51c38f7d0bb56e [07:54:31 CEST] <furq> the setup i use is there fwiw [07:54:43 CEST] <furq> you can ignore the lua bit if you don't need stream authentication [07:55:26 CEST] <sarmiena_> cool. thanks for the help furq ! [07:57:38 CEST] <furq> how are you currently viewing the streams btw [07:57:49 CEST] <sarmiena_> just via web browser [07:57:57 CEST] <furq> if it's in a browser then you'll need flash to view rtmp streams [07:58:07 CEST] <sarmiena_> gotcha. that shouldnt be a problem [07:58:16 CEST] <furq> ok good [07:58:32 CEST] <furq> nginx-rtmp can also do hls and dash muxing, but the latency will be horrible [07:58:35 CEST] <sarmiena_> i just want to stream video of something on my property for fun [07:58:45 CEST] <sarmiena_> whats that stuff? [07:58:58 CEST] <sarmiena_> hls and dash muxing [07:59:05 CEST] <furq> https://en.wikipedia.org/wiki/HTTP_Live_Streaming and https://en.wikipedia.org/wiki/Dynamic_Adaptive_Streaming_over_HTTP [08:00:25 CEST] <sarmiena_> so looks like those things are basically what i want to do [08:00:36 CEST] <sarmiena_> so theres no hope? [08:01:08 CEST] <furq> you'll struggle to get <5sec latency with HLS [08:01:17 CEST] <furq> i've never used DASH but it works in a similar way so i'd expect it to be the same [08:01:29 CEST] <sarmiena_> what else can be done? [08:01:44 CEST] <sarmiena_> stream it directly to a player? [08:01:53 CEST] <sarmiena_> like quicktime or something? [08:02:07 CEST] <furq> if you use nginx-rtmp then most players can play rtmp urls [08:02:39 CEST] <furq> you could also stream directly over udp with ffmpeg, but since it's not a proper server it'll only serve one client and stop running when the client disconnects [08:04:09 CEST] <sarmiena_> what do you mean by stream directly over udp? like to a player directly? [08:04:38 CEST] <furq> -i whatever udp://0.0.0.0:1234 [08:04:42 CEST] <furq> then open localhost:1234 in your player [08:04:52 CEST] <furq> or your lan ip or whatever [08:05:03 CEST] <furq> er [08:05:09 CEST] <furq> -i whatever -f mpegts udp://0.0.0.0:1234 [08:05:19 CEST] <sarmiena_> ok [08:05:32 CEST] <sarmiena_> and so with the nginx-rtmp thing. thats HLS, right? [08:05:42 CEST] <furq> that's rtmp but it can also do hls and dash [08:05:58 CEST] <furq> rtmp should do what you want [08:06:10 CEST] <sarmiena_> the way you have it setup, its hls, yeah? [08:06:18 CEST] <furq> it does both [08:06:25 CEST] <furq> you can just disable hls if you don't need it [08:07:09 CEST] <sarmiena_> i see. so say i do your setup. exact same setup. how do i view the stream? go to the browser? [08:08:14 CEST] <furq> play rtmp://yourip:1935/live/streamname in your player or in a flash video player [08:08:58 CEST] <sarmiena_> ok. i think its enough for me to start messing around. you in here a lot? [08:09:08 CEST] <furq> sure [08:09:14 CEST] <sarmiena_> cool [08:09:29 CEST] <sarmiena_> thanks for the help dude [13:45:10 CEST] <andrey_utkin_> fflogger8 !uncut me [13:46:23 CEST] <andrey_utkin_> oh how to get that scripted request of uncut output [15:29:53 CEST] <sopparus> hello [15:30:03 CEST] <sopparus> anyone know if nvenc works on freebsd? [15:39:20 CEST] <BtbN> sopparus, if the nvidia driver ships with the nvenc library, i'd guess so. [15:59:44 CEST] <andrey_utkin_> is it a known issue that ffmpeg ends with zero retcode even though it gets a lot of h264 decoding errors and "-err_detect explode" is given? have just tried with 2.8.6 and recent git master [16:03:00 CEST] <andrey_utkin_> it'd be much better if it returned failure retcode in such case, for git-bisect purpose [16:54:08 CEST] <prelude2004c_Zzz> hey everyone.. can i get some help [16:55:15 CEST] <prelude2004c_Zzz> http://pastebin.com/VVZ9GtJC [16:55:35 CEST] <prelude2004c_Zzz> can someone have a look and tell me what i am doing wrong. Why does the second stream never come through [17:45:55 CEST] <SpeakerToMeat> sfan5: Btw to revive a one month old discussion and then kill it because it's useless. Dad and I work in digital cinema and a lot of cinema and video professionals do not know that containers exist and about codecs [17:46:35 CEST] <SpeakerToMeat> Question, I could swear there was a guide about installing non-libav ffmpeg on debian, but I can't find it... does anybody have the url perchance? [17:46:40 CEST] <sfan5> i know that those exist [17:46:46 CEST] <SpeakerToMeat> I'd like to avoid the dependency "hell" of compiling from source. [17:46:48 CEST] <sfan5> i should get a job as video professional :P [17:49:05 CEST] <SpeakerToMeat> Wait or am I wrong and is jessie or backports to jessie that's back onto ffmpeg? [18:05:35 CEST] <SpeakerToMeat> Yes, it's in jessie-backports [18:12:57 CEST] <r3volver> hiii!!!! [18:13:01 CEST] <r3volver> ffmpeg -f alsa -r 16000 -i hw:2,0 -f video4linux2 -s 800x600 -i /dev/video0 -r 30 -f avi -vcodec mpeg4 -vtag xvid -qscale 0 -acodec libmp3lame -ab 96k output.avi [18:13:26 CEST] <r3volver> this command does not work because the audio hw ID is wrong [18:13:32 CEST] <r3volver> how can i discover the right id? [18:13:44 CEST] <r3volver> [alsa @ 0x563158d4eca0] cannot open audio device hw:2,0 (No such file or directory) [18:13:47 CEST] <r3volver> hw:2,0: Input/output error [18:14:03 CEST] <r3volver> if i remove the audio configuration in the command, it works fine. [18:14:04 CEST] <r3volver> =D [18:14:36 CEST] <sopparus> aplay -l I think [18:14:45 CEST] <sopparus> something simular [18:15:25 CEST] <r3volver> sopparus: its from the alsa suite right? [18:15:32 CEST] <r3volver> sopparus: outside ffmpeg [18:15:44 CEST] <sopparus> yes [18:16:11 CEST] <r3volver> it worked. thank you very much [18:16:13 CEST] <r3volver> =D [18:16:17 CEST] <r3volver> testing it right now [18:34:12 CEST] <rabelais> I'm trying to bring an overlay in to a video at a specific time and have it fade in. I was able to get the overlay to happen at the appropriate time (e.g., 3 seconds after the start of the video), but I can't seem to also get the fade in to happen as well, it says that fade doesn't support enable. Is what I'm hoping to do possible? [19:53:57 CEST] <prelude2004c> hey guys.. not sure i got an answer the last time but i am having some trouble with audio [19:54:04 CEST] <prelude2004c> see : http://pastebin.com/MSvYUQBa [19:54:18 CEST] <prelude2004c> not sure why the 2nd audio channel never gets encoded.. any ideas? [20:43:17 CEST] <DHE> prelude2004c: what does "-map i:0x1022" do? [20:44:38 CEST] <DHE> in fact, show us the real comand-line and not just cherry-picked output only [21:51:13 CEST] <PacketPusha> I can's quite seem to get this figured out, I am using "ffmpeg -y -threads 8 -i "video.mp4" -an -sn -vsync 0 -vframes 10 -s 854x480 -vcodec mjpeg -pix_fmt yuvj420p -b:v 2000 -bt 20M "test_%d.jpg" -ss 10" so far but I want to skip the first 10 seconds and then every 30 seconds get 10 frames as images. [21:51:17 CEST] <PacketPusha> Suggestions? [21:54:40 CEST] <prelude2004c> hey DHE sorry for the delayed response [21:56:20 CEST] <prelude2004c> please see full command here [21:56:21 CEST] <prelude2004c> http://pastebin.com/tVi1BhQU [22:11:06 CEST] <prelude2004c> so , anyone able to help ? [22:11:12 CEST] <prelude2004c> i am stuck on this last item [22:22:28 CEST] <jcd> Hi! [22:22:34 CEST] <jcd> Is anyone listening? [22:23:22 CEST] <jcd> I need perform phase reversal on an audio track [22:23:30 CEST] <jcd> (actually lots of them) [22:23:46 CEST] <jcd> but I haven't found a way to do it in the documentation [22:24:12 CEST] <jcd> Has anyone got any idea where to look? [22:24:29 CEST] <TAFB> I want ffmpeg to stream a short video in a loop put insert random .mp3 audio songs, possible? [22:24:40 CEST] <TAFB> but insert... [22:27:30 CEST] <TAFB2> I want ffmpeg to stream a short video in a loop but insert random .mp3 audio songs, possible? [22:47:30 CEST] <livingBEEF> Is it at all possible to get size of a stream with ffprobe or something? As far as I can tell, the only way is to extract the stream with -c:copy and look at the file size.. [22:47:53 CEST] <livingBEEF> Which isn't exactly practical, especially with bigger files [22:55:35 CEST] <Snoo20664> you can use mediainfo to get the stream sizes [22:58:45 CEST] <livingBEEF> This seems to work "ffmpeg -i <vid> -c copy -map <stream> -f <fmt> - | wc -c" [22:59:13 CEST] <livingBEEF> Thanks for the tip though. [23:00:50 CEST] <livingBEEF> Still, it would be nice if ffprobe could calculate avg bitrate when hard bitrate is not present. [00:00:00 CEST] --- Mon Apr 18 2016

1 0

ffmpeg-devel.log.20160416
by burek 17 Apr '16

17 Apr '16

[02:02:38 CEST] <cone-069> ffmpeg 03Michael Niedermayer 07master:c4fad0277991: avformat/matroskaenc: Undo bits_per_coded_sample change as bits_per_raw_sample is available again [02:02:38 CEST] <cone-069> ffmpeg 03Michael Niedermayer 07master:60517c3ad66f: avfilter/af_hdcd: Fix informations typo [02:29:06 CEST] <rcombs> atomnuker: how do PCEs work in ADTS AAC-LC? [02:29:33 CEST] <rcombs> or, actually, specifically ER AAC-LC [02:30:52 CEST] <rcombs> looks like it's just not supported in the decoder afaict [02:31:26 CEST] <rcombs> (no I don't have a sample, just looking through the code and noticed there's no handling for it in that code path) [02:34:21 CEST] <atomnuker> for adts they should work just normally [02:37:30 CEST] <rcombs> there's PCE handling in aac_decode_frame_int, but I don't see any in aac_decode_er_frame [02:37:55 CEST] <atomnuker> yeah, seems ER only uses channel tags [02:38:36 CEST] <atomnuker> could be just that it's not implemented though [02:38:39 CEST] <rcombs> does PCE in ER just not exist, or isyeah [02:38:45 CEST] <atomnuker> I'll take a look at the specs [02:39:16 CEST] <rcombs> cool [02:43:01 CEST] <atomnuker> rcombs: nope, seems like they're not supported [02:43:12 CEST] <rcombs> cool, thanks [02:43:41 CEST] <atomnuker> "Please note, that due to this modification coupling_channel_element(), data_stream_element(), program_config_element(), and fill_element() are not supported within the error resilient bitstream payload syntax." [04:17:39 CEST] <jamrial> nice, release candidate for gcc 6 [04:17:48 CEST] <jamrial> and it doesn't miscompile ffmpeg :p [04:17:53 CEST] <jamrial> final should be out next week [13:27:12 CEST] <Daemon404> [22:07] <+rcombs> Daemon404: nevcairiel: how far out do you suppose the new BSF API is? <-- when someone helps me figure out merge stuff for hevc/h264 and maybe help port [13:27:15 CEST] <Daemon404> (its in a branch) [13:27:37 CEST] <rcombs> maybe I'll take a look [13:27:42 CEST] <rcombs> tomorrow or so [13:27:47 CEST] <Daemon404> https://github.com/dwbuiten/FFmpeg/commits/bsf [13:28:41 CEST] <Daemon404> its not a ton of work, but i dont have the best grasp on bsfs. [13:28:45 CEST] <wm4> what's still missing? I also wanted to take a look [13:29:02 CEST] <Daemon404> todo is in the commit message for the two top commits there [13:29:09 CEST] <Daemon404> (the only 2 new ones) [13:35:23 CEST] <nevcairiel> I'll look at h264 and hevc, it should be no big trouble [13:35:57 CEST] <wm4> uh then I guess I'll continue being lazy and useless [13:36:56 CEST] <Daemon404> wm4, feel free to port shit [13:37:04 CEST] <Daemon404> saturdays are fairly busy for me [13:37:09 CEST] <Daemon404> (chore day, yay) [14:02:51 CEST] <nevcairiel> Daemon404: i came up with an evil plan, remove the stupid argument from the bsfs themself and instead hack it into the compat layer for the old API, that way it magically dies with it one day [14:18:14 CEST] <Daemon404> .. lulz [14:18:40 CEST] <Daemon404> perverted. [14:48:40 CEST] <nevcairiel> i may have fixed h264 and a few others, but still a bunch left that may need attention [14:49:07 CEST] <JEEB> nice [16:28:02 CEST] <durandal_170> can someone backport my apedec fix to all other branches? [16:40:59 CEST] <cone-603> ffmpeg 03Tobias Rapp 07master:2aad631a818c: avfilter: add readvitc filter [17:06:14 CEST] <nevcairiel> BBB: do we have a fate test for vp9_superframe? [17:06:25 CEST] <BBB> not that I know [17:06:49 CEST] <BBB> ffmpeg -i anyvp9file -c:v copy out.ivf [17:06:52 CEST] <BBB> should suffice [17:17:07 CEST] <nevcairiel> guess it works post-conversion then [17:17:11 CEST] <nevcairiel> produces identical output in any case [17:17:26 CEST] <nevcairiel> i should probably break it to see if it even affects the output =p [17:18:42 CEST] <nevcairiel> oh ok it immediately crys about broken timestamps if the bsf isnt used, suppose thats "normal"? [17:18:52 CEST] <BBB> right, thats what the bsf fixes [17:19:08 CEST] <BBB> not the timestamps, but the splitting of a superframe into two subframes with identical timestamps [17:19:22 CEST] <BBB> the bsf merges them back together so you have one frame with a unique timestamp (instead of two frames with the same tS) [17:20:11 CEST] <nevcairiel> feel like looking over https://github.com/dwbuiten/FFmpeg/commit/0895961d15e2ecf24845c7444fe2128e8… ? its kinda inflated due to changing the name of the context variable to a "standard" that every bsf now uses, so meh.. [17:20:19 CEST] <nevcairiel> the real changes are only how input and output works [17:21:57 CEST] <nevcairiel> i wonder if it should return something special when it "eats" a packet [17:22:03 CEST] Action: nevcairiel checks the API more [17:22:24 CEST] <BBB> the rename is a little confusing [17:22:26 CEST] <BBB> but it looks ok [17:22:47 CEST] <BBB> make sure that after -c:v copy, decoding the source and dest file both give identical framemd5s [17:22:55 CEST] <BBB> e.g. ffmpeg -i in -c:v copy out [17:23:00 CEST] <BBB> ffmpeg -i in -f framemd5 a [17:23:04 CEST] <BBB> ffmpeg -i out -f framemd5 b [17:23:08 CEST] <BBB> diff -q a b [17:23:11 CEST] <nevcairiel> i diffed the output file bit-wise and it was identical to before [17:23:11 CEST] <BBB> or so [17:23:19 CEST] <BBB> ok thats good enough [17:23:19 CEST] <BBB> tnx [17:24:56 CEST] <nevcairiel> i think its supposed ro return EAGAIN when it has no output [17:24:59 CEST] Action: nevcairiel changes [17:27:14 CEST] <wm4> yes it is [17:28:33 CEST] <nevcairiel> its the only filter that does this, amazing that the libav api definition even cared for that case =p [17:29:06 CEST] <nevcairiel> after these changes we should really improve fate coverage of bsf [17:29:10 CEST] <nevcairiel> its barely tested at all [17:30:09 CEST] <Daemon404> yep [17:30:14 CEST] Action: Daemon404 home now [17:30:41 CEST] <JEEB> it even has one bug that makes some things actually fail when used in real life, unless the prefix length issue got fixed [17:30:51 CEST] <JEEB> (the h264 bsf at least) [17:31:07 CEST] <nevcairiel> well fate tests dont find bugs, they just prevent regressions [17:31:48 CEST] <nevcairiel> Daemon404: i left mp3_header_decompress for you, ima go grab foodz noaw [17:34:18 CEST] <nevcairiel> and the aac bsf with its extradata is still broken, i can look at that after food [17:34:35 CEST] <Daemon404> ok [17:34:41 CEST] <nevcairiel> should probably pick the test rcombs wrote for that into master quickly, it should be independent from the remainder of his set [17:35:06 CEST] <Daemon404> i might wait till after dinner too to poke [17:35:12 CEST] <Daemon404> but ill look now [17:35:23 CEST] <nevcairiel> the mp3 one is probably not hard [17:35:28 CEST] <nevcairiel> didnt even open the file yet [17:37:36 CEST] <Daemon404> sure [20:23:03 CEST] <nevcairiel> Daemon404: pushed a fix for aac as well [20:24:12 CEST] <Daemon404> ok [20:24:17 CEST] <Daemon404> im just about to eat [20:24:23 CEST] <Daemon404> so i havent poked mp3 yet [20:30:31 CEST] <nevcairiel> we should really work on migrating the internal components that use bsf to the new api after, so many hacks accumulating for it [20:31:40 CEST] <nevcairiel> at least autobsf [21:01:25 CEST] <petrurares> Hi there! I want to save a new frame side data, concretely AVMasteringDisplayMetadata. I tried to put some random data in doc/examples/decoding_encoding.c example but I can't retrieve back such data [21:03:03 CEST] <petrurares> any idea how to do this? I want to make a test but firstly I need to know how to do that. Thanks :) [21:11:09 CEST] <petrurares> /msg nickserv register naturis psincraian(a)gmail.com [21:47:39 CEST] <Compn> petrurares : stick around, i think the devs are all sleeping atm [21:47:42 CEST] <Compn> doh [21:51:45 CEST] <petrurares> lol :') Thanks Compn, for today I'm done [00:00:00 CEST] --- Sun Apr 17 2016

1 0

ffmpeg.log.20160416
by burek 17 Apr '16

17 Apr '16

[00:00:50 CEST] <c_14> I think only vp9 has a lossless mode [00:01:17 CEST] <c_14> You can use -c:v libvpx-vp9 and then add -lossless [00:01:58 CEST] <iive> h264 for sure have lossless mode [00:02:17 CEST] <iive> x264 too [00:03:55 CEST] <snakeryslug> `ffmpeg -i input.gif -c:v libvpx-vp9 -lossless output.webm` says i need an output file, so lossless must take a parameter. [00:04:05 CEST] <furq> -lossless 1 [00:06:05 CEST] <snakeryslug> thanks. that worked, but the output webm isn't as good as the input gif. [00:12:42 CEST] <c_14> rgb to yuv conversion maybe? [01:31:29 CEST] <durango> Is there any option to cut the sound off once the video/images are done cycling? [01:31:56 CEST] <pzich> -shortest ? [01:34:49 CEST] <durango> perfecto, ty [02:17:24 CEST] <Prelude2004c> hey guys.. good day [02:17:29 CEST] <Prelude2004c> looking for help with " ] Non-monotonous DTS in output stream 0:0; previous: 8585097540, current: 1197762652; changing to 8585097541. This may result in incorrect timestamps in the output file. " [02:17:35 CEST] <Prelude2004c> not sure what to do about it [02:18:03 CEST] <TD-Linux> are you making the input file? [02:18:36 CEST] <Prelude2004c> http://pastebin.com/wCbhxvDc [02:18:43 CEST] <Prelude2004c> no, its a UDP source from satelite [02:19:21 CEST] <Prelude2004c> i have tired to put vsync everywhere i could find to try and fix this.. [02:19:24 CEST] <Prelude2004c> even start_zero i tried [02:33:05 CEST] <TD-Linux> I suspect that's unavoidable with TS, though someone else might know better [03:41:45 CEST] <FinboySlick> Hello people. I've got image buffers in raw formats (mosly RGB) that I would like to put together in a compressed video with libav. I'm trying to figure out two things: 1- Is there a method to describe the data (how many R, G and B bits, the order they're in) and be given a value that libav would understand as a source image format. 2- convert that source image data to a format that the (say x264) encoding functions will be able to deal with. [03:43:28 CEST] <FinboySlick> I'm mostly looking for a general direction to look into here. Google isn't being too helpful. [03:47:26 CEST] <furq> FinboySlick: are you using the api or the command-line [03:47:47 CEST] <FinboySlick> furq: I'm writing C code. So API. [03:48:20 CEST] <furq> well on the plus side you're in the right channel [03:48:31 CEST] <furq> on the down side i've barely used the api [03:48:52 CEST] <FinboySlick> furq: Hehe, I peeked in #devel but managed to read the topic ;) [03:49:05 CEST] <FinboySlick> I mean #ffmpeg-devel [03:49:25 CEST] <furq> well you said libav so i thought you might be using the false ffmpeg [03:49:51 CEST] <FinboySlick> furq: The issue I'm trying to solve isn't really ffmpeg-specific, but I figure there has to be code figuring this stuff up in libav already. [03:50:27 CEST] <furq> afaik if you were using the cli you'd use rawvideo with the appropriate pix_fmt set (ffmpeg -pix_fmts) [03:50:33 CEST] <furq> so i assume it's the same in the cli, just much more verbose [03:50:37 CEST] <furq> the same in the api [03:52:48 CEST] <FinboySlick> furq: My problem is that "pix_fmt" is essentially a number. I still need some sort of get_pix_fmt(num_red_bits, num_green_bits, num_blue_bits, order) that will spew out a pix_fmt value. [03:53:23 CEST] <FinboySlick> Because when I want to encode a frame, I need to specify a pix_fmt number. [03:55:07 CEST] <FinboySlick> Maybe I should look at the x11grab code, it has to figure this out too. [04:54:39 CEST] <FinboySlick> av_pix_fmt_desc_get_id() looks like it might do what I want. [07:39:39 CEST] <jml2> hi [07:39:57 CEST] <jml2> cannot transcode an .ts file to a .wav , anyone want to jab at this one? thanks [07:42:06 CEST] <furq> that's not an error message [07:47:55 CEST] <jml2> thing is i have a problematic hardware encoder --- the options I use for 3 gig files work, but not for 9 gig ones [07:48:15 CEST] <jml2> (options given to ffmpeg) [07:48:50 CEST] <jml2> avconv -analyzeduration 2147483647 -probesize 2147483647 -start_at_zero -ignore_unknown -copy_unknown -copytb 0 -async 1 -avoid_negative_ts 1 -seek_timestamp 1 -hwaccel vdpau -i source.ts -vn -sn -dn -acodec pcm_s24le -b:a 512k -ac 2 out.wav [07:49:23 CEST] <furq> have you tried with ffmpeg [07:49:29 CEST] <jml2> same with ffmpeg [07:50:02 CEST] Action: jml2 (http://pastebin.com/5cyuLKCx) [07:51:29 CEST] <furq> can you extract the aac track without transcoding [07:51:47 CEST] <jml2> (ffmeg - http://pastebin.com/HdpmRd64 ) [07:52:54 CEST] <jml2> same [07:53:04 CEST] <jml2> ffmpeg -analyzeduration 2147483647 -probesize 2147483647 -hwaccel vdpau -i source.ts -vn -acodec alac -bsf:a aac_adtstoasc -ab 384k -ar 48000 sound.m4 [07:53:13 CEST] <jml2> well that's alac, [07:53:14 CEST] <furq> i said without transcoding [07:53:16 CEST] <furq> -c:a copy [07:53:31 CEST] <furq> that might get around the need for analyzeduration and probesize [07:53:57 CEST] <jml2> i can take out video.. but not audio [07:54:08 CEST] <jml2> i never tried 5+ gig.. I know under 5 i have no issue [07:54:25 CEST] <jml2> i havent looked at hbrake and vlc for this 9gig yet [07:55:23 CEST] <furq> also analyzeduration isn't a uint32 in recent versions of ffmpeg [07:55:29 CEST] <furq> try setting that to 9000M [07:55:31 CEST] <jml2> so far vlc is extracting it, awaiting its output [07:55:47 CEST] <furq> looks like probesize isn't either [07:55:48 CEST] <jml2> furq, how do i use its max? [07:55:58 CEST] <jml2> furq, there's an exposed range showing e^ [07:56:30 CEST] <jml2> ( Value -1.000000 for parameter 'analyzeduration' out of range [0 - 9.22337e+18] ) [07:56:36 CEST] <jml2> can I use the max? [07:56:48 CEST] <furq> no idea, i've never needed to set it [07:56:53 CEST] <furq> if the file is 9GB then 9000M should work fine [07:57:21 CEST] <jml2> the .wav output is 16 mb.. [07:57:23 CEST] <jml2> (vlc-out.wav) [07:57:38 CEST] <jml2> it's able to extract audio.. [07:57:44 CEST] <jml2> so i know its do-able [07:59:16 CEST] <jml2> its 161mb, and so far its audio works --- (vlc-out.wav) [07:59:29 CEST] <jml2> looks like that 9000M for ffmpeg-out.wav might do the trick.. [14:08:03 CEST] <TheGreatDoc> Hi [14:09:00 CEST] Last message repeated 1 time(s). [14:09:37 CEST] <TheGreatDoc> Im having a little issue with video/audio desycn doing a streaming. The source is a DeckLink SDI 4K [14:09:53 CEST] <TheGreatDoc> Over time, audio start desyncing with video [14:14:27 CEST] <CruX|_> hi, can I convert flac -> ape with ffmpeg ? [14:14:36 CEST] <furq> no [14:14:42 CEST] <furq> ape is decode only [14:14:53 CEST] <CruX|_> if not which is the best loosles codec that can use ffmpeg ? [14:14:55 CEST] <CruX|_> flac ? [14:15:11 CEST] <furq> flac is what i use [14:15:30 CEST] <CruX|_> ok thx [14:16:59 CEST] <furq> CruX|_: ffmpeg -codecs | grep \ .EA..S [14:17:05 CEST] <furq> will list all lossless audio encoders [14:21:00 CEST] <CruX|_> great [14:21:05 CEST] <CruX|_> for i in *; do ffmpeg -y -i "$i" -vn -acodec wavpack "$i.wavpack"; done [14:21:16 CEST] <CruX|_> btw what's wrong in this command ? [14:22:37 CEST] <furq> looks fine to me other than the fact that wavpack is normally .wv [14:22:51 CEST] <furq> and also you're using wavpack in 2016 [14:23:34 CEST] <CruX|_> I want to test all loosless formats [14:23:48 CEST] <furq> https://xiph.org/flac/comparison.html [14:23:52 CEST] <furq> there's a nice comparison there [14:31:17 CEST] <CruX|_> furq: nice thanks [15:56:18 CEST] <dongs> is there some ffmpeg default filter that can do 3d transform to video [15:56:29 CEST] <dongs> like, tilt/rotate it slightly [15:57:04 CEST] <dongs> http://i.ytimg.com/vi/uVPNjEFsuQs/maxresdefault.jpg something like this etc [16:13:19 CEST] <durandal_170> perspective [16:13:55 CEST] <dongs> 3d warp / perspective yeah. [16:14:00 CEST] <durandal_170> and remap if you have maps... [16:14:13 CEST] <dongs> perspective is a filter? [16:14:29 CEST] <dongs> nice [18:08:18 CEST] <evan_> Hi all, is it possible to use ffmpeg to re-encode and stream a video file on the fly from a VPS server to a pc that is behind nat (adsl) if I'm not able to open the ports on my adsl router .. ? [23:35:06 CEST] <__raven> hi [23:35:52 CEST] <__raven> how to concatenate several .mov chunks (same type) into one video file ("the right way")? ^^ [23:36:04 CEST] <__raven> i just get "invalid data found when processing input" at the moment [23:36:33 CEST] <__raven> could that be caused since concat just works for containerless files? [23:37:30 CEST] <BtbN> remux them all to ts or mkv, and concat those. [23:38:07 CEST] <__raven> BtbN: pipe possible this way? [23:38:15 CEST] <BtbN> what? [23:38:56 CEST] <__raven> doing that in one step: ffmpeg remux | ffmpeg concat.mkv? [23:39:36 CEST] <BtbN> how would you pipe more then one file? oO [23:39:52 CEST] <BtbN> and if you only have one file, what exactly do you want to concat? [23:49:14 CEST] <kepstin> for several 'mov' files which all contain the same audio/video formats, you'd probably want to use the "concat demuxer" https://www.ffmpeg.org/ffmpeg-formats.html#concat-1 [23:49:56 CEST] <BtbN> it doesn't work exactly well for mp4 [23:50:09 CEST] <kepstin> concat demuxer should work perfectly fine in mp4 [23:50:20 CEST] <JEEB> really? [23:50:25 CEST] <kepstin> it reads each file individually with a proper format-aware demuxer [23:50:31 CEST] <kepstin> don't get it confused with the concat protocol [23:50:35 CEST] <JEEB> right [23:50:45 CEST] <JEEB> protocol, demuxer and filter :P [23:51:02 CEST] <JEEB> I mostly used the filter because it seemed like the thing that might fuck up least [23:51:06 CEST] <BtbN> I had even the demuxer screw up on mp4 files. remuxing to ts fixed it [23:51:31 CEST] <kepstin> well, filter will always work, but requires re-encoding. [23:52:03 CEST] <BtbN> filter also isn't exactly nice when you want to concat several 100 files [23:52:18 CEST] <JEEB> yeah, thankfully for my use case that was a requirement and I think I supported up to 5 or so files [00:00:00 CEST] --- Sun Apr 17 2016

1 0

ffmpeg-devel.log.20160415
by burek 16 Apr '16

16 Apr '16

[00:02:45 CEST] <durandal_1707> mgraczyk: channel layout structs are very limited... [00:03:16 CEST] <durandal_1707> the current ones... [00:06:49 CEST] <wm4_> mgraczyk: what us ambisonic? [00:06:58 CEST] <wm4_> object audio? [00:07:55 CEST] <nevcairiel> yes [00:08:07 CEST] <nevcairiel> but i dont believe it would fit into our channel layouts [00:08:18 CEST] <nevcairiel> any user code would have to special case it since its not actually speakers [00:10:21 CEST] <wm4_> yeah, something else would be needed [00:10:42 CEST] <wm4_> AFAIK ffmpeg can use channel count without channel layout (while libav can't) [00:11:03 CEST] <nevcairiel> we can, fate even tests it [00:11:20 CEST] <nevcairiel> (i never understood why libav forces a layout) [00:13:18 CEST] <wm4_> normally you know the layout, and then the channel count is redundant [00:14:45 CEST] <nevcairiel> but when you dont, i would prefer avcodec not inventing fake information [00:16:25 CEST] <wm4_> true [00:16:48 CEST] <wm4_> elenril had a WIP patch to extend channel layouts to include various fucked up cases, but I guess it'll never happen [00:17:07 CEST] <nevcairiel> you mean the complex struct? [00:17:27 CEST] <wm4_> yes [00:20:25 CEST] <mgraczyk> It's not quite object audio [00:20:54 CEST] <mgraczyk> With Ambisonics, the channels have specific meaning as they do in 5.1 or stereo setups, for example. [00:21:21 CEST] <mgraczyk> It's just that the channels do not correspond to physical speakers, but rather to a direction/shape [00:22:21 CEST] <mgraczyk> I was not able to encode with Opus without a channel layout being imposed. Maybe I missed an option? For example, how can I encode a 16 channel WAV file with libopusenc in such a way that the result is 16 channels? [00:22:37 CEST] <wm4_> there's currently absolutely no way in ffmpeg to represent this [00:22:54 CEST] <wm4_> hm good question [00:23:31 CEST] <nevcairiel> dont think you can [00:23:45 CEST] <wm4_> should be easy with the API (except using the API is not easy), as long as the encoder doesn't require a channel layout [00:24:07 CEST] <nevcairiel> our encode API enforces a channel layout requirement [00:24:12 CEST] <mgraczyk> At the very least, I would like to be able to tell ffmpeg "The channel structure is unknown to you, so tell the codec that there is no channel layout" [00:24:30 CEST] <wm4_> too bad then [00:24:42 CEST] <mgraczyk> I think then we could add another option that means "this input is ambisonics" [00:25:19 CEST] <wm4_> or just something to disable this check [00:26:30 CEST] <mgraczyk> Yeah to be specific, libopusenc.c:445 specifies that Opus uses a channel_layout in ff_vorbis_channel_layouts [00:26:39 CEST] <mgraczyk> But there is no "none" option because these are bitvectors [00:26:57 CEST] <mgraczyk> I think we could add the possibility for "none" which would mean that the channels have no special meaning [00:27:18 CEST] <nevcairiel> none would be zero, which is already the terminator of these lists, so thats not possible [00:27:32 CEST] <nevcairiel> if a encoder supports any channel layout, it could just not provide a channel layout listing [00:28:08 CEST] <mgraczyk> but the same number of input channels can have multiple meanings, so with no channel layout listing it would not be possible to determine which way to interpret the input channels. [00:28:33 CEST] <mgraczyk> eg, quadrophonic vs. R+L+C+LFE [00:28:38 CEST] <nevcairiel> using a special value for channel_layout is also likely to trip all sorts of things since many things may rely on the bitcount of the layout to equal the channel count [00:29:33 CEST] <nevcairiel> the encoder not offering a list of supported layouts doesnt stop the user from providing one =p [00:30:51 CEST] <nevcairiel> the only way i can think of would be adding a codec capability that it accepts a unknown channel layout [00:31:07 CEST] <nevcairiel> hm looking at the code [00:31:13 CEST] <wm4_> it could use the 30 or so high bits to represent "unknown" channel layouts [00:31:16 CEST] <nevcairiel> you should be able to just pass avctx->channel_layout = 0 [00:31:29 CEST] <nevcairiel> it will warn, but not fail [00:32:17 CEST] <nevcairiel> it only fails if you try to specify a layout and it doesnt accept it [00:32:42 CEST] <mgraczyk> Just to play around with that, is there any way from the command line to set channel_layout to zero? [00:33:07 CEST] <nevcairiel> -channel_layout 0? [00:33:30 CEST] <mgraczyk> Unknown channel layout: 0 [00:34:19 CEST] <mgraczyk> We also need to change a few things on and after libopusenc.c:167 to address the FIXME and prevent accessing those arrays when channels > 8 [00:35:02 CEST] <nevcairiel> guess ffmpeg.c prevents that, but thats where i dont care anymore =p [00:35:06 CEST] <nevcairiel> avcodec accepts 0 [00:35:53 CEST] <cone-544> ffmpeg 03Michael Niedermayer 07master:21acc4db5fa3: avcodec: Add bits_per_raw_sample to AVCodecParameters [00:36:45 CEST] <wm4_> mgraczyk: extending the max. channel count beyond 8 would be pretty orthogonal I guess [00:37:22 CEST] <mgraczyk> It looks like the parsing is done in libavutil/channel_layout.c, not ffmpeg.c, right? [00:37:49 CEST] <mgraczyk> Why orthogonal? [00:38:16 CEST] <mgraczyk> It seems that if we were to allow codecs to accept "unknown" or "don't care" channel layouts, the channel count wouldn't matter anymore, right? [00:38:32 CEST] <mgraczyk> We shouldn't need "Unknown but 4 channels" in addition to "Unknown but 6 channels" [00:38:52 CEST] <mgraczyk> I think if we add "unknown layout" it should be any number of channels, or a number up to a max. [00:40:05 CEST] <wm4_> I mean, there can be classic channel layouts with more than 8 channels (although I've never seen one) [00:40:47 CEST] <mgraczyk> Sure, and Opus doesn't have any special ones so for Opus in particular > 8 would mean "unknown" for now. [00:43:11 CEST] <mgraczyk> What about AV_CH_LAYOUT_NATIVE? I'm not sure I understand the meaning of that one, but maybe we could just add that to ff_vorbis_channel_layouts? [00:43:36 CEST] <mgraczyk> When passed to Opus, it would mean to do whatever the right thing is for the number of channels passed in. So for > 8, it would mean no layout (each stream coded independently). [00:43:42 CEST] <nevcairiel> like I said, as soon as a channel layout is set, its required for the bitcount to match the number of channels [00:44:01 CEST] <nevcairiel> adding odd special cases is not wise [00:44:04 CEST] <nevcairiel> rather go with unset then [00:44:44 CEST] <mgraczyk> Well it's an odd special case that is already there. [00:45:01 CEST] <nevcairiel> its not used for channel_layout though, only for request_channel_layout, it says as much in the docs =p [00:45:02 CEST] <mgraczyk> If that were passed to Opus, what would it mean? [00:45:07 CEST] <mgraczyk> Ah okay [00:49:22 CEST] <mgraczyk> Okay, let's start from basics then. The natural non-special case thing to do would be to assign a specific position in each bit vector to each ambisonic channel. [00:49:47 CEST] <mgraczyk> Just as is done for "FRONT_LEFT", there would be an "AMBISONIC_L0_M1" channel [00:49:50 CEST] <mgraczyk> for example [00:50:17 CEST] <mgraczyk> The problem is that there would be at least 16 of these, using valuable bits in this bitvector in a somewhat wasteful way. [00:50:24 CEST] <nevcairiel> except that there can be hundreds of those and we really dont want to fill our already half-full 64-bit bitmask with such bits [00:50:29 CEST] <mgraczyk> Yeah [00:50:47 CEST] <mgraczyk> So maybe there is a way we could easily expand the size to a multibyte encoding, unicode style? [00:51:20 CEST] <mgraczyk> Reserve a bit to mean "also look at this longer vector", but do so in such a way that existing codecs can completely ignore the bit and be okay. [00:51:24 CEST] <nevcairiel> not really, i would just go with unset layout, like I said before, avcodec can deal with unset just fine [00:52:10 CEST] <mgraczyk> Okay, so you are suggesting modifying av_get_channel_layout in libavutil/channel_layout.c so that it can parse "none" or "0" or something like that? [00:54:13 CEST] <nevcairiel> thats just a silly ffmpeg.c limitation of not supporting numeric channel layouts [00:54:57 CEST] <nevcairiel> and no, that change wouldnt help since ffmpeg.c specifically checks if its non-null [00:55:06 CEST] <nevcairiel> or rather ffmpeg_opt.c [00:55:31 CEST] <mgraczyk> Yeah but that would be easy to change, right? [00:55:39 CEST] <nevcairiel> you would think so, but not really [00:55:43 CEST] <mgraczyk> Just return a status flag and take an [out] arg that has the actually channel layout [00:55:56 CEST] <nevcairiel> thats new public API [00:55:57 CEST] <nevcairiel> which is evil [00:56:16 CEST] <mgraczyk> Why? [00:56:24 CEST] <mgraczyk> You can add a new function and not change the old one? [00:56:27 CEST] <mgraczyk> right? [00:56:49 CEST] <nevcairiel> Of course, but before we add new public functions you need to convince us that its really super useful to have it [00:57:00 CEST] <nevcairiel> because once we do have it, we cant get rid of it anymore [00:57:33 CEST] <mgraczyk> Okay, if you don't want to modify or augment the API, how can I get this available to the command line? [00:58:07 CEST] <mgraczyk> Would it be better to special case certain ambisonics? For instance, I could add a "first order ambisonics" channel layout with 4 channels. [01:00:46 CEST] <nevcairiel> Like I explained earlier, you can't do that, channel layouts need to have 4 bits set if they represent 4 channel, or 16 bits set for 16 channel [01:02:11 CEST] <mgraczyk> Right, I forgot that. Any suggestions then? [01:02:29 CEST] <nevcairiel> you can of course pass it using existing channel layouts to opus, and add an opus option to inform it that its ambisonic [01:03:29 CEST] <mgraczyk> Okay, so we would change libopusenc to forgo indexing into those arrays whenever the channel count is too high or this new option is provided or something like that? [01:04:04 CEST] <nevcairiel> it sounded like opus cant even do this right now anyway, so maybe you should finish making opus support this first :) [01:04:33 CEST] <mgraczyk> Opus can independently code any number of channels <255 [01:04:54 CEST] <mgraczyk> It can also couple any of those 127 pairs [01:05:01 CEST] <nevcairiel> sure, but you still need to signal somehow that the decoder has to interpret it in a certain way [01:05:10 CEST] <nevcairiel> just giving it 16 channel doesnt inform it of anything [01:05:47 CEST] <mgraczyk> Yes that is true. [01:06:02 CEST] <mgraczyk> However, encoding 16 channels independently is better than not being able to encode 16 channels at once. [01:06:03 CEST] <nevcairiel> otherwise it sounds like a rather hacky "proprietary" solution where you have to know what the file contains to interpret it, and possibly not something we should claim ffmpeg can create =p [01:06:38 CEST] <mgraczyk> Nothing here will be proprietary, I promise! [01:07:21 CEST] <nevcairiel> i quoted it for a reason just meaning that if the file doesnt contain all information needed to fully decode and interpet it, its not going to make many people happy [01:07:57 CEST] <mgraczyk> Yes that is true. The Opus bitstream leaves a lot of information implicit, so you can do a lot of special case compression by only modifying the encoder. [01:08:00 CEST] <nevcairiel> and coming up with your own "specification" is generally not a good idea [01:08:17 CEST] <nevcairiel> did you try to talk to Xiph about ambisonic? [01:08:24 CEST] <mgraczyk> Yes [01:08:29 CEST] <mgraczyk> We are in communication [01:08:56 CEST] <mgraczyk> For now we are doing as much as we can without modifying the bitstream or the decoder. [01:09:08 CEST] <mgraczyk> So I think changes to ffmpeg will only need to be configuration changes. [01:09:28 CEST] <mgraczyk> Or rather, adding ways to pass configuration to the codec. [01:12:35 CEST] <wm4_> well its just a question of metadata isn't it [01:12:58 CEST] <wm4_> so in place of whatever opus/ogg uses for channel layouts there should be some for ambisonic [01:13:33 CEST] <nevcairiel> maybe ffmpeg -ac 16 works? I don't know what it does with the channel layout then [01:13:35 CEST] <wm4_> it was quite horrible when flac (and maybe vorbis?) didn't have anything standard even for classic channel layouts [01:13:41 CEST] <wm4_> don't repeat that mistake [01:17:32 CEST] <nevcairiel> we also have various channel layouts for 4, 8, 16 channels, i dont know what channel counts ambisonic would really need, but quad, octagonal and hexadecagonal sound generic enough to use for ambisonic in combiation with a opus private option [01:19:08 CEST] <mgraczyk> Yes that should be okay for the short term. [01:19:25 CEST] <mgraczyk> We would also like to encode 25, 36, 49, etc channel streams [01:19:50 CEST] <mgraczyk> Probably not for a year or so though [01:20:26 CEST] <mgraczyk> Okay so I will work on a patch which modified libopusenc to accept 16 channels and a special ambisonic option. [01:20:37 CEST] <nevcairiel> before thinking about actually encoding things, it would be nice to have an official xiph approved specification how to signal this in a opus stream, anything else is just super annoying in the long run, as wm4_ explained we had this with flac for a while because they didnt even specifiy ordinary 7 and 8 channel layouts [01:21:00 CEST] <mgraczyk> I do not plan to modify the bitstream right now. [01:21:18 CEST] <nevcairiel> so "inofficial" specifications showed up using tags, which was just stupid [01:21:28 CEST] <mgraczyk> The header will contain a new channel_mapping value [01:22:54 CEST] <mgraczyk> libopusenc.c:96 will change, specifically [01:23:08 CEST] <mgraczyk> Right now it is incorrect anyway [01:23:22 CEST] <mgraczyk> It should be 0 for coupled stereo [01:23:52 CEST] <nevcairiel> its only written for more than stereo [01:23:56 CEST] <mgraczyk> In fact, for now 255 for >8 channels is totally fine with me. [01:24:01 CEST] <mgraczyk> Ah yes you're right [01:24:20 CEST] <nevcairiel> the question is not what is fine with you, but what lets decoders identify such files [01:24:24 CEST] <nevcairiel> because without it, its just a hackjob [01:24:59 CEST] <mgraczyk> Well the decoder doesn't care whether it is uncoupled or ambisonic, because the decoder doesn't know what ambisonics are [01:25:13 CEST] <mgraczyk> So I think the best thing to do would be to use "255" rather than a new channel mapping [01:25:20 CEST] <mgraczyk> Until the decoder is aware of ambisonics [01:25:45 CEST] <nevcairiel> so i give you a random opus file with 4 audio channels, how do you know if its just 4 random channel or ambisonics? [01:25:46 CEST] <nevcairiel> you dont [01:25:49 CEST] <nevcairiel> and thats a huge problem [01:26:08 CEST] <mgraczyk> yes, there will be a new channel mapping for that [01:26:17 CEST] <mgraczyk> But as of right now, the problem is unsolved as well. [01:26:26 CEST] <mgraczyk> I'm not proposing anything that would make the current state worse [01:26:42 CEST] <nevcairiel> if ffmpeg can start creating such files, its worse! =P [01:27:07 CEST] <mgraczyk> It's much worse if ffmpeg can't create such files though. [01:27:21 CEST] <mgraczyk> Because we need to be able to create them. [01:27:28 CEST] <mgraczyk> I agree, we should do this the right way. [01:27:39 CEST] <nevcairiel> then do so [01:27:54 CEST] <nevcairiel> and not a half-finished solution because it was easy :) [01:28:00 CEST] <nevcairiel> such files will haunt us for years [01:28:15 CEST] <nevcairiel> encoding and muxing files should be extremely strict to avoid creating a mess for the future [01:28:19 CEST] <mgraczyk> It's not about easy or difficult, it's about how long it will take to get a Xiph stamp on a channel mapping number. [01:28:41 CEST] <mgraczyk> I don't want to take shortcuts, but I'd prefer to get something working before it is perfect. [01:28:59 CEST] <mgraczyk> So I'm going to work on a patch and I think your guidance has helped me figure out what that ought to be [01:29:30 CEST] <mgraczyk> You don't have to accept it until everything is cohesive [01:31:08 CEST] <nevcairiel> at least opus already implements a way to define different channel mapping strategies, so a bitstream change as such is not needed, only a new defined value once the definition of it can be worked out [01:31:25 CEST] <nevcairiel> most other formats do not [01:32:39 CEST] <mgraczyk> Yes that is fortunate. I am going to propose to them that we call "mapping_family == 2" (L96) mean ambisonics and that we let channel_mapping (L99) determine the meaning of each ambisonic channel [01:32:45 CEST] <mgraczyk> Does that sound reasonable? [05:01:33 CEST] <cone-083> ffmpeg 03James Almer 07master:bb505cd5057c: avformat/hashenc: simplify hash_write_trailer [12:17:05 CEST] <cone-141> ffmpeg 03Benjamin Steffes 07master:0cfe16437fce: lavfi/af_hdcd: Implement high definition audio cd filtering. [12:19:33 CEST] <atomnuker> finally! [12:20:05 CEST] <atomnuker> now if only there was an hdcd detection filter [12:21:28 CEST] <rcombs> nice tables [12:23:27 CEST] <wm4> holy shit half of the file is tables [12:24:13 CEST] <rcombs> probably more byte-wise [12:24:50 CEST] <wm4> will surely add a few KBs of binary size [12:25:03 CEST] <rcombs> yum [14:47:25 CEST] <Daemon404> http://git.videolan.org/?p=ffmpeg.git;a=commitdiff;h=0cfe16437fce4c2090d395… [14:47:29 CEST] <Daemon404> why the heck is this in lavfi? [14:48:00 CEST] <wm4> because someone wrote a patch [14:50:37 CEST] <ubitux> Daemon404: where would you have it? [14:51:27 CEST] <RiCON> as a bsf, maybe? [14:52:22 CEST] <Daemon404> ubitux, see my mail [14:52:28 CEST] <Daemon404> it's literally documented as a decoder. [14:53:05 CEST] <ubitux> so as a pcm decoder? [14:53:11 CEST] <ubitux> sounds limited [14:53:13 CEST] <Daemon404> we have one for e.g. bluray pcm [14:53:36 CEST] <Daemon404> er sorry, dvd pcm [14:53:55 CEST] <ubitux> ok, well no opinion really [15:05:07 CEST] <wm4> ubitux: sent a patch for the subtitle thing [15:05:18 CEST] <ubitux> i think i answered [15:05:34 CEST] <ubitux> ml is being slow to accept my mails again [15:05:54 CEST] <wm4> seems so [15:06:08 CEST] <cone-889> ffmpeg 03Michael Niedermayer 07master:a7fa1e38ca76: avcodec/utils: Fix duration of ATRAC3 packets [15:07:10 CEST] <nevcairiel> Daemon404: except you could potentially have hdcd content in any other lossless container since its basically just pcm data, having it as a filter seems most logical the way this particular format works [15:07:24 CEST] <nevcairiel> also its been on the ML for weeks, you are late 0p [15:11:37 CEST] <Daemon404> nevcairiel, i said i was late ;p [15:12:09 CEST] <Daemon404> BBB, someoe mentioned you could port vp9_superframe to the new bsf api [15:12:12 CEST] <Daemon404> still feel like it? [15:12:20 CEST] <BBB> sure [15:12:24 CEST] <BBB> what do I need to do? [15:13:06 CEST] <Daemon404> WIP merge is here: https://github.com/dwbuiten/FFmpeg/commits/bsf [15:13:33 CEST] <Daemon404> https://github.com/dwbuiten/FFmpeg/commit/33d18982fa03feb061c8f744a4f0a9175… [15:13:37 CEST] <Daemon404> for example conversions [15:15:15 CEST] <Daemon404> maybe look at a simple one like chomp to see how it sdone [15:15:45 CEST] <nevcairiel> the most important thing that changes is how input and output of packets works [15:17:02 CEST] <nevcairiel> so you can more cleanly do m:n scenarios in a bsf, which should help your filter [15:19:49 CEST] <Daemon404> nevcairiel, i fixed the hevc bsf today... seems i can simply remove that crappy nablet hack [15:19:52 CEST] <Daemon404> it doesnt do anything anymore [15:20:07 CEST] <Daemon404> afaict [15:20:36 CEST] <nevcairiel> the problem it "fixed" before was that executing it twice would try to convert the extradata twice, which resulted in failure the second time around since it wrote the new one to the avctx [15:20:53 CEST] <ubitux> michaelni: frame_bytes / ba>0 ? frame_bytes / ba : 1 [15:20:57 CEST] <ubitux> can you please add some braces? [15:20:59 CEST] <Daemon404> nevcairiel, and i dont think that happens anymore [15:21:13 CEST] <Daemon404> unless i am mistaken? [15:21:27 CEST] <nevcairiel> from what I can tell the new bsf api has an input and output codecpar [15:21:33 CEST] <nevcairiel> how those are used is another question then [15:21:45 CEST] <ubitux> michaelni: your spacing suggests ba>0 has the highest priority, which means you could /0 [15:22:01 CEST] <nevcairiel> if the output is then used to overwrite the input codecpar at some point, the same problem would happen [15:22:35 CEST] <nevcairiel> no clue what the fallback code from the old API does [15:23:33 CEST] <Daemon404> nevcairiel, im trying to figure out how the hell youve even manage to trigger this in correct code [15:23:46 CEST] <Daemon404> you'd* [15:24:38 CEST] <nevcairiel> use ffmpeg and specify h264/hevc_qsv as the decoder, it will then be used both during probing and during decoding, but during probing it would already have converted the extradata [15:24:45 CEST] <nevcairiel> its the problem when a bsf is used from within a decoder =p [15:25:31 CEST] <Daemon404> im not sure where to even look to find out [15:25:35 CEST] <Daemon404> this is so convoluted [15:25:38 CEST] <nevcairiel> anyway all usage of the bsfs within avcodec and avformat should be converted to the new API after its available and this can then be avoided, hopefully [15:26:12 CEST] <Daemon404> welli cant tell if youre saying i should remove the code or not [15:26:15 CEST] <Daemon404> ;) [15:26:28 CEST] <nevcairiel> you should ensure it still works [15:26:34 CEST] <nevcairiel> if the code is needed or not i dont know [15:26:45 CEST] <Daemon404> and i dont know how the hell to test if it works [15:26:49 CEST] <Daemon404> i dont have any qsv stuff [15:28:52 CEST] <Daemon404> so uh? [15:29:33 CEST] <nevcairiel> qsv has a software decoder plugin you can use =p [15:29:45 CEST] <Daemon404> sigh [15:30:23 CEST] <Daemon404> lemme guess [15:30:29 CEST] <Daemon404> this is going to be a big pain in the ass to setup [15:30:43 CEST] <nevcairiel> dunno [15:31:08 CEST] <nevcairiel> should just be grabbing the plugin binary and the dispatcher library and headers, the second is available on github from luca [15:31:38 CEST] <Daemon404> its really hard to motivate myself to do this when i dont give a shit avbout qsv or nablet hacks [15:32:45 CEST] <Daemon404> nevcairiel, also does this appyl to hevc [15:32:46 CEST] <wm4> "nablet hacks" never caught my eye before, but maybe I was looking away [15:32:57 CEST] <Daemon404> does qsv have a software hevc decoder [15:33:00 CEST] <Daemon404> if not, i cant test this [15:33:08 CEST] <nevcairiel> well h264 and hevc are exactly the same in this regard [15:33:13 CEST] <nevcairiel> fix it in h264 and do the same for hevc [15:33:19 CEST] <Daemon404> i dont really want to fix h264 [15:33:22 CEST] <nevcairiel> the qsv decoder code is even shared [15:33:23 CEST] <Daemon404> outs is not easy to merge [15:33:28 CEST] <Daemon404> i was hoping someone lse would. [15:33:30 CEST] <Daemon404> else* [15:33:32 CEST] <Daemon404> i'd probably scerw it up [15:34:22 CEST] <Daemon404> im probably not even going to. someone else can help me or it wont get done. [15:34:25 CEST] <Daemon404> simple. [15:38:43 CEST] <Daemon404> i pushed the resolved hevc conflicts, without the sps option in it, to my bsf branch. someone feel free to test, or try and fix the h264 bsf. [15:38:48 CEST] Action: Daemon404 is not going to [15:46:26 CEST] <durandal_1707> SPAM, BAN HIM [15:49:13 CEST] <cone-889> ffmpeg 03Michael Niedermayer 07master:bcf936eaeae8: avcodec/utils: Add braces to framecount computation [15:57:11 CEST] <wm4> possibly I can take a look at the BSF shit later [15:58:12 CEST] <ubitux> michaelni: thx [18:24:16 CEST] <omerjerk> is there a utility function in ffmpeg to create a float value out of sign, exponent and mantissa ? [18:25:24 CEST] <durandal_1707> no, but that is trivial in C [18:26:15 CEST] <durandal_1707> use union [18:27:55 CEST] <omerjerk> I'm doing it like - sign << 31 | exp << 23 | mantissa [18:28:00 CEST] <omerjerk> I hope this is correct ? [18:28:24 CEST] <omerjerk> I then convert the integer to float using av_int2float [18:29:45 CEST] <nevcairiel> manually building a float sure sounds wrong [18:32:26 CEST] <omerjerk> I have written the code up to the point where I've sign bit, exponent, and mantissa. [18:32:39 CEST] <omerjerk> but the final float is coming out to be wrong. [18:36:20 CEST] <durandal_1707> omerjerk: use casting [18:39:55 CEST] <omerjerk> like float a = (float) some_int; ? [18:41:06 CEST] <Daemon404> just what exactly are you doing that requires manually building a float [18:43:54 CEST] <omerjerk> I'm working on implementing the floating point format support in the ALS decoder that we have. [18:46:03 CEST] <durandal_1707> omerjerk: just use different pointer to frame extended data [18:47:14 CEST] <Daemon404> omerjerk, is it float for intermediate only, or for final output? [18:48:04 CEST] <omerjerk> both kind of. [18:48:32 CEST] <omerjerk> I create a float variable, assign to an array, then write the array to the output. [18:49:53 CEST] <omerjerk> I'm creating the float like this - https://github.com/omerjerk/FFmpeg/blob/float/libavcodec/alsdec.c#L1761 [18:50:12 CEST] <durandal_1707> you dont need to create float variable [18:51:56 CEST] <nevcairiel> if anything you should just use the mantissa and multiply it with the exponent appropriately to create the value, bitshifting to manually create the memory representation of a float seems terrible [18:53:49 CEST] <omerjerk> okay. Thank you everyone. I can try this. [18:57:18 CEST] <durandal_1707> this must bit bitexact [18:57:35 CEST] <durandal_1707> so just write them as ints [19:06:24 CEST] <cone-889> ffmpeg 03Michael Niedermayer 07master:f07923253a1b: avcodec/avcodec: Document signed/unsignedness of sample formats in relation to bits_per_raw_sample [19:09:17 CEST] <kurosu> omerjerk, your sign is an unsigned char, shifting it left by 31 is likely not going to work [19:10:34 CEST] <kurosu> (and e is an int but I haven't checked if that might cause issues) [19:12:37 CEST] <omerjerk> oops [19:12:56 CEST] <omerjerk> thanks for pointing it out. [19:14:01 CEST] <kurosu> also, it would be prettier if you declared, when possible, variables in the codeblock where they are used [19:14:59 CEST] <kurosu> *at the start of the codeblock [20:56:49 CEST] <omerjerk> I'm almost done with the my patch now. :) [20:57:01 CEST] <omerjerk> I'm properly able to decode the float values and print to the console. [20:57:10 CEST] <omerjerk> Just had some doubt with writing to the file [20:57:21 CEST] <omerjerk> Here's how I'm doing it currently - https://github.com/omerjerk/FFmpeg/blob/float/libavcodec/alsdec.c#L2004 [20:57:24 CEST] <omerjerk> is this okay ? [20:57:47 CEST] <omerjerk> because the output file is almost half the size of what it should actually be. [21:05:12 CEST] <BBB> I think I found an actual bug [21:05:45 CEST] <BBB> I havent bisected exactly, but a h264 raw file (annexb) used to be detected as 30fps, 30tbr, and is (as of recently) being detected as 25fps,30tbr [21:05:50 CEST] <BBB> which causes all kind of weird issues [21:05:56 CEST] <BBB> is this a known regression recently? [21:07:00 CEST] <wm4> is it due to codecpar? [21:07:50 CEST] <cone-889> ffmpeg 03Carl Eugen Hoyos 07master:139cbeb75e0f: lavc/dds: Fix GRAY8A decoding. [21:09:38 CEST] <durandal_1707> omerjerk: how you decode file? [21:10:06 CEST] <durandal_1707> by default codec for wav is 16bit pcm [21:10:53 CEST] <omerjerk> but the float data us 32 bit [21:10:57 CEST] <omerjerk> *is [21:12:03 CEST] <omerjerk> I've the final values inside raw_float[][] array. [21:13:13 CEST] <jamrial> BBB: probably codecpar related. maybe it used to read that info from the stream but now from the demuxer only, which seems to be hardcoded as 25 for all raw demuxers [21:13:29 CEST] <BBB> that should probably be fixed [21:13:42 CEST] <jamrial> yeah [21:13:45 CEST] <jamrial> try using -framerate 30 for the input file as workaround for now [21:17:00 CEST] <omerjerk> durandal_1707: I tried copying the bits from float to int and use the old code, even then it didn't work. (there's code written for 32 bit integer already) [21:17:10 CEST] <BBB> jamrial: should I file a bug? [21:17:19 CEST] <jamrial> yeah [21:18:26 CEST] <durandal_1707> omerjerk: hard to follow you... [21:20:11 CEST] <omerjerk> oh. sorry for that. Right now, I just want to write 32-bit integer to the output file. [21:23:51 CEST] <cone-889> ffmpeg 03Carl Eugen Hoyos 07master:6d39132fd4af: lavf/bink: Support Monkey Island 4 (SMUSH) files. [21:26:47 CEST] <cone-889> ffmpeg 03Carl Eugen Hoyos 07master:a676c29af5ee: lavf/bink: Cosmetics: Reindent after last commit. [21:35:39 CEST] <BBB> jamrial: ^^ [22:20:47 CEST] <Threads> can i ask when compiling ffmpeg where does it store "Lavf56.30.100" this written application data in the source files [22:25:26 CEST] <Shiz> why? [22:27:49 CEST] <Daemon404> he probably wants to remove the muxer info from files he creates. [22:32:11 CEST] <Threads> well remove or atleast change it in some way so it looks abit nicer even just putting a space between lavf 56* [22:32:41 CEST] <Threads> if its a subject that can't be talked about i understand [22:33:48 CEST] <Shiz> it's pretty easy to find in an obvious place anyway [23:07:29 CEST] <rcombs> Daemon404: nevcairiel: how far out do you suppose the new BSF API is? [23:24:19 CEST] <rcombs> also, is there a new _parser_ API on the way? [23:27:35 CEST] <wm4> no [23:28:16 CEST] <rcombs> alright, won't worry about those then [00:00:00 CEST] --- Sat Apr 16 2016

1 0

ffmpeg.log.20160415
by burek 16 Apr '16

16 Apr '16

[03:31:21 CEST] <k_sze[work]> What are the relative merits between compiling FFmpeg on Windows with MSVC vs with MinGW? [03:36:15 CEST] <furq> one of them doesn't require you to have MSVC installed [03:53:14 CEST] <k_sze[work]> And I'm a bit confused about yasm and SDL. [03:53:48 CEST] <k_sze[work]> Here it says to download yasm from a web site: https://trac.ffmpeg.org/wiki/CompilationGuide/MinGW#yasm [03:54:34 CEST] <k_sze[work]> But here I'm told to install something related to yasm and SDL using pacman. [03:54:46 CEST] <k_sze[work]> Does the pacman command already cover what I need? [03:56:16 CEST] <furq> if you're using msys2 then sure [03:56:34 CEST] <furq> original msys doesn't have a package manager [03:56:47 CEST] <k_sze[work]> I meant this: http://ffmpeg.org/platform.html#toc-Native-Windows-compilation-using-MSYS2 [03:56:51 CEST] <k_sze[work]> I forgot to paste the link. [03:56:57 CEST] <k_sze[work]> ok [04:13:45 CEST] <DHE> as long as you get a copy of yasm of at least the minimum version it's fine [09:48:38 CEST] <k_sze[work]> So if I understand correctly, x264 in ffmpeg 3.0.1 now supports threading? [09:48:51 CEST] <k_sze[work]> (whereas it didn't in e.g. ffmpeg 2.5.3) [09:48:53 CEST] <furq_> it's always supported threading [09:49:36 CEST] <k_sze[work]> because if I run `ffmpeg -help encoder=libx264` with ffmpeg 2.5.3, it says "Threading capabilities: no" [09:49:45 CEST] <k_sze[work]> zeranoe's build, btw. [09:50:50 CEST] <furq_> i'm not sure what that refers to [09:51:02 CEST] <furq_> i get the same with the zeranoe build but it definitely does multithreaded encoding [09:52:01 CEST] <k_sze[work]> At some point I need to figure out how to build ffmpeg with mingw-w64 [09:52:19 CEST] <furq_> under msys or cross-compiling [10:00:48 CEST] <k_sze[work]> hmm, this has slice_count, threads, and thread_type options. http://ffmpeg.org/ffmpeg-codecs.html#Codec-Options [10:02:02 CEST] <furq_> you don't usually need to pass -threads to x264, it'll autodetect [10:02:08 CEST] <furq_> and i don't think the other two apply to x264 [10:13:44 CEST] <brontosaurusrex> My terminal is sliglty bork when ffmpeg exists, what was the solution to this? [10:14:55 CEST] <furq> ^L [10:21:06 CEST] <BtbN> stty sane [10:21:25 CEST] <BtbN> on a clean exit that shouldn't happen though [10:22:31 CEST] <brontosaurusrex> stty sane < that means anything? [10:23:11 CEST] <BtbN> see man stty if you want to know what exactly it means [10:23:16 CEST] <BtbN> it fixes your terminal though [10:23:25 CEST] <brontosaurusrex> oh, ok, so it is a command :) [10:46:23 CEST] <k_sze[work]> furq: thread_type appears specifically under the x264 section as well: http://ffmpeg.org/ffmpeg-codecs.html#toc-Options-23 [10:48:18 CEST] <furq> fun [10:48:27 CEST] <furq> i guess sliced threading is only useful for low-latency encoding [11:19:30 CEST] <k_sze[work]> x264 does *not* do slicing by default? [11:23:51 CEST] <k_sze[work]> And my computer has 4 logical cores. Why does ffmpeg do threads=6 by default? [11:24:12 CEST] <c_14> I think it's 1.5 * numcpu [11:25:17 CEST] <lukesan> morning all [11:28:25 CEST] <furq> k_sze[work]: no, the default is frame-based threading [11:28:46 CEST] <furq> which is faster but has worse latency [11:29:10 CEST] <k_sze[work]> c_14 is that in case some threads are waiting for IO, the other threads can do work? [11:30:06 CEST] <c_14> encoder threads usually don't do io [11:30:30 CEST] <c_14> It's probably because switching to an inactive thread is faster than reinitializing the current thread [11:30:44 CEST] <k_sze[work]> ok [11:31:03 CEST] <c_14> *much io [11:31:21 CEST] <c_14> Everything does io in some sense [11:44:30 CEST] <hc2p> hey guys, i'm sending rtmp to nginx-rtmp, pushing it to ffmpeg -i rtmp://... then connection handshake fails with [11:44:35 CEST] <hc2p> [rtmp @ 0x37fc6e0] Erroneous C2 Message epoch does not match up with C1 epoch [11:44:35 CEST] <hc2p> [rtmp @ 0x37fc6e0] Erroneous C2 Message random does not match up [11:45:18 CEST] <hc2p> sorry, to be precise it's `ffmpeg -listen 1 -i rtmp://0.0.0.0:1935/live/test` [11:46:58 CEST] <hc2p> i have the feeling i'm missing smth really obvious [11:49:56 CEST] <hc2p> http://pastebin.com/fFb2nxc9 http://pastebin.com/Xy76zBEx [12:22:35 CEST] <realies> would ffmpeg treat .ts files differently than .mp4 [12:22:43 CEST] <realies> eg. do I need a container? [12:26:18 CEST] <k_sze[work]> Does anybody know what's the *default* -context when using ffv1 level 1? I can't seem to find it documented anywhere. [12:26:29 CEST] <k_sze[work]> And it's not shown when I run the ffmpeg command. [12:38:49 CEST] <DHE> realies: .ts is a container, it's called mpegts [12:39:18 CEST] <realies> DHE so it encapsulates the h264 inside? [12:39:33 CEST] <DHE> yes [12:39:47 CEST] <realies> DHE, right, ty [13:09:27 CEST] <hrw> hi [13:10:30 CEST] <hrw> can ffprobe ignore what it does not know and continue? http://paste.fedoraproject.org/356045/71858214 is "ffprobe /dev/sr0" with some dvd movie which I want to convert to h264 file [14:30:24 CEST] <BtbN> hrw, looks like it did continue? [14:31:01 CEST] <BtbN> the unknown stuff is in stream 0, and it still listed the other stream [14:31:42 CEST] <hrw> BtbN: I excepted audio tracks to be listed but it is $%#$%@ dvd disk so anything can be borken [14:32:07 CEST] <BtbN> or the first track it found only has video [14:32:24 CEST] <hrw> or [14:32:41 CEST] <hrw> anyway it is last time when I rip dvd for someone d: [14:34:18 CEST] <BtbN> there are better tools to rip a dvd than the ffmpeg cli. [14:37:21 CEST] <hrw> BtbN: used several of them. all faild [14:37:42 CEST] <BtbN> sounds like a broken disk to me [14:40:46 CEST] <hrw> BtbN: 'mplayer -dvd-device /dev/sr0 dvd://' is capable of playing fine. mencoder with same info generates out of sync stuff. handbrake gives soup of pixels [14:41:33 CEST] <BtbN> so of all tools available, you chose the ones least suited for ripping a DVD. [14:41:39 CEST] <hrw> ;D [14:42:13 CEST] <hrw> BtbN: dvd ripping during other work == not enough time to dig for all tools [14:43:42 CEST] <Snoo20664> hello all [14:44:00 CEST] <Snoo20664> ffmpeg -ss 00:01:00 -i video.mp4 -to 00:02:00 -c copy cut.mp4 [14:44:31 CEST] <Snoo20664> this ^ will cut out 1 minuit of video from 1m to 2m and save as cut.mp4 [14:44:52 CEST] <Snoo20664> how can i omit that and keep the proceeding and following sections of video [14:45:10 CEST] <BtbN> uhm, don't do anything to the input file? [14:45:14 CEST] <Snoo20664> so like i want to keep the first min, omit mins 1 to 2, and keep min 2 to 3 [14:45:24 CEST] <Snoo20664> i want to cut out an advert BtbN [14:45:33 CEST] <Snoo20664> at 1min to 2min in this example [14:45:56 CEST] <BtbN> make one video with only the first minute, and one starting at 2 minutes, and then concat those. [14:46:04 CEST] <Snoo20664> yeah [14:46:16 CEST] <Snoo20664> i was hoping there was a simpler/faster way [14:47:31 CEST] <Snoo20664> like the opposite of -ss and -to would be great [14:49:06 CEST] <BtbN> that's not easily possible, as the data that would product is non-contiguous. [14:49:58 CEST] <hrw> BtbN: so which tool would you suggest to try? [15:36:58 CEST] <Snoo20664> BtbN can you help [15:37:40 CEST] <Snoo20664> ffmpeg -ss 00:14:46 -i test.ts -to 00:28:17 -vcodec copy -acodec copy temp\%seg%.ts [15:38:00 CEST] <Snoo20664> instead of cutting from 00:14:46 to 00:28:17 [15:38:32 CEST] <Snoo20664> it cuts from 00:14:46, but doesn't end at 28:17, the video is just 28:17 long [15:42:32 CEST] <Snoo20664> it's using the -to time as a duration like -t instead of an endpoint [15:44:06 CEST] <Snoo20664> i think what it's doing is i set the start point as 00:14:46 with -ss 00:14:46 [15:44:45 CEST] <Snoo20664> and it jumps to that point and then set's it to 00:00:00, so when i set my end point of 00:28:17 it counts to that from it's new zero point [15:44:57 CEST] <Snoo20664> there has to be someway round this [15:47:04 CEST] <Snoo20664> got it [15:47:12 CEST] <Snoo20664> -copyts to maintain timestamps [15:47:20 CEST] <Snoo20664> ^should be a default setting tbh [16:55:08 CEST] <yzT> is this diagram correct? http://i64.tinypic.com/289kfo1.png [16:57:36 CEST] <Snoo20664> i don't think so [16:58:25 CEST] <yzT> what's wrong? [16:59:28 CEST] <yzT> mmmm well, the packager should connect to the CENC and the CENC to the website, right? [17:20:52 CEST] <satinder___> how we can capture current image from /dev/video0 [17:21:01 CEST] <satinder___> please anyone can help [17:26:41 CEST] <satinder___> hi please any know how we can capture a single current image from /dev/video0 with ffmpeg [17:26:54 CEST] <satinder___> ?? [17:30:13 CEST] <Snoo20664> ffmpeg -i /dev/video0 -frames:v 1 out2.jpg [17:30:31 CEST] <Snoo20664> should do it for you satinder___ [17:39:16 CEST] <satinder___> Snoo20664 -frame:v unrecognized option [17:43:49 CEST] <satinder___> ?? [18:30:34 CEST] <realies> is it possible to playback 9 clips in a 3x3 grid simultaneously [18:41:34 CEST] <benbro> how much bandwidth I need for live 720p@30fps VP8 video? [18:44:25 CEST] <DHE> depends on quality needs. 4 or 5 megabit/sec for video might be a good starting point [19:08:09 CEST] <benbro> DHE: I was hoping for 1-1.5 Mbps :) [19:09:38 CEST] <durandal_1707> realies: yes [19:16:58 CEST] <DHE> benbro: it depends on your content. cartoons might work out. a hockey game would not. [19:20:46 CEST] <benbro> DHE: I'm shooting a meeting [19:21:02 CEST] <DHE> ... 1.5 might work then. stationary camera and all that [19:21:25 CEST] <DHE> all I can say is, give it a try. see how it turns out [19:42:54 CEST] <Kuroonehalf> Yo guys. I currently have some really poorly ripped dvds that for some reason are quite large and I'd like to do something about. They're 480p, around an hour each, but somehow each one takes up around 1.2GB. They were oririnally in avi and I reencoded to mp4 but they're still way larger in size than it looks like they should be [19:43:02 CEST] <Kuroonehalf> is there perhaps some option ffmpeg has to deal with this? [19:43:15 CEST] <blue_misfit101> re-rip them :) [19:43:30 CEST] <Kuroonehalf> I was hoping that wasn't the only option :p [19:44:10 CEST] <kepstin> well, you can probably just re-encode them using a lower bitrate/lower quality setting/better codec, if the source looks decent. [19:44:48 CEST] <kepstin> obviously if you can start from the original dvd rather than this multiple-re-encoded file, that would be best. [20:17:04 CEST] <realies> durandal_1707, how? [20:17:51 CEST] <durandal_1707> hstack and vstack combo [20:20:43 CEST] <realies> I need to look into this [20:20:55 CEST] <realies> any ideas if it's resource intensive? [20:21:50 CEST] <realies> I want to make a scrubable/seekable 3x3 grid, tried with CasparCG, but it seems to not be able to handle the situation effectively [20:26:46 CEST] <durandal_1707> ffmpeg is for transcoding not playback [20:27:16 CEST] <furq_> you can use hstack with ffplay [20:27:22 CEST] <durandal_1707> you gona use libavfilter api? [20:27:26 CEST] <furq> whether it performs acceptably i have no idea [20:27:35 CEST] <realies> I thought ffplay functionality is embedded in ffmpeg [20:27:38 CEST] <furq> and you won't be able to seek the videos individually [20:27:51 CEST] <durandal_1707> not possible with ffplay [20:27:56 CEST] <realies> furq the idea is for them to stay in sync [20:31:08 CEST] <durandal_1707> mpv uses ffmpeg libs for playback [20:31:38 CEST] <furq> oh [20:31:54 CEST] <furq> so you can use hstack/vstack with ffplay but you can't provide multiple input files [20:32:26 CEST] <furq> so that's no good [20:33:41 CEST] <realies> any ideas for making that happen? [20:34:06 CEST] <furq> and i don't think you can use filter_complex with mpv [20:34:37 CEST] <realies> : / [20:35:36 CEST] <furq> as far as using ffmpeg goes i think your only option is to transcode it and stream/pipe the output to a player [20:35:56 CEST] <furq> or use the api [20:36:36 CEST] <realies> I wonder what the latency would be when transcoding [20:36:55 CEST] <furq> if you're piping it to a player then there's no reason to not use rawvideo [20:37:03 CEST] <furq> so it shouldn't be too bad [20:37:05 CEST] <realies> prob not pleasable for realtime operations [20:37:22 CEST] <realies> I'll try it out though [20:37:32 CEST] <furq> i guess you'll be limited by the decoding speed [20:41:24 CEST] <realies> can the API be used with .NET? [20:42:28 CEST] <furq> not to my knowledge [20:42:40 CEST] <furq> googling turns up a few wrappers but they're all really old and alpha versions [20:42:47 CEST] <realies> yeah... [20:42:58 CEST] <realies> bummer haha [20:42:58 CEST] <furq> which is par for the course for ffmpeg bindings [20:44:01 CEST] <realies> hm... https://github.com/Microsoft/FFmpegInterop [20:44:30 CEST] <furq> One of the advantages of this approach is that audio and video synchronization is handled by the Windows media pipeline. [20:44:39 CEST] <furq> i don't think microsoft know what "advantages" means [20:44:56 CEST] <realies> im a newb and don't get that [20:45:07 CEST] <realies> wm pipelines suck? [20:45:39 CEST] <furq> i assume that means directshow [20:46:02 CEST] <furq> i was always told that if you can't say anything nice about something then you should say nothing at all [20:46:12 CEST] <realies> :D [20:46:34 CEST] <realies> what does ffplay create in it's window then? [20:46:41 CEST] <furq> it's an SDL window [20:46:42 CEST] <realies> *playback window [20:47:09 CEST] <furq> it works exactly the same on every platform afaik [20:47:13 CEST] <realies> my manager was mentioning something about sdl today... [20:48:09 CEST] <furq> it looks like this ffmpeginterop lib only does basic decoding and playback stuff anyway [20:48:30 CEST] <furq> based on me looking at the source directory for about 20 seconds [20:48:37 CEST] <realies> haha [20:50:03 CEST] <realies> I guess making a wrapper for the ffmpeg binary is also a thing [20:50:10 CEST] <realies> a funny one [22:02:26 CEST] <blue_misfit101> how do you specify a qpfile when using libx264 in ffmpeg? [23:59:29 CEST] <snakeryslug> i want lossless gif to webm. is that possible? i've done `ffmpeg -i input.gif -c:v libvpx -crf 4 -b:v 32000K output.webm` but the output looks worse. [00:00:00 CEST] --- Sat Apr 16 2016

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