Ffmpeg-devel-irc

Download

ffmpeg-devel-irc@ffmpeg.org

October 2017

1 participants
62 discussions

ffmpeg-devel.log.20171010
by burek 11 Oct '17

11 Oct '17

[00:24:47 CEST] <cone-330> ffmpeg 03Jun Zhao 07master:71e2ec017a1b: lavc: enable hwaccel_flags option [00:24:48 CEST] <cone-330> ffmpeg 03Jun Zhao 07master:217a723b4e05: lavc/vaapi_decode: fix profile search when profile mismatch is allowed [00:27:51 CEST] <jamrial> jkqxz: https://pastebin.com/raw/8kTUy4gk [00:28:10 CEST] <jamrial> guess the range you hardcoded (0, 4) is wrong? [00:28:45 CEST] <jamrial> this is in pps rext [00:33:01 CEST] <jkqxz> "When present, the value of log2_max_transform_skip_block_size_minus2 shall be less than or equal to MaxTbLog2SizeY - 2." [00:34:12 CEST] <jkqxz> "The CVS shall not contain data that result in MaxTbLog2SizeY greater than Min( CtbLog2SizeY, 5 )." [00:34:30 CEST] <jkqxz> And isn't CtbLog2SizeY bounded above by 6? (64x64) [00:36:24 CEST] <jamrial> i have no idea [00:36:32 CEST] <jamrial> but does this mean the conformance sample is not in fact conformant? :p [00:37:06 CEST] <jkqxz> Does the trace look otherwise sensible? [00:38:41 CEST] <jamrial> seems fine otherwise [00:39:45 CEST] <jamrial> found this btw when i was trying to add cbs tests using the rext samples. all work and the output is bitexact, except for that one where parsing fails because of that one field [00:43:25 CEST] <jkqxz> Well, the rest of that stream has log2_min_luma_coding_block_size_minus3 = 1, log2_diff_max_min_luma_coding_block_size = 2. [00:43:53 CEST] <jkqxz> So MinCbLog2SizeY = 4 and CtbLog2SizeY = 6. [00:44:30 CEST] <jkqxz> So log2_max_transform_skip_block_size_minus2 can't be greater than 4. [00:44:35 CEST] <jkqxz> I wonder what the reference decoder says. [00:45:42 CEST] <jkqxz> " 233 log2_max_transform_skip_block_size_minus2 ue(v) : 6" [00:45:43 CEST] <jkqxz> Hmm. [00:46:46 CEST] <jkqxz> I must be getting something wrong here... [00:52:54 CEST] <jkqxz> It's not nonsensical for it to be higher, it just means "transform_skip_flag is always present if it can be". [00:54:50 CEST] <jkqxz> <https://hevc.hhi.fraunhofer.de/trac/hevc/ticket/1363> [00:55:06 CEST] <jkqxz> Maybe it really is out-of-range in that sample and noone ever noticed. [01:04:39 CEST] <jamrial> jkqxz: probably. there's a new sample replacing this one, called PERSIST_RPARAM_A_RExt_Sony_3, where this field has a value of 3 [01:04:46 CEST] <jkqxz> Thinking further, min(x, 5) is because the largest defined transform is 32x32. Given that, offering the possibility of transform skip for anything greater than 32x32 makes no sense, so log2_max_transform_skip_block_size_minus2 should be bounded above at 3. [01:04:54 CEST] <jkqxz> So the sample is just wrong? [01:05:20 CEST] <jamrial> i have no idea, but it very well could be, judging by that [01:05:23 CEST] <jamrial> http://wftp3.itu.int/av-arch/jctvc-site/bitstream_exchange/draft_conformanc… [01:05:39 CEST] <jamrial> since it's been superseeded... [01:06:21 CEST] <jkqxz> Right, in the explanatory text for that it has "Maximum transform skip log2 maximum size set to 5 (32x32) (previous revisions had the value of 8).". [01:06:42 CEST] <jamrial> mystery solved :p [01:06:45 CEST] <jkqxz> Yay! [01:06:51 CEST] <jamrial> maybe we should add this sample to the fate suite [01:07:30 CEST] <jkqxz> Would be nice if it could replace the invalid one... [01:07:39 CEST] <jamrial> it can't. old releases and all that [01:07:46 CEST] <jkqxz> But I guess that's not really allowed, yeah. [01:09:39 CEST] <jamrial> michaelni, nevcairiel: can you upload http://wftp3.itu.int/av-arch/jctvc-site/bitstream_exchange/draft_conformanc… (only the .bit file inside the zip) to the hevc-conformance folder? [01:10:47 CEST] <jkqxz> I've corrected the bounds to [0, 3] locally. [01:11:42 CEST] <jkqxz> What tests are you intending to add? I avoided running across everything in FATE because I thought people would say that was excessive, hence the current subset of streams hopefully covering all header cases. [01:12:01 CEST] <jkqxz> Normal cases, though, not rext. [01:25:15 CEST] <jamrial> jkqxz: tried all the rext ones (HEVC_SAMPLES_422_* and HEVC_SAMPLES_444_* in hevc.mak) [01:25:27 CEST] <jamrial> and only that one tripped with that field [01:25:43 CEST] <jamrial> the rest were bitexact after passing through hevc_metadata [01:25:56 CEST] <jamrial> so imo, add one of each [01:28:04 CEST] <jamrial> none had the extensions you don't support (3d, multilayer, etc) [01:39:57 CEST] <michaelni> jamrial, bit file uploded [01:40:12 CEST] <jamrial> michaelni: thanks [02:42:31 CEST] <cone-330> ffmpeg 03James Almer 07master:fde3bb16f90a: build: prevent SDL2 from polluting global cflags and extralibs [03:46:44 CEST] <cone-330> ffmpeg 03James Almer 07master:17ba9e123f1e: fate: update fate-api reference files after 71e2ec017a [17:38:48 CEST] <wm4> what does -Wbool-operation do? [17:38:58 CEST] <wm4> of course nobody knows because Carl Eugen Hoyos can't write commit messages [17:43:17 CEST] <jkqxz> Compiler tells you you are naughty for writing things like { ~(a == b) } or { !a & b }. [17:52:53 CEST] <BtbN> wm4, do you still want to rename stuff to nvdec btw.? I wanted to rename cuvid for a while, since nvdec is what nvidia calls it everywhere now. And the new thing would be a perfect chance to just call it nvdec from the beginning. [17:54:12 CEST] <wm4> jkqxz: ???? [17:54:30 CEST] <wm4> jkqxz: i.e. compiler devs being patronizing? [17:54:49 CEST] <wm4> BtbN: if we remove dummy hwaccels (Libav patch pending), then that won't be necessary for now [17:55:17 CEST] <BtbN> Yeah, I know it's not technically neccesary. [17:55:33 CEST] <wm4> or do you want the Libav thing renamed? [17:55:48 CEST] <BtbN> That was my idea, yes [17:56:07 CEST] <BtbN> Just calling it nvdec instead of cuvid. Avoids confusion between the current cuvid, and does the rename I wanted to do eventually anyway [17:58:24 CEST] <wm4> would it be ok to rename the ffmpeg cuvid source file anyway, and commit the Libav thing under the libav name, then rename it in a separate commit? I'm worried about git's rename tracking [17:58:38 CEST] <BtbN> yeah, of course [17:59:06 CEST] <BtbN> keeping the history like that sounds like a good idea to me [17:59:45 CEST] <Emerica> I have modified mpegtsenc to support user specified pid values based on the the stream index. ie -mpegts_pids "0=1311,1=1411" I'm using avstrtok to breakup the input string and store the pid values to an array. is this the wrong approach in bringing in the input values? [18:00:40 CEST] <BtbN> wouldn't sscanf be easier? [18:00:59 CEST] <Emerica> I don't know enough to know, but now I'll look :) [18:31:57 CEST] <BtbN> wm4, what I'm kinda missing in the libav-cuvid is any kind of capability check. With ffmpeg own h264 parser in place, the information should all be readily available. [18:32:09 CEST] <BtbN> But I guess I can just add one once it's merged [18:33:37 CEST] <wm4> hm I assume it relies on h264 calling this only for generally supported profiles? still seems rather fishy [18:34:09 CEST] <BtbN> It just assumes the hardware supports 10bit, no check on that whatsoever [18:34:20 CEST] <BtbN> and the resolution and so on [18:35:11 CEST] <nevcairiel> its a normal hwaccel in the future, it should implement checks like any ohter hwaccels [18:35:45 CEST] <wm4> yeah, that seems lacking [18:36:27 CEST] <BtbN> I'd not change the cherry-picked commit too much though. That stuff can be added later [18:38:01 CEST] <wm4> is the change to dynamic loading within the cherry-pick commit still ok? [18:38:12 CEST] <BtbN> Yeah, that's essential for it to work at all [18:39:02 CEST] <BtbN> There's also a bunch of if-defery and configure checks for features tied to that. No point in dealing with that. [18:40:31 CEST] <wm4> yes, it'd need to temporarily introduce full linking support [18:45:38 CEST] <BtbN> changing the libav cherry-pick so much that it actually works seems fine enough. Additions like capability checks are ffmpeg patches on top of it. [18:46:00 CEST] <BtbN> Also allows for them to be maybe merged back to libav if anyone cares. [19:20:25 CEST] <ticktack> Hey! I am new to ffmpeg and want to work on some beginner stuff, any suggestion or any issue i can work on [19:24:15 CEST] <JEEB> ticktack: start with compiling it, then think of what interests you within it or a problem that you can feel yourself [19:24:31 CEST] <JEEB> usually if you yourself are affected it's better [19:54:34 CEST] <Compn> doh he left [20:12:48 CEST] <cone-404> ffmpeg 03Marton Balint 07master:ff6de6b180fd: Makefile: generate stripped CLI tools directly instead of copying unstripped ones first [20:12:49 CEST] <cone-404> ffmpeg 03Devin Heitmueller 07master:278588cd0bf7: libavdevice/decklink: add support for -sources and -sinks arguments [20:12:50 CEST] <cone-404> ffmpeg 03Devin Heitmueller 07master:77f7d710e086: libavdevice/decklink: add support for 10-bit output for Decklink SDI [20:17:12 CEST] <jamrial> michaelni: it should now be fine to release 3.4 [20:34:44 CEST] <durandal_1707> cehoyos is very short sighted persona [20:36:32 CEST] <BBB> now now now [20:36:40 CEST] <BBB> lets remain nice [22:21:19 CEST] <BBB> this helmut guy I suspect check_disable_warning has a bug very dry :D [22:30:37 CEST] <wm4> isn't it that clang simply doesn't error out on unknown warnings and thus the check succeeds [22:30:52 CEST] <wm4> so not really an error, just sabotage by the compiler devs [22:34:18 CEST] <JEEB> yea I noticed the unknown options just got used :) [22:57:25 CEST] <jkqxz> Is anyone reverting the second -Wno-bool-operation patch? It seems pretty clear now that it's not acceptable with a lot of versions of clang (only apple ones?). [23:02:05 CEST] <BBB> maybe adjust the compiler check so it catches the specific warning from clang also, and the flag is not inserted? [23:02:15 CEST] <BBB> that seems the whole goal of that configure check [23:02:35 CEST] <jkqxz> I do not have such a clang. Do you? [23:06:16 CEST] <BBB> $ gcc -Wno-bool-operation /dev/null -c -o /tmp/x.o && echo success [23:06:17 CEST] <BBB> clang: warning: /dev/null: 'linker' input unused [-Wunused-command-line-argument] [23:06:18 CEST] <BBB> clang: warning: argument unused during compilation: '-Wno-bool-operation' [-Wunused-command-line-argument] [23:06:19 CEST] <BBB> success [23:06:20 CEST] <BBB> ? [23:07:10 CEST] <BBB> (Apple LLVM version 8.1.0 (clang-802.0.42)) [23:08:25 CEST] <BBB> $ gcc -Wno-bool-operation /dev/null -c -o /tmp/x.o -Werror=unused-command-line-argument && echo success [23:08:25 CEST] <BBB> clang: error: /dev/null: 'linker' input unused [-Werror,-Wunused-command-line-argument] [23:08:26 CEST] <BBB> but [23:08:30 CEST] <BBB> $ gcc -Wno-bool-operation /tmp/x.c -c -o /tmp/x.o -Werror=unused-command-line-argument && echo success [23:08:31 CEST] <BBB> warning: unknown warning option '-Wno-bool-operation'; did you mean '-Wno-bool-conversion'? [-Wunknown-warning-option] [23:08:32 CEST] <BBB> 1 warning generated. [23:08:33 CEST] <BBB> success [23:08:37 CEST] <BBB> so thats not very useful [23:12:32 CEST] <wbs> BBB: try -Werror=unknown-warning-option [23:12:40 CEST] <JEEB> ah [23:12:49 CEST] <BBB> woohoo [23:12:58 CEST] <BBB> wbs: you are genious [23:13:33 CEST] <wbs> (that particular behaviour in clang is rather annoying) [23:13:53 CEST] <BBB> but why didnt the original message say that? (I admit I ignored the second message) [23:14:15 CEST] <BBB> it sounds almost as if theyre using a state machine for error reporting [23:20:13 CEST] <cone-629> ffmpeg 03Gyan Doshi 07master:d251effe7320: doc/filters: note minimum resolution for pixscope [23:29:34 CEST] <cone-629> ffmpeg 03Marton Balint 07master:f280575a0fc9: configure: fix decklink dependencies [23:39:56 CEST] <thebombzen> zimg supports sRGB transfer characteristics, but the zscale filter wrapper does not. Is this just an oversight? should I send a patch to add this feature? [23:48:38 CEST] <wm4> ask durandal_1707, but most likely yes [23:50:12 CEST] <thebombzen> turns out it would be easy to support, but you'd need to add SRGB to "enum AVColorTransferCharacteristic" in libavutil/pixfmt.h [23:50:55 CEST] <thebombzen> the comment says "These values match the ones defined by ISO/IEC 23001-8_2013 § 7.2." so I'm not sure where or if it would be a good idea to add it immediately [23:51:36 CEST] <jkqxz> Is the transfer you want actually the same as one of the others? [23:52:59 CEST] <jkqxz> (If you don't feel like paying 158 swiss fracs for the standard, you may wish to know that it happens to be identical to ITU H.273.) [23:53:29 CEST] <wm4> thebombzen: you could add it as value over 256 lol [23:53:41 CEST] <wm4> because the bitstream field in h264/hevc is only 8 bit wide [23:54:09 CEST] <wm4> most likely this would trigger a very unproductive bikeshed flamewar though [00:00:00 CEST] --- Wed Oct 11 2017

1 0

ffmpeg.log.20171010
by burek 11 Oct '17

11 Oct '17

[01:08:50 CEST] <kevinn> Hi! I am having trouble getting av_parser_parse2 to work correctly. I am compressing a movie with x264 with key_int = 1. So every frame that is generated by x264 should be immediately render-able. However av_parser_parse2 is returning the following: pkt->size = 0 signifying that it doesn't have anything to render [01:08:59 CEST] <kevinn> any help would be greatly appreciated! [01:10:26 CEST] <kevinn> This is in reference to the first frame that is produced by x264 [01:23:50 CEST] <teratorn> kevinn: you have to feed some frames first usually before you get any out due to codec latency.. but if they are all keyframes.. who knows? [01:24:32 CEST] <kevinn> Ya that's what I thought... It should output right off the bat [01:26:58 CEST] <DHE> still, set tune=zerolatency to be sure [01:32:30 CEST] <kevinn> hmm, okay I'll try that now [01:34:21 CEST] <kevinn> just to be clear you mean this right? [01:34:23 CEST] <kevinn> av_opt_set(mCodec->priv_data, "tune", "zerolatency", 0); [01:35:59 CEST] <kevinn> DHE: is there anyway to check to see if my x264 frame is actually a keyframe? [01:36:09 CEST] <kevinn> I set that tune and it is still not working [01:40:07 CEST] <DHE> the AVPacket has a flag for keyframes [01:44:09 CEST] <kevinn> okay it is not a keyframe [01:44:31 CEST] <kevinn> DHE: does that mean x264 is not producing the correct buffers? [01:45:03 CEST] <kevinn> I noticed that the very next frame produced by x264 has all the correct data inside of it [01:45:14 CEST] <kevinn> curious what was even in the first frame then... [01:55:54 CEST] <kevinn> DHE: can you think of any reason why the first frame produced by x264 is not a keyframe? [01:57:03 CEST] <DHE> sorry, was playing video games [01:57:25 CEST] <DHE> even if you ask for all keyframes, x264 will buffer frames for bitrate estimation. [01:57:30 CEST] <DHE> use tune=zerolatency if this matters to you [01:58:25 CEST] <kevinn> I have both ends tuned to zerolatency though [01:58:32 CEST] <kevinn> both libav and x264 [01:58:58 CEST] <kevinn> How do I make x264 not be behind by a frame like this [02:01:37 CEST] <teratorn> kevinn: might just be an implementation detail of the codec [02:02:27 CEST] <JEEB> kevinn: are you fucking sure it's not just parameter sets first? [02:02:43 CEST] <JEEB> and then actual pictures [02:04:02 CEST] <kevinn> ahh, I didn't know the first frame is always the parameter set, sorry, I will account for this going forward [02:04:31 CEST] <JEEB> it depends, and if you're goddamn using libavcodec just look at the goddamn wrapper [02:05:45 CEST] <JEEB> since x264_encoder_headers is only called under avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER I would guess it's in either the first packet or the first one is the parameter sets [02:06:55 CEST] <DHE> wait, he's using x264 directly? [02:07:09 CEST] <JEEB> fuck if I fucking know. he was saying something on x264dev and something here [02:07:22 CEST] <JEEB> it's really fucking tiring and he couldn't even bother to check what he's outputting [02:07:25 CEST] <JEEB> for fuck's sake [02:07:34 CEST] <DHE> ah that explains the cursing then [02:07:35 CEST] <DHE> carry on [02:07:47 CEST] <JEEB> DHE: my comment regarding the headers function was from libx264.c [02:07:50 CEST] <JEEB> in libavcodec [02:08:17 CEST] <JEEB> which leads me to think that you get the parameter sets with the initial encoded frame(s) [02:08:42 CEST] <JEEB> and I know the libavcodec parser at least separates parameter sets into their own packets [02:08:52 CEST] <JEEB> so if you have PPS/SPS/VPS in HEVC they're all separate [02:10:30 CEST] <kevinn> DHE: yes I am using x264 directly... should I not? [02:10:46 CEST] <JEEB> nothing bad with that [02:11:27 CEST] <JEEB> the reason why you're so irritating is somewhere completely different [02:12:22 CEST] <DHE> yeah. I just assumed you were using ffmpeg's libavcodec wrapper around it as literally everyone else does [02:12:49 CEST] <JEEB> you're of the species that instead of checking "what on earth x264 returned to me" decided to start playing with parameter [02:12:52 CEST] <JEEB> *parameters [02:13:06 CEST] <JEEB> (he was starting to ask around for min-keyint etc on x264dev) [02:13:31 CEST] <JEEB> anyways, I hope you all the good [02:14:06 CEST] <kevinn> I am sorry I really did not want to frustrate you JEEB, I am all good now thank you. [02:14:21 CEST] <kevinn> I will check that wrapper more carefully too before I ask anymore questions [02:14:28 CEST] <kevinn> Didn't know it existed [02:14:43 CEST] <JEEB> the wrapper quite honestly doesn't do too much. just maps the buffers to AVPackets and such [02:15:02 CEST] <JEEB> the problem really is that you made zero attempt at looking at what was going on it seems [02:15:21 CEST] <JEEB> like, you could have done something like -show_frames with ffprobe? [02:15:27 CEST] <JEEB> then found out that there are no frames in there [02:15:39 CEST] <JEEB> then maybe thought that AVC needs initialization data? [02:16:03 CEST] <JEEB> and there are other parsers for AVC bit streams of course [02:16:23 CEST] <kevinn> what should I use for reference then? I checked the flag that AVPackets returned and it said that the frame wasn't a keyframe. I was told on x264dev that if I set keyint=1 then all frames would be a keyframe, hence my confusion [02:16:39 CEST] <JEEB> BY CHECKING THE FUCKING DATA [02:16:57 CEST] <JEEB> vittu perkele, I will just stop wasting my time on this [02:17:32 CEST] <furq> you know it's bad when the swearing is finnish [02:17:39 CEST] <kevinn> I am sorry JEEB, I promise next time I will check the data. [02:17:57 CEST] <kevinn> thank you for the help though [04:29:31 CEST] <eric_hill> trying to extract a video frame from a CCTV camera .DAV file - ffmpeg -y -i in.dav -vcodec libx264 -ss 30 -qscale:v 2 -vframes 1 out.jpg [04:29:49 CEST] <eric_hill> jpeg looks like an h264 frame, not a jpeg. What am I doing wrong? [04:32:32 CEST] <kepstin> eric_hill: you have "-vcodec libx264" in there, which is telling it to encode an h264 frame [04:32:43 CEST] <kepstin> i'm surprised that doesn't error out, tbh. [04:35:48 CEST] <eric_hill> hmm. If I don't specify it, ffmpeg has trouble figuring out the source .dav is x264 encoded. It's apparently some proprietary Chinese one-off codec... [04:38:44 CEST] <eric_hill> kepstin, spot on. Removing that worked great. Facepalm. Tx. [04:45:32 CEST] <Johnjay> JEEB: what is playing with parameters and why is it bad lol? [05:55:04 CEST] <LunixA380> Hello guys. I have a problem with ffmpeg. It uses 22 GiB (and more, actually it used all the RAM of the VM and then crashed by OOM) hile usually, when I transcode file, it runs in order of hundred of megabytes used. [05:55:47 CEST] <LunixA380> I guess it is because I try to concat 700+ .ts files (because of HLS technology) [06:13:38 CEST] <stevenliu> demux hls [06:13:45 CEST] <stevenliu> or concat format? [06:16:25 CEST] <LunixA380> concat filter_complex [06:17:30 CEST] <LunixA380> stevenliu: https://paste.fedoraproject.org/paste/qo1OLGPG3TntAY2mQ8KsqA [06:17:58 CEST] <LunixA380> When I say "22 GiB" this is not a joke, and despite so much RAM it crashes :/ [06:20:00 CEST] <LunixA380> With a complex filter I can't do -codec copy, otherwise I would have ruled out (or not) the encoder [06:20:05 CEST] <akkad> did you buy Gibabyte ram? or gigabyte ram? [06:20:34 CEST] <LunixA380> I talk ion Gibibyte actually. 22 Gibibyte. I don't buy anything, it's cloud though. [06:20:43 CEST] <LunixA380> I don't have so much RAM at my disposal [06:48:24 CEST] <LunixA380> Any idea? [07:35:48 CEST] <LunixA380> The filtergraph feels simple. I have tried to put on loglevel debug, pretty unhelpful. I wonder if scaling the slices before concat is better than concat and scale, but I don't see how. On Internet people says "high memory usage is because ffmpeg thinks there is a cycle and basically stores all the video/audio raw in memory" [07:36:00 CEST] <LunixA380> Obviously, if it does that, it'll run out of memory [07:36:21 CEST] <LunixA380> I suspect it does, but I don't have proofs from, say, a log message (if such log message exists) [08:36:08 CEST] <JEEB> Johnjay_: read the darn context, which was this guy complaining that his first received data from x264 was not a random access picture. instead of taking a look wtf x264 returned him (most likely the headers) he started randomly poking parameters [09:33:47 CEST] <LunixA380> After investigation, the problem seems to be the concat filter. [09:34:08 CEST] <JEEB> funky, that filter has been rather stable for me in 2015 or so [09:34:16 CEST] <JEEB> although I was concatenating very simple things [09:34:24 CEST] <LunixA380> Well, yes, but [09:34:34 CEST] <JEEB> (MPEG-2 Video, AVC + raw audio in MXF) [09:34:34 CEST] <LunixA380> I have tried to just *extract the sound* (audio streams) and leave videos on the side for now. And only for audio it uses [09:34:40 CEST] <LunixA380> it uses 700 MiB of RAM [09:35:25 CEST] <LunixA380> I know libvorbis enough to say it will never use so much RAM on its own, and if audio already uses 700 MiB, I easily understand how video can go beyond 22 GiB. [09:35:54 CEST] <JEEB> the amount of memory used depends on the amount of buffering done [09:36:02 CEST] <JEEB> that depends on the filter chain and inputs and a few other things [09:36:09 CEST] <JEEB> what I've mostly seen I don't think went that high [09:36:14 CEST] <JEEB> but that was a while ago [09:36:36 CEST] <JEEB> and the synchronization logic IIRC was partly reworked lately, among other things [09:36:55 CEST] <LunixA380> I know it depends of a lot of things [09:37:13 CEST] <LunixA380> But the thing I did is: ffmpeg -i [many files] -filter_complex concat audio and that's it [09:37:30 CEST] <furq> can you not use the hls demuxer [09:37:41 CEST] <JEEB> oh, HLS demuxer :D [09:37:44 CEST] <JEEB> that funny little thing [09:38:37 CEST] <LunixA380> IMO the fact there is many input files is the cause (because the combined size on-disk of .ts files are fair) [09:39:53 CEST] <JEEB> well I'm not sure how many decoders and demuxers are being brought up :) [09:40:02 CEST] <JEEB> even if you only utilize the audio tracks [09:40:05 CEST] <JEEB> in the concat filter [09:40:17 CEST] <JEEB> also if it's MPEG-TS you can literally just concatenate those together :D [09:40:21 CEST] <furq> i mean if this is mpegts then you almost certainly don't need to use the...yeah [09:40:38 CEST] <furq> if you have the m3u8 then i assume you can just use that as the input and it'll do something approaching the right thing [09:40:43 CEST] <LunixA380> JEEB: I've searched enough Google to know I can do that, I'm literally on it since hours [09:40:48 CEST] <furq> but failing that you can just cat them [09:41:18 CEST] <furq> LunixA380: by "literally concatenate" he means cat file1.ts file2.ts > out.ts [09:41:22 CEST] <furq> except with more than two files obv [09:41:27 CEST] <LunixA380> I know [09:41:41 CEST] <furq> it sounds like the filter is just being dumb about that many inputs [09:42:25 CEST] <LunixA380> But it is told it can desync sound/files between them. I guess it's case-by-case, and I've reviewed the file: they share same aspect ratio etc., but NOT the bitrate. [09:42:52 CEST] <furq> they don't need to be the same bitrate unless it's supposed to be cbr [09:43:19 CEST] <LunixA380> No, anyway I just want to reconvert them. [09:43:29 CEST] <LunixA380> So CBR or not I'll redo them from top to bottom [09:44:26 CEST] <LunixA380> OK then let's try the "simple solution I reject to do since hours because I fear that it would destroy the file" [09:44:44 CEST] <furq> i take it you don't have the m3u8 [09:44:50 CEST] <furq> because i would've thought that was the simplest solution [09:46:09 CEST] <LunixA380> I have the m3u8 [09:46:19 CEST] <LunixA380> I have even 2 [09:46:44 CEST] <furq> if it has relative paths then i would assume you can just do -i foo.m3u8 out.mp4 [09:46:57 CEST] <furq> i've never worked with an m3u8 locally though [09:47:14 CEST] <LunixA380> And I do what, the concat? [09:47:28 CEST] <furq> no, literally just that [09:47:32 CEST] <LunixA380> Ah. [09:47:37 CEST] <LunixA380> I'm going to try [09:47:37 CEST] <furq> given an m3u8 as input it'll process all the files in the playlist in order [09:53:28 CEST] <LunixA380> Well. It _kinda_ works [09:53:36 CEST] <LunixA380> I've a lot of "Non-monotonous DTS in output stream 0" [09:55:59 CEST] <LunixA380> Memory usage: insignificant. I can't even give a number given it's so small [09:56:08 CEST] <LunixA380> I confirm the concat filter make FFMpeg fully cray [09:56:11 CEST] <LunixA380> crazy [09:56:49 CEST] <LunixA380> 29 MiB of RAM used vs. 700 with concat filter (both in audio only situation) [10:02:10 CEST] <Fyr> guys, what container is less CPU-consuming for FFMPEG? [10:02:28 CEST] <Fyr> I noticed that muxing into matroska is faster than muxing into MP4. [10:02:43 CEST] <LunixA380> Same codecs? [10:02:48 CEST] <Fyr> yes [10:03:13 CEST] <LunixA380> I think ffmpeg has benchmarks flag but I don't know much and whether I'm right [10:03:13 CEST] <Fyr> especially, MP4 is much slower if I set faststart. [10:04:30 CEST] <JEEB> yes, since you're essentially going through the full file again [10:04:40 CEST] <JEEB> although that's not CPU intensive [10:04:42 CEST] <JEEB> it's IO intensive [10:04:51 CEST] <Fyr> oh [10:05:03 CEST] <JEEB> faststart means that the index it would otherwise write at the end [10:05:11 CEST] <Fyr> JEEB, so, what container is designed to be fast? [10:05:32 CEST] <JEEB> it will take its contents into memory and start rewriting the file again with its contents having the index in the beginning [10:05:36 CEST] <JEEB> that is what "faststart" is [10:05:45 CEST] <JEEB> its contents -> the index's [10:06:06 CEST] <JEEB> Fyr: uhh, we're going so far into IO bound territory I'm not sure [10:06:19 CEST] <LunixA380> faststart has index written at the beginning if I recall correctly [10:06:35 CEST] <JEEB> it can't be written in the beginning to start with, it has to do another pass of muxing [10:06:36 CEST] <LunixA380> However I think webm has the same constraints by default (doesn't matter if you put a flag or not) [10:06:51 CEST] <LunixA380> Yes [10:06:59 CEST] <JEEB> no, matroska is fundamentally different in its structures so at least in theory it doesn't need a "faststart" [10:07:01 CEST] <LunixA380> Because ffmpeg can't know in advance what the index will contain [10:07:03 CEST] <JEEB> and webm is matroska [10:07:07 CEST] <Fyr> I also noticed that TS is faster than MKV. [10:07:07 CEST] <JEEB> well, d'uh [10:07:31 CEST] <JEEB> Fyr: if you're really being blocked by the IO then I congratulate you for having a very fast workflow [10:07:36 CEST] <JEEB> or well, not blocked I mean [10:07:46 CEST] <JEEB> if you're *CPU* bound with those that's a woah [10:08:15 CEST] <LunixA380> It's possible if ou do your video files inside tmpfs [10:10:44 CEST] <LunixA380> "Non-monotonous DTS in output stream 0:0" how top investigate that issue now? [10:15:05 CEST] <Nacht> Does anyone know that the concat process of FFMPEG has any checks in it to see if it's all correct ? [10:15:30 CEST] <Nacht> As it, that you're not missing a small portion of data due to the system having a hickup [10:23:38 CEST] <rabbe> hi, need urgent help [10:24:03 CEST] <rabbe> i've created an rtmp stream through nginx [10:24:08 CEST] <rabbe> now i need to record that stream [10:24:40 CEST] <rabbe> when i do the recording the timestamps in ffmpeg does not correspond to real world time [10:25:27 CEST] <rabbe> so when i quit recording i don't have a video with the real amount of time that the recording really lasted [10:30:50 CEST] <LunixA380> Nacht: ffmpeg is not affected by system load if you don't do streaming, the only hiccups you may encounter is if the file is actually failed, or if the software/hardware is [10:30:59 CEST] <LunixA380> But in normal circumstances, you should be OK. [10:31:26 CEST] <LunixA380> Have to go, bye [10:47:30 CEST] <thebombzen> lol it still says "Non-monotonous" [10:47:39 CEST] <thebombzen> I was wondering when someone would change it to "monotonic" [10:47:55 CEST] <thebombzen> monotonous is "speaking in a monotone" [10:48:08 CEST] <thebombzen> "monotonic" means "nondecreasing" [10:52:53 CEST] <rabbe> using a capture card, i create this stream: ffmpeg -r 60 -f v4l2 -video_size hd1080 -i /dev/video0 -c:a aac -ac 2 -b:a 128k -c:v libx264 -pix_fmt yuv420p -preset ultrafast -tune zerolatency -x264-params "nal-hrd=cbr" -b:v 5000k -minrate 5000k -maxrate 5000k -bufsize 10000k -g 60 -f flv rtmp://192.168.0.16/live/test [10:53:02 CEST] <rabbe> how do i record the rtmp stream? [10:53:20 CEST] <rabbe> into mp4 file [10:54:36 CEST] <stevenliu> ffmpeg -r 60 -f v4l2 -video_size hd1080 -i /dev/video0 -c:a aac -ac 2 -b:a 128k -c:v libx264 -pix_fmt yuv420p -preset ultrafast -tune zerolatency -x264-params "nal-hrd=cbr" -b:v 5000k -minrate 5000k -maxrate 5000k -bufsize 10000k -g 60 - | ffmpeg -i - -c copy -f flv rtmp://192.168.0.16/live/test -c copy -f mp4 -movflags faststart test.mp4 [10:55:48 CEST] <rabbe> but i only need to record sometimes [10:56:00 CEST] <rabbe> whereas the rtmp feed should be created all the time [10:56:10 CEST] <rabbe> forgot to say that [10:56:46 CEST] <stevenliu> crontab [10:57:08 CEST] <rabbe> recording times should be decided by robot [10:57:16 CEST] <rabbe> i just need an ffmpeg command to do it [10:57:52 CEST] <titbang> https://superuser.com/questions/901339/continuous-streaming-from-an-empty-p… [10:57:57 CEST] <rabbe> the one i use now seems to lag in time.. so 10 seconds real time will become maybe 2 seconds video file [10:58:12 CEST] <rabbe> ok, i'll check [11:08:22 CEST] <rabbe_> titbang: that is not the same thing? [11:11:25 CEST] <furq> rabbe_: you can use the tee muxer if you want to output an encoded stream to multiple places [11:11:33 CEST] <furq> there's no way to turn one on or off on demand though [11:11:38 CEST] <furq> !muxer tee @rabbe_ [11:11:38 CEST] <nfobot> rabbe_: http://ffmpeg.org/ffmpeg-formats.html#tee-1 [11:12:20 CEST] <furq> also you probably don't want to use mp4 because you'll get an unplayable file if any of the inputs or outputs drops out [11:12:29 CEST] <furq> use mkv, mpegts, flv etc [11:13:56 CEST] <rabbe_> this works, but gives me too short video: ffmpeg -i rtmp://localhost/live/test -c:v libx264 -crf 18 -vf format=yuv420p /usr/local/nginx/html/video/output.mp4 [11:14:07 CEST] <stevenliu> you can use crontab or write a crontab task scheduler by your self to start record [11:15:02 CEST] <rabbe_> when i run that the timestamps are not real time, if i kill the process when the time stamps are at 00:00:03 i will get a 3 second video, but in reality it might have recorded during 10 seconds [11:15:05 CEST] <stevenliu> 1st, what's the info of the rtmp streams ? [11:15:51 CEST] <rabbe_> ok, hold i will go offline to connect to that network [11:20:08 CEST] <rabbe> https://pastebin.com/T8ftApJC [11:21:20 CEST] <rabbe> i have a newer ffmpeg on the server than on this computer that i ran the command at [11:22:28 CEST] <_julian> hi [11:22:52 CEST] <rabbe> the plan is to have some Uython code which will spawn a process with ffmpeg, then kill it CTRL-C style.. that should work, right? [11:23:03 CEST] <stevenliu> you need not transcode, if you want record, just use -c copy should be ok [11:23:30 CEST] <stevenliu> oh, no, don't use ctrl+c [11:23:51 CEST] <furq> rabbe: ^C will absolutely not work with mp4 [11:24:09 CEST] <rabbe> ok, so something like: ffmpeg -i rtmp://localhost/live/test -c:copy /usr/local/nginx/html/video/output.mp4 ? [11:24:15 CEST] <furq> -c copy [11:24:23 CEST] <rabbe> ok, so something like: ffmpeg -i rtmp://localhost/live/test -c copy /usr/local/nginx/html/video/output.mp4 ? [11:24:28 CEST] <furq> and like i said, don't use mp4 unless you want unplayable files [11:24:29 CEST] <stevenliu> if you want record to mp4, maybe you should record to ts or flv at frist, that can use ctrl+c, complete record to flv or ts, you can remux the ts or flv to mp4 and the end [11:24:34 CEST] <furq> ^ [11:24:59 CEST] <_julian> I'm having an issue with an aac_latm audio track inside mpegts. If I open that stream and run avformat_find_stream_info on it, everything seems fine and I get continuous pts out for each packet. If I skip avformat_find_stream_info, as this does not work well for mpegts live streams I do get lots of packets with pts==0 out of the codec. In both cases AVStream->needs_parsing is AVSTREAM_PARSE_FULL [11:25:08 CEST] <_julian> any thoughts what could cause this difference in behaviour? [11:25:11 CEST] <stevenliu> ffmpeg -i rtmp://localhost/live/test -c copy /usr/local/nginx/html/video/output.ts ; ffmpeg -i output.ts -c copy output.mp4; [11:25:46 CEST] <rabbe> okay i will try that [11:30:39 CEST] <rabbe> now the timestamps are ahead :) should i have -r 60 somewhere? [11:31:03 CEST] <rabbe> i get really high framerates [11:31:44 CEST] <stevenliu> Stream #0:0: Video: h264 (Constrained Baseline), yuv420p, 1920x1080, 4999 kb/s, 62.50 fps, 60 tbr, 1k tbn, 120 tbc [11:31:48 CEST] <stevenliu> this is the stream info [11:31:53 CEST] <stevenliu> 62.50 fps [11:32:31 CEST] <rabbe> yes, i use -r 60 to create the rtmp: ffmpeg -r 60 -f v4l2 -video_size hd1080 -i /dev/video0 -c:a aac -ac 2 -b:a 128k -c:v libx264 -pix_fmt yuv420p -preset ultrafast -tune zerolatency -x264-params "nal-hrd=cbr" -b:v 5000k -minrate 5000k -maxrate 5000k -bufsize 10000k -g 60 -f flv rtmp://192.168.0.16/live/test [11:32:50 CEST] <rabbe> so i should use 62.5 when recording the rtmp? [11:33:21 CEST] <stevenliu> no need [11:33:29 CEST] <stevenliu> -c copy can do everything [11:34:05 CEST] <rabbe> so the frame rate info when recording can be like > 100 fps? [11:34:29 CEST] <stevenliu> BTW: -b:v 5000k -minrate 5000k -maxrate 5000k -bufsize 10000k cannot control the bitrate , you should set the -bufsize small [11:34:59 CEST] <stevenliu> yes [11:35:05 CEST] <rabbe> okay [11:35:52 CEST] <stevenliu> ffmpeg -r 60 -f v4l2 -video_size hd1080 -i /dev/video0 -c:a aac -ac 2 -b:a 128k -c:v libx264 -pix_fmt yuv420p -preset ultrafast -tune zerolatency -x264-params "nal-hrd=cbr" -b:v 5000k -minrate 5000k -maxrate 5000k -bufsize 1600k -g 90 -r:v 30 -f flv rtmp://192.168.0.16/live/test [11:36:22 CEST] <stevenliu> it will 3s one GOP, and the fps is 30 [11:37:52 CEST] <stevenliu> the video quality it very terrible ultrafast is too fast, but the quality is bad [11:38:49 CEST] <stevenliu> if you don't care the quality, you can use it, if you care it, maybe insert some B slice is better, but the lantency maybe not good [11:39:19 CEST] <stevenliu> you can assessment it [11:39:41 CEST] <rabbe> no, please give me advice :) [11:40:12 CEST] <rabbe> i want good quality, as long as it will work over network [11:40:53 CEST] <stevenliu> medium will better, but it will cost lots of cpu cores, if you use soft encode [11:41:09 CEST] <stevenliu> maybe 8 cores will be cost [11:41:43 CEST] <rabbe> ok [11:42:35 CEST] <stevenliu> GOP you can set 2s -g 60 -r:v 30 [11:43:00 CEST] <rabbe> but ok, you say i should first skip these? -b:v 5000k -minrate 5000k -maxrate 5000k [11:43:19 CEST] <stevenliu> no [11:44:16 CEST] <stevenliu> ffmpeg -r 60 -f v4l2 -video_size hd1080 -i /dev/video0 -c:a aac -ac 2 -b:a 128k -c:v libx264 -pix_fmt yuv420p -preset medium -tune zerolatency -x264-params "nal-hrd=cbr" -b:v 5000k -minrate 5000k -maxrate 5000k -bufsize 1600k -g 60 -bf 15 -r:v 30 -f flv rtmp://192.168.0.16/live/test [11:44:49 CEST] <rabbe> -bufsize small == bufsize 1600k? [11:45:17 CEST] <stevenliu> no [11:46:13 CEST] <stevenliu> -bufsize 1600k? [11:46:22 CEST] <stevenliu> control the buffer size of encoding [11:46:35 CEST] <furq> fyi -minrate doesn't actually do anything with x264 [11:46:55 CEST] <rabbe> ok [11:47:48 CEST] <beo> hello, i have a question, if i concatenate mp3 files with ffmpeg, does it reencodes them? [11:47:59 CEST] <furq> also you don't have an audio stream so those audio options are pointless [11:48:07 CEST] <furq> beo: not if you use -c copy [11:48:49 CEST] <beo> right, ty [11:49:12 CEST] <stevenliu> maybe that is not important now bufsize, really CBR cannot toooo large than the bitrate you set, maybe no decoder or transport network care that, if you use DVB, that is useful [11:49:13 CEST] <rabbe> but i do get audio from the gopro, although not needed it can be nice to process it ok? [11:49:24 CEST] <stevenliu> if just TCP, that is no matter [11:50:55 CEST] <furq> rabbe: you'd need to capture the audio separately [11:51:30 CEST] <rabbe> but it's included in the hdmi signal, isn't it? [11:51:47 CEST] <furq> are you actually getting an audio stream [11:52:39 CEST] <rabbe> i think so [11:52:48 CEST] <rabbe> not able to test right now [11:52:56 CEST] <beo> what does "-safe 0" do exactly? [11:55:48 CEST] <rabbe> but this is your current suggestion? ffmpeg -r 60 -f v4l2 -video_size hd1080 -i /dev/video0 -c:a aac -ac 2 -b:a 128k -c:v libx264 -pix_fmt yuv420p -preset medium -tune zerolatency -x264-params "nal-hrd=cbr" -b:v 5000k -minrate 5000k -maxrate 5000k -bufsize 1600k -g 60 -bf 15 -r:v 30 -f flv rtmp://192.168.0.16/live/test [11:56:12 CEST] <rabbe> then record with: ffmpeg -i rtmp://localhost/live/test -c copy /usr/local/nginx/html/video/output.ts ; ffmpeg -i output.ts -c copy output.mp4; [11:57:06 CEST] <stevenliu> -bufsize 5000k [11:57:28 CEST] <stevenliu> if you don't care the bitrate large ~ small ~ large [11:58:07 CEST] <rabbe> i start recording in python by starting a process running "ffmpeg -i rtmp://localhost/live/test -c copy /usr/local/nginx/html/video/output.ts", then kill it and start a new process running "ffmpeg -i output.ts -c copy output.mp4;" [11:58:14 CEST] <rabbe> okay [11:58:19 CEST] <stevenliu> ok [11:58:53 CEST] <rabbe> at this moment i just want it to work [11:59:11 CEST] <rabbe> but customer is coming tomorrow, so there is some time to tweak [11:59:11 CEST] <stevenliu> It should work :D [11:59:15 CEST] <rabbe> :) [11:59:35 CEST] <stevenliu> inner peace :D [11:59:39 CEST] <rabbe> hehe [11:59:49 CEST] <rabbe> *reading my mantra* [11:59:55 CEST] <rabbe> now lunch [12:00:15 CEST] <stevenliu> now dinner :D [12:00:23 CEST] <rabbe> cheers :) [12:04:47 CEST] <beo> im getting "Invalid input file index 1 while processing metadata maps" on "ffmpeg -f concat -safe 0 -i mylist.txt -c copy -map_metadata 1 output.mp3" can someone tell me why? [12:08:10 CEST] <beo> it works with -map_metadata 0 and output file seems fine, but there are many of these warnings during concatenation -> https://puu.sh/xUy0o/20d0e55fc0.png [12:15:14 CEST] <beo> hm probably nothing to worry about [13:15:26 CEST] <rabbe> where can i use faststart? where i'm creating the flv over rtmp or where i'm recording the rtmp to .ts or when i convert .ts to mp4? [13:16:25 CEST] <rabbe> i would like to cut down the time before start recording the rtmp [13:19:11 CEST] <BtbN> movflags faststart are for mov/mp4 files. [13:19:28 CEST] <BtbN> And it has nothing to do with that. [13:19:33 CEST] <rabbe> ok [13:20:19 CEST] <rabbe> the ffmpeg command to start recording the rtmp, can it start faster after listing the codecs and stuff? [13:21:09 CEST] <BtbN> it will start as soon as it gets data, not sure what you mean. [13:21:27 CEST] <rabbe> ok [13:21:46 CEST] <BtbN> Why not just instruct nginx to record itself? [13:21:52 CEST] <BtbN> Putting ffmpeg into there seems unneccesary [13:22:35 CEST] <rabbe> because the rtmp created will be running all the time, whereas i only want to record at specific times [13:23:42 CEST] <BtbN> So use the controls it has to toggle the recording. [13:24:21 CEST] <rabbe> hm, okay.. can that convert the stream to some suitable format? [13:24:33 CEST] <BtbN> It will just dump the flv which rtmp essentially is [13:24:50 CEST] <rabbe> we do want to have some way of knowing if we're recording or not [13:25:37 CEST] <BtbN> https://github.com/arut/nginx-rtmp-module/wiki/Control-module [13:46:26 CEST] <beo> https://trac.ffmpeg.org/wiki/Concatenate how to avoid problems if filename in mylist.txt contains "'" ? [13:46:48 CEST] <BtbN> escape the ' [13:47:28 CEST] <BtbN> f = f.replace("'", "'\\''") [13:50:46 CEST] <beo> in a bat-file? [13:50:47 CEST] <beo> ok [13:53:34 CEST] <BtbN> That's python, but probably works similar in a lot of languages. [13:56:19 CEST] <rabbe> using this to get the rtmp: ffmpeg -i rtmp://192.168.0.16/live/test -c copy /usr/local/nginx/html/video/output.ts gets me error: https://pastebin.com/AzrymAru [15:02:33 CEST] <_julian> it appears my problem is caused by the aac_latm parser splits packets into smaller ones (probably audio frames), but does not insert pts on the new packets... [15:03:12 CEST] <_julian> I'm just wondering why, when find_stream_info was called the packets which are fed to the parser are already exactly at frame sizes and each has a pts. [15:14:45 CEST] <fristonio> hey there i am new to ffmpeg development community and would like to work on some beginner issues, any help would be appreciated [15:16:05 CEST] <DHE> fristonio: got any itches that need scratching? (as the saying goes) [15:16:15 CEST] <DHE> though first thing would be to read the contributing guide on the web site [15:21:23 CEST] <fristonio> I have gone through contributing guide, just need some beginner issues to work upon [15:26:46 CEST] <DHE> had any problems or complaints of your own? [17:11:16 CEST] <NapoleonWils0n> hi all [17:12:02 CEST] <NapoleonWils0n> im just wondering how the timeout option works [17:12:32 CEST] <NapoleonWils0n> is it possible to get ffmpeg to try record for 20 seconds before it quits [17:13:15 CEST] <NapoleonWils0n> trying to record stream that is a bit flaky and thought i could get ffmpeg to try recording for say 20 seconds before it gives up [17:26:47 CEST] <Alex__> I have a small inssue compling ffmpeg on centos 7.4 : Update for 3.2.8 or earlier commits, i am able to ./configure without an issue, allthough when i make i get the following error: [17:27:35 CEST] <Alex__> error: /usr/include/bits/mathdef.h:28:1: error: expected , or ; before typedef typedef float float_t; /* `float' expressions are evaluated as `float'. */ [17:29:03 CEST] <Alex__> gcc -I. -I./ -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -I./compat/float -I./compat/atomics/gcc -DZLIB_CONST -DHAVE_AV_CONFIG_H -std=c11 -fomit-frame-pointer -pthread -D_REENTRANT -I/usr/include/SDL2 -g -Wdeclaration-after-statement -Wall -Wdisabled-optimization -Wpointer-arith -Wredundant-decls -Wwrite-strings -Wtype-limits -Wundef -Wmissing-prototypes -Wno-pointer-to-in [17:29:20 CEST] <Alex__> -body -Wno-parentheses -Wno-switch -Wno-format-zero-length -Wno-pointer-sign -O3 -fno-math-errno -fno-signed-zeros -fno-tree-vectorize -Werror=format-security -Werror=implicit-function-declaration -Werror=missing-prototypes -Werror=return-type -Werror=vla -Wformat -fdiagnostics-color=auto -Wno-maybe-uninitialized -MMD -MF libavdevice/alldevices.d -MT libavdevice/alldevices.o -c -o libavdevice/alldevices.o libavdevice/alldevices.c [17:29:34 CEST] <Alex__> In file included from /usr/include/math.h:46:0, from ./libavutil/common.h:36, from ./libavutil/avutil.h:296, from ./libavutil/log.h:25, from libavdevice/avdevice.h:48, from libavdevice/alldevices.c:23: /usr/include/bits/mathdef.h:28:1: error: expected , or ; before typedef typedef float float_t; /* `float' expressions are evaluate [17:30:04 CEST] <Johnjay_> Okay I admit there is a use for pastebin [17:30:36 CEST] <Alex__> https://pastebin.com/ZgrAXmT1 [17:32:39 CEST] <Johnjay_> well Alex I compiled the latest snapshot just fine on msys2 [17:32:49 CEST] <Johnjay_> so maybe try that? [17:33:26 CEST] <Alex__> msys2? [17:33:44 CEST] <Johnjay_> linux build environment for windows [17:34:50 CEST] <Alex__> ah, i am tryin to get it to work on centos 7.4 [17:35:02 CEST] <Alex__> i dont have an issue compling on centos 7.3 [17:35:12 CEST] <Johnjay_> ok... so did you download the latest snapshot? [17:35:42 CEST] <Alex__> latest release 3.2 [17:37:16 CEST] <Alex__> i will try with https://github.com/FFmpeg/FFmpeg/tree/release/3.3 [17:41:18 CEST] <Alex__> nope same issue [17:42:52 CEST] <Johnjay_> well maybe furq or JEEB can figure it out then [17:43:16 CEST] <Johnjay_> what gcc version do you have at least? [17:43:40 CEST] <Alex__> one sec, rebooting system [17:44:06 CEST] <Alex__> gcc version 4.8.5 20150623 (Red Hat 4.8.5-16) (GCC) [18:28:01 CEST] <lethalwp> i'm trying to play an UHD-HEVC.mkv file with mpv on recent ffmpeg with open source amdgpu. But it always fallback to software decoding, do you know if there's a way to hardware decode yuv420p10? it worked last year before some yet unknown ffmpeg changes [18:28:26 CEST] <BtbN> Is that vaapi? [18:28:35 CEST] <lethalwp> tried vaapi & vdpau [18:28:48 CEST] <BtbN> I didn't know those even had software fallback [18:29:10 CEST] <lethalwp> it's mpv or ffmpeg that fallbacks on software decode [18:29:27 CEST] <BtbN> try with ffmpeg cli directly. [18:30:01 CEST] <BtbN> I'm not sure if vaapi and/or vdpau even support 10 bit at all though [18:30:16 CEST] <jkqxz> Support was only added to mesa at the beginning of this year, so it working before seems slightly unlikely. [18:30:38 CEST] <jkqxz> It should work now with VAAPI on GCN 4/5 (Polaris or Vega). [18:30:59 CEST] <jkqxz> If it doesn't, paste an ffmpeg log somewhere. [18:32:34 CEST] <lethalwp> last year i've added some trivial #HWVAAPI patch to ffmpeg and it worked. I now see they changed a lot of code. But i run llvm-git-copr. So trying to pinpoint why it doesn't work [18:42:55 CEST] <lethalwp> jkqxz: my actual ffmpeg command line to test is : # FFREPORT="level=32" ./ffmpeg -vaapi_device /dev/dri/renderD128 -i /home/lethalwp/hevcds_2160p60_Main10_20M.mp4 -f null - nothing in the logs shows if it uses hw decoding or cpu [18:43:08 CEST] <lethalwp> (it does open va libraries) [18:44:35 CEST] <jkqxz> That doesn't tell it to use the device for anything. [18:54:42 CEST] <lethalwp> jkqxz: i got it, in an earlier mpv version: hwdec=vaapi + vo opengl worked in hardware decode. Now it has to ba hwdec&vo=vaapi ... so it changed but still works :) [18:58:42 CEST] <jkqxz> -vo=vaapi is really a test output, though it's mostly usable if you don't mind colours being wrong. [19:01:09 CEST] <jkqxz> GL support via vaapi-copy works now, support via interop for Mesa is pending. [00:00:00 CEST] --- Wed Oct 11 2017

1 0

ffmpeg-devel.log.20171009
by burek 10 Oct '17

10 Oct '17

[01:50:41 CEST] <cone-284> ffmpeg 03Mark Thompson 07master:9c7a71145b88: configure: Add config option for libva2 (VAAPI 1) [01:50:42 CEST] <cone-284> ffmpeg 03Mark Thompson 07master:bd211bb866f8: vaapi: Remove H.264 baseline profile [01:50:43 CEST] <cone-284> ffmpeg 03Mark Thompson 07master:e339411691e3: vaapi: Always free parameter buffers after vaEndPicture() with libva2 [01:50:44 CEST] <cone-284> ffmpeg 03Mark Thompson 07master:5f3978866879: hwcontext_vaapi: Factorise out common connection code [01:50:45 CEST] <cone-284> ffmpeg 03Mark Thompson 07master:f3602875b325: hwcontext_vaapi: Set message callbacks on internally-created devices [01:50:46 CEST] <cone-284> ffmpeg 03Mark Thompson 07master:309d660775e2: hwcontext: Perform usual initialisation on derived device contexts [01:50:47 CEST] <cone-284> ffmpeg 03Mark Thompson 07master:b2f256a9f5db: hwcontext_vaapi: Add support for mapping to DRM objects [01:59:36 CEST] <cone-284> ffmpeg 03Mark Thompson 07master:c154c34c1b37: Revert "configure: Disable -Wbool-operation." [02:24:12 CEST] <cone-284> ffmpeg 03Carl Eugen Hoyos 07master:c585b4e9f0b3: configure: Disable -Wbool-operation using check_disable_warning(). [03:01:56 CEST] <cone-284> ffmpeg 03Carl Eugen Hoyos 07master:14b15539398f: lavf/adp: Fix the probe function on systems with signed char. [09:58:53 CEST] <durandal_170> how new ffmpeg release will be named? [13:35:45 CEST] <jkqxz> wm4: That's the encoder, not the decoder. And yes, it does in the decoder (unless you have the allow-profile-mismatch flag to override it). [13:37:53 CEST] <wm4> ooh [13:38:12 CEST] <wm4> that should be fine then [15:56:16 CEST] <cone-068> ffmpeg 03foo86 07master:23990950e35e: avcodec/dca_parser: revert to conservative sync distance estimation [15:56:16 CEST] <cone-068> ffmpeg 03foo86 07master:73789b85a759: avcodec/dca_core: always limit frame size to data size [17:06:48 CEST] <cone-068> ffmpeg 03bnnm 07master:1954e625b1d6: avcodec/wmaprodec: support multichannel XMA stream configurations [19:57:53 CEST] <BBB> wbs: http://www.bbc.com/news/technology-41551546 < does that mean the presentation @ VDD was futile? :( [19:59:10 CEST] <wm4> wasn't this long known [20:00:02 CEST] <jkqxz> Aren't we still expecting desktop-Windows-on-aarch64 real-soon-now? [20:00:06 CEST] <JEEB> ^ [20:00:24 CEST] <JEEB> I thought the presentation was mostly about the non-mobile aarch64 [20:00:35 CEST] <BBB> desktop-win-on-aarch64? [20:00:37 CEST] <BBB> is that a thing? [20:01:00 CEST] <JEEB> yea, binaries have already poked their head up [20:01:33 CEST] <JEEB> not a full image yet though, just additional binaries that wbs found :) and then Certain People seem to have SDKs [20:02:11 CEST] <JEEB> BBB: after all, they did already have an armv7 thing for quite a while [20:03:20 CEST] <jamrial> does amd still have plans for their arm stuff, or are they focused on ryzen? [20:04:45 CEST] <wm4> (was the ryzen bug with gcc compilation found yet?) [20:05:23 CEST] <wm4> would be nice to try amd CPU+GPU instead of the old intel+nvidia crap [20:05:34 CEST] <wm4> (nvidia still don't support wayland properly) [20:05:51 CEST] <BtbN> There was no Ryzen bug with gcc i think [20:05:52 CEST] <jkqxz> They haven't talked about it at all in the last year. I assume it's dead. [20:05:53 CEST] <BtbN> just broken chips [20:06:15 CEST] <jkqxz> (The AMD custom ARM core.) [20:06:16 CEST] <BtbN> jkqxz, they recently published a draft of the new API and had a whole conference about it. [20:06:46 CEST] <BtbN> wm4, now if only AMD would sell useable GPUs... [20:06:59 CEST] <wm4> I thought vega or something was good [20:07:03 CEST] <BtbN> But everything they offer is inferior to what nvidia has for the same price [20:07:06 CEST] <BtbN> lol, Vega is a joke [20:07:14 CEST] <wm4> what new API? [20:07:22 CEST] <jamrial> BtbN: how so? [20:07:24 CEST] <BtbN> Vega is more expensive than a faster nvidia card, and uses twice as much power. [20:08:10 CEST] <wm4> well, can't comment on that [20:08:12 CEST] <jamrial> i think i read that undervolting helped vega 56 in a lot of scenarios [20:08:17 CEST] <wm4> what does faster mean? game benchmarks? [20:08:31 CEST] <BtbN> Overall benchmarks [20:08:47 CEST] <wm4> nvidia almost surely have better GL drivers [20:08:48 CEST] <jkqxz> BtbN: New API for what, sorry? [20:08:51 CEST] <wm4> but fortunately vulkan exists [20:08:52 CEST] <BtbN> Wayland [20:09:07 CEST] <wm4> if you mean nvidia's eglstream idea for wayland, please no [20:09:08 CEST] <BtbN> The Vega64 is roughly somewhere between a GTX1070 and a GTX1080, but uses over 2 times the power a 1080 uses. [20:09:25 CEST] <atomnuker> that's why its a monster and I want one [20:09:40 CEST] <BtbN> just so you don't need a space heater? [20:09:47 CEST] <atomnuker> though I guess I could settle for an rx460 too, that thing is a monster too [20:09:50 CEST] <jamrial> it's not like you can buy a vega anyway. crypto currency miners made sure of that [20:10:01 CEST] <wm4> lol [20:10:30 CEST] <jkqxz> BtbN: For what, though? It's significantly better than that for compute (including mining, of course). [20:10:38 CEST] <wm4> btw. nvidia's power management is persistently broken [20:10:57 CEST] <wm4> I can't play video or play a game from 1996 without stuttering, unless I set the card to high perf mode [20:12:19 CEST] <jamrial> atomnuker: i wouldn't get a rx460, be it gaming or video decoding. amd starts being a good purchase with the rx470/570 [20:12:30 CEST] <jamrial> if you can get them at a decent price because, again, miners [20:15:05 CEST] <atomnuker> I like how amazon is ridiculously biased here, you search for rx570 and you get a few results with the rest being nvidia gtx 1050 or some other nvidia card [20:15:19 CEST] <atomnuker> but if you look for nvidia its pages and pages of nvidia without a single amd card [20:15:38 CEST] <BtbN> The algorithm favors what people actually end up buying [20:19:00 CEST] <atomnuker> true, most are windows users who only play minecraft and don't mind installing +1gb drivers with fancy guis [20:19:38 CEST] <BtbN> The Windows-Drivers for AMD and NVidia are equally horrible [20:19:50 CEST] <BtbN> I prefer the nvidia one, but they are both horribly overloaded [20:20:59 CEST] <atomnuker> I bet they look horrible too, just conservative spreadsheets of options, back in the day this was norma - http://img.hexus.net/v2/sff/biostar/200t/Egg.jpg [20:21:05 CEST] <atomnuker> *normal [20:22:04 CEST] <wm4> wat [20:22:21 CEST] <atomnuker> actually nevermind, http://www.guru3d.com/articles-pages/geforce-gtx-1080-overclocking-guide-wi… still shows overclock programs look crazy [20:22:34 CEST] <BtbN> Dealt with the AMD one at work recently. They are horrible. It took me over 10 minutes to find where to rotate the screen [20:22:56 CEST] <BtbN> Afterburner is a 3rd party tool. [20:22:57 CEST] <wm4> lol gamer shit [20:23:03 CEST] <jamrial> BtbN: best thing was when amd removed the old .net control panel in favor of the new one [20:23:18 CEST] <jamrial> 200mb off the installer shaved in one release [20:23:26 CEST] <nevcairiel> afterburner is the only tool that works properly however, but their new fancy skins are always dumb [20:24:25 CEST] <atomnuker> its a blank canvas for programmers to express their graphical design creativity, like this one here -http://i.imgur.com/9UGgO.jpg [20:25:46 CEST] <atomnuker> (though in 90% of the cases they end up copying car manufacturers because its too hard) [20:54:51 CEST] <wbs> BBB: as JEEB said, that should be unrelated. they're allegedly working on windows-on-arm64 for both server and laptop use [21:10:44 CEST] <wm4> arm servers? [21:10:50 CEST] <wbs> yes [21:11:00 CEST] <wm4> I thought everyone recognized that as nonsense [21:11:17 CEST] <wbs> that doesn't stop people from pouring money and effort into it [21:11:21 CEST] <JEEB> ^ [21:33:35 CEST] <kierank> wm4: what's wrong with arm servers [21:36:34 CEST] <jkqxz> Nothing intrinsically, but if they are going to put shitty cores running slow in the things then they better have fuckloads of them. This is a thing that ARM server vendors have historically failed at. [21:37:08 CEST] <jkqxz> (And is why Cavium is basically the only one worth considering from those currently around.) [21:37:59 CEST] <wm4> kierank: so far arm seems to have a good reputation for low energy use, but not exactly for being very powerful? [21:38:14 CEST] <kierank> sure, but ok for most webserving and crap, no [21:38:22 CEST] <wm4> and I don't think anyone ever tried to build servers with Atoms (when they were alive) [21:52:31 CEST] <nevcairiel> actually such things exist [21:53:07 CEST] <nevcairiel> they use Atom SoCs with high core count (16 or so) for NAS servers [21:53:28 CEST] <JEEB> yea, 8+ [21:53:43 CEST] <nevcairiel> not every server necessarily needs high cpu performance [21:54:09 CEST] <wm4> yeah, but there are ARM NASes too [21:54:31 CEST] <kierank> my nas is atom based [21:54:43 CEST] <JEEB> atom > ARM in various cases [21:54:53 CEST] <jkqxz> That's really just because of the rest of the things on the SoC. (E.g. <https://ark.intel.com/products/97927/> -> 16 SATA ports, 4x10GbE.) [21:55:24 CEST] <llogan> speaking of nas, is synology still distributing a modified, nonfree ffmpeg? [21:55:43 CEST] <TD-Linux> socionext has a new 24-core cortex-a53 board on 96boards [00:00:00 CEST] --- Tue Oct 10 2017

1 0

ffmpeg.log.20171009
by burek 10 Oct '17

10 Oct '17

[00:09:19 CEST] <kerio> how are you sending it to the tv? [00:09:40 CEST] <kerio> whoops that was way in the backscroll nvm [00:10:06 CEST] <JEEB> the TV I have supports both DLNA and miracast. not sure which the win10 built-in thing is using but probably miracast [00:10:42 CEST] <kerio> isn't miracast just screen mirroring [00:10:47 CEST] <JEEB> no [00:11:30 CEST] <JEEB> although I do remember that being one of the primary use cases for it when I was working at a company implementing it [00:12:35 CEST] <JEEB> (basically I'm pretty sure the "Cast to Device" thing in Win10 uses Miracast) [00:12:57 CEST] <JEEB> although it also supports the screen sharing via it [00:19:21 CEST] <kerio> yea but [00:19:32 CEST] <kerio> doesn't that mean that it's supposed to be transcoded? [00:24:57 CEST] <JEEB> no [00:25:08 CEST] <JEEB> I mean, what's going over the pipe is H.264 and pals anyways [00:38:41 CEST] <Djfe> hi, is it possible to tell ffmpeg to ignore 404s? I'm getting 404s while downloading an HLS over VPN and ffmpeg just skips to the next file of the playlist instead of retrying (at least it seems so). I know that 404s mean "the file is unavailable right now, but there might be something here available in the future" (in terms of HLS context) [00:45:35 CEST] <Djfe> it's used in a vod context and not a livestream,so the segments won't disappear [00:50:24 CEST] <DHE> no. that's broken. the file should be available, then the playlist is updated. vod is entirely more stringent [00:54:56 CEST] <Djfe> ok thx [00:56:01 CEST] <Djfe> bye [01:35:54 CEST] <bodqhrohro1> Are now any plans of OnTextData flv packet implementation? [02:57:59 CEST] <hendry> how do i quickly convert Matroska data to an mp4 that FCPX understands? [03:02:51 CEST] <tdr> quickly is such a relative word, the gods would say 1000 yrs wasnt so long :) [03:06:29 CEST] <DHE> I assume it means no transcoding. ffmpeg -i input.mkv -c copy output.mp4 [03:11:40 CEST] <Johnjay> question [03:11:52 CEST] <Johnjay> what's the smallest size mp4 file that would probably play in VLC/wmp? [03:11:54 CEST] <Johnjay> 200 KB? [03:12:26 CEST] <Johnjay> by play I mean something like show the first frame [03:27:31 CEST] <hendry> DHE: Could not find tag for codec pcm_s16le in stream #1, codec not currently supported in container <-- when i try `ffmpeg -i input.mkv -c copy output.mp4` [03:32:20 CEST] <relaxed> hendry: try output.mov [03:37:09 CEST] <hendry> relaxed: the resulting output.mov is just black in the import pane of FCPX [03:48:09 CEST] <DHE> then you need to transcode the audio. aac codec at 64k per channel (eg: 128k for stereo) as bitrate shoul dbe okay [03:49:46 CEST] <hendry> DHE: for MP4 ? [03:51:33 CEST] <hendry> DHE: ffmpeg -i input.mkv -c copy -acodec aac output.mp4 # doesn't work on FCPX. No audio and black [04:42:17 CEST] <FishPencil> Is there a way to combine a bunch of video thumbnails into one big image, say x images wide and y images high? [04:45:28 CEST] <hendry> FishPencil: probably with convert http://dabase.com/e/01046/ [04:47:27 CEST] <FishPencil> So extract the frames with FFmpeg and create a large image with imagemagick? I'm surprised FFmpeg can't do this [04:48:42 CEST] <furq> you can do it with ffmpeg [04:48:44 CEST] <furq> !filter tile [04:48:44 CEST] <nfobot> furq: http://ffmpeg.org/ffmpeg-filters.html#tile [04:51:52 CEST] <FishPencil> Nice, is it then possible to divide up the entire stream into n number of thumbnails, tiling them in the same go? [04:54:43 CEST] <furq> there's a thumbnail filter but you'll need to know the total number of frames in the file first [04:54:59 CEST] <furq> also it buffers the entire selection in memory so you probably want to decimate or scale that beforehand [04:55:05 CEST] <furq> it's generally not ideal [04:58:50 CEST] <FishPencil> for tile do I need to know how many frames, or can I just have it do a set width amount and keep going down till it's done? [07:20:11 CEST] <echelon> hey guys [07:20:32 CEST] <echelon> is there a way to distribute encoding across network? like distcc? [08:46:52 CEST] <frzs> Hi! I'm using ffprobe it to filter valid video files from a folder, but I found something strange: if I run the following command: " ffprobe.exe myimage.jpg " ffprobe will show " Duration: 00:00:00.04 " . [08:47:34 CEST] <frzs> The problem is that in my case this means a valid video file, as I'm checking if(duration > 0) for a valid video file [08:47:49 CEST] <e38383> 1 frame has this duration [08:48:28 CEST] <e38383> maybe check if there is a video part [09:14:35 CEST] <panpeter> good morning @ all :) i am totally new to ffmpeg (used gstreamer before). Nvidia says ffmpeg supports nvenc and nvdec completely. Im running ubuntu 16.04 with a gtx 750 (which supports at least second generation nvenc) if ill want to de- and encode an rtsp-stream in my own application, do i have to compile ffmpeg on my own or can i use ffmpeg from the default ubuntu repositories? how can i... [09:14:37 CEST] <panpeter> ...check nvenc is working correctly? [09:18:48 CEST] <stevenliu> ffmpeg -h encoder=h264 [09:20:43 CEST] <panpeter> stevenliu: this prints a wall of text (options) for the following encoders(?): libx264, libx264rgb [09:20:47 CEST] <Nacht> panpeter: If you run ffmpeg, you see in the banner what configuration its build with [09:20:53 CEST] <Nacht> panpeter: it should say --enable-nvenc [09:21:43 CEST] <panpeter> Nacht: so it seems the default ubuntu package does not contain it [09:22:23 CEST] <Nacht> panpeter: I'm not 100% sure tho. If I read the page about it, it does say the following. "ffmpeg configured without --disable-nvenc " [09:22:37 CEST] <stevenliu> look at the banner [09:22:38 CEST] <Nacht> So I'm not sure if enable-nvenc is on by default [09:22:55 CEST] <stevenliu> configuration: --enable-fontconfig --enable-gpl --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libspeex --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-version3 --cc='ccache gcc' --enable-nonfree [09:23:03 CEST] <stevenliu> for example, my ffmpeg banner is this [09:23:15 CEST] <Nacht> But simplest to test it with, just get a file and try to encode it [09:25:27 CEST] <panpeter> Yes, "--disable-nvenc" is not activated, but also "--enable-nvenc" is not activated, it seems as a default it is disabled [09:28:00 CEST] <panpeter> https://trac.ffmpeg.org/wiki/HWAccelIntro says if not --disable-nvenc is set, it will compiled with nvenc support :D [09:28:31 CEST] <panpeter> but also it seems i need the NVIDIA Video Codec SDK [09:30:25 CEST] <panpeter> Okay and i think shipping the NVIDIA Video Codec SDK would be a license flaw. So it seems nvidia doesn't want normal end-customers to have encoding and decoding capabilities used under linux systems [09:32:50 CEST] <Nacht> From what I've heard here, using the GPU doesn't really improve much. Not experianced it myself yet though [09:33:50 CEST] <panpeter> Nacht: ill do not need a rendered fps improvement, ill only need to hold the cpu free from rendering load [09:37:22 CEST] <stevenliu> use nvenc is better more than CPU [09:38:22 CEST] <Nacht> panpeter: I'm not sure how easy it is to get into the developer program. (ie: just signing up). But this document will help in the steps further: http://developer.download.nvidia.com/compute/redist/ffmpeg/1511-patch/FFMPE… [09:38:29 CEST] <stevenliu> speex is high than cpu, for exmaple, tesla K20 , speex is 20x , but cpu is 1x, 1080p 10M to 1080p 2.5M [09:39:54 CEST] <Nacht> panpeter: This might be intresting: http://git.videolan.org/?p=ffmpeg.git;a=commit;h=325e56479ff64c884f3bcccf92… [09:40:11 CEST] <Nacht> panpeter: Try the following: ffmpeg -encoders 2>/dev/null | grep nvenc [09:41:46 CEST] <panpeter> Nacht: it prints nothing, also "ffmpeg -h encoder=h264_nvenc" and "ffmpeg -h encoder=hevc_nvenc" deliver "Codec xyz is not recognized by FFmpeg." [09:42:04 CEST] <Nacht> Hmm. Doesn't seem you have the encoder installed then. [09:42:33 CEST] <Nacht> panpeter: According to this: https://askubuntu.com/questions/778100/how-to-install-compile-nvenc-in-ubun… [09:42:39 CEST] <Nacht> if you compile it yourself, you will have it [09:43:15 CEST] <panpeter> Nacht: the main problem is, ill have to ship a working hw-accelerated software to customers, so i think shipping the needed nvidia video codec SDK is restricted, but this may has to be checked by a lawyer [09:43:50 CEST] <Nacht> panpeter: Yeah, that's def worth a check then. Might even be worth signing up as a developer at nvidia then [09:44:28 CEST] <panpeter> Nacht: i already have done signed up as a developer ;) [09:45:13 CEST] <panpeter> Nacht: ah i found some thing that goes into the right direction http://git.videolan.org/?p=ffmpeg.git;a=commit;h=325e56479ff64c884f3bcccf92… [09:46:45 CEST] <panpeter> Nacht: maybe only my ffpmeg version is to old. I think i can check out the rest myself, thank you very much [13:05:18 CEST] <paradoxspiral> Is there an option when seeking with ms precision that will seek to the nearest frame? I'd like to seek to e.g. 1.86085. Or do I have to convert to the nearest frame myself? [13:06:45 CEST] <JEEB> do note that if you want frame-exact seeks then you want something like ffms2 [13:06:54 CEST] <JEEB> which creates indexes based on FFmpeg's libraries and decoders [13:07:19 CEST] <JEEB> because just seeking with av_seek_frame is not always frame-exact depending on the format. [13:07:43 CEST] <JEEB> so if you need frame-exact seeks and don't want to reinvent wheels then take a look at ffms2 [13:07:53 CEST] <JEEB> https://github.com/FFMS/ffms2 [13:07:57 CEST] <paradoxspiral> Well, I'd like to keep it more simple, since this is for a Lua mpv script. [13:08:20 CEST] <JEEB> well then you don't have access to the API at all [13:08:25 CEST] <JEEB> libavformat/-codec [13:08:26 CEST] <paradoxspiral> yeah :/ [13:08:35 CEST] <JEEB> I'm not sure what you were expecting to do [13:08:54 CEST] <paradoxspiral> I'm invoking ffmpeg via os.execute [13:09:11 CEST] <JEEB> because screenshots with mpv aren't possible or something? [13:09:30 CEST] <JEEB> I'd like to know the actual use case without the mpv lua script part in general. [13:10:05 CEST] <paradoxspiral> The script creates clips of the current video, and the start/end point is in the form of second.ms [13:10:23 CEST] <JEEB> do you need stream copy or do you always re-encode? [13:10:28 CEST] <paradoxspiral> re-encode [13:10:39 CEST] <JEEB> ok, then if you are OK with slowness you can use -ss after -i [13:10:43 CEST] <JEEB> which is slow but frame-exact [13:10:55 CEST] <JEEB> or well, it should be frame-exact :P (since it decodes A->B until that point) [13:11:33 CEST] <paradoxspiral> Does that re-encode the vid upto that point? [13:11:42 CEST] <JEEB> no, it just decodes [13:11:53 CEST] <paradoxspiral> then it's not tooo bad I gues [13:11:55 CEST] <paradoxspiral> +s [13:12:13 CEST] <JEEB> as in, what you really need is seek + decode, but in that case using ffms2 through some cli app/script would make more sense :P [13:12:28 CEST] <JEEB> so I guess "just wait until it decodes isn't /that/ bad" [13:13:37 CEST] <paradoxspiral> Yeah^^, thanks :3 [13:15:07 CEST] <JEEB> basically in the API of FFmpeg you can do av_seke_frame with the AVSEEK_FLAG_BACKWARDS , which should give you the closest random access picture from before the point. except when the format is not easily exactly seekable (say, MPEG-TS). of course you then have to decode until your point... at which point you're recreating parts of ffms2 :D [14:46:12 CEST] <MacroMan> Is there anything I can do to lower output file size, other than reducing the sample rate (just audio)? [14:46:29 CEST] <MacroMan> Sorry, bitrate, not sample rate [14:46:47 CEST] <Mavrik> Lower the bitrate and number of channels [14:46:55 CEST] <MacroMan> I'm already down to 32kbps, but any lower I lose audio quality. [14:47:10 CEST] <Mavrik> Which format? [14:47:13 CEST] <MacroMan> mp3 [14:47:23 CEST] <MacroMan> I think I'm at the floor limit, but I thought I'd ask [14:47:26 CEST] <Mavrik> Can you choose another one? [14:47:29 CEST] <Mavrik> Yeah, 32k is really low [14:47:42 CEST] <Mavrik> Although switching to HE-AACv2 will probably give you quite a boost in quality [14:49:54 CEST] <MacroMan> Mavrik: You'll have to give me a few minutes to read up on HE-AACv2 [14:50:00 CEST] <MacroMan> I have no idea what that is. [14:50:21 CEST] <JEEB> a profile for AAC made for low bit rate audio [14:50:41 CEST] <Mavrik> MacroMan: it's a high efficiency version of AAC made for <64kbit usecases [14:50:43 CEST] <JEEB> although personally if possible I'd utilize opus instead if there is no device support required [14:51:01 CEST] <Mavrik> It sounded significantly better for me than MP3/AAC-LC at those low bitrates and it's quite widely supported still [14:53:49 CEST] <MacroMan> Is there a command flag to set to HE-AACv2 or opus? [14:54:31 CEST] <MacroMan> Ah, think I've found it: `-c:a libfdk_aac -profile:a aac_he_v2` [14:58:12 CEST] <Mavrik> MacroMan: yep, that looks right [14:58:23 CEST] <Mavrik> you'll need to use HE-AAC (without v2) for mono content [14:58:51 CEST] <MacroMan> OK thanks. Do I have to compile ffmpeg myself for HE-AAC support? I'm getting this error: 'Unknown encoder 'libfdk_aac'' [14:59:30 CEST] <JEEB> yes, fdk-aac isn't compatible with FFmpeg's license and thus you can't distribute the two linked together in binary form [14:59:39 CEST] <Mavrik> Does the builtin AAC support he-aac? [14:59:46 CEST] <JEEB> no idea, I would guess not [14:59:51 CEST] <Mavrik> MacroMan: FDK is an AAC encoding library, similar to LAME for mp3 [14:59:52 CEST] <JEEB> atomnuker would know [14:59:55 CEST] <MacroMan> Damn. I didn't really want to get this deep into this project. [15:00:01 CEST] <JEEB> MacroMan: what's your client [15:00:02 CEST] <Mavrik> Then just use mp3 :) [15:00:06 CEST] <JEEB> as in what will play back the thing [15:00:20 CEST] <MacroMan> I'll stick with that for now, thanks [15:00:50 CEST] <atomnuker> use opus lol [15:01:40 CEST] <Mavrik> Yeah, use opus if target supports it :P [15:01:46 CEST] <Mavrik> Do mobile devices have decoders yet? [15:02:04 CEST] <MacroMan> It only needs to be supported in Chrome and windows media player [15:02:12 CEST] <MacroMan> I'll have a look at opus [15:04:49 CEST] <JEEB> chrome supports opus just fine, WMP probably not [15:08:29 CEST] <MacroMan> Unfortunately, the opus file isn't any smaller than mp3. I'll stick with that for now. Thanks for all the advice. [15:08:54 CEST] <JEEB> well if you were setting a low bit rate only the quality would be better [15:09:08 CEST] <JEEB> of course not the file size since you are defining the bit rate [15:09:15 CEST] <JEEB> bit rate over duration is the file size [15:37:26 CEST] <MacroMan> Wow, actually managed to lower the bitrate to 8kbps using opus, and it's still alright quality. [15:37:37 CEST] <redrabbit> sample? [15:37:40 CEST] <MacroMan> And about a 10th the file size of wav [15:37:54 CEST] <redrabbit> you mean 80k right [15:38:28 CEST] <MacroMan> These are voice recordings though, so the quality is not top priority. [15:38:35 CEST] <MacroMan> What ever 7.0287kbit/s is [15:38:43 CEST] <MacroMan> 7kbps? [15:39:24 CEST] <redrabbit> kbit/s =/= kbps [15:39:40 CEST] <MacroMan> Ah OK. Is that 70k then? [15:40:22 CEST] <redrabbit> im also confused now [15:40:22 CEST] <redrabbit> :p [15:41:18 CEST] <MacroMan> kbit is kilobit and kb is kilobyte, I think. [15:41:33 CEST] <MacroMan> Correct me if I'm wrong. [15:42:08 CEST] <redrabbit> https://en.wikipedia.org/wiki/Data_rate_units [15:42:55 CEST] <redrabbit> even more confusing lmao [15:43:01 CEST] <MacroMan> kilobit per second (symbol kbit/s or kb/s, often abbreviated "kbps") [15:43:16 CEST] <MacroMan> So Wikipedia are saying kbit/s === kbps [15:43:53 CEST] <MacroMan> Bytes is kB/s [15:44:43 CEST] <JEEB> that's a whole mess, just get used to kilobits (per second) when talking of bit rates [15:44:57 CEST] <JEEB> (and dividing/multiplying by 1000) [15:45:30 CEST] <MacroMan> lol. Whatever the correct symbol, I set the bitrate to 8 and it sounds fine, with an excellent file size. [15:46:32 CEST] <Nacht> B = byte b = bit [15:47:47 CEST] <redrabbit> 8kbps for audio sounds low [15:48:05 CEST] <redrabbit> wav is 1440k [15:48:43 CEST] <redrabbit> so maybe its really 8 kBps [15:48:58 CEST] <JEEB> if it's -b:a 8k that's 8 kilobits [15:49:03 CEST] <JEEB> and that's an OK bit rate for opus [15:49:11 CEST] <JEEB> for certain use cases [15:49:19 CEST] <MacroMan> From the man: Target bitrate in kbit/sec (6-256 per channel) [15:49:20 CEST] <JEEB> it won't keep the audio the same, but it will not be horrible horrible stuff [15:49:42 CEST] <redrabbit> can you up a small sample [15:49:56 CEST] <MacroMan> You'll need to give me a few minutes [15:50:09 CEST] <MacroMan> Most of the recordings are confidential [15:50:12 CEST] <redrabbit> JEEB: he's saying "about a 10th the file size of wav" [15:50:39 CEST] <JEEB> well you're thinking his source was of specific rate or bit depth [15:50:42 CEST] <JEEB> which it might not be [15:50:50 CEST] <MacroMan> Yes, opus: 0.8mb vs wav 15.1mb. [15:50:54 CEST] <redrabbit> yep i might be another bitrate and just mono [15:51:04 CEST] <JEEB> stop calling sample rate a bit rate [15:51:24 CEST] <MacroMan> The original file doesn't have a bitrate as it's wav. The sample rate was only 8000Hz though [15:51:31 CEST] <JEEB> there you have it :P [15:51:34 CEST] <redrabbit> ! [15:51:36 CEST] <redrabbit> ok [15:52:02 CEST] <JEEB> MacroMan: it does have a constant bit rate. sample rate * bits per sample * channel count [15:52:08 CEST] <MacroMan> These files are produced by Asterisk btw, which is super weird with the way it handles audio [15:52:36 CEST] <JEEB> but what redrabbit was mentioning as "bit rate" is not bit rate :P (sample rate by itself) [15:53:28 CEST] <MacroMan> JEEB: Didn't know that. Ubuntu tells me n/a on the bitrate for wav files for some reason. [15:54:03 CEST] <JEEB> well it's a constant amount of bits per second because it's not compressed, so it doesn't have a *compressed* bit rate as such :P [15:54:12 CEST] <JEEB> but it does have a bit rate according to the conten [15:54:13 CEST] <JEEB> *content [15:54:19 CEST] <JEEB> raw amount of bits [15:58:03 CEST] <MacroMan> Ah I see. Thanks. [16:24:12 CEST] <wouterstreamit> I am trying to emulate avcodec_decode_video2 using avcodec_receive_frame. Am I correct that I should expect AVERROR(EAGAIN) if there is no data available yet and should poll until there is data available? [17:57:54 CEST] <tomreyn> hi there. does ffmpeg already support the (proprietary) D(H)AV file format from Dahua (as used by their CCTV cameras)? [17:59:20 CEST] <tomreyn> I have ffmpeg 2.8.11 here (ubuntu's) but -formats does not suggest to support it. I came across GPLv4 software http://tanidvr.sourceforge.net/ which states to support it, though. [17:59:27 CEST] <tomreyn> *GPLv3 [17:59:55 CEST] <Johnjay> ok is there a simpler way to add silence to end of audio than this? [18:00:05 CEST] <Johnjay> I don't mean creating a temp file 10SecSilence.mp3 or whatever [18:00:07 CEST] <Johnjay> https://pastebin.com/UykBUzx0 [18:05:49 CEST] <durandal_1707> Johnjay: see apad filter [18:06:50 CEST] <Johnjay> thanks [18:09:30 CEST] <Johnjay> it wants samples, so would I do 48000 * N for N seconds of padding? [18:11:34 CEST] <Johnjay> I have a webm audio input and need a mp3 output with padding at the end [18:17:16 CEST] <durandal_1707> yes [21:17:48 CEST] <fsphil> is there a way to include subtitles in an MKV file but not have them display by default? [21:25:16 CEST] <Fenrirthviti> fsphil: Pretty sure that's a player setting, not something you can control in the source file. [21:32:59 CEST] <CoreX> fsphil if i remember its -disposition:s:0 0 [21:33:14 CEST] <fsphil> trying that now [21:38:13 CEST] <fsphil> no, no difference [21:43:59 CEST] <nostrora> Google use ffmpeg for Youtube ? [21:44:14 CEST] <durandal_1707> no [21:44:40 CEST] <nostrora> there is an parameter i can add to ffmpeg to delete input file if encoding is correcty done ? [21:44:43 CEST] <JEEB> you forgot them patching main() ? [21:45:05 CEST] <JEEB> also how GOOG is now trying to send in all of their patches :P [21:45:23 CEST] <durandal_1707> whut.? [21:45:39 CEST] <relaxed> didn't they use a custom mencoder at on point? [21:45:42 CEST] <JEEB> yes [21:45:50 CEST] <relaxed> one* [21:45:51 CEST] <JEEB> then they were basically renaming ffmpeg.c's main() [21:45:55 CEST] <JEEB> and calling it from their app [21:45:59 CEST] <JEEB> and/or library [21:53:51 CEST] <nostrora> there is an parameter i can add to ffmpeg to delete input file if encoding is correcty done ? [21:54:10 CEST] <JEEB> no [21:54:24 CEST] <JEEB> use ffprobe with json output, decide by yourself [21:56:11 CEST] <nostrora> ok :) [23:43:42 CEST] <rabbe> anyone have a magewell capture card (1080P @ 60FPS -> USB3)? which ffmpeg command for saving mp4 to hdd will give me max quality? [23:44:48 CEST] <dystopia_> capturing as raw video will give you best quality [23:44:54 CEST] <dystopia_> but you probably don't want to do that [23:45:41 CEST] <rabbe> no, probably h.264+aac .mp4 is best? [23:46:04 CEST] <rabbe> i mean, best quality per reasonable size [23:46:46 CEST] <rabbe> or mkv? [23:47:00 CEST] <dystopia_> mkv and mp4 are just containers [23:47:06 CEST] <dystopia_> what is it you are capturing? [23:47:11 CEST] <dystopia_> tv, video games? [23:47:26 CEST] <rabbe> yeah, so lets go with mp4. i think the customer will like that most [23:47:37 CEST] <rabbe> it will be from a gopro mounted on a robot arm [23:48:02 CEST] <rabbe> but the signal will go through a whdi link [23:48:14 CEST] <llogan> how will the customer use these files? just for viewing, or uploading to YouTube, or editing, etc. [23:48:35 CEST] <rabbe> just for viewing [23:49:09 CEST] <llogan> i'm guessing you want something like: ffmpeg -i input -c:v libx264 -crf 18 -vf format=yuv420p -metadata title="Robot Video" output.mp4 [23:49:29 CEST] <llogan> see https://trac.ffmpeg.org/wiki/Encode/H.264 [23:49:37 CEST] <rabbe> alright [23:49:49 CEST] <rabbe> thanks [23:50:51 CEST] <rabbe> i did some testing and then i needed to tell the resolution [23:51:30 CEST] <rabbe> but i did some check on magewells web page now and found they have some software to set up the resolution etc [23:51:54 CEST] <rabbe> (i mean i had to specify in the ffmpeg command about the resolution) [00:00:00 CEST] --- Tue Oct 10 2017

1 0

ffmpeg-devel.log.20171008
by burek 09 Oct '17

09 Oct '17

[01:14:07 CEST] <cone-493> ffmpeg 03Carl Eugen Hoyos 07master:d96d65dfebdd: lavu/utils: Use "__asm__" like everywhere else in the codebase. [13:37:34 CEST] <ubitux> why is libopenjpg such a mess... T_T [13:38:01 CEST] <JEEB> is that about the patch to add support for latest upstream or something? [13:38:07 CEST] <JEEB> also didn't it have that AGPL fork thing? [13:38:08 CEST] <ubitux> yeah [13:38:27 CEST] <ubitux> like, everytime there is a new release we have to add a ton of crazy check to the configure [13:38:31 CEST] <BtbN> did not break API or ABI, still renamed everything. [13:38:32 CEST] <ubitux> wtf is going on [13:38:36 CEST] <JEEB> lol [13:38:47 CEST] <ubitux> it's like it's an entire new project after every release [13:38:55 CEST] <ubitux> even though that's the same shit [13:39:15 CEST] <ubitux> -#if HAVE_OPENJPEG_2_2_OPENJPEG_H || HAVE_OPENJPEG_2_1_OPENJPEG_H || HAVE_OPENJPEG_2_0_OPENJPEG_H [13:39:17 CEST] <ubitux> +#if HAVE_OPENJPEG_2_3_OPENJPEG_H || HAVE_OPENJPEG_2_2_OPENJPEG_H || HAVE_OPENJPEG_2_1_OPENJPEG_H || HAVE_OPENJPEG_2_0_OPENJPEG_H [13:39:21 CEST] <ubitux> how long is this gonna go? [13:39:24 CEST] <JEEB> :| [13:39:47 CEST] <BtbN> have configure auto-generate it based on what version it finds [13:39:58 CEST] <BtbN> and only code in actual api differences [13:40:08 CEST] <ubitux> have you seen the libopenjpeg check? [13:40:17 CEST] <BtbN> yep [14:40:05 CEST] <RiCON> ubitux: with the removal of openjpeg 1.x after 3.4 it can probably get simpler [14:52:28 CEST] <jkqxz> On the comment that the new libopenjpeg "contains many security-related fixes": should we infer from that that the old versions are unmaintained with known security issues and should therefore be dropped? [14:53:08 CEST] <JEEB> sounds like it at least [17:13:58 CEST] <cone-630> ffmpeg 03Michael Bradshaw 07master:41d6d6270243: lavc: add support for OpenJPEG 2.3.0 [17:14:41 CEST] <ubitux> ... ? [17:16:13 CEST] <durandal_1707> wut? [21:23:32 CEST] <relaxed> 3.4 will support libopenjpeg 2.3.0 ? [21:24:07 CEST] <jamrial> judging by the latest commit, yes [22:03:22 CEST] <nevcairiel> jkqxz: considering recent talk on the subject, did you consider a CBS filter for h264 to re-add x264 version metadata? [22:04:52 CEST] <nevcairiel> although perhaps one doesnt even need CBS for that, its not filtering an existing bitstream as such, just adding a new SEI [22:06:14 CEST] <jamrial> nevcairiel: the existing h264 metadata bsf can do that already [22:06:24 CEST] <nevcairiel> oh [22:06:41 CEST] <jamrial> the sei_user_data option adds an unregistered user data SEI with an arbitrary string [22:07:05 CEST] <nevcairiel> oh yeah, just a bit hard to use [22:07:10 CEST] <jkqxz> Yeah, you just want 'h264_metadata=sei_user_data=x264-uuid-which-is-long+x264 - core 42'. [22:07:19 CEST] <jkqxz> Could be added to the docs? [22:08:10 CEST] <jamrial> jkqxz: going to send a new version of the cbs set, or just push the one in the ml? [22:08:25 CEST] <jamrial> were there outstanding changes that need a new revision? [22:08:35 CEST] <nevcairiel> he just posted a new one [22:08:39 CEST] <nevcairiel> hence why I remembered to ask [22:08:49 CEST] <jamrial> oh [22:08:53 CEST] <jamrial> my bad :p [22:09:01 CEST] <jkqxz> Though note that it doesn't directly work on the sample in whatever the bug is on trac, because it only inserts the SEI in a unit with an SPS and the extradata isn't inline. [22:09:37 CEST] <jkqxz> Not sure what to do to fix that. Dumping to a raw stream and reading back gives you the right extradata and then it works. [22:10:08 CEST] <jkqxz> Or it could just unconditionally insert it in the first access unit, maybe. [22:10:27 CEST] <nevcairiel> thats where the x264 data would be anyway [22:10:33 CEST] <nevcairiel> and only there, its not ever repeated later [22:12:28 CEST] <jamrial> h264 sei messages are never in extradata anyway. they are not in avcc and we don't extract them from raw streams into extradata either [22:14:51 CEST] <jkqxz> Oh, libx264 in lavc actually has code to extract that sei from the headers. [22:20:57 CEST] <jkqxz> Yeah, always stuffing it in the first access unit is probably the best answer. [22:28:20 CEST] <jkqxz> Yeah, that works nicely with the file on trac. [22:29:21 CEST] <jamrial> that'd make a good fate test, btw [22:30:07 CEST] <jamrial> for both the metadata filter and the backwards compat code for these old files [22:33:04 CEST] <jkqxz> Do we have a test in fate already for that? [22:35:17 CEST] <jamrial> don't think so, the whole compat code for these files is pretty recent [22:36:00 CEST] <jkqxz> I mean, I thought one was added with it? That would then want the SEI-stripping BSF to make the input, and I'm not sure I want to make that... [22:36:13 CEST] <jamrial> http://coverage.ffmpeg.org/libavcodec/h264_cabac.c.gcov.html or maybe we do [22:36:23 CEST] <jkqxz> (Tools which can only be used for evil, and all that.) [22:36:32 CEST] <Luigi12> michaelni: not sure if you'd heard about this http://openwall.com/lists/oss-security/2017/10/02/1 but just letting you know if not [22:36:40 CEST] <Luigi12> sorry for the double-ping, just moving this to the -devel channel where it belongs [22:37:09 CEST] <jamrial> jkqxz: no, i mean, adding a sample encoded with old x264 that had the sei stripped, like in that ticket, then using the metadata filter to add a fake x264 sei so it decodes [22:37:56 CEST] <JEEB> Luigi12: you might want to note what that is in general [22:38:06 CEST] <jkqxz> I was thinking if there already is one then we could remove the SEI from it programmatically and avoid adding another sample. [22:38:07 CEST] <cone-284> ffmpeg 03Marton Balint 07master:41569bbc66d8: ffmpeg: always use single threaded decoding for attached pictures [22:38:13 CEST] <Luigi12> JEEB: Internet Bug Bounty project. They were going to include ffmpeg, but they needed to hear from upstream before OK'ing it. [22:38:25 CEST] <JEEB> well that still doesn't explain what that is [22:38:31 CEST] <JEEB> and what are the requirements from the project [22:38:38 CEST] <JEEB> and/or anything else like that [22:39:05 CEST] <jamrial> jkqxz: only if changing the sample doesn't breaks fate with old releases [22:39:19 CEST] <jamrial> they all use the same rsync samples suite [22:39:26 CEST] <Luigi12> JEEB: you really need to read the mailing list thread for full context [22:39:34 CEST] <Luigi12> it's not that long [22:39:53 CEST] <jkqxz> I can't find a test. Maybe there actually isn't one. [22:40:25 CEST] <jamrial> there is a sample, though, judging by the coverage link above [22:41:02 CEST] <Luigi12> JEEB: short version is it's a project that pays bounties for security bugs with a PoC, but only for approved projects. They'd like to add ffmpeg, but need to hear from upstream first. [22:41:31 CEST] <Luigi12> they've already added libav, but obviously it'd be much more useful for people to audit ffmpeg [22:41:34 CEST] <JEEB> well that part I kind of get without going further but the reason why they'd have to have explicit approval from the OSS project is interesting [22:41:45 CEST] <JEEB> as in, what is there that might not be wanted by a project [22:41:52 CEST] <jkqxz> That suggests that there is only a bad sample, and no good one? [22:42:31 CEST] <jamrial> yeah [22:42:46 CEST] <jkqxz> (And is that test really correct? x264_build is -1 by default, which will invoke the bad path.) [22:43:37 CEST] <Luigi12> JEEB: they don't explicitly address that question in the thread, but it sounds like they wanted to be sure that the project is interested in and prepared to deal with reports of security issues [22:43:40 CEST] <jkqxz> Hmm, maybe it's a random conformance sample, and the decode is just wrong. [22:43:49 CEST] <jamrial> jkqxz: i think that's why it checks for 151U [22:44:19 CEST] <Luigi12> I know that FFmpeg has always been good about handling security issues in the past, but I guess they don't want to assume (since they're shelling out money) [22:44:19 CEST] <jkqxz> Oh, integer promotion. That's totally opaque. [22:45:17 CEST] <jamrial> wouldn't be libavcodec otherwise :P [22:59:08 CEST] <cone-284> ffmpeg 03Carl Eugen Hoyos 07master:f49c129dd494: lavd/decklink_dec: Do not claim to output transparency information. [23:09:02 CEST] <cone-284> ffmpeg 03Carl Eugen Hoyos 07master:c2d155e11ee5: configure: Disable -Wbool-operation. [23:16:30 CEST] <cone-284> ffmpeg 03Carl Eugen Hoyos 07master:cd01b3cbcf3b: lavu/opt: Use "&&" instead of "*" in boolean expression. [23:36:16 CEST] <michaelni> Luigi12, ive found the mail and replied, yes we of course we want to be part of bug bounty stuff [23:47:44 CEST] <cone-284> ffmpeg 03James Almer 07master:65c3a32836f4: configure: disable libxcb dependent features if libxcb is not enabled [00:00:00 CEST] --- Mon Oct 9 2017

1 0

ffmpeg.log.20171008
by burek 09 Oct '17

09 Oct '17

[00:02:57 CEST] <kiroma> I hope the SDK didn't overwrite any file... [00:09:43 CEST] <kiroma> I deleted every file installed by the sdk but it's still throwing the error [00:09:51 CEST] <kiroma> and I've still got 2 render devices [00:12:23 CEST] <jkqxz> Did you remove the extra environment variables from your shell config? [00:12:57 CEST] <kiroma> Oh wait, I inspected install script ant there are some files left over in kernel modules. [00:25:44 CEST] <kiroma> can you use hwaccel in ffplay? [00:38:18 CEST] <Johnjay> can ffmpeg to told to cut at a keyframe when extracting audio? [00:38:35 CEST] <Johnjay> i.e. something like "go to 1:47 in the video and then seek to the closest key frame" [00:50:47 CEST] <Johnjay> What does mean Application provided invalid, non monotonically increas ing dts to muxer in stream 0? [00:52:09 CEST] <klaxa> dts means decoding timestamp (as opposed to pts, presentation timestamp), the dts should obviously not jump around [00:53:07 CEST] <Johnjay> I created 2 mp3 files and then concated several copies of them together with -f concat [00:53:13 CEST] <Johnjay> when -c copy didn't work I changed it to -c mp3 [00:53:47 CEST] <klaxa> huh [00:54:48 CEST] <klaxa> well what's your complete command line + output? (use pastebin or something) [00:55:56 CEST] <Johnjay> it's not long enough for that [00:56:03 CEST] <Johnjay> ffmpeg -f concat -safe 0 -i list1.txt test2.mp3 [00:56:14 CEST] <klaxa> the output sure is long enough for pastebin [00:56:15 CEST] <Johnjay> It must be something to do with the way i created the 2 files maybe? idk [00:56:35 CEST] <Johnjay> oh lol [00:56:42 CEST] <Johnjay> my command window didn't get all of it. [00:56:46 CEST] <Johnjay> just that same error over and over [00:57:26 CEST] <Johnjay> https://pastebin.com/rTZXdu1B [00:58:11 CEST] <klaxa> hmm... does the resulting file play though? [00:58:30 CEST] <klaxa> as it seems it completed without an actual error [01:01:36 CEST] <Johnjay> yeah [01:01:58 CEST] <Johnjay> I used anullsrc for the first file and I clipped the second file [01:02:01 CEST] <Johnjay> So it's just a warning? [01:02:05 CEST] <Johnjay> maybe to do with vbr mp3? [01:02:21 CEST] <Johnjay> since it failed to produce a usable file with -c copy [01:02:38 CEST] <Johnjay> it was too short and didn't have the second file in it [01:03:00 CEST] <klaxa> shouldn't have to do with vbr [01:03:37 CEST] <klaxa> ah, but you probably can't randomly concat mp3s with different encodings maybe? although that is specified on a frame basis... hmm... [01:03:48 CEST] <Johnjay> i can't just randomly concat mp3s together? [01:03:58 CEST] <klaxa> well you should be able to [01:04:06 CEST] <BtbN> mp3 is special and wonky [01:04:24 CEST] <klaxa> it would be a problem if one is mono and one is stereo for starters [01:04:48 CEST] <klaxa> i think? [01:05:09 CEST] <Johnjay> they're both stereo [01:05:21 CEST] <Johnjay> one's 48k and the other is 44k [01:05:24 CEST] <klaxa> i was talking about the general case [01:05:34 CEST] <klaxa> ah, that could also already make it a problem [01:05:57 CEST] <Johnjay> the silence anullsrc generated was 44k [01:06:04 CEST] <Johnjay> the downloaded file I split up is 48khz [01:06:09 CEST] <klaxa> although i don't know, like i said, as far as i know mp3 negotiates all of this information in each mp3 frame [01:08:18 CEST] <Johnjay> I redid the silence file as 48k and reran the concat command. still gives that error [01:09:05 CEST] <Johnjay> both s16p, 128kb/s,stereo 48k [01:14:57 CEST] <Johnjay> hmm [01:15:02 CEST] <Johnjay> there might be a simpler way to do what i'm doing [01:15:26 CEST] <Johnjay> can I tell ffmpeg to take a file F and at times t1,t2,t3,...,tn insert some silence? [01:28:59 CEST] <Johnjay> I guess I'll have to split it up then use concat filter [01:33:24 CEST] <JEEB> furq: for your reference I had no idea there was a filter to copy the thing to proper alignment [01:33:31 CEST] <furq> lol oh [01:33:38 CEST] <furq> well yeah that worked fine and didn't seem particularly slow [01:36:13 CEST] <JEEB> yea, it shouldn't really [01:36:22 CEST] <JEEB> if vapoursynth was able to just do it for cropping as well [01:51:35 CEST] <Johnjay> furq can I do that insertion thing? [01:51:44 CEST] <Johnjay> or do I have to split and concat ? [02:02:50 CEST] <jerome-> I guess it should be possible to use the output of a ffserver as an input for fffmpeg? [02:04:02 CEST] <JEEB> you'd be surprised if someone here knew about ffserver [02:04:14 CEST] <jerome-> :) [02:05:11 CEST] <jerome-> well, I tried, but ffmpeg gets stuck and doesn't give any output [02:08:35 CEST] <jerome-> and when I add the input format, then ffmpeg is not happy: http://uu.zoy.org/p/8vLXqw#x=lIJWAKgxDgBcjsQA [02:37:12 CEST] <Asher_> there is a way to render subtitle on video stream from subtitle stream srt embeded in same mkv ? [02:46:15 CEST] <c_14> subtitles filter [02:46:55 CEST] <c_14> https://trac.ffmpeg.org/wiki/HowToBurnSubtitlesIntoVideo [02:46:57 CEST] <Asher_> Hello , i tryed but i have 2 problems . [02:48:57 CEST] <Asher_> First my mkv source is an HTTP Link . because it's used to do on the fly transcoding . and subtitle don't take http src link ? and second i tryed subtitle with local file it's working , but the subtitle are not sync time ( they start from 0 ) insteed of starting at a time ( -ss ) . [02:49:29 CEST] <Asher_> any way ? [02:50:02 CEST] <c_14> The first one, no (sadly). The second, if you put -ss after the input filename or I think -copyts works as well [02:50:37 CEST] <Asher_> oh ok [02:50:55 CEST] <c_14> you need a file input for the subtitles filter because libavfilter still doesn't support subtitle streams [02:51:21 CEST] <Asher_> i tryed to add after the input filename , but if i start like 10 min after the start of the movie i need to wait some time before the movie start [02:51:30 CEST] <Asher_> and about the first problem [02:51:36 CEST] <Asher_> There is another way ? [02:51:43 CEST] <Asher_> or no way at all ? [02:51:44 CEST] <c_14> yeah, try -copyts can't remember if it works or not [02:51:57 CEST] <c_14> No, the only thing you can do is download the http stream into a file and use that as input [02:52:09 CEST] <Asher_> hum i see [02:52:15 CEST] <Asher_> :/ [02:52:20 CEST] <c_14> well [02:52:32 CEST] <c_14> You can try feeding the stream url into the subtitles filter as a file= [02:52:49 CEST] <c_14> It should probably just call avopen2 so it might work [02:53:23 CEST] <Asher_> hum like file=http://xxxxx.mkv ? [02:53:35 CEST] <c_14> yeah, -vf subtitles=file=http:// [02:53:59 CEST] <c_14> eh, it might be filename= [02:56:06 CEST] <Asher_> [subtitles @ 0x1afc2a0] Unable to parse option value "//192.168.1.34" as image size [02:56:22 CEST] <c_14> put it in single quotes [02:56:27 CEST] <c_14> filename='http://' [02:59:33 CEST] <Asher_> ok [02:59:35 CEST] <Asher_> trying [03:19:32 CEST] <Asher_> @c_14 [03:19:37 CEST] <Asher_> the single cote don't work also [03:20:21 CEST] <c_14> error? [03:21:21 CEST] <Asher_> [subtitles @ 0x182bfe0] Unable to parse option value "//192.168.1.34" as image size Last message repeated 1 times [subtitles @ 0x182bfe0] Error setting option original_size to value //192.168.1.34. [Parsed_subtitles_0 @ 0x17b52a0] Error applying options to the filter. [AVFilterGraph @ 0x1821f00] Error initializing filter 'subtitles' with args 'filename=http://192.168.1.34:81/test.mkv' [03:21:38 CEST] <c_14> try a \ before the : [03:23:16 CEST] <Asher_> [subtitles @ 0x13dffe0] Unable to parse option value "//192.168.1.34" as image size Last message repeated 1 times [subtitles @ 0x13dffe0] Error setting option original_size to value //192.168.1.34. [Parsed_subtitles_0 @ 0x13692a0] Error applying options to the filter. [AVFilterGraph @ 0x13d5f00] Error initializing filter 'subtitles' with args 'filename=http://192.168.1.34:81/test.mkv' Error opening filters! [03:23:25 CEST] <Asher_> same error [03:23:36 CEST] <c_14> then I'm out of ideas [03:24:43 CEST] <Asher_> i wanted to do something like with the stream [03:25:12 CEST] <Asher_> :/ [03:25:27 CEST] <Asher_> something with the stream* [03:25:39 CEST] <Asher_> but i don't think subtitles take http link [03:26:15 CEST] <Asher_> c_14 , do you know another way to make the subtitle in the video stream in live ? [03:26:56 CEST] <c_14> not really, no [03:27:11 CEST] <Asher_> :( , or ffmpeg alternative who can do this ? [03:31:05 CEST] <Asher_> c_14 [03:31:07 CEST] <Asher_> seems i got it [03:31:18 CEST] <Asher_> escaped with \\before all : [03:31:21 CEST] <Asher_> did the work [05:32:05 CEST] <Johnjay> hmm [05:32:09 CEST] <Johnjay> i might be being dumb [05:32:25 CEST] <Johnjay> but is there a way to insert a sound or file at certain times? [05:32:31 CEST] <Johnjay> instead of splitting the file and re-concatting it? [06:19:02 CEST] <postmodern> i noticed ffmpeg has zmq support. i was wondering if there was a way to change the -vf value in real-time, for live video effects? [06:26:26 CEST] <furq> postmodern: that's pretty much all you can do with it afaik [06:26:33 CEST] <furq> !filter zmq @postmodern [06:26:33 CEST] <nfobot> postmodern: http://ffmpeg.org/ffmpeg-filters.html#zmq_002c-azmq [06:34:12 CEST] <postmodern> eggsellent [06:38:53 CEST] <postmodern> furq, is there a list of al TARGETs? Or do they all start with the Parsed_ prefix? [06:54:37 CEST] <postmodern> furq, can you change the entire filter string and have that re-parsed? [09:50:22 CEST] <rabbe> ok, i've got ffmpeg grabbing my webcam, sending flv over rtmp to nginx which creates mpeg dash that i receive in my browser. how do i reduce latency, at the web server or ffmpeg? ffmpeg -f v4l2 -i /dev/video1 -c:a aac -ac 2 -b:a 128k -c:v libx264 -pix_fmt yuv420p -profile:v baseline -preset ultrafast -tune zerolatency -vsync cfr -x264-params "nal-hrd=cbr" -b:v 5000k -minrate 5000k -maxrate 5000k -bufsize 10000k -g 60 -s [09:50:22 CEST] <rabbe> 1920x1080 -f flv rtmp://localhost/live/test [12:58:35 CEST] <Kiicki> I'm curious about something. Handbrake got this 2 pass and turbo option when doing average bitrate. What would that be equivalent to with meGUI? Handbrake on left, meGUI on right https://i.imgur.com/iYQNrVQ.png [13:01:14 CEST] <DHE> not really an ffmpeg question. I'd suggest automatic 2pass [14:07:08 CEST] <nostrora> Hello! I'm new with FFMPEG and video encoding in general. i want to encode MP4 to webm(vp9+opus) the source file have 5M v bitrate. i have to set bitrate to 5M ? my question is. xM bitrate in mp4 is equivalent of xM bitrate in vp9 ? [14:08:43 CEST] <JEEB> when you re-encode you can forget the input bit rates altogether [14:08:59 CEST] <JEEB> and no, first of all you're comparing container to a video format [14:09:04 CEST] <JEEB> mp4 can contain a wide range of stuff [14:09:16 CEST] <nostrora> JEEB: i mean, h264 sorry [14:09:20 CEST] <JEEB> and whomever made your input clip could either have encoded it with good compression or bad compression [14:09:25 CEST] <JEEB> or something in the middle [14:09:37 CEST] <JEEB> so really, just stop thinking of your input file's file size or bit rate [14:09:44 CEST] <JEEB> it's irrelevant when you decode the pictures [14:09:53 CEST] <JEEB> what matters is if your output file looks good enough [14:10:01 CEST] <nostrora> JEEB: i talking about output video bitrate [14:10:10 CEST] <JEEB> yes, but you were comparing it to the input one [14:10:30 CEST] <JEEB> and I was saying "you don't know how the input was encoded and many other things, so the output vs input doesn't matter" [14:10:48 CEST] <JEEB> it generally just doesn't, what matters is that you encode something that it looks good enough to you on the output side [14:11:07 CEST] <JEEB> I recommend using whatever is mapped to the -crf option and tweak that with whatever speed-related options there are for libvpx-vp9 [14:11:17 CEST] <nostrora> JEEB: ok, i don't know how to choice the output bitrate for VP9 [14:11:24 CEST] <JEEB> that's why you don't [14:11:27 CEST] <e38383> overall vp9 should have a slightly better encoding as h264, if you have the same bitrate it should be fairly equal (just my opinion and in no way is anything JEEB just said wrong) [14:11:44 CEST] <JEEB> you generally don't set a pre-defined bit rate if you are not following a limitation [14:11:55 CEST] <JEEB> what you want is a certain quality level instead [14:12:15 CEST] <JEEB> e38383: that is way too generalized [14:12:28 CEST] <JEEB> I mean, you and me or him cannot know how well compressed that H.264 clip was [14:12:46 CEST] <JEEB> so we cannnot know if compared to the input bit rate how well libvpx's vp9 encoder (at which settings?!) could compress it [14:12:59 CEST] <nostrora> better solution i to just don't set output bitrate ? [14:13:11 CEST] <e38383> JEEB: true, but it might give a general hint in what direction it will go [14:13:14 CEST] <e38383> nostrora: definitly [14:13:54 CEST] <JEEB> nostrora: drop the -b:v to zero and otherwise try these https://developers.google.com/media/vp9/settings/vod/ [14:14:03 CEST] <JEEB> it says to use whatever is mapped to the misnomer of "crf" [14:14:13 CEST] <JEEB> and then it sets -quality good -speed 0 [14:14:23 CEST] <JEEB> those seem to be the speed VS compression parameters [14:14:31 CEST] <JEEB> and of course if you don't want to rescale, remove the vf scale [14:15:11 CEST] <JEEB> nostrora: and then you tweak the CRF parameter up/down to change the output quality [14:15:32 CEST] <JEEB> with x264 it was up for less bits and down for more bits [14:16:57 CEST] <nostrora> can we preview the current quality setting without encode all clip ? (because it's take long time) [14:17:04 CEST] <JEEB> yes [14:17:06 CEST] <nostrora> like a screenshot or somethink like, just a frame [14:17:17 CEST] <JEEB> well usually you want to limit to like 1200 frames of content [14:17:19 CEST] <JEEB> and play it through [14:17:23 CEST] <JEEB> 1200-2500 frames [14:17:24 CEST] <nostrora> jeeb oh cool :) [14:17:42 CEST] <JEEB> -t 105 I think is 105 seconds [14:17:55 CEST] <JEEB> or -vframes 2500 for exact frame count [14:18:03 CEST] <nostrora> Is that true CPU encoding is better than GPU ? [14:18:19 CEST] <JEEB> GPU is not really GPU it's the built-in decoder chips [14:18:25 CEST] <JEEB> which you nowadays also do have in CPUs as well [14:18:41 CEST] <JEEB> but the whole HW decoding thing is "it can work, BUT" [14:18:54 CEST] <JEEB> if you want to know the gritty details I can link you the mpv manual entry [14:19:13 CEST] <JEEB> which has all the various cases of how HW decoding, even when the decoding part works, can cause worse rendering [14:20:27 CEST] <JEEB> nostrora: also if you want to test a part of the clip not in the beginning of the thing [14:20:32 CEST] <JEEB> you can add -ss SECONDS [14:20:35 CEST] <JEEB> before -i [14:20:45 CEST] <JEEB> so it seeks there [14:20:56 CEST] <nostrora> JEEB: \o/ [14:21:00 CEST] <JEEB> of course you will want to remove that when doing the final encode together with the -t / -vframes [14:21:26 CEST] <JEEB> so you can see which part of the clip you'd be especially interested, encode that ~2500 frames of stuff and then keep checking [14:21:48 CEST] <JEEB> and at the end of the day you will know which CRF value at those specific compression parameters are "good enough" for you [14:22:01 CEST] <JEEB> and you should be able to use them with similar content [14:22:37 CEST] <JEEB> at least that was the idea of the original CRF for libx264. "similar" libvpx options for vp8/9 were just mapped to the "crf" parameter [15:15:43 CEST] <Sellerie> Hi, I am currently trying to update the ancient linux version of the PS2 to more up to date packages. For that I need to find out if there went work of porting tools e.g. ffmpeg from the standard packages to the ps2 package included (source) [15:16:04 CEST] <Sellerie> For that I need to get some older package versions to compare [15:16:20 CEST] <Sellerie> Can I find the source package of ffmpeg 0.58 anywhere? [15:16:51 CEST] <JEEB> on the git most likely [15:17:09 CEST] <JEEB> although I remember building FFmpeg in like 2011 for the PSP pretty much vanilla [15:17:16 CEST] <JEEB> I just had to disable the command line tools for obvious reasons [15:17:34 CEST] <Sellerie> Hmm, I could actually just get the source package and compile it on the ps2 locally [15:17:49 CEST] <Sellerie> And just test whats broken [15:18:22 CEST] <JEEB> yea, I would start with that. the basic cross-compilation parameters are... [15:19:12 CEST] <JEEB> --arch=ARCH --enable-cross-compile --target-os=SOMETHING --cross-prefix=your-cross-prefix- [15:19:28 CEST] <Sellerie> ohoh [15:19:32 CEST] <JEEB> that way the build tools with your-cross-prefix (like your-cross-prefix-gcc) get used [15:19:44 CEST] <Sellerie> I cant find a gcc in that package collection lol [15:20:26 CEST] <JEEB> but yea, the last time I was poking around MIPS FFmpeg I could get it compiled [15:20:38 CEST] <JEEB> I just had to disable command line apps because those aren't a thing on the PSP [15:20:50 CEST] <JEEB> (you make libraries with specific entry points instead, IIRC) [15:21:12 CEST] <Sellerie> https://github.com/ps2dev/ps2toolchain I found that thing. Guess that will work [15:21:35 CEST] <JEEB> woah [15:21:39 CEST] <JEEB> that's even older than the PSP stuff [15:21:49 CEST] <JEEB> PSP I think got up until... 4.6 or 4.9 GCC [15:21:52 CEST] <Sellerie> PS2 was released in 2001 :) [15:22:20 CEST] <JEEB> yes, but neither is using the official sdk anyways [15:22:29 CEST] <JEEB> it's just what people have kept up maintaining [15:22:33 CEST] <Sellerie> Ye [15:22:47 CEST] <JEEB> ok, gcc 4.9.3 for the PSP it seems [15:22:56 CEST] Action: JEEB looked at https://github.com/pspdev/psptoolchain/tree/master/patches [15:23:46 CEST] <JEEB> I think one of the guys tried to upstream the PSP target but I think he stopped due to authorship issues or so [15:23:54 CEST] <Sellerie> Nice. The requirements say that I need a gcc or clang for that [15:24:17 CEST] <JEEB> well yes, the toolchain will be running on your normal PC creating MIPS binaries [15:25:02 CEST] <JEEB> &41 [15:25:05 CEST] <Sellerie> God this project will result in ultimate Frankenstein linux [15:25:36 CEST] <JEEB> well if it's sane it will make the sysroot somewhere and then have the binaries cross-compilers with a cross-prefix [15:25:41 CEST] <JEEB> that's how you usually keep things separated [15:26:01 CEST] <JEEB> for example a lot of distros after all package ARM, mingw-w64 toolchains [15:26:06 CEST] <Sellerie> Maybe I should get my phat PS2 to work again, I can get a goddamn harddrive for that [15:26:14 CEST] <JEEB> so you get for example x86_64-w64-mingw32-gcc etc [15:26:38 CEST] <JEEB> you're not meant to just have a "gcc" that targets another arch [15:26:49 CEST] <Sellerie> I guess [15:27:49 CEST] <Sellerie> So lets create some toolchain [15:28:33 CEST] <JEEB> if you want to be extra safe, use systemd-nspawn or whatever your preferred way of spawning a new kernel namespace under a chroot is [15:31:12 CEST] <Sellerie> I guess I need to read about crosscompiling anyway at first [15:36:03 CEST] <Sellerie> Oh I see there is a cmake included... how far do I get with that? [15:36:53 CEST] <Sellerie> Damn gcc 2.9.5 [15:44:19 CEST] <Sellerie> Thanks for your advise JEEB! [16:16:46 CEST] <bodqhrohro1> Can I encode a video with bitrate less than 1 kbps using libvpx-9? The bitrate still keeps too high when I pass every of minrate, maxrate, b:v and bufsize [16:17:32 CEST] <JEEB> that's more of a question to #vp8 than here, but I would guess no [16:17:43 CEST] <JEEB> many APIs are limited to kilobits [16:17:52 CEST] <JEEB> for example x264 does that as well [16:18:09 CEST] <JEEB> so when libavcodec passes the bit rate to it, it divides it by 1000 [16:18:13 CEST] <JEEB> so you get kilobits instead of bits [16:18:51 CEST] <bodqhrohro1> When I tried to limit opus to too small value it raised an error that only from 500 to 256000 bps. With vp9 I don't see anything like this [16:20:11 CEST] <JEEB> https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/libvpxenc.c#L520..L… [16:20:20 CEST] <JEEB> this is all that libavcodec does when passing the bit rate to libvpx [16:20:36 CEST] <JEEB> libvpx seems to be limited to kilobits per second, yes [16:20:58 CEST] <BtbN> Just show a black screen on the player and put some random noise on it, that will be about the same visual quality than less than 1kbps vp9 [16:21:37 CEST] <JEEB> bodqhrohro1: in theory the wrapper in libavcodec could check if the nonzero value was less than one thousand, but uhh [16:25:51 CEST] <bodqhrohro1> So if vp9 doesn't, is there a video codec that supports any low bitrate? [16:27:20 CEST] <bodqhrohro1> 12 kbps would be probably enough too [16:28:23 CEST] <JEEB> that amount can be set in many encoders but I don't know many that would keep that bit rate considering the amount of headers for each frame that are required [16:28:40 CEST] <JEEB> so unless you have a very low frame rate you can *very* easily go over that [16:29:11 CEST] <BtbN> that's way less than you usually use for audio [16:29:20 CEST] <BtbN> you are not going to get any kind of useful video with that low of a bitrate [16:29:40 CEST] <JEEB> as far as I know only x264 actually has a rate control that will try its darnest to keep your set bit rate, but even it cannot fight the amount that goes onto headers [16:29:54 CEST] <bodqhrohro1> I've set 4 FPS and 86x48 resolution so probably at least some moving squares of corresponding colours will still be seen [16:30:59 CEST] <BtbN> so that's 256 byte per picture on average. [16:35:00 CEST] <bodqhrohro1> And is MKV the only container to fit Opus and H.264 together? No anything supported with browsers? [16:36:23 CEST] <dystopia_> anyone know how i can demux the audio from ea vp6 [16:36:29 CEST] <dystopia_> ffmpeg doesn't see the track [16:36:30 CEST] <JEEB> I think opus in ISOBMFF (colloquially mp4) is standardized now [16:37:07 CEST] <JEEB> at least it was heavily being drafted https://www.opus-codec.org/docs/opus_in_isobmff.html [16:37:18 CEST] <JEEB> and FFmpeg seems to support it [16:37:26 CEST] <JEEB> but no idea if the browsers do, I think mozilla had interest in it [16:37:50 CEST] <bodqhrohro1> Mine still doesn't [16:37:50 CEST] <bodqhrohro1> [mp4 @ 0x560078d54de0] Could not find tag for codec opus in stream #1, codec not currently supported in container [16:37:51 CEST] <bodqhrohro1> Was it added in the latest versions? [16:38:07 CEST] <JEEB> I see a patch in March, 2017 [16:38:31 CEST] <JEEB> also if it's experimental it's likely you have to enable -strict experimental [16:38:51 CEST] <JEEB> and yea, it was merged in April [16:39:02 CEST] <JEEB> and strict experimental is required [16:40:04 CEST] <JEEB> bodqhrohro1: I wouldn't necessarily expect all the things to support that yet, but it's being standardized [16:40:48 CEST] <JEEB> mozilla was rather interested in both FLAC and Opus in ISOBMFF standardization [17:05:26 CEST] <Ducky^> hi all, I'm trying to use ffmpeg to encode captured raw video in realtime [17:05:29 CEST] <Ducky^> if I run: [17:05:33 CEST] <Ducky^> ./bmdcapture -C 0 -m 13 -V 3 -p rgb10 -F nut -f pipe:1 | ffmpeg -re -i pipe:0 -framerate 25 -c:v libx264 -preset ultrafast -maxrate 3000k -bufsize 256k -c:a aac -b:a 160k -ac 2 -ar 44100 -f mpegts udp://10.249.0.14:1234 [17:05:43 CEST] <Ducky^> it will work for a while, but eventually the process will stop [17:06:03 CEST] <Ducky^> while running the speed is just under 1x so it doesn't seem to be quite doing it in realtime [17:06:09 CEST] <c_14> get rid of the -re [17:06:13 CEST] <JEEB> you shouldn't try to use -re, the encoder will be limited by the amount of data given to it [17:06:23 CEST] <JEEB> -re is a horrible hack :D [17:06:37 CEST] <Ducky^> ok, I've removed -re [17:06:43 CEST] <Ducky^> I think the problem still occurs without re though [17:06:46 CEST] <JEEB> (and if you have funkyness in input timestamps you *will* get funzies like "sleeping for 8+ hours" [17:06:59 CEST] <JEEB> also cool that it supports NUT [17:07:03 CEST] <JEEB> the capture thing that is [17:07:13 CEST] <JEEB> has timestamps and video/audio data [17:10:00 CEST] <Ducky^> the process died after a few minutes of running [17:10:07 CEST] <Ducky^> any thoughts on how to keep it going? [17:10:38 CEST] <JEEB> Ducky^: also if it's 10bit RGB the conversion to YCbCr might be slow [17:12:15 CEST] <JEEB> Ducky^: also I'm not sure why you're setting an input frame rate if you have NUT input which *should* have timestamps already [17:12:35 CEST] <JEEB> but yea, I would guess the x264 encoder shouldn't be slowing you down, or the AAC encoder [17:12:44 CEST] <JEEB> although.. what's with that buffer size? [17:12:45 CEST] <Ducky^> the input framerate is 50 and I thought if I downscaled it to 25 ffmpeg would keep up beeter [17:12:53 CEST] <Ducky^> better* [17:13:10 CEST] <JEEB> you're doing 3000 kbps maxrate and with 256 kilobit buffer [17:13:11 CEST] <Ducky^> JEEB: higher buffer sizes that people recommend seem to break it [17:13:21 CEST] <JEEB> that sounds coincidental [17:13:27 CEST] <Ducky^> I can't tell why, but a 2M buffer size I can't see any video at all over the network [17:13:40 CEST] <JEEB> there's a transfer rate option in the UDP output I think [17:13:44 CEST] <JEEB> if that's the problem [17:14:24 CEST] <JEEB> so that even if the encoder outputs more bits suddenly, they don't get spammed out of UDP right away [17:14:39 CEST] <JEEB> but in a more stable manner, if the network doesn't like it otherwise [17:15:28 CEST] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=blobdiff;f=libavformat/udp.c;h=8699… [17:15:50 CEST] <JEEB> so basically -bitrate and -burst_bits I think, and setting only one of those should be enough [17:17:06 CEST] <JEEB> Ducky^: I would recommend looking at 10bit YCbCr from the capture card first, then increasing debug logging level to see which part is the one that gives up in ffmpeg.c [17:18:33 CEST] <JEEB> the networks I've dealt with have dealt with the bursts you might get with a higher buffer size just fine, but the UDP bitrate and burst_size options should enable you to limit the output in case the network doesn't like the difference that a bigger buffer lets the encoder do [17:20:49 CEST] <Ducky^> thanks JEEB, I'm seeing if I can get the hdmi input to be ycbr [17:21:30 CEST] <JEEB> well you were setting -p rgb10 so I expected you'd be able to poke it to be something else as well [17:21:41 CEST] <JEEB> not that I know what on earth you're using for capture :D [17:21:44 CEST] <Ducky^> it actually looks like the capture software can only support yuv8, yuv10 or rgb10 [17:22:02 CEST] <JEEB> there's no digital yuv so that's YCbCr [17:22:14 CEST] <JEEB> people just still call it YUV [17:22:16 CEST] <JEEB> colloquially [17:22:18 CEST] <Ducky^> right, ok [17:22:59 CEST] <Ducky^> now I just need to see if I can get the raspberry pi to output at ycbr, it seems like there is an option but the pi still reports rgb [17:23:05 CEST] <Ducky^> but strangely the colours changed [17:23:39 CEST] <JEEB> yea, the desktop will still be rendered in RGB [17:23:53 CEST] <JEEB> and then whatever is pushing the picture over will most likely be in control of the conversion YCbCr [17:23:56 CEST] <JEEB> *conversion to [17:27:27 CEST] <Ducky^> hmm, I can only get a picture from the capture card if I set pixel encoding to be rgb10 [17:28:01 CEST] <BtbN> if that's the format it captures [17:31:19 CEST] Action: TrIPpY *waves* HI [17:31:57 CEST] <TrIPpY> Hello. I have a ffmeg and ffplay use case involving capturing webcam images. Is anybody willing to help? [17:33:15 CEST] <JEEB> just ask your actual questions [17:34:41 CEST] <Ducky^> hooray, got yuv capture working! [17:34:48 CEST] <Ducky^> lets see if it keeps running [17:35:10 CEST] <Ducky^> speed is at 1x now [17:35:32 CEST] <TrIPpY> OK. I'm actually trying to sort of like trying to use my webcam to digitalize documents (i do know they have scanners for that). When i look at the image like so: [17:35:41 CEST] <TrIPpY> ffplay -f v4l2 -framerate 10 /dev/video0 [17:36:04 CEST] <TrIPpY> the characters on the page are readable [17:36:59 CEST] <TrIPpY> to have no loss of info i capture it to rawvideo like this: [17:37:03 CEST] <TrIPpY> ffmpeg -y -f v4l2 -framerate 10 -i /dev/video0 -c:v rawvideo -pix_fmt yuyv422 out.yuv [17:37:56 CEST] <TrIPpY> camera pixel format seems to be yuyv422 when listing capabilities with ffplay [17:38:53 CEST] <TrIPpY> after capture i play in ffplay like this: [17:38:59 CEST] <TrIPpY> ffplay -fdebug 1 -video_size 640x480 -pix_fmt yuyv422 -framerate 10 out.yuv [17:39:36 CEST] <klaxa> it'll be hard to digitize documents at 640x480... [17:40:24 CEST] <TrIPpY> i know but when live playing in ffplay it's readable [17:40:37 CEST] <TrIPpY> just about anyway [17:40:49 CEST] <TrIPpY> the unwanted thing in the log is this: [17:40:54 CEST] <TrIPpY> [ffplay_crop @ 0x7f18a4002aa0] auto-inserting filter 'auto-inserted scaler 0' between the filter 'ffplay_buffer' and the filter 'ffplay_crop' [17:40:54 CEST] <TrIPpY> [AVFilterGraph @ 0x7f18a4000d40] query_formats: 3 queried, 1 merged, 1 already done, 0 delayed [17:40:54 CEST] <TrIPpY> [auto-inserted scaler 0 @ 0x7f18a4002160] w:640 h:480 fmt:yuyv422 sar:0/1 -> w:640 h:480 fmt:yuv420p sar:0/1 flags:0x4 [17:42:17 CEST] <TrIPpY> know, is it possible to not have the scaler auto inserted OR set the pixel format to yuyv422 because the conversion to yuv420p, I think makes it unreadable hwile the live image is much better [17:43:20 CEST] <TrIPpY> capturing it to video is a test btw. Ultimately i'd like it to go to a photo format. [17:45:00 CEST] <klaxa> i would think the filter is inserted for viewing only? i don't think you should worry about that too much? [17:46:57 CEST] <TrIPpY> same effect (somewhat) also shows when capturing to bmp's or tiff's [17:47:34 CEST] <TrIPpY> i'd would actually like to see the capture of the same quality as the live image in ffplay. [17:48:11 CEST] <TrIPpY> capturing to rawvideo is as lossless as possible, is that right? [17:49:12 CEST] <Ducky^> has anyone here used ffmpeg to send video into wowza engine? [17:49:30 CEST] <klaxa> usually yes, codec copying is just as lossless though [17:49:48 CEST] <klaxa> most webcams don't send rawvideo but encoded frames, often mjpeg or h264 [17:49:52 CEST] <Ducky^> I can open a video file with VLC and send it to local port udp 10010, which wowza is configure to watch [17:49:55 CEST] <Ducky^> the video will appear [17:50:00 CEST] <TrIPpY> ok. ok [17:50:14 CEST] <Ducky^> but with ffmpeg there is either no video or it's just a static glitchy image [17:51:45 CEST] <TrIPpY> so i should only change the -vcodec to mjpeg? [17:52:21 CEST] <klaxa> no that would re-encode to mjpeg, use -c:v copy or -c copy [17:53:27 CEST] <TrIPpY> i'll try that. see if it makes a difference. Thanks KLaxa [17:56:20 CEST] <klaxa> also, jpeg (-c:v libopenjpeg) supports yuv422p (afaict it should be possible to convert yuyv422 to yuv422p without loss) [18:02:58 CEST] <TrIPpY> Ok. Thanks again Klaxa. [18:03:12 CEST] <TrIPpY> i try that too [18:57:47 CEST] <brotherBox> hi guys. i have a program that asks me to recompile libswscale with -fPIC, however, i am already passing --enable-shared and it is still throwing that message. is there any way i can see how libsws is compiled? [18:57:58 CEST] <brotherBox> *how libswscale is compiled [18:58:38 CEST] <JEEB> --enable-pic if PIC is needed on architectures where it's not the default for shared libraries [18:59:04 CEST] <JEEB> (it's not default for 32bit x86 because it kills off some required registers for SIMD optimizations [18:59:29 CEST] <JEEB> but the thing can also be whatever it's depending on, of course. not necessarily FFmpeg's own libraries. [18:59:46 CEST] <JEEB> basically clean your build root and re-configure with enable-pic and you should get PIC [18:59:58 CEST] <brotherBox> alright, i'll try it and report back [19:16:11 CEST] <brotherBox> wow that did it [19:16:13 CEST] <brotherBox> thank you! [19:33:58 CEST] <BtbN> that error should only ever show up when you are linking statically [19:55:06 CEST] <thebombzen> how does interal AAC stack up with fdk-aac nowadays? should I still prefer fdk-aac if I have it? [19:56:49 CEST] <BtbN> see "the bug": https://trac.ffmpeg.org/ticket/2686 [19:57:03 CEST] <JEEB> I don't think it's on par yet, but generally it's good enough. as in, when running live transcoding with it, I don't want to switch it [19:58:03 CEST] <BtbN> I think it's pretty much indistiguishable for AAC LC [20:00:44 CEST] <JEEB> not like it supports HE, right? [20:34:31 CEST] <jiffe> is there a way to find out what the max bitrate of an mkv is? [20:35:06 CEST] <JEEB> depends on which level. are you interested in the maximum amount of bits per picture or verification of VBV/HRD or something? [20:35:58 CEST] <jiffe> I'm having buffering issues streaming to my TV so I'm trying to find out the max rate that would be sent to the tv [20:36:32 CEST] <JEEB> well the amount of it being sent is not necessarily the max amount of bits per second (calculated over what?) in the file [20:36:52 CEST] <JEEB> also for some reason my TV had issues with MPEG-TS, but if I remuxed to mp4 stuff worked somewhat better [20:37:02 CEST] <JEEB> this was with an average of around 40mbps though :P [20:37:42 CEST] <jiffe> yeah, reading up on similar issues people have had with plex and samsung TVs they've pointed to low bandwidth restrictions placed by the tv [20:38:23 CEST] <JEEB> I don't use plex but rather the built-in stuff in windows (10) [20:39:06 CEST] <JEEB> anyways, I'm not sure if you will find the details from the file side, you would have to look into what the logic on the TV's side is to know. otherwise you would be doing arbitrary decisions [20:39:23 CEST] <JEEB> esp. given the fact that you don't know how quickly the content is being transferred and in what sort of buffers [20:39:46 CEST] <JEEB> "what is the maximum bit rate according to X" is hard to answer if the X is unknown [20:39:53 CEST] <jiffe> well I know that is buffers for about 3-4 seconds and runs for another 15 seconds before buffering again in some scenes [20:40:16 CEST] <JEEB> to give you some glimmer of hope you can check the strings in the file if it's an encode done with x264 [20:40:40 CEST] <JEEB> or if you know it's a straight remux from a blu-ray (as in, streams are as-is unchanged) [20:40:59 CEST] <JEEB> for x264 you check the output of `strings file.mkv |grep "x264"` [20:41:04 CEST] <jiffe> yeah all my rips are done with makemkv straight from the blurays [20:41:06 CEST] <JEEB> and hope you have maxrate and bufsize there [20:41:09 CEST] <Mavrik> Some TVs (hi Sony!) have ethernet ports connected via internal USB 2.0 bus -_- [20:41:09 CEST] <JEEB> ok [20:41:16 CEST] <JEEB> jiffe: for blu-ray it's simple [20:41:23 CEST] <JEEB> 40mbps maxrate and 40mbps bufsize [20:41:27 CEST] <JEEB> the video will never go over that [20:41:46 CEST] <JEEB> *40megabit bufsize [20:42:03 CEST] <JEEB> so roughly a second's buffer within which the maximum bit rate is 40 megabits per second [20:42:15 CEST] <jiffe> gotcha [20:42:31 CEST] <JEEB> but that is only in the case of straight stream rips, as in you don't do re-encodes [20:42:46 CEST] <JEEB> jiffe: with samsung I had better experience remuxing to mp4 btw [20:43:00 CEST] <JEEB> also that enables you to use windows 10's own miracast/dlna thing [20:43:17 CEST] <JEEB> `ffmpeg -i in.mkv -c copy -movflags faststart out.mp4` [20:43:19 CEST] <jiffe> yeah, this all runs on linux [20:43:28 CEST] <jiffe> I can try mp4 and see if that makes a difference [20:43:40 CEST] <JEEB> yea, for some reason the TV liked mp4 compared to the original m2ts [20:43:54 CEST] <JEEB> I haven't been able to try matroska since windows thinks it needs to transcode that to wmv3 [20:44:06 CEST] <JEEB> and I have been way too lazy to search for a sane miracast or dlna thing [20:44:14 CEST] <JEEB> that just passes everything as-is to the device [20:46:20 CEST] <e38383> minidlna is very basic [20:46:25 CEST] <fahadash> I mapped an mp3 audio to my video using -map, the audio is shorter than the video; I want audio to start repeating as soon as it finishes, how do I do it? [20:49:21 CEST] <jiffe> that re-encoded pretty quick [20:49:42 CEST] <JEEB> no re-encode [20:49:44 CEST] <JEEB> just remultiplex [20:49:49 CEST] <JEEB> -c copy does that [20:50:28 CEST] <JEEB> oh, and I forgot to say that audio tracks aren't calculated into that 40mbps/40 megabit maxrate/bufsize [20:50:31 CEST] <JEEB> that's just for the video track [20:59:21 CEST] <jiffe> no difference with the mp4, buffers the same [21:00:17 CEST] <lindylex> When do i ad the resize to get this to resize the photos ffmpeg -r 60 -f image2 -s 1080x810 -start_number 633 -i IMG_%04d.JPG -vcodec libx264 -crf 15 -pix_fmt yuv420p test.mp4 [21:01:47 CEST] <jiffe> I have an email in to samsung to find out what kind of bandwidth this tv has [21:03:23 CEST] <JEEB> jiffe: also when I play with miracast I don't have any buffering stuff on the TV side [21:03:31 CEST] <lindylex> jiffe : is your smart tv not that smart? [21:03:35 CEST] <JEEB> as in, I have no visible buffering levels [21:04:14 CEST] <JEEB> lindylex: -vf 'scale=w=WIDTH:h=height,format=yuv420p' should handle both the pix_fmt and the scaling [21:04:31 CEST] <lindylex> JEBB : Thanks [21:04:35 CEST] <JEEB> width or height can be -2 for "automatical according to the aspect ratio" [21:04:41 CEST] <JEEB> (and modulo 2) [21:04:48 CEST] <JEEB> module 2 is needed for 4:2:0 [21:04:53 CEST] <JEEB> because chroma is in 2x2 blocks [21:05:27 CEST] <lindylex> How can I automatically convert width to 1080 with calculating the height? [21:07:03 CEST] <JEEB> do you really mean width to 1080? [21:07:16 CEST] <lindylex> yes [21:07:17 CEST] <JEEB> anyways, scale=w=1080:h=-2 [21:07:27 CEST] <JEEB> and then add ,format=yuv420p for the pixel format [21:07:34 CEST] <JEEB> format is a kind of a metafilter :P [21:07:36 CEST] <lindylex> oh ok thanks. Now I understand what you are saying. [21:07:43 CEST] <BtbN> that's a very unusual width. If you want 1080p, 1080 is the height. [21:08:02 CEST] <lindylex> 1080 X 720 [21:14:57 CEST] <Johnjay> why is 1.5 the aspect ratio for hd? seems a bit large to me [22:02:00 CEST] <bodqhrohro1> What values can I set rc_strategy and b_strategy to? [22:04:22 CEST] <furq> for what codec [22:30:16 CEST] <Luigi12> michaelni: not sure if you'd heard about this http://openwall.com/lists/oss-security/2017/10/02/1 but just letting you know if not [22:33:43 CEST] <bodqhrohro1> furq: H.264 [22:33:59 CEST] <durandal_1707> Luigi12: whats that? [22:34:01 CEST] <furq> b_strategy is b-adapt, and rc_strategy doesn't exist for x264 afaik [22:34:21 CEST] <bodqhrohro1> For what codecs does it exist? [22:34:34 CEST] <Luigi12> durandal_1707: Internet Bug Bounty project. They were going to include ffmpeg, but they needed to hear from upstream before OK'ing it. [22:34:40 CEST] <Luigi12> they already OK'd libav [22:34:55 CEST] <Luigi12> which seems silly since nobody uses that anymore. It'd be much more useful to have ffmpeg in there. [22:36:05 CEST] <JEEB> Luigi12: that is probably more useful to ask on -devel [22:36:14 CEST] <Luigi12> true [22:44:18 CEST] <plitter> Is it possible to change settings during a stream? I want to create content and want the background and webcam to be on at the same time. But would like to be able to move the webcam around if I need to show something at that part of the screen. The same if I have graphics on the stream I would like to be able to move it. I can imagine that one solution would be to put settings in a config file and make [22:44:20 CEST] <plitter> ffmpeg continuously read it? [22:45:12 CEST] <JEEB> sounds like you want to create your own API client for the APIs on which FFmpeg is built on [22:45:30 CEST] <JEEB> or you just want to separate your overlay/whatever generation from the encoding process [22:45:36 CEST] <furq> you could maybe do that with zmq [22:46:29 CEST] <Johnjay> hey furq, jeeb [22:46:37 CEST] <Johnjay> is there a way to do that thing [22:46:44 CEST] <furq> no [22:46:46 CEST] <Johnjay> about inserting a file within another file at time t1,t2,t3? [22:46:59 CEST] <Johnjay> ok i'll split the file up then. [22:47:07 CEST] <Johnjay> i needed to add intros and outros at regular intervals for a podcast [22:47:13 CEST] <Johnjay> but the times are not always at exact times [22:47:29 CEST] <plitter> JEEB: I want to do both at the same time, so I don't have to do postproduction [22:47:35 CEST] <furq> there is probably a way to do it in a single command but i imagine it'd be more complicated than just split+concat [22:47:54 CEST] <Johnjay> right [22:48:06 CEST] <Johnjay> i wanted to do something like feed it a list of times and insert file A at time B [22:48:20 CEST] <furq> the way i'm thinking is something like trim,setpts;trim,setpts;trim,setpts;amerge with anullsrc [22:48:21 CEST] <Johnjay> filter_complex looks scary though [22:48:42 CEST] <Johnjay> pts = presentation time stamp right? [22:48:42 CEST] <JEEB> plitter: dynamic input means that you either want to create your own API client or you separate your overlay/whatever you're doing with the input(s) from the transcoding process (which could be ffmpeg.c for example) [22:48:44 CEST] <furq> right [22:49:14 CEST] <furq> !filter zmq @plitter [22:49:14 CEST] <nfobot> plitter: http://ffmpeg.org/ffmpeg-filters.html#zmq_002c-azmq [22:49:33 CEST] <furq> plitter: you can send commands to the overlay filter with that, but don't ask me exactly how [22:49:37 CEST] <furq> i'm not sure anyone in here has ever used it [22:52:25 CEST] <plitter> furq: thanks, I've seen someone saying they managed to do different text on a stream, I'll see if that is also applicable here http://ffmpeg.gusari.org/viewtopic.php?f=11&t=1897 [00:00:00 CEST] --- Mon Oct 9 2017

1 0

ffmpeg-devel.log.20171007
by burek 08 Oct '17

08 Oct '17

[00:02:13 CEST] <jkqxz> On patch 1/2, I was really just wanting to know that it's actually been tested in that mode now. [00:02:48 CEST] <jkqxz> ... and on what. [00:02:50 CEST] <jkqxz> (Hint hint wiki page outlining supported devices.) [00:04:20 CEST] <ldts> did I answer your question or you expected some other info? yes imx6 is single plane (it was the Kodi team who raised the issue) [00:05:36 CEST] <jkqxz> Yeah, if it's been tested now that's fine. [00:07:50 CEST] <ldts> ok [00:13:52 CEST] <ldts> still building? [00:14:15 CEST] <jkqxz> Almost done. [00:14:58 CEST] <jkqxz> wm4: Which hardware decoding patches do you actually want here? [00:18:52 CEST] <jamrial> jkqxz: inline asm is annoying like that when you're register starved [00:19:39 CEST] <jkqxz> ldts: <http://sprunge.us/dJBC>, <http://sprunge.us/PeIG> [00:21:43 CEST] <ldts> jkqxz: um...ok. and that is a debug message, correct? [00:22:40 CEST] <jkqxz> Yes. [00:23:19 CEST] <jkqxz> ff_v4l2_m2m_codec_end() has been called with an AVCodecContext containing a NULL priv_data. [00:23:57 CEST] <ldts> I wonder why I dont see it on my board...ok [00:24:50 CEST] <wm4> jkqxz: minimum the frame pool info one I sent to Libav, ideally the cuvid one, and possibly yours [00:29:06 CEST] <jkqxz> ldts: Do you want me to do anything else with this? (It's 100% repeatable, I don't think I said that earlier.) [00:29:56 CEST] <ldts> jkqxz: I cant reproduce on my board...but I guess it makes sense. when the codec is closed the private data must be deallocated inmmediatly right? [00:30:35 CEST] <ldts> so on the second call, the private data is null... [00:30:56 CEST] <ldts> I wonder why that doesnt happen on arm64 (maybe the memory is managed differently) [00:31:14 CEST] <jkqxz> Yes <http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/utils.c;h=9551f31…>. [00:33:07 CEST] <ldts> ok so you are right, the patch doesnt fix it... [00:34:16 CEST] <ldts> I'll think of something else then ...but av_free should be setting the pointers to null on all architectures [00:35:43 CEST] <jkqxz> Ok. I'll push the other one on its own, then. [00:36:24 CEST] <ldts> yes..I suppose there is no way to delay the deletion of the private data? [00:37:32 CEST] <jkqxz> No. I think you need to keep references to it from the buffers which are outstanding (see hwcontext, where the frames context do all of this). [00:39:04 CEST] <Compn> wm4 : you mean you want the release delayed until cuvid is in ? [00:39:15 CEST] <wm4> Compn: yes [00:39:16 CEST] <Compn> wm4 : "I vote the release until this has happened." [00:39:22 CEST] <Compn> you ate a word somewhere [00:39:27 CEST] <ldts> but the buffers to keep a refence (See v4l2_free_buffer that can get called after the codec was closed) [00:39:29 CEST] <wm4> ah [00:39:50 CEST] <ldts> s/to/do [00:39:50 CEST] <Compn> just a friendly reminder :) [00:39:55 CEST] <Compn> i dont want people to be confused :) [00:40:32 CEST] <jkqxz> Can I accidentally the release as well? [00:40:52 CEST] <ldts> ? [00:41:00 CEST] <Compn> ehe [00:41:20 CEST] <Compn> its a recurring interenet joke when someone the word. [00:43:05 CEST] <ldts> jkqxz: so after the codec has been closed, v4l2_free_buffer can be called; it keeps a reference to the buffer which provides a pointer to the AVCtx via the private data (V4L2m2mContext) [00:48:33 CEST] <cone-507> ffmpeg 03Jorge Ramirez-Ortiz 07master:2a31ad7d6063: avcodec/v4l2: fix single plane decoding [00:50:48 CEST] <ldts> jkqxz: I can have something ready in 20 minutes. will you be around? [00:53:53 CEST] <jkqxz> For a bit. I do sleep sometimes. [00:55:36 CEST] <ldts> thanks a lot, just 10 minutes [01:00:47 CEST] <ldts> testing on my end now [01:01:57 CEST] <ldts> looks good on my side [01:02:31 CEST] <ldts> https://hastebin.com/iroruqomuq.php [01:03:41 CEST] <ldts> obviously no need to clean the build...so should be fast. again many thanks for doing this. I'll get my power supply soon! [01:04:40 CEST] <jkqxz> Is that link right? It's a blank page, with just some javascript in the source. [01:05:17 CEST] <ldts> yes it works for me: https://hastebin.com/iroruqomuq.php [01:05:43 CEST] <ldts> http://paste.ubuntu.com/25688854/ [01:05:48 CEST] <ldts> what about that one? [01:06:08 CEST] <ldts> this goes on top of the previous patch by the way [01:06:24 CEST] <jkqxz> That's better. [01:06:29 CEST] <ldts> so no need to remove the other one... [01:12:05 CEST] <jkqxz> Looks better. [01:12:36 CEST] <ldts> ok. should I send a new patch then? [01:12:49 CEST] <ldts> works also ok on my hardware [01:13:52 CEST] <jkqxz> Right, it doesn't crash, but there is still something nasty going on. [01:14:09 CEST] <jkqxz> <http://sprunge.us/cKSB> (and some others). [01:14:57 CEST] <ldts> how do I Get that info? [01:15:27 CEST] <ldts> I'll get the supply and work on my odroid then... [01:16:26 CEST] <jkqxz> That's valgrind. Is it valgrind-clean on aarch64? [01:17:08 CEST] <ldts> sorry I havent been checking it after the initial implementation [01:18:22 CEST] <ldts> will test now [01:19:47 CEST] <ldts> same command than before? [01:20:40 CEST] <jkqxz> Yes. [01:20:56 CEST] <ldts> --leak-check=yes? [01:22:35 CEST] <jkqxz> Should be irrelevant, but I ran with full. [01:23:13 CEST] <jkqxz> Looks some packets are being leaked as well. [01:23:58 CEST] <ldts> http://paste.ubuntu.com/25688964/ [01:24:46 CEST] <ldts> ok I'll have to look into this more carefully [01:25:20 CEST] <jkqxz> But you didn't have any invalid reads? [01:25:29 CEST] <ldts> yes I did [01:25:39 CEST] <ldts> same than yours [01:25:56 CEST] <jkqxz> Right, ok. [01:27:28 CEST] <ldts> I think the segfault fix should go in while I fix this. what do you think? [01:31:45 CEST] <jkqxz> That this problem hasn't been understood properly. [01:34:12 CEST] <ldts> isn't the problem statement to avoid closing the file descriptor until the last buffer reference is released? but ok, maybe we should discuss on the ML [01:51:27 CEST] <ldts> jkqxz: interesting the same code on 3.3.3 gives http://paste.ubuntu.com/25689107/ [01:51:48 CEST] <ldts> jkqxz: same code, same test [01:52:25 CEST] <ldts> jkqxz: instead of http://paste.ubuntu.com/25688964/ [03:00:04 CEST] <cone-507> ffmpeg 03James Almer 07master:cc5b7601f7c2: avformat/mp3enc: flush buffered packets if referencing fails [03:43:12 CEST] <cone-507> ffmpeg 03Carl Eugen Hoyos 07master:a20f64bee235: lavf/img2dec: Auto-detect svg images. [08:06:32 CEST] <rcombs> has anyone considered [08:06:37 CEST] <rcombs> using nasm's preprocessor [08:06:38 CEST] <rcombs> for ARM [11:21:39 CEST] <cone-957> ffmpeg 03Sasi Inguva 07master:123f6dc6b563: lavfi/avfilter.c: Correct guess_status_pts to account for differing link timebases. [14:05:50 CEST] <ubitux> ffplay is broken with ffmpeg -lavfi testsrc2=d=5:r=100 -r 100 -y /tmp/fps100.gif [14:06:48 CEST] <ubitux> (the color are fucked but i'm talking about the timings) [14:07:00 CEST] <ubitux> it's fine in firefox [14:07:23 CEST] <ubitux> ah my bad, it's not [14:07:47 CEST] <ubitux> i though a delay of 1 meant a sleep of 1/100 sec [14:07:57 CEST] <ubitux> there is something weird going on here [20:49:41 CEST] <cone-605> ffmpeg 03Carl Eugen Hoyos 07master:50462e3e5e32: lavf/sdp: Fix MIME-type for big-endian G.726. [20:49:41 CEST] <cone-605> ffmpeg 03Carl Eugen Hoyos 07master:2386cfc1ae29: lavf/rtpenc: Add support for little-endian G.726. [20:50:17 CEST] <cone-605> ffmpeg 03Carl Eugen Hoyos 07master:5d3e935728c1: lavfi: Rename local variables "main" as "master". [23:28:13 CEST] <BBB> j-b: ping [23:28:42 CEST] <j-b> BBB: pong? [23:35:04 CEST] <atomnuker> speaking of which, j-b, are the vdd talks going to be uploaded to youtube? [23:35:27 CEST] <j-b> atomnuker: uhh, why not [23:35:57 CEST] <durandal_1707> effort! [00:00:00 CEST] --- Sun Oct 8 2017

1 0

ffmpeg.log.20171007
by burek 08 Oct '17

08 Oct '17

[00:13:06 CEST] <llogan> LiftLeft: i mentioned xwininfo, not xpdyinfo [00:14:09 CEST] <LiftLeft> oh [00:19:00 CEST] <LiftLeft> llogan: do I gave ffmpeg somehow? [00:19:05 CEST] <LiftLeft> *give [00:20:02 CEST] <LiftLeft> llogan: do I give ffmpeg the window's hex number, somehow? [00:35:23 CEST] <llogan> you can click the window and xwininfo will provide info [00:35:38 CEST] <llogan> LiftLeft: ^ [00:36:35 CEST] <LiftLeft> I know that [00:36:42 CEST] <LiftLeft> what do I give ffmpeg? [00:36:47 CEST] <llogan> or provide name or id.... [00:36:57 CEST] <llogan> geometry as shown in the second link i gave you [00:39:36 CEST] <LiftLeft> how do I give it the id? [00:43:24 CEST] <llogan> what's "it"? [00:43:32 CEST] <LiftLeft> ffmpeg [00:44:19 CEST] <llogan> you can't [00:44:43 CEST] <LiftLeft> ? [00:45:27 CEST] <LiftLeft> am I suppose to crop it or something? [00:45:28 CEST] <llogan> you can only provide display number, screen number, x, y, width, height. [00:46:10 CEST] <LiftLeft> I need to stream a minimized window [00:46:31 CEST] <llogan> i don't think you can do that with x11grab as far as i know [00:46:36 CEST] <LiftLeft> ok [00:46:58 CEST] <LiftLeft> what command line program should I use to stream? [00:47:14 CEST] <llogan> i don't know [00:58:38 CEST] <LiftLeft> I think I can just use obs [00:58:45 CEST] <LiftLeft> command parameters [01:24:49 CEST] <jerome-> I'm looking for help to stream my usb webcam to youtube [01:25:13 CEST] <jerome-> anyone tried? [01:30:55 CEST] <LiftLeft> jerome-: use obs [01:31:40 CEST] <jerome-> the camear is on pi, with no screen [01:32:15 CEST] <LiftLeft> I dunno [01:38:17 CEST] <llogan> jerome-: omx may be your best bet [01:38:41 CEST] <llogan> make sure your ffmpeg is compiled with --enable-omx-rpi [01:39:29 CEST] <jerome-> thanks, I'm looking into that [01:46:17 CEST] <llogan> jerome-: http://trac.ffmpeg.org/wiki/EncodingForStreamingSites may be of use, but it has no OpenMax info [02:47:44 CEST] <cryptodechange> If aq-strength 'Reduces blocking and blurring in flat and textured areas', why is it lower for the animation preset which makes use of a lot of 'flat areas' [02:48:06 CEST] <cryptodechange> Is it because it's overkill and unnecessary, and lower values uses less bitrate? [03:11:38 CEST] <BunnyHelp_> so, erm [03:12:02 CEST] <BunnyHelp_> i've been having an issue with extracting subtitles from an mp4 [03:12:10 CEST] <BunnyHelp_> "Unsupported codec with id 100359 for input stream 3" [03:12:31 CEST] <BunnyHelp_> is this chat even alive? :p [03:17:12 CEST] <DHE> it's been 10 minutes since you joined... geez... [06:48:37 CEST] <rabbe> how to remove ffmpeg completely, no matter how it was installed? [07:21:32 CEST] <tdr> that is an impossible question [08:13:47 CEST] <akkad> not really [08:14:35 CEST] <akkad> `which ffmpeg` + ldd should dtrt [08:18:37 CEST] <hiihiii> hello [08:19:10 CEST] <hiihiii> I'm trying to stream to youtube but always get [08:19:23 CEST] <hiihiii> RTMP_Connect0, failed to connect socket. 111 (Connection refused) [08:20:00 CEST] <hiihiii> rtmp://a.rtmp.youtube.com/live2/<PASS>: Unknown error occurred [08:20:26 CEST] <hiihiii> it used to work but no. hm? [08:21:09 CEST] <hiihiii> ffmpeg -re -i test.mp4 -threads 1 -c copy -f flv rtmp://a.rtmp.youtube.com/live2/<PASS> [08:23:06 CEST] <diverdude> Hi, i have to encode a 4200×2800 grayscale uint8 using h264. At what framerate will the h264 implementation in ffmpeg allow me to encode this signal? [08:28:20 CEST] <hiya> how I use ffmpeg to convert videos to webm? [08:28:43 CEST] <hiya> keep the same or comparable audio/video quality? [08:28:54 CEST] <hiya> but when I use the command, it said something about audio [08:29:18 CEST] <hiya> [libopus @ 0x55c8c3b72020] No bit rate set. Defaulting to 96000 bps. [08:29:54 CEST] <hiya> 96 kb/s is much lower than source Stream #0:1[0x1c0]: Audio: mp2, 48000 Hz, stereo, s16p, 384 kb/s [08:30:08 CEST] <hiya> and now it is becoming Stream #0:1: Audio: opus (libopus), 48000 Hz, stereo, s16, 96 kb/s [11:34:26 CEST] <dan3wik> Can ffmpeg download twitch streams? [11:49:55 CEST] <durandal_1707> yes [12:28:01 CEST] <diverdude> Hi, i have to encode a 4200×2800 grayscale uint8 using h264. At what framerate will the h264 implementation in ffmpeg allow me to encode this signal? [12:30:52 CEST] <Sharkigator> i guess the framerate is up to you? [17:46:19 CEST] <ren0v0> Hi, i'm trying to convert .mkv to .mp4 to play on LG TV, but i'm getting this error > Could not find tag for codec truehd in stream #1, codec not currently supported in container [17:48:19 CEST] <DHE> mp4 doesn't support all codecs. you'll have to transcode the video to another codec (most likely h264) [17:50:23 CEST] <ren0v0> DHE, maybe i should provide my problem in another way then. My LG tv will play this .mkv file, its 4k HEVC. However, it gives an error about the audio (but video plays). How can i convert my file so it will play, i guess i want to reencode the audio or something [17:50:45 CEST] <DHE> oh.. if it supports mkv then that's great. just convert the audio to something it supports... [17:50:52 CEST] <ren0v0> i tried "-vcodec copy -acodec aac -strict -2 -b:a 384k output.mp4", but get this error > [aac @ 0x24c68a0] Unsupported number of channels: 8 [17:51:07 CEST] <DHE> 8 channels? [17:51:10 CEST] <ren0v0> The audio is ATMOS [17:51:11 CEST] <JEEB> 7.1 I guess [17:51:12 CEST] <BtbN> well, the 8 channels are probably why your TV does not like the audio [17:51:14 CEST] <ren0v0> yea :S [17:51:21 CEST] <DHE> also strict -2 isn't needed since the current versions of ffmpeg. [17:51:27 CEST] <JEEB> switch to 5.1 or so? [17:51:31 CEST] <ren0v0> DHE ok i just found it on SO [17:51:45 CEST] <JEEB> of course [17:51:46 CEST] <JEEB> lol [17:51:50 CEST] <ren0v0> :D [17:51:58 CEST] <ren0v0> How can i "switch to 5.1" ? [17:52:03 CEST] <JEEB> anyways, -af "format=5.1" [17:52:05 CEST] <JEEB> I think [17:52:18 CEST] <JEEB> that should downmix the 7.1 to 5.1 [17:52:25 CEST] <ren0v0> So would i use -codec copy -af "format=5.1" ? [17:52:35 CEST] <JEEB> and then the -c:a aac [17:52:36 CEST] <DHE> -af aformat=channel_layouts=[checking] [17:52:44 CEST] <JEEB> I think it's called 5.1 [17:53:00 CEST] <JEEB> that much i remember checking [17:53:04 CEST] <JEEB> for the channel layouts [17:53:21 CEST] <e38383> if it's just a tv and you only need stereo you can also just use -ac 2 (audio channels) [17:53:29 CEST] <JEEB> yea [17:53:59 CEST] <DHE> yeah, "5.1" is a valid value [17:54:04 CEST] <DHE> or "stereo" for 2-channel [17:54:59 CEST] <JEEB> yup [17:55:16 CEST] <JEEB> e38383: I prefer setting the output channel layout instead of just channel count [17:56:04 CEST] <e38383> JEEB: definitly sounds like the better idea, I just wanted to provide a quick solution (one that's easier to remember) [17:56:33 CEST] <ren0v0> -af is frames, is that actually right ? [17:56:39 CEST] <DHE> "audio filter" [17:57:36 CEST] <ren0v0> it doesn't like this > -vcodec copy -af "format=5.1" -c:a aac -b:a 384k output.mp4 says 5.1 is unsupported [17:57:50 CEST] <ren0v0> sorry these filters are way over my head [17:58:16 CEST] <DHE> aformat=channel_layouts=5.1 [17:58:17 CEST] <BtbN> Doesn't format expect a video pixel format? [17:58:25 CEST] <DHE> there's distinct filters for audio and video [17:58:58 CEST] <DHE> filters whose names begin with 'a' are often audio filters that have a similar behaviour to their video counterparts [17:59:02 CEST] <ren0v0> DHE, aformat=channel_layouts=5.1: Invalid argument [18:00:27 CEST] <DHE> works for me... maybe you have an old version? [18:01:14 CEST] <ren0v0> can you paste full command ? [18:01:31 CEST] <BtbN> 384k for aac is also super exzessive [18:01:32 CEST] <ren0v0> o wait, i'm doing mp4 maybe thats why :D [18:01:51 CEST] <DHE> BtbN: if it's 5.1, it's not so bad.. for stereo, yes [18:01:51 CEST] <JEEB> no [18:01:55 CEST] <JEEB> mp4 is just fine [18:02:01 CEST] <BtbN> oh, right. It's tons of channels [18:02:04 CEST] <JEEB> the container can take AAC and AVC [18:02:06 CEST] <BtbN> yeah, probably fine then [18:02:27 CEST] <ren0v0> this is version 7:2.8.11-0ubuntu0.16.04.1 [18:02:33 CEST] <JEEB> yea, that's old [18:02:33 CEST] <BtbN> that's ancient [18:02:39 CEST] <ren0v0> ah ok let me update [18:03:05 CEST] <DHE> ren0v0: using a test video with no sound as input and generated silence for audio: ./ffmpeg_g -f lavfi -i anullsrc -i test2.mp4 -shortest -af aformat=channel_layouts=5.1 -c:a aac -b:a 384k -c:v copy test4.mp4 [18:04:11 CEST] <ren0v0> ok running now :) [18:04:43 CEST] <ren0v0> is AAC the best codec when original is ATOM TrueHD ? [18:04:46 CEST] <ren0v0> ATMOS* [18:04:56 CEST] <ren0v0> Dolby Atmos TrueHD 7.1 @ 3951 kbps [18:05:01 CEST] <ren0v0> i want to keep as much of original as possible [18:05:04 CEST] <DHE> that's not relevant. the objective is a codec highly likely to be supported by a TV [18:05:33 CEST] <ren0v0> Well to be honest i don't know why TV cares because it should be passing through to my AMP which supports the lot [18:05:41 CEST] <ren0v0> ah, but not over optical [18:05:52 CEST] <ren0v0> well ARC doesn't work so i dont have a choice [18:34:34 CEST] <Johnjay> i have no idea what codecs my tv supports [18:39:11 CEST] <DHE> typically mpeg2, h264 and if 4k add h265 [18:39:32 CEST] <DHE> those are what are used in broadcast for the mostpart. anything on top of that is a bonus. maybe mpeg4/divx/xvid [18:43:03 CEST] <JEEB> mpeg-4 part 2 over broadcast is something I've seen and I've actually WTF'd at a few times [18:43:22 CEST] <JEEB> of course end users won't be seeing those internal things [18:43:25 CEST] <JEEB> but still [18:43:26 CEST] <JEEB> lol [18:43:40 CEST] <JEEB> "things you'd think you'd never see in the wild" [18:47:14 CEST] <thebombzen> yuvJ444p [18:52:02 CEST] <DHE> I have a settopbox that supports vp9. or so the manual says. intended for youtube. no idea if it supports 4k on it [20:09:53 CEST] <ren0v0> DHE after a long while i got this [matroska,webm @ 0x5621e57c4720] Read error time=01:46:06.01 bitrate=31244.5kbits/s speed=5.06x :( [20:09:58 CEST] <ren0v0> any ideas? [20:11:14 CEST] <e38383> I guess the file is truncated [20:11:58 CEST] <ren0v0> how do you mean? the original ? [20:56:21 CEST] <hiihiii> I'm trying to stream over youtube -re -i input.mp4 -threads 1 -c copy -f flv rtmp://a.rtmp.youtube.com/live2/key-key-key-key but I'm getting : [20:57:18 CEST] <hiihiii> https://thepasteb.in/p/66hVYmOZQ1rFW [21:06:40 CEST] <furq> JEEB: i thought i would let you know that i literally just figured out that you meant i could do crop,copy,zscale [21:06:43 CEST] <furq> so thanks [21:08:39 CEST] <hiihiii> also when I run this command from my other machine it runs ok [21:09:03 CEST] <furq> is it the same input file [21:09:13 CEST] <hiihiii> yes [21:09:44 CEST] <hiihiii> same command same everything [21:11:50 CEST] <hiihiii> wait... IIRC I read it doesn't support stream copy?! [21:12:20 CEST] <hiihiii> well the other command worked so... idk [21:14:56 CEST] <furq> it supports stream copy if the input file has the right codecs [22:16:29 CEST] <Johnjay> crap i don't think i have the flite filter on my zaranoe build on windows [22:17:54 CEST] <Johnjay> yeah no such filter [22:18:29 CEST] <Johnjay> i wanted to take the youtube vids i'm downloading the au dio for [22:18:39 CEST] <Johnjay> And add a small text-to-speech at the start saying the title [22:20:25 CEST] <vpb1099> hello, could anyone tell me how to attenuate 20 dB every frequency above 8hkz, it looks like i should use https://ffmpeg.org/ffmpeg-filters.html#lowpass but i dont quite understand its parameters [22:20:45 CEST] <vpb1099> *from every frequency [22:41:04 CEST] <kiroma> How can I pass "-fno-strict-aliasing"? [22:41:46 CEST] <kiroma> I'm compiling with --enable-lto, and gcc is throwing warnings that code may be misoptimized unless -fno-strict-aliasing is passed. [22:42:01 CEST] <kiroma> But adding it to CFLAGS/CXXFLAGS/LDFLAGS doesn't fix it. [22:42:22 CEST] <furq> kiroma: --extra-cflags [22:43:03 CEST] <kiroma> That's what I did. [22:43:47 CEST] <Brendan_> Hi. I'm looking to try speed up my H.264 transcoding (I am using windows). I was hoping to use Intel QSV and have installed the community edition of their Media Server Studio. I do not have the skills to compile ffmpeg from source. What are my options? [22:44:14 CEST] <kiroma> Do you have an NVidia/AMD GPU? [22:44:37 CEST] <Brendan_> Testing on my laptop... Intel CPU only [22:45:42 CEST] <jkqxz> If you are on Windows, use the builds from Zeranoe. If you are on Linux, build from source with the distribution VAAPI stuff. [22:46:11 CEST] <Brendan_> Ok let me try that. Thanks [22:46:59 CEST] <kiroma> Use codec h264_qsv [22:47:22 CEST] <kiroma> This codec should utilize Intel's QuickSync video encoder. [22:48:11 CEST] <Brendan_> Brilliant! Works [22:48:37 CEST] <Brendan_> Thanks... used Zeranoe build [22:54:17 CEST] <furq> is ffmpeg's ac3 encoder any good [22:54:44 CEST] <furq> i notice aften was forked from ffmpeg and then hasn't been updated for six years, so i'm guessing ffmpeg is better than aften now [22:54:46 CEST] <durandal_1707> its the best [22:59:25 CEST] <furq> i'm going to assume that's not sarcasm because it would make my life a lot easier if it isn't [23:01:02 CEST] <atomnuker> no, the opposite, aften was merged into ffmpeg [23:01:16 CEST] <furq> it was split from ffmpeg first wasn't it [23:01:19 CEST] <atomnuker> nope [23:01:32 CEST] <furq> http://aften.sourceforge.net/ [23:01:35 CEST] <furq> The Aften project started as a stand-alone commandline version of the FFmpeg AC-3 encoder. [23:02:01 CEST] <durandal_1707> later merged and aften abandoned [23:02:04 CEST] <furq> but yeah i noticed from the commit history that it all got merged back in after aften died [23:03:21 CEST] <furq> that's good, it saves me pissing about with eac3to [23:10:25 CEST] <vpb1099> how to calculate octaves between two frequencies ? [23:14:17 CEST] <kiroma> Can you enable mfx on Ubuntu, or does Intel's SDK not support it? [23:15:41 CEST] <durandal_1707> vpb1099: i found something on google [23:16:23 CEST] <vpb1099> durandal_1707: i could not [23:16:59 CEST] <durandal_1707> http://www.daqarta.com/dw_0moo.htm [23:18:21 CEST] <vpb1099> durandal_1707: thank you [23:18:36 CEST] <kiroma> On which version of Intel's SDK has ffmpeg based? [23:21:53 CEST] <BtbN> none. [23:22:07 CEST] <jkqxz> kiroma: It is strongly recommended not to use libmfx on Linux. It is only suitable for some constrained server setups, and there is little to recomment it even then. [23:22:14 CEST] <BtbN> also, you don't want to deal with Intels stuff on Linux. It's bad enough on Windows [23:22:51 CEST] <kiroma> Oh [23:23:04 CEST] <kiroma> I could've asked before I installed their "sdk" [23:23:24 CEST] <BtbN> you also need to patch your kernel for the SDK to work [23:23:30 CEST] <BtbN> and some system libs [23:23:31 CEST] <kiroma> Oh god [23:24:13 CEST] <BtbN> vaapi just works, there is really no reason to deal with the qsv stuff on linux [23:24:35 CEST] <kiroma> vaapi works with Intel's iGPUs? [23:25:06 CEST] <BtbN> Not sure what you mean, it only works with Intel GPUs, and I don't think they have anything that's not called an iGPU? [23:25:39 CEST] <kiroma> Oh, I thought it was AMD's API. [23:25:58 CEST] <kiroma> iGPU is an abbrevation of ingtegrated GPU. [23:25:59 CEST] <BtbN> some of the open source drivers also implement it in parts [23:26:15 CEST] <BtbN> but it only really works reliably on Intel GPUs [23:45:36 CEST] <kiroma> When I try to use vaapi it throws an error. [23:46:21 CEST] <kiroma> https://pastebin.com/5a0HLWMy [23:47:41 CEST] <jkqxz> kiroma: <https://trac.ffmpeg.org/wiki/Hardware/VAAPI> [23:48:24 CEST] <kiroma> Oh, thanks. [23:53:29 CEST] <kiroma> -init_hw_device fails and I have renderD128 and renderD129. [23:54:28 CEST] <jkqxz> Post log somewhere. [23:54:30 CEST] <vpb1099> how does ffmpeg's equalizer works, will gain=-20 clip everything above 0.1 ? [23:55:15 CEST] <kiroma> https://pastebin.com/cbzVKizv [23:56:04 CEST] <jkqxz> You installed the media SDK. Don't do that, it breaks the normal stuff. [23:56:16 CEST] <kiroma> Oh god [23:56:40 CEST] <jkqxz> Also it isn't a package, so you need to track down the stuff it created and remove it. [23:56:50 CEST] <kiroma> Yeah I know. [23:56:56 CEST] <kiroma> It's going to be painful [00:00:00 CEST] --- Sun Oct 8 2017

1 0

ffmpeg-devel.log.20171006
by burek 07 Oct '17

07 Oct '17

[01:02:24 CEST] <jkqxz> wm4: I think draw_horiz_band really is fatal to this. It appears to be an eldritch horror from the past (and won't work with modern codecs or threading), but I imagine some people will be attached to it. [01:22:31 CEST] <jkqxz> Hmm, the wrapped_avframe decoder is also evil in this respect. (And what idiot added that one?) [01:29:28 CEST] <wm4> retarded shit [01:29:44 CEST] <wm4> I value a cuvid hwaccel and working videotoolbox more [01:31:22 CEST] <jkqxz> Sounds like you need opaque_opaque_ref. [01:32:57 CEST] <wm4> or fuck this shit and just use Libav [02:43:28 CEST] <cone-673> ffmpeg 03Kaustubh Raste 07master:51ebce7d7d61: avcodec/mips: Cleanup unused functions [05:24:53 CEST] <cone-726> ffmpeg 03Luca Barbato 07master:1e2783726570: rtsp: Move message parsing to a separate function [05:54:41 CEST] <rcombs> looking into some mechanisms to handle poor OTA input better [05:54:53 CEST] <rcombs> specifically, the case of timestamps coming in corrupted [05:55:32 CEST] <rcombs> probably gonna go with an option to discard input timestamps if AV_PKT_FLAG_CORRUPT is set, and also set AV_PKT_FLAG_CORRUPT when the MPEGTS TEI flag is set [10:16:41 CEST] <ilyanov> ffmpeg 3.3.3 no longer has the file "avformat-55.dll" contained in it. Why? [10:42:03 CEST] <nevcairiel> thats because its 56 now [10:44:16 CEST] <nevcairiel> or actually 57 already [10:49:43 CEST] <ilyanov> nevcairiel here is output from a ffprobe on an m4a file [10:49:49 CEST] <ilyanov> http://dark-code.bulix.org/tfov1m-205560 [10:50:05 CEST] <ilyanov> "encoder : X Lossless Decoder 20151214, QuickTime 7.7.3" [10:50:22 CEST] <ilyanov> I think that ffmpeg cannot decode this format. true? [10:50:41 CEST] <nevcairiel> its just alac, should work fine [10:50:48 CEST] <nevcairiel> also, user support is in #ffmpeg, not here [17:13:47 CEST] <cone-310> ffmpeg 03Tobias Rapp 07master:62bdec806ecd: avfilter/vf_fps: add eof_action filter option [20:36:33 CEST] <cone-113> ffmpeg 03Marton Balint 07master:7d141e2cacb4: avdevice/decklink_dec: fix extracting luma [22:12:35 CEST] <ubitux> nevcairiel: the ER doesn't work with libav, the recovery code was dropped/never introduced [22:12:44 CEST] <ubitux> at least the reconstruction part [22:12:56 CEST] <ubitux> (i remember because it was pretty annoying with the merges) [22:12:59 CEST] <nevcairiel> heh [22:13:13 CEST] <nevcairiel> well that sample shows that it actually does something useful [22:32:37 CEST] <jamrial> ubitux: did you have time to test the merge? [22:32:40 CEST] <jamrial> i'll not push it until 3.4 is branched off master, btw [22:32:46 CEST] <jamrial> so no hurry [22:32:52 CEST] <jamrial> even if we test it enough right now, there's always the chance some specific library/module configuration could end up missing one ldflag [22:32:57 CEST] <ubitux> nope sorry not yet [22:33:04 CEST] <ubitux> that's the extralib thing? [22:33:05 CEST] <jamrial> and i don't want to cherry pick such fixes for 3.4 [22:33:07 CEST] <jamrial> yeah [22:33:10 CEST] <ubitux> i had a quick look but that's all [22:35:16 CEST] <jamrial> ok, thanks [22:35:50 CEST] <jamrial> we're pretty close to the bump :D [22:36:12 CEST] <jamrial> spring? cleaning season upcoming [22:52:14 CEST] <wm4> I want the hardware decoding stuff in 3.4 [23:39:07 CEST] <ldts> jkqxz: hi [23:45:50 CEST] <jkqxz> ldts: Hi. [23:46:13 CEST] <ldts> ah I was just sending the result of your test on my boards with some info [23:46:31 CEST] <ldts> I dont see the problem with the patch applied [23:47:29 CEST] <jkqxz> I'm just doing a clean build now. [23:47:36 CEST] <ldts> ah ok. thanks [23:48:14 CEST] <jkqxz> (Takes a while to build.) [23:48:33 CEST] <ldts> yeah..do you use ccache? [23:48:34 CEST] <jkqxz> Do you have an exynos to test on? [23:49:02 CEST] <ldts> I have one but cant find the freaking supply...I have a tonne for 5 v 2A but no 5V 4A so it wont boot... [23:49:41 CEST] <ldts> unfortunately this morning we didnt test on exynos (the guy had to leave) so we only tested on imx6 and db410 [23:50:05 CEST] <ldts> but I think this particular use case should be independent from the hardware [23:50:23 CEST] <ldts> this is the string I want to see: [h264_v4l2m2m @ 0xaaaadd3bbb50] ff_v4l2m2m_codec_end leaving pending buffers [23:51:08 CEST] <ldts> it means we deteced the user wanted to close the codec with a pending reference [23:51:16 CEST] <ldts> so we wont close the file descriptor [23:51:26 CEST] <ldts> then when the buffer is released we do a proper close [23:52:02 CEST] <ldts> [I will buy a supply next week when I relocate to my new office - moving 1000Km north) [23:52:12 CEST] <jkqxz> It happens in both cases - see the log I sent for one with it failing. [23:52:39 CEST] <ldts> yes in one case it was an error in the other case it was a debug message [23:52:50 CEST] <ldts> can you differentiate them in the trace? different colors? [23:53:50 CEST] <jkqxz> Gah, it doesn't build with --disable-optimizations. Oh well, so much for debuggability... [23:54:06 CEST] <wm4> lol [23:54:22 CEST] <ldts> ? I always build : ./configure --disable-optimizations --disable-stripping [23:54:25 CEST] <jkqxz> (asm impossible constraint things.) [23:54:38 CEST] <jkqxz> "src/libavutil/arm/intmath.h:77:5: error: impossible constraint in asm" [23:54:39 CEST] <ldts> I am building on arm64 [23:54:52 CEST] <ldts> all the boards I use are arm64... [23:55:06 CEST] <jkqxz> This is 32. (Hence also the PRId64 patch I sent yesterday...) [23:55:13 CEST] <ldts> yeah, thanks for that! [23:56:11 CEST] <jkqxz> I definitely had the message in both cases. I don't remember if it was a different colour. [23:58:31 CEST] <ldts> yes but in the case where is an error message I made the mistake of removing the return 0 (when I went from patch v12 to patch v13) [23:59:26 CEST] <ldts> I will be very surprised if this doesnt work on your board...the use case is very straightforward [23:59:44 CEST] <ldts> but I'll wait. this has to be fixed obviously [00:00:00 CEST] --- Sat Oct 7 2017

1 0

ffmpeg.log.20171006
by burek 07 Oct '17

07 Oct '17

[00:07:48 CEST] <Sebas_> nn all [04:06:43 CEST] <vigneshk> Hi. I am using ffmpeg to decode frames from video file. How do I at the same time get the time pointer for each decoded frame ? [07:09:59 CEST] <rabbe> is there any free webrtc server i can use to stream live video? [07:10:21 CEST] <rabbe> from a capturecard [07:15:12 CEST] <rabbe> or can you just open a browser window on the server? [09:34:49 CEST] <rabbe> is one-to-one/one-to-many webRTC any good? low latency/full hd quality? [09:37:08 CEST] <ilyanov> I have a perfectly well-formated m4a file, that opens and plays beautifully in VLC media player. Now I want to convert it to wav. I tried faad.exe first it said {Unable to find correct AAC sound track in the MP4 file.} After modifying the file in a hex editor, (remove the metadata header) it still failed with {Error: Gain control not yet implemented} [09:37:34 CEST] <ilyanov> So then I tried fre:ac. It refused to even open the file , reporting only {Unknown error}. What is going on here? What software should I use instead? [09:50:17 CEST] <Scanx_Taz> Hello hello :) [09:50:42 CEST] <Scanx_Taz> god bless you, I've found your IRC channel on the ffmpeg official webpage [09:51:10 CEST] <Scanx_Taz> I've a small question for you : I'm trying to install Kerberos.io which is using ffmpeg libavcodec.so.56 libraries [09:51:17 CEST] <Scanx_Taz> using Raspbian Stretch [09:51:30 CEST] <Scanx_Taz> the official package provided is the v57 [09:51:53 CEST] <Scanx_Taz> I've then downloaded the .56 on the ffmpeg "old versions" pages, configures it, make it and install it [09:52:18 CEST] <Scanx_Taz> but, when I launch Kerberos.io, I'm still receiving that message : [09:52:39 CEST] <Scanx_Taz> https://thepasteb.in/p/3lh7zz6414yc1 [09:52:57 CEST] <Scanx_Taz> how can I be sure that the install morked ? [09:53:02 CEST] <Scanx_Taz> worked sorry [09:55:18 CEST] <Scanx_Taz> when I type ldconfig -v | grep libavcodec.so, I'm receiving that : [09:55:32 CEST] <Scanx_Taz> https://thepasteb.in/p/8qhOppDNoGRH0 [09:56:48 CEST] <tdr> Scanx_Taz, can you find libavcodec.so.56 ? [09:58:00 CEST] <Scanx_Taz> huh.... No (shame on me) [09:58:36 CEST] <Scanx_Taz> (missing something, wait...) [09:59:48 CEST] <tdr> your ldconfig -v | grep libavcodec should find one version if ffmpeg is there [10:00:01 CEST] <tdr> relax your grep to find any libavcodec vs one version [10:00:38 CEST] <Scanx_Taz> https://thepasteb.in/p/98hREELQj74Tk [10:00:43 CEST] <Scanx_Taz> it doesn't find anything [10:00:49 CEST] <Scanx_Taz> playing that command at the root [10:01:14 CEST] <Scanx_Taz> brb [10:01:45 CEST] <tdr> do a listing of the files your ffmpeg provided, the ones in that paste arent [10:02:21 CEST] <tdr> well unless you were building them there and never did a make install. assumed you were using a binary version though [10:05:10 CEST] <ilyanov> ??! --" Audio files that are DRM-protected to work only in particular software cannot be imported." [10:05:16 CEST] <ilyanov> This is the reason? [10:14:57 CEST] <ilyanov> ffmpeg 3.3.3 no longer has the file "avformat-55.dll" contained in it. Why? [10:21:58 CEST] <JEEB> ilyanov: because the library API was changed and the API version got updated? [10:22:11 CEST] <JEEB> if we just changed the API and didn't change the version you wouldn't be happy either now would you? [10:22:55 CEST] <ilyanov> JEEB Well FFMPEG has lost all articulation with audacity then [10:23:13 CEST] <JEEB> you will have to make sure your application works with the new API headers and link it against it again basically [10:23:18 CEST] <tdr> does your audactiy have an update? [10:24:27 CEST] <ilyanov> I have the latest version from 2015 [10:24:40 CEST] <tdr> newest audacity only requres >=ffmpeg-1.2 [10:25:08 CEST] <JEEB> it's a windows binary so it's linked against something [10:25:10 CEST] <tdr> so that seems pretty flexible [10:25:25 CEST] <JEEB> yes, but they'd have to build a new binary for ilyanov if he wants to use the latest FFmpeg. Boo-hoo [10:25:38 CEST] <JEEB> ilyanov: build it yourself or just use an older FFmpeg until Audacity makes a windows update [10:25:48 CEST] <JEEB> go request an update from them if you have an actual need [10:25:49 CEST] <hiihiii> hello [10:25:59 CEST] <hiihiii> how do you pipe ffmpeg with zip ? [10:26:04 CEST] <ilyanov> thanks folr the help.. but I get this funny feeling I need to install ubuntu onto a machine just to convert a little m4a file into WAV [10:26:17 CEST] <ilyanov> I don't doubt this is possible, just seems like a lot of run-around [10:27:10 CEST] <JEEB> you would probably get audacity linked against a newer FFmpeg with the latest Ubuntu (17.04 or 17.10 beta), but I don't think you will gain much [10:27:20 CEST] <JEEB> so just grab the older FFmpeg and use it unless you have an issue [10:27:26 CEST] <JEEB> and if you have an issue go poke Audacity's support [10:27:40 CEST] <ilyanov> Okay i do not _need_ to use audacity. So if this train-of-thought isn't going anywher, I can chalk it down with the other failuers I posted earlier. [10:27:56 CEST] <JEEB> lol [10:28:16 CEST] <ilyanov> So faad.exe cannot open m4a files. (even though it advertises such functionality). Faad.exe cannot even open m4a files with the metadata header cut out of them. [10:28:27 CEST] <ilyanov> fre:ac cannot open m4a files. "Uknown error" [10:28:37 CEST] <ilyanov> Audacity cannot open m4a files. [10:28:54 CEST] <JEEB> is latest FFmpeg's ffprobe able to open it? [10:29:00 CEST] <JEEB> ffprobe file.m4a [10:29:08 CEST] <JEEB> does it give you streams properly [10:29:13 CEST] <JEEB> if yes, then go complain at audacity [10:29:21 CEST] <ilyanov> JEEB VLC player opens and plays it beautifuly. Next question. [10:29:22 CEST] <JEEB> or if you just need wav out of it [10:29:40 CEST] <JEEB> VLC is unrelated but then many recent FFmpeg versions are likely to be able ot open it [10:29:43 CEST] <JEEB> *to [10:29:53 CEST] <JEEB> if you just want a wav to edit [10:29:59 CEST] <tdr> JEEB, for fun, i looked at the audactity for windows site, it links to a site to get ffmpeg that has ffmpeg-win-2.2.2.exe ... [10:30:00 CEST] <JEEB> ffmpeg -i perkele.m4a out.wav [10:30:11 CEST] <JEEB> try that :P [10:31:25 CEST] <JEEB> that will include the encoder delay samples in the beginning but I think that's the least of your problems at this point :P (and almost nothing handles that correctly anyways - if the file even has the information regarding it) [10:37:17 CEST] <ilyanov> ffprobe output http://dark-code.bulix.org/tfov1m-205560 [10:51:34 CEST] <Bear10> Does anyone know if it's possible to sync audio and video from two different udp ports? [10:56:45 CEST] <rabbe> JEEB: whats your opinion on Kurento? tried it? [11:11:49 CEST] <Scanx_taz> brb [11:11:49 CEST] <Scanx_taz> tdr [11:11:49 CEST] <Scanx_taz> do a listing of the files your ffmpeg provided, the ones in that paste arent [11:11:49 CEST] <Scanx_taz> Déconnecté du réseau IRC. [11:11:49 CEST] <Scanx_taz> Scanx_Taz [11:11:50 CEST] <Scanx_taz> back [11:11:50 CEST] <Scanx_taz> sorry [11:11:51 CEST] <Scanx_taz> what do you mean ? [11:11:51 CEST] <Scanx_taz> a ls -R on the root parent of the ffmpeg source folder ? [11:11:52 CEST] <Scanx_taz> Scanx_Taz [11:11:52 CEST] <Scanx_taz> https://thepasteb.in/p/Q1hBjjMwJNJh8 [11:11:53 CEST] <Scanx_taz> Scanx_Taz [11:11:53 CEST] <Scanx_taz> sorry, escape characters in it [11:12:13 CEST] <Scanx_taz> sorry, internet connection was broken for a long moment [11:19:02 CEST] <Commander_Cody> I'd like to cut out part of an audio file, without changing the codec. [11:19:22 CEST] <Commander_Cody> How can I determine which exact commands are necessary to get the old audio codec? [11:21:56 CEST] <ilyanov> JEEB that was the silver bullet [11:39:57 CEST] <Sebas_> I am having this exact problem: https://trac.ffmpeg.org/ticket/5760 with the latest version of ffmpeg. [11:40:50 CEST] <Sebas_> Seems ffmpeg is unable to produce non-muxed webm chunked files, needed for bitrate adaptivity in browser. [11:41:28 CEST] <Sebas_> or could anyone point me to a working example? [12:54:25 CEST] <Bear10> Anyone know if you by default an overlay is transparent? [12:54:59 CEST] <Bear10> I'm sending a udp input into an overlay (currently it's a transparent image) but I get a black background [13:07:34 CEST] <DHE> Bear10: the top layered image should have transparency defined within it (eg: PNG of a logo or something) or you'll have a hard rectangular border [13:19:41 CEST] <Bear10> DHE: hmm ok it seems like I got it it was actually colorkey what I was looking for for a video [13:33:53 CEST] <Bear10> by any chance would you know how to have an overlay occupy full screen? [14:02:11 CEST] <Fyr> guys, I have 800 Gb of PNG files. what lossless image format will have smaller size than the PNG format? [14:03:22 CEST] <Fyr> is JPEG able not to loose quality? [14:05:27 CEST] <Sebas_> probally offtopic: perhaps BMP files and 7zip? [14:06:27 CEST] <Sharkigator> Fyr: you can also try to recompress the png files [14:06:41 CEST] <Sharkigator> <http://optipng.sourceforge.net/> [14:08:27 CEST] <Fyr> Sharkigator, thanks. [15:13:20 CEST] <Brian_> im making a command where im merging multiple videos and i want to add an audio background music to it [15:13:29 CEST] <Brian_> im trying this command but ffmpeg freezes after a while: https://pastebin.com/raw/QmBCndba [15:13:39 CEST] <Brian_> anyone know how do add audio to -complex_filter? [15:13:46 CEST] <Brian_> all other video files are muted [15:16:44 CEST] <Fyr> Brian_, why are you using filter_complex for muxing? [15:17:12 CEST] <Fyr> convert the video, if you are converting, then mux. [15:17:49 CEST] <Brian_> for simplicity i pasted a command here with just 1 video [15:17:50 CEST] <Fyr> sequential work will allow you to control the result at each stage of conversion. [15:18:07 CEST] <Brian_> also removed overlay part which i added to the filter [15:18:23 CEST] <Fyr> 1) convert/merge/cut/concat the video [15:18:23 CEST] <Fyr> 2) mux it with audio [15:19:15 CEST] <Brian_> so first create the video i want, than add audio with an other command? [15:19:21 CEST] <Fyr> yes [15:19:49 CEST] <Fyr> it's simple, without usage of -filter_complex. [15:19:49 CEST] <Brian_> it cant be done with one command? Im using ffmpeg on android so encoding is slow [15:20:06 CEST] <Fyr> Brian_, muxing doesn't require much CPU. [15:20:14 CEST] <Fyr> it's just copying. [15:20:47 CEST] <Brian_> alright, i'll try. Thanks [15:33:17 CEST] <dan3wik> When I specify "-f v4l2 -i /dev/video0" I get a slightly higher cpu usage than if I just specify "-i /dev/video0". [15:34:13 CEST] <dan3wik> Without the -f v4l2, it still uses the video4linux2 format but seems to bring single core usage from 55% average to 45% average. [15:39:40 CEST] <Bear10> anyone know how to POST to a listening ffmpeg server on http? [15:42:42 CEST] <Bear10> I keep getting invalid data [15:52:53 CEST] <kepstin> dan3wik: too many variables to say anything. cpu usage probably depends mostly on the encoder being used, which can depend on the video contents, for example [15:54:17 CEST] <dan3wik> FFmpeg shows that the input format and codec is v4l2 and yuv420 with and without the -f specified, but has different cpu usage. [15:54:56 CEST] <dan3wik> So, it is decoding and encoding to identical format and codecs. [16:11:25 CEST] <ayum> Hi, when I using tee muxer to create multiple output, I found that bitrate is "N/A". once I disabled tee muxer, the bitrate show correct again. how can I use tee muxer and show bitrate correctly? [16:13:11 CEST] <BtbN> You can't [16:13:44 CEST] <BtbN> The tee muxer does not create any actual output, so it can't measure any output bitrate [16:14:31 CEST] <ayum> @BtbN, I want to modify the source code to supports this, is it possible? [16:14:55 CEST] <BtbN> It does not have that information, so unless you come up with a massive ugly hack, no [16:15:59 CEST] <ayum> @BtbN, so far, my solution is using 2 ffmpeg, first ffmpeg use pipe output to second ffmpeg, so it will show the bitrate. then I use tee muxer in second ffmpeg. [16:16:23 CEST] <BtbN> Why do you even care so much about that bitrate reading? It's not vital for any functionality. [16:16:26 CEST] <ayum> but second ffmpeg will use about 4% CPU resource. [16:16:57 CEST] <ayum> @BtbN, I am writing a web interface for ffmpeg, so I want to display bitrate if possible [16:25:53 CEST] <DHE> the tee multiplexer might be aware of the per-stream bitrate (eg: video bitrate as set in the encoder) but it can't measure muxing overhead, etc. [16:30:48 CEST] <Fyr> what is tee multiplexor for? [16:30:57 CEST] <Fyr> google doesn't answer. [16:31:19 CEST] <Fyr> only a few entries concerning bugs in this muxer. [16:39:05 CEST] <iive> Fyr: try `man ffmpeg-formats` then '/' to search 'q' to quit [16:48:56 CEST] <Fyr> iive, "No manual entry for ffmpeg-formats" [16:50:24 CEST] <Fyr> Ubuntu 16.04.3 LTS. [16:52:39 CEST] <relaxed> https://ffmpeg.org/ffmpeg-formats.html#tee-1 [16:57:45 CEST] <Fyr> thanks [16:58:20 CEST] <Fyr> why is there image2 and no image? [17:00:36 CEST] <DHE> things change over time. the old API for decoding audio was avcodec_decode_audio4 [17:27:05 CEST] <BytesBacon> Is there a easier way to go about doing zones? Some tool, ... [19:07:20 CEST] <Kiicki> Could anyone explain to me what this NBC is? I have tried to convert my audio to that same setting without luck. It doesn't say NBC for me, but Explicit. https://i.imgur.com/Ztqi5dp.png [19:11:26 CEST] <Sharkigator> non backward compatible? [19:13:40 CEST] <Kiicki> I try to get the same result but I get this https://i.imgur.com/HZ1Coqp.png [19:14:15 CEST] <Sharkigator> why do you need the exact same format? [19:14:31 CEST] <Sharkigator> and what does ffprobe say? [19:14:32 CEST] <Kiicki> I'm just curious how it got there [19:14:41 CEST] <Kiicki> I use meGUI at the moment [19:14:48 CEST] <Kiicki> as it can support HE-AAC [19:14:52 CEST] <Kiicki> audio [19:20:56 CEST] <Sharkigator> Kiicki are you using the same container? [19:22:03 CEST] <Kiicki> Yes, MKV [19:59:32 CEST] <utack> has h265/hevc decoding speed been improved from ffmpeg 3.2 to 3.3? i know 3.2 made it faster with some assembly magic, but it seems even faster now [20:00:11 CEST] <utack> or am i confusing the two, and that was 3.3? [20:00:48 CEST] <JEEB> not sure at which point additional asm was added, but it's still not exactl as fast as it could be [20:00:52 CEST] <furq> uh [20:01:10 CEST] <furq> is it just me or does crop,zscale cause a segfault with crop x > 0 [20:01:11 CEST] <JEEB> nobody really cares about HEVC decoding that much unfortunately :< [20:01:27 CEST] <furq> tried three different builds on two different OSs and it crashes every time [20:01:30 CEST] <JEEB> furq: it can be, zimg has interesting buffer alignment requirements [20:01:39 CEST] <furq> lol [20:01:42 CEST] <furq> "great" [20:01:54 CEST] <JEEB> and if you crop those might break [20:02:06 CEST] <JEEB> I remember it also breaking with lavfi generated input [20:02:15 CEST] <furq> yeah it seems to break a lot [20:02:37 CEST] <furq> i assumed my build was broken in some way but i just tried it with a build from ports and it's identically broken [20:02:41 CEST] <furq> so i guess it's just not very good [20:02:57 CEST] <JEEB> well, you get this sort of stuff when different things have different alignment requirements [20:02:59 CEST] <furq> looks like i'm using swscale then [20:03:09 CEST] <JEEB> (of course a bug in zimg is also possible, but usually it's the alignment) [20:03:18 CEST] <furq> yeah alignment would make sense [20:03:33 CEST] <furq> it works fine with x = 0 [20:03:38 CEST] <furq> and y > 0 [20:04:05 CEST] <JEEB> I wonder how vapoursynth handles it since it has crop [20:04:17 CEST] <JEEB> and it uses zimg for the internal scaling stuff [20:05:46 CEST] <furq> aha [20:05:55 CEST] <furq> yeah it seems to work if crop x is mod32 [20:06:02 CEST] <furq> what a lot of fun [20:07:17 CEST] Action: JEEB pokes Myrs for shits and giggles [20:07:29 CEST] <furq> so i could pad,crop,zscale, or i could zscale,crop, or i could obviously do neither of those and just crop,scale [20:08:13 CEST] <JEEB> http://up-cat.net/p/8c00c16e [20:08:37 CEST] <JEEB> you can clearly guess it's the author since he's the only one calling it z.lib :3 [20:08:44 CEST] <furq> lol [20:09:08 CEST] <furq> he has a big dartboard in his house with a picture of mark adler on it [20:09:41 CEST] <furq> https://upload.wikimedia.org/wikipedia/commons/7/7d/Mark_Adler.jpg [20:09:41 CEST] <furq> ladies [20:18:14 CEST] <kepstin> hmm. we run into a lot of weird issues because of the zero-copy optimizations in crop, eh. [20:18:44 CEST] <kepstin> fixing it by realigning the data requires copying it to a new avframe, basically :/ [20:18:58 CEST] <kepstin> or moving it within the existing frame i guess [20:46:36 CEST] <zyme> I've got a dumb question, as it may be to vague to answer, but I use a streaming program which uses ffmpeg, and I've found recently that a lot of HEVC videos are having routine instances of, well lets just say "I can see the squares!" for a few seconds, combined with overall fuzziness, then back to normal, but this will continue on again and again routinely throughout some videos (even if I downscale the resolution)- [20:46:36 CEST] <zyme> whereas MPC-HC w/K-lite codec pack mega shows a perfectly sharp high quality image throughout. -- Generally I'm just curious what conditions [c/w]ould cause this phenomenon, any ideas? [20:47:45 CEST] <JEEB> guess what MPC-HC's decoder uses :P [20:47:47 CEST] <JEEB> FFmpeg [20:48:03 CEST] <zyme> I'm pretty sure it uses ffdshow [20:48:03 CEST] <JEEB> anyways, you'd have to look into the logs of the given application but it just looks like it has nothing to do with your input format [20:48:17 CEST] <furq> i wouldn't want to assume that "k-lite codec pack mega" hasn't fucked with the internal decoders in some way [20:48:20 CEST] <JEEB> pretty sure it doesn't, and FFDShow was the first DShow filter to use FFmpeg [20:48:38 CEST] <furq> but yeah any version of mpc-hc from the last few years uses lav filters internally [20:48:39 CEST] <JEEB> so even if it was FFDShow-tryouts, which is highly unlikely as LAV Video is a thing [20:49:08 CEST] <JEEB> because even clsid (k-lite's maintainer) can't be as stupid as to put -tryouts (which has been more or less dead since like 2011-12) as the default decoder [20:49:12 CEST] <JEEB> but as I said, evne if it was [20:49:15 CEST] <JEEB> that would still be FFmpeg-based [20:49:23 CEST] <JEEB> but as I was writing [20:49:35 CEST] <zyme> I was actually suprised it worked, since it has that big stability warning next to my gpu enabled HEVC decoding option - lol. [20:49:37 CEST] <furq> it's windows, so i also wouldn't assume this has been updated in the last five years [20:50:19 CEST] <JEEB> zyme: anyways you'd have to look into your logs for that streaming application :P most likely it is not pushing enough bits through or is using not good enough encoder / compression settings [20:50:45 CEST] <JEEB> unless you are missing bits of the stream when playing or something [20:51:03 CEST] <JEEB> aka "we have no fucking way of knowing by your words where the flying fuck it's cocking itself up" [20:51:09 CEST] <zyme> I doubt the logs will revel much, though I can force it into a debugging and event trace mode and hope for the best [20:51:42 CEST] <JEEB> because if it's a streaming thing which I would guess means re-encoding, I would say it's decoding and then re-encoding for the stream [20:52:38 CEST] <JEEB> and if it's periodic it sounds like keyframe pumping - although it can be the decoder somehow cocking up. you literally cannot limit any options out because there's zarro information [20:52:51 CEST] <JEEB> although decoding of HEVC has been pretty stable for the bast three years [20:52:55 CEST] <JEEB> slow, but stable [20:53:00 CEST] <zyme> reencoding barely, It doesn't Encode after it decodes, more of a high bitrate no effort lossless format conversion into h.264 if anything [20:53:17 CEST] <kepstin> yeah, sounds like keyframe pumping in the source encoder to me. [20:53:30 CEST] <zyme> slaughters the performance of true transcoding apps like plex [20:54:05 CEST] <JEEB> sounds like it's using some crap ASIC on your thing or just set x264 to superfast, and most likely sounds like it's definitely not using a lot of bit rate :P [20:54:13 CEST] <JEEB> if it's keyframe pulsing/pumping [20:54:33 CEST] <JEEB> but as I said, fuck if I know. it has nothing to do with FFmpeg even if it's using it. we can't possibly the fuck know what the flying fuck it's doing [20:54:53 CEST] <JEEB> and I really don't know hwo the fuck you expected [21:15:27 CEST] <abel-abel> Hello! I'm trying to add some text to video and saw this thread: https://stackoverflow.com/questions/17623676/text-on-video-ffmpeg [21:16:07 CEST] <abel-abel> When I tried this, I found my video's quality is much lower than before. [21:18:38 CEST] <abel-abel> Is there any way to solve this? [21:18:44 CEST] <abel-abel> Thank you in advance. [21:20:52 CEST] <abel-abel> ffmpeg -i input.flv -vf drawtext="fontfile=/path/to/font.ttf: text='Stack Overflow': fontcolor=white: fontsize=24: box=1: boxcolor=black(a)0.5: boxborderw=5: x=(w-text_w)/2: y=(h-text_h)/2" -codec:a copy output.flv [21:22:59 CEST] <furq> you need to set the video codec or else it'll use flv1 or something terrible [21:27:01 CEST] <abel-abel> furq, which value should I set for the video codec? [21:27:25 CEST] <furq> presumably libx264 [21:28:04 CEST] <abel-abel> you mean it's the default value? Should I set it to another value? [21:38:14 CEST] <llogan> abel-abel: he means you need to add "-c:v libx264" [21:38:55 CEST] <relaxed> abel-abel: https://trac.ffmpeg.org/wiki/Encode/H.264 [21:39:14 CEST] <abel-abel> llogan, Thanks. Unknown encoder 'libx264'. I'm rebuilding ffmpeg with X264 option enabled now. [21:40:48 CEST] <abel-abel> Besides, I saw `Metadata: encoder : Lavf57.56.101` in the log. Maybe I should use this instead of libx264? [21:41:26 CEST] <llogan> no [21:43:15 CEST] <abel-abel> llogan, ok I will try libx264 option when build of ffmpeg finishes [21:46:54 CEST] <abel-abel> relaxed, llogan, It works! Thank you so much! [21:47:31 CEST] <abel-abel> Maybe It's very necessary for me learn some ffmpeg! ;) [21:47:41 CEST] <abel-abel> Interesting [21:48:31 CEST] <JASTON> hey gang, does anyone know if the hls muxer supports the extraction/transcoding of subtitles? I'm having this issue: https://pastebin.com/8CCMZqD4 [21:48:39 CEST] <JASTON> if i remove the hls flags it works as expected [21:54:14 CEST] <BtbN> You want to make a hls playlist... with just subtitles? oO [21:56:21 CEST] <JASTON> yeah [21:56:54 CEST] <JASTON> trying to separate everything out into separate files so that they can be nicely organized and switchable by the client [21:58:02 CEST] <JASTON> not ruling out that it might be a bad approach :) [22:01:32 CEST] <JASTON> to put it a better way, im trying to isolate all streams (video,audio,subtitles) into their own playlists, only running into issues with subtitles [22:05:08 CEST] <BtbN> I doubt you can put anything else than mpeg-ts or fragmented mp4 in hls. [22:07:02 CEST] <saml> is there a good ffmpeg ubuntu 14.04 repo? with all latest security patches [22:11:17 CEST] <JASTON> BtbN: well its interesting. if i send all streams to the muxer "-map:0", it'll go above and beyond to transcode the subtitles into vtt files with their own m3u8 playlist, pretty much what i want to do, while lumping all the video/audio tracks into another set of files. i just wanted to see if i could invoke the behavior more explicitly [22:11:35 CEST] <JASTON> ill use this behavior as a workaround i guess [22:19:19 CEST] <llogan> saml: just download it https://johnvansickle.com/ffmpeg/ throw it in /usr/local/bin or ~/bin [22:20:41 CEST] <BtbN> ubuntu does use the latest version of the branch it uses. So you get security patches. [22:21:23 CEST] <saml> llogan, ah thanks. that's actually linked from ffmpeg. i thought it was some private build with virus [22:22:25 CEST] <BtbN> it is a private build [22:22:29 CEST] <BtbN> there are no official builds. [22:23:03 CEST] <saml> http://ffmpeg.org/download.html this page links to that johnvansickle [22:23:19 CEST] <BtbN> yes, so? [22:23:46 CEST] <saml> so i want to be part of that web ring [22:25:03 CEST] <llogan> "web ring"...brings back memories of geocities, etc. [23:14:41 CEST] <cryptodechange> Quite pleased with these results [23:14:45 CEST] <cryptodechange> https://imgur.com/a/tMxw1 [23:15:42 CEST] <cryptodechange> It seems that the original had a few small blocking artifacts around outlines, the encode kind of smoothed them out in some areas, while adding them in others (zoom in on Goku's collar in the second comparison) [23:16:09 CEST] <cryptodechange> But quickly flipping between layers, simply cannot notice the difference [23:21:54 CEST] <LiftLeft> How can I stream one window with ffmpeg? [23:24:45 CEST] <llogan> depends on your OS [23:24:53 CEST] <LiftLeft> I'm using ubuntu [23:25:36 CEST] <llogan> x11grab. you can use xwininfo to get the window parameters. [23:29:49 CEST] <llogan> see http://ffmpeg.org/ffmpeg-devices.html#x11grab http://trac.ffmpeg.org/wiki/EncodingForStreamingSites#Streamingyourdesktop [23:53:53 CEST] <LiftLeft> llogan: am I suppose to find the window id from xdpyinfo? [23:54:04 CEST] <LiftLeft> and them give the ffmpeg the id [23:54:31 CEST] <LiftLeft> xdpyinfo doesn't seem to tell me that information [00:00:00 CEST] --- Sat Oct 7 2017

1 0